Merge tag 'sound-4.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "It's been busy summer weeks and hence lots of changes, partly for a
  few new drivers and partly for a wide range of fixes.

  Here are highlights:

  ALSA Core:
   - Fix rawmidi buffer management, code cleanup / refactoring
   - Fix the SG-buffer page handling with incorrect fallback size
   - Fix the stall at virmidi trigger callback with a large buffer; also
     offloading and code-refactoring along with it
   - Various ALSA sequencer code cleanups

  ASoC:
   - Deploy the standard snd_pcm_stop_xrun() helper in several drivers
   - Support for providing name prefixes to generic component nodes
   - Quite a few fixes for DPCM as it gains a bit wider use and more
     robust testing
   - Generalization of the DIO2125 support to a simple amplifier driver
   - Accessory detection support for the audio graph card
   - DT support for PXA AC'97 devices
   - Quirks for a number of new x86 systems
   - Support for AM Logic Meson, Everest ES7154, Intel systems with
     RT5682, Qualcomm QDSP6 and WCD9335, Realtek RT5682 and TI TAS5707

  HD-audio:
   - Code refactoring in HD-audio ext codec codes to drop own classes;
     preliminary works for the upcoming legacy codec support
   - Generalized DRM audio component for the upcoming radeon / amdgpu
     support
   - Unification of mic mute-LED and GPIO support for various codecs
   - Further improvement of CA0132 codec support including Recon3D
   - Proper vga_switcheroo handling for AMD i-GPU
   - Update of model list in documentation
   - Fixups for another HP Spectre x360, Conexant codecs, power-save
     blacklist update

  USB-audio:
   - Fix the invalid sample rate setup with external clock
   - Support of UAC3 selector units and processing units
   - Basic UAC3 power-domain support
   - Support for Encore mDSD and Thesycon-based DSD devices
   - Preparation for future complete callback changes

  Firewire:
   - Add support for MOTU Traveler

  Misc:
   - The endianess notation fixes in various drivers
   - Add fall-through comment in lots of drivers
   - Various sparse warning fixes, e.g. about PCM format types"

* tag 'sound-4.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (529 commits)
  ASoC: adav80x: mark expected switch fall-through
  ASoC: da7219: Add delays to capture path to remove DC offset noise
  ALSA: usb-audio: Mark expected switch fall-through
  ALSA: mixart: Mark expected switch fall-through
  ALSA: opl3: Mark expected switch fall-through
  ALSA: hda/ca0132 - Add exit commands for Recon3D
  ALSA: hda/ca0132 - Change mixer controls for Recon3D
  ALSA: hda/ca0132 - Add Recon3D input and output select commands
  ALSA: hda/ca0132 - Add DSP setup defaults for Recon3D
  ALSA: hda/ca0132 - Add Recon3D startup functions and setup
  ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio
  ALSA: hda/ca0132 - Add Recon3D pincfg
  ALSA: hda/ca0132 - Add quirk ID and enum for Recon3D
  ALSA: hda/ca0132 - Add alt_functions unsolicited response
  ALSA: hda/ca0132 - Clean up ca0132_init function.
  ALSA: hda/ca0132 - Create mmio gpio function to make code clearer
  ASoC: wm_adsp: Make DSP name configurable by codec driver
  ASoC: wm_adsp: Declare firmware controls from codec driver
  ASoC: max98373: Added software reset register to readable registers
  ASoC: wm_adsp: Correct DSP pointer for preloader control
  ...

498 files changed, 19417 insertions, 6533 deletions

diff --git a/Documentation/devicetree/bindings/sound/ac97-bus.txt b/Documentation/devicetree/bindings/sound/ac97-bus.txt
new file mode 100644
index 000000000000..103c428f2595
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ac97-bus.txt
@@ -0,0 +1,32 @@
+Generic AC97 Device Properties
+
+This documents describes the devicetree bindings for an ac97 controller child
+node describing ac97 codecs.
+
+Required properties:
+-compatible : Must be "ac97,vendor_id1,vendor_id2
+	      The ids shall be the 4 characters hexadecimal encoding, such as
+	      given by "%04x" formatting of printf
+-reg	    : Must be the ac97 codec number, between 0 and 3
+
+Example:
+ac97: sound@40500000 {
+	compatible = "marvell,pxa270-ac97";
+	reg = < 0x40500000 0x1000 >;
+	interrupts = <14>;
+	reset-gpios = <&gpio 95 GPIO_ACTIVE_HIGH>;
+	#sound-dai-cells = <1>;
+	pinctrl-names = "default";
+	pinctrl-0 = < &pinctrl_ac97_default >;
+	clocks = <&clks CLK_AC97>, <&clks CLK_AC97CONF>;
+	clock-names = "AC97CLK", "AC97CONFCLK";
+
+	#address-cells = <1>;
+	#size-cells = <0>;
+	audio-codec@0 {
+		reg = <0>;
+		compatible = "ac97,574d,4c13";
+		clocks = <&fixed_wm9713_clock>;
+		clock-names = "ac97_clk";
+	}
+};
diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt
new file mode 100644
index 000000000000..3dfc2515e5c6
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt
@@ -0,0 +1,23 @@
+* Amlogic Audio FIFO controllers
+
+Required properties:
+- compatible: 'amlogic,axg-toddr' or
+	      'amlogic,axg-frddr'
+- reg: physical base address of the controller and length of memory
+       mapped region.
+- interrupts: interrupt specifier for the fifo.
+- clocks: phandle to the fifo peripheral clock provided by the audio
+	  clock controller.
+- resets: phandle to memory ARB line provided by the arb reset controller.
+- #sound-dai-cells: must be 0.
+
+Example of FRDDR A on the A113 SoC:
+
+frddr_a: audio-controller@1c0 {
+	compatible = "amlogic,axg-frddr";
+	reg = <0x0 0x1c0 0x0 0x1c>;
+	#sound-dai-cells = <0>;
+	interrupts = <GIC_SPI 88 IRQ_TYPE_EDGE_RISING>;
+	clocks = <&clkc_audio AUD_CLKID_FRDDR_A>;
+	resets = <&arb AXG_ARB_FRDDR_A>;
+};
diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt
new file mode 100644
index 000000000000..80b411296480
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt
@@ -0,0 +1,124 @@
+Amlogic AXG sound card:
+
+Required properties:
+
+- compatible: "amlogic,axg-sound-card"
+- model : User specified audio sound card name, one string
+
+Optional properties:
+
+- audio-aux-devs : List of phandles pointing to auxiliary devices
+- audio-widgets : Please refer to widgets.txt.
+- audio-routing : A list of the connections between audio components.
+
+Subnodes:
+
+- dai-link: Container for dai-link level properties and the CODEC
+	    sub-nodes. There should be at least one (and probably more)
+	    subnode of this type.
+
+Required dai-link properties:
+
+- sound-dai: phandle and port of the CPU DAI.
+
+Required TDM Backend dai-link properties:
+- dai-format : CPU/CODEC common audio format
+
+Optional TDM Backend dai-link properties:
+- dai-tdm-slot-rx-mask-{0,1,2,3}: Receive direction slot masks
+- dai-tdm-slot-tx-mask-{0,1,2,3}: Transmit direction slot masks
+				  When omitted, mask is assumed to have to no
+				  slots. A valid must have at one slot, so at
+				  least one these mask should be provided with
+				  an enabled slot.
+- dai-tdm-slot-num : Please refer to tdm-slot.txt.
+		     If omitted, slot number is set to accommodate the largest
+		     mask provided.
+- dai-tdm-slot-width : Please refer to tdm-slot.txt. default to 32 if omitted.
+- mclk-fs : Multiplication factor between stream rate and mclk
+
+Backend dai-link subnodes:
+
+- codec: dai-link representing backend links should have at least one subnode.
+	 One subnode for each codec of the dai-link.
+	 dai-link representing frontend links have no codec, therefore have no
+	 subnodes
+
+Required codec subnodes properties:
+
+- sound-dai: phandle and port of the CODEC DAI.
+
+Optional codec subnodes properties:
+
+- dai-tdm-slot-tx-mask : Please refer to tdm-slot.txt.
+- dai-tdm-slot-rx-mask : Please refer to tdm-slot.txt.
+
+Example:
+
+sound {
+	compatible = "amlogic,axg-sound-card";
+	model = "AXG-S420";
+	audio-aux-devs = <&tdmin_a>, <&tdmout_c>;
+	audio-widgets = "Line", "Lineout",
+			"Line", "Linein",
+			"Speaker", "Speaker1 Left",
+			"Speaker", "Speaker1 Right";
+			"Speaker", "Speaker2 Left",
+			"Speaker", "Speaker2 Right";
+	audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2",
+			"SPDIFOUT IN 0", "FRDDR_A OUT 3",
+			"TDM_C Playback", "TDMOUT_C OUT",
+			"TDMIN_A IN 2", "TDM_C Capture",
+			"TDMIN_A IN 5", "TDM_C Loopback",
+			"TODDR_A IN 0", "TDMIN_A OUT",
+			"Lineout", "Lineout AOUTL",
+			"Lineout", "Lineout AOUTR",
+			"Speaker1 Left", "SPK1 OUT_A",
+			"Speaker2 Left", "SPK2 OUT_A",
+			"Speaker1 Right", "SPK1 OUT_B",
+			"Speaker2 Right", "SPK2 OUT_B",
+			"Linein AINL", "Linein",
+			"Linein AINR", "Linein";
+
+	dai-link@0 {
+		sound-dai = <&frddr_a>;
+	};
+
+	dai-link@1 {
+		sound-dai = <&toddr_a>;
+	};
+
+	dai-link@2 {
+		sound-dai = <&tdmif_c>;
+		dai-format = "i2s";
+		dai-tdm-slot-tx-mask-2 = <1 1>;
+		dai-tdm-slot-tx-mask-3 = <1 1>;
+		dai-tdm-slot-rx-mask-1 = <1 1>;
+		mclk-fs = <256>;
+
+		codec@0 {
+			sound-dai = <&lineout>;
+		};
+
+		codec@1 {
+			sound-dai = <&speaker_amp1>;
+		};
+
+		codec@2 {
+			sound-dai = <&speaker_amp2>;
+		};
+
+		codec@3 {
+			sound-dai = <&linein>;
+		};
+
+	};
+
+	dai-link@3 {
+		sound-dai = <&spdifout>;
+
+		codec {
+			sound-dai = <&spdif_dit>;
+		};
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt
new file mode 100644
index 000000000000..521c38ad89e7
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt
@@ -0,0 +1,20 @@
+* Amlogic Audio SPDIF Output
+
+Required properties:
+- compatible: 'amlogic,axg-spdifout'
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+  * "pclk" : peripheral clock.
+  * "mclk" : master clock
+- #sound-dai-cells: must be 0.
+
+Example on the A113 SoC:
+
+spdifout: audio-controller@480 {
+	compatible = "amlogic,axg-spdifout";
+	reg = <0x0 0x480 0x0 0x50>;
+	#sound-dai-cells = <0>;
+	clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>,
+		 <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>;
+	clock-names = "pclk", "mclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt
new file mode 100644
index 000000000000..1c1b7490554e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt
@@ -0,0 +1,28 @@
+* Amlogic Audio TDM formatters
+
+Required properties:
+- compatible: 'amlogic,axg-tdmin' or
+	      'amlogic,axg-tdmout'
+- reg: physical base address of the controller and length of memory
+       mapped region.
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+  * "pclk"     : peripheral clock.
+  * "sclk"     : bit clock.
+  * "sclk_sel" : bit clock input multiplexer.
+  * "lrclk"    : sample clock
+  * "lrclk_sel": sample clock input multiplexer
+
+Example of TDMOUT_A on the A113 SoC:
+
+tdmout_a: audio-controller@500 {
+	compatible = "amlogic,axg-tdmout";
+	reg = <0x0 0x500 0x0 0x40>;
+	clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>;
+	clock-names = "pclk", "sclk", "sclk_sel",
+		      "lrclk", "lrclk_sel";
+};
diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt
new file mode 100644
index 000000000000..cabfb26a5f22
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt
@@ -0,0 +1,22 @@
+* Amlogic Audio TDM Interfaces
+
+Required properties:
+- compatible: 'amlogic,axg-tdm-iface'
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+  * "sclk" : bit clock.
+  * "lrclk": sample clock
+  * "mclk" : master clock
+	     -> optional if the interface is in clock slave mode.
+- #sound-dai-cells: must be 0.
+
+Example of TDM_A on the A113 SoC:
+
+tdmif_a: audio-controller@0 {
+	compatible = "amlogic,axg-tdm-iface";
+	#sound-dai-cells = <0>;
+	clocks = <&clkc_audio AUD_CLKID_MST_A_MCLK>,
+		 <&clkc_audio AUD_CLKID_MST_A_SCLK>,
+		 <&clkc_audio AUD_CLKID_MST_A_LRCLK>;
+	clock-names = "mclk", "sclk", "lrclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/atmel-i2s.txt b/Documentation/devicetree/bindings/sound/atmel-i2s.txt
index 735368b8a73f..40549f496a81 100644
--- a/Documentation/devicetree/bindings/sound/atmel-i2s.txt
+++ b/Documentation/devicetree/bindings/sound/atmel-i2s.txt
@@ -15,7 +15,6 @@ Required properties:
 - clock-names:    Should be one of each entry matching the clocks phandles list:
                   - "pclk" (peripheral clock) Required.
                   - "gclk" (generated clock) Optional (1).
-                  - "aclk" (Audio PLL clock) Optional (1).
                   - "muxclk" (I2S mux clock) Optional (1).
 
 Optional properties:
@@ -23,9 +22,9 @@ Optional properties:
 - princtrl-names: Should contain only one value - "default".
 
 
-(1) : Only the peripheral clock is required. The generated clock, the Audio
-      PLL clock adn the I2S mux clock are optional and should only be set
-      together, when Master Mode is required.
+(1) : Only the peripheral clock is required. The generated clock and the I2S
+      mux clock are optional and should only be set together, when Master Mode
+      is required.
 
 Example:
 
@@ -40,8 +39,8 @@ Example:
 			(AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) |
 			 AT91_XDMAC_DT_PERID(32))>;
 		dma-names = "tx", "rx";
-		clocks = <&i2s0_clk>, <&i2s0_gclk>, <&audio_pll_pmc>, <&i2s0muxck>;
-		clock-names = "pclk", "gclk", "aclk", "muxclk";
+		clocks = <&i2s0_clk>, <&i2s0_gclk>, <&i2s0muxck>;
+		clock-names = "pclk", "gclk", "muxclk";
 		pinctrl-names = "default";
 		pinctrl-0 = <&pinctrl_i2s0_default>;
 	};
diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-card.txt
index d04ea3b1a1dd..7e63e53a901c 100644
--- a/Documentation/devicetree/bindings/sound/audio-graph-card.txt
+++ b/Documentation/devicetree/bindings/sound/audio-graph-card.txt
@@ -18,6 +18,8 @@ Below are same as Simple-Card.
 - bitclock-inversion
 - frame-inversion
 - mclk-fs
+- hp-det-gpio
+- mic-det-gpio
 - dai-tdm-slot-num
 - dai-tdm-slot-width
 - clocks / system-clock-frequency
diff --git a/Documentation/devicetree/bindings/sound/dioo,dio2125.txt b/Documentation/devicetree/bindings/sound/dioo,dio2125.txt
deleted file mode 100644
index 63dbfe0f11d0..000000000000
--- a/Documentation/devicetree/bindings/sound/dioo,dio2125.txt
+++ /dev/null
@@ -1,12 +0,0 @@
-DIO2125 Audio Driver
-
-Required properties:
-- compatible : "dioo,dio2125"
-- enable-gpios : the gpio connected to the enable pin of the dio2125
-
-Example:
-
-amp: analog-amplifier {
-	compatible = "dioo,dio2125";
-	enable-gpios = <&gpio GPIOH_3 0>;
-};
diff --git a/Documentation/devicetree/bindings/sound/everest,es7134.txt b/Documentation/devicetree/bindings/sound/everest,es7134.txt
index 5495a3cb8b7b..091666069bde 100644
--- a/Documentation/devicetree/bindings/sound/everest,es7134.txt
+++ b/Documentation/devicetree/bindings/sound/everest,es7134.txt
@@ -1,10 +1,15 @@
 ES7134 i2s DA converter
 
 Required properties:
-- compatible : "everest,es7134" or "everest,es7144"
+- compatible : "everest,es7134" or
+               "everest,es7144" or
+	       "everest,es7154"
+- VDD-supply : regulator phandle for the VDD supply
+- PVDD-supply: regulator phandle for the PVDD supply for the es7154
 
 Example:
 
 i2s_codec: external-codec {
 	compatible = "everest,es7134";
+	VDD-supply = <&vcc_5v>;
 };
diff --git a/Documentation/devicetree/bindings/sound/everest,es7241.txt b/Documentation/devicetree/bindings/sound/everest,es7241.txt
new file mode 100644
index 000000000000..28f82cf4959f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/everest,es7241.txt
@@ -0,0 +1,28 @@
+ES7241 i2s AD converter
+
+Required properties:
+- compatible : "everest,es7241"
+- VDDP-supply: regulator phandle for the VDDA supply
+- VDDA-supply: regulator phandle for the VDDP supply
+- VDDD-supply: regulator phandle for the VDDD supply
+
+Optional properties:
+- reset-gpios: gpio connected to the reset pin
+- m0-gpios   : gpio connected to the m0 pin
+- m1-gpios   : gpio connected to the m1 pin
+- everest,sdout-pull-down:
+   Format used by the serial interface is controlled by pulling
+   the sdout. If the sdout is pulled down, leftj format is used.
+   If this property is not provided, sdout is assumed to pulled
+   up and i2s format is used
+
+Example:
+
+linein: audio-codec@2 {
+	#sound-dai-cells = <0>;
+	compatible = "everest,es7241";
+	VDDA-supply = <&vcc_3v3>;
+	VDDP-supply = <&vcc_3v3>;
+	VDDD-supply = <&vcc_3v3>;
+	reset-gpios = <&gpio GPIOH_42>;
+};
diff --git a/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt b/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt
new file mode 100644
index 000000000000..2ea85d5be6a4
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt
@@ -0,0 +1,27 @@
+Marvell PXA2xx audio complex
+
+This descriptions matches the AC97 controller found in pxa2xx and pxa3xx series.
+
+Required properties:
+  - compatible: should be one of the following:
+    "marvell,pxa250-ac97"
+    "marvell,pxa270-ac97"
+    "marvell,pxa300-ac97"
+  - reg: device MMIO address space
+  - interrupts: single interrupt generated by AC97 IP
+  - clocks: input clock of the AC97 IP, refer to clock-bindings.txt
+
+Optional properties:
+  - pinctrl-names, pinctrl-0: refer to pinctrl-bindings.txt
+  - reset-gpios: gpio used for AC97 reset, refer to gpio.txt
+
+Example:
+	ac97: sound@40500000 {
+		compatible = "marvell,pxa250-ac97";
+		reg = < 0x40500000 0x1000 >;
+		interrupts = <14>;
+		reset-gpios = <&gpio 113 GPIO_ACTIVE_HIGH>;
+		#sound-dai-cells = <1>;
+		pinctrl-names = "default";
+		pinctrl-0 = < &pmux_ac97_default >;
+	};
diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt
index 74c9ba6c2823..93b982e9419f 100644
--- a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt
+++ b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt
@@ -5,6 +5,14 @@ Required properties:
 	compatible	Must be "mrvl,pxa-ssp-dai"
 	port		A phandle reference to a PXA ssp upstream device
 
+Optional properties:
+
+	clock-names
+	clocks		Through "clock-names" and "clocks", external clocks
+			can be configured. If a clock names "extclk" exists,
+			it will be set to the mclk rate of the audio stream
+			and be used as clock provider of the DAI.
+
 Example:
 
 	/* upstream device */
diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt
deleted file mode 100644
index 551fbb8348c2..000000000000
--- a/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt
+++ /dev/null
@@ -1,15 +0,0 @@
-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";
-	};
-
diff --git a/Documentation/devicetree/bindings/sound/name-prefix.txt b/Documentation/devicetree/bindings/sound/name-prefix.txt
new file mode 100644
index 000000000000..645775908657
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/name-prefix.txt
@@ -0,0 +1,24 @@
+Name prefix:
+
+Card implementing the routing property define the connection between
+audio components as list of string pair. Component using the same
+sink/source names may use the name prefix property to prepend the
+name of their sinks/sources with the provided string.
+
+Optional name prefix property:
+- sound-name-prefix : string using as prefix for the sink/source names of
+		      the component.
+
+Example: Two instances of the same component.
+
+amp0: analog-amplifier@0 {
+	compatible = "simple-audio-amplifier";
+	enable-gpios = <&gpio GPIOH_3 0>;
+	sound-name-prefix = "FRONT";
+};
+
+amp1: analog-amplifier@1 {
+	compatible = "simple-audio-amplifier";
+	enable-gpios = <&gpio GPIOH_4 0>;
+	sound-name-prefix = "BACK";
+};
diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8096.txt b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt
index c7600a93ab39..c814e867850f 100644
--- a/Documentation/devicetree/bindings/sound/qcom,apq8096.txt
+++ b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt
@@ -7,7 +7,7 @@ This binding describes the APQ8096 sound card, which uses qdsp for audio.
 	Value type: <stringlist>
 	Definition: must be "qcom,apq8096-sndcard"
 
-- qcom,audio-routing:
+- audio-routing:
 	Usage: Optional
 	Value type: <stringlist>
 	Definition:  A list of the connections between audio components.
@@ -49,6 +49,12 @@ This binding describes the APQ8096 sound card, which uses qdsp for audio.
 			"DMIC1"
 			"DMIC2"
 			"DMIC3"
+
+- model:
+	Usage: required
+	Value type: <stringlist>
+	Definition: The user-visible name of this sound card.
+
 = dailinks
 Each subnode of sndcard represents either a dailink, and subnodes of each
 dailinks would be cpu/codec/platform dais.
@@ -79,11 +85,16 @@ dailinks would be cpu/codec/platform dais.
 	Value type: <phandle with arguments>
 	Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node.
 
+Obsolete:
+	qcom,model: String for soundcard name (Use model instead)
+	qcom,audio-routing: A list of the connections between audio components.
+			    (Use audio-routing instead)
+
 Example:
 
 audio {
 	compatible = "qcom,apq8096-sndcard";
-	qcom,model = "DB820c";
+	model = "DB820c";
 
 	mm1-dai-link {
 		link-name = "MultiMedia1";
diff --git a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt
index cb709e5dbc44..bbae426cdfb1 100644
--- a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt
+++ b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt
@@ -18,6 +18,11 @@ used by the apr service device.
 = ADM routing
 "routing" subnode of the ADM node represents adm routing specific configuration
 
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6adm-routing".
+
 - #sound-dai-cells
 	Usage: required
 	Value type: <u32>
@@ -28,6 +33,7 @@ q6adm@8 {
 	compatible = "qcom,q6adm";
 	reg = <APR_SVC_ADM>;
 	q6routing: routing {
+		compatible = "qcom,q6adm-routing";
 		#sound-dai-cells = <0>;
 	};
 };
diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
index bdbf87df8c0b..a8179409c194 100644
--- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
+++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
@@ -17,6 +17,11 @@ used by all apr services. Must contain the following properties.
 subnode of "dais" representing board specific dai setup.
 "dais" node should have following properties followed by dai children.
 
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6afe-dais"
+
 - #sound-dai-cells
 	Usage: required
 	Value type: <u32>
@@ -100,6 +105,7 @@ q6afe@4 {
 	reg = <APR_SVC_AFE>;
 
 	dais {
+		compatible = "qcom,q6afe-dais";
 		#sound-dai-cells = <1>;
 		#address-cells = <1>;
 		#size-cells = <0>;
diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt
index 2178eb91146f..f9c7bd8c1bc0 100644
--- a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt
+++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt
@@ -17,6 +17,11 @@ used by the apr service device.
 = ASM DAIs (Digial Audio Interface)
 "dais" subnode of the ASM node represents dai specific configuration
 
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6asm-dais".
+
 - #sound-dai-cells
 	Usage: required
 	Value type: <u32>
@@ -28,6 +33,7 @@ q6asm@7 {
 	compatible = "qcom,q6asm";
 	reg = <APR_SVC_ASM>;
 	q6asmdai: dais {
+		compatible = "qcom,q6asm-dais";
 		#sound-dai-cells = <1>;
 	};
 };
diff --git a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt
new file mode 100644
index 000000000000..408c4837e6d5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt
@@ -0,0 +1,80 @@
+* Qualcomm Technologies Inc. SDM845 ASoC sound card driver
+
+This binding describes the SDM845 sound card, which uses qdsp for audio.
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,sdm845-sndcard"
+
+- audio-routing:
+	Usage: Optional
+	Value type: <stringlist>
+	Definition:  A list of the connections between audio components.
+		  Each entry is a pair of strings, the first being the
+		  connection's sink, the second being the connection's
+		  source. Valid names could be power supplies, MicBias
+		  of codec and the jacks on the board.
+
+- model:
+	Usage: required
+	Value type: <stringlist>
+	Definition: The user-visible name of this sound card.
+
+= dailinks
+Each subnode of sndcard represents either a dailink, and subnodes of each
+dailinks would be cpu/codec/platform dais.
+
+- link-name:
+	Usage: required
+	Value type: <string>
+	Definition: User friendly name for dai link
+
+= CPU, PLATFORM, CODEC dais subnodes
+- cpu:
+	Usage: required
+	Value type: <subnode>
+	Definition: cpu dai sub-node
+
+- codec:
+	Usage: required
+	Value type: <subnode>
+	Definition: codec dai sub-node
+
+- platform:
+	Usage: Optional
+	Value type: <subnode>
+	Definition: platform dai sub-node
+
+- sound-dai:
+	Usage: required
+	Value type: <phandle>
+	Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node.
+
+Example:
+
+audio {
+	compatible = "qcom,sdm845-sndcard";
+	model = "sdm845-snd-card";
+	pinctrl-names = "default", "sleep";
+	pinctrl-0 = <&pri_mi2s_active &pri_mi2s_ws_active>;
+	pinctrl-1 = <&pri_mi2s_sleep &pri_mi2s_ws_sleep>;
+
+	mm1-dai-link {
+		link-name = "MultiMedia1";
+		cpu {
+			sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>;
+		};
+	};
+
+	pri-mi2s-dai-link {
+		link-name = "PRI MI2S Playback";
+		cpu {
+			sound-dai = <&q6afedai PRIMARY_MI2S_RX>;
+		};
+
+		platform {
+			sound-dai = <&q6routing>;
+		};
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt
new file mode 100644
index 000000000000..1d8d49e30af7
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt
@@ -0,0 +1,123 @@
+QCOM WCD9335 Codec
+
+Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC, supports
+Qualcomm Technologies, Inc. (QTI) multimedia solutions, including
+the MSM8996, MSM8976, and MSM8956 chipsets. It has in-built
+Soundwire controller, interrupt mux. It supports both I2S/I2C and
+SLIMbus audio interfaces.
+
+Required properties with SLIMbus Interface:
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: For SLIMbus interface it should be "slimMID,PID",
+		    textual representation of Manufacturer ID, Product Code,
+		    shall be in lower case hexadecimal with leading zeroes
+		    suppressed.  Refer to slimbus/bus.txt for details.
+		    Should be:
+		    "slim217,1a0" for MSM8996 and APQ8096 SoCs with SLIMbus.
+
+- reg
+	Usage: required
+	Value type: <u32 u32>
+	Definition: Should be ('Device index', 'Instance ID')
+
+- interrupts
+	Usage: required
+	Value type: <prop-encoded-array>
+	Definition: Interrupts via WCD INTR1 and INTR2 pins
+
+- interrupt-names:
+	Usage: required
+	Value type: <String array>
+	Definition: Interrupt names of WCD INTR1 and INTR2
+	Should be: "intr1", "intr2"
+
+- reset-gpio:
+	Usage: required
+	Value type: <String Array>
+	Definition: Reset gpio line
+
+- qcom,ifd:
+	Usage: required
+	Value type: <phandle>
+	Definition: SLIM interface device
+
+- clocks:
+	Usage: required
+	Value type: <prop-encoded-array>
+	Definition: See clock-bindings.txt section "consumers". List of
+                three clock specifiers for mclk, mclk2 and slimbus clock.
+
+- clock-names:
+	Usage: required
+	Value type: <string>
+	Definition: Must contain "mclk", "mclk2" and "slimbus" strings.
+
+- vdd-buck-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V buck supply
+
+- vdd-buck-sido-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V SIDO buck supply
+
+- vdd-rx-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V rx supply
+
+- vdd-tx-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V tx supply
+
+- vdd-vbat-supply:
+	Usage: Optional
+	Value type: <phandle>
+	Definition: Should contain a reference to the vbat supply
+
+- vdd-micbias-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the micbias supply
+
+- vdd-io-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V io supply
+
+- interrupt-controller:
+	Usage: required
+	Definition: Indicating that this is a interrupt controller
+
+- #interrupt-cells:
+	Usage: required
+	Value type: <int>
+	Definition: should be 1
+
+#sound-dai-cells
+	Usage: required
+	Value type: <u32>
+	Definition: Must be 1
+
+codec@1{
+	compatible = "slim217,1a0";
+	reg  = <1 0>;
+	interrupts = <&msmgpio 54 IRQ_TYPE_LEVEL_HIGH>;
+	interrupt-names = "intr2"
+	reset-gpio = <&msmgpio 64 0>;
+	qcom,ifd  = <&wc9335_ifd>;
+	clock-names = "mclk", "native";
+	clocks = <&rpmcc RPM_SMD_DIV_CLK1>,
+		 <&rpmcc RPM_SMD_BB_CLK1>;
+	vdd-buck-supply = <&pm8994_s4>;
+	vdd-rx-supply = <&pm8994_s4>;
+	vdd-buck-sido-supply = <&pm8994_s4>;
+	vdd-tx-supply = <&pm8994_s4>;
+	vdd-io-supply = <&pm8994_s4>;
+	#sound-dai-cells = <1>;
+}
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index b86d790f630f..9e764270c36b 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -352,6 +352,7 @@ Required properties:
 				    - "renesas,rcar_sound-r8a7794" (R-Car E2)
 				    - "renesas,rcar_sound-r8a7795" (R-Car H3)
 				    - "renesas,rcar_sound-r8a7796" (R-Car M3-W)
+				    - "renesas,rcar_sound-r8a77965" (R-Car M3-N)
 - reg				: Should contain the register physical address.
 				  required register is
 				   SRU/ADG/SSI      if generation1
diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
index b208a752576c..54aefab71f2c 100644
--- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
+++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
@@ -7,6 +7,7 @@ Required properties:
 
 - compatible: should be one of the following:
    - "rockchip,rk3066-i2s": for rk3066
+   - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30
    - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036
    - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188
    - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228
diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt
new file mode 100644
index 000000000000..312e9a129530
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5682.txt
@@ -0,0 +1,50 @@
+RT5682 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5682" or "realtek,rt5682i"
+
+- reg : The I2C address of the device.
+
+Optional properties:
+
+- interrupts : The CODEC's interrupt output.
+
+- realtek,dmic1-data-pin
+  0: dmic1 is not used
+  1: using GPIO2 pin as dmic1 data pin
+  2: using GPIO5 pin as dmic1 data pin
+
+- realtek,dmic1-clk-pin
+  0: using GPIO1 pin as dmic1 clock pin
+  1: using GPIO3 pin as dmic1 clock pin
+
+- realtek,jd-src
+  0: No JD is used
+  1: using JD1 as JD source
+
+- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
+
+Pins on the device (for linking into audio routes) for RT5682:
+
+  * DMIC L1
+  * DMIC R1
+  * IN1P
+  * HPOL
+  * HPOR
+
+Example:
+
+rt5682 {
+	compatible = "realtek,rt5682i";
+	reg = <0x1a>;
+	interrupt-parent = <&gpio>;
+	interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>;
+	realtek,ldo1-en-gpios =
+		<&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>;
+	realtek,dmic1-data-pin = <1>;
+	realtek,dmic1-clk-pin = <1>;
+	realtek,jd-src = <1>;
+};
diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt
index 0f214457476f..9c58f724396a 100644
--- a/Documentation/devicetree/bindings/sound/sgtl5000.txt
+++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt
@@ -17,7 +17,7 @@ Optional properties:
 
 - VDDD-supply : the regulator provider of VDDD
 
-- micbias-resistor-k-ohms : the bias resistor to be used in kOmhs
+- micbias-resistor-k-ohms : the bias resistor to be used in kOhms
 	The resistor can take values of 2k, 4k or 8k.
 	If set to 0 it will be off.
 	If this node is not mentioned or if the value is unknown, then
diff --git a/Documentation/devicetree/bindings/sound/simple-amplifier.txt b/Documentation/devicetree/bindings/sound/simple-amplifier.txt
new file mode 100644
index 000000000000..8647edae7af0
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/simple-amplifier.txt
@@ -0,0 +1,12 @@
+Simple Amplifier Audio Driver
+
+Required properties:
+- compatible : "dioo,dio2125" or "simple-audio-amplifier"
+- enable-gpios : the gpio connected to the enable pin of the simple amplifier
+
+Example:
+
+amp: analog-amplifier {
+	compatible = "simple-audio-amplifier";
+	enable-gpios = <&gpio GPIOH_3 0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/tas571x.txt b/Documentation/devicetree/bindings/sound/tas571x.txt
index b4959f10b74b..7c8fd37c2f9e 100644
--- a/Documentation/devicetree/bindings/sound/tas571x.txt
+++ b/Documentation/devicetree/bindings/sound/tas571x.txt
@@ -7,6 +7,7 @@ powerdown (optional).
 Required properties:
 
 - compatible: should be one of the following:
+  - "ti,tas5707"
   - "ti,tas5711",
   - "ti,tas5717",
   - "ti,tas5719",
diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst
index 4d83c1c0ca04..4a3cecc8ad38 100644
--- a/Documentation/sound/alsa-configuration.rst
+++ b/Documentation/sound/alsa-configuration.rst
@@ -1568,7 +1568,7 @@ joystick_io
 The driver requires firmware files ``turtlebeach/msndinit.bin`` and
 ``turtlebeach/msndperm.bin`` in the proper firmware directory.
 
-See Documentation/sound/oss/MultiSound for important information
+See Documentation/sound/cards/multisound.sh for important information
 about this driver.  Note that it has been discontinued, but the 
 Voyetra Turtle Beach knowledge base entry for it is still available
 at
diff --git a/Documentation/sound/cards/multisound.sh b/Documentation/sound/cards/multisound.sh
new file mode 100755
index 000000000000..a915a1affcde
--- /dev/null
+++ b/Documentation/sound/cards/multisound.sh
@@ -0,0 +1,1139 @@
+#! /bin/sh
+#
+#  Turtle Beach MultiSound Driver Notes
+#  -- Andrew Veliath <andrewtv@usa.net>
+#
+#  Last update:                      September 10, 1998
+#  Corresponding msnd driver:        0.8.3
+#
+# ** This file is a README (top part) and shell archive (bottom part).
+#    The corresponding archived utility sources can be unpacked by
+#    running `sh MultiSound' (the utilities are only needed for the
+#    Pinnacle and Fiji cards). **
+#
+#
+#  -=-=- Getting Firmware -=-=-
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  See the section `Obtaining and Creating Firmware Files' in this
+#  document for instructions on obtaining the necessary firmware
+#  files.
+#
+#
+#  Supported Features
+#  ~~~~~~~~~~~~~~~~~~
+#
+#  Currently, full-duplex digital audio (/dev/dsp only, /dev/audio is
+#  not currently available) and mixer functionality (/dev/mixer) are
+#  supported (memory mapped digital audio is not yet supported).
+#  Digital transfers and monitoring can be done as well if you have
+#  the digital daughterboard (see the section on using the S/PDIF port
+#  for more information).
+#
+#  Support for the Turtle Beach MultiSound Hurricane architecture is
+#  composed of the following modules (these can also operate compiled
+#  into the kernel):
+#
+#  snd-msnd-lib           - MultiSound base (requires snd)
+#
+#  snd-msnd-classic       - Base audio/mixer support for Classic, Monetery and
+#                           Tahiti cards
+#
+#  snd-msnd-pinnacle      - Base audio/mixer support for Pinnacle and Fiji cards
+#
+#
+#  Important Notes - Read Before Using
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  The firmware files are not included (may change in future).  You
+#  must obtain these images from Turtle Beach (they are included in
+#  the MultiSound Development Kits), and place them in /etc/sound for
+#  example, and give the full paths in the Linux configuration.  If
+#  you are compiling in support for the MultiSound driver rather than
+#  using it as a module, these firmware files must be accessible
+#  during kernel compilation.
+#
+#  Please note these files must be binary files, not assembler.  See
+#  the section later in this document for instructions to obtain these
+#  files.
+#
+#
+#  Configuring Card Resources
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  ** This section is very important, as your card may not work at all
+#     or your machine may crash if you do not do this correctly. **
+#
+#  * Classic/Monterey/Tahiti
+#
+#  These cards are configured through the driver snd-msnd-classic.  You must
+#  know the io port, then the driver will select the irq and memory resources
+#  on the card.  It is up to you to know if these are free locations or now,
+#  a conflict can lock the machine up.
+#
+#  * Pinnacle/Fiji
+#
+#  The Pinnacle and Fiji cards have an extra config port, either
+#  0x250, 0x260 or 0x270.  This port can be disabled to have the card
+#  configured strictly through PnP, however you lose the ability to
+#  access the IDE controller and joystick devices on this card when
+#  using PnP.  The included pinnaclecfg program in this shell archive
+#  can be used to configure the card in non-PnP mode, and in PnP mode
+#  you can use isapnptools.  These are described briefly here.
+#
+#  pinnaclecfg is not required; you can use the snd-msnd-pinnacle module
+#  to fully configure the card as well.  However, pinnaclecfg can be
+#  used to change the resource values of a particular device after the
+#  snd-msnd-pinnacle module has been loaded.  If you are compiling the
+#  driver into the kernel, you must set these values during compile
+#  time, however other peripheral resource values can be changed with
+#  the pinnaclecfg program after the kernel is loaded.
+#
+#
+#  *** PnP mode
+#
+#  Use pnpdump to obtain a sample configuration if you can; I was able
+#  to obtain one with the command `pnpdump 1 0x203' -- this may vary
+#  for you (running pnpdump by itself did not work for me).  Then,
+#  edit this file and use isapnp to uncomment and set the card values.
+#  Use these values when inserting the snd-msnd-pinnacle module.  Using
+#  this method, you can set the resources for the DSP and the Kurzweil
+#  synth (Pinnacle).  Since Linux does not directly support PnP
+#  devices, you may have difficulty when using the card in PnP mode
+#  when it the driver is compiled into the kernel.  Using non-PnP mode
+#  is preferable in this case.
+#
+#  Here is an example mypinnacle.conf for isapnp that sets the card to
+#  io base 0x210, irq 5 and mem 0xd8000, and also sets the Kurzweil
+#  synth to 0x330 and irq 9 (may need editing for your system):
+#
+#  (READPORT 0x0203)
+#  (CSN 2)
+#  (IDENTIFY *)
+#
+#  # DSP
+#  (CONFIGURE BVJ0440/-1 (LD 0
+#          (INT 0 (IRQ 5 (MODE +E))) (IO 0 (BASE 0x0210)) (MEM 0 (BASE 0x0d8000))
+#          (ACT Y)))
+#
+#  # Kurzweil Synth (Pinnacle Only)
+#  (CONFIGURE BVJ0440/-1 (LD 1
+#          (IO 0 (BASE 0x0330)) (INT 0 (IRQ 9 (MODE +E)))
+#          (ACT Y)))
+#
+#  (WAITFORKEY)
+#
+#
+#  *** Non-PnP mode
+#
+#  The second way is by running the card in non-PnP mode.  This
+#  actually has some advantages in that you can access some other
+#  devices on the card, such as the joystick and IDE controller.  To
+#  configure the card, unpack this shell archive and build the
+#  pinnaclecfg program.  Using this program, you can assign the
+#  resource values to the card's devices, or disable the devices.  As
+#  an alternative to using pinnaclecfg, you can specify many of the
+#  configuration values when loading the snd-msnd-pinnacle module (or
+#  during kernel configuration when compiling the driver into the
+#  kernel).
+#
+#  If you specify cfg=0x250 for the snd-msnd-pinnacle module, it
+#  automatically configure the card to the given io, irq and memory
+#  values using that config port (the config port is jumper selectable
+#  on the card to 0x250, 0x260 or 0x270).
+#
+#  See the `snd-msnd-pinnacle Additional Options' section below for more
+#  information on these parameters (also, if you compile the driver
+#  directly into the kernel, these extra parameters can be useful
+#  here).
+#
+#
+# ** It is very easy to cause problems in your machine if you choose a
+#    resource value which is incorrect. **
+#
+#
+#  Examples
+#  ~~~~~~~~
+#
+#  * MultiSound Classic/Monterey/Tahiti:
+#
+#  modprobe snd
+#  insmod snd-msnd-lib
+#  insmod snd-msnd-classic io=0x290 irq=7 mem=0xd0000
+#
+#  * MultiSound Pinnacle in PnP mode:
+#
+#  modprobe snd
+#  insmod snd-msnd-lib
+#  isapnp mypinnacle.conf
+#  insmod snd-msnd-pinnacle io=0x210 irq=5 mem=0xd8000 <-- match mypinnacle.conf values
+#
+#  * MultiSound Pinnacle in non-PnP mode (replace 0x250 with your configuration port,
+#    one of 0x250, 0x260 or 0x270):
+#
+#  modprobe snd
+#  insmod snd-msnd-lib
+#  insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000
+#
+# * To use the MPU-compatible Kurzweil synth on the Pinnacle in PnP
+#   mode, add the following (assumes you did `isapnp mypinnacle.conf'):
+#
+#  insmod snd
+#  insmod mpu401 io=0x330 irq=9                    <-- match mypinnacle.conf values
+#
+# * To use the MPU-compatible Kurzweil synth on the Pinnacle in non-PnP
+#   mode, add the following.  Note how we first configure the peripheral's
+#   resources, _then_ install a Linux driver for it:
+#
+#  insmod snd
+#  pinnaclecfg 0x250 mpu 0x330 9
+#  insmod mpu401 io=0x330 irq=9
+#
+#  -- OR you can use the following sequence without pinnaclecfg in non-PnP mode:
+#
+#  modprobe snd
+#  insmod snd-msnd-lib
+#  insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 mpu_io=0x330 mpu_irq=9
+#  insmod snd
+#  insmod mpu401 io=0x330 irq=9
+#
+# * To setup the joystick port on the Pinnacle in non-PnP mode (though
+#   you have to find the actual Linux joystick driver elsewhere), you
+#   can use pinnaclecfg:
+#
+#   pinnaclecfg 0x250 joystick 0x200
+#
+#  -- OR you can configure this using snd-msnd-pinnacle with the following:
+#
+#  modprobe snd
+#  insmod snd-msnd-lib
+#  insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 joystick_io=0x200
+#
+#
+#  snd-msnd-classic, snd-msnd-pinnacle Required Options
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  If the following options are not given, the module will not load.
+#  Examine the kernel message log for informative error messages.
+#  WARNING--probing isn't supported so try to make sure you have the
+#  correct shared memory area, otherwise you may experience problems.
+#
+#  io                   I/O base of DSP, e.g. io=0x210
+#  irq                  IRQ number, e.g. irq=5
+#  mem                  Shared memory area, e.g. mem=0xd8000
+#
+#
+#  snd-msnd-classic, snd-msnd-pinnacle Additional Options
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  fifosize             The digital audio FIFOs, in kilobytes.  If not
+#                       specified, the default will be used.  Increasing
+#                       this value will reduce the chance of a FIFO
+#                       underflow at the expense of increasing overall
+#                       latency.  For example, fifosize=512 will
+#                       allocate 512kB read and write FIFOs (1MB total).
+#                       While this may reduce dropouts, a heavy machine
+#                       load will undoubtedly starve the FIFO of data
+#                       and you will eventually get dropouts.  One
+#                       option is to alter the scheduling priority of
+#                       the playback process, using `nice' or some form
+#                       of POSIX soft real-time scheduling.
+#
+#  calibrate_signal     Setting this to one calibrates the ADCs to the
+#                       signal, zero calibrates to the card (defaults
+#                       to zero).
+#
+#
+#  snd-msnd-pinnacle Additional Options
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  digital              Specify digital=1 to enable the S/PDIF input
+#                       if you have the digital daughterboard
+#                       adapter. This will enable access to the
+#                       DIGITAL1 input for the soundcard in the mixer.
+#                       Some mixer programs might have trouble setting
+#                       the DIGITAL1 source as an input.  If you have
+#                       trouble, you can try the setdigital.c program
+#                       at the bottom of this document.
+#
+#  cfg                  Non-PnP configuration port for the Pinnacle
+#                       and Fiji (typically 0x250, 0x260 or 0x270,
+#                       depending on the jumper configuration).  If
+#                       this option is omitted, then it is assumed
+#                       that the card is in PnP mode, and that the
+#                       specified DSP resource values are already
+#                       configured with PnP (i.e. it won't attempt to
+#                       do any sort of configuration).
+#
+#  When the Pinnacle is in non-PnP mode, you can use the following
+#  options to configure particular devices.  If a full specification
+#  for a device is not given, then the device is not configured.  Note
+#  that you still must use a Linux driver for any of these devices
+#  once their resources are setup (such as the Linux joystick driver,
+#  or the MPU401 driver from OSS for the Kurzweil synth).
+#
+#  mpu_io               I/O port of MPU (on-board Kurzweil synth)
+#  mpu_irq              IRQ of MPU (on-board Kurzweil synth)
+#  ide_io0		First I/O port of IDE controller
+#  ide_io1		Second I/O port of IDE controller
+#  ide_irq		IRQ IDE controller
+#  joystick_io          I/O port of joystick
+#
+#
+#  Obtaining and Creating Firmware Files
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#       For the Classic/Tahiti/Monterey
+#       ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  Download to /tmp and unzip the following file from Turtle Beach:
+#
+#       ftp://ftp.voyetra.com/pub/tbs/msndcl/msndvkit.zip
+#
+#  When unzipped, unzip the file named MsndFiles.zip.  Then copy the
+#  following firmware files to /etc/sound (note the file renaming):
+#
+#    cp DSPCODE/MSNDINIT.BIN /etc/sound/msndinit.bin
+#    cp DSPCODE/MSNDPERM.REB /etc/sound/msndperm.bin
+#
+#  When configuring the Linux kernel, specify /etc/sound/msndinit.bin and
+#  /etc/sound/msndperm.bin for the two firmware files (Linux kernel
+#  versions older than 2.2 do not ask for firmware paths, and are
+#  hardcoded to /etc/sound).
+#
+#  If you are compiling the driver into the kernel, these files must
+#  be accessible during compilation, but will not be needed later.
+#  The files must remain, however, if the driver is used as a module.
+#
+#
+#       For the Pinnacle/Fiji
+#       ~~~~~~~~~~~~~~~~~~~~~
+#
+#  Download to /tmp and unzip the following file from Turtle Beach (be
+#  sure to use the entire URL; some have had trouble navigating to the
+#  URL):
+#
+#       ftp://ftp.voyetra.com/pub/tbs/pinn/pnddk100.zip
+#
+#  Unpack this shell archive, and run make in the created directory
+#  (you need a C compiler and flex to build the utilities).  This
+#  should give you the executables conv, pinnaclecfg and setdigital.
+#  conv is only used temporarily here to create the firmware files,
+#  while pinnaclecfg is used to configure the Pinnacle or Fiji card in
+#  non-PnP mode, and setdigital can be used to set the S/PDIF input on
+#  the mixer (pinnaclecfg and setdigital should be copied to a
+#  convenient place, possibly run during system initialization).
+#
+#  To generating the firmware files with the `conv' program, we create
+#  the binary firmware files by doing the following conversion
+#  (assuming the archive unpacked into a directory named PINNDDK):
+#
+#    ./conv < PINNDDK/dspcode/pndspini.asm > /etc/sound/pndspini.bin
+#    ./conv < PINNDDK/dspcode/pndsperm.asm > /etc/sound/pndsperm.bin
+#
+#  The conv (and conv.l) program is not needed after conversion and can
+#  be safely deleted.  Then, when configuring the Linux kernel, specify
+#  /etc/sound/pndspini.bin and /etc/sound/pndsperm.bin for the two
+#  firmware files (Linux kernel versions older than 2.2 do not ask for
+#  firmware paths, and are hardcoded to /etc/sound).
+#
+#  If you are compiling the driver into the kernel, these files must
+#  be accessible during compilation, but will not be needed later.
+#  The files must remain, however, if the driver is used as a module.
+#
+#
+#  Using Digital I/O with the S/PDIF Port
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  If you have a Pinnacle or Fiji with the digital daughterboard and
+#  want to set it as the input source, you can use this program if you
+#  have trouble trying to do it with a mixer program (be sure to
+#  insert the module with the digital=1 option, or say Y to the option
+#  during compiled-in kernel operation).  Upon selection of the S/PDIF
+#  port, you should be able monitor and record from it.
+#
+#  There is something to note about using the S/PDIF port.  Digital
+#  timing is taken from the digital signal, so if a signal is not
+#  connected to the port and it is selected as recording input, you
+#  will find PCM playback to be distorted in playback rate.  Also,
+#  attempting to record at a sampling rate other than the DAT rate may
+#  be problematic (i.e. trying to record at 8000Hz when the DAT signal
+#  is 44100Hz).  If you have a problem with this, set the recording
+#  input to analog if you need to record at a rate other than that of
+#  the DAT rate.
+#
+#
+#  -- Shell archive attached below, just run `sh MultiSound' to extract.
+#     Contains Pinnacle/Fiji utilities to convert firmware, configure
+#     in non-PnP mode, and select the DIGITAL1 input for the mixer.
+#
+#
+#!/bin/sh
+# This is a shell archive (produced by GNU sharutils 4.2).
+# To extract the files from this archive, save it to some FILE, remove
+# everything before the `!/bin/sh' line above, then type `sh FILE'.
+#
+# Made on 1998-12-04 10:07 EST by <andrewtv@ztransform.velsoft.com>.
+# Source directory was `/home/andrewtv/programming/pinnacle/pinnacle'.
+#
+# Existing files will *not* be overwritten unless `-c' is specified.
+#
+# This shar contains:
+# length mode       name
+# ------ ---------- ------------------------------------------
+#   2064 -rw-rw-r-- MultiSound.d/setdigital.c
+#  10224 -rw-rw-r-- MultiSound.d/pinnaclecfg.c
+#    106 -rw-rw-r-- MultiSound.d/Makefile
+#    146 -rw-rw-r-- MultiSound.d/conv.l
+#   1491 -rw-rw-r-- MultiSound.d/msndreset.c
+#
+save_IFS="${IFS}"
+IFS="${IFS}:"
+gettext_dir=FAILED
+locale_dir=FAILED
+first_param="$1"
+for dir in $PATH
+do
+  if test "$gettext_dir" = FAILED && test -f $dir/gettext \
+     && ($dir/gettext --version >/dev/null 2>&1)
+  then
+    set `$dir/gettext --version 2>&1`
+    if test "$3" = GNU
+    then
+      gettext_dir=$dir
+    fi
+  fi
+  if test "$locale_dir" = FAILED && test -f $dir/shar \
+     && ($dir/shar --print-text-domain-dir >/dev/null 2>&1)
+  then
+    locale_dir=`$dir/shar --print-text-domain-dir`
+  fi
+done
+IFS="$save_IFS"
+if test "$locale_dir" = FAILED || test "$gettext_dir" = FAILED
+then
+  echo=echo
+else
+  TEXTDOMAINDIR=$locale_dir
+  export TEXTDOMAINDIR
+  TEXTDOMAIN=sharutils
+  export TEXTDOMAIN
+  echo="$gettext_dir/gettext -s"
+fi
+touch -am 1231235999 $$.touch >/dev/null 2>&1
+if test ! -f 1231235999 && test -f $$.touch; then
+  shar_touch=touch
+else
+  shar_touch=:
+  echo
+  $echo 'WARNING: not restoring timestamps.  Consider getting and'
+  $echo "installing GNU \`touch', distributed in GNU File Utilities..."
+  echo
+fi
+rm -f 1231235999 $$.touch
+#
+if mkdir _sh01426; then
+  $echo 'x -' 'creating lock directory'
+else
+  $echo 'failed to create lock directory'
+  exit 1
+fi
+# ============= MultiSound.d/setdigital.c ==============
+if test ! -d 'MultiSound.d'; then
+  $echo 'x -' 'creating directory' 'MultiSound.d'
+  mkdir 'MultiSound.d'
+fi
+if test -f 'MultiSound.d/setdigital.c' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/setdigital.c' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/setdigital.c' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/setdigital.c' &&
+/*********************************************************************
+X *
+X * setdigital.c - sets the DIGITAL1 input for a mixer
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <sys/soundcard.h>
+X
+int main(int argc, char *argv[])
+{
+X	int fd;
+X	unsigned long recmask, recsrc;
+X
+X	if (argc != 2) {
+X		fprintf(stderr, "usage: setdigital <mixer device>\n");
+X		exit(1);
+X	}
+X
+X	if ((fd = open(argv[1], O_RDWR)) < 0) {
+X		perror(argv[1]);
+X		exit(1);
+X	}
+X
+X	if (ioctl(fd, SOUND_MIXER_READ_RECMASK, &recmask) < 0) {
+X		fprintf(stderr, "error: ioctl read recording mask failed\n");
+X		perror("ioctl");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	if (!(recmask & SOUND_MASK_DIGITAL1)) {
+X		fprintf(stderr, "error: cannot find DIGITAL1 device in mixer\n");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	if (ioctl(fd, SOUND_MIXER_READ_RECSRC, &recsrc) < 0) {
+X		fprintf(stderr, "error: ioctl read recording source failed\n");
+X		perror("ioctl");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	recsrc |= SOUND_MASK_DIGITAL1;
+X
+X	if (ioctl(fd, SOUND_MIXER_WRITE_RECSRC, &recsrc) < 0) {
+X		fprintf(stderr, "error: ioctl write recording source failed\n");
+X		perror("ioctl");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	close(fd);
+X
+X	return 0;
+}
+SHAR_EOF
+  $shar_touch -am 1204092598 'MultiSound.d/setdigital.c' &&
+  chmod 0664 'MultiSound.d/setdigital.c' ||
+  $echo 'restore of' 'MultiSound.d/setdigital.c' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/setdigital.c:' 'MD5 check failed'
+e87217fc3e71288102ba41fd81f71ec4  MultiSound.d/setdigital.c
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/setdigital.c'`"
+    test 2064 -eq "$shar_count" ||
+    $echo 'MultiSound.d/setdigital.c:' 'original size' '2064,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/pinnaclecfg.c ==============
+if test -f 'MultiSound.d/pinnaclecfg.c' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/pinnaclecfg.c' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/pinnaclecfg.c' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/pinnaclecfg.c' &&
+/*********************************************************************
+X *
+X * pinnaclecfg.c - Pinnacle/Fiji Device Configuration Program
+X *
+X * This is for NON-PnP mode only.  For PnP mode, use isapnptools.
+X *
+X * This is Linux-specific, and must be run with root permissions.
+X *
+X * Part of the Turtle Beach MultiSound Sound Card Driver for Linux
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <errno.h>
+#include <unistd.h>
+#include <asm/types.h>
+#include <sys/io.h>
+X
+#define IREG_LOGDEVICE		0x07
+#define IREG_ACTIVATE		0x30
+#define LD_ACTIVATE		0x01
+#define LD_DISACTIVATE		0x00
+#define IREG_EECONTROL		0x3F
+#define IREG_MEMBASEHI		0x40
+#define IREG_MEMBASELO		0x41
+#define IREG_MEMCONTROL		0x42
+#define IREG_MEMRANGEHI		0x43
+#define IREG_MEMRANGELO		0x44
+#define MEMTYPE_8BIT		0x00
+#define MEMTYPE_16BIT		0x02
+#define MEMTYPE_RANGE		0x00
+#define MEMTYPE_HIADDR		0x01
+#define IREG_IO0_BASEHI		0x60
+#define IREG_IO0_BASELO		0x61
+#define IREG_IO1_BASEHI		0x62
+#define IREG_IO1_BASELO		0x63
+#define IREG_IRQ_NUMBER		0x70
+#define IREG_IRQ_TYPE		0x71
+#define IRQTYPE_HIGH		0x02
+#define IRQTYPE_LOW		0x00
+#define IRQTYPE_LEVEL		0x01
+#define IRQTYPE_EDGE		0x00
+X
+#define HIBYTE(w)		((BYTE)(((WORD)(w) >> 8) & 0xFF))
+#define LOBYTE(w)		((BYTE)(w))
+#define MAKEWORD(low,hi)	((WORD)(((BYTE)(low))|(((WORD)((BYTE)(hi)))<<8)))
+X
+typedef __u8			BYTE;
+typedef __u16			USHORT;
+typedef __u16			WORD;
+X
+static int config_port = -1;
+X
+static int msnd_write_cfg(int cfg, int reg, int value)
+{
+X	outb(reg, cfg);
+X	outb(value, cfg + 1);
+X	if (value != inb(cfg + 1)) {
+X		fprintf(stderr, "error: msnd_write_cfg: I/O error\n");
+X		return -EIO;
+X	}
+X	return 0;
+}
+X
+static int msnd_read_cfg(int cfg, int reg)
+{
+X	outb(reg, cfg);
+X	return inb(cfg + 1);
+}
+X
+static int msnd_write_cfg_io0(int cfg, int num, WORD io)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io)))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io)))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_io0(int cfg, int num, WORD *io)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X
+X	*io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO0_BASELO),
+X		       msnd_read_cfg(cfg, IREG_IO0_BASEHI));
+X
+X	return 0;
+}
+X
+static int msnd_write_cfg_io1(int cfg, int num, WORD io)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io)))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io)))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_io1(int cfg, int num, WORD *io)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X
+X	*io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO1_BASELO),
+X		       msnd_read_cfg(cfg, IREG_IO1_BASEHI));
+X
+X	return 0;
+}
+X
+static int msnd_write_cfg_irq(int cfg, int num, WORD irq)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq)))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_irq(int cfg, int num, WORD *irq)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X
+X	*irq = msnd_read_cfg(cfg, IREG_IRQ_NUMBER);
+X
+X	return 0;
+}
+X
+static int msnd_write_cfg_mem(int cfg, int num, int mem)
+{
+X	WORD wmem;
+X
+X	mem >>= 8;
+X	mem &= 0xfff;
+X	wmem = (WORD)mem;
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem)))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem)))
+X		return -EIO;
+X	if (wmem && msnd_write_cfg(cfg, IREG_MEMCONTROL, (MEMTYPE_HIADDR | MEMTYPE_16BIT)))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_mem(int cfg, int num, int *mem)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X
+X	*mem = MAKEWORD(msnd_read_cfg(cfg, IREG_MEMBASELO),
+X			msnd_read_cfg(cfg, IREG_MEMBASEHI));
+X	*mem <<= 8;
+X
+X	return 0;
+}
+X
+static int msnd_activate_logical(int cfg, int num)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_write_cfg_logical(int cfg, int num, WORD io0, WORD io1, WORD irq, int mem)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg_io0(cfg, num, io0))
+X		return -EIO;
+X	if (msnd_write_cfg_io1(cfg, num, io1))
+X		return -EIO;
+X	if (msnd_write_cfg_irq(cfg, num, irq))
+X		return -EIO;
+X	if (msnd_write_cfg_mem(cfg, num, mem))
+X		return -EIO;
+X	if (msnd_activate_logical(cfg, num))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_logical(int cfg, int num, WORD *io0, WORD *io1, WORD *irq, int *mem)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_read_cfg_io0(cfg, num, io0))
+X		return -EIO;
+X	if (msnd_read_cfg_io1(cfg, num, io1))
+X		return -EIO;
+X	if (msnd_read_cfg_irq(cfg, num, irq))
+X		return -EIO;
+X	if (msnd_read_cfg_mem(cfg, num, mem))
+X		return -EIO;
+X	return 0;
+}
+X
+static void usage(void)
+{
+X	fprintf(stderr,
+X		"\n"
+X		"pinnaclecfg 1.0\n"
+X		"\n"
+X		"usage: pinnaclecfg <config port> [device config]\n"
+X		"\n"
+X		"This is for use with the card in NON-PnP mode only.\n"
+X		"\n"
+X		"Available devices (not all available for Fiji):\n"
+X		"\n"
+X		"        Device                       Description\n"
+X		"        -------------------------------------------------------------------\n"
+X		"        reset                        Reset all devices (i.e. disable)\n"
+X		"        show                         Display current device configurations\n"
+X		"\n"
+X		"        dsp <io> <irq> <mem>         Audio device\n"
+X		"        mpu <io> <irq>               Internal Kurzweil synth\n"
+X		"        ide <io0> <io1> <irq>        On-board IDE controller\n"
+X		"        joystick <io>                Joystick port\n"
+X		"\n");
+X	exit(1);
+}
+X
+static int cfg_reset(void)
+{
+X	int i;
+X
+X	for (i = 0; i < 4; ++i)
+X		msnd_write_cfg_logical(config_port, i, 0, 0, 0, 0);
+X
+X	return 0;
+}
+X
+static int cfg_show(void)
+{
+X	int i;
+X	int count = 0;
+X
+X	for (i = 0; i < 4; ++i) {
+X		WORD io0, io1, irq;
+X		int mem;
+X		msnd_read_cfg_logical(config_port, i, &io0, &io1, &irq, &mem);
+X		switch (i) {
+X		case 0:
+X			if (io0 || irq || mem) {
+X				printf("dsp 0x%x %d 0x%x\n", io0, irq, mem);
+X				++count;
+X			}
+X			break;
+X		case 1:
+X			if (io0 || irq) {
+X				printf("mpu 0x%x %d\n", io0, irq);
+X				++count;
+X			}
+X			break;
+X		case 2:
+X			if (io0 || io1 || irq) {
+X				printf("ide 0x%x 0x%x %d\n", io0, io1, irq);
+X				++count;
+X			}
+X			break;
+X		case 3:
+X			if (io0) {
+X				printf("joystick 0x%x\n", io0);
+X				++count;
+X			}
+X			break;
+X		}
+X	}
+X
+X	if (count == 0)
+X		fprintf(stderr, "no devices configured\n");
+X
+X	return 0;
+}
+X
+static int cfg_dsp(int argc, char *argv[])
+{
+X	int io, irq, mem;
+X
+X	if (argc < 3 ||
+X	    sscanf(argv[0], "0x%x", &io) != 1 ||
+X	    sscanf(argv[1], "%d", &irq) != 1 ||
+X	    sscanf(argv[2], "0x%x", &mem) != 1)
+X		usage();
+X
+X	if (!(io == 0x290 ||
+X	      io == 0x260 ||
+X	      io == 0x250 ||
+X	      io == 0x240 ||
+X	      io == 0x230 ||
+X	      io == 0x220 ||
+X	      io == 0x210 ||
+X	      io == 0x3e0)) {
+X		fprintf(stderr, "error: io must be one of "
+X			"210, 220, 230, 240, 250, 260, 290, or 3E0\n");
+X		usage();
+X	}
+X
+X	if (!(irq == 5 ||
+X	      irq == 7 ||
+X	      irq == 9 ||
+X	      irq == 10 ||
+X	      irq == 11 ||
+X	      irq == 12)) {
+X		fprintf(stderr, "error: irq must be one of "
+X			"5, 7, 9, 10, 11 or 12\n");
+X		usage();
+X	}
+X
+X	if (!(mem == 0xb0000 ||
+X	      mem == 0xc8000 ||
+X	      mem == 0xd0000 ||
+X	      mem == 0xd8000 ||
+X	      mem == 0xe0000 ||
+X	      mem == 0xe8000)) {
+X		fprintf(stderr, "error: mem must be one of "
+X			"0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000\n");
+X		usage();
+X	}
+X
+X	return msnd_write_cfg_logical(config_port, 0, io, 0, irq, mem);
+}
+X
+static int cfg_mpu(int argc, char *argv[])
+{
+X	int io, irq;
+X
+X	if (argc < 2 ||
+X	    sscanf(argv[0], "0x%x", &io) != 1 ||
+X	    sscanf(argv[1], "%d", &irq) != 1)
+X		usage();
+X
+X	return msnd_write_cfg_logical(config_port, 1, io, 0, irq, 0);
+}
+X
+static int cfg_ide(int argc, char *argv[])
+{
+X	int io0, io1, irq;
+X
+X	if (argc < 3 ||
+X	    sscanf(argv[0], "0x%x", &io0) != 1 ||
+X	    sscanf(argv[0], "0x%x", &io1) != 1 ||
+X	    sscanf(argv[1], "%d", &irq) != 1)
+X		usage();
+X
+X	return msnd_write_cfg_logical(config_port, 2, io0, io1, irq, 0);
+}
+X
+static int cfg_joystick(int argc, char *argv[])
+{
+X	int io;
+X
+X	if (argc < 1 ||
+X	    sscanf(argv[0], "0x%x", &io) != 1)
+X		usage();
+X
+X	return msnd_write_cfg_logical(config_port, 3, io, 0, 0, 0);
+}
+X
+int main(int argc, char *argv[])
+{
+X	char *device;
+X	int rv = 0;
+X
+X	--argc; ++argv;
+X
+X	if (argc < 2)
+X		usage();
+X
+X	sscanf(argv[0], "0x%x", &config_port);
+X	if (config_port != 0x250 && config_port != 0x260 && config_port != 0x270) {
+X		fprintf(stderr, "error: <config port> must be 0x250, 0x260 or 0x270\n");
+X		exit(1);
+X	}
+X	if (ioperm(config_port, 2, 1)) {
+X		perror("ioperm");
+X		fprintf(stderr, "note: pinnaclecfg must be run as root\n");
+X		exit(1);
+X	}
+X	device = argv[1];
+X
+X	argc -= 2; argv += 2;
+X
+X	if (strcmp(device, "reset") == 0)
+X		rv = cfg_reset();
+X	else if (strcmp(device, "show") == 0)
+X		rv = cfg_show();
+X	else if (strcmp(device, "dsp") == 0)
+X		rv = cfg_dsp(argc, argv);
+X	else if (strcmp(device, "mpu") == 0)
+X		rv = cfg_mpu(argc, argv);
+X	else if (strcmp(device, "ide") == 0)
+X		rv = cfg_ide(argc, argv);
+X	else if (strcmp(device, "joystick") == 0)
+X		rv = cfg_joystick(argc, argv);
+X	else {
+X		fprintf(stderr, "error: unknown device %s\n", device);
+X		usage();
+X	}
+X
+X	if (rv)
+X		fprintf(stderr, "error: device configuration failed\n");
+X
+X	return 0;
+}
+SHAR_EOF
+  $shar_touch -am 1204092598 'MultiSound.d/pinnaclecfg.c' &&
+  chmod 0664 'MultiSound.d/pinnaclecfg.c' ||
+  $echo 'restore of' 'MultiSound.d/pinnaclecfg.c' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/pinnaclecfg.c:' 'MD5 check failed'
+366bdf27f0db767a3c7921d0a6db20fe  MultiSound.d/pinnaclecfg.c
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/pinnaclecfg.c'`"
+    test 10224 -eq "$shar_count" ||
+    $echo 'MultiSound.d/pinnaclecfg.c:' 'original size' '10224,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/Makefile ==============
+if test -f 'MultiSound.d/Makefile' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/Makefile' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/Makefile' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/Makefile' &&
+CC	= gcc
+CFLAGS	= -O
+PROGS	= setdigital msndreset pinnaclecfg conv
+X
+all: $(PROGS)
+X
+clean:
+X	rm -f $(PROGS)
+SHAR_EOF
+  $shar_touch -am 1204092398 'MultiSound.d/Makefile' &&
+  chmod 0664 'MultiSound.d/Makefile' ||
+  $echo 'restore of' 'MultiSound.d/Makefile' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/Makefile:' 'MD5 check failed'
+76ca8bb44e3882edcf79c97df6c81845  MultiSound.d/Makefile
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/Makefile'`"
+    test 106 -eq "$shar_count" ||
+    $echo 'MultiSound.d/Makefile:' 'original size' '106,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/conv.l ==============
+if test -f 'MultiSound.d/conv.l' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/conv.l' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/conv.l' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/conv.l' &&
+%%
+[ \n\t,\r]
+\;.*
+DB
+[0-9A-Fa-f]+H	{ int n; sscanf(yytext, "%xH", &n); printf("%c", n); }
+%%
+int yywrap() { return 1; }
+void main() { yylex(); }
+SHAR_EOF
+  $shar_touch -am 0828231798 'MultiSound.d/conv.l' &&
+  chmod 0664 'MultiSound.d/conv.l' ||
+  $echo 'restore of' 'MultiSound.d/conv.l' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/conv.l:' 'MD5 check failed'
+d2411fc32cd71a00dcdc1f009e858dd2  MultiSound.d/conv.l
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/conv.l'`"
+    test 146 -eq "$shar_count" ||
+    $echo 'MultiSound.d/conv.l:' 'original size' '146,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/msndreset.c ==============
+if test -f 'MultiSound.d/msndreset.c' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/msndreset.c' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/msndreset.c' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/msndreset.c' &&
+/*********************************************************************
+X *
+X * msndreset.c - resets the MultiSound card
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <sys/soundcard.h>
+X
+int main(int argc, char *argv[])
+{
+X	int fd;
+X
+X	if (argc != 2) {
+X		fprintf(stderr, "usage: msndreset <mixer device>\n");
+X		exit(1);
+X	}
+X
+X	if ((fd = open(argv[1], O_RDWR)) < 0) {
+X		perror(argv[1]);
+X		exit(1);
+X	}
+X
+X	if (ioctl(fd, SOUND_MIXER_PRIVATE1, 0) < 0) {
+X		fprintf(stderr, "error: msnd ioctl reset failed\n");
+X		perror("ioctl");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	close(fd);
+X
+X	return 0;
+}
+SHAR_EOF
+  $shar_touch -am 1204100698 'MultiSound.d/msndreset.c' &&
+  chmod 0664 'MultiSound.d/msndreset.c' ||
+  $echo 'restore of' 'MultiSound.d/msndreset.c' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/msndreset.c:' 'MD5 check failed'
+c52f876521084e8eb25e12e01dcccb8a  MultiSound.d/msndreset.c
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/msndreset.c'`"
+    test 1491 -eq "$shar_count" ||
+    $echo 'MultiSound.d/msndreset.c:' 'original size' '1491,' 'current size' "$shar_count!"
+  fi
+fi
+rm -fr _sh01426
+exit 0
diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst
index 7c2d37571af0..e06238131f77 100644
--- a/Documentation/sound/hd-audio/models.rst
+++ b/Documentation/sound/hd-audio/models.rst
@@ -34,6 +34,22 @@ ALC262
 ======
 inv-dmic
     Inverted internal mic workaround
+fsc-h270
+    Fixups for Fujitsu-Siemens Celsius H270
+fsc-s7110
+    Fixups for Fujitsu-Siemens Lifebook S7110
+hp-z200
+    Fixups for HP Z200
+tyan
+    Fixups for Tyan Thunder n6650W
+lenovo-3000
+    Fixups for Lenovo 3000
+benq
+    Fixups for Benq ED8
+benq-t31
+    Fixups for Benq T31
+bayleybay
+    Fixups for Intel BayleyBay
 
 ALC267/268
 ==========
@@ -41,6 +57,8 @@ inv-dmic
     Inverted internal mic workaround
 hp-eapd
     Disable HP EAPD on NID 0x15
+spdif
+    Enable SPDIF output on NID 0x1e
 
 ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models)
 ===================================================================
@@ -70,6 +88,10 @@ dell-headset-multi
     Headset jack, which can also be used as mic-in
 dell-headset-dock
     Headset jack (without mic-in), and also dock I/O
+dell-headset3
+    Headset jack (without mic-in), and also dock I/O, variant 3
+dell-headset4
+    Headset jack (without mic-in), and also dock I/O, variant 4
 alc283-dac-wcaps
     Fixups for Chromebook with ALC283
 alc283-sense-combo
@@ -80,15 +102,173 @@ tpt440
     Lenovo Thinkpad T440s setup
 tpt460
     Lenovo Thinkpad T460/560 setup
+tpt470-dock
+    Lenovo Thinkpad T470 dock setup
 dual-codecs
     Lenovo laptops with dual codecs
 alc700-ref
     Intel reference board with ALC700 codec
+vaio
+    Pin fixups for Sony VAIO laptops
+dell-m101z
+    COEF setup for Dell M101z
+asus-g73jw
+    Subwoofer pin fixup for ASUS G73JW
+lenovo-eapd
+    Inversed EAPD setup for Lenovo laptops
+sony-hweq
+    H/W EQ COEF setup for Sony laptops
+pcm44k
+    Fixed PCM 44kHz constraints (for buggy devices)
+lifebook
+    Dock pin fixups for Fujitsu Lifebook
+lifebook-extmic
+    Headset mic fixup for Fujitsu Lifebook
+lifebook-hp-pin
+    Headphone pin fixup for Fujitsu Lifebook
+lifebook-u7x7
+    Lifebook U7x7 fixups
+alc269vb-amic
+    ALC269VB analog mic pin fixups
+alc269vb-dmic
+    ALC269VB digital mic pin fixups
+hp-mute-led-mic1
+    Mute LED via Mic1 pin on HP
+hp-mute-led-mic2
+    Mute LED via Mic2 pin on HP
+hp-mute-led-mic3
+    Mute LED via Mic3 pin on HP
+hp-gpio-mic1
+    GPIO + Mic1 pin LED on HP
+hp-line1-mic1
+    Mute LED via Line1 + Mic1 pins on HP
+noshutup
+    Skip shutup callback
+sony-nomic
+    Headset mic fixup for Sony laptops
+aspire-headset-mic
+    Headset pin fixup for Acer Aspire
+asus-x101
+    ASUS X101 fixups
+acer-ao7xx
+    Acer AO7xx fixups
+acer-aspire-e1
+    Acer Aspire E1 fixups
+acer-ac700
+    Acer AC700 fixups
+limit-mic-boost
+    Limit internal mic boost on Lenovo machines
+asus-zenbook
+    ASUS Zenbook fixups
+asus-zenbook-ux31a
+    ASUS Zenbook UX31A fixups
+ordissimo
+    Ordissimo EVE2 (or Malata PC-B1303) fixups
+asus-tx300
+    ASUS TX300 fixups
+alc283-int-mic
+    ALC283 COEF setup for Lenovo machines
+mono-speakers
+    Subwoofer and headset fixupes for Dell Inspiron
+alc290-subwoofer
+    Subwoofer fixups for Dell Vostro
+thinkpad
+    Binding with thinkpad_acpi driver for Lenovo machines
+dmic-thinkpad
+    thinkpad_acpi binding + digital mic support
+alc255-acer
+    ALC255 fixups on Acer machines
+alc255-asus
+    ALC255 fixups on ASUS machines
+alc255-dell1
+    ALC255 fixups on Dell machines
+alc255-dell2
+    ALC255 fixups on Dell machines, variant 2
+alc293-dell1
+    ALC293 fixups on Dell machines
+alc283-headset
+    Headset pin fixups on ALC283
+aspire-v5
+    Acer Aspire V5 fixups
+hp-gpio4
+    GPIO and Mic1 pin mute LED fixups for HP
+hp-gpio-led
+    GPIO mute LEDs on HP
+hp-gpio2-hotkey
+    GPIO mute LED with hot key handling on HP
+hp-dock-pins
+    GPIO mute LEDs and dock support on HP
+hp-dock-gpio-mic
+    GPIO, Mic mute LED and dock support on HP
+hp-9480m
+    HP 9480m fixups
+alc288-dell1
+    ALC288 fixups on Dell machines
+alc288-dell-xps13
+    ALC288 fixups on Dell XPS13
+dell-e7x
+    Dell E7x fixups
+alc293-dell
+    ALC293 fixups on Dell machines
+alc298-dell1
+    ALC298 fixups on Dell machines
+alc298-dell-aio
+    ALC298 fixups on Dell AIO machines
+alc275-dell-xps
+    ALC275 fixups on Dell XPS models
+alc256-dell-xps13
+    ALC256 fixups on Dell XPS13
+lenovo-spk-noise
+    Workaround for speaker noise on Lenovo machines
+lenovo-hotkey
+    Hot-key support via Mic2 pin on Lenovo machines
+dell-spk-noise
+    Workaround for speaker noise on Dell machines
+alc255-dell1
+    ALC255 fixups on Dell machines
+alc295-disable-dac3
+    Disable DAC3 routing on ALC295
+alc280-hp-headset
+    HP Elitebook fixups
+alc221-hp-mic
+    Front mic pin fixup on HP machines
+alc298-spk-volume
+    Speaker pin routing workaround on ALC298
+dell-inspiron-7559
+    Dell Inspiron 7559 fixups
+ativ-book
+    Samsung Ativ book 8 fixups
+alc221-hp-mic
+    ALC221 headset fixups on HP machines
+alc256-asus-mic
+    ALC256 fixups on ASUS machines
+alc256-asus-aio
+    ALC256 fixups on ASUS AIO machines
+alc233-eapd
+    ALC233 fixups on ASUS machines
+alc294-lenovo-mic
+    ALC294 Mic pin fixup for Lenovo AIO machines
+alc225-wyse
+    Dell Wyse fixups
+alc274-dell-aio
+    ALC274 fixups on Dell AIO machines
+alc255-dummy-lineout
+    Dell Precision 3930 fixups
+alc255-dell-headset"},
+    Dell Precision 3630 fixups
+alc295-hp-x360
+    HP Spectre X360 fixups
 
 ALC66x/67x/892
 ==============
+aspire
+    Subwoofer pin fixup for Aspire laptops
+ideapad
+    Subwoofer pin fixup for Ideapad laptops
 mario
     Chromebook mario model fixup
+hp-rp5800
+    Headphone pin fixup for HP RP5800
 asus-mode1
     ASUS
 asus-mode2
@@ -105,10 +285,40 @@ asus-mode7
     ASUS
 asus-mode8
     ASUS
+zotac-z68
+    Front HP fixup for Zotac Z68
 inv-dmic
     Inverted internal mic workaround
+alc662-headset-multi
+    Dell headset jack, which can also be used as mic-in (ALC662)
 dell-headset-multi
     Headset jack, which can also be used as mic-in
+alc662-headset
+    Headset mode support on ALC662
+alc668-headset
+    Headset mode support on ALC668
+bass16
+    Bass speaker fixup on pin 0x16
+bass1a
+    Bass speaker fixup on pin 0x1a
+automute
+    Auto-mute fixups for ALC668
+dell-xps13
+    Dell XPS13 fixups
+asus-nx50
+    ASUS Nx50 fixups
+asus-nx51
+    ASUS Nx51 fixups
+alc891-headset
+    Headset mode support on ALC891
+alc891-headset-multi
+    Dell headset jack, which can also be used as mic-in (ALC891)
+acer-veriton
+    Acer Veriton speaker pin fixup
+asrock-mobo
+    Fix invalid 0x15 / 0x16 pins
+usi-headset
+    Headset support on USI machines
 dual-codecs
     Lenovo laptops with dual codecs
 
@@ -116,20 +326,70 @@ ALC680
 ======
 N/A
 
-ALC88x/898/1150
-======================
+ALC88x/898/1150/1220
+====================
+abit-aw9d
+    Pin fixups for Abit AW9D-MAX
+lenovo-y530
+    Pin fixups for Lenovo Y530
+acer-aspire-7736
+    Fixup for Acer Aspire 7736
+asus-w90v
+    Pin fixup for ASUS W90V
+cd
+    Enable audio CD pin NID 0x1c
+no-front-hp
+    Disable front HP pin NID 0x1b
+vaio-tt
+    Pin fixup for VAIO TT
+eee1601
+    COEF setups for ASUS Eee 1601
+alc882-eapd
+    Change EAPD COEF mode on ALC882
+alc883-eapd
+    Change EAPD COEF mode on ALC883
+gpio1
+    Enable GPIO1
+gpio2
+    Enable GPIO2
+gpio3
+    Enable GPIO3
+alc889-coef
+    Setup ALC889 COEF
+asus-w2jc
+    Fixups for ASUS W2JC
 acer-aspire-4930g
     Acer Aspire 4930G/5930G/6530G/6930G/7730G
 acer-aspire-8930g
     Acer Aspire 8330G/6935G
 acer-aspire
     Acer Aspire others
+macpro-gpio
+    GPIO setup for Mac Pro
+dac-route
+    Workaround for DAC routing on Acer Aspire
+mbp-vref
+    Vref setup for Macbook Pro
+imac91-vref
+    Vref setup for iMac 9,1
+mba11-vref
+    Vref setup for MacBook Air 1,1
+mba21-vref
+    Vref setup for MacBook Air 2,1
+mp11-vref
+    Vref setup for Mac Pro 1,1
+mp41-vref
+    Vref setup for Mac Pro 4,1
 inv-dmic
     Inverted internal mic workaround
 no-primary-hp
     VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC)
+asus-bass
+    Bass speaker setup for ASUS ET2700
 dual-codecs
     ALC1220 dual codecs for Gaming mobos
+clevo-p950
+    Fixups for Clevo P950
 
 ALC861/660
 ==========
diff --git a/Documentation/sound/soc/dpcm.rst b/Documentation/sound/soc/dpcm.rst
index 395e5a516282..fe61e02277f8 100644
--- a/Documentation/sound/soc/dpcm.rst
+++ b/Documentation/sound/soc/dpcm.rst
@@ -254,9 +254,7 @@ configuration.
 	channels->min = channels->max = 2;
 
 	/* set DAI0 to 16 bit */
-	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
-				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
-				    SNDRV_PCM_FORMAT_S16_LE);
+	params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
 	return 0;
   }
 
diff --git a/MAINTAINERS b/MAINTAINERS
index 3d6e7e7b30d5..634039e7f083 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -13585,6 +13585,13 @@ L:	linux-block@vger.kernel.org
 S:	Maintained
 F:	drivers/block/skd*[ch]
 
+STI AUDIO (ASoC) DRIVERS
+M:	Arnaud Pouliquen <arnaud.pouliquen@st.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Maintained
+F:	Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt
+F:	sound/soc/sti/
+
 STI CEC DRIVER
 M:	Benjamin Gaignard <benjamin.gaignard@linaro.org>
 S:	Maintained
@@ -13598,6 +13605,14 @@ T:	git git://linuxtv.org/media_tree.git
 S:	Maintained
 F:	drivers/media/usb/stk1160/
 
+STM32 AUDIO (ASoC) DRIVERS
+M:	Olivier Moysan <olivier.moysan@st.com>
+M:	Arnaud Pouliquen <arnaud.pouliquen@st.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Maintained
+F:	Documentation/devicetree/bindings/sound/st,stm32-*.txt
+F:	sound/soc/stm/
+
 STM32 TIMER/LPTIMER DRIVERS
 M:	Fabrice Gasnier <fabrice.gasnier@st.com>
 S:	Maintained
diff --git a/arch/arm/mach-pxa/devices.c b/arch/arm/mach-pxa/devices.c
index d7c9a8476d57..5a16ea74e28a 100644
--- a/arch/arm/mach-pxa/devices.c
+++ b/arch/arm/mach-pxa/devices.c
@@ -4,6 +4,7 @@
 #include <linux/init.h>
 #include <linux/platform_device.h>
 #include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
 #include <linux/spi/pxa2xx_spi.h>
 #include <linux/platform_data/i2c-pxa.h>
 
@@ -59,16 +60,6 @@ static struct resource pxamci_resources[] = {
 		.end	= IRQ_MMC,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		.start	= 21,
-		.end	= 21,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		.start	= 22,
-		.end	= 22,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 static u64 pxamci_dmamask = 0xffffffffUL;
@@ -406,16 +397,6 @@ static struct resource pxa_ir_resources[] = {
 		.end	= 0x40700023,
 		.flags  = IORESOURCE_MEM,
 	},
-	[5] = {
-		.start  = 17,
-		.end	= 17,
-		.flags  = IORESOURCE_DMA,
-	},
-	[6] = {
-		.start  = 18,
-		.end	= 18,
-		.flags  = IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa_device_ficp = {
@@ -544,18 +525,6 @@ static struct resource pxa25x_resource_ssp[] = {
 		.end	= IRQ_SSP,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for RX */
-		.start	= 13,
-		.end	= 13,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for TX */
-		.start	= 14,
-		.end	= 14,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa25x_device_ssp = {
@@ -582,18 +551,6 @@ static struct resource pxa25x_resource_nssp[] = {
 		.end	= IRQ_NSSP,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for RX */
-		.start	= 15,
-		.end	= 15,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for TX */
-		.start	= 16,
-		.end	= 16,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa25x_device_nssp = {
@@ -620,18 +577,6 @@ static struct resource pxa25x_resource_assp[] = {
 		.end	= IRQ_ASSP,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for RX */
-		.start	= 23,
-		.end	= 23,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for TX */
-		.start	= 24,
-		.end	= 24,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa25x_device_assp = {
@@ -750,18 +695,6 @@ static struct resource pxa27x_resource_ssp1[] = {
 		.end	= IRQ_SSP,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for RX */
-		.start	= 13,
-		.end	= 13,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for TX */
-		.start	= 14,
-		.end	= 14,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa27x_device_ssp1 = {
@@ -788,18 +721,6 @@ static struct resource pxa27x_resource_ssp2[] = {
 		.end	= IRQ_SSP2,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for RX */
-		.start	= 15,
-		.end	= 15,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for TX */
-		.start	= 16,
-		.end	= 16,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa27x_device_ssp2 = {
@@ -826,18 +747,6 @@ static struct resource pxa27x_resource_ssp3[] = {
 		.end	= IRQ_SSP3,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for RX */
-		.start	= 66,
-		.end	= 66,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for TX */
-		.start	= 67,
-		.end	= 67,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa27x_device_ssp3 = {
@@ -894,16 +803,6 @@ static struct resource pxa3xx_resources_mci2[] = {
 		.end	= IRQ_MMC2,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		.start	= 93,
-		.end	= 93,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		.start	= 94,
-		.end	= 94,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa3xx_device_mci2 = {
@@ -933,16 +832,6 @@ static struct resource pxa3xx_resources_mci3[] = {
 		.end	= IRQ_MMC3,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		.start	= 100,
-		.end	= 100,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		.start	= 101,
-		.end	= 101,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa3xx_device_mci3 = {
@@ -1020,18 +909,6 @@ static struct resource pxa3xx_resources_nand[] = {
 		.end	= IRQ_NAND,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for Data DMA */
-		.start	= 97,
-		.end	= 97,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for Command DMA */
-		.start	= 99,
-		.end	= 99,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 static u64 pxa3xx_nand_dma_mask = DMA_BIT_MASK(32);
@@ -1065,18 +942,6 @@ static struct resource pxa3xx_resource_ssp4[] = {
 		.end	= IRQ_SSP4,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for RX */
-		.start	= 2,
-		.end	= 2,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for TX */
-		.start	= 3,
-		.end	= 3,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 /*
@@ -1202,11 +1067,6 @@ void __init pxa2xx_set_spi_info(unsigned id, struct pxa2xx_spi_master *info)
 	platform_device_add(pd);
 }
 
-static struct mmp_dma_platdata pxa_dma_pdata = {
-	.dma_channels	= 0,
-	.nb_requestors	= 0,
-};
-
 static struct resource pxa_dma_resource[] = {
 	[0] = {
 		.start	= 0x40000000,
@@ -1233,9 +1093,7 @@ static struct platform_device pxa2xx_pxa_dma = {
 	.resource	= pxa_dma_resource,
 };
 
-void __init pxa2xx_set_dmac_info(int nb_channels, int nb_requestors)
+void __init pxa2xx_set_dmac_info(struct mmp_dma_platdata *dma_pdata)
 {
-	pxa_dma_pdata.dma_channels = nb_channels;
-	pxa_dma_pdata.nb_requestors = nb_requestors;
-	pxa_register_device(&pxa2xx_pxa_dma, &pxa_dma_pdata);
+	pxa_register_device(&pxa2xx_pxa_dma, dma_pdata);
 }
diff --git a/arch/arm/mach-pxa/devices.h b/arch/arm/mach-pxa/devices.h
index 11263f7c455b..498b07bc6a3e 100644
--- a/arch/arm/mach-pxa/devices.h
+++ b/arch/arm/mach-pxa/devices.h
@@ -1,4 +1,8 @@
 /* SPDX-License-Identifier: GPL-2.0 */
+#define PDMA_FILTER_PARAM(_prio, _requestor) (&(struct pxad_param) { \
+	.prio = PXAD_PRIO_##_prio, .drcmr = _requestor })
+struct mmp_dma_platdata;
+
 extern struct platform_device pxa_device_pmu;
 extern struct platform_device pxa_device_mci;
 extern struct platform_device pxa3xx_device_mci2;
@@ -55,7 +59,7 @@ extern struct platform_device pxa3xx_device_gpio;
 extern struct platform_device pxa93x_device_gpio;
 
 void __init pxa_register_device(struct platform_device *dev, void *data);
-void __init pxa2xx_set_dmac_info(int nb_channels, int nb_requestors);
+void __init pxa2xx_set_dmac_info(struct mmp_dma_platdata *dma_pdata);
 
 struct i2c_pxa_platform_data;
 extern void pxa_set_i2c_info(struct i2c_pxa_platform_data *info);
diff --git a/arch/arm/mach-pxa/pxa25x.c b/arch/arm/mach-pxa/pxa25x.c
index ba431fad5c47..ab8808ce7e21 100644
--- a/arch/arm/mach-pxa/pxa25x.c
+++ b/arch/arm/mach-pxa/pxa25x.c
@@ -16,6 +16,8 @@
  * initialization stuff for PXA machines which can be overridden later if
  * need be.
  */
+#include <linux/dmaengine.h>
+#include <linux/dma/pxa-dma.h>
 #include <linux/gpio.h>
 #include <linux/gpio-pxa.h>
 #include <linux/module.h>
@@ -26,6 +28,7 @@
 #include <linux/syscore_ops.h>
 #include <linux/irq.h>
 #include <linux/irqchip.h>
+#include <linux/platform_data/mmp_dma.h>
 
 #include <asm/mach/map.h>
 #include <asm/suspend.h>
@@ -201,6 +204,39 @@ static struct platform_device *pxa25x_devices[] __initdata = {
 	&pxa_device_asoc_platform,
 };
 
+static const struct dma_slave_map pxa25x_slave_map[] = {
+	/* PXA25x, PXA27x and PXA3xx common entries */
+	{ "pxa2xx-ac97", "pcm_pcm_mic_mono", PDMA_FILTER_PARAM(LOWEST, 8) },
+	{ "pxa2xx-ac97", "pcm_pcm_aux_mono_in", PDMA_FILTER_PARAM(LOWEST, 9) },
+	{ "pxa2xx-ac97", "pcm_pcm_aux_mono_out",
+	  PDMA_FILTER_PARAM(LOWEST, 10) },
+	{ "pxa2xx-ac97", "pcm_pcm_stereo_in", PDMA_FILTER_PARAM(LOWEST, 11) },
+	{ "pxa2xx-ac97", "pcm_pcm_stereo_out", PDMA_FILTER_PARAM(LOWEST, 12) },
+	{ "pxa-ssp-dai.1", "rx", PDMA_FILTER_PARAM(LOWEST, 13) },
+	{ "pxa-ssp-dai.1", "tx", PDMA_FILTER_PARAM(LOWEST, 14) },
+	{ "pxa-ssp-dai.2", "rx", PDMA_FILTER_PARAM(LOWEST, 15) },
+	{ "pxa-ssp-dai.2", "tx", PDMA_FILTER_PARAM(LOWEST, 16) },
+	{ "pxa2xx-ir", "rx", PDMA_FILTER_PARAM(LOWEST, 17) },
+	{ "pxa2xx-ir", "tx", PDMA_FILTER_PARAM(LOWEST, 18) },
+	{ "pxa2xx-mci.0", "rx", PDMA_FILTER_PARAM(LOWEST, 21) },
+	{ "pxa2xx-mci.0", "tx", PDMA_FILTER_PARAM(LOWEST, 22) },
+
+	/* PXA25x specific map */
+	{ "pxa25x-ssp.0", "rx", PDMA_FILTER_PARAM(LOWEST, 13) },
+	{ "pxa25x-ssp.0", "tx", PDMA_FILTER_PARAM(LOWEST, 14) },
+	{ "pxa25x-nssp.1", "rx", PDMA_FILTER_PARAM(LOWEST, 15) },
+	{ "pxa25x-nssp.1", "tx", PDMA_FILTER_PARAM(LOWEST, 16) },
+	{ "pxa25x-nssp.2", "rx", PDMA_FILTER_PARAM(LOWEST, 23) },
+	{ "pxa25x-nssp.2", "tx", PDMA_FILTER_PARAM(LOWEST, 24) },
+};
+
+static struct mmp_dma_platdata pxa25x_dma_pdata = {
+	.dma_channels	= 16,
+	.nb_requestors	= 40,
+	.slave_map	= pxa25x_slave_map,
+	.slave_map_cnt	= ARRAY_SIZE(pxa25x_slave_map),
+};
+
 static int __init pxa25x_init(void)
 {
 	int ret = 0;
@@ -215,7 +251,7 @@ static int __init pxa25x_init(void)
 		register_syscore_ops(&pxa2xx_mfp_syscore_ops);
 
 		if (!of_have_populated_dt()) {
-			pxa2xx_set_dmac_info(16, 40);
+			pxa2xx_set_dmac_info(&pxa25x_dma_pdata);
 			pxa_register_device(&pxa25x_device_gpio, &pxa25x_gpio_info);
 			ret = platform_add_devices(pxa25x_devices,
 						   ARRAY_SIZE(pxa25x_devices));
diff --git a/arch/arm/mach-pxa/pxa27x.c b/arch/arm/mach-pxa/pxa27x.c
index 0c06f383ad52..5a8990a9313d 100644
--- a/arch/arm/mach-pxa/pxa27x.c
+++ b/arch/arm/mach-pxa/pxa27x.c
@@ -11,6 +11,8 @@
  * it under the terms of the GNU General Public License version 2 as
  * published by the Free Software Foundation.
  */
+#include <linux/dmaengine.h>
+#include <linux/dma/pxa-dma.h>
 #include <linux/gpio.h>
 #include <linux/gpio-pxa.h>
 #include <linux/module.h>
@@ -23,6 +25,7 @@
 #include <linux/io.h>
 #include <linux/irq.h>
 #include <linux/platform_data/i2c-pxa.h>
+#include <linux/platform_data/mmp_dma.h>
 
 #include <asm/mach/map.h>
 #include <mach/hardware.h>
@@ -297,6 +300,40 @@ static struct platform_device *devices[] __initdata = {
 	&pxa27x_device_pwm1,
 };
 
+static const struct dma_slave_map pxa27x_slave_map[] = {
+	/* PXA25x, PXA27x and PXA3xx common entries */
+	{ "pxa2xx-ac97", "pcm_pcm_mic_mono", PDMA_FILTER_PARAM(LOWEST, 8) },
+	{ "pxa2xx-ac97", "pcm_pcm_aux_mono_in", PDMA_FILTER_PARAM(LOWEST, 9) },
+	{ "pxa2xx-ac97", "pcm_pcm_aux_mono_out",
+	  PDMA_FILTER_PARAM(LOWEST, 10) },
+	{ "pxa2xx-ac97", "pcm_pcm_stereo_in", PDMA_FILTER_PARAM(LOWEST, 11) },
+	{ "pxa2xx-ac97", "pcm_pcm_stereo_out", PDMA_FILTER_PARAM(LOWEST, 12) },
+	{ "pxa-ssp-dai.0", "rx", PDMA_FILTER_PARAM(LOWEST, 13) },
+	{ "pxa-ssp-dai.0", "tx", PDMA_FILTER_PARAM(LOWEST, 14) },
+	{ "pxa-ssp-dai.1", "rx", PDMA_FILTER_PARAM(LOWEST, 15) },
+	{ "pxa-ssp-dai.1", "tx", PDMA_FILTER_PARAM(LOWEST, 16) },
+	{ "pxa2xx-ir", "rx", PDMA_FILTER_PARAM(LOWEST, 17) },
+	{ "pxa2xx-ir", "tx", PDMA_FILTER_PARAM(LOWEST, 18) },
+	{ "pxa2xx-mci.0", "rx", PDMA_FILTER_PARAM(LOWEST, 21) },
+	{ "pxa2xx-mci.0", "tx", PDMA_FILTER_PARAM(LOWEST, 22) },
+	{ "pxa-ssp-dai.2", "rx", PDMA_FILTER_PARAM(LOWEST, 66) },
+	{ "pxa-ssp-dai.2", "tx", PDMA_FILTER_PARAM(LOWEST, 67) },
+
+	/* PXA27x specific map */
+	{ "pxa2xx-i2s", "rx", PDMA_FILTER_PARAM(LOWEST, 2) },
+	{ "pxa2xx-i2s", "tx", PDMA_FILTER_PARAM(LOWEST, 3) },
+	{ "pxa27x-camera.0", "CI_Y", PDMA_FILTER_PARAM(HIGHEST, 68) },
+	{ "pxa27x-camera.0", "CI_U", PDMA_FILTER_PARAM(HIGHEST, 69) },
+	{ "pxa27x-camera.0", "CI_V", PDMA_FILTER_PARAM(HIGHEST, 70) },
+};
+
+static struct mmp_dma_platdata pxa27x_dma_pdata = {
+	.dma_channels	= 32,
+	.nb_requestors	= 75,
+	.slave_map	= pxa27x_slave_map,
+	.slave_map_cnt	= ARRAY_SIZE(pxa27x_slave_map),
+};
+
 static int __init pxa27x_init(void)
 {
 	int ret = 0;
@@ -313,7 +350,7 @@ static int __init pxa27x_init(void)
 		if (!of_have_populated_dt()) {
 			pxa_register_device(&pxa27x_device_gpio,
 					    &pxa27x_gpio_info);
-			pxa2xx_set_dmac_info(32, 75);
+			pxa2xx_set_dmac_info(&pxa27x_dma_pdata);
 			ret = platform_add_devices(devices,
 						   ARRAY_SIZE(devices));
 		}
diff --git a/arch/arm/mach-pxa/pxa3xx.c b/arch/arm/mach-pxa/pxa3xx.c
index 8c64f93b669b..df9c8970adcf 100644
--- a/arch/arm/mach-pxa/pxa3xx.c
+++ b/arch/arm/mach-pxa/pxa3xx.c
@@ -12,6 +12,8 @@
  * it under the terms of the GNU General Public License version 2 as
  * published by the Free Software Foundation.
  */
+#include <linux/dmaengine.h>
+#include <linux/dma/pxa-dma.h>
 #include <linux/module.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
@@ -24,6 +26,7 @@
 #include <linux/of.h>
 #include <linux/syscore_ops.h>
 #include <linux/platform_data/i2c-pxa.h>
+#include <linux/platform_data/mmp_dma.h>
 
 #include <asm/mach/map.h>
 #include <asm/suspend.h>
@@ -421,6 +424,42 @@ static struct platform_device *devices[] __initdata = {
 	&pxa27x_device_pwm1,
 };
 
+static const struct dma_slave_map pxa3xx_slave_map[] = {
+	/* PXA25x, PXA27x and PXA3xx common entries */
+	{ "pxa2xx-ac97", "pcm_pcm_mic_mono", PDMA_FILTER_PARAM(LOWEST, 8) },
+	{ "pxa2xx-ac97", "pcm_pcm_aux_mono_in", PDMA_FILTER_PARAM(LOWEST, 9) },
+	{ "pxa2xx-ac97", "pcm_pcm_aux_mono_out",
+	  PDMA_FILTER_PARAM(LOWEST, 10) },
+	{ "pxa2xx-ac97", "pcm_pcm_stereo_in", PDMA_FILTER_PARAM(LOWEST, 11) },
+	{ "pxa2xx-ac97", "pcm_pcm_stereo_out", PDMA_FILTER_PARAM(LOWEST, 12) },
+	{ "pxa-ssp-dai.0", "rx", PDMA_FILTER_PARAM(LOWEST, 13) },
+	{ "pxa-ssp-dai.0", "tx", PDMA_FILTER_PARAM(LOWEST, 14) },
+	{ "pxa-ssp-dai.1", "rx", PDMA_FILTER_PARAM(LOWEST, 15) },
+	{ "pxa-ssp-dai.1", "tx", PDMA_FILTER_PARAM(LOWEST, 16) },
+	{ "pxa2xx-ir", "rx", PDMA_FILTER_PARAM(LOWEST, 17) },
+	{ "pxa2xx-ir", "tx", PDMA_FILTER_PARAM(LOWEST, 18) },
+	{ "pxa2xx-mci.0", "rx", PDMA_FILTER_PARAM(LOWEST, 21) },
+	{ "pxa2xx-mci.0", "tx", PDMA_FILTER_PARAM(LOWEST, 22) },
+	{ "pxa-ssp-dai.2", "rx", PDMA_FILTER_PARAM(LOWEST, 66) },
+	{ "pxa-ssp-dai.2", "tx", PDMA_FILTER_PARAM(LOWEST, 67) },
+
+	/* PXA3xx specific map */
+	{ "pxa-ssp-dai.3", "rx", PDMA_FILTER_PARAM(LOWEST, 2) },
+	{ "pxa-ssp-dai.3", "tx", PDMA_FILTER_PARAM(LOWEST, 3) },
+	{ "pxa2xx-mci.1", "rx", PDMA_FILTER_PARAM(LOWEST, 93) },
+	{ "pxa2xx-mci.1", "tx", PDMA_FILTER_PARAM(LOWEST, 94) },
+	{ "pxa3xx-nand", "data", PDMA_FILTER_PARAM(LOWEST, 97) },
+	{ "pxa2xx-mci.2", "rx", PDMA_FILTER_PARAM(LOWEST, 100) },
+	{ "pxa2xx-mci.2", "tx", PDMA_FILTER_PARAM(LOWEST, 101) },
+};
+
+static struct mmp_dma_platdata pxa3xx_dma_pdata = {
+	.dma_channels	= 32,
+	.nb_requestors	= 100,
+	.slave_map	= pxa3xx_slave_map,
+	.slave_map_cnt	= ARRAY_SIZE(pxa3xx_slave_map),
+};
+
 static int __init pxa3xx_init(void)
 {
 	int ret = 0;
@@ -456,7 +495,7 @@ static int __init pxa3xx_init(void)
 		if (of_have_populated_dt())
 			return 0;
 
-		pxa2xx_set_dmac_info(32, 100);
+		pxa2xx_set_dmac_info(&pxa3xx_dma_pdata);
 		ret = platform_add_devices(devices, ARRAY_SIZE(devices));
 		if (ret)
 			return ret;
diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c
index ba13f793fbce..ed36dcab80f1 100644
--- a/arch/arm/plat-pxa/ssp.c
+++ b/arch/arm/plat-pxa/ssp.c
@@ -127,53 +127,6 @@ static int pxa_ssp_probe(struct platform_device *pdev)
 	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;
-		int ret;
-
-		/*
-		 * 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 */
-		ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells",
-						 0, &dma_spec);
-
-		if (ret) {
-			dev_err(dev, "Can't parse dmas property\n");
-			return -ENODEV;
-		}
-		ssp->drcmr_rx = dma_spec.args[0];
-		of_node_put(dma_spec.np);
-
-		/* tx */
-		ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells",
-						 1, &dma_spec);
-		if (ret) {
-			dev_err(dev, "Can't parse dmas property\n");
-			return -ENODEV;
-		}
-		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(dev, "no SSP TX DRCMR defined\n");
-			return -ENODEV;
-		}
-		ssp->drcmr_tx = res->start;
-	}
-
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (res == NULL) {
 		dev_err(dev, "no memory resource defined\n");
diff --git a/drivers/ata/pata_pxa.c b/drivers/ata/pata_pxa.c
index f6c46e9a4dc0..e8b6a2e464c9 100644
--- a/drivers/ata/pata_pxa.c
+++ b/drivers/ata/pata_pxa.c
@@ -25,7 +25,6 @@
 #include <linux/libata.h>
 #include <linux/platform_device.h>
 #include <linux/dmaengine.h>
-#include <linux/dma/pxa-dma.h>
 #include <linux/gpio.h>
 #include <linux/slab.h>
 #include <linux/completion.h>
@@ -180,8 +179,6 @@ static int pxa_ata_probe(struct platform_device *pdev)
 	struct resource *irq_res;
 	struct pata_pxa_pdata *pdata = dev_get_platdata(&pdev->dev);
 	struct dma_slave_config	config;
-	dma_cap_mask_t mask;
-	struct pxad_param param;
 	int ret = 0;
 
 	/*
@@ -278,10 +275,6 @@ static int pxa_ata_probe(struct platform_device *pdev)
 
 	ap->private_data = data;
 
-	dma_cap_zero(mask);
-	dma_cap_set(DMA_SLAVE, mask);
-	param.prio = PXAD_PRIO_LOWEST;
-	param.drcmr = pdata->dma_dreq;
 	memset(&config, 0, sizeof(config));
 	config.src_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
 	config.dst_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
@@ -294,8 +287,7 @@ static int pxa_ata_probe(struct platform_device *pdev)
 	 * Request the DMA channel
 	 */
 	data->dma_chan =
-		dma_request_slave_channel_compat(mask, pxad_filter_fn,
-						 &param, &pdev->dev, "data");
+		dma_request_slave_channel(&pdev->dev, "data");
 	if (!data->dma_chan)
 		return -EBUSY;
 	ret = dmaengine_slave_config(data->dma_chan, &config);
diff --git a/drivers/clk/clk.c b/drivers/clk/clk.c
index e2ed078abd90..976f59e11f9a 100644
--- a/drivers/clk/clk.c
+++ b/drivers/clk/clk.c
@@ -67,6 +67,7 @@ struct clk_core {
 	unsigned long		max_rate;
 	unsigned long		accuracy;
 	int			phase;
+	struct clk_duty		duty;
 	struct hlist_head	children;
 	struct hlist_node	child_node;
 	struct hlist_head	clks;
@@ -2401,6 +2402,172 @@ int clk_get_phase(struct clk *clk)
 }
 EXPORT_SYMBOL_GPL(clk_get_phase);
 
+static void clk_core_reset_duty_cycle_nolock(struct clk_core *core)
+{
+	/* Assume a default value of 50% */
+	core->duty.num = 1;
+	core->duty.den = 2;
+}
+
+static int clk_core_update_duty_cycle_parent_nolock(struct clk_core *core);
+
+static int clk_core_update_duty_cycle_nolock(struct clk_core *core)
+{
+	struct clk_duty *duty = &core->duty;
+	int ret = 0;
+
+	if (!core->ops->get_duty_cycle)
+		return clk_core_update_duty_cycle_parent_nolock(core);
+
+	ret = core->ops->get_duty_cycle(core->hw, duty);
+	if (ret)
+		goto reset;
+
+	/* Don't trust the clock provider too much */
+	if (duty->den == 0 || duty->num > duty->den) {
+		ret = -EINVAL;
+		goto reset;
+	}
+
+	return 0;
+
+reset:
+	clk_core_reset_duty_cycle_nolock(core);
+	return ret;
+}
+
+static int clk_core_update_duty_cycle_parent_nolock(struct clk_core *core)
+{
+	int ret = 0;
+
+	if (core->parent &&
+	    core->flags & CLK_DUTY_CYCLE_PARENT) {
+		ret = clk_core_update_duty_cycle_nolock(core->parent);
+		memcpy(&core->duty, &core->parent->duty, sizeof(core->duty));
+	} else {
+		clk_core_reset_duty_cycle_nolock(core);
+	}
+
+	return ret;
+}
+
+static int clk_core_set_duty_cycle_parent_nolock(struct clk_core *core,
+						 struct clk_duty *duty);
+
+static int clk_core_set_duty_cycle_nolock(struct clk_core *core,
+					  struct clk_duty *duty)
+{
+	int ret;
+
+	lockdep_assert_held(&prepare_lock);
+
+	if (clk_core_rate_is_protected(core))
+		return -EBUSY;
+
+	trace_clk_set_duty_cycle(core, duty);
+
+	if (!core->ops->set_duty_cycle)
+		return clk_core_set_duty_cycle_parent_nolock(core, duty);
+
+	ret = core->ops->set_duty_cycle(core->hw, duty);
+	if (!ret)
+		memcpy(&core->duty, duty, sizeof(*duty));
+
+	trace_clk_set_duty_cycle_complete(core, duty);
+
+	return ret;
+}
+
+static int clk_core_set_duty_cycle_parent_nolock(struct clk_core *core,
+						 struct clk_duty *duty)
+{
+	int ret = 0;
+
+	if (core->parent &&
+	    core->flags & (CLK_DUTY_CYCLE_PARENT | CLK_SET_RATE_PARENT)) {
+		ret = clk_core_set_duty_cycle_nolock(core->parent, duty);
+		memcpy(&core->duty, &core->parent->duty, sizeof(core->duty));
+	}
+
+	return ret;
+}
+
+/**
+ * clk_set_duty_cycle - adjust the duty cycle ratio of a clock signal
+ * @clk: clock signal source
+ * @num: numerator of the duty cycle ratio to be applied
+ * @den: denominator of the duty cycle ratio to be applied
+ *
+ * Apply the duty cycle ratio if the ratio is valid and the clock can
+ * perform this operation
+ *
+ * Returns (0) on success, a negative errno otherwise.
+ */
+int clk_set_duty_cycle(struct clk *clk, unsigned int num, unsigned int den)
+{
+	int ret;
+	struct clk_duty duty;
+
+	if (!clk)
+		return 0;
+
+	/* sanity check the ratio */
+	if (den == 0 || num > den)
+		return -EINVAL;
+
+	duty.num = num;
+	duty.den = den;
+
+	clk_prepare_lock();
+
+	if (clk->exclusive_count)
+		clk_core_rate_unprotect(clk->core);
+
+	ret = clk_core_set_duty_cycle_nolock(clk->core, &duty);
+
+	if (clk->exclusive_count)
+		clk_core_rate_protect(clk->core);
+
+	clk_prepare_unlock();
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(clk_set_duty_cycle);
+
+static int clk_core_get_scaled_duty_cycle(struct clk_core *core,
+					  unsigned int scale)
+{
+	struct clk_duty *duty = &core->duty;
+	int ret;
+
+	clk_prepare_lock();
+
+	ret = clk_core_update_duty_cycle_nolock(core);
+	if (!ret)
+		ret = mult_frac(scale, duty->num, duty->den);
+
+	clk_prepare_unlock();
+
+	return ret;
+}
+
+/**
+ * clk_get_scaled_duty_cycle - return the duty cycle ratio of a clock signal
+ * @clk: clock signal source
+ * @scale: scaling factor to be applied to represent the ratio as an integer
+ *
+ * Returns the duty cycle ratio of a clock node multiplied by the provided
+ * scaling factor, or negative errno on error.
+ */
+int clk_get_scaled_duty_cycle(struct clk *clk, unsigned int scale)
+{
+	if (!clk)
+		return 0;
+
+	return clk_core_get_scaled_duty_cycle(clk->core, scale);
+}
+EXPORT_SYMBOL_GPL(clk_get_scaled_duty_cycle);
+
 /**
  * clk_is_match - check if two clk's point to the same hardware clock
  * @p: clk compared against q
@@ -2454,12 +2621,13 @@ static void clk_summary_show_one(struct seq_file *s, struct clk_core *c,
 	if (!c)
 		return;
 
-	seq_printf(s, "%*s%-*s %7d %8d %8d %11lu %10lu %-3d\n",
+	seq_printf(s, "%*s%-*s %7d %8d %8d %11lu %10lu %5d %6d\n",
 		   level * 3 + 1, "",
 		   30 - level * 3, c->name,
 		   c->enable_count, c->prepare_count, c->protect_count,
 		   clk_core_get_rate(c), clk_core_get_accuracy(c),
-		   clk_core_get_phase(c));
+		   clk_core_get_phase(c),
+		   clk_core_get_scaled_duty_cycle(c, 100000));
 }
 
 static void clk_summary_show_subtree(struct seq_file *s, struct clk_core *c,
@@ -2481,9 +2649,9 @@ static int clk_summary_show(struct seq_file *s, void *data)
 	struct clk_core *c;
 	struct hlist_head **lists = (struct hlist_head **)s->private;
 
-	seq_puts(s, "                                 enable  prepare  protect                               \n");
-	seq_puts(s, "   clock                          count    count    count        rate   accuracy   phase\n");
-	seq_puts(s, "----------------------------------------------------------------------------------------\n");
+	seq_puts(s, "                                 enable  prepare  protect                                duty\n");
+	seq_puts(s, "   clock                          count    count    count        rate   accuracy phase  cycle\n");
+	seq_puts(s, "---------------------------------------------------------------------------------------------\n");
 
 	clk_prepare_lock();
 
@@ -2510,6 +2678,8 @@ static void clk_dump_one(struct seq_file *s, struct clk_core *c, int level)
 	seq_printf(s, "\"rate\": %lu,", clk_core_get_rate(c));
 	seq_printf(s, "\"accuracy\": %lu,", clk_core_get_accuracy(c));
 	seq_printf(s, "\"phase\": %d", clk_core_get_phase(c));
+	seq_printf(s, "\"duty_cycle\": %u",
+		   clk_core_get_scaled_duty_cycle(c, 100000));
 }
 
 static void clk_dump_subtree(struct seq_file *s, struct clk_core *c, int level)
@@ -2571,6 +2741,7 @@ static const struct {
 	ENTRY(CLK_SET_RATE_UNGATE),
 	ENTRY(CLK_IS_CRITICAL),
 	ENTRY(CLK_OPS_PARENT_ENABLE),
+	ENTRY(CLK_DUTY_CYCLE_PARENT),
 #undef ENTRY
 };
 
@@ -2609,6 +2780,17 @@ static int possible_parents_show(struct seq_file *s, void *data)
 }
 DEFINE_SHOW_ATTRIBUTE(possible_parents);
 
+static int clk_duty_cycle_show(struct seq_file *s, void *data)
+{
+	struct clk_core *core = s->private;
+	struct clk_duty *duty = &core->duty;
+
+	seq_printf(s, "%u/%u\n", duty->num, duty->den);
+
+	return 0;
+}
+DEFINE_SHOW_ATTRIBUTE(clk_duty_cycle);
+
 static void clk_debug_create_one(struct clk_core *core, struct dentry *pdentry)
 {
 	struct dentry *root;
@@ -2627,6 +2809,8 @@ static void clk_debug_create_one(struct clk_core *core, struct dentry *pdentry)
 	debugfs_create_u32("clk_enable_count", 0444, root, &core->enable_count);
 	debugfs_create_u32("clk_protect_count", 0444, root, &core->protect_count);
 	debugfs_create_u32("clk_notifier_count", 0444, root, &core->notifier_count);
+	debugfs_create_file("clk_duty_cycle", 0444, root, core,
+			    &clk_duty_cycle_fops);
 
 	if (core->num_parents > 1)
 		debugfs_create_file("clk_possible_parents", 0444, root, core,
@@ -2845,6 +3029,11 @@ static int __clk_core_init(struct clk_core *core)
 		core->phase = 0;
 
 	/*
+	 * Set clk's duty cycle.
+	 */
+	clk_core_update_duty_cycle_nolock(core);
+
+	/*
 	 * Set clk's rate.  The preferred method is to use .recalc_rate.  For
 	 * simple clocks and lazy developers the default fallback is to use the
 	 * parent's rate.  If a clock doesn't have a parent (or is orphaned)
diff --git a/drivers/dma/pxa_dma.c b/drivers/dma/pxa_dma.c
index b53fb618bbf6..b31c28b67ad3 100644
--- a/drivers/dma/pxa_dma.c
+++ b/drivers/dma/pxa_dma.c
@@ -179,6 +179,8 @@ static unsigned int pxad_drcmr(unsigned int line)
 	return 0x1000 + line * 4;
 }
 
+bool pxad_filter_fn(struct dma_chan *chan, void *param);
+
 /*
  * Debug fs
  */
@@ -760,6 +762,8 @@ static void pxad_free_chan_resources(struct dma_chan *dchan)
 	dma_pool_destroy(chan->desc_pool);
 	chan->desc_pool = NULL;
 
+	chan->drcmr = U32_MAX;
+	chan->prio = PXAD_PRIO_LOWEST;
 }
 
 static void pxad_free_desc(struct virt_dma_desc *vd)
@@ -1384,6 +1388,9 @@ static int pxad_init_dmadev(struct platform_device *op,
 		c = devm_kzalloc(&op->dev, sizeof(*c), GFP_KERNEL);
 		if (!c)
 			return -ENOMEM;
+
+		c->drcmr = U32_MAX;
+		c->prio = PXAD_PRIO_LOWEST;
 		c->vc.desc_free = pxad_free_desc;
 		vchan_init(&c->vc, &pdev->slave);
 		init_waitqueue_head(&c->wq_state);
@@ -1396,9 +1403,10 @@ static int pxad_probe(struct platform_device *op)
 {
 	struct pxad_device *pdev;
 	const struct of_device_id *of_id;
+	const struct dma_slave_map *slave_map = NULL;
 	struct mmp_dma_platdata *pdata = dev_get_platdata(&op->dev);
 	struct resource *iores;
-	int ret, dma_channels = 0, nb_requestors = 0;
+	int ret, dma_channels = 0, nb_requestors = 0, slave_map_cnt = 0;
 	const enum dma_slave_buswidth widths =
 		DMA_SLAVE_BUSWIDTH_1_BYTE   | DMA_SLAVE_BUSWIDTH_2_BYTES |
 		DMA_SLAVE_BUSWIDTH_4_BYTES;
@@ -1429,6 +1437,8 @@ static int pxad_probe(struct platform_device *op)
 	} else if (pdata && pdata->dma_channels) {
 		dma_channels = pdata->dma_channels;
 		nb_requestors = pdata->nb_requestors;
+		slave_map = pdata->slave_map;
+		slave_map_cnt = pdata->slave_map_cnt;
 	} else {
 		dma_channels = 32;	/* default 32 channel */
 	}
@@ -1440,6 +1450,9 @@ static int pxad_probe(struct platform_device *op)
 	pdev->slave.device_prep_dma_memcpy = pxad_prep_memcpy;
 	pdev->slave.device_prep_slave_sg = pxad_prep_slave_sg;
 	pdev->slave.device_prep_dma_cyclic = pxad_prep_dma_cyclic;
+	pdev->slave.filter.map = slave_map;
+	pdev->slave.filter.mapcnt = slave_map_cnt;
+	pdev->slave.filter.fn = pxad_filter_fn;
 
 	pdev->slave.copy_align = PDMA_ALIGNMENT;
 	pdev->slave.src_addr_widths = widths;
diff --git a/drivers/gpu/drm/i915/Kconfig b/drivers/gpu/drm/i915/Kconfig
index dfd95889f4b7..5c607f2c707b 100644
--- a/drivers/gpu/drm/i915/Kconfig
+++ b/drivers/gpu/drm/i915/Kconfig
@@ -23,6 +23,7 @@ config DRM_I915
 	select SYNC_FILE
 	select IOSF_MBI
 	select CRC32
+	select SND_HDA_I915 if SND_HDA_CORE
 	help
 	  Choose this option if you have a system that has "Intel Graphics
 	  Media Accelerator" or "HD Graphics" integrated graphics,
diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c
index 3ea566f99450..7dd5605d94ae 100644
--- a/drivers/gpu/drm/i915/intel_audio.c
+++ b/drivers/gpu/drm/i915/intel_audio.c
@@ -639,11 +639,12 @@ void intel_audio_codec_enable(struct intel_encoder *encoder,
 	dev_priv->av_enc_map[pipe] = encoder;
 	mutex_unlock(&dev_priv->av_mutex);
 
-	if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) {
+	if (acomp && acomp->base.audio_ops &&
+	    acomp->base.audio_ops->pin_eld_notify) {
 		/* audio drivers expect pipe = -1 to indicate Non-MST cases */
 		if (!intel_crtc_has_type(crtc_state, INTEL_OUTPUT_DP_MST))
 			pipe = -1;
-		acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr,
+		acomp->base.audio_ops->pin_eld_notify(acomp->base.audio_ops->audio_ptr,
 						 (int) port, (int) pipe);
 	}
 
@@ -681,11 +682,12 @@ void intel_audio_codec_disable(struct intel_encoder *encoder,
 	dev_priv->av_enc_map[pipe] = NULL;
 	mutex_unlock(&dev_priv->av_mutex);
 
-	if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) {
+	if (acomp && acomp->base.audio_ops &&
+	    acomp->base.audio_ops->pin_eld_notify) {
 		/* audio drivers expect pipe = -1 to indicate Non-MST cases */
 		if (!intel_crtc_has_type(old_crtc_state, INTEL_OUTPUT_DP_MST))
 			pipe = -1;
-		acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr,
+		acomp->base.audio_ops->pin_eld_notify(acomp->base.audio_ops->audio_ptr,
 						 (int) port, (int) pipe);
 	}
 
@@ -880,7 +882,7 @@ static int i915_audio_component_get_eld(struct device *kdev, int port,
 	return ret;
 }
 
-static const struct i915_audio_component_ops i915_audio_component_ops = {
+static const struct drm_audio_component_ops i915_audio_component_ops = {
 	.owner		= THIS_MODULE,
 	.get_power	= i915_audio_component_get_power,
 	.put_power	= i915_audio_component_put_power,
@@ -897,12 +899,12 @@ static int i915_audio_component_bind(struct device *i915_kdev,
 	struct drm_i915_private *dev_priv = kdev_to_i915(i915_kdev);
 	int i;
 
-	if (WARN_ON(acomp->ops || acomp->dev))
+	if (WARN_ON(acomp->base.ops || acomp->base.dev))
 		return -EEXIST;
 
 	drm_modeset_lock_all(&dev_priv->drm);
-	acomp->ops = &i915_audio_component_ops;
-	acomp->dev = i915_kdev;
+	acomp->base.ops = &i915_audio_component_ops;
+	acomp->base.dev = i915_kdev;
 	BUILD_BUG_ON(MAX_PORTS != I915_MAX_PORTS);
 	for (i = 0; i < ARRAY_SIZE(acomp->aud_sample_rate); i++)
 		acomp->aud_sample_rate[i] = 0;
@@ -919,8 +921,8 @@ static void i915_audio_component_unbind(struct device *i915_kdev,
 	struct drm_i915_private *dev_priv = kdev_to_i915(i915_kdev);
 
 	drm_modeset_lock_all(&dev_priv->drm);
-	acomp->ops = NULL;
-	acomp->dev = NULL;
+	acomp->base.ops = NULL;
+	acomp->base.dev = NULL;
 	dev_priv->audio_component = NULL;
 	drm_modeset_unlock_all(&dev_priv->drm);
 }
diff --git a/drivers/gpu/vga/vga_switcheroo.c b/drivers/gpu/vga/vga_switcheroo.c
index fc4adf3d34e8..a96bf46bc483 100644
--- a/drivers/gpu/vga/vga_switcheroo.c
+++ b/drivers/gpu/vga/vga_switcheroo.c
@@ -103,9 +103,11 @@
  *	runtime pm. If true, writing ON and OFF to the vga_switcheroo debugfs
  *	interface is a no-op so as not to interfere with runtime pm
  * @list: client list
+ * @vga_dev: pci device, indicate which GPU is bound to current audio client
  *
  * Registered client. A client can be either a GPU or an audio device on a GPU.
- * For audio clients, the @fb_info and @active members are bogus.
+ * For audio clients, the @fb_info and @active members are bogus. For GPU
+ * clients, the @vga_dev is bogus.
  */
 struct vga_switcheroo_client {
 	struct pci_dev *pdev;
@@ -116,6 +118,7 @@ struct vga_switcheroo_client {
 	bool active;
 	bool driver_power_control;
 	struct list_head list;
+	struct pci_dev *vga_dev;
 };
 
 /*
@@ -161,9 +164,8 @@ struct vgasr_priv {
 };
 
 #define ID_BIT_AUDIO		0x100
-#define client_is_audio(c)	((c)->id & ID_BIT_AUDIO)
-#define client_is_vga(c)	((c)->id == VGA_SWITCHEROO_UNKNOWN_ID || \
-				 !client_is_audio(c))
+#define client_is_audio(c)		((c)->id & ID_BIT_AUDIO)
+#define client_is_vga(c)		(!client_is_audio(c))
 #define client_id(c)		((c)->id & ~ID_BIT_AUDIO)
 
 static int vga_switcheroo_debugfs_init(struct vgasr_priv *priv);
@@ -192,14 +194,29 @@ static void vga_switcheroo_enable(void)
 		vgasr_priv.handler->init();
 
 	list_for_each_entry(client, &vgasr_priv.clients, list) {
-		if (client->id != VGA_SWITCHEROO_UNKNOWN_ID)
+		if (!client_is_vga(client) ||
+		     client_id(client) != VGA_SWITCHEROO_UNKNOWN_ID)
 			continue;
+
 		ret = vgasr_priv.handler->get_client_id(client->pdev);
 		if (ret < 0)
 			return;
 
 		client->id = ret;
 	}
+
+	list_for_each_entry(client, &vgasr_priv.clients, list) {
+		if (!client_is_audio(client) ||
+		     client_id(client) != VGA_SWITCHEROO_UNKNOWN_ID)
+			continue;
+
+		ret = vgasr_priv.handler->get_client_id(client->vga_dev);
+		if (ret < 0)
+			return;
+
+		client->id = ret | ID_BIT_AUDIO;
+	}
+
 	vga_switcheroo_debugfs_init(&vgasr_priv);
 	vgasr_priv.active = true;
 }
@@ -272,7 +289,9 @@ EXPORT_SYMBOL(vga_switcheroo_handler_flags);
 
 static int register_client(struct pci_dev *pdev,
 			   const struct vga_switcheroo_client_ops *ops,
-			   enum vga_switcheroo_client_id id, bool active,
+			   enum vga_switcheroo_client_id id,
+			   struct pci_dev *vga_dev,
+			   bool active,
 			   bool driver_power_control)
 {
 	struct vga_switcheroo_client *client;
@@ -287,6 +306,7 @@ static int register_client(struct pci_dev *pdev,
 	client->id = id;
 	client->active = active;
 	client->driver_power_control = driver_power_control;
+	client->vga_dev = vga_dev;
 
 	mutex_lock(&vgasr_mutex);
 	list_add_tail(&client->list, &vgasr_priv.clients);
@@ -319,7 +339,7 @@ int vga_switcheroo_register_client(struct pci_dev *pdev,
 				   const struct vga_switcheroo_client_ops *ops,
 				   bool driver_power_control)
 {
-	return register_client(pdev, ops, VGA_SWITCHEROO_UNKNOWN_ID,
+	return register_client(pdev, ops, VGA_SWITCHEROO_UNKNOWN_ID, NULL,
 			       pdev == vga_default_device(),
 			       driver_power_control);
 }
@@ -329,19 +349,40 @@ EXPORT_SYMBOL(vga_switcheroo_register_client);
  * vga_switcheroo_register_audio_client - register audio client
  * @pdev: client pci device
  * @ops: client callbacks
- * @id: client identifier
+ * @vga_dev:  pci device which is bound to current audio client
  *
  * Register audio client (audio device on a GPU). The client is assumed
  * to use runtime PM. Beforehand, vga_switcheroo_client_probe_defer()
  * shall be called to ensure that all prerequisites are met.
  *
- * Return: 0 on success, -ENOMEM on memory allocation error.
+ * Return: 0 on success, -ENOMEM on memory allocation error, -EINVAL on getting
+ * client id error.
  */
 int vga_switcheroo_register_audio_client(struct pci_dev *pdev,
 			const struct vga_switcheroo_client_ops *ops,
-			enum vga_switcheroo_client_id id)
+			struct pci_dev *vga_dev)
 {
-	return register_client(pdev, ops, id | ID_BIT_AUDIO, false, true);
+	enum vga_switcheroo_client_id id = VGA_SWITCHEROO_UNKNOWN_ID;
+
+	/*
+	 * if vga_switcheroo has enabled, that mean two GPU clients and also
+	 * handler are registered. Get audio client id from bound GPU client
+	 * id directly, otherwise, set it as VGA_SWITCHEROO_UNKNOWN_ID,
+	 * it will set to correct id in later when vga_switcheroo_enable()
+	 * is called.
+	 */
+	mutex_lock(&vgasr_mutex);
+	if (vgasr_priv.active) {
+		id = vgasr_priv.handler->get_client_id(vga_dev);
+		if (id < 0) {
+			mutex_unlock(&vgasr_mutex);
+			return -EINVAL;
+		}
+	}
+	mutex_unlock(&vgasr_mutex);
+
+	return register_client(pdev, ops, id | ID_BIT_AUDIO, vga_dev,
+			       false, true);
 }
 EXPORT_SYMBOL(vga_switcheroo_register_audio_client);
 
diff --git a/drivers/media/platform/pxa_camera.c b/drivers/media/platform/pxa_camera.c
index d85ffbfb7c1f..b6e9e93bde7a 100644
--- a/drivers/media/platform/pxa_camera.c
+++ b/drivers/media/platform/pxa_camera.c
@@ -2375,8 +2375,6 @@ static int pxa_camera_probe(struct platform_device *pdev)
 		.src_maxburst = 8,
 		.direction = DMA_DEV_TO_MEM,
 	};
-	dma_cap_mask_t mask;
-	struct pxad_param params;
 	char clk_name[V4L2_CLK_NAME_SIZE];
 	int irq;
 	int err = 0, i;
@@ -2450,34 +2448,20 @@ static int pxa_camera_probe(struct platform_device *pdev)
 	pcdev->base = base;
 
 	/* request dma */
-	dma_cap_zero(mask);
-	dma_cap_set(DMA_SLAVE, mask);
-	dma_cap_set(DMA_PRIVATE, mask);
-
-	params.prio = 0;
-	params.drcmr = 68;
-	pcdev->dma_chans[0] =
-		dma_request_slave_channel_compat(mask, pxad_filter_fn,
-						 &params, &pdev->dev, "CI_Y");
+	pcdev->dma_chans[0] = dma_request_slave_channel(&pdev->dev, "CI_Y");
 	if (!pcdev->dma_chans[0]) {
 		dev_err(&pdev->dev, "Can't request DMA for Y\n");
 		return -ENODEV;
 	}
 
-	params.drcmr = 69;
-	pcdev->dma_chans[1] =
-		dma_request_slave_channel_compat(mask, pxad_filter_fn,
-						 &params, &pdev->dev, "CI_U");
+	pcdev->dma_chans[1] = dma_request_slave_channel(&pdev->dev, "CI_U");
 	if (!pcdev->dma_chans[1]) {
 		dev_err(&pdev->dev, "Can't request DMA for Y\n");
 		err = -ENODEV;
 		goto exit_free_dma_y;
 	}
 
-	params.drcmr = 70;
-	pcdev->dma_chans[2] =
-		dma_request_slave_channel_compat(mask, pxad_filter_fn,
-						 &params, &pdev->dev, "CI_V");
+	pcdev->dma_chans[2] = dma_request_slave_channel(&pdev->dev, "CI_V");
 	if (!pcdev->dma_chans[2]) {
 		dev_err(&pdev->dev, "Can't request DMA for V\n");
 		err = -ENODEV;
diff --git a/drivers/mmc/host/pxamci.c b/drivers/mmc/host/pxamci.c
index c763b404510f..6c94474e36f4 100644
--- a/drivers/mmc/host/pxamci.c
+++ b/drivers/mmc/host/pxamci.c
@@ -24,7 +24,6 @@
 #include <linux/interrupt.h>
 #include <linux/dmaengine.h>
 #include <linux/dma-mapping.h>
-#include <linux/dma/pxa-dma.h>
 #include <linux/clk.h>
 #include <linux/err.h>
 #include <linux/mmc/host.h>
@@ -637,10 +636,8 @@ static int pxamci_probe(struct platform_device *pdev)
 {
 	struct mmc_host *mmc;
 	struct pxamci_host *host = NULL;
-	struct resource *r, *dmarx, *dmatx;
-	struct pxad_param param_rx, param_tx;
+	struct resource *r;
 	int ret, irq, gpio_cd = -1, gpio_ro = -1, gpio_power = -1;
-	dma_cap_mask_t mask;
 
 	ret = pxamci_of_init(pdev);
 	if (ret)
@@ -739,34 +736,14 @@ static int pxamci_probe(struct platform_device *pdev)
 
 	platform_set_drvdata(pdev, mmc);
 
-	if (!pdev->dev.of_node) {
-		dmarx = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-		dmatx = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-		if (!dmarx || !dmatx) {
-			ret = -ENXIO;
-			goto out;
-		}
-		param_rx.prio = PXAD_PRIO_LOWEST;
-		param_rx.drcmr = dmarx->start;
-		param_tx.prio = PXAD_PRIO_LOWEST;
-		param_tx.drcmr = dmatx->start;
-	}
-
-	dma_cap_zero(mask);
-	dma_cap_set(DMA_SLAVE, mask);
-
-	host->dma_chan_rx =
-		dma_request_slave_channel_compat(mask, pxad_filter_fn,
-						 &param_rx, &pdev->dev, "rx");
+	host->dma_chan_rx = dma_request_slave_channel(&pdev->dev, "rx");
 	if (host->dma_chan_rx == NULL) {
 		dev_err(&pdev->dev, "unable to request rx dma channel\n");
 		ret = -ENODEV;
 		goto out;
 	}
 
-	host->dma_chan_tx =
-		dma_request_slave_channel_compat(mask, pxad_filter_fn,
-						 &param_tx,  &pdev->dev, "tx");
+	host->dma_chan_tx = dma_request_slave_channel(&pdev->dev, "tx");
 	if (host->dma_chan_tx == NULL) {
 		dev_err(&pdev->dev, "unable to request tx dma channel\n");
 		ret = -ENODEV;
diff --git a/drivers/mtd/nand/raw/marvell_nand.c b/drivers/mtd/nand/raw/marvell_nand.c
index 218e09431d3d..7af4d6213ee5 100644
--- a/drivers/mtd/nand/raw/marvell_nand.c
+++ b/drivers/mtd/nand/raw/marvell_nand.c
@@ -2618,8 +2618,6 @@ static int marvell_nfc_init_dma(struct marvell_nfc *nfc)
 						    dev);
 	struct dma_slave_config config = {};
 	struct resource *r;
-	dma_cap_mask_t mask;
-	struct pxad_param param;
 	int ret;
 
 	if (!IS_ENABLED(CONFIG_PXA_DMA)) {
@@ -2632,20 +2630,7 @@ static int marvell_nfc_init_dma(struct marvell_nfc *nfc)
 	if (ret)
 		return ret;
 
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!r) {
-		dev_err(nfc->dev, "No resource defined for data DMA\n");
-		return -ENXIO;
-	}
-
-	param.drcmr = r->start;
-	param.prio = PXAD_PRIO_LOWEST;
-	dma_cap_zero(mask);
-	dma_cap_set(DMA_SLAVE, mask);
-	nfc->dma_chan =
-		dma_request_slave_channel_compat(mask, pxad_filter_fn,
-						 &param, nfc->dev,
-						 "data");
+	nfc->dma_chan =	dma_request_slave_channel(nfc->dev, "data");
 	if (!nfc->dma_chan) {
 		dev_err(nfc->dev,
 			"Unable to request data DMA channel\n");
diff --git a/drivers/staging/most/sound/sound.c b/drivers/staging/most/sound/sound.c
index 04c18323c2ea..89b02fc305b8 100644
--- a/drivers/staging/most/sound/sound.c
+++ b/drivers/staging/most/sound/sound.c
@@ -457,7 +457,6 @@ static const struct snd_pcm_ops pcm_ops = {
 	.trigger    = pcm_trigger,
 	.pointer    = pcm_pointer,
 	.page       = snd_pcm_lib_get_vmalloc_page,
-	.mmap       = snd_pcm_lib_mmap_vmalloc,
 };
 
 static int split_arg_list(char *buf, char **card_name, u16 *ch_num,
diff --git a/include/drm/drm_audio_component.h b/include/drm/drm_audio_component.h
new file mode 100644
index 000000000000..4923b00328c1
--- /dev/null
+++ b/include/drm/drm_audio_component.h
@@ -0,0 +1,118 @@
+// SPDX-License-Identifier: MIT
+// Copyright © 2014 Intel Corporation
+
+#ifndef _DRM_AUDIO_COMPONENT_H_
+#define _DRM_AUDIO_COMPONENT_H_
+
+struct drm_audio_component;
+
+/**
+ * struct drm_audio_component_ops - Ops implemented by DRM driver, called by hda driver
+ */
+struct drm_audio_component_ops {
+	/**
+	 * @owner: drm module to pin down
+	 */
+	struct module *owner;
+	/**
+	 * @get_power: get the POWER_DOMAIN_AUDIO power well
+	 *
+	 * Request the power well to be turned on.
+	 */
+	void (*get_power)(struct device *);
+	/**
+	 * @put_power: put the POWER_DOMAIN_AUDIO power well
+	 *
+	 * Allow the power well to be turned off.
+	 */
+	void (*put_power)(struct device *);
+	/**
+	 * @codec_wake_override: Enable/disable codec wake signal
+	 */
+	void (*codec_wake_override)(struct device *, bool enable);
+	/**
+	 * @get_cdclk_freq: Get the Core Display Clock in kHz
+	 */
+	int (*get_cdclk_freq)(struct device *);
+	/**
+	 * @sync_audio_rate: set n/cts based on the sample rate
+	 *
+	 * Called from audio driver. After audio driver sets the
+	 * sample rate, it will call this function to set n/cts
+	 */
+	int (*sync_audio_rate)(struct device *, int port, int pipe, int rate);
+	/**
+	 * @get_eld: fill the audio state and ELD bytes for the given port
+	 *
+	 * Called from audio driver to get the HDMI/DP audio state of the given
+	 * digital port, and also fetch ELD bytes to the given pointer.
+	 *
+	 * It returns the byte size of the original ELD (not the actually
+	 * copied size), zero for an invalid ELD, or a negative error code.
+	 *
+	 * Note that the returned size may be over @max_bytes.  Then it
+	 * implies that only a part of ELD has been copied to the buffer.
+	 */
+	int (*get_eld)(struct device *, int port, int pipe, bool *enabled,
+		       unsigned char *buf, int max_bytes);
+};
+
+/**
+ * struct drm_audio_component_audio_ops - Ops implemented by hda driver, called by DRM driver
+ */
+struct drm_audio_component_audio_ops {
+	/**
+	 * @audio_ptr: Pointer to be used in call to pin_eld_notify
+	 */
+	void *audio_ptr;
+	/**
+	 * @pin_eld_notify: Notify the HDA driver that pin sense and/or ELD information has changed
+	 *
+	 * Called when the DRM driver has set up audio pipeline or has just
+	 * begun to tear it down. This allows the HDA driver to update its
+	 * status accordingly (even when the HDA controller is in power save
+	 * mode).
+	 */
+	void (*pin_eld_notify)(void *audio_ptr, int port, int pipe);
+	/**
+	 * @pin2port: Check and convert from pin node to port number
+	 *
+	 * Called by HDA driver to check and convert from the pin widget node
+	 * number to a port number in the graphics side.
+	 */
+	int (*pin2port)(void *audio_ptr, int pin);
+	/**
+	 * @master_bind: (Optional) component master bind callback
+	 *
+	 * Called at binding master component, for HDA codec-specific
+	 * handling of dynamic binding.
+	 */
+	int (*master_bind)(struct device *dev, struct drm_audio_component *);
+	/**
+	 * @master_unbind: (Optional) component master unbind callback
+	 *
+	 * Called at unbinding master component, for HDA codec-specific
+	 * handling of dynamic unbinding.
+	 */
+	void (*master_unbind)(struct device *dev, struct drm_audio_component *);
+};
+
+/**
+ * struct drm_audio_component - Used for direct communication between DRM and hda drivers
+ */
+struct drm_audio_component {
+	/**
+	 * @dev: DRM device, used as parameter for ops
+	 */
+	struct device *dev;
+	/**
+	 * @ops: Ops implemented by DRM driver, called by hda driver
+	 */
+	const struct drm_audio_component_ops *ops;
+	/**
+	 * @audio_ops: Ops implemented by hda driver, called by DRM driver
+	 */
+	const struct drm_audio_component_audio_ops *audio_ops;
+};
+
+#endif /* _DRM_AUDIO_COMPONENT_H_ */
diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h
index 346b1f5cb180..fca22d463e1b 100644
--- a/include/drm/i915_component.h
+++ b/include/drm/i915_component.h
@@ -24,101 +24,26 @@
 #ifndef _I915_COMPONENT_H_
 #define _I915_COMPONENT_H_
 
+#include "drm_audio_component.h"
+
 /* MAX_PORT is the number of port
  * It must be sync with I915_MAX_PORTS defined i915_drv.h
  */
 #define MAX_PORTS 6
 
 /**
- * struct i915_audio_component_ops - Ops implemented by i915 driver, called by hda driver
- */
-struct i915_audio_component_ops {
-	/**
-	 * @owner: i915 module
-	 */
-	struct module *owner;
-	/**
-	 * @get_power: get the POWER_DOMAIN_AUDIO power well
-	 *
-	 * Request the power well to be turned on.
-	 */
-	void (*get_power)(struct device *);
-	/**
-	 * @put_power: put the POWER_DOMAIN_AUDIO power well
-	 *
-	 * Allow the power well to be turned off.
-	 */
-	void (*put_power)(struct device *);
-	/**
-	 * @codec_wake_override: Enable/disable codec wake signal
-	 */
-	void (*codec_wake_override)(struct device *, bool enable);
-	/**
-	 * @get_cdclk_freq: Get the Core Display Clock in kHz
-	 */
-	int (*get_cdclk_freq)(struct device *);
-	/**
-	 * @sync_audio_rate: set n/cts based on the sample rate
-	 *
-	 * Called from audio driver. After audio driver sets the
-	 * sample rate, it will call this function to set n/cts
-	 */
-	int (*sync_audio_rate)(struct device *, int port, int pipe, int rate);
-	/**
-	 * @get_eld: fill the audio state and ELD bytes for the given port
-	 *
-	 * Called from audio driver to get the HDMI/DP audio state of the given
-	 * digital port, and also fetch ELD bytes to the given pointer.
-	 *
-	 * It returns the byte size of the original ELD (not the actually
-	 * copied size), zero for an invalid ELD, or a negative error code.
-	 *
-	 * Note that the returned size may be over @max_bytes.  Then it
-	 * implies that only a part of ELD has been copied to the buffer.
-	 */
-	int (*get_eld)(struct device *, int port, int pipe, bool *enabled,
-		       unsigned char *buf, int max_bytes);
-};
-
-/**
- * struct i915_audio_component_audio_ops - Ops implemented by hda driver, called by i915 driver
- */
-struct i915_audio_component_audio_ops {
-	/**
-	 * @audio_ptr: Pointer to be used in call to pin_eld_notify
-	 */
-	void *audio_ptr;
-	/**
-	 * @pin_eld_notify: Notify the HDA driver that pin sense and/or ELD information has changed
-	 *
-	 * Called when the i915 driver has set up audio pipeline or has just
-	 * begun to tear it down. This allows the HDA driver to update its
-	 * status accordingly (even when the HDA controller is in power save
-	 * mode).
-	 */
-	void (*pin_eld_notify)(void *audio_ptr, int port, int pipe);
-};
-
-/**
  * struct i915_audio_component - Used for direct communication between i915 and hda drivers
  */
 struct i915_audio_component {
 	/**
-	 * @dev: i915 device, used as parameter for ops
+	 * @base: the drm_audio_component base class
 	 */
-	struct device *dev;
+	struct drm_audio_component	base;
+
 	/**
 	 * @aud_sample_rate: the array of audio sample rate per port
 	 */
 	int aud_sample_rate[MAX_PORTS];
-	/**
-	 * @ops: Ops implemented by i915 driver, called by hda driver
-	 */
-	const struct i915_audio_component_ops *ops;
-	/**
-	 * @audio_ops: Ops implemented by hda driver, called by i915 driver
-	 */
-	const struct i915_audio_component_audio_ops *audio_ops;
 };
 
 #endif /* _I915_COMPONENT_H_ */
diff --git a/include/linux/clk-provider.h b/include/linux/clk-provider.h
index b7cfa037e593..08b1aa70a38d 100644
--- a/include/linux/clk-provider.h
+++ b/include/linux/clk-provider.h
@@ -38,6 +38,8 @@
 #define CLK_IS_CRITICAL		BIT(11) /* do not gate, ever */
 /* parents need enable during gate/ungate, set rate and re-parent */
 #define CLK_OPS_PARENT_ENABLE	BIT(12)
+/* duty cycle call may be forwarded to the parent clock */
+#define CLK_DUTY_CYCLE_PARENT	BIT(13)
 
 struct clk;
 struct clk_hw;
@@ -67,6 +69,17 @@ struct clk_rate_request {
 };
 
 /**
+ * struct clk_duty - Struture encoding the duty cycle ratio of a clock
+ *
+ * @num:	Numerator of the duty cycle ratio
+ * @den:	Denominator of the duty cycle ratio
+ */
+struct clk_duty {
+	unsigned int num;
+	unsigned int den;
+};
+
+/**
  * struct clk_ops -  Callback operations for hardware clocks; these are to
  * be provided by the clock implementation, and will be called by drivers
  * through the clk_* api.
@@ -169,6 +182,15 @@ struct clk_rate_request {
  *		by the second argument. Valid values for degrees are
  *		0-359. Return 0 on success, otherwise -EERROR.
  *
+ * @get_duty_cycle: Queries the hardware to get the current duty cycle ratio
+ *              of a clock. Returned values denominator cannot be 0 and must be
+ *              superior or equal to the numerator.
+ *
+ * @set_duty_cycle: Apply the duty cycle ratio to this clock signal specified by
+ *              the numerator (2nd argurment) and denominator (3rd  argument).
+ *              Argument must be a valid ratio (denominator > 0
+ *              and >= numerator) Return 0 on success, otherwise -EERROR.
+ *
  * @init:	Perform platform-specific initialization magic.
  *		This is not not used by any of the basic clock types.
  *		Please consider other ways of solving initialization problems
@@ -218,6 +240,10 @@ struct clk_ops {
 					   unsigned long parent_accuracy);
 	int		(*get_phase)(struct clk_hw *hw);
 	int		(*set_phase)(struct clk_hw *hw, int degrees);
+	int		(*get_duty_cycle)(struct clk_hw *hw,
+					  struct clk_duty *duty);
+	int		(*set_duty_cycle)(struct clk_hw *hw,
+					  struct clk_duty *duty);
 	void		(*init)(struct clk_hw *hw);
 	void		(*debug_init)(struct clk_hw *hw, struct dentry *dentry);
 };
diff --git a/include/linux/clk.h b/include/linux/clk.h
index 0dbd0885b2c2..4f750c481b82 100644
--- a/include/linux/clk.h
+++ b/include/linux/clk.h
@@ -142,6 +142,27 @@ int clk_set_phase(struct clk *clk, int degrees);
 int clk_get_phase(struct clk *clk);
 
 /**
+ * clk_set_duty_cycle - adjust the duty cycle ratio of a clock signal
+ * @clk: clock signal source
+ * @num: numerator of the duty cycle ratio to be applied
+ * @den: denominator of the duty cycle ratio to be applied
+ *
+ * Adjust the duty cycle of a clock signal by the specified ratio. Returns 0 on
+ * success, -EERROR otherwise.
+ */
+int clk_set_duty_cycle(struct clk *clk, unsigned int num, unsigned int den);
+
+/**
+ * clk_get_duty_cycle - return the duty cycle ratio of a clock signal
+ * @clk: clock signal source
+ * @scale: scaling factor to be applied to represent the ratio as an integer
+ *
+ * Returns the duty cycle ratio multiplied by the scale provided, otherwise
+ * returns -EERROR.
+ */
+int clk_get_scaled_duty_cycle(struct clk *clk, unsigned int scale);
+
+/**
  * clk_is_match - check if two clk's point to the same hardware clock
  * @p: clk compared against q
  * @q: clk compared against p
@@ -183,6 +204,18 @@ static inline long clk_get_phase(struct clk *clk)
 	return -ENOTSUPP;
 }
 
+static inline int clk_set_duty_cycle(struct clk *clk, unsigned int num,
+				     unsigned int den)
+{
+	return -ENOTSUPP;
+}
+
+static inline unsigned int clk_get_scaled_duty_cycle(struct clk *clk,
+						     unsigned int scale)
+{
+	return 0;
+}
+
 static inline bool clk_is_match(const struct clk *p, const struct clk *q)
 {
 	return p == q;
diff --git a/include/linux/dma/pxa-dma.h b/include/linux/dma/pxa-dma.h
index e56ec7af4fd7..9fc594f69eff 100644
--- a/include/linux/dma/pxa-dma.h
+++ b/include/linux/dma/pxa-dma.h
@@ -9,6 +9,15 @@ enum pxad_chan_prio {
 	PXAD_PRIO_LOWEST,
 };
 
+/**
+ * struct pxad_param - dma channel request parameters
+ * @drcmr: requestor line number
+ * @prio: minimal mandatory priority of the channel
+ *
+ * If a requested channel is granted, its priority will be at least @prio,
+ * ie. if PXAD_PRIO_LOW is required, the requested channel will be either
+ * PXAD_PRIO_LOW, PXAD_PRIO_NORMAL or PXAD_PRIO_HIGHEST.
+ */
 struct pxad_param {
 	unsigned int drcmr;
 	enum pxad_chan_prio prio;
diff --git a/include/linux/platform_data/mmp_dma.h b/include/linux/platform_data/mmp_dma.h
index d1397c8ed94e..6397b9c8149a 100644
--- a/include/linux/platform_data/mmp_dma.h
+++ b/include/linux/platform_data/mmp_dma.h
@@ -12,9 +12,13 @@
 #ifndef MMP_DMA_H
 #define MMP_DMA_H
 
+struct dma_slave_map;
+
 struct mmp_dma_platdata {
 	int dma_channels;
 	int nb_requestors;
+	int slave_map_cnt;
+	const struct dma_slave_map *slave_map;
 };
 
 #endif /* MMP_DMA_H */
diff --git a/include/linux/pxa2xx_ssp.h b/include/linux/pxa2xx_ssp.h
index 8461b18e4608..13b4244d44c1 100644
--- a/include/linux/pxa2xx_ssp.h
+++ b/include/linux/pxa2xx_ssp.h
@@ -171,6 +171,14 @@
 #define SSACD_SCDB		(1 << 3)	/* SSPSYSCLK Divider Bypass */
 #define SSACD_ACPS(x)		((x) << 4)	/* Audio clock PLL select */
 #define SSACD_ACDS(x)		((x) << 0)	/* Audio clock divider select */
+#define SSACD_ACDS_1		(0)
+#define SSACD_ACDS_2		(1)
+#define SSACD_ACDS_4		(2)
+#define SSACD_ACDS_8		(3)
+#define SSACD_ACDS_16		(4)
+#define SSACD_ACDS_32		(5)
+#define SSACD_SCDB_4X		(0)
+#define SSACD_SCDB_1X		(1)
 #define SSACD_SCDX8		(1 << 7)	/* SYSCLK division ratio select */
 
 /* LPSS SSP */
@@ -212,8 +220,6 @@ struct ssp_device {
 	int		type;
 	int		use_count;
 	int		irq;
-	int		drcmr_rx;
-	int		drcmr_tx;
 
 	struct device_node	*of_node;
 };
diff --git a/include/linux/usb/audio-v3.h b/include/linux/usb/audio-v3.h
index a710e28b5215..6b708434b7f9 100644
--- a/include/linux/usb/audio-v3.h
+++ b/include/linux/usb/audio-v3.h
@@ -387,6 +387,12 @@ struct uac3_interrupt_data_msg {
 #define UAC3_CONNECTORS			0x0f
 #define UAC3_POWER_DOMAIN		0x10
 
+/* A.20 PROCESSING UNIT PROCESS TYPES */
+#define UAC3_PROCESS_UNDEFINED		0x00
+#define UAC3_PROCESS_UP_DOWNMIX		0x01
+#define UAC3_PROCESS_STEREO_EXTENDER	0x02
+#define UAC3_PROCESS_MULTI_FUNCTION	0x03
+
 /* A.22 AUDIO CLASS-SPECIFIC REQUEST CODES */
 /* see audio-v2.h for the rest, which is identical to v2 */
 #define UAC3_CS_REQ_INTEN			0x04
@@ -406,6 +412,15 @@ struct uac3_interrupt_data_msg {
 #define UAC3_TE_OVERFLOW			0x04
 #define UAC3_TE_LATENCY 			0x05
 
+/* A.23.10 PROCESSING UNITS CONTROL SELECTROS */
+
+/* Up/Down Mixer */
+#define UAC3_UD_MODE_SELECT			0x01
+
+/* Stereo Extender */
+#define UAC3_EXT_WIDTH_CONTROL			0x01
+
+
 /* BADD predefined Unit/Terminal values */
 #define UAC3_BADD_IT_ID1	1  /* Input Terminal ID1: bTerminalID = 1 */
 #define UAC3_BADD_FU_ID2	2  /* Feature Unit ID2: bUnitID = 2 */
@@ -432,4 +447,8 @@ struct uac3_interrupt_data_msg {
 /* BADD sample rate is always fixed to 48kHz */
 #define UAC3_BADD_SAMPLING_RATE				48000
 
+/* BADD power domains recovery times in 50us increments */
+#define UAC3_BADD_PD_RECOVER_D1D0			0x0258	/* 30ms */
+#define UAC3_BADD_PD_RECOVER_D2D0			0x1770	/* 300ms */
+
 #endif /* __LINUX_USB_AUDIO_V3_H */
diff --git a/include/linux/vga_switcheroo.h b/include/linux/vga_switcheroo.h
index 77f0f0af3a71..a34539b7f750 100644
--- a/include/linux/vga_switcheroo.h
+++ b/include/linux/vga_switcheroo.h
@@ -84,8 +84,8 @@ enum vga_switcheroo_state {
  * Client identifier. Audio clients use the same identifier & 0x100.
  */
 enum vga_switcheroo_client_id {
-	VGA_SWITCHEROO_UNKNOWN_ID = -1,
-	VGA_SWITCHEROO_IGD,
+	VGA_SWITCHEROO_UNKNOWN_ID = 0x1000,
+	VGA_SWITCHEROO_IGD = 0,
 	VGA_SWITCHEROO_DIS,
 	VGA_SWITCHEROO_MAX_CLIENTS,
 };
@@ -151,7 +151,7 @@ int vga_switcheroo_register_client(struct pci_dev *dev,
 				   bool driver_power_control);
 int vga_switcheroo_register_audio_client(struct pci_dev *pdev,
 					 const struct vga_switcheroo_client_ops *ops,
-					 enum vga_switcheroo_client_id id);
+					 struct pci_dev *vga_dev);
 
 void vga_switcheroo_client_fb_set(struct pci_dev *dev,
 				  struct fb_info *info);
@@ -180,7 +180,7 @@ static inline int vga_switcheroo_register_handler(const struct vga_switcheroo_ha
 		enum vga_switcheroo_handler_flags_t handler_flags) { return 0; }
 static inline int vga_switcheroo_register_audio_client(struct pci_dev *pdev,
 	const struct vga_switcheroo_client_ops *ops,
-	enum vga_switcheroo_client_id id) { return 0; }
+	struct pci_dev *vga_dev) { return 0; }
 static inline void vga_switcheroo_unregister_handler(void) {}
 static inline enum vga_switcheroo_handler_flags_t vga_switcheroo_handler_flags(void) { return 0; }
 static inline int vga_switcheroo_lock_ddc(struct pci_dev *pdev) { return -ENODEV; }
diff --git a/include/sound/ac97/codec.h b/include/sound/ac97/codec.h
index ec04be9ab119..9792d25fa369 100644
--- a/include/sound/ac97/codec.h
+++ b/include/sound/ac97/codec.h
@@ -1,10 +1,8 @@
-/*
- *  Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr>
+/* SPDX-License-Identifier: GPL-2.0
  *
- * 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.
+ *  Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr>
  */
+
 #ifndef __SOUND_AC97_CODEC2_H
 #define __SOUND_AC97_CODEC2_H
 
diff --git a/include/sound/ac97/compat.h b/include/sound/ac97/compat.h
index 1351cba40048..57e19afa31ab 100644
--- a/include/sound/ac97/compat.h
+++ b/include/sound/ac97/compat.h
@@ -1,14 +1,11 @@
-/*
- *  Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr>
+/* SPDX-License-Identifier: GPL-2.0
  *
- * 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.
+ *  Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr>
  *
  * This file is for backward compatibility with snd_ac97 structure and its
  * multiple usages, such as the snd_ac97_bus and snd_ac97_build_ops.
- *
  */
+
 #ifndef AC97_COMPAT_H
 #define AC97_COMPAT_H
 
diff --git a/include/sound/ac97/controller.h b/include/sound/ac97/controller.h
index b36ecdd64f14..06b5afb7fa6b 100644
--- a/include/sound/ac97/controller.h
+++ b/include/sound/ac97/controller.h
@@ -1,10 +1,8 @@
-/*
- *  Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr>
+/* SPDX-License-Identifier: GPL-2.0
  *
- * 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.
+ * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr>
  */
+
 #ifndef AC97_CONTROLLER_H
 #define AC97_CONTROLLER_H
 
diff --git a/include/sound/ac97/regs.h b/include/sound/ac97/regs.h
index 9a4fa0c3264a..843f73f3705a 100644
--- a/include/sound/ac97/regs.h
+++ b/include/sound/ac97/regs.h
@@ -1,27 +1,11 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0+
+ *
  *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *  Universal interface for Audio Codec '97
  *
  *  For more details look to AC '97 component specification revision 2.1
  *  by Intel Corporation (http://developer.intel.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.
- *
- *   This program is distributed in the hope that it will be useful,
- *   but WITHOUT ANY WARRANTY; without even the implied warranty of
- *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- *   GNU General Public License for more details.
- *
- *   You should have received a copy of the GNU General Public License
- *   along with this program; if not, write to the Free Software
- *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
- *
  */
-
 /*
  *  AC'97 codec registers
  */
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 89d311a503d3..cc383991c0fe 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -1,30 +1,15 @@
-#ifndef __SOUND_AC97_CODEC_H
-#define __SOUND_AC97_CODEC_H
-
-/*
+/* SPDX-License-Identifier: GPL-2.0+
+ *
  *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *  Universal interface for Audio Codec '97
  *
  *  For more details look to AC '97 component specification revision 2.1
  *  by Intel Corporation (http://developer.intel.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.
- *
- *   This program is distributed in the hope that it will be useful,
- *   but WITHOUT ANY WARRANTY; without even the implied warranty of
- *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- *   GNU General Public License for more details.
- *
- *   You should have received a copy of the GNU General Public License
- *   along with this program; if not, write to the Free Software
- *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
- *
  */
 
+#ifndef __SOUND_AC97_CODEC_H
+#define __SOUND_AC97_CODEC_H
+
 #include <linux/bitops.h>
 #include <linux/device.h>
 #include <linux/workqueue.h>
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
index 9924bc9cbc7c..ea8c93bbb0e0 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -1,27 +1,12 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  *  compress_driver.h - compress offload driver definations
  *
  *  Copyright (C) 2011 Intel Corporation
  *  Authors:	Vinod Koul <vinod.koul@linux.intel.com>
  *		Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.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; version 2 of the License.
- *
- *  This program is distributed in the hope that it will be useful, but
- *  WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  General Public License for more details.
- *
- *  You should have received a copy of the GNU General Public License along
- *  with this program; if not, write to the Free Software Foundation, Inc.,
- *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
- *
- * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
- *
  */
+
 #ifndef __COMPRESS_DRIVER_H
 #define __COMPRESS_DRIVER_H
 
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index e3481eebdd98..2c4cfaa135a6 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -1,17 +1,9 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0+
+ *
  *  Copyright (C) 2012, Analog Devices Inc.
  *	Author: Lars-Peter Clausen <lars@metafoo.de>
- *
- *  This program is free software; you can redistribute it and/or modify it
- *  under  the terms of the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the License, or (at your
- *  option) any later version.
- *
- *  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.,
- *  675 Mass Ave, Cambridge, MA 02139, USA.
- *
  */
+
 #ifndef __SOUND_DMAENGINE_PCM_H__
 #define __SOUND_DMAENGINE_PCM_H__
 
diff --git a/include/sound/hda_component.h b/include/sound/hda_component.h
new file mode 100644
index 000000000000..78626cde7081
--- /dev/null
+++ b/include/sound/hda_component.h
@@ -0,0 +1,61 @@
+// SPDX-License-Identifier: GPL-2.0
+// HD-Audio helpers to sync with DRM driver
+
+#ifndef __SOUND_HDA_COMPONENT_H
+#define __SOUND_HDA_COMPONENT_H
+
+#include <drm/drm_audio_component.h>
+
+#ifdef CONFIG_SND_HDA_COMPONENT
+int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable);
+int snd_hdac_display_power(struct hdac_bus *bus, bool enable);
+int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid,
+			     int dev_id, int rate);
+int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id,
+			   bool *audio_enabled, char *buffer, int max_bytes);
+int snd_hdac_acomp_init(struct hdac_bus *bus,
+			const struct drm_audio_component_audio_ops *aops,
+			int (*match_master)(struct device *, void *),
+			size_t extra_size);
+int snd_hdac_acomp_exit(struct hdac_bus *bus);
+int snd_hdac_acomp_register_notifier(struct hdac_bus *bus,
+				    const struct drm_audio_component_audio_ops *ops);
+#else
+static inline int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable)
+{
+	return 0;
+}
+static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable)
+{
+	return 0;
+}
+static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec,
+					   hda_nid_t nid, int dev_id, int rate)
+{
+	return 0;
+}
+static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid,
+					 int dev_id, bool *audio_enabled,
+					 char *buffer, int max_bytes)
+{
+	return -ENODEV;
+}
+static inline int snd_hdac_acomp_init(struct hdac_bus *bus,
+				      const struct drm_audio_component_audio_ops *aops,
+				      int (*match_master)(struct device *, void *),
+				      size_t extra_size)
+{
+	return -ENODEV;
+}
+static inline int snd_hdac_acomp_exit(struct hdac_bus *bus)
+{
+	return 0;
+}
+static inline int snd_hdac_acomp_register_notifier(struct hdac_bus *bus,
+						  const struct drm_audio_component_audio_ops *ops)
+{
+	return -ENODEV;
+}
+#endif
+
+#endif /* __SOUND_HDA_COMPONENT_H */
diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h
index a94f5b6f92ac..6b79614a893b 100644
--- a/include/sound/hda_i915.h
+++ b/include/sound/hda_i915.h
@@ -5,54 +5,23 @@
 #ifndef __SOUND_HDA_I915_H
 #define __SOUND_HDA_I915_H
 
-#include <drm/i915_component.h>
+#include "hda_component.h"
 
 #ifdef CONFIG_SND_HDA_I915
-int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable);
-int snd_hdac_display_power(struct hdac_bus *bus, bool enable);
 void snd_hdac_i915_set_bclk(struct hdac_bus *bus);
-int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid,
-			     int dev_id, int rate);
-int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id,
-			   bool *audio_enabled, char *buffer, int max_bytes);
 int snd_hdac_i915_init(struct hdac_bus *bus);
-int snd_hdac_i915_exit(struct hdac_bus *bus);
-int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *);
 #else
-static inline int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable)
-{
-	return 0;
-}
-static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable)
-{
-	return 0;
-}
 static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus)
 {
 }
-static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec,
-					   hda_nid_t nid, int dev_id, int rate)
-{
-	return 0;
-}
-static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid,
-					 int dev_id, bool *audio_enabled,
-					 char *buffer, int max_bytes)
-{
-	return -ENODEV;
-}
 static inline int snd_hdac_i915_init(struct hdac_bus *bus)
 {
 	return -ENODEV;
 }
+#endif
 static inline int snd_hdac_i915_exit(struct hdac_bus *bus)
 {
-	return 0;
+	return snd_hdac_acomp_exit(bus);
 }
-static inline int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *ops)
-{
-	return -ENODEV;
-}
-#endif
 
 #endif /* __SOUND_HDA_I915_H */
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index c052afc27547..6f1e1f3b3063 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -8,8 +8,10 @@
 
 #include <linux/device.h>
 #include <linux/interrupt.h>
+#include <linux/pm_runtime.h>
 #include <linux/timecounter.h>
 #include <sound/core.h>
+#include <sound/pcm.h>
 #include <sound/memalloc.h>
 #include <sound/hda_verbs.h>
 #include <drm/i915_component.h>
@@ -132,7 +134,7 @@ int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid,
 			   hda_nid_t *start_id);
 unsigned int snd_hdac_calc_stream_format(unsigned int rate,
 					 unsigned int channels,
-					 unsigned int format,
+					 snd_pcm_format_t format,
 					 unsigned int maxbps,
 					 unsigned short spdif_ctls);
 int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid,
@@ -171,12 +173,38 @@ int snd_hdac_power_down(struct hdac_device *codec);
 int snd_hdac_power_up_pm(struct hdac_device *codec);
 int snd_hdac_power_down_pm(struct hdac_device *codec);
 int snd_hdac_keep_power_up(struct hdac_device *codec);
+
+/* call this at entering into suspend/resume callbacks in codec driver */
+static inline void snd_hdac_enter_pm(struct hdac_device *codec)
+{
+	atomic_inc(&codec->in_pm);
+}
+
+/* call this at leaving from suspend/resume callbacks in codec driver */
+static inline void snd_hdac_leave_pm(struct hdac_device *codec)
+{
+	atomic_dec(&codec->in_pm);
+}
+
+static inline bool snd_hdac_is_in_pm(struct hdac_device *codec)
+{
+	return atomic_read(&codec->in_pm);
+}
+
+static inline bool snd_hdac_is_power_on(struct hdac_device *codec)
+{
+	return !pm_runtime_suspended(&codec->dev);
+}
 #else
 static inline int snd_hdac_power_up(struct hdac_device *codec) { return 0; }
 static inline int snd_hdac_power_down(struct hdac_device *codec) { return 0; }
 static inline int snd_hdac_power_up_pm(struct hdac_device *codec) { return 0; }
 static inline int snd_hdac_power_down_pm(struct hdac_device *codec) { return 0; }
 static inline int snd_hdac_keep_power_up(struct hdac_device *codec) { return 0; }
+static inline void snd_hdac_enter_pm(struct hdac_device *codec) {}
+static inline void snd_hdac_leave_pm(struct hdac_device *codec) {}
+static inline bool snd_hdac_is_in_pm(struct hdac_device *codec) { return 0; }
+static inline bool snd_hdac_is_power_on(struct hdac_device *codec) { return 1; }
 #endif
 
 /*
@@ -188,6 +216,11 @@ struct hdac_driver {
 	const struct hda_device_id *id_table;
 	int (*match)(struct hdac_device *dev, struct hdac_driver *drv);
 	void (*unsol_event)(struct hdac_device *dev, unsigned int event);
+
+	/* fields used by ext bus APIs */
+	int (*probe)(struct hdac_device *dev);
+	int (*remove)(struct hdac_device *dev);
+	void (*shutdown)(struct hdac_device *dev);
 };
 
 #define drv_to_hdac_driver(_drv) container_of(_drv, struct hdac_driver, driver)
@@ -209,6 +242,14 @@ struct hdac_bus_ops {
 };
 
 /*
+ * ops used for ASoC HDA codec drivers
+ */
+struct hdac_ext_bus_ops {
+	int (*hdev_attach)(struct hdac_device *hdev);
+	int (*hdev_detach)(struct hdac_device *hdev);
+};
+
+/*
  * Lowlevel I/O operators
  */
 struct hdac_io_ops {
@@ -250,11 +291,17 @@ struct hdac_rb {
  * @mlcap: MultiLink capabilities pointer
  * @gtscap: gts capabilities pointer
  * @drsmcap: dma resume capabilities pointer
+ * @num_streams: streams supported
+ * @idx: HDA link index
+ * @hlink_list: link list of HDA links
+ * @lock: lock for link mgmt
+ * @cmd_dma_state: state of cmd DMAs: CORB and RIRB
  */
 struct hdac_bus {
 	struct device *dev;
 	const struct hdac_bus_ops *ops;
 	const struct hdac_io_ops *io_ops;
+	const struct hdac_ext_bus_ops *ext_ops;
 
 	/* h/w resources */
 	unsigned long addr;
@@ -314,9 +361,19 @@ struct hdac_bus {
 	spinlock_t reg_lock;
 	struct mutex cmd_mutex;
 
-	/* i915 component interface */
-	struct i915_audio_component *audio_component;
-	int i915_power_refcount;
+	/* DRM component interface */
+	struct drm_audio_component *audio_component;
+	int drm_power_refcount;
+
+	/* parameters required for enhanced capabilities */
+	int num_streams;
+	int idx;
+
+	struct list_head hlink_list;
+
+	struct mutex lock;
+	bool cmd_dma_state;
+
 };
 
 int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,
diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h
index 9c14e21dda85..f34aced69ca8 100644
--- a/include/sound/hdaudio_ext.h
+++ b/include/sound/hdaudio_ext.h
@@ -4,38 +4,16 @@
 
 #include <sound/hdaudio.h>
 
-/**
- * hdac_ext_bus: HDAC extended bus for extended HDA caps
- *
- * @bus: hdac bus
- * @num_streams: streams supported
- * @hlink_list: link list of HDA links
- * @lock: lock for link mgmt
- * @cmd_dma_state: state of cmd DMAs: CORB and RIRB
- */
-struct hdac_ext_bus {
-	struct hdac_bus bus;
-	int num_streams;
-	int idx;
-
-	struct list_head hlink_list;
-
-	struct mutex lock;
-	bool cmd_dma_state;
-};
-
-int snd_hdac_ext_bus_init(struct hdac_ext_bus *sbus, struct device *dev,
+int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev,
 		      const struct hdac_bus_ops *ops,
-		      const struct hdac_io_ops *io_ops);
+		      const struct hdac_io_ops *io_ops,
+		      const struct hdac_ext_bus_ops *ext_ops);
 
-void snd_hdac_ext_bus_exit(struct hdac_ext_bus *sbus);
-int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *sbus, int addr);
+void snd_hdac_ext_bus_exit(struct hdac_bus *bus);
+int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr,
+						struct hdac_device *hdev);
 void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev);
-void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus);
-
-#define ebus_to_hbus(ebus)	(&(ebus)->bus)
-#define hbus_to_ebus(_bus) \
-	container_of(_bus, struct hdac_ext_bus, bus)
+void snd_hdac_ext_bus_device_remove(struct hdac_bus *bus);
 
 #define HDA_CODEC_REV_EXT_ENTRY(_vid, _rev, _name, drv_data) \
 	{ .vendor_id = (_vid), .rev_id = (_rev), .name = (_name), \
@@ -44,14 +22,14 @@ void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus);
 #define HDA_CODEC_EXT_ENTRY(_vid, _revid, _name, _drv_data) \
 	HDA_CODEC_REV_EXT_ENTRY(_vid, _revid, _name, _drv_data)
 
-void snd_hdac_ext_bus_ppcap_enable(struct hdac_ext_bus *chip, bool enable);
-void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_ext_bus *chip, bool enable);
+void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *chip, bool enable);
+void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *chip, bool enable);
 
-void snd_hdac_ext_stream_spbcap_enable(struct hdac_ext_bus *chip,
+void snd_hdac_ext_stream_spbcap_enable(struct hdac_bus *chip,
 				 bool enable, int index);
 
-int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *bus);
-struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *bus,
+int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_bus *bus);
+struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_bus *bus,
 						const char *codec_name);
 
 enum hdac_ext_stream_type {
@@ -100,28 +78,28 @@ struct hdac_ext_stream {
 #define stream_to_hdac_ext_stream(s) \
 	container_of(s, struct hdac_ext_stream, hstream)
 
-void snd_hdac_ext_stream_init(struct hdac_ext_bus *bus,
+void snd_hdac_ext_stream_init(struct hdac_bus *bus,
 				struct hdac_ext_stream *stream, int idx,
 				int direction, int tag);
-int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx,
+int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx,
 		int num_stream, int dir);
-void snd_hdac_stream_free_all(struct hdac_ext_bus *ebus);
-void snd_hdac_link_free_all(struct hdac_ext_bus *ebus);
-struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_ext_bus *bus,
+void snd_hdac_stream_free_all(struct hdac_bus *bus);
+void snd_hdac_link_free_all(struct hdac_bus *bus);
+struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_bus *bus,
 					   struct snd_pcm_substream *substream,
 					   int type);
 void snd_hdac_ext_stream_release(struct hdac_ext_stream *azx_dev, int type);
-void snd_hdac_ext_stream_decouple(struct hdac_ext_bus *bus,
+void snd_hdac_ext_stream_decouple(struct hdac_bus *bus,
 				struct hdac_ext_stream *azx_dev, bool decouple);
-void snd_hdac_ext_stop_streams(struct hdac_ext_bus *sbus);
+void snd_hdac_ext_stop_streams(struct hdac_bus *bus);
 
-int snd_hdac_ext_stream_set_spib(struct hdac_ext_bus *ebus,
+int snd_hdac_ext_stream_set_spib(struct hdac_bus *bus,
 				 struct hdac_ext_stream *stream, u32 value);
-int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_ext_bus *ebus,
+int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_bus *bus,
 				 struct hdac_ext_stream *stream);
-void snd_hdac_ext_stream_drsm_enable(struct hdac_ext_bus *ebus,
+void snd_hdac_ext_stream_drsm_enable(struct hdac_bus *bus,
 				bool enable, int index);
-int snd_hdac_ext_stream_set_dpibr(struct hdac_ext_bus *ebus,
+int snd_hdac_ext_stream_set_dpibr(struct hdac_bus *bus,
 				struct hdac_ext_stream *stream, u32 value);
 int snd_hdac_ext_stream_set_lpib(struct hdac_ext_stream *stream, u32 value);
 
@@ -144,17 +122,15 @@ struct hdac_ext_link {
 
 int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link);
 int snd_hdac_ext_bus_link_power_down(struct hdac_ext_link *link);
-int snd_hdac_ext_bus_link_power_up_all(struct hdac_ext_bus *ebus);
-int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus);
+int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus);
+int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus);
 void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link,
 				 int stream);
 void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link,
 				 int stream);
 
-int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus,
-				struct hdac_ext_link *link);
-int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus,
-				struct hdac_ext_link *link);
+int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, struct hdac_ext_link *link);
+int snd_hdac_ext_bus_link_put(struct hdac_bus *bus, struct hdac_ext_link *link);
 
 /* update register macro */
 #define snd_hdac_updatel(addr, reg, mask, val)		\
@@ -181,53 +157,12 @@ struct hda_dai_map {
 	u32	maxbps;
 };
 
-#define HDA_MAX_NIDS 16
-
-/**
- * struct hdac_ext_device - HDAC Ext device
- *
- * @hdac: hdac core device
- * @nid_list - the dai map which matches the dai-name with the nid
- * @map_cur_idx - the idx in use in dai_map
- * @ops - the hda codec ops common to all codec drivers
- * @pvt_data - private data, for asoc contains asoc codec object
- */
-struct hdac_ext_device {
-	struct hdac_device hdev;
-	struct hdac_ext_bus *ebus;
-
-	/* soc-dai to nid map */
-	struct hda_dai_map nid_list[HDA_MAX_NIDS];
-	unsigned int map_cur_idx;
-
-	/* codec ops */
-	struct hdac_ext_codec_ops ops;
-
-	struct snd_card *card;
-	void *scodec;
-	void *private_data;
-};
-
 struct hdac_ext_dma_params {
 	u32 format;
 	u8 stream_tag;
 };
-#define to_ehdac_device(dev) (container_of((dev), \
-				 struct hdac_ext_device, hdev))
-/*
- * HD-audio codec base driver
- */
-struct hdac_ext_driver {
-	struct hdac_driver hdac;
-
-	int	(*probe)(struct hdac_ext_device *dev);
-	int	(*remove)(struct hdac_ext_device *dev);
-	void	(*shutdown)(struct hdac_ext_device *dev);
-};
-
-int snd_hda_ext_driver_register(struct hdac_ext_driver *drv);
-void snd_hda_ext_driver_unregister(struct hdac_ext_driver *drv);
 
-#define to_ehdac_driver(_drv) container_of(_drv, struct hdac_ext_driver, hdac)
+int snd_hda_ext_driver_register(struct hdac_driver *drv);
+void snd_hda_ext_driver_unregister(struct hdac_driver *drv);
 
 #endif /* __SOUND_HDAUDIO_EXT_H */
diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h
index 9c3db3dce32b..67561b997915 100644
--- a/include/sound/memalloc.h
+++ b/include/sound/memalloc.h
@@ -24,6 +24,8 @@
 #ifndef __SOUND_MEMALLOC_H
 #define __SOUND_MEMALLOC_H
 
+#include <asm/page.h>
+
 struct device;
 
 /*
@@ -67,6 +69,14 @@ struct snd_dma_buffer {
 	void *private_data;	/* private for allocator; don't touch */
 };
 
+/*
+ * return the pages matching with the given byte size
+ */
+static inline unsigned int snd_sgbuf_aligned_pages(size_t size)
+{
+	return (size + PAGE_SIZE - 1) >> PAGE_SHIFT;
+}
+
 #ifdef CONFIG_SND_DMA_SGBUF
 /*
  * Scatter-Gather generic device pages
@@ -91,14 +101,6 @@ struct snd_sg_buf {
 };
 
 /*
- * return the pages matching with the given byte size
- */
-static inline unsigned int snd_sgbuf_aligned_pages(size_t size)
-{
-	return (size + PAGE_SIZE - 1) >> PAGE_SHIFT;
-}
-
-/*
  * return the physical address at the corresponding offset
  */
 static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index e054c583d3b3..d6bd3caf6878 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -462,6 +462,7 @@ struct snd_pcm_substream {
         /* -- timer section -- */
 	struct snd_timer *timer;		/* timer */
 	unsigned timer_running: 1;	/* time is running */
+	long wait_time;	/* time in ms for R/W to wait for avail */
 	/* -- next substream -- */
 	struct snd_pcm_substream *next;
 	/* -- linked substreams -- */
@@ -1089,14 +1090,14 @@ static inline snd_pcm_sframes_t
 snd_pcm_lib_write(struct snd_pcm_substream *substream,
 		  const void __user *buf, snd_pcm_uframes_t frames)
 {
-	return __snd_pcm_lib_xfer(substream, (void *)buf, true, frames, false);
+	return __snd_pcm_lib_xfer(substream, (void __force *)buf, true, frames, false);
 }
 
 static inline snd_pcm_sframes_t
 snd_pcm_lib_read(struct snd_pcm_substream *substream,
 		 void __user *buf, snd_pcm_uframes_t frames)
 {
-	return __snd_pcm_lib_xfer(substream, (void *)buf, true, frames, false);
+	return __snd_pcm_lib_xfer(substream, (void __force *)buf, true, frames, false);
 }
 
 static inline snd_pcm_sframes_t
@@ -1341,8 +1342,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s
 #define snd_pcm_lib_mmap_iomem	NULL
 #endif
 
-#define snd_pcm_lib_mmap_vmalloc NULL
-
 /**
  * snd_pcm_limit_isa_dma_size - Get the max size fitting with ISA DMA transfer
  * @dma: DMA number
diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h
index c704357775fc..2dd37cada7c0 100644
--- a/include/sound/pcm_params.h
+++ b/include/sound/pcm_params.h
@@ -87,6 +87,13 @@ static inline void snd_mask_set(struct snd_mask *mask, unsigned int val)
 	mask->bits[MASK_OFS(val)] |= MASK_BIT(val);
 }
 
+/* Most of drivers need only this one */
+static inline void snd_mask_set_format(struct snd_mask *mask,
+				       snd_pcm_format_t format)
+{
+	snd_mask_set(mask, (__force unsigned int)format);
+}
+
 static inline void snd_mask_reset(struct snd_mask *mask, unsigned int val)
 {
 	mask->bits[MASK_OFS(val)] &= ~MASK_BIT(val);
@@ -369,8 +376,7 @@ static inline int params_physical_width(const struct snd_pcm_hw_params *p)
 static inline void
 params_set_format(struct snd_pcm_hw_params *p, snd_pcm_format_t fmt)
 {
-	snd_mask_set(hw_param_mask(p, SNDRV_PCM_HW_PARAM_FORMAT),
-		(__force int)fmt);
+	snd_mask_set_format(hw_param_mask(p, SNDRV_PCM_HW_PARAM_FORMAT), fmt);
 }
 
 #endif /* __SOUND_PCM_PARAMS_H */
diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h
index 63f75450d3db..6758fc12fa84 100644
--- a/include/sound/pxa2xx-lib.h
+++ b/include/sound/pxa2xx-lib.h
@@ -8,20 +8,23 @@
 /* PCM */
 struct snd_pcm_substream;
 struct snd_pcm_hw_params;
+struct snd_soc_pcm_runtime;
 struct snd_pcm;
 
-extern int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
+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);
+extern int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream);
 extern int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd);
 extern snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream);
-extern int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream);
-extern int __pxa2xx_pcm_open(struct snd_pcm_substream *substream);
-extern int __pxa2xx_pcm_close(struct snd_pcm_substream *substream);
+extern int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream);
+extern int pxa2xx_pcm_open(struct snd_pcm_substream *substream);
+extern int pxa2xx_pcm_close(struct snd_pcm_substream *substream);
 extern int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream,
 	struct vm_area_struct *vma);
 extern int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream);
 extern void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm);
+extern int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd);
+extern const struct snd_pcm_ops pxa2xx_pcm_ops;
 
 /* AC97 */
 
diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h
new file mode 100644
index 000000000000..0251797ab438
--- /dev/null
+++ b/include/sound/rt5682.h
@@ -0,0 +1,40 @@
+/*
+ * linux/sound/rt5682.h -- Platform data for RT5682
+ *
+ * Copyright 2018 Realtek Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_RT5682_H
+#define __LINUX_SND_RT5682_H
+
+enum rt5682_dmic1_data_pin {
+	RT5682_DMIC1_NULL,
+	RT5682_DMIC1_DATA_GPIO2,
+	RT5682_DMIC1_DATA_GPIO5,
+};
+
+enum rt5682_dmic1_clk_pin {
+	RT5682_DMIC1_CLK_GPIO1,
+	RT5682_DMIC1_CLK_GPIO3,
+};
+
+enum rt5682_jd_src {
+	RT5682_JD_NULL,
+	RT5682_JD1,
+};
+
+struct rt5682_platform_data {
+
+	int ldo1_en; /* GPIO for LDO1_EN */
+
+	enum rt5682_dmic1_data_pin dmic1_data_pin;
+	enum rt5682_dmic1_clk_pin dmic1_clk_pin;
+	enum rt5682_jd_src jd_src;
+};
+
+#endif
+
diff --git a/include/sound/sb16_csp.h b/include/sound/sb16_csp.h
index c7c7788005e4..7817e88bd08d 100644
--- a/include/sound/sb16_csp.h
+++ b/include/sound/sb16_csp.h
@@ -46,7 +46,7 @@ enum {
 struct snd_sb_csp_ops {
 	int (*csp_use) (struct snd_sb_csp * p);
 	int (*csp_unuse) (struct snd_sb_csp * p);
-	int (*csp_autoload) (struct snd_sb_csp * p, int pcm_sfmt, int play_rec_mode);
+	int (*csp_autoload) (struct snd_sb_csp * p, snd_pcm_format_t pcm_sfmt, int play_rec_mode);
 	int (*csp_start) (struct snd_sb_csp * p, int sample_width, int channels);
 	int (*csp_stop) (struct snd_sb_csp * p);
 	int (*csp_qsound_transfer) (struct snd_sb_csp * p);
diff --git a/include/sound/seq_midi_event.h b/include/sound/seq_midi_event.h
index e40f43e6fc7b..2f135bccf457 100644
--- a/include/sound/seq_midi_event.h
+++ b/include/sound/seq_midi_event.h
@@ -43,10 +43,8 @@ void snd_midi_event_free(struct snd_midi_event *dev);
 void snd_midi_event_reset_encode(struct snd_midi_event *dev);
 void snd_midi_event_reset_decode(struct snd_midi_event *dev);
 void snd_midi_event_no_status(struct snd_midi_event *dev, int on);
-/* encode from byte stream - return number of written bytes if success */
-long snd_midi_event_encode(struct snd_midi_event *dev, unsigned char *buf, long count,
-			   struct snd_seq_event *ev);
-int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, struct snd_seq_event *ev);
+bool snd_midi_event_encode_byte(struct snd_midi_event *dev, unsigned char c,
+				struct snd_seq_event *ev);
 /* decode from event to bytes - return number of written bytes if success */
 long snd_midi_event_decode(struct snd_midi_event *dev, unsigned char *buf, long count,
 			   struct snd_seq_event *ev);
diff --git a/include/sound/seq_virmidi.h b/include/sound/seq_virmidi.h
index 695257ae64ac..796ce7772213 100644
--- a/include/sound/seq_virmidi.h
+++ b/include/sound/seq_virmidi.h
@@ -36,11 +36,12 @@ struct snd_virmidi {
 	int seq_mode;
 	int client;
 	int port;
-	unsigned int trigger: 1;
+	bool trigger;
 	struct snd_midi_event *parser;
 	struct snd_seq_event event;
 	struct snd_virmidi_dev *rdev;
 	struct snd_rawmidi_substream *substream;
+	struct work_struct output_work;
 };
 
 #define SNDRV_VIRMIDI_SUBSCRIBE		(1<<0)
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
index 7a9710b4b799..89eafe23ef88 100644
--- a/include/sound/sh_fsi.h
+++ b/include/sound/sh_fsi.h
@@ -1,16 +1,13 @@
-#ifndef __SOUND_FSI_H
-#define __SOUND_FSI_H
-
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * Fifo-attached Serial Interface (FSI) support for SH7724
  *
  * Copyright (C) 2009 Renesas Solutions Corp.
  * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
+#ifndef __SOUND_FSI_H
+#define __SOUND_FSI_H
+
 #include <linux/clk.h>
 #include <sound/soc.h>
 
diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h
index a6a2e1547092..d264e5463f22 100644
--- a/include/sound/simple_card.h
+++ b/include/sound/simple_card.h
@@ -1,12 +1,9 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * ASoC simple sound card support
  *
  * Copyright (C) 2012 Renesas Solutions Corp.
  * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #ifndef __SIMPLE_CARD_H
diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h
index 7e25afce6566..8bc5e2d8b13c 100644
--- a/include/sound/simple_card_utils.h
+++ b/include/sound/simple_card_utils.h
@@ -1,17 +1,20 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * simple_card_utils.h
  *
  * Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
+
 #ifndef __SIMPLE_CARD_UTILS_H
 #define __SIMPLE_CARD_UTILS_H
 
 #include <sound/soc.h>
 
+#define asoc_simple_card_init_hp(card, sjack, prefix) \
+	asoc_simple_card_init_jack(card, sjack, 1, prefix)
+#define asoc_simple_card_init_mic(card, sjack, prefix) \
+	asoc_simple_card_init_jack(card, sjack, 0, prefix)
+
 struct asoc_simple_dai {
 	const char *name;
 	unsigned int sysclk;
@@ -28,6 +31,12 @@ struct asoc_simple_card_data {
 	u32 convert_channels;
 };
 
+struct asoc_simple_jack {
+	struct snd_soc_jack jack;
+	struct snd_soc_jack_pin pin;
+	struct snd_soc_jack_gpio gpio;
+};
+
 int asoc_simple_card_parse_daifmt(struct device *dev,
 				  struct device_node *node,
 				  struct device_node *codec,
@@ -107,4 +116,8 @@ int asoc_simple_card_of_parse_routing(struct snd_soc_card *card,
 int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card,
 				      char *prefix);
 
+int asoc_simple_card_init_jack(struct snd_soc_card *card,
+			       struct asoc_simple_jack *sjack,
+			       int is_hp, char *prefix);
+
 #endif /* __SIMPLE_CARD_UTILS_H */
diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h
index 9da6388c20a1..bb1d24b703fb 100644
--- a/include/sound/soc-acpi-intel-match.h
+++ b/include/sound/soc-acpi-intel-match.h
@@ -1,16 +1,6 @@
-
-/*
- * Copyright (C) 2017, Intel Corporation. All rights reserved.
- *
- * 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.
+/* SPDX-License-Identifier: GPL-2.0
  *
+ * Copyright (C) 2017, Intel Corporation. All rights reserved.
  */
 
 #ifndef __LINUX_SND_SOC_ACPI_INTEL_MATCH_H
@@ -29,5 +19,10 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_legacy_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[];
 
 #endif
diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h
index 082224275f52..e45b2330d16a 100644
--- a/include/sound/soc-acpi.h
+++ b/include/sound/soc-acpi.h
@@ -1,15 +1,6 @@
-/*
- * Copyright (C) 2013-15, Intel Corporation. All rights reserved.
- *
- * 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.
+/* SPDX-License-Identifier: GPL-2.0
  *
+ * Copyright (C) 2013-15, Intel Corporation. All rights reserved.
  */
 
 #ifndef __LINUX_SND_SOC_ACPI_H
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index e6f8c40ed43c..f5d70041108f 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -1,12 +1,9 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * linux/sound/soc-dai.h -- ALSA SoC Layer
  *
  * Copyright:	2005-2008 Wolfson Microelectronics. PLC.
  *
- * 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.
- *
  * Digital Audio Interface (DAI) API.
  */
 
@@ -141,6 +138,11 @@ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
 			     int direction);
 
+
+int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
+		unsigned int *tx_num, unsigned int *tx_slot,
+		unsigned int *rx_num, unsigned int *rx_slot);
+
 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
 
 struct snd_soc_dai_ops {
@@ -168,6 +170,9 @@ struct snd_soc_dai_ops {
 	int (*set_channel_map)(struct snd_soc_dai *dai,
 		unsigned int tx_num, unsigned int *tx_slot,
 		unsigned int rx_num, unsigned int *rx_slot);
+	int (*get_channel_map)(struct snd_soc_dai *dai,
+			unsigned int *tx_num, unsigned int *tx_slot,
+			unsigned int *rx_num, unsigned int *rx_slot);
 	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
 
 	int (*set_sdw_stream)(struct snd_soc_dai *dai,
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index a6ce2de4e20a..af9ef16cc34d 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -1,13 +1,10 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * linux/sound/soc-dapm.h -- ALSA SoC Dynamic Audio Power Management
  *
- * Author:		Liam Girdwood
- * Created:		Aug 11th 2005
+ * Author:	Liam Girdwood
+ * Created:	Aug 11th 2005
  * Copyright:	Wolfson Microelectronics. PLC.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #ifndef __LINUX_SND_SOC_DAPM_H
diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h
index 806059052bfc..9bb92f187af8 100644
--- a/include/sound/soc-dpcm.h
+++ b/include/sound/soc-dpcm.h
@@ -1,11 +1,8 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * linux/sound/soc-dpcm.h -- ALSA SoC Dynamic PCM Support
  *
  * Author:		Liam Girdwood <lrg@ti.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #ifndef __LINUX_SND_SOC_DPCM_H
diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h
index f552c3f56368..fa4b8413d2e2 100644
--- a/include/sound/soc-topology.h
+++ b/include/sound/soc-topology.h
@@ -1,13 +1,10 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * linux/sound/soc-topology.h -- ALSA SoC Firmware Controls and DAPM
  *
  * Copyright (C) 2012 Texas Instruments Inc.
  * Copyright (C) 2015 Intel Corporation.
  *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
  * Simple file API to load FW that includes mixers, coefficients, DAPM graphs,
  * algorithms, equalisers, DAIs, widgets, FE caps, BE caps, codec link caps etc.
  */
@@ -30,6 +27,9 @@ struct snd_soc_dapm_context;
 struct snd_soc_card;
 struct snd_kcontrol_new;
 struct snd_soc_dai_link;
+struct snd_soc_dai_driver;
+struct snd_soc_dai;
+struct snd_soc_dapm_route;
 
 /* object scan be loaded and unloaded in groups with identfying indexes */
 #define SND_SOC_TPLG_INDEX_ALL	0	/* ID that matches all FW objects */
@@ -109,35 +109,44 @@ struct snd_soc_tplg_widget_events {
 struct snd_soc_tplg_ops {
 
 	/* external kcontrol init - used for any driver specific init */
-	int (*control_load)(struct snd_soc_component *,
+	int (*control_load)(struct snd_soc_component *, int index,
 		struct snd_kcontrol_new *, struct snd_soc_tplg_ctl_hdr *);
 	int (*control_unload)(struct snd_soc_component *,
 		struct snd_soc_dobj *);
 
+	/* DAPM graph route element loading and unloading */
+	int (*dapm_route_load)(struct snd_soc_component *, int index,
+		struct snd_soc_dapm_route *route);
+	int (*dapm_route_unload)(struct snd_soc_component *,
+		struct snd_soc_dobj *);
+
 	/* external widget init - used for any driver specific init */
-	int (*widget_load)(struct snd_soc_component *,
+	int (*widget_load)(struct snd_soc_component *, int index,
 		struct snd_soc_dapm_widget *,
 		struct snd_soc_tplg_dapm_widget *);
-	int (*widget_ready)(struct snd_soc_component *,
+	int (*widget_ready)(struct snd_soc_component *, int index,
 		struct snd_soc_dapm_widget *,
 		struct snd_soc_tplg_dapm_widget *);
 	int (*widget_unload)(struct snd_soc_component *,
 		struct snd_soc_dobj *);
 
 	/* FE DAI - used for any driver specific init */
-	int (*dai_load)(struct snd_soc_component *,
-		struct snd_soc_dai_driver *dai_drv);
+	int (*dai_load)(struct snd_soc_component *, int index,
+		struct snd_soc_dai_driver *dai_drv,
+		struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai);
+
 	int (*dai_unload)(struct snd_soc_component *,
 		struct snd_soc_dobj *);
 
 	/* DAI link - used for any driver specific init */
-	int (*link_load)(struct snd_soc_component *,
-		struct snd_soc_dai_link *link);
+	int (*link_load)(struct snd_soc_component *, int index,
+		struct snd_soc_dai_link *link,
+		struct snd_soc_tplg_link_config *cfg);
 	int (*link_unload)(struct snd_soc_component *,
 		struct snd_soc_dobj *);
 
 	/* callback to handle vendor bespoke data */
-	int (*vendor_load)(struct snd_soc_component *,
+	int (*vendor_load)(struct snd_soc_component *, int index,
 		struct snd_soc_tplg_hdr *);
 	int (*vendor_unload)(struct snd_soc_component *,
 		struct snd_soc_tplg_hdr *);
@@ -146,7 +155,7 @@ struct snd_soc_tplg_ops {
 	void (*complete)(struct snd_soc_component *);
 
 	/* manifest - optional to inform component of manifest */
-	int (*manifest)(struct snd_soc_component *,
+	int (*manifest)(struct snd_soc_component *, int index,
 		struct snd_soc_tplg_manifest *);
 
 	/* vendor specific kcontrol handlers available for binding */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 1378dcd2128a..41cec42fb456 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1,13 +1,10 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * linux/sound/soc.h -- ALSA SoC Layer
  *
- * Author:		Liam Girdwood
- * Created:		Aug 11th 2005
+ * Author:	Liam Girdwood
+ * Created:	Aug 11th 2005
  * Copyright:	Wolfson Microelectronics. PLC.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #ifndef __LINUX_SND_SOC_H
@@ -806,6 +803,14 @@ struct snd_soc_component_driver {
 	unsigned int use_pmdown_time:1; /* care pmdown_time at stop */
 	unsigned int endianness:1;
 	unsigned int non_legacy_dai_naming:1;
+
+	/* this component uses topology and ignore machine driver FEs */
+	const char *ignore_machine;
+	const char *topology_name_prefix;
+	int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd,
+				  struct snd_pcm_hw_params *params);
+	bool use_dai_pcm_id;	/* use the DAI link PCM ID as PCM device number */
+	int be_pcm_base;	/* base device ID for all BE PCMs */
 };
 
 struct snd_soc_component {
@@ -957,10 +962,17 @@ struct snd_soc_dai_link {
 
 	/* DPCM used FE & BE merged format */
 	unsigned int dpcm_merged_format:1;
+	/* DPCM used FE & BE merged channel */
+	unsigned int dpcm_merged_chan:1;
+	/* DPCM used FE & BE merged rate */
+	unsigned int dpcm_merged_rate:1;
 
 	/* pmdown_time is ignored at stop */
 	unsigned int ignore_pmdown_time:1;
 
+	/* Do not create a PCM for this DAI link (Backend link) */
+	unsigned int ignore:1;
+
 	struct list_head list; /* DAI link list of the soc card */
 	struct snd_soc_dobj dobj; /* For topology */
 };
@@ -1000,6 +1012,7 @@ struct snd_soc_card {
 	const char *long_name;
 	const char *driver_name;
 	char dmi_longname[80];
+	char topology_shortname[32];
 
 	struct device *dev;
 	struct snd_card *snd_card;
@@ -1009,6 +1022,7 @@ struct snd_soc_card {
 	struct mutex dapm_mutex;
 
 	bool instantiated;
+	bool topology_shortname_created;
 
 	int (*probe)(struct snd_soc_card *card);
 	int (*late_probe)(struct snd_soc_card *card);
@@ -1412,6 +1426,9 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card,
 			       const char *propname);
 int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
 					  const char *propname);
+int snd_soc_of_get_slot_mask(struct device_node *np,
+			     const char *prop_name,
+			     unsigned int *mask);
 int snd_soc_of_parse_tdm_slot(struct device_node *np,
 			      unsigned int *tx_mask,
 			      unsigned int *rx_mask,
diff --git a/include/trace/events/clk.h b/include/trace/events/clk.h
index 2cd449328aee..9004ffff7f32 100644
--- a/include/trace/events/clk.h
+++ b/include/trace/events/clk.h
@@ -192,6 +192,42 @@ DEFINE_EVENT(clk_phase, clk_set_phase_complete,
 	TP_ARGS(core, phase)
 );
 
+DECLARE_EVENT_CLASS(clk_duty_cycle,
+
+	TP_PROTO(struct clk_core *core, struct clk_duty *duty),
+
+	TP_ARGS(core, duty),
+
+	TP_STRUCT__entry(
+		__string(        name,           core->name              )
+		__field( unsigned int,           num                     )
+		__field( unsigned int,           den                     )
+	),
+
+	TP_fast_assign(
+		__assign_str(name, core->name);
+		__entry->num = duty->num;
+		__entry->den = duty->den;
+	),
+
+	TP_printk("%s %u/%u", __get_str(name), (unsigned int)__entry->num,
+		  (unsigned int)__entry->den)
+);
+
+DEFINE_EVENT(clk_duty_cycle, clk_set_duty_cycle,
+
+	TP_PROTO(struct clk_core *core, struct clk_duty *duty),
+
+	TP_ARGS(core, duty)
+);
+
+DEFINE_EVENT(clk_duty_cycle, clk_set_duty_cycle_complete,
+
+	TP_PROTO(struct clk_core *core, struct clk_duty *duty),
+
+	TP_ARGS(core, duty)
+);
+
 #endif /* _TRACE_CLK_H */
 
 /* This part must be outside protection */
diff --git a/include/uapi/linux/usb/audio.h b/include/uapi/linux/usb/audio.h
index 74e520fb944f..ddc5396800aa 100644
--- a/include/uapi/linux/usb/audio.h
+++ b/include/uapi/linux/usb/audio.h
@@ -390,33 +390,64 @@ static inline __u8 uac_processing_unit_iChannelNames(struct uac_processing_unit_
 static inline __u8 uac_processing_unit_bControlSize(struct uac_processing_unit_descriptor *desc,
 						    int protocol)
 {
-	return (protocol == UAC_VERSION_1) ?
-		desc->baSourceID[desc->bNrInPins + 4] :
-		2; /* in UAC2, this value is constant */
+	switch (protocol) {
+	case UAC_VERSION_1:
+		return desc->baSourceID[desc->bNrInPins + 4];
+	case UAC_VERSION_2:
+		return 2; /* in UAC2, this value is constant */
+	case UAC_VERSION_3:
+		return 4; /* in UAC3, this value is constant */
+	default:
+		return 1;
+	}
 }
 
 static inline __u8 *uac_processing_unit_bmControls(struct uac_processing_unit_descriptor *desc,
 						   int protocol)
 {
-	return (protocol == UAC_VERSION_1) ?
-		&desc->baSourceID[desc->bNrInPins + 5] :
-		&desc->baSourceID[desc->bNrInPins + 6];
+	switch (protocol) {
+	case UAC_VERSION_1:
+		return &desc->baSourceID[desc->bNrInPins + 5];
+	case UAC_VERSION_2:
+		return &desc->baSourceID[desc->bNrInPins + 6];
+	case UAC_VERSION_3:
+		return &desc->baSourceID[desc->bNrInPins + 2];
+	default:
+		return NULL;
+	}
 }
 
 static inline __u8 uac_processing_unit_iProcessing(struct uac_processing_unit_descriptor *desc,
 						   int protocol)
 {
 	__u8 control_size = uac_processing_unit_bControlSize(desc, protocol);
-	return *(uac_processing_unit_bmControls(desc, protocol)
-			+ control_size);
+
+	switch (protocol) {
+	case UAC_VERSION_1:
+	case UAC_VERSION_2:
+	default:
+		return *(uac_processing_unit_bmControls(desc, protocol)
+			 + control_size);
+	case UAC_VERSION_3:
+		return 0; /* UAC3 does not have this field */
+	}
 }
 
 static inline __u8 *uac_processing_unit_specific(struct uac_processing_unit_descriptor *desc,
 						 int protocol)
 {
 	__u8 control_size = uac_processing_unit_bControlSize(desc, protocol);
-	return uac_processing_unit_bmControls(desc, protocol)
+
+	switch (protocol) {
+	case UAC_VERSION_1:
+	case UAC_VERSION_2:
+	default:
+		return uac_processing_unit_bmControls(desc, protocol)
 			+ control_size + 1;
+	case UAC_VERSION_3:
+		return uac_processing_unit_bmControls(desc, protocol)
+			+ control_size;
+	}
 }
 
 /* 4.5.2 Class-Specific AS Interface Descriptor */
diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c
index 31f858eceffc..7a0dfca03a57 100644
--- a/sound/ac97/bus.c
+++ b/sound/ac97/bus.c
@@ -13,6 +13,7 @@
 #include <linux/idr.h>
 #include <linux/list.h>
 #include <linux/mutex.h>
+#include <linux/of.h>
 #include <linux/pm.h>
 #include <linux/pm_runtime.h>
 #include <linux/slab.h>
@@ -68,6 +69,27 @@ ac97_codec_find(struct ac97_controller *ac97_ctrl, unsigned int codec_num)
 	return ac97_ctrl->codecs[codec_num];
 }
 
+static struct device_node *
+ac97_of_get_child_device(struct ac97_controller *ac97_ctrl, int idx,
+			 unsigned int vendor_id)
+{
+	struct device_node *node;
+	u32 reg;
+	char compat[] = "ac97,0000,0000";
+
+	snprintf(compat, sizeof(compat), "ac97,%04x,%04x",
+		 vendor_id >> 16, vendor_id & 0xffff);
+
+	for_each_child_of_node(ac97_ctrl->parent->of_node, node) {
+		if ((idx != of_property_read_u32(node, "reg", &reg)) ||
+		    !of_device_is_compatible(node, compat))
+			continue;
+		return of_node_get(node);
+	}
+
+	return NULL;
+}
+
 static void ac97_codec_release(struct device *dev)
 {
 	struct ac97_codec_device *adev;
@@ -76,6 +98,7 @@ static void ac97_codec_release(struct device *dev)
 	adev = to_ac97_device(dev);
 	ac97_ctrl = adev->ac97_ctrl;
 	ac97_ctrl->codecs[adev->num] = NULL;
+	of_node_put(dev->of_node);
 	kfree(adev);
 }
 
@@ -98,6 +121,8 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx,
 
 	device_initialize(&codec->dev);
 	dev_set_name(&codec->dev, "%s:%u", dev_name(ac97_ctrl->parent), idx);
+	codec->dev.of_node = ac97_of_get_child_device(ac97_ctrl, idx,
+						      vendor_id);
 
 	ret = device_add(&codec->dev);
 	if (ret)
@@ -105,6 +130,7 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx,
 
 	return 0;
 err_free_codec:
+	of_node_put(codec->dev.of_node);
 	put_device(&codec->dev);
 	kfree(codec);
 	ac97_ctrl->codecs[idx] = NULL;
diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c
index 71960089e207..65557421fe0b 100644
--- a/sound/aoa/core/gpio-feature.c
+++ b/sound/aoa/core/gpio-feature.c
@@ -88,8 +88,10 @@ static struct device_node *get_gpio(char *name,
 	}
 
 	reg = of_get_property(np, "reg", NULL);
-	if (!reg)
+	if (!reg) {
+		of_node_put(np);
 		return NULL;
+	}
 
 	*gpioptr = *reg;
 
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index 65171f6657a2..5fbd47a9177e 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -17,14 +17,9 @@ config SND_ARMAACI
 	select SND_PCM
 	select SND_AC97_CODEC
 
-config SND_PXA2XX_PCM
-	tristate
-	select SND_PCM
-
 config SND_PXA2XX_AC97
 	tristate "AC97 driver for the Intel PXA2xx chip"
 	depends on ARCH_PXA
-	select SND_PXA2XX_PCM
 	select SND_AC97_CODEC
 	select SND_PXA2XX_LIB
 	select SND_PXA2XX_LIB_AC97
diff --git a/sound/arm/Makefile b/sound/arm/Makefile
index e10d5b169565..34c769489877 100644
--- a/sound/arm/Makefile
+++ b/sound/arm/Makefile
@@ -6,9 +6,6 @@
 obj-$(CONFIG_SND_ARMAACI)	+= snd-aaci.o
 snd-aaci-objs			:= aaci.o
 
-obj-$(CONFIG_SND_PXA2XX_PCM)	+= snd-pxa2xx-pcm.o
-snd-pxa2xx-pcm-objs		:= pxa2xx-pcm.o
-
 obj-$(CONFIG_SND_PXA2XX_LIB)	+= snd-pxa2xx-lib.o
 snd-pxa2xx-lib-y		:= pxa2xx-pcm-lib.o
 snd-pxa2xx-lib-$(CONFIG_SND_PXA2XX_LIB_AC97)	+= pxa2xx-ac97-lib.o
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 5950a9e218d9..8eafd3d3dff6 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -19,6 +19,7 @@
 #include <linux/module.h>
 #include <linux/io.h>
 #include <linux/gpio.h>
+#include <linux/of_gpio.h>
 
 #include <sound/pxa2xx-lib.h>
 
@@ -337,6 +338,17 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev)
 			dev_err(&dev->dev, "Invalid reset GPIO %d\n",
 				pdata->reset_gpio);
 		}
+	} else if (!pdata && dev->dev.of_node) {
+		pdata = devm_kzalloc(&dev->dev, sizeof(*pdata), GFP_KERNEL);
+		if (!pdata)
+			return -ENOMEM;
+		pdata->reset_gpio = of_get_named_gpio(dev->dev.of_node,
+						      "reset-gpios", 0);
+		if (pdata->reset_gpio == -ENOENT)
+			pdata->reset_gpio = -1;
+		else if (pdata->reset_gpio < 0)
+			return pdata->reset_gpio;
+		reset_gpio = pdata->reset_gpio;
 	} else {
 		if (cpu_is_pxa27x())
 			reset_gpio = 113;
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 4bc244c40f80..1f72672262d0 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -15,7 +15,7 @@
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/dmaengine.h>
-#include <linux/dma/pxa-dma.h>
+#include <linux/dma-mapping.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -27,8 +27,6 @@
 #include <mach/regs-ac97.h>
 #include <mach/audio.h>
 
-#include "pxa2xx-pcm.h"
-
 static void pxa2xx_ac97_legacy_reset(struct snd_ac97 *ac97)
 {
 	if (!pxa2xx_ac97_try_cold_reset())
@@ -63,61 +61,46 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
 	.reset	= pxa2xx_ac97_legacy_reset,
 };
 
-static struct pxad_param pxa2xx_ac97_pcm_out_req = {
-	.prio = PXAD_PRIO_LOWEST,
-	.drcmr = 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 pxad_param pxa2xx_ac97_pcm_in_req = {
-	.prio = PXAD_PRIO_LOWEST,
-	.drcmr = 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;
 static struct snd_ac97 *pxa2xx_ac97_ac97;
 
-static int pxa2xx_ac97_pcm_startup(struct snd_pcm_substream *substream)
+static int pxa2xx_ac97_pcm_open(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	pxa2xx_audio_ops_t *platform_ops;
-	int r;
+	int ret, i;
+
+	ret = pxa2xx_pcm_open(substream);
+	if (ret)
+		return ret;
 
 	runtime->hw.channels_min = 2;
 	runtime->hw.channels_max = 2;
 
-	r = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
-	    AC97_RATES_FRONT_DAC : AC97_RATES_ADC;
-	runtime->hw.rates = pxa2xx_ac97_ac97->rates[r];
+	i = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+		AC97_RATES_FRONT_DAC : AC97_RATES_ADC;
+	runtime->hw.rates = pxa2xx_ac97_ac97->rates[i];
 	snd_pcm_limit_hw_rates(runtime);
 
-       	platform_ops = substream->pcm->card->dev->platform_data;
-	if (platform_ops && platform_ops->startup)
-		return platform_ops->startup(substream, platform_ops->priv);
-	else
-		return 0;
+	platform_ops = substream->pcm->card->dev->platform_data;
+	if (platform_ops && platform_ops->startup) {
+		ret = platform_ops->startup(substream, platform_ops->priv);
+		if (ret < 0)
+			pxa2xx_pcm_close(substream);
+	}
+
+	return ret;
 }
 
-static void pxa2xx_ac97_pcm_shutdown(struct snd_pcm_substream *substream)
+static int pxa2xx_ac97_pcm_close(struct snd_pcm_substream *substream)
 {
 	pxa2xx_audio_ops_t *platform_ops;
 
-       	platform_ops = substream->pcm->card->dev->platform_data;
+	platform_ops = substream->pcm->card->dev->platform_data;
 	if (platform_ops && platform_ops->shutdown)
 		platform_ops->shutdown(substream, platform_ops->priv);
+
+	return 0;
 }
 
 static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream)
@@ -125,17 +108,15 @@ static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
 		  AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE;
+	int ret;
+
+	ret = pxa2xx_pcm_prepare(substream);
+	if (ret < 0)
+		return ret;
+
 	return snd_ac97_set_rate(pxa2xx_ac97_ac97, reg, runtime->rate);
 }
 
-static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = {
-	.playback_params	= &pxa2xx_ac97_pcm_out,
-	.capture_params		= &pxa2xx_ac97_pcm_in,
-	.startup		= pxa2xx_ac97_pcm_startup,
-	.shutdown		= pxa2xx_ac97_pcm_shutdown,
-	.prepare		= pxa2xx_ac97_pcm_prepare,
-};
-
 #ifdef CONFIG_PM_SLEEP
 
 static int pxa2xx_ac97_do_suspend(struct snd_card *card)
@@ -193,6 +174,53 @@ static int pxa2xx_ac97_resume(struct device *dev)
 static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume);
 #endif
 
+static const struct snd_pcm_ops pxa2xx_ac97_pcm_ops = {
+	.open		= pxa2xx_ac97_pcm_open,
+	.close		= pxa2xx_ac97_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= pxa2xx_pcm_hw_params,
+	.hw_free	= pxa2xx_pcm_hw_free,
+	.prepare	= pxa2xx_ac97_pcm_prepare,
+	.trigger	= pxa2xx_pcm_trigger,
+	.pointer	= pxa2xx_pcm_pointer,
+	.mmap		= pxa2xx_pcm_mmap,
+};
+
+
+static int pxa2xx_ac97_pcm_new(struct snd_card *card)
+{
+	struct snd_pcm *pcm;
+	int stream, ret;
+
+	ret = snd_pcm_new(card, "PXA2xx-PCM", 0, 1, 1, &pcm);
+	if (ret)
+		goto out;
+
+	pcm->private_free = pxa2xx_pcm_free_dma_buffers;
+
+	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+	if (ret)
+		goto out;
+
+	stream = SNDRV_PCM_STREAM_PLAYBACK;
+	snd_pcm_set_ops(pcm, stream, &pxa2xx_ac97_pcm_ops);
+	ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream);
+	if (ret)
+		goto out;
+
+	stream = SNDRV_PCM_STREAM_CAPTURE;
+	snd_pcm_set_ops(pcm, stream, &pxa2xx_ac97_pcm_ops);
+	ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream);
+	if (ret)
+		goto out;
+
+	pxa2xx_ac97_pcm = pcm;
+	ret = 0;
+
+ out:
+	return ret;
+}
+
 static int pxa2xx_ac97_probe(struct platform_device *dev)
 {
 	struct snd_card *card;
@@ -214,7 +242,7 @@ static int pxa2xx_ac97_probe(struct platform_device *dev)
 
 	strlcpy(card->driver, dev->dev.driver->name, sizeof(card->driver));
 
-	ret = pxa2xx_pcm_new(card, &pxa2xx_ac97_pcm_client, &pxa2xx_ac97_pcm);
+	ret = pxa2xx_ac97_pcm_new(card);
 	if (ret)
 		goto err;
 
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index e8da3b8ee721..7931789d4a9f 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -16,8 +16,6 @@
 #include <sound/pxa2xx-lib.h>
 #include <sound/dmaengine_pcm.h>
 
-#include "pxa2xx-pcm.h"
-
 static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
@@ -25,8 +23,8 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
 				  SNDRV_PCM_INFO_PAUSE |
 				  SNDRV_PCM_INFO_RESUME,
 	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
-					SNDRV_PCM_FMTBIT_S24_LE |
-					SNDRV_PCM_FMTBIT_S32_LE,
+				  SNDRV_PCM_FMTBIT_S24_LE |
+				  SNDRV_PCM_FMTBIT_S32_LE,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 8192 - 32,
 	.periods_min		= 1,
@@ -35,8 +33,8 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
 	.fifo_size		= 32,
 };
 
-int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *params)
 {
 	struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -64,14 +62,14 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 
 	return 0;
 }
-EXPORT_SYMBOL(__pxa2xx_pcm_hw_params);
+EXPORT_SYMBOL(pxa2xx_pcm_hw_params);
 
-int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
+int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	snd_pcm_set_runtime_buffer(substream, NULL);
 	return 0;
 }
-EXPORT_SYMBOL(__pxa2xx_pcm_hw_free);
+EXPORT_SYMBOL(pxa2xx_pcm_hw_free);
 
 int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 {
@@ -86,13 +84,13 @@ pxa2xx_pcm_pointer(struct snd_pcm_substream *substream)
 }
 EXPORT_SYMBOL(pxa2xx_pcm_pointer);
 
-int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
+int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
 {
 	return 0;
 }
-EXPORT_SYMBOL(__pxa2xx_pcm_prepare);
+EXPORT_SYMBOL(pxa2xx_pcm_prepare);
 
-int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
+int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_pcm_runtime *runtime = substream->runtime;
@@ -125,17 +123,17 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
 	if (ret < 0)
 		return ret;
 
-	return snd_dmaengine_pcm_open_request_chan(substream,
-					pxad_filter_fn,
-					dma_params->filter_data);
+	return snd_dmaengine_pcm_open(
+		substream, dma_request_slave_channel(rtd->cpu_dai->dev,
+						     dma_params->chan_name));
 }
-EXPORT_SYMBOL(__pxa2xx_pcm_open);
+EXPORT_SYMBOL(pxa2xx_pcm_open);
 
-int __pxa2xx_pcm_close(struct snd_pcm_substream *substream)
+int pxa2xx_pcm_close(struct snd_pcm_substream *substream)
 {
 	return snd_dmaengine_pcm_close_release_chan(substream);
 }
-EXPORT_SYMBOL(__pxa2xx_pcm_close);
+EXPORT_SYMBOL(pxa2xx_pcm_close);
 
 int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream,
 	struct vm_area_struct *vma)
@@ -181,6 +179,47 @@ void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 }
 EXPORT_SYMBOL(pxa2xx_pcm_free_dma_buffers);
 
+int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_pcm *pcm = rtd->pcm;
+	int ret;
+
+	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+	if (ret)
+		return ret;
+
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_PLAYBACK);
+		if (ret)
+			goto out;
+	}
+
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_CAPTURE);
+		if (ret)
+			goto out;
+	}
+ out:
+	return ret;
+}
+EXPORT_SYMBOL(pxa2xx_soc_pcm_new);
+
+const struct snd_pcm_ops pxa2xx_pcm_ops = {
+	.open		= pxa2xx_pcm_open,
+	.close		= pxa2xx_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= pxa2xx_pcm_hw_params,
+	.hw_free	= pxa2xx_pcm_hw_free,
+	.prepare	= pxa2xx_pcm_prepare,
+	.trigger	= pxa2xx_pcm_trigger,
+	.pointer	= pxa2xx_pcm_pointer,
+	.mmap		= pxa2xx_pcm_mmap,
+};
+EXPORT_SYMBOL(pxa2xx_pcm_ops);
+
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("Intel PXA2xx sound library");
 MODULE_LICENSE("GPL");
diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c
deleted file mode 100644
index 1c6f4b436de3..000000000000
--- a/sound/arm/pxa2xx-pcm.c
+++ /dev/null
@@ -1,129 +0,0 @@
-/*
- * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip
- *
- * Author:	Nicolas Pitre
- * Created:	Nov 30, 2004
- * Copyright:	(C) 2004 MontaVista Software, Inc.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#include <linux/module.h>
-#include <linux/dma-mapping.h>
-#include <linux/dmaengine.h>
-
-#include <mach/dma.h>
-
-#include <sound/core.h>
-#include <sound/pxa2xx-lib.h>
-#include <sound/dmaengine_pcm.h>
-
-#include "pxa2xx-pcm.h"
-
-static int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
-{
-	struct pxa2xx_pcm_client *client = substream->private_data;
-
-	__pxa2xx_pcm_prepare(substream);
-
-	return client->prepare(substream);
-}
-
-static int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
-{
-	struct pxa2xx_pcm_client *client = substream->private_data;
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct pxa2xx_runtime_data *rtd;
-	int ret;
-
-	ret = __pxa2xx_pcm_open(substream);
-	if (ret)
-		goto out;
-
-	rtd = runtime->private_data;
-
-	rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
-		      client->playback_params : client->capture_params;
-
-	ret = client->startup(substream);
-	if (!ret)
-		goto err2;
-
-	return 0;
-
- err2:
-	__pxa2xx_pcm_close(substream);
- out:
-	return ret;
-}
-
-static int pxa2xx_pcm_close(struct snd_pcm_substream *substream)
-{
-	struct pxa2xx_pcm_client *client = substream->private_data;
-
-	client->shutdown(substream);
-
-	return __pxa2xx_pcm_close(substream);
-}
-
-static const struct snd_pcm_ops pxa2xx_pcm_ops = {
-	.open		= pxa2xx_pcm_open,
-	.close		= pxa2xx_pcm_close,
-	.ioctl		= snd_pcm_lib_ioctl,
-	.hw_params	= __pxa2xx_pcm_hw_params,
-	.hw_free	= __pxa2xx_pcm_hw_free,
-	.prepare	= pxa2xx_pcm_prepare,
-	.trigger	= pxa2xx_pcm_trigger,
-	.pointer	= pxa2xx_pcm_pointer,
-	.mmap		= pxa2xx_pcm_mmap,
-};
-
-int pxa2xx_pcm_new(struct snd_card *card, struct pxa2xx_pcm_client *client,
-		   struct snd_pcm **rpcm)
-{
-	struct snd_pcm *pcm;
-	int play = client->playback_params ? 1 : 0;
-	int capt = client->capture_params ? 1 : 0;
-	int ret;
-
-	ret = snd_pcm_new(card, "PXA2xx-PCM", 0, play, capt, &pcm);
-	if (ret)
-		goto out;
-
-	pcm->private_data = client;
-	pcm->private_free = pxa2xx_pcm_free_dma_buffers;
-
-	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
-	if (ret)
-		goto out;
-
-	if (play) {
-		int stream = SNDRV_PCM_STREAM_PLAYBACK;
-		snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops);
-		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream);
-		if (ret)
-			goto out;
-	}
-	if (capt) {
-		int stream = SNDRV_PCM_STREAM_CAPTURE;
-		snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops);
-		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream);
-		if (ret)
-			goto out;
-	}
-
-	if (rpcm)
-		*rpcm = pcm;
-	ret = 0;
-
- out:
-	return ret;
-}
-
-EXPORT_SYMBOL(pxa2xx_pcm_new);
-
-MODULE_AUTHOR("Nicolas Pitre");
-MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
-MODULE_LICENSE("GPL");
diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h
deleted file mode 100644
index 8fa2b7c9e6b8..000000000000
--- a/sound/arm/pxa2xx-pcm.h
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip
- *
- * Author:	Nicolas Pitre
- * Created:	Nov 30, 2004
- * Copyright:	MontaVista Software, Inc.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-struct pxa2xx_runtime_data {
-	int dma_ch;
-	struct snd_dmaengine_dai_dma_data *params;
-};
-
-struct pxa2xx_pcm_client {
-	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 *);
-};
-
-extern int pxa2xx_pcm_new(struct snd_card *, struct pxa2xx_pcm_client *, struct snd_pcm **);
-
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 4b01a37c836e..26b5e245b074 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -1160,18 +1160,6 @@ int snd_compress_deregister(struct snd_compr *device)
 }
 EXPORT_SYMBOL_GPL(snd_compress_deregister);
 
-static int __init snd_compress_init(void)
-{
-	return 0;
-}
-
-static void __exit snd_compress_exit(void)
-{
-}
-
-module_init(snd_compress_init);
-module_exit(snd_compress_exit);
-
 MODULE_DESCRIPTION("ALSA Compressed offload framework");
 MODULE_AUTHOR("Vinod Koul <vinod.koul@linux.intel.com>");
 MODULE_LICENSE("GPL v2");
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 7f89d3c79a4b..753d5fc4b284 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -242,16 +242,12 @@ int snd_dma_alloc_pages_fallback(int type, struct device *device, size_t size,
 	int err;
 
 	while ((err = snd_dma_alloc_pages(type, device, size, dmab)) < 0) {
-		size_t aligned_size;
 		if (err != -ENOMEM)
 			return err;
 		if (size <= PAGE_SIZE)
 			return -ENOMEM;
-		aligned_size = PAGE_SIZE << get_order(size);
-		if (size != aligned_size)
-			size = aligned_size;
-		else
-			size >>= 1;
+		size >>= 1;
+		size = PAGE_SIZE << get_order(size);
 	}
 	if (! dmab->area)
 		return -ENOMEM;
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 905a53c1cde5..f8d4a419f3af 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -1851,7 +1851,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file)
 	format_mask = hw_param_mask_c(params, SNDRV_PCM_HW_PARAM_FORMAT);
 	for (fmt = 0; fmt < 32; ++fmt) {
 		if (snd_mask_test(format_mask, fmt)) {
-			int f = snd_pcm_oss_format_to(fmt);
+			int f = snd_pcm_oss_format_to((__force snd_pcm_format_t)fmt);
 			if (f >= 0)
 				formats |= f;
 		}
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index 85a56af104bd..0391cb1a4f19 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -281,10 +281,10 @@ static int snd_pcm_plug_formats(const struct snd_mask *mask,
 		       SNDRV_PCM_FMTBIT_U32_BE | SNDRV_PCM_FMTBIT_S32_BE);
 	snd_mask_set(&formats, (__force int)SNDRV_PCM_FORMAT_MU_LAW);
 	
-	if (formats.bits[0] & (u32)linfmts)
-		formats.bits[0] |= (u32)linfmts;
-	if (formats.bits[1] & (u32)(linfmts >> 32))
-		formats.bits[1] |= (u32)(linfmts >> 32);
+	if (formats.bits[0] & lower_32_bits(linfmts))
+		formats.bits[0] |= lower_32_bits(linfmts);
+	if (formats.bits[1] & upper_32_bits(linfmts))
+		formats.bits[1] |= upper_32_bits(linfmts);
 	return snd_mask_test(&formats, (__force int)format);
 }
 
@@ -353,6 +353,7 @@ snd_pcm_format_t snd_pcm_plug_slave_format(snd_pcm_format_t format,
 				if (snd_mask_test(format_mask, (__force int)format1))
 					return format1;
 			}
+			/* fall through */
 		default:
 			return (__force snd_pcm_format_t)-EINVAL;
 		}
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index c352bfb973cc..fdb9b92fc8d6 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -492,13 +492,8 @@ static void snd_pcm_xrun_injection_write(struct snd_info_entry *entry,
 					 struct snd_info_buffer *buffer)
 {
 	struct snd_pcm_substream *substream = entry->private_data;
-	struct snd_pcm_runtime *runtime;
 
-	snd_pcm_stream_lock_irq(substream);
-	runtime = substream->runtime;
-	if (runtime && runtime->status->state == SNDRV_PCM_STATE_RUNNING)
-		snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-	snd_pcm_stream_unlock_irq(substream);
+	snd_pcm_stop_xrun(substream);
 }
 
 static void snd_pcm_xrun_debug_read(struct snd_info_entry *entry,
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 44b5ae833082..4e6110d778bd 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -153,7 +153,8 @@ EXPORT_SYMBOL(snd_pcm_debug_name);
 			dump_stack();				\
 	} while (0)
 
-static void xrun(struct snd_pcm_substream *substream)
+/* call with stream lock held */
+void __snd_pcm_xrun(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
@@ -201,7 +202,7 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream,
 		}
 	} else {
 		if (avail >= runtime->stop_threshold) {
-			xrun(substream);
+			__snd_pcm_xrun(substream);
 			return -EPIPE;
 		}
 	}
@@ -297,7 +298,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 	}
 
 	if (pos == SNDRV_PCM_POS_XRUN) {
-		xrun(substream);
+		__snd_pcm_xrun(substream);
 		return -EPIPE;
 	}
 	if (pos >= runtime->buffer_size) {
@@ -626,27 +627,33 @@ EXPORT_SYMBOL(snd_interval_refine);
 
 static int snd_interval_refine_first(struct snd_interval *i)
 {
+	const unsigned int last_max = i->max;
+
 	if (snd_BUG_ON(snd_interval_empty(i)))
 		return -EINVAL;
 	if (snd_interval_single(i))
 		return 0;
 	i->max = i->min;
-	i->openmax = i->openmin;
-	if (i->openmax)
+	if (i->openmin)
 		i->max++;
+	/* only exclude max value if also excluded before refine */
+	i->openmax = (i->openmax && i->max >= last_max);
 	return 1;
 }
 
 static int snd_interval_refine_last(struct snd_interval *i)
 {
+	const unsigned int last_min = i->min;
+
 	if (snd_BUG_ON(snd_interval_empty(i)))
 		return -EINVAL;
 	if (snd_interval_single(i))
 		return 0;
 	i->min = i->max;
-	i->openmin = i->openmax;
-	if (i->openmin)
+	if (i->openmax)
 		i->min--;
+	/* only exclude min value if also excluded before refine */
+	i->openmin = (i->openmin && i->min <= last_min);
 	return 1;
 }
 
@@ -1832,12 +1839,19 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 	if (runtime->no_period_wakeup)
 		wait_time = MAX_SCHEDULE_TIMEOUT;
 	else {
-		wait_time = 10;
-		if (runtime->rate) {
-			long t = runtime->period_size * 2 / runtime->rate;
-			wait_time = max(t, wait_time);
+		/* use wait time from substream if available */
+		if (substream->wait_time) {
+			wait_time = substream->wait_time;
+		} else {
+			wait_time = 10;
+
+			if (runtime->rate) {
+				long t = runtime->period_size * 2 /
+					 runtime->rate;
+				wait_time = max(t, wait_time);
+			}
+			wait_time = msecs_to_jiffies(wait_time * 1000);
 		}
-		wait_time = msecs_to_jiffies(wait_time * 1000);
 	}
 
 	for (;;) {
diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h
index 7a499d02df6c..c515612969a4 100644
--- a/sound/core/pcm_local.h
+++ b/sound/core/pcm_local.h
@@ -65,4 +65,6 @@ static inline void snd_pcm_timer_init(struct snd_pcm_substream *substream) {}
 static inline void snd_pcm_timer_done(struct snd_pcm_substream *substream) {}
 #endif
 
+void __snd_pcm_xrun(struct snd_pcm_substream *substream);
+
 #endif	/* __SOUND_CORE_PCM_LOCAL_H */
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index cecc79772c94..66c90f486af9 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1337,13 +1337,12 @@ int snd_pcm_drain_done(struct snd_pcm_substream *substream)
 int snd_pcm_stop_xrun(struct snd_pcm_substream *substream)
 {
 	unsigned long flags;
-	int ret = 0;
 
 	snd_pcm_stream_lock_irqsave(substream, flags);
-	if (snd_pcm_running(substream))
-		ret = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+	if (substream->runtime && snd_pcm_running(substream))
+		__snd_pcm_xrun(substream);
 	snd_pcm_stream_unlock_irqrestore(substream, flags);
-	return ret;
+	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_pcm_stop_xrun);
 
@@ -1591,7 +1590,8 @@ static int snd_pcm_xrun(struct snd_pcm_substream *substream)
 		result = 0;	/* already there */
 		break;
 	case SNDRV_PCM_STATE_RUNNING:
-		result = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+		__snd_pcm_xrun(substream);
+		result = 0;
 		break;
 	default:
 		result = -EBADFD;
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index b53026a72e73..69517e18ef07 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -29,6 +29,7 @@
 #include <linux/mutex.h>
 #include <linux/module.h>
 #include <linux/delay.h>
+#include <linux/mm.h>
 #include <sound/rawmidi.h>
 #include <sound/info.h>
 #include <sound/control.h>
@@ -88,6 +89,7 @@ static inline unsigned short snd_rawmidi_file_flags(struct file *file)
 static inline int snd_rawmidi_ready(struct snd_rawmidi_substream *substream)
 {
 	struct snd_rawmidi_runtime *runtime = substream->runtime;
+
 	return runtime->avail >= runtime->avail_min;
 }
 
@@ -95,6 +97,7 @@ static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substre
 					   size_t count)
 {
 	struct snd_rawmidi_runtime *runtime = substream->runtime;
+
 	return runtime->avail >= runtime->avail_min &&
 	       (!substream->append || runtime->avail >= count);
 }
@@ -103,6 +106,7 @@ static void snd_rawmidi_input_event_work(struct work_struct *work)
 {
 	struct snd_rawmidi_runtime *runtime =
 		container_of(work, struct snd_rawmidi_runtime, event_work);
+
 	if (runtime->event)
 		runtime->event(runtime->substream);
 }
@@ -111,7 +115,8 @@ static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream)
 {
 	struct snd_rawmidi_runtime *runtime;
 
-	if ((runtime = kzalloc(sizeof(*runtime), GFP_KERNEL)) == NULL)
+	runtime = kzalloc(sizeof(*runtime), GFP_KERNEL);
+	if (!runtime)
 		return -ENOMEM;
 	runtime->substream = substream;
 	spin_lock_init(&runtime->lock);
@@ -124,7 +129,8 @@ static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream)
 		runtime->avail = 0;
 	else
 		runtime->avail = runtime->buffer_size;
-	if ((runtime->buffer = kmalloc(runtime->buffer_size, GFP_KERNEL)) == NULL) {
+	runtime->buffer = kvmalloc(runtime->buffer_size, GFP_KERNEL);
+	if (!runtime->buffer) {
 		kfree(runtime);
 		return -ENOMEM;
 	}
@@ -137,13 +143,13 @@ static int snd_rawmidi_runtime_free(struct snd_rawmidi_substream *substream)
 {
 	struct snd_rawmidi_runtime *runtime = substream->runtime;
 
-	kfree(runtime->buffer);
+	kvfree(runtime->buffer);
 	kfree(runtime);
 	substream->runtime = NULL;
 	return 0;
 }
 
-static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *substream,int up)
+static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *substream, int up)
 {
 	if (!substream->opened)
 		return;
@@ -159,17 +165,28 @@ static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, i
 		cancel_work_sync(&substream->runtime->event_work);
 }
 
-int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream)
+static void __reset_runtime_ptrs(struct snd_rawmidi_runtime *runtime,
+				 bool is_input)
+{
+	runtime->drain = 0;
+	runtime->appl_ptr = runtime->hw_ptr = 0;
+	runtime->avail = is_input ? 0 : runtime->buffer_size;
+}
+
+static void reset_runtime_ptrs(struct snd_rawmidi_runtime *runtime,
+			       bool is_input)
 {
 	unsigned long flags;
-	struct snd_rawmidi_runtime *runtime = substream->runtime;
 
-	snd_rawmidi_output_trigger(substream, 0);
-	runtime->drain = 0;
 	spin_lock_irqsave(&runtime->lock, flags);
-	runtime->appl_ptr = runtime->hw_ptr = 0;
-	runtime->avail = runtime->buffer_size;
+	__reset_runtime_ptrs(runtime, is_input);
 	spin_unlock_irqrestore(&runtime->lock, flags);
+}
+
+int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream)
+{
+	snd_rawmidi_output_trigger(substream, 0);
+	reset_runtime_ptrs(substream->runtime, false);
 	return 0;
 }
 EXPORT_SYMBOL(snd_rawmidi_drop_output);
@@ -208,15 +225,8 @@ EXPORT_SYMBOL(snd_rawmidi_drain_output);
 
 int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream)
 {
-	unsigned long flags;
-	struct snd_rawmidi_runtime *runtime = substream->runtime;
-
 	snd_rawmidi_input_trigger(substream, 0);
-	runtime->drain = 0;
-	spin_lock_irqsave(&runtime->lock, flags);
-	runtime->appl_ptr = runtime->hw_ptr = 0;
-	runtime->avail = 0;
-	spin_unlock_irqrestore(&runtime->lock, flags);
+	reset_runtime_ptrs(substream->runtime, true);
 	return 0;
 }
 EXPORT_SYMBOL(snd_rawmidi_drain_input);
@@ -330,25 +340,23 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode,
 
 /* called from sound/core/seq/seq_midi.c */
 int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice,
-			    int mode, struct snd_rawmidi_file * rfile)
+			    int mode, struct snd_rawmidi_file *rfile)
 {
 	struct snd_rawmidi *rmidi;
-	int err;
+	int err = 0;
 
 	if (snd_BUG_ON(!rfile))
 		return -EINVAL;
 
 	mutex_lock(&register_mutex);
 	rmidi = snd_rawmidi_search(card, device);
-	if (rmidi == NULL) {
-		mutex_unlock(&register_mutex);
-		return -ENODEV;
-	}
-	if (!try_module_get(rmidi->card->module)) {
-		mutex_unlock(&register_mutex);
-		return -ENXIO;
-	}
+	if (!rmidi)
+		err = -ENODEV;
+	else if (!try_module_get(rmidi->card->module))
+		err = -ENXIO;
 	mutex_unlock(&register_mutex);
+	if (err < 0)
+		return err;
 
 	mutex_lock(&rmidi->open_mutex);
 	err = rawmidi_open_priv(rmidi, subdevice, mode, rfile);
@@ -370,7 +378,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file)
 	struct snd_rawmidi_file *rawmidi_file = NULL;
 	wait_queue_entry_t wait;
 
-	if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) 
+	if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK))
 		return -EINVAL;		/* invalid combination */
 
 	err = nonseekable_open(inode, file);
@@ -520,7 +528,7 @@ int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile)
 
 	if (snd_BUG_ON(!rfile))
 		return -ENXIO;
-	
+
 	rmidi = rfile->rmidi;
 	rawmidi_release_priv(rfile);
 	module_put(rmidi->card->module);
@@ -548,7 +556,7 @@ static int snd_rawmidi_info(struct snd_rawmidi_substream *substream,
 			    struct snd_rawmidi_info *info)
 {
 	struct snd_rawmidi *rmidi;
-	
+
 	if (substream == NULL)
 		return -ENODEV;
 	rmidi = substream->rmidi;
@@ -568,11 +576,13 @@ static int snd_rawmidi_info(struct snd_rawmidi_substream *substream,
 }
 
 static int snd_rawmidi_info_user(struct snd_rawmidi_substream *substream,
-				 struct snd_rawmidi_info __user * _info)
+				 struct snd_rawmidi_info __user *_info)
 {
 	struct snd_rawmidi_info info;
 	int err;
-	if ((err = snd_rawmidi_info(substream, &info)) < 0)
+
+	err = snd_rawmidi_info(substream, &info);
+	if (err < 0)
 		return err;
 	if (copy_to_user(_info, &info, sizeof(struct snd_rawmidi_info)))
 		return -EFAULT;
@@ -619,85 +629,68 @@ static int snd_rawmidi_info_select_user(struct snd_card *card,
 {
 	int err;
 	struct snd_rawmidi_info info;
+
 	if (get_user(info.device, &_info->device))
 		return -EFAULT;
 	if (get_user(info.stream, &_info->stream))
 		return -EFAULT;
 	if (get_user(info.subdevice, &_info->subdevice))
 		return -EFAULT;
-	if ((err = snd_rawmidi_info_select(card, &info)) < 0)
+	err = snd_rawmidi_info_select(card, &info);
+	if (err < 0)
 		return err;
 	if (copy_to_user(_info, &info, sizeof(struct snd_rawmidi_info)))
 		return -EFAULT;
 	return 0;
 }
 
-int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream,
-			      struct snd_rawmidi_params * params)
+static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime,
+				 struct snd_rawmidi_params *params,
+				 bool is_input)
 {
 	char *newbuf, *oldbuf;
-	struct snd_rawmidi_runtime *runtime = substream->runtime;
-	
-	if (substream->append && substream->use_count > 1)
-		return -EBUSY;
-	snd_rawmidi_drain_output(substream);
-	if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) {
+
+	if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L)
 		return -EINVAL;
-	}
-	if (params->avail_min < 1 || params->avail_min > params->buffer_size) {
+	if (params->avail_min < 1 || params->avail_min > params->buffer_size)
 		return -EINVAL;
-	}
 	if (params->buffer_size != runtime->buffer_size) {
-		newbuf = kmalloc(params->buffer_size, GFP_KERNEL);
+		newbuf = kvmalloc(params->buffer_size, GFP_KERNEL);
 		if (!newbuf)
 			return -ENOMEM;
 		spin_lock_irq(&runtime->lock);
 		oldbuf = runtime->buffer;
 		runtime->buffer = newbuf;
 		runtime->buffer_size = params->buffer_size;
-		runtime->avail = runtime->buffer_size;
-		runtime->appl_ptr = runtime->hw_ptr = 0;
+		__reset_runtime_ptrs(runtime, is_input);
 		spin_unlock_irq(&runtime->lock);
-		kfree(oldbuf);
+		kvfree(oldbuf);
 	}
 	runtime->avail_min = params->avail_min;
-	substream->active_sensing = !params->no_active_sensing;
 	return 0;
 }
+
+int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream,
+			      struct snd_rawmidi_params *params)
+{
+	if (substream->append && substream->use_count > 1)
+		return -EBUSY;
+	snd_rawmidi_drain_output(substream);
+	substream->active_sensing = !params->no_active_sensing;
+	return resize_runtime_buffer(substream->runtime, params, false);
+}
 EXPORT_SYMBOL(snd_rawmidi_output_params);
 
 int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream,
-			     struct snd_rawmidi_params * params)
+			     struct snd_rawmidi_params *params)
 {
-	char *newbuf, *oldbuf;
-	struct snd_rawmidi_runtime *runtime = substream->runtime;
-
 	snd_rawmidi_drain_input(substream);
-	if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) {
-		return -EINVAL;
-	}
-	if (params->avail_min < 1 || params->avail_min > params->buffer_size) {
-		return -EINVAL;
-	}
-	if (params->buffer_size != runtime->buffer_size) {
-		newbuf = kmalloc(params->buffer_size, GFP_KERNEL);
-		if (!newbuf)
-			return -ENOMEM;
-		spin_lock_irq(&runtime->lock);
-		oldbuf = runtime->buffer;
-		runtime->buffer = newbuf;
-		runtime->buffer_size = params->buffer_size;
-		runtime->appl_ptr = runtime->hw_ptr = 0;
-		spin_unlock_irq(&runtime->lock);
-		kfree(oldbuf);
-	}
-	runtime->avail_min = params->avail_min;
-	return 0;
+	return resize_runtime_buffer(substream->runtime, params, true);
 }
 EXPORT_SYMBOL(snd_rawmidi_input_params);
 
 static int snd_rawmidi_output_status(struct snd_rawmidi_substream *substream,
-				     struct snd_rawmidi_status * status)
+				     struct snd_rawmidi_status *status)
 {
 	struct snd_rawmidi_runtime *runtime = substream->runtime;
 
@@ -710,7 +703,7 @@ static int snd_rawmidi_output_status(struct snd_rawmidi_substream *substream,
 }
 
 static int snd_rawmidi_input_status(struct snd_rawmidi_substream *substream,
-				    struct snd_rawmidi_status * status)
+				    struct snd_rawmidi_status *status)
 {
 	struct snd_rawmidi_runtime *runtime = substream->runtime;
 
@@ -739,6 +732,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long
 	{
 		int stream;
 		struct snd_rawmidi_info __user *info = argp;
+
 		if (get_user(stream, &info->stream))
 			return -EFAULT;
 		switch (stream) {
@@ -753,6 +747,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long
 	case SNDRV_RAWMIDI_IOCTL_PARAMS:
 	{
 		struct snd_rawmidi_params params;
+
 		if (copy_from_user(&params, argp, sizeof(struct snd_rawmidi_params)))
 			return -EFAULT;
 		switch (params.stream) {
@@ -772,6 +767,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long
 	{
 		int err = 0;
 		struct snd_rawmidi_status status;
+
 		if (copy_from_user(&status, argp, sizeof(struct snd_rawmidi_status)))
 			return -EFAULT;
 		switch (status.stream) {
@@ -797,6 +793,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long
 	case SNDRV_RAWMIDI_IOCTL_DROP:
 	{
 		int val;
+
 		if (get_user(val, (int __user *) argp))
 			return -EFAULT;
 		switch (val) {
@@ -811,6 +808,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long
 	case SNDRV_RAWMIDI_IOCTL_DRAIN:
 	{
 		int val;
+
 		if (get_user(val, (int __user *) argp))
 			return -EFAULT;
 		switch (val) {
@@ -844,7 +842,7 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card,
 	case SNDRV_CTL_IOCTL_RAWMIDI_NEXT_DEVICE:
 	{
 		int device;
-		
+
 		if (get_user(device, (int __user *)argp))
 			return -EFAULT;
 		if (device >= SNDRV_RAWMIDI_DEVICES) /* next device is -1 */
@@ -866,7 +864,7 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card,
 	case SNDRV_CTL_IOCTL_RAWMIDI_PREFER_SUBDEVICE:
 	{
 		int val;
-		
+
 		if (get_user(val, (int __user *)argp))
 			return -EFAULT;
 		control->preferred_subdevice[SND_CTL_SUBDEV_RAWMIDI] = val;
@@ -1020,6 +1018,7 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun
 		spin_lock_irq(&runtime->lock);
 		while (!snd_rawmidi_ready(substream)) {
 			wait_queue_entry_t wait;
+
 			if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) {
 				spin_unlock_irq(&runtime->lock);
 				return result > 0 ? result : -EAGAIN;
@@ -1072,7 +1071,7 @@ int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream)
 	spin_lock_irqsave(&runtime->lock, flags);
 	result = runtime->avail >= runtime->buffer_size;
 	spin_unlock_irqrestore(&runtime->lock, flags);
-	return result;		
+	return result;
 }
 EXPORT_SYMBOL(snd_rawmidi_transmit_empty);
 
@@ -1210,7 +1209,7 @@ EXPORT_SYMBOL(snd_rawmidi_transmit_ack);
  * @substream: the rawmidi substream
  * @buffer: the buffer pointer
  * @count: the data size to transfer
- * 
+ *
  * Copies data from the buffer to the device and advances the pointer.
  *
  * Return: The copied size if successful, or a negative error code on failure.
@@ -1324,6 +1323,7 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf,
 		spin_lock_irq(&runtime->lock);
 		while (!snd_rawmidi_ready_append(substream, count)) {
 			wait_queue_entry_t wait;
+
 			if (file->f_flags & O_NONBLOCK) {
 				spin_unlock_irq(&runtime->lock);
 				return result > 0 ? result : -EAGAIN;
@@ -1357,6 +1357,7 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf,
 		while (runtime->avail != runtime->buffer_size) {
 			wait_queue_entry_t wait;
 			unsigned int last_avail = runtime->avail;
+
 			init_waitqueue_entry(&wait, current);
 			add_wait_queue(&runtime->sleep, &wait);
 			set_current_state(TASK_INTERRUPTIBLE);
@@ -1374,7 +1375,7 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf,
 	return result;
 }
 
-static __poll_t snd_rawmidi_poll(struct file *file, poll_table * wait)
+static __poll_t snd_rawmidi_poll(struct file *file, poll_table *wait)
 {
 	struct snd_rawmidi_file *rfile;
 	struct snd_rawmidi_runtime *runtime;
@@ -1411,7 +1412,6 @@ static __poll_t snd_rawmidi_poll(struct file *file, poll_table * wait)
 #endif
 
 /*
-
  */
 
 static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry,
@@ -1479,8 +1479,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry,
  *  Register functions
  */
 
-static const struct file_operations snd_rawmidi_f_ops =
-{
+static const struct file_operations snd_rawmidi_f_ops = {
 	.owner =	THIS_MODULE,
 	.read =		snd_rawmidi_read,
 	.write =	snd_rawmidi_write,
@@ -1535,7 +1534,7 @@ static void release_rawmidi_device(struct device *dev)
  */
 int snd_rawmidi_new(struct snd_card *card, char *id, int device,
 		    int output_count, int input_count,
-		    struct snd_rawmidi ** rrawmidi)
+		    struct snd_rawmidi **rrawmidi)
 {
 	struct snd_rawmidi *rmidi;
 	int err;
@@ -1566,27 +1565,29 @@ int snd_rawmidi_new(struct snd_card *card, char *id, int device,
 	rmidi->dev.release = release_rawmidi_device;
 	dev_set_name(&rmidi->dev, "midiC%iD%i", card->number, device);
 
-	if ((err = snd_rawmidi_alloc_substreams(rmidi,
-						&rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT],
-						SNDRV_RAWMIDI_STREAM_INPUT,
-						input_count)) < 0) {
-		snd_rawmidi_free(rmidi);
-		return err;
-	}
-	if ((err = snd_rawmidi_alloc_substreams(rmidi,
-						&rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT],
-						SNDRV_RAWMIDI_STREAM_OUTPUT,
-						output_count)) < 0) {
-		snd_rawmidi_free(rmidi);
-		return err;
-	}
-	if ((err = snd_device_new(card, SNDRV_DEV_RAWMIDI, rmidi, &ops)) < 0) {
-		snd_rawmidi_free(rmidi);
-		return err;
-	}
+	err = snd_rawmidi_alloc_substreams(rmidi,
+					   &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT],
+					   SNDRV_RAWMIDI_STREAM_INPUT,
+					   input_count);
+	if (err < 0)
+		goto error;
+	err = snd_rawmidi_alloc_substreams(rmidi,
+					   &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT],
+					   SNDRV_RAWMIDI_STREAM_OUTPUT,
+					   output_count);
+	if (err < 0)
+		goto error;
+	err = snd_device_new(card, SNDRV_DEV_RAWMIDI, rmidi, &ops);
+	if (err < 0)
+		goto error;
+
 	if (rrawmidi)
 		*rrawmidi = rmidi;
 	return 0;
+
+ error:
+	snd_rawmidi_free(rmidi);
+	return err;
 }
 EXPORT_SYMBOL(snd_rawmidi_new);
 
@@ -1624,6 +1625,7 @@ static int snd_rawmidi_free(struct snd_rawmidi *rmidi)
 static int snd_rawmidi_dev_free(struct snd_device *device)
 {
 	struct snd_rawmidi *rmidi = device->device_data;
+
 	return snd_rawmidi_free(rmidi);
 }
 
@@ -1631,6 +1633,7 @@ static int snd_rawmidi_dev_free(struct snd_device *device)
 static void snd_rawmidi_dev_seq_free(struct snd_seq_device *device)
 {
 	struct snd_rawmidi *rmidi = device->private_data;
+
 	rmidi->seq_dev = NULL;
 }
 #endif
@@ -1644,30 +1647,27 @@ static int snd_rawmidi_dev_register(struct snd_device *device)
 
 	if (rmidi->device >= SNDRV_RAWMIDI_DEVICES)
 		return -ENOMEM;
+	err = 0;
 	mutex_lock(&register_mutex);
-	if (snd_rawmidi_search(rmidi->card, rmidi->device)) {
-		mutex_unlock(&register_mutex);
-		return -EBUSY;
-	}
-	list_add_tail(&rmidi->list, &snd_rawmidi_devices);
+	if (snd_rawmidi_search(rmidi->card, rmidi->device))
+		err = -EBUSY;
+	else
+		list_add_tail(&rmidi->list, &snd_rawmidi_devices);
 	mutex_unlock(&register_mutex);
+	if (err < 0)
+		return err;
+
 	err = snd_register_device(SNDRV_DEVICE_TYPE_RAWMIDI,
 				  rmidi->card, rmidi->device,
 				  &snd_rawmidi_f_ops, rmidi, &rmidi->dev);
 	if (err < 0) {
 		rmidi_err(rmidi, "unable to register\n");
-		mutex_lock(&register_mutex);
-		list_del(&rmidi->list);
-		mutex_unlock(&register_mutex);
-		return err;
+		goto error;
 	}
-	if (rmidi->ops && rmidi->ops->dev_register &&
-	    (err = rmidi->ops->dev_register(rmidi)) < 0) {
-		snd_unregister_device(&rmidi->dev);
-		mutex_lock(&register_mutex);
-		list_del(&rmidi->list);
-		mutex_unlock(&register_mutex);
-		return err;
+	if (rmidi->ops && rmidi->ops->dev_register) {
+		err = rmidi->ops->dev_register(rmidi);
+		if (err < 0)
+			goto error_unregister;
 	}
 #ifdef CONFIG_SND_OSSEMUL
 	rmidi->ossreg = 0;
@@ -1719,6 +1719,14 @@ static int snd_rawmidi_dev_register(struct snd_device *device)
 	}
 #endif
 	return 0;
+
+ error_unregister:
+	snd_unregister_device(&rmidi->dev);
+ error:
+	mutex_lock(&register_mutex);
+	list_del(&rmidi->list);
+	mutex_unlock(&register_mutex);
+	return err;
 }
 
 static int snd_rawmidi_dev_disconnect(struct snd_device *device)
@@ -1732,6 +1740,7 @@ static int snd_rawmidi_dev_disconnect(struct snd_device *device)
 	list_del_init(&rmidi->list);
 	for (dir = 0; dir < 2; dir++) {
 		struct snd_rawmidi_substream *s;
+
 		list_for_each_entry(s, &rmidi->streams[dir].substreams, list) {
 			if (s->runtime)
 				wake_up(&s->runtime->sleep);
@@ -1769,7 +1778,7 @@ void snd_rawmidi_set_ops(struct snd_rawmidi *rmidi, int stream,
 			 const struct snd_rawmidi_ops *ops)
 {
 	struct snd_rawmidi_substream *substream;
-	
+
 	list_for_each_entry(substream, &rmidi->streams[stream].substreams, list)
 		substream->ops = ops;
 }
diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c
index 5f64d0d88320..e1f44fc86885 100644
--- a/sound/core/seq/oss/seq_oss.c
+++ b/sound/core/seq/oss/seq_oss.c
@@ -203,7 +203,7 @@ odev_poll(struct file *file, poll_table * wait)
 	struct seq_oss_devinfo *dp;
 	dp = file->private_data;
 	if (snd_BUG_ON(!dp))
-		return -ENXIO;
+		return EPOLLERR;
 	return snd_seq_oss_poll(dp, file, wait);
 }
 
diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c
index 9debd1b8fd28..0d5f8b16d057 100644
--- a/sound/core/seq/oss/seq_oss_midi.c
+++ b/sound/core/seq/oss/seq_oss_midi.c
@@ -637,7 +637,7 @@ snd_seq_oss_midi_putc(struct seq_oss_devinfo *dp, int dev, unsigned char c, stru
 
 	if ((mdev = get_mididev(dp, dev)) == NULL)
 		return -ENODEV;
-	if (snd_midi_event_encode_byte(mdev->coder, c, ev) > 0) {
+	if (snd_midi_event_encode_byte(mdev->coder, c, ev)) {
 		snd_seq_oss_fill_addr(dp, ev, mdev->client, mdev->port);
 		snd_use_lock_free(&mdev->use_lock);
 		return 0;
diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c
index 4f24ea9fad93..ba127c22539a 100644
--- a/sound/core/seq/oss/seq_oss_timer.c
+++ b/sound/core/seq/oss/seq_oss_timer.c
@@ -92,7 +92,7 @@ snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev)
 		case TMR_WAIT_REL:
 			parm += rec->cur_tick;
 			rec->realtime = 0;
-			/* continue to next */
+			/* fall through and continue to next */
 		case TMR_WAIT_ABS:
 			if (parm == 0) {
 				rec->realtime = 1;
diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c
index 639544b4fb04..7de98d71f2aa 100644
--- a/sound/core/seq/seq.c
+++ b/sound/core/seq/seq.c
@@ -84,30 +84,32 @@ static int __init alsa_seq_init(void)
 {
 	int err;
 
-	if ((err = client_init_data()) < 0)
-		goto error;
-
-	/* init memory, room for selected events */
-	if ((err = snd_sequencer_memory_init()) < 0)
-		goto error;
-
-	/* init event queues */
-	if ((err = snd_seq_queues_init()) < 0)
+	err = client_init_data();
+	if (err < 0)
 		goto error;
 
 	/* register sequencer device */
-	if ((err = snd_sequencer_device_init()) < 0)
+	err = snd_sequencer_device_init();
+	if (err < 0)
 		goto error;
 
 	/* register proc interface */
-	if ((err = snd_seq_info_init()) < 0)
-		goto error;
+	err = snd_seq_info_init();
+	if (err < 0)
+		goto error_device;
 
 	/* register our internal client */
-	if ((err = snd_seq_system_client_init()) < 0)
-		goto error;
+	err = snd_seq_system_client_init();
+	if (err < 0)
+		goto error_info;
 
 	snd_seq_autoload_init();
+	return 0;
+
+ error_info:
+	snd_seq_info_done();
+ error_device:
+	snd_sequencer_device_done();
  error:
 	return err;
 }
@@ -126,9 +128,6 @@ static void __exit alsa_seq_exit(void)
 	/* unregister sequencer device */
 	snd_sequencer_device_done();
 
-	/* release event memory */
-	snd_sequencer_memory_done();
-
 	snd_seq_autoload_exit();
 }
 
diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c
index 56ca78423040..92e6524a3a9d 100644
--- a/sound/core/seq/seq_clientmgr.c
+++ b/sound/core/seq/seq_clientmgr.c
@@ -311,10 +311,9 @@ static int snd_seq_open(struct inode *inode, struct file *file)
 	if (err < 0)
 		return err;
 
-	if (mutex_lock_interruptible(&register_mutex))
-		return -ERESTARTSYS;
+	mutex_lock(&register_mutex);
 	client = seq_create_client1(-1, SNDRV_SEQ_DEFAULT_EVENTS);
-	if (client == NULL) {
+	if (!client) {
 		mutex_unlock(&register_mutex);
 		return -ENOMEM;	/* failure code */
 	}
@@ -1101,7 +1100,7 @@ static __poll_t snd_seq_poll(struct file *file, poll_table * wait)
 
 	/* check client structures are in place */
 	if (snd_BUG_ON(!client))
-		return -ENXIO;
+		return EPOLLERR;
 
 	if ((snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_INPUT) &&
 	    client->data.user.fifo) {
@@ -1704,10 +1703,7 @@ static int snd_seq_ioctl_get_queue_timer(struct snd_seq_client *client,
 	if (queue == NULL)
 		return -EINVAL;
 
-	if (mutex_lock_interruptible(&queue->timer_mutex)) {
-		queuefree(queue);
-		return -ERESTARTSYS;
-	}
+	mutex_lock(&queue->timer_mutex);
 	tmr = queue->timer;
 	memset(timer, 0, sizeof(*timer));
 	timer->queue = queue->queue;
@@ -1741,10 +1737,7 @@ static int snd_seq_ioctl_set_queue_timer(struct snd_seq_client *client,
 		q = queueptr(timer->queue);
 		if (q == NULL)
 			return -ENXIO;
-		if (mutex_lock_interruptible(&q->timer_mutex)) {
-			queuefree(q);
-			return -ERESTARTSYS;
-		}
+		mutex_lock(&q->timer_mutex);
 		tmr = q->timer;
 		snd_seq_queue_timer_close(timer->queue);
 		tmr->type = timer->type;
@@ -2180,8 +2173,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index,
 	if (card == NULL && client_index >= SNDRV_SEQ_GLOBAL_CLIENTS)
 		return -EINVAL;
 
-	if (mutex_lock_interruptible(&register_mutex))
-		return -ERESTARTSYS;
+	mutex_lock(&register_mutex);
 
 	if (card) {
 		client_index += SNDRV_SEQ_GLOBAL_CLIENTS
@@ -2522,19 +2514,15 @@ int __init snd_sequencer_device_init(void)
 	snd_device_initialize(&seq_dev, NULL);
 	dev_set_name(&seq_dev, "seq");
 
-	if (mutex_lock_interruptible(&register_mutex))
-		return -ERESTARTSYS;
-
+	mutex_lock(&register_mutex);
 	err = snd_register_device(SNDRV_DEVICE_TYPE_SEQUENCER, NULL, 0,
 				  &snd_seq_f_ops, NULL, &seq_dev);
+	mutex_unlock(&register_mutex);
 	if (err < 0) {
-		mutex_unlock(&register_mutex);
 		put_device(&seq_dev);
 		return err;
 	}
 	
-	mutex_unlock(&register_mutex);
-
 	return 0;
 }
 
@@ -2543,7 +2531,7 @@ int __init snd_sequencer_device_init(void)
 /* 
  * unregister sequencer device 
  */
-void __exit snd_sequencer_device_done(void)
+void snd_sequencer_device_done(void)
 {
 	snd_unregister_device(&seq_dev);
 	put_device(&seq_dev);
diff --git a/sound/core/seq/seq_info.c b/sound/core/seq/seq_info.c
index 97015447b9b3..b27fedd435b6 100644
--- a/sound/core/seq/seq_info.c
+++ b/sound/core/seq/seq_info.c
@@ -50,7 +50,7 @@ create_info_entry(char *name, void (*read)(struct snd_info_entry *,
 	return entry;
 }
 
-static void free_info_entries(void)
+void snd_seq_info_done(void)
 {
 	snd_info_free_entry(queues_entry);
 	snd_info_free_entry(clients_entry);
@@ -70,12 +70,6 @@ int __init snd_seq_info_init(void)
 	return 0;
 
  error:
-	free_info_entries();
+	snd_seq_info_done();
 	return -ENOMEM;
 }
-
-int __exit snd_seq_info_done(void)
-{
-	free_info_entries();
-	return 0;
-}
diff --git a/sound/core/seq/seq_info.h b/sound/core/seq/seq_info.h
index f8549f81a645..2cdf8f6e63f5 100644
--- a/sound/core/seq/seq_info.h
+++ b/sound/core/seq/seq_info.h
@@ -30,11 +30,11 @@ void snd_seq_info_queues_read(struct snd_info_entry *entry, struct snd_info_buff
 
 
 #ifdef CONFIG_SND_PROC_FS
-int snd_seq_info_init( void );
-int snd_seq_info_done( void );
+int snd_seq_info_init(void);
+void snd_seq_info_done(void);
 #else
 static inline int snd_seq_info_init(void) { return 0; }
-static inline int snd_seq_info_done(void) { return 0; }
+static inline void snd_seq_info_done(void) {}
 #endif
 
 #endif
diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c
index a4c8543176b2..5b0388202bac 100644
--- a/sound/core/seq/seq_memory.c
+++ b/sound/core/seq/seq_memory.c
@@ -504,18 +504,6 @@ int snd_seq_pool_delete(struct snd_seq_pool **ppool)
 	return 0;
 }
 
-/* initialize sequencer memory */
-int __init snd_sequencer_memory_init(void)
-{
-	return 0;
-}
-
-/* release sequencer memory */
-void __exit snd_sequencer_memory_done(void)
-{
-}
-
-
 /* exported to seq_clientmgr.c */
 void snd_seq_info_pool(struct snd_info_buffer *buffer,
 		       struct snd_seq_pool *pool, char *space)
diff --git a/sound/core/seq/seq_memory.h b/sound/core/seq/seq_memory.h
index 3abe306c394a..1292fe91f02e 100644
--- a/sound/core/seq/seq_memory.h
+++ b/sound/core/seq/seq_memory.h
@@ -94,12 +94,6 @@ struct snd_seq_pool *snd_seq_pool_new(int poolsize);
 /* remove pool */
 int snd_seq_pool_delete(struct snd_seq_pool **pool);
 
-/* init memory */
-int snd_sequencer_memory_init(void);
-            
-/* release event memory */
-void snd_sequencer_memory_done(void);
-
 /* polling */
 int snd_seq_pool_poll_wait(struct snd_seq_pool *pool, struct file *file, poll_table *wait);
 
diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c
index 5dd0ee258359..9e0dabd3ce5f 100644
--- a/sound/core/seq/seq_midi.c
+++ b/sound/core/seq/seq_midi.c
@@ -78,7 +78,7 @@ static void snd_midi_input_event(struct snd_rawmidi_substream *substream)
 	struct seq_midisynth *msynth;
 	struct snd_seq_event ev;
 	char buf[16], *pbuf;
-	long res, count;
+	long res;
 
 	if (substream == NULL)
 		return;
@@ -94,19 +94,15 @@ static void snd_midi_input_event(struct snd_rawmidi_substream *substream)
 		if (msynth->parser == NULL)
 			continue;
 		pbuf = buf;
-		while (res > 0) {
-			count = snd_midi_event_encode(msynth->parser, pbuf, res, &ev);
-			if (count < 0)
-				break;
-			pbuf += count;
-			res -= count;
-			if (ev.type != SNDRV_SEQ_EVENT_NONE) {
-				ev.source.port = msynth->seq_port;
-				ev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS;
-				snd_seq_kernel_client_dispatch(msynth->seq_client, &ev, 1, 0);
-				/* clear event and reset header */
-				memset(&ev, 0, sizeof(ev));
-			}
+		while (res-- > 0) {
+			if (!snd_midi_event_encode_byte(msynth->parser,
+							*pbuf++, &ev))
+				continue;
+			ev.source.port = msynth->seq_port;
+			ev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS;
+			snd_seq_kernel_client_dispatch(msynth->seq_client, &ev, 1, 0);
+			/* clear event and reset header */
+			memset(&ev, 0, sizeof(ev));
 		}
 	}
 }
diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c
index 288f839a554b..c1975dd31871 100644
--- a/sound/core/seq/seq_midi_emul.c
+++ b/sound/core/seq/seq_midi_emul.c
@@ -318,7 +318,7 @@ do_control(struct snd_midi_op *ops, void *drv, struct snd_midi_channel_set *chse
 		break;
 	case MIDI_CTL_MSB_DATA_ENTRY:
 		chan->control[MIDI_CTL_LSB_DATA_ENTRY] = 0;
-		/* go through here */
+		/* fall through */
 	case MIDI_CTL_LSB_DATA_ENTRY:
 		if (chan->param_type == SNDRV_MIDI_PARAM_TYPE_REGISTERED)
 			rpn(ops, drv, chan, chset);
@@ -728,15 +728,3 @@ void snd_midi_channel_free_set(struct snd_midi_channel_set *chset)
 	kfree(chset);
 }
 EXPORT_SYMBOL(snd_midi_channel_free_set);
-
-static int __init alsa_seq_midi_emul_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_seq_midi_emul_exit(void)
-{
-}
-
-module_init(alsa_seq_midi_emul_init)
-module_exit(alsa_seq_midi_emul_exit)
diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c
index 90bbbdbeba03..b11419537062 100644
--- a/sound/core/seq/seq_midi_event.c
+++ b/sound/core/seq/seq_midi_event.c
@@ -175,14 +175,6 @@ void snd_midi_event_reset_decode(struct snd_midi_event *dev)
 }
 EXPORT_SYMBOL(snd_midi_event_reset_decode);
 
-#if 0
-void snd_midi_event_init(struct snd_midi_event *dev)
-{
-	snd_midi_event_reset_encode(dev);
-	snd_midi_event_reset_decode(dev);
-}
-#endif  /*  0  */
-
 void snd_midi_event_no_status(struct snd_midi_event *dev, int on)
 {
 	dev->nostat = on ? 1 : 0;
@@ -190,69 +182,16 @@ void snd_midi_event_no_status(struct snd_midi_event *dev, int on)
 EXPORT_SYMBOL(snd_midi_event_no_status);
 
 /*
- * resize buffer
- */
-#if 0
-int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize)
-{
-	unsigned char *new_buf, *old_buf;
-	unsigned long flags;
-
-	if (bufsize == dev->bufsize)
-		return 0;
-	new_buf = kmalloc(bufsize, GFP_KERNEL);
-	if (new_buf == NULL)
-		return -ENOMEM;
-	spin_lock_irqsave(&dev->lock, flags);
-	old_buf = dev->buf;
-	dev->buf = new_buf;
-	dev->bufsize = bufsize;
-	reset_encode(dev);
-	spin_unlock_irqrestore(&dev->lock, flags);
-	kfree(old_buf);
-	return 0;
-}
-#endif  /*  0  */
-
-/*
- *  read bytes and encode to sequencer event if finished
- *  return the size of encoded bytes
- */
-long snd_midi_event_encode(struct snd_midi_event *dev, unsigned char *buf, long count,
-			   struct snd_seq_event *ev)
-{
-	long result = 0;
-	int rc;
-
-	ev->type = SNDRV_SEQ_EVENT_NONE;
-
-	while (count-- > 0) {
-		rc = snd_midi_event_encode_byte(dev, *buf++, ev);
-		result++;
-		if (rc < 0)
-			return rc;
-		else if (rc > 0)
-			return result;
-	}
-
-	return result;
-}
-EXPORT_SYMBOL(snd_midi_event_encode);
-
-/*
  *  read one byte and encode to sequencer event:
- *  return 1 if MIDI bytes are encoded to an event
- *         0 data is not finished
- *         negative for error
+ *  return true if MIDI bytes are encoded to an event
+ *         false data is not finished
  */
-int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c,
-			       struct snd_seq_event *ev)
+bool snd_midi_event_encode_byte(struct snd_midi_event *dev, unsigned char c,
+				struct snd_seq_event *ev)
 {
-	int rc = 0;
+	bool rc = false;
 	unsigned long flags;
 
-	c &= 0xff;
-
 	if (c >= MIDI_CMD_COMMON_CLOCK) {
 		/* real-time event */
 		ev->type = status_event[ST_SPECIAL + c - 0xf0].event;
@@ -293,7 +232,7 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c,
 			status_event[dev->type].encode(dev, ev);
 		if (dev->type >= ST_SPECIAL)
 			dev->type = ST_INVALID;
-		rc = 1;
+		rc = true;
 	} else 	if (dev->type == ST_SYSEX) {
 		if (c == MIDI_CMD_COMMON_SYSEX_END ||
 		    dev->read >= dev->bufsize) {
@@ -306,7 +245,7 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c,
 				dev->read = 0; /* continue to parse */
 			else
 				reset_encode(dev); /* all parsed */
-			rc = 1;
+			rc = true;
 		}
 	}
 
@@ -531,15 +470,3 @@ static int extra_decode_xrpn(struct snd_midi_event *dev, unsigned char *buf,
 	}
 	return idx;
 }
-
-static int __init alsa_seq_midi_event_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_seq_midi_event_exit(void)
-{
-}
-
-module_init(alsa_seq_midi_event_init)
-module_exit(alsa_seq_midi_event_exit)
diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c
index b377f5048352..3b3ac96f1f5f 100644
--- a/sound/core/seq/seq_queue.c
+++ b/sound/core/seq/seq_queue.c
@@ -159,18 +159,8 @@ static void queue_delete(struct snd_seq_queue *q)
 
 /*----------------------------------------------------------------*/
 
-/* setup queues */
-int __init snd_seq_queues_init(void)
-{
-	/*
-	memset(queue_list, 0, sizeof(queue_list));
-	num_queues = 0;
-	*/
-	return 0;
-}
-
 /* delete all existing queues */
-void __exit snd_seq_queues_delete(void)
+void snd_seq_queues_delete(void)
 {
 	int i;
 
diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h
index 719093489a2c..e006fc8e3a36 100644
--- a/sound/core/seq/seq_queue.h
+++ b/sound/core/seq/seq_queue.h
@@ -63,9 +63,6 @@ struct snd_seq_queue {
 /* get the number of current queues */
 int snd_seq_queue_get_cur_queues(void);
 
-/* init queues structure */
-int snd_seq_queues_init(void);
-
 /* delete queues */ 
 void snd_seq_queues_delete(void);
 
@@ -112,28 +109,4 @@ int snd_seq_queue_is_used(int queueid, int client);
 
 int snd_seq_control_queue(struct snd_seq_event *ev, int atomic, int hop);
 
-/*
- * 64bit division - for sync stuff..
- */
-#if defined(i386) || defined(i486)
-
-#define udiv_qrnnd(q, r, n1, n0, d) \
-  __asm__ ("divl %4"		\
-	   : "=a" ((u32)(q)),	\
-	     "=d" ((u32)(r))	\
-	   : "0" ((u32)(n0)),	\
-	     "1" ((u32)(n1)),	\
-	     "rm" ((u32)(d)))
-
-#define u64_div(x,y,q) do {u32 __tmp; udiv_qrnnd(q, __tmp, (x)>>32, x, y);} while (0)
-#define u64_mod(x,y,r) do {u32 __tmp; udiv_qrnnd(__tmp, q, (x)>>32, x, y);} while (0)
-#define u64_divmod(x,y,q,r) udiv_qrnnd(q, r, (x)>>32, x, y)
-
-#else
-#define u64_div(x,y,q)	((q) = (u32)((u64)(x) / (u64)(y)))
-#define u64_mod(x,y,r)	((r) = (u32)((u64)(x) % (u64)(y)))
-#define u64_divmod(x,y,q,r) (u64_div(x,y,q), u64_mod(x,y,r))
-#endif
-
-
 #endif
diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c
index 289ae6bb81d9..a2f1c6b58693 100644
--- a/sound/core/seq/seq_virmidi.c
+++ b/sound/core/seq/seq_virmidi.c
@@ -89,7 +89,7 @@ static int snd_virmidi_dev_receive_event(struct snd_virmidi_dev *rdev,
 	else
 		down_read(&rdev->filelist_sem);
 	list_for_each_entry(vmidi, &rdev->filelist, list) {
-		if (!vmidi->trigger)
+		if (!READ_ONCE(vmidi->trigger))
 			continue;
 		if (ev->type == SNDRV_SEQ_EVENT_SYSEX) {
 			if ((ev->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) != SNDRV_SEQ_EVENT_LENGTH_VARIABLE)
@@ -110,23 +110,6 @@ static int snd_virmidi_dev_receive_event(struct snd_virmidi_dev *rdev,
 }
 
 /*
- * receive an event from the remote virmidi port
- *
- * for rawmidi inputs, you can call this function from the event
- * handler of a remote port which is attached to the virmidi via
- * SNDRV_VIRMIDI_SEQ_ATTACH.
- */
-#if 0
-int snd_virmidi_receive(struct snd_rawmidi *rmidi, struct snd_seq_event *ev)
-{
-	struct snd_virmidi_dev *rdev;
-
-	rdev = rmidi->private_data;
-	return snd_virmidi_dev_receive_event(rdev, ev, true);
-}
-#endif  /*  0  */
-
-/*
  * event handler of virmidi port
  */
 static int snd_virmidi_event_input(struct snd_seq_event *ev, int direct,
@@ -147,68 +130,62 @@ static void snd_virmidi_input_trigger(struct snd_rawmidi_substream *substream, i
 {
 	struct snd_virmidi *vmidi = substream->runtime->private_data;
 
-	if (up) {
-		vmidi->trigger = 1;
-	} else {
-		vmidi->trigger = 0;
-	}
+	WRITE_ONCE(vmidi->trigger, !!up);
 }
 
-/*
- * trigger rawmidi stream for output
+/* process rawmidi bytes and send events;
+ * we need no lock here for vmidi->event since it's handled only in this work
  */
-static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+static void snd_vmidi_output_work(struct work_struct *work)
 {
-	struct snd_virmidi *vmidi = substream->runtime->private_data;
-	int count, res;
-	unsigned char buf[32], *pbuf;
-	unsigned long flags;
-
-	if (up) {
-		vmidi->trigger = 1;
-		if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH &&
-		    !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) {
-			while (snd_rawmidi_transmit(substream, buf,
-						    sizeof(buf)) > 0) {
-				/* ignored */
-			}
-			return;
-		}
-		spin_lock_irqsave(&substream->runtime->lock, flags);
+	struct snd_virmidi *vmidi;
+	struct snd_rawmidi_substream *substream;
+	unsigned char input;
+	int ret;
+
+	vmidi = container_of(work, struct snd_virmidi, output_work);
+	substream = vmidi->substream;
+
+	/* discard the outputs in dispatch mode unless subscribed */
+	if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH &&
+	    !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) {
+		while (!snd_rawmidi_transmit_empty(substream))
+			snd_rawmidi_transmit_ack(substream, 1);
+		return;
+	}
+
+	while (READ_ONCE(vmidi->trigger)) {
+		if (snd_rawmidi_transmit(substream, &input, 1) != 1)
+			break;
+		if (!snd_midi_event_encode_byte(vmidi->parser, input,
+						&vmidi->event))
+			continue;
 		if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) {
-			if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0)
-				goto out;
+			ret = snd_seq_kernel_client_dispatch(vmidi->client,
+							     &vmidi->event,
+							     false, 0);
 			vmidi->event.type = SNDRV_SEQ_EVENT_NONE;
-		}
-		while (1) {
-			count = __snd_rawmidi_transmit_peek(substream, buf, sizeof(buf));
-			if (count <= 0)
+			if (ret < 0)
 				break;
-			pbuf = buf;
-			while (count > 0) {
-				res = snd_midi_event_encode(vmidi->parser, pbuf, count, &vmidi->event);
-				if (res < 0) {
-					snd_midi_event_reset_encode(vmidi->parser);
-					continue;
-				}
-				__snd_rawmidi_transmit_ack(substream, res);
-				pbuf += res;
-				count -= res;
-				if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) {
-					if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0)
-						goto out;
-					vmidi->event.type = SNDRV_SEQ_EVENT_NONE;
-				}
-			}
 		}
-	out:
-		spin_unlock_irqrestore(&substream->runtime->lock, flags);
-	} else {
-		vmidi->trigger = 0;
+		/* rawmidi input might be huge, allow to have a break */
+		cond_resched();
 	}
 }
 
 /*
+ * trigger rawmidi stream for output
+ */
+static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+	struct snd_virmidi *vmidi = substream->runtime->private_data;
+
+	WRITE_ONCE(vmidi->trigger, !!up);
+	if (up)
+		queue_work(system_highpri_wq, &vmidi->output_work);
+}
+
+/*
  * open rawmidi handle for input
  */
 static int snd_virmidi_input_open(struct snd_rawmidi_substream *substream)
@@ -260,6 +237,7 @@ static int snd_virmidi_output_open(struct snd_rawmidi_substream *substream)
 	vmidi->port = rdev->port;
 	snd_virmidi_init_event(vmidi, &vmidi->event);
 	vmidi->rdev = rdev;
+	INIT_WORK(&vmidi->output_work, snd_vmidi_output_work);
 	runtime->private_data = vmidi;
 	return 0;
 }
@@ -289,6 +267,9 @@ static int snd_virmidi_input_close(struct snd_rawmidi_substream *substream)
 static int snd_virmidi_output_close(struct snd_rawmidi_substream *substream)
 {
 	struct snd_virmidi *vmidi = substream->runtime->private_data;
+
+	WRITE_ONCE(vmidi->trigger, false); /* to be sure */
+	cancel_work_sync(&vmidi->output_work);
 	snd_midi_event_free(vmidi->parser);
 	substream->runtime->private_data = NULL;
 	kfree(vmidi);
@@ -546,19 +527,3 @@ int snd_virmidi_new(struct snd_card *card, int device, struct snd_rawmidi **rrmi
 	return 0;
 }
 EXPORT_SYMBOL(snd_virmidi_new);
-
-/*
- *  ENTRY functions
- */
-
-static int __init alsa_virmidi_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_virmidi_exit(void)
-{
-}
-
-module_init(alsa_virmidi_init)
-module_exit(alsa_virmidi_exit)
diff --git a/sound/core/timer.c b/sound/core/timer.c
index b6f076bbc72d..61a0cec6e1f6 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -883,6 +883,11 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid,
 
 	if (snd_BUG_ON(!tid))
 		return -EINVAL;
+	if (tid->dev_class == SNDRV_TIMER_CLASS_CARD ||
+	    tid->dev_class == SNDRV_TIMER_CLASS_PCM) {
+		if (WARN_ON(!card))
+			return -EINVAL;
+	}
 	if (rtimer)
 		*rtimer = NULL;
 	timer = kzalloc(sizeof(*timer), GFP_KERNEL);
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 78a2fdc38531..1e34e6381baa 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -778,7 +778,6 @@ static const struct snd_pcm_ops loopback_pcm_ops = {
 	.trigger =	loopback_trigger,
 	.pointer =	loopback_pointer,
 	.page =		snd_pcm_lib_get_vmalloc_page,
-	.mmap =		snd_pcm_lib_mmap_vmalloc,
 };
 
 static int loopback_pcm_new(struct loopback *loopback,
diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c
index 3e745f47dd2f..dae26e856b26 100644
--- a/sound/drivers/mpu401/mpu401_uart.c
+++ b/sound/drivers/mpu401/mpu401_uart.c
@@ -617,19 +617,3 @@ free_device:
 }
 
 EXPORT_SYMBOL(snd_mpu401_uart_new);
-
-/*
- *  INIT part
- */
-
-static int __init alsa_mpu401_uart_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_mpu401_uart_exit(void)
-{
-}
-
-module_init(alsa_mpu401_uart_init)
-module_exit(alsa_mpu401_uart_exit)
diff --git a/sound/drivers/opl3/opl3_drums.c b/sound/drivers/opl3/opl3_drums.c
index 73694380734a..14929822956c 100644
--- a/sound/drivers/opl3/opl3_drums.c
+++ b/sound/drivers/opl3/opl3_drums.c
@@ -21,8 +21,6 @@
 
 #include "opl3_voice.h"
 
-extern char snd_opl3_regmap[MAX_OPL2_VOICES][4];
-
 static char snd_opl3_drum_table[47] =
 {
 	OPL3_BASSDRUM_ON,  OPL3_BASSDRUM_ON,  OPL3_HIHAT_ON,	/* 35 - 37 */
diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c
index 588963d6be28..cf86c36c7c3b 100644
--- a/sound/drivers/opl3/opl3_lib.c
+++ b/sound/drivers/opl3/opl3_lib.c
@@ -31,13 +31,12 @@
 #include <linux/slab.h>
 #include <linux/ioport.h>
 #include <sound/minors.h>
+#include "opl3_voice.h"
 
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Hannu Savolainen 1993-1996, Rob Hooft");
 MODULE_DESCRIPTION("Routines for control of AdLib FM cards (OPL2/OPL3/OPL4 chips)");
 MODULE_LICENSE("GPL");
 
-extern char snd_opl3_regmap[MAX_OPL2_VOICES][4];
-
 static void snd_opl2_command(struct snd_opl3 * opl3, unsigned short cmd, unsigned char val)
 {
 	unsigned long flags;
@@ -539,19 +538,3 @@ int snd_opl3_hwdep_new(struct snd_opl3 * opl3,
 }
 
 EXPORT_SYMBOL(snd_opl3_hwdep_new);
-
-/*
- *  INIT part
- */
-
-static int __init alsa_opl3_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_opl3_exit(void)
-{
-}
-
-module_init(alsa_opl3_init)
-module_exit(alsa_opl3_exit)
diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
index bb3f3a5a6951..a33cb744e96c 100644
--- a/sound/drivers/opl3/opl3_midi.c
+++ b/sound/drivers/opl3/opl3_midi.c
@@ -25,10 +25,6 @@
 #include "opl3_voice.h"
 #include <sound/asoundef.h>
 
-extern char snd_opl3_regmap[MAX_OPL2_VOICES][4];
-
-extern bool use_internal_drums;
-
 static void snd_opl3_note_off_unsafe(void *p, int note, int vel,
 				     struct snd_midi_channel *chan);
 /*
@@ -372,6 +368,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
 			instr_4op = 1;
 			break;
 		}
+		/* fall through */
 	default:
 		spin_unlock_irqrestore(&opl3->voice_lock, flags);
 		return;
@@ -721,9 +718,6 @@ void snd_opl3_note_off(void *p, int note, int vel,
  */
 void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *chan)
 {
-  	struct snd_opl3 *opl3;
-
-	opl3 = p;
 #ifdef DEBUG_MIDI
 	snd_printk(KERN_DEBUG "Key pressure, ch#: %i, inst#: %i\n",
 		   chan->number, chan->midi_program);
@@ -735,9 +729,6 @@ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *cha
  */
 void snd_opl3_terminate_note(void *p, int note, struct snd_midi_channel *chan)
 {
-  	struct snd_opl3 *opl3;
-
-	opl3 = p;
 #ifdef DEBUG_MIDI
 	snd_printk(KERN_DEBUG "Terminate note, ch#: %i, inst#: %i\n",
 		   chan->number, chan->midi_program);
@@ -861,9 +852,6 @@ void snd_opl3_control(void *p, int type, struct snd_midi_channel *chan)
 void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan,
 		   struct snd_midi_channel_set *chset)
 {
-  	struct snd_opl3 *opl3;
-
-	opl3 = p;
 #ifdef DEBUG_MIDI
 	snd_printk(KERN_DEBUG "NRPN, ch#: %i, inst#: %i\n",
 		   chan->number, chan->midi_program);
@@ -876,9 +864,6 @@ void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan,
 void snd_opl3_sysex(void *p, unsigned char *buf, int len,
 		    int parsed, struct snd_midi_channel_set *chset)
 {
-  	struct snd_opl3 *opl3;
-
-	opl3 = p;
 #ifdef DEBUG_MIDI
 	snd_printk(KERN_DEBUG "SYSEX\n");
 #endif
diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c
index 22c3e4bca220..869220ced4ed 100644
--- a/sound/drivers/opl3/opl3_oss.c
+++ b/sound/drivers/opl3/opl3_oss.c
@@ -29,8 +29,6 @@ static int snd_opl3_reset_seq_oss(struct snd_seq_oss_arg *arg);
 
 /* operators */
 
-extern struct snd_midi_op opl3_ops;
-
 static struct snd_seq_oss_callback oss_callback = {
 	.owner = 	THIS_MODULE,
 	.open =		snd_opl3_open_seq_oss,
@@ -233,11 +231,8 @@ static int snd_opl3_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format,
 static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd,
 				  unsigned long ioarg)
 {
-	struct snd_opl3 *opl3;
-
 	if (snd_BUG_ON(!arg))
 		return -ENXIO;
-	opl3 = arg->private_data;
 	switch (cmd) {
 		case SNDCTL_FM_LOAD_INSTR:
 			snd_printk(KERN_ERR "OPL3: "
@@ -261,11 +256,8 @@ static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd,
 /* reset device */
 static int snd_opl3_reset_seq_oss(struct snd_seq_oss_arg *arg)
 {
-	struct snd_opl3 *opl3;
-
 	if (snd_BUG_ON(!arg))
 		return -ENXIO;
-	opl3 = arg->private_data;
 
 	return 0;
 }
diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c
index 42920a243328..d522925fc5c0 100644
--- a/sound/drivers/opl3/opl3_synth.c
+++ b/sound/drivers/opl3/opl3_synth.c
@@ -24,6 +24,7 @@
 #include <linux/nospec.h>
 #include <sound/opl3.h>
 #include <sound/asound_fm.h>
+#include "opl3_voice.h"
 
 #if IS_ENABLED(CONFIG_SND_SEQUENCER)
 #define OPL3_SUPPORT_SYNTH
diff --git a/sound/drivers/opl3/opl3_voice.h b/sound/drivers/opl3/opl3_voice.h
index a2445163008e..5b02bd49fde4 100644
--- a/sound/drivers/opl3/opl3_voice.h
+++ b/sound/drivers/opl3/opl3_voice.h
@@ -52,4 +52,8 @@ void snd_opl3_free_seq_oss(struct snd_opl3 *opl3);
 #define snd_opl3_free_seq_oss(opl3) /* NOP */
 #endif
 
+extern char snd_opl3_regmap[MAX_OPL2_VOICES][4];
+extern bool use_internal_drums;
+extern struct snd_midi_op opl3_ops;
+
 #endif
diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c
index db76a5bf2bd2..819d2dce2a19 100644
--- a/sound/drivers/opl4/opl4_lib.c
+++ b/sound/drivers/opl4/opl4_lib.c
@@ -263,15 +263,3 @@ int snd_opl4_create(struct snd_card *card,
 }
 
 EXPORT_SYMBOL(snd_opl4_create);
-
-static int __init alsa_opl4_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_opl4_exit(void)
-{
-}
-
-module_init(alsa_opl4_init)
-module_exit(alsa_opl4_exit)
diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c
index 121357397a6d..04368dd59a4c 100644
--- a/sound/drivers/vx/vx_core.c
+++ b/sound/drivers/vx/vx_core.c
@@ -815,18 +815,3 @@ struct vx_core *snd_vx_create(struct snd_card *card, struct snd_vx_hardware *hw,
 }
 
 EXPORT_SYMBOL(snd_vx_create);
-
-/*
- * module entries
- */
-static int __init alsa_vx_core_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_vx_core_exit(void)
-{
-}
-
-module_init(alsa_vx_core_init)
-module_exit(alsa_vx_core_exit)
diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c
index 380a028469c4..ba80f459bdc5 100644
--- a/sound/drivers/vx/vx_pcm.c
+++ b/sound/drivers/vx/vx_pcm.c
@@ -883,7 +883,6 @@ static const struct snd_pcm_ops vx_pcm_playback_ops = {
 	.trigger =	vx_pcm_trigger,
 	.pointer =	vx_pcm_playback_pointer,
 	.page =		snd_pcm_lib_get_vmalloc_page,
-	.mmap =		snd_pcm_lib_mmap_vmalloc,
 };
 
 
@@ -1105,7 +1104,6 @@ static const struct snd_pcm_ops vx_pcm_capture_ops = {
 	.trigger =	vx_pcm_trigger,
 	.pointer =	vx_pcm_capture_pointer,
 	.page =		snd_pcm_lib_get_vmalloc_page,
-	.mmap =		snd_pcm_lib_mmap_vmalloc,
 };
 
 
diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c
index e6adab3ef42e..ea9b86450580 100644
--- a/sound/firewire/bebob/bebob_pcm.c
+++ b/sound/firewire/bebob/bebob_pcm.c
@@ -373,7 +373,6 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob)
 		.pointer	= pcm_playback_pointer,
 		.ack		= pcm_playback_ack,
 		.page		= snd_pcm_lib_get_vmalloc_page,
-		.mmap		= snd_pcm_lib_mmap_vmalloc,
 	};
 	struct snd_pcm *pcm;
 	int err;
diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c
index b2efb1c71a98..218292bdace6 100644
--- a/sound/firewire/dice/dice-alesis.c
+++ b/sound/firewire/dice/dice-alesis.c
@@ -37,7 +37,7 @@ int snd_dice_detect_alesis_formats(struct snd_dice *dice)
 				MAX_STREAMS * SND_DICE_RATE_MODE_COUNT *
 				sizeof(unsigned int));
 	} else {
-		memcpy(dice->rx_pcm_chs, alesis_io26_tx_pcm_chs,
+		memcpy(dice->tx_pcm_chs, alesis_io26_tx_pcm_chs,
 				MAX_STREAMS * SND_DICE_RATE_MODE_COUNT *
 				sizeof(unsigned int));
 	}
diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c
index 80351b29fe0d..bb3ef5ff3488 100644
--- a/sound/firewire/dice/dice-pcm.c
+++ b/sound/firewire/dice/dice-pcm.c
@@ -412,7 +412,6 @@ int snd_dice_create_pcm(struct snd_dice *dice)
 		.pointer   = capture_pointer,
 		.ack       = capture_ack,
 		.page      = snd_pcm_lib_get_vmalloc_page,
-		.mmap      = snd_pcm_lib_mmap_vmalloc,
 	};
 	static const struct snd_pcm_ops playback_ops = {
 		.open      = pcm_open,
@@ -425,7 +424,6 @@ int snd_dice_create_pcm(struct snd_dice *dice)
 		.pointer   = playback_pointer,
 		.ack       = playback_ack,
 		.page      = snd_pcm_lib_get_vmalloc_page,
-		.mmap      = snd_pcm_lib_mmap_vmalloc,
 	};
 	struct snd_pcm *pcm;
 	unsigned int capture, playback;
diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c
index 796f4b4645f5..fdcff0460c53 100644
--- a/sound/firewire/digi00x/digi00x-pcm.c
+++ b/sound/firewire/digi00x/digi00x-pcm.c
@@ -352,7 +352,6 @@ int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x)
 		.pointer	= pcm_playback_pointer,
 		.ack		= pcm_playback_ack,
 		.page		= snd_pcm_lib_get_vmalloc_page,
-		.mmap		= snd_pcm_lib_mmap_vmalloc,
 	};
 	struct snd_pcm *pcm;
 	int err;
diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c
index e3c16308363d..bf47f9ec8703 100644
--- a/sound/firewire/fireface/ff-pcm.c
+++ b/sound/firewire/fireface/ff-pcm.c
@@ -383,7 +383,6 @@ int snd_ff_create_pcm_devices(struct snd_ff *ff)
 		.pointer	= pcm_playback_pointer,
 		.ack		= pcm_playback_ack,
 		.page		= snd_pcm_lib_get_vmalloc_page,
-		.mmap		= snd_pcm_lib_mmap_vmalloc,
 	};
 	struct snd_pcm *pcm;
 	int err;
diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c
index 40faed5e6968..aed566d82726 100644
--- a/sound/firewire/fireworks/fireworks_pcm.c
+++ b/sound/firewire/fireworks/fireworks_pcm.c
@@ -397,7 +397,6 @@ int snd_efw_create_pcm_devices(struct snd_efw *efw)
 		.pointer	= pcm_playback_pointer,
 		.ack		= pcm_playback_ack,
 		.page		= snd_pcm_lib_get_vmalloc_page,
-		.mmap		= snd_pcm_lib_mmap_vmalloc,
 	};
 	struct snd_pcm *pcm;
 	int err;
diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index 3919e186a30b..30957477e005 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -454,7 +454,6 @@ static int isight_create_pcm(struct isight *isight)
 		.trigger   = isight_trigger,
 		.pointer   = isight_pointer,
 		.page      = snd_pcm_lib_get_vmalloc_page,
-		.mmap      = snd_pcm_lib_mmap_vmalloc,
 	};
 	struct snd_pcm *pcm;
 	int err;
diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c
index 4330220890e8..ab69d7e6ac05 100644
--- a/sound/firewire/motu/motu-pcm.c
+++ b/sound/firewire/motu/motu-pcm.c
@@ -363,7 +363,6 @@ int snd_motu_create_pcm_devices(struct snd_motu *motu)
 		.pointer   = capture_pointer,
 		.ack       = capture_ack,
 		.page      = snd_pcm_lib_get_vmalloc_page,
-		.mmap      = snd_pcm_lib_mmap_vmalloc,
 	};
 	static const struct snd_pcm_ops playback_ops = {
 		.open      = pcm_open,
@@ -376,7 +375,6 @@ int snd_motu_create_pcm_devices(struct snd_motu *motu)
 		.pointer   = playback_pointer,
 		.ack       = playback_ack,
 		.page      = snd_pcm_lib_get_vmalloc_page,
-		.mmap      = snd_pcm_lib_mmap_vmalloc,
 	};
 	struct snd_pcm *pcm;
 	int err;
diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c
index 525b746330be..453fc29fade7 100644
--- a/sound/firewire/motu/motu-protocol-v2.c
+++ b/sound/firewire/motu/motu-protocol-v2.c
@@ -13,6 +13,8 @@
 #define  V2_CLOCK_RATE_SHIFT			3
 #define  V2_CLOCK_SRC_MASK			0x00000007
 #define  V2_CLOCK_SRC_SHIFT			0
+#define  V2_CLOCK_TRAVELER_FETCH_DISABLE	0x04000000
+#define  V2_CLOCK_TRAVELER_FETCH_ENABLE		0x03000000
 
 #define V2_IN_OUT_CONF_OFFSET			0x0c04
 #define  V2_OPT_OUT_IFACE_MASK			0x00000c00
@@ -66,6 +68,11 @@ static int v2_set_clock_rate(struct snd_motu *motu, unsigned int rate)
 	data &= ~V2_CLOCK_RATE_MASK;
 	data |= i << V2_CLOCK_RATE_SHIFT;
 
+	if (motu->spec == &snd_motu_spec_traveler) {
+		data &= ~V2_CLOCK_TRAVELER_FETCH_ENABLE;
+		data |= V2_CLOCK_TRAVELER_FETCH_DISABLE;
+	}
+
 	reg = cpu_to_be32(data);
 	return snd_motu_transaction_write(motu, V2_CLOCK_STATUS_OFFSET, &reg,
 					  sizeof(reg));
@@ -121,8 +128,31 @@ static int v2_get_clock_source(struct snd_motu *motu,
 
 static int v2_switch_fetching_mode(struct snd_motu *motu, bool enable)
 {
-	/* V2 protocol doesn't have this feature. */
-	return 0;
+	__be32 reg;
+	u32 data;
+	int err = 0;
+
+	if (motu->spec == &snd_motu_spec_traveler) {
+		err = snd_motu_transaction_read(motu, V2_CLOCK_STATUS_OFFSET,
+						&reg, sizeof(reg));
+		if (err < 0)
+			return err;
+		data = be32_to_cpu(reg);
+
+		data &= ~(V2_CLOCK_TRAVELER_FETCH_DISABLE |
+			  V2_CLOCK_TRAVELER_FETCH_ENABLE);
+
+		if (enable)
+			data |= V2_CLOCK_TRAVELER_FETCH_ENABLE;
+		else
+			data |= V2_CLOCK_TRAVELER_FETCH_DISABLE;
+
+		reg = cpu_to_be32(data);
+		err = snd_motu_transaction_write(motu, V2_CLOCK_STATUS_OFFSET,
+						 &reg, sizeof(reg));
+	}
+
+	return err;
 }
 
 static void calculate_fixed_part(struct snd_motu_packet_format *formats,
@@ -149,11 +179,20 @@ static void calculate_fixed_part(struct snd_motu_packet_format *formats,
 			pcm_chunks[1] += 2;
 		}
 	} else {
-		/*
-		 * Packets to v2 units transfer main-out-1/2 and phone-out-1/2.
-		 */
-		pcm_chunks[0] += 4;
-		pcm_chunks[1] += 4;
+		if (flags & SND_MOTU_SPEC_RX_SEPARETED_MAIN) {
+			pcm_chunks[0] += 2;
+			pcm_chunks[1] += 2;
+		}
+
+		// Packets to v2 units include 2 chunks for phone 1/2, except
+		// for 176.4/192.0 kHz.
+		pcm_chunks[0] += 2;
+		pcm_chunks[1] += 2;
+	}
+
+	if (flags & SND_MOTU_SPEC_HAS_AESEBU_IFACE) {
+		pcm_chunks[0] += 2;
+		pcm_chunks[1] += 2;
 	}
 
 	/*
@@ -164,19 +203,16 @@ static void calculate_fixed_part(struct snd_motu_packet_format *formats,
 	pcm_chunks[0] += 2;
 	pcm_chunks[1] += 2;
 
-	/* This part should be multiples of 4. */
-	formats->fixed_part_pcm_chunks[0] = round_up(2 + pcm_chunks[0], 4) - 2;
-	formats->fixed_part_pcm_chunks[1] = round_up(2 + pcm_chunks[1], 4) - 2;
-	if (flags & SND_MOTU_SPEC_SUPPORT_CLOCK_X4)
-		formats->fixed_part_pcm_chunks[2] =
-					round_up(2 + pcm_chunks[2], 4) - 2;
+	formats->fixed_part_pcm_chunks[0] = pcm_chunks[0];
+	formats->fixed_part_pcm_chunks[1] = pcm_chunks[1];
+	formats->fixed_part_pcm_chunks[2] = pcm_chunks[2];
 }
 
 static void calculate_differed_part(struct snd_motu_packet_format *formats,
 				    enum snd_motu_spec_flags flags,
 				    u32 data, u32 mask, u32 shift)
 {
-	unsigned char pcm_chunks[3] = {0, 0};
+	unsigned char pcm_chunks[2] = {0, 0};
 
 	/*
 	 * When optical interfaces are configured for S/PDIF (TOSLINK),
diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c
index c7cd9864dc4d..7cc80a05e91f 100644
--- a/sound/firewire/motu/motu-protocol-v3.c
+++ b/sound/firewire/motu/motu-protocol-v3.c
@@ -188,11 +188,20 @@ static void calculate_fixed_part(struct snd_motu_packet_format *formats,
 			pcm_chunks[1] += 2;
 		}
 	} else {
-		/*
-		 * Packets to v2 units transfer main-out-1/2 and phone-out-1/2.
-		 */
-		pcm_chunks[0] += 4;
-		pcm_chunks[1] += 4;
+		if (flags & SND_MOTU_SPEC_RX_SEPARETED_MAIN) {
+			pcm_chunks[0] += 2;
+			pcm_chunks[1] += 2;
+		}
+
+		// Packets to v3 units include 2 chunks for phone 1/2, except
+		// for 176.4/192.0 kHz.
+		pcm_chunks[0] += 2;
+		pcm_chunks[1] += 2;
+	}
+
+	if (flags & SND_MOTU_SPEC_HAS_AESEBU_IFACE) {
+		pcm_chunks[0] += 2;
+		pcm_chunks[1] += 2;
 	}
 
 	/*
diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c
index 0d6b526105ab..300d31b6f191 100644
--- a/sound/firewire/motu/motu.c
+++ b/sound/firewire/motu/motu.c
@@ -200,6 +200,22 @@ static const struct snd_motu_spec motu_828mk2 = {
 	.flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 |
 		 SND_MOTU_SPEC_TX_MICINST_CHUNK |
 		 SND_MOTU_SPEC_TX_RETURN_CHUNK |
+		 SND_MOTU_SPEC_RX_SEPARETED_MAIN |
+		 SND_MOTU_SPEC_HAS_OPT_IFACE_A |
+		 SND_MOTU_SPEC_RX_MIDI_2ND_Q |
+		 SND_MOTU_SPEC_TX_MIDI_2ND_Q,
+
+	.analog_in_ports = 8,
+	.analog_out_ports = 8,
+};
+
+const struct snd_motu_spec snd_motu_spec_traveler = {
+	.name = "Traveler",
+	.protocol = &snd_motu_protocol_v2,
+	.flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 |
+		 SND_MOTU_SPEC_SUPPORT_CLOCK_X4 |
+		 SND_MOTU_SPEC_TX_RETURN_CHUNK |
+		 SND_MOTU_SPEC_HAS_AESEBU_IFACE |
 		 SND_MOTU_SPEC_HAS_OPT_IFACE_A |
 		 SND_MOTU_SPEC_RX_MIDI_2ND_Q |
 		 SND_MOTU_SPEC_TX_MIDI_2ND_Q,
@@ -216,6 +232,7 @@ static const struct snd_motu_spec motu_828mk3 = {
 		 SND_MOTU_SPEC_TX_MICINST_CHUNK |
 		 SND_MOTU_SPEC_TX_RETURN_CHUNK |
 		 SND_MOTU_SPEC_TX_REVERB_CHUNK |
+		 SND_MOTU_SPEC_RX_SEPARETED_MAIN |
 		 SND_MOTU_SPEC_HAS_OPT_IFACE_A |
 		 SND_MOTU_SPEC_HAS_OPT_IFACE_B |
 		 SND_MOTU_SPEC_RX_MIDI_3RD_Q |
@@ -231,6 +248,7 @@ static const struct snd_motu_spec motu_audio_express = {
 	.flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 |
 		 SND_MOTU_SPEC_TX_MICINST_CHUNK |
 		 SND_MOTU_SPEC_TX_RETURN_CHUNK |
+		 SND_MOTU_SPEC_RX_SEPARETED_MAIN |
 		 SND_MOTU_SPEC_RX_MIDI_2ND_Q |
 		 SND_MOTU_SPEC_TX_MIDI_3RD_Q,
 	.analog_in_ports = 2,
@@ -250,6 +268,7 @@ static const struct snd_motu_spec motu_audio_express = {
 
 static const struct ieee1394_device_id motu_id_table[] = {
 	SND_MOTU_DEV_ENTRY(0x101800, &motu_828mk2),
+	SND_MOTU_DEV_ENTRY(0x107800, &snd_motu_spec_traveler),
 	SND_MOTU_DEV_ENTRY(0x106800, &motu_828mk3),	/* FireWire only. */
 	SND_MOTU_DEV_ENTRY(0x100800, &motu_828mk3),	/* Hybrid. */
 	SND_MOTU_DEV_ENTRY(0x104800, &motu_audio_express),
diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h
index 4b23cf337c4b..fd5327d30ab1 100644
--- a/sound/firewire/motu/motu.h
+++ b/sound/firewire/motu/motu.h
@@ -79,13 +79,14 @@ enum snd_motu_spec_flags {
 	SND_MOTU_SPEC_TX_MICINST_CHUNK	= 0x0004,
 	SND_MOTU_SPEC_TX_RETURN_CHUNK	= 0x0008,
 	SND_MOTU_SPEC_TX_REVERB_CHUNK	= 0x0010,
-	SND_MOTU_SPEC_TX_AESEBU_CHUNK	= 0x0020,
+	SND_MOTU_SPEC_HAS_AESEBU_IFACE	= 0x0020,
 	SND_MOTU_SPEC_HAS_OPT_IFACE_A	= 0x0040,
 	SND_MOTU_SPEC_HAS_OPT_IFACE_B	= 0x0080,
 	SND_MOTU_SPEC_RX_MIDI_2ND_Q	= 0x0100,
 	SND_MOTU_SPEC_RX_MIDI_3RD_Q	= 0x0200,
 	SND_MOTU_SPEC_TX_MIDI_2ND_Q	= 0x0400,
 	SND_MOTU_SPEC_TX_MIDI_3RD_Q	= 0x0800,
+	SND_MOTU_SPEC_RX_SEPARETED_MAIN	= 0x1000,
 };
 
 #define SND_MOTU_CLOCK_RATE_COUNT	6
@@ -128,6 +129,8 @@ struct snd_motu_spec {
 extern const struct snd_motu_protocol snd_motu_protocol_v2;
 extern const struct snd_motu_protocol snd_motu_protocol_v3;
 
+extern const struct snd_motu_spec snd_motu_spec_traveler;
+
 int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit,
 		    enum amdtp_stream_direction dir,
 		    const struct snd_motu_protocol *const protocol);
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c
index 3dd46285c0e2..b3f6503dd34d 100644
--- a/sound/firewire/oxfw/oxfw-pcm.c
+++ b/sound/firewire/oxfw/oxfw-pcm.c
@@ -389,7 +389,6 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw)
 		.pointer   = pcm_capture_pointer,
 		.ack       = pcm_capture_ack,
 		.page      = snd_pcm_lib_get_vmalloc_page,
-		.mmap      = snd_pcm_lib_mmap_vmalloc,
 	};
 	static const struct snd_pcm_ops playback_ops = {
 		.open      = pcm_open,
@@ -402,7 +401,6 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw)
 		.pointer   = pcm_playback_pointer,
 		.ack       = pcm_playback_ack,
 		.page      = snd_pcm_lib_get_vmalloc_page,
-		.mmap      = snd_pcm_lib_mmap_vmalloc,
 	};
 	struct snd_pcm *pcm;
 	unsigned int cap = 0;
diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c
index 6ec8ec634d4d..e4cc8990e195 100644
--- a/sound/firewire/tascam/tascam-pcm.c
+++ b/sound/firewire/tascam/tascam-pcm.c
@@ -279,7 +279,6 @@ int snd_tscm_create_pcm_devices(struct snd_tscm *tscm)
 		.pointer	= pcm_playback_pointer,
 		.ack		= pcm_playback_ack,
 		.page		= snd_pcm_lib_get_vmalloc_page,
-		.mmap		= snd_pcm_lib_mmap_vmalloc,
 	};
 	struct snd_pcm *pcm;
 	int err;
diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig
index 3129546398d0..2d90e11b3eaa 100644
--- a/sound/hda/Kconfig
+++ b/sound/hda/Kconfig
@@ -5,11 +5,12 @@ config SND_HDA_CORE
 config SND_HDA_DSP_LOADER
 	bool
 
+config SND_HDA_COMPONENT
+	bool
+
 config SND_HDA_I915
 	bool
-	default y
-	depends on DRM_I915
-	depends on SND_HDA_CORE
+	select SND_HDA_COMPONENT
 
 config SND_HDA_EXT_CORE
        tristate
diff --git a/sound/hda/Makefile b/sound/hda/Makefile
index e4e726f2ce98..2160202e2dc1 100644
--- a/sound/hda/Makefile
+++ b/sound/hda/Makefile
@@ -6,6 +6,7 @@ snd-hda-core-objs += trace.o
 CFLAGS_trace.o := -I$(src)
 
 # for sync with i915 gfx driver
+snd-hda-core-$(CONFIG_SND_HDA_COMPONENT) += hdac_component.o
 snd-hda-core-$(CONFIG_SND_HDA_I915) += hdac_i915.o
 
 obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o
diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c
index 0daf31383084..9c37d9af3023 100644
--- a/sound/hda/ext/hdac_ext_bus.c
+++ b/sound/hda/ext/hdac_ext_bus.c
@@ -87,9 +87,10 @@ static const struct hdac_io_ops hdac_ext_default_io = {
  *
  * Returns 0 if successful, or a negative error code.
  */
-int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev,
+int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev,
 			const struct hdac_bus_ops *ops,
-			const struct hdac_io_ops *io_ops)
+			const struct hdac_io_ops *io_ops,
+			const struct hdac_ext_bus_ops *ext_ops)
 {
 	int ret;
 	static int idx;
@@ -98,15 +99,16 @@ int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev,
 	if (io_ops == NULL)
 		io_ops = &hdac_ext_default_io;
 
-	ret = snd_hdac_bus_init(&ebus->bus, dev, ops, io_ops);
+	ret = snd_hdac_bus_init(bus, dev, ops, io_ops);
 	if (ret < 0)
 		return ret;
 
-	INIT_LIST_HEAD(&ebus->hlink_list);
-	ebus->idx = idx++;
+	bus->ext_ops = ext_ops;
+	INIT_LIST_HEAD(&bus->hlink_list);
+	bus->idx = idx++;
 
-	mutex_init(&ebus->lock);
-	ebus->cmd_dma_state = true;
+	mutex_init(&bus->lock);
+	bus->cmd_dma_state = true;
 
 	return 0;
 }
@@ -116,10 +118,10 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init);
  * snd_hdac_ext_bus_exit - clean up a HD-audio extended bus
  * @ebus: the pointer to extended bus object
  */
-void snd_hdac_ext_bus_exit(struct hdac_ext_bus *ebus)
+void snd_hdac_ext_bus_exit(struct hdac_bus *bus)
 {
-	snd_hdac_bus_exit(&ebus->bus);
-	WARN_ON(!list_empty(&ebus->hlink_list));
+	snd_hdac_bus_exit(bus);
+	WARN_ON(!list_empty(&bus->hlink_list));
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_exit);
 
@@ -135,21 +137,15 @@ static void default_release(struct device *dev)
  *
  * Returns zero for success or a negative error code.
  */
-int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr)
+int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr,
+					struct hdac_device *hdev)
 {
-	struct hdac_ext_device *edev;
-	struct hdac_device *hdev = NULL;
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
 	char name[15];
 	int ret;
 
-	edev = kzalloc(sizeof(*edev), GFP_KERNEL);
-	if (!edev)
-		return -ENOMEM;
-	hdev = &edev->hdev;
-	edev->ebus = ebus;
+	hdev->bus = bus;
 
-	snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr);
+	snprintf(name, sizeof(name), "ehdaudio%dD%d", bus->idx, addr);
 
 	ret  = snd_hdac_device_init(hdev, bus, name, addr);
 	if (ret < 0) {
@@ -176,10 +172,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_init);
  */
 void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev)
 {
-	struct hdac_ext_device *edev = to_ehdac_device(hdev);
-
 	snd_hdac_device_exit(hdev);
-	kfree(edev);
+	kfree(hdev);
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_exit);
 
@@ -188,14 +182,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_exit);
  *
  * @ebus: HD-audio extended bus
  */
-void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus)
+void snd_hdac_ext_bus_device_remove(struct hdac_bus *bus)
 {
 	struct hdac_device *codec, *__codec;
 	/*
 	 * we need to remove all the codec devices objects created in the
 	 * snd_hdac_ext_bus_device_init
 	 */
-	list_for_each_entry_safe(codec, __codec, &ebus->bus.codec_list, list) {
+	list_for_each_entry_safe(codec, __codec, &bus->codec_list, list) {
 		snd_hdac_device_unregister(codec);
 		put_device(&codec->dev);
 	}
@@ -204,35 +198,31 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_remove);
 #define dev_to_hdac(dev) (container_of((dev), \
 			struct hdac_device, dev))
 
-static inline struct hdac_ext_driver *get_edrv(struct device *dev)
+static inline struct hdac_driver *get_hdrv(struct device *dev)
 {
 	struct hdac_driver *hdrv = drv_to_hdac_driver(dev->driver);
-	struct hdac_ext_driver *edrv = to_ehdac_driver(hdrv);
-
-	return edrv;
+	return hdrv;
 }
 
-static inline struct hdac_ext_device *get_edev(struct device *dev)
+static inline struct hdac_device *get_hdev(struct device *dev)
 {
 	struct hdac_device *hdev = dev_to_hdac_dev(dev);
-	struct hdac_ext_device *edev = to_ehdac_device(hdev);
-
-	return edev;
+	return hdev;
 }
 
 static int hda_ext_drv_probe(struct device *dev)
 {
-	return (get_edrv(dev))->probe(get_edev(dev));
+	return (get_hdrv(dev))->probe(get_hdev(dev));
 }
 
 static int hdac_ext_drv_remove(struct device *dev)
 {
-	return (get_edrv(dev))->remove(get_edev(dev));
+	return (get_hdrv(dev))->remove(get_hdev(dev));
 }
 
 static void hdac_ext_drv_shutdown(struct device *dev)
 {
-	return (get_edrv(dev))->shutdown(get_edev(dev));
+	return (get_hdrv(dev))->shutdown(get_hdev(dev));
 }
 
 /**
@@ -240,20 +230,20 @@ static void hdac_ext_drv_shutdown(struct device *dev)
  *
  * @drv: ext hda driver structure
  */
-int snd_hda_ext_driver_register(struct hdac_ext_driver *drv)
+int snd_hda_ext_driver_register(struct hdac_driver *drv)
 {
-	drv->hdac.type = HDA_DEV_ASOC;
-	drv->hdac.driver.bus = &snd_hda_bus_type;
+	drv->type = HDA_DEV_ASOC;
+	drv->driver.bus = &snd_hda_bus_type;
 	/* we use default match */
 
 	if (drv->probe)
-		drv->hdac.driver.probe = hda_ext_drv_probe;
+		drv->driver.probe = hda_ext_drv_probe;
 	if (drv->remove)
-		drv->hdac.driver.remove = hdac_ext_drv_remove;
+		drv->driver.remove = hdac_ext_drv_remove;
 	if (drv->shutdown)
-		drv->hdac.driver.shutdown = hdac_ext_drv_shutdown;
+		drv->driver.shutdown = hdac_ext_drv_shutdown;
 
-	return driver_register(&drv->hdac.driver);
+	return driver_register(&drv->driver);
 }
 EXPORT_SYMBOL_GPL(snd_hda_ext_driver_register);
 
@@ -262,8 +252,8 @@ EXPORT_SYMBOL_GPL(snd_hda_ext_driver_register);
  *
  * @drv: ext hda driver structure
  */
-void snd_hda_ext_driver_unregister(struct hdac_ext_driver *drv)
+void snd_hda_ext_driver_unregister(struct hdac_driver *drv)
 {
-	driver_unregister(&drv->hdac.driver);
+	driver_unregister(&drv->driver);
 }
 EXPORT_SYMBOL_GPL(snd_hda_ext_driver_unregister);
diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c
index 84f3b8168716..5bc4a1d587d4 100644
--- a/sound/hda/ext/hdac_ext_controller.c
+++ b/sound/hda/ext/hdac_ext_controller.c
@@ -39,9 +39,8 @@
  * @ebus: HD-audio extended core bus
  * @enable: flag to turn on/off the capability
  */
-void snd_hdac_ext_bus_ppcap_enable(struct hdac_ext_bus *ebus, bool enable)
+void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *bus, bool enable)
 {
-	struct hdac_bus *bus = &ebus->bus;
 
 	if (!bus->ppcap) {
 		dev_err(bus->dev, "Address of PP capability is NULL");
@@ -60,9 +59,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_enable);
  * @ebus: HD-audio extended core bus
  * @enable: flag to enable/disable interrupt
  */
-void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_ext_bus *ebus, bool enable)
+void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *bus, bool enable)
 {
-	struct hdac_bus *bus = &ebus->bus;
 
 	if (!bus->ppcap) {
 		dev_err(bus->dev, "Address of PP capability is NULL\n");
@@ -89,12 +87,11 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_int_enable);
  * in hlink_list of extended hdac bus
  * Note: this will be freed on bus exit by driver
  */
-int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus)
+int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_bus *bus)
 {
 	int idx;
 	u32 link_count;
 	struct hdac_ext_link *hlink;
-	struct hdac_bus *bus = &ebus->bus;
 
 	link_count = readl(bus->mlcap + AZX_REG_ML_MLCD) + 1;
 
@@ -114,7 +111,7 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus)
 		/* since link in On, update the ref */
 		hlink->ref_count = 1;
 
-		list_add_tail(&hlink->list, &ebus->hlink_list);
+		list_add_tail(&hlink->list, &bus->hlink_list);
 	}
 
 	return 0;
@@ -127,12 +124,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_get_ml_capabilities);
  * @ebus: HD-audio ext core bus
  */
 
-void snd_hdac_link_free_all(struct hdac_ext_bus *ebus)
+void snd_hdac_link_free_all(struct hdac_bus *bus)
 {
 	struct hdac_ext_link *l;
 
-	while (!list_empty(&ebus->hlink_list)) {
-		l = list_first_entry(&ebus->hlink_list, struct hdac_ext_link, list);
+	while (!list_empty(&bus->hlink_list)) {
+		l = list_first_entry(&bus->hlink_list, struct hdac_ext_link, list);
 		list_del(&l->list);
 		kfree(l);
 	}
@@ -144,7 +141,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_link_free_all);
  * @ebus: HD-audio extended core bus
  * @codec_name: codec name
  */
-struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *ebus,
+struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_bus *bus,
 						 const char *codec_name)
 {
 	int i;
@@ -153,10 +150,10 @@ struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *ebus,
 
 	if (sscanf(codec_name, "ehdaudio%dD%d", &bus_idx, &addr) != 2)
 		return NULL;
-	if (ebus->idx != bus_idx)
+	if (bus->idx != bus_idx)
 		return NULL;
 
-	list_for_each_entry(hlink, &ebus->hlink_list, list) {
+	list_for_each_entry(hlink, &bus->hlink_list, list) {
 		for (i = 0; i < HDA_MAX_CODECS; i++) {
 			if (hlink->lsdiid & (0x1 << addr))
 				return hlink;
@@ -219,12 +216,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down);
  * snd_hdac_ext_bus_link_power_up_all -power up all hda link
  * @ebus: HD-audio extended bus
  */
-int snd_hdac_ext_bus_link_power_up_all(struct hdac_ext_bus *ebus)
+int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus)
 {
 	struct hdac_ext_link *hlink = NULL;
 	int ret;
 
-	list_for_each_entry(hlink, &ebus->hlink_list, list) {
+	list_for_each_entry(hlink, &bus->hlink_list, list) {
 		snd_hdac_updatel(hlink->ml_addr,
 				AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA);
 		ret = check_hdac_link_power_active(hlink, true);
@@ -240,12 +237,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_up_all);
  * snd_hdac_ext_bus_link_power_down_all -power down all hda link
  * @ebus: HD-audio extended bus
  */
-int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus)
+int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus)
 {
 	struct hdac_ext_link *hlink = NULL;
 	int ret;
 
-	list_for_each_entry(hlink, &ebus->hlink_list, list) {
+	list_for_each_entry(hlink, &bus->hlink_list, list) {
 		snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0);
 		ret = check_hdac_link_power_active(hlink, false);
 		if (ret < 0)
@@ -256,39 +253,48 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus)
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all);
 
-int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus,
+int snd_hdac_ext_bus_link_get(struct hdac_bus *bus,
 				struct hdac_ext_link *link)
 {
 	int ret = 0;
 
-	mutex_lock(&ebus->lock);
+	mutex_lock(&bus->lock);
 
 	/*
 	 * if we move from 0 to 1, count will be 1 so power up this link
 	 * as well, also check the dma status and trigger that
 	 */
 	if (++link->ref_count == 1) {
-		if (!ebus->cmd_dma_state) {
-			snd_hdac_bus_init_cmd_io(&ebus->bus);
-			ebus->cmd_dma_state = true;
+		if (!bus->cmd_dma_state) {
+			snd_hdac_bus_init_cmd_io(bus);
+			bus->cmd_dma_state = true;
 		}
 
 		ret = snd_hdac_ext_bus_link_power_up(link);
+
+		/*
+		 *  wait for 521usec for codec to report status
+		 *  HDA spec section 4.3 - Codec Discovery
+		 */
+		udelay(521);
+		bus->codec_mask = snd_hdac_chip_readw(bus, STATESTS);
+		dev_dbg(bus->dev, "codec_mask = 0x%lx\n", bus->codec_mask);
+		snd_hdac_chip_writew(bus, STATESTS, bus->codec_mask);
 	}
 
-	mutex_unlock(&ebus->lock);
+	mutex_unlock(&bus->lock);
 	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_get);
 
-int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus,
+int snd_hdac_ext_bus_link_put(struct hdac_bus *bus,
 				struct hdac_ext_link *link)
 {
 	int ret = 0;
 	struct hdac_ext_link *hlink;
 	bool link_up = false;
 
-	mutex_lock(&ebus->lock);
+	mutex_lock(&bus->lock);
 
 	/*
 	 * if we move from 1 to 0, count will be 0
@@ -301,7 +307,7 @@ int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus,
 		 * now check if all links are off, if so turn off
 		 * cmd dma as well
 		 */
-		list_for_each_entry(hlink, &ebus->hlink_list, list) {
+		list_for_each_entry(hlink, &bus->hlink_list, list) {
 			if (hlink->ref_count) {
 				link_up = true;
 				break;
@@ -309,12 +315,12 @@ int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus,
 		}
 
 		if (!link_up) {
-			snd_hdac_bus_stop_cmd_io(&ebus->bus);
-			ebus->cmd_dma_state = false;
+			snd_hdac_bus_stop_cmd_io(bus);
+			bus->cmd_dma_state = false;
 		}
 	}
 
-	mutex_unlock(&ebus->lock);
+	mutex_unlock(&bus->lock);
 	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_put);
diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c
index c96d7a7a36af..1bd27576db98 100644
--- a/sound/hda/ext/hdac_ext_stream.c
+++ b/sound/hda/ext/hdac_ext_stream.c
@@ -25,7 +25,7 @@
 
 /**
  * snd_hdac_ext_stream_init - initialize each stream (aka device)
- * @ebus: HD-audio ext core bus
+ * @bus: HD-audio core bus
  * @stream: HD-audio ext core stream object to initialize
  * @idx: stream index number
  * @direction: stream direction (SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE)
@@ -34,18 +34,16 @@
  * initialize the stream, if ppcap is enabled then init those and then
  * invoke hdac stream initialization routine
  */
-void snd_hdac_ext_stream_init(struct hdac_ext_bus *ebus,
+void snd_hdac_ext_stream_init(struct hdac_bus *bus,
 				struct hdac_ext_stream *stream,
 				int idx, int direction, int tag)
 {
-	struct hdac_bus *bus = &ebus->bus;
-
 	if (bus->ppcap) {
 		stream->pphc_addr = bus->ppcap + AZX_PPHC_BASE +
 				AZX_PPHC_INTERVAL * idx;
 
 		stream->pplc_addr = bus->ppcap + AZX_PPLC_BASE +
-				AZX_PPLC_MULTI * ebus->num_streams +
+				AZX_PPLC_MULTI * bus->num_streams +
 				AZX_PPLC_INTERVAL * idx;
 	}
 
@@ -71,12 +69,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init);
 /**
  * snd_hdac_ext_stream_init_all - create and initialize the stream objects
  *   for an extended hda bus
- * @ebus: HD-audio ext core bus
+ * @bus: HD-audio core bus
  * @start_idx: start index for streams
  * @num_stream: number of streams to initialize
  * @dir: direction of streams
  */
-int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx,
+int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx,
 		int num_stream, int dir)
 {
 	int stream_tag = 0;
@@ -88,7 +86,7 @@ int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx,
 		if (!stream)
 			return -ENOMEM;
 		tag = ++stream_tag;
-		snd_hdac_ext_stream_init(ebus, stream, idx, dir, tag);
+		snd_hdac_ext_stream_init(bus, stream, idx, dir, tag);
 		idx++;
 	}
 
@@ -100,17 +98,16 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init_all);
 /**
  * snd_hdac_stream_free_all - free hdac extended stream objects
  *
- * @ebus: HD-audio ext core bus
+ * @bus: HD-audio core bus
  */
-void snd_hdac_stream_free_all(struct hdac_ext_bus *ebus)
+void snd_hdac_stream_free_all(struct hdac_bus *bus)
 {
 	struct hdac_stream *s, *_s;
 	struct hdac_ext_stream *stream;
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
 
 	list_for_each_entry_safe(s, _s, &bus->stream_list, list) {
 		stream = stream_to_hdac_ext_stream(s);
-		snd_hdac_ext_stream_decouple(ebus, stream, false);
+		snd_hdac_ext_stream_decouple(bus, stream, false);
 		list_del(&s->list);
 		kfree(stream);
 	}
@@ -119,15 +116,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_stream_free_all);
 
 /**
  * snd_hdac_ext_stream_decouple - decouple the hdac stream
- * @ebus: HD-audio ext core bus
+ * @bus: HD-audio core bus
  * @stream: HD-audio ext core stream object to initialize
  * @decouple: flag to decouple
  */
-void snd_hdac_ext_stream_decouple(struct hdac_ext_bus *ebus,
+void snd_hdac_ext_stream_decouple(struct hdac_bus *bus,
 				struct hdac_ext_stream *stream, bool decouple)
 {
 	struct hdac_stream *hstream = &stream->hstream;
-	struct hdac_bus *bus = &ebus->bus;
 	u32 val;
 	int mask = AZX_PPCTL_PROCEN(hstream->index);
 
@@ -251,19 +247,18 @@ void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link,
 EXPORT_SYMBOL_GPL(snd_hdac_ext_link_clear_stream_id);
 
 static struct hdac_ext_stream *
-hdac_ext_link_stream_assign(struct hdac_ext_bus *ebus,
+hdac_ext_link_stream_assign(struct hdac_bus *bus,
 				struct snd_pcm_substream *substream)
 {
 	struct hdac_ext_stream *res = NULL;
 	struct hdac_stream *stream = NULL;
-	struct hdac_bus *hbus = &ebus->bus;
 
-	if (!hbus->ppcap) {
-		dev_err(hbus->dev, "stream type not supported\n");
+	if (!bus->ppcap) {
+		dev_err(bus->dev, "stream type not supported\n");
 		return NULL;
 	}
 
-	list_for_each_entry(stream, &hbus->stream_list, list) {
+	list_for_each_entry(stream, &bus->stream_list, list) {
 		struct hdac_ext_stream *hstream = container_of(stream,
 						struct hdac_ext_stream,
 						hstream);
@@ -277,34 +272,33 @@ hdac_ext_link_stream_assign(struct hdac_ext_bus *ebus,
 		}
 
 		if (!hstream->link_locked) {
-			snd_hdac_ext_stream_decouple(ebus, hstream, true);
+			snd_hdac_ext_stream_decouple(bus, hstream, true);
 			res = hstream;
 			break;
 		}
 	}
 	if (res) {
-		spin_lock_irq(&hbus->reg_lock);
+		spin_lock_irq(&bus->reg_lock);
 		res->link_locked = 1;
 		res->link_substream = substream;
-		spin_unlock_irq(&hbus->reg_lock);
+		spin_unlock_irq(&bus->reg_lock);
 	}
 	return res;
 }
 
 static struct hdac_ext_stream *
-hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus,
+hdac_ext_host_stream_assign(struct hdac_bus *bus,
 				struct snd_pcm_substream *substream)
 {
 	struct hdac_ext_stream *res = NULL;
 	struct hdac_stream *stream = NULL;
-	struct hdac_bus *hbus = &ebus->bus;
 
-	if (!hbus->ppcap) {
-		dev_err(hbus->dev, "stream type not supported\n");
+	if (!bus->ppcap) {
+		dev_err(bus->dev, "stream type not supported\n");
 		return NULL;
 	}
 
-	list_for_each_entry(stream, &hbus->stream_list, list) {
+	list_for_each_entry(stream, &bus->stream_list, list) {
 		struct hdac_ext_stream *hstream = container_of(stream,
 						struct hdac_ext_stream,
 						hstream);
@@ -313,17 +307,17 @@ hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus,
 
 		if (!stream->opened) {
 			if (!hstream->decoupled)
-				snd_hdac_ext_stream_decouple(ebus, hstream, true);
+				snd_hdac_ext_stream_decouple(bus, hstream, true);
 			res = hstream;
 			break;
 		}
 	}
 	if (res) {
-		spin_lock_irq(&hbus->reg_lock);
+		spin_lock_irq(&bus->reg_lock);
 		res->hstream.opened = 1;
 		res->hstream.running = 0;
 		res->hstream.substream = substream;
-		spin_unlock_irq(&hbus->reg_lock);
+		spin_unlock_irq(&bus->reg_lock);
 	}
 
 	return res;
@@ -331,7 +325,7 @@ hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus,
 
 /**
  * snd_hdac_ext_stream_assign - assign a stream for the PCM
- * @ebus: HD-audio ext core bus
+ * @bus: HD-audio core bus
  * @substream: PCM substream to assign
  * @type: type of stream (coupled, host or link stream)
  *
@@ -346,27 +340,26 @@ hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus,
  * the same stream object when it's used beforehand.  when a stream is
  * decoupled, it becomes a host stream and link stream.
  */
-struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_ext_bus *ebus,
+struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_bus *bus,
 					   struct snd_pcm_substream *substream,
 					   int type)
 {
 	struct hdac_ext_stream *hstream = NULL;
 	struct hdac_stream *stream = NULL;
-	struct hdac_bus *hbus = &ebus->bus;
 
 	switch (type) {
 	case HDAC_EXT_STREAM_TYPE_COUPLED:
-		stream = snd_hdac_stream_assign(hbus, substream);
+		stream = snd_hdac_stream_assign(bus, substream);
 		if (stream)
 			hstream = container_of(stream,
 					struct hdac_ext_stream, hstream);
 		return hstream;
 
 	case HDAC_EXT_STREAM_TYPE_HOST:
-		return hdac_ext_host_stream_assign(ebus, substream);
+		return hdac_ext_host_stream_assign(bus, substream);
 
 	case HDAC_EXT_STREAM_TYPE_LINK:
-		return hdac_ext_link_stream_assign(ebus, substream);
+		return hdac_ext_link_stream_assign(bus, substream);
 
 	default:
 		return NULL;
@@ -384,7 +377,6 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_assign);
 void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type)
 {
 	struct hdac_bus *bus = stream->hstream.bus;
-	struct hdac_ext_bus *ebus = hbus_to_ebus(bus);
 
 	switch (type) {
 	case HDAC_EXT_STREAM_TYPE_COUPLED:
@@ -393,13 +385,13 @@ void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type)
 
 	case HDAC_EXT_STREAM_TYPE_HOST:
 		if (stream->decoupled && !stream->link_locked)
-			snd_hdac_ext_stream_decouple(ebus, stream, false);
+			snd_hdac_ext_stream_decouple(bus, stream, false);
 		snd_hdac_stream_release(&stream->hstream);
 		break;
 
 	case HDAC_EXT_STREAM_TYPE_LINK:
 		if (stream->decoupled && !stream->hstream.opened)
-			snd_hdac_ext_stream_decouple(ebus, stream, false);
+			snd_hdac_ext_stream_decouple(bus, stream, false);
 		spin_lock_irq(&bus->reg_lock);
 		stream->link_locked = 0;
 		stream->link_substream = NULL;
@@ -415,16 +407,15 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_release);
 
 /**
  * snd_hdac_ext_stream_spbcap_enable - enable SPIB for a stream
- * @ebus: HD-audio ext core bus
+ * @bus: HD-audio core bus
  * @enable: flag to enable/disable SPIB
  * @index: stream index for which SPIB need to be enabled
  */
-void snd_hdac_ext_stream_spbcap_enable(struct hdac_ext_bus *ebus,
+void snd_hdac_ext_stream_spbcap_enable(struct hdac_bus *bus,
 				 bool enable, int index)
 {
 	u32 mask = 0;
 	u32 register_mask = 0;
-	struct hdac_bus *bus = &ebus->bus;
 
 	if (!bus->spbcap) {
 		dev_err(bus->dev, "Address of SPB capability is NULL\n");
@@ -446,14 +437,13 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_spbcap_enable);
 
 /**
  * snd_hdac_ext_stream_set_spib - sets the spib value of a stream
- * @ebus: HD-audio ext core bus
+ * @bus: HD-audio core bus
  * @stream: hdac_ext_stream
  * @value: spib value to set
  */
-int snd_hdac_ext_stream_set_spib(struct hdac_ext_bus *ebus,
+int snd_hdac_ext_stream_set_spib(struct hdac_bus *bus,
 				 struct hdac_ext_stream *stream, u32 value)
 {
-	struct hdac_bus *bus = &ebus->bus;
 
 	if (!bus->spbcap) {
 		dev_err(bus->dev, "Address of SPB capability is NULL\n");
@@ -468,15 +458,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_set_spib);
 
 /**
  * snd_hdac_ext_stream_get_spbmaxfifo - gets the spib value of a stream
- * @ebus: HD-audio ext core bus
+ * @bus: HD-audio core bus
  * @stream: hdac_ext_stream
  *
  * Return maxfifo for the stream
  */
-int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_ext_bus *ebus,
+int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_bus *bus,
 				 struct hdac_ext_stream *stream)
 {
-	struct hdac_bus *bus = &ebus->bus;
 
 	if (!bus->spbcap) {
 		dev_err(bus->dev, "Address of SPB capability is NULL\n");
@@ -490,11 +479,10 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_get_spbmaxfifo);
 
 /**
  * snd_hdac_ext_stop_streams - stop all stream if running
- * @ebus: HD-audio ext core bus
+ * @bus: HD-audio core bus
  */
-void snd_hdac_ext_stop_streams(struct hdac_ext_bus *ebus)
+void snd_hdac_ext_stop_streams(struct hdac_bus *bus)
 {
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
 	struct hdac_stream *stream;
 
 	if (bus->chip_init) {
@@ -507,16 +495,15 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stop_streams);
 
 /**
  * snd_hdac_ext_stream_drsm_enable - enable DMA resume for a stream
- * @ebus: HD-audio ext core bus
+ * @bus: HD-audio core bus
  * @enable: flag to enable/disable DRSM
  * @index: stream index for which DRSM need to be enabled
  */
-void snd_hdac_ext_stream_drsm_enable(struct hdac_ext_bus *ebus,
+void snd_hdac_ext_stream_drsm_enable(struct hdac_bus *bus,
 				bool enable, int index)
 {
 	u32 mask = 0;
 	u32 register_mask = 0;
-	struct hdac_bus *bus = &ebus->bus;
 
 	if (!bus->drsmcap) {
 		dev_err(bus->dev, "Address of DRSM capability is NULL\n");
@@ -538,14 +525,13 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_drsm_enable);
 
 /**
  * snd_hdac_ext_stream_set_dpibr - sets the dpibr value of a stream
- * @ebus: HD-audio ext core bus
+ * @bus: HD-audio core bus
  * @stream: hdac_ext_stream
  * @value: dpib value to set
  */
-int snd_hdac_ext_stream_set_dpibr(struct hdac_ext_bus *ebus,
+int snd_hdac_ext_stream_set_dpibr(struct hdac_bus *bus,
 				 struct hdac_ext_stream *stream, u32 value)
 {
-	struct hdac_bus *bus = &ebus->bus;
 
 	if (!bus->drsmcap) {
 		dev_err(bus->dev, "Address of DRSM capability is NULL\n");
@@ -560,7 +546,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_set_dpibr);
 
 /**
  * snd_hdac_ext_stream_set_lpib - sets the lpib value of a stream
- * @ebus: HD-audio ext core bus
+ * @bus: HD-audio core bus
  * @stream: hdac_ext_stream
  * @value: lpib value to set
  */
diff --git a/sound/hda/hdac_component.c b/sound/hda/hdac_component.c
new file mode 100644
index 000000000000..6e46a9c73aed
--- /dev/null
+++ b/sound/hda/hdac_component.c
@@ -0,0 +1,335 @@
+// SPDX-License-Identifier: GPL-2.0
+// hdac_component.c - routines for sync between HD-A core and DRM driver
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/pci.h>
+#include <linux/component.h>
+#include <sound/core.h>
+#include <sound/hdaudio.h>
+#include <sound/hda_component.h>
+#include <sound/hda_register.h>
+
+static void hdac_acomp_release(struct device *dev, void *res)
+{
+}
+
+static struct drm_audio_component *hdac_get_acomp(struct device *dev)
+{
+	return devres_find(dev, hdac_acomp_release, NULL, NULL);
+}
+
+/**
+ * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup
+ * @bus: HDA core bus
+ * @enable: enable or disable the wakeup
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with graphics driver.
+ *
+ * This function should be called during the chip reset, also called at
+ * resume for updating STATESTS register read.
+ *
+ * Returns zero for success or a negative error code.
+ */
+int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable)
+{
+	struct drm_audio_component *acomp = bus->audio_component;
+
+	if (!acomp || !acomp->ops)
+		return -ENODEV;
+
+	if (!acomp->ops->codec_wake_override)
+		return 0;
+
+	dev_dbg(bus->dev, "%s codec wakeup\n",
+		enable ? "enable" : "disable");
+
+	acomp->ops->codec_wake_override(acomp->dev, enable);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup);
+
+/**
+ * snd_hdac_display_power - Power up / down the power refcount
+ * @bus: HDA core bus
+ * @enable: power up or down
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with graphics driver.
+ *
+ * This function manages a refcount and calls the get_power() and
+ * put_power() ops accordingly, toggling the codec wakeup, too.
+ *
+ * Returns zero for success or a negative error code.
+ */
+int snd_hdac_display_power(struct hdac_bus *bus, bool enable)
+{
+	struct drm_audio_component *acomp = bus->audio_component;
+
+	if (!acomp || !acomp->ops)
+		return -ENODEV;
+
+	dev_dbg(bus->dev, "display power %s\n",
+		enable ? "enable" : "disable");
+
+	if (enable) {
+		if (!bus->drm_power_refcount++) {
+			if (acomp->ops->get_power)
+				acomp->ops->get_power(acomp->dev);
+			snd_hdac_set_codec_wakeup(bus, true);
+			snd_hdac_set_codec_wakeup(bus, false);
+		}
+	} else {
+		WARN_ON(!bus->drm_power_refcount);
+		if (!--bus->drm_power_refcount)
+			if (acomp->ops->put_power)
+				acomp->ops->put_power(acomp->dev);
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_display_power);
+
+/**
+ * snd_hdac_sync_audio_rate - Set N/CTS based on the sample rate
+ * @codec: HDA codec
+ * @nid: the pin widget NID
+ * @dev_id: device identifier
+ * @rate: the sample rate to set
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with graphics driver.
+ *
+ * This function sets N/CTS value based on the given sample rate.
+ * Returns zero for success, or a negative error code.
+ */
+int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid,
+			     int dev_id, int rate)
+{
+	struct hdac_bus *bus = codec->bus;
+	struct drm_audio_component *acomp = bus->audio_component;
+	int port, pipe;
+
+	if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate)
+		return -ENODEV;
+	port = nid;
+	if (acomp->audio_ops && acomp->audio_ops->pin2port) {
+		port = acomp->audio_ops->pin2port(codec, nid);
+		if (port < 0)
+			return -EINVAL;
+	}
+	pipe = dev_id;
+	return acomp->ops->sync_audio_rate(acomp->dev, port, pipe, rate);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate);
+
+/**
+ * snd_hdac_acomp_get_eld - Get the audio state and ELD via component
+ * @codec: HDA codec
+ * @nid: the pin widget NID
+ * @dev_id: device identifier
+ * @audio_enabled: the pointer to store the current audio state
+ * @buffer: the buffer pointer to store ELD bytes
+ * @max_bytes: the max bytes to be stored on @buffer
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with graphics driver.
+ *
+ * This function queries the current state of the audio on the given
+ * digital port and fetches the ELD bytes onto the given buffer.
+ * It returns the number of bytes for the total ELD data, zero for
+ * invalid ELD, or a negative error code.
+ *
+ * The return size is the total bytes required for the whole ELD bytes,
+ * thus it may be over @max_bytes.  If it's over @max_bytes, it implies
+ * that only a part of ELD bytes have been fetched.
+ */
+int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id,
+			   bool *audio_enabled, char *buffer, int max_bytes)
+{
+	struct hdac_bus *bus = codec->bus;
+	struct drm_audio_component *acomp = bus->audio_component;
+	int port, pipe;
+
+	if (!acomp || !acomp->ops || !acomp->ops->get_eld)
+		return -ENODEV;
+
+	port = nid;
+	if (acomp->audio_ops && acomp->audio_ops->pin2port) {
+		port = acomp->audio_ops->pin2port(codec, nid);
+		if (port < 0)
+			return -EINVAL;
+	}
+	pipe = dev_id;
+	return acomp->ops->get_eld(acomp->dev, port, pipe, audio_enabled,
+				   buffer, max_bytes);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld);
+
+static int hdac_component_master_bind(struct device *dev)
+{
+	struct drm_audio_component *acomp = hdac_get_acomp(dev);
+	int ret;
+
+	if (WARN_ON(!acomp))
+		return -EINVAL;
+
+	ret = component_bind_all(dev, acomp);
+	if (ret < 0)
+		return ret;
+
+	if (WARN_ON(!(acomp->dev && acomp->ops))) {
+		ret = -EINVAL;
+		goto out_unbind;
+	}
+
+	/* pin the module to avoid dynamic unbinding, but only if given */
+	if (!try_module_get(acomp->ops->owner)) {
+		ret = -ENODEV;
+		goto out_unbind;
+	}
+
+	if (acomp->audio_ops && acomp->audio_ops->master_bind) {
+		ret = acomp->audio_ops->master_bind(dev, acomp);
+		if (ret < 0)
+			goto module_put;
+	}
+
+	return 0;
+
+ module_put:
+	module_put(acomp->ops->owner);
+out_unbind:
+	component_unbind_all(dev, acomp);
+
+	return ret;
+}
+
+static void hdac_component_master_unbind(struct device *dev)
+{
+	struct drm_audio_component *acomp = hdac_get_acomp(dev);
+
+	if (acomp->audio_ops && acomp->audio_ops->master_unbind)
+		acomp->audio_ops->master_unbind(dev, acomp);
+	module_put(acomp->ops->owner);
+	component_unbind_all(dev, acomp);
+	WARN_ON(acomp->ops || acomp->dev);
+}
+
+static const struct component_master_ops hdac_component_master_ops = {
+	.bind = hdac_component_master_bind,
+	.unbind = hdac_component_master_unbind,
+};
+
+/**
+ * snd_hdac_acomp_register_notifier - Register audio component ops
+ * @bus: HDA core bus
+ * @aops: audio component ops
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with graphics driver.
+ *
+ * This function sets the given ops to be called by the graphics driver.
+ *
+ * Returns zero for success or a negative error code.
+ */
+int snd_hdac_acomp_register_notifier(struct hdac_bus *bus,
+				    const struct drm_audio_component_audio_ops *aops)
+{
+	if (!bus->audio_component)
+		return -ENODEV;
+
+	bus->audio_component->audio_ops = aops;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_acomp_register_notifier);
+
+/**
+ * snd_hdac_acomp_init - Initialize audio component
+ * @bus: HDA core bus
+ * @match_master: match function for finding components
+ * @extra_size: Extra bytes to allocate
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with graphics driver.
+ *
+ * This function initializes and sets up the audio component to communicate
+ * with graphics driver.
+ *
+ * Unlike snd_hdac_i915_init(), this function doesn't synchronize with the
+ * binding with the DRM component.  Each caller needs to sync via master_bind
+ * audio_ops.
+ *
+ * Returns zero for success or a negative error code.
+ */
+int snd_hdac_acomp_init(struct hdac_bus *bus,
+			const struct drm_audio_component_audio_ops *aops,
+			int (*match_master)(struct device *, void *),
+			size_t extra_size)
+{
+	struct component_match *match = NULL;
+	struct device *dev = bus->dev;
+	struct drm_audio_component *acomp;
+	int ret;
+
+	if (WARN_ON(hdac_get_acomp(dev)))
+		return -EBUSY;
+
+	acomp = devres_alloc(hdac_acomp_release, sizeof(*acomp) + extra_size,
+			     GFP_KERNEL);
+	if (!acomp)
+		return -ENOMEM;
+	acomp->audio_ops = aops;
+	bus->audio_component = acomp;
+	devres_add(dev, acomp);
+
+	component_match_add(dev, &match, match_master, bus);
+	ret = component_master_add_with_match(dev, &hdac_component_master_ops,
+					      match);
+	if (ret < 0)
+		goto out_err;
+
+	return 0;
+
+out_err:
+	bus->audio_component = NULL;
+	devres_destroy(dev, hdac_acomp_release, NULL, NULL);
+	dev_info(dev, "failed to add audio component master (%d)\n", ret);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_acomp_init);
+
+/**
+ * snd_hdac_acomp_exit - Finalize audio component
+ * @bus: HDA core bus
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with graphics driver.
+ *
+ * This function releases the audio component that has been used.
+ *
+ * Returns zero for success or a negative error code.
+ */
+int snd_hdac_acomp_exit(struct hdac_bus *bus)
+{
+	struct device *dev = bus->dev;
+	struct drm_audio_component *acomp = bus->audio_component;
+
+	if (!acomp)
+		return 0;
+
+	WARN_ON(bus->drm_power_refcount);
+	if (bus->drm_power_refcount > 0 && acomp->ops)
+		acomp->ops->put_power(acomp->dev);
+
+	component_master_del(dev, &hdac_component_master_ops);
+
+	bus->audio_component = NULL;
+	devres_destroy(dev, hdac_acomp_release, NULL, NULL);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_acomp_exit);
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index 7ba100bb1c3f..dbf02a3a8d2f 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -738,7 +738,7 @@ static struct hda_rate_tbl rate_bits[] = {
  */
 unsigned int snd_hdac_calc_stream_format(unsigned int rate,
 					 unsigned int channels,
-					 unsigned int format,
+					 snd_pcm_format_t format,
 					 unsigned int maxbps,
 					 unsigned short spdif_ctls)
 {
diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index cbe818eda336..b5282cbbe489 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -15,88 +15,12 @@
 #include <linux/init.h>
 #include <linux/module.h>
 #include <linux/pci.h>
-#include <linux/component.h>
-#include <drm/i915_component.h>
 #include <sound/core.h>
 #include <sound/hdaudio.h>
 #include <sound/hda_i915.h>
 #include <sound/hda_register.h>
 
-static struct i915_audio_component *hdac_acomp;
-
-/**
- * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup
- * @bus: HDA core bus
- * @enable: enable or disable the wakeup
- *
- * This function is supposed to be used only by a HD-audio controller
- * driver that needs the interaction with i915 graphics.
- *
- * This function should be called during the chip reset, also called at
- * resume for updating STATESTS register read.
- *
- * Returns zero for success or a negative error code.
- */
-int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable)
-{
-	struct i915_audio_component *acomp = bus->audio_component;
-
-	if (!acomp || !acomp->ops)
-		return -ENODEV;
-
-	if (!acomp->ops->codec_wake_override) {
-		dev_warn(bus->dev,
-			"Invalid codec wake callback\n");
-		return 0;
-	}
-
-	dev_dbg(bus->dev, "%s codec wakeup\n",
-		enable ? "enable" : "disable");
-
-	acomp->ops->codec_wake_override(acomp->dev, enable);
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup);
-
-/**
- * snd_hdac_display_power - Power up / down the power refcount
- * @bus: HDA core bus
- * @enable: power up or down
- *
- * This function is supposed to be used only by a HD-audio controller
- * driver that needs the interaction with i915 graphics.
- *
- * This function manages a refcount and calls the i915 get_power() and
- * put_power() ops accordingly, toggling the codec wakeup, too.
- *
- * Returns zero for success or a negative error code.
- */
-int snd_hdac_display_power(struct hdac_bus *bus, bool enable)
-{
-	struct i915_audio_component *acomp = bus->audio_component;
-
-	if (!acomp || !acomp->ops)
-		return -ENODEV;
-
-	dev_dbg(bus->dev, "display power %s\n",
-		enable ? "enable" : "disable");
-
-	if (enable) {
-		if (!bus->i915_power_refcount++) {
-			acomp->ops->get_power(acomp->dev);
-			snd_hdac_set_codec_wakeup(bus, true);
-			snd_hdac_set_codec_wakeup(bus, false);
-		}
-	} else {
-		WARN_ON(!bus->i915_power_refcount);
-		if (!--bus->i915_power_refcount)
-			acomp->ops->put_power(acomp->dev);
-	}
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(snd_hdac_display_power);
+static struct completion bind_complete;
 
 #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \
 				((pci)->device == 0x0c0c) || \
@@ -119,7 +43,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_display_power);
  */
 void snd_hdac_i915_set_bclk(struct hdac_bus *bus)
 {
-	struct i915_audio_component *acomp = bus->audio_component;
+	struct drm_audio_component *acomp = bus->audio_component;
 	struct pci_dev *pci = to_pci_dev(bus->dev);
 	int cdclk_freq;
 	unsigned int bclk_m, bclk_n;
@@ -158,181 +82,11 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus)
 }
 EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk);
 
-/* There is a fixed mapping between audio pin node and display port.
- * on SNB, IVY, HSW, BSW, SKL, BXT, KBL:
- * Pin Widget 5 - PORT B (port = 1 in i915 driver)
- * Pin Widget 6 - PORT C (port = 2 in i915 driver)
- * Pin Widget 7 - PORT D (port = 3 in i915 driver)
- *
- * on VLV, ILK:
- * Pin Widget 4 - PORT B (port = 1 in i915 driver)
- * Pin Widget 5 - PORT C (port = 2 in i915 driver)
- * Pin Widget 6 - PORT D (port = 3 in i915 driver)
- */
-static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid)
-{
-	int base_nid;
-
-	switch (codec->vendor_id) {
-	case 0x80860054: /* ILK */
-	case 0x80862804: /* ILK */
-	case 0x80862882: /* VLV */
-		base_nid = 3;
-		break;
-	default:
-		base_nid = 4;
-		break;
-	}
-
-	if (WARN_ON(pin_nid <= base_nid || pin_nid > base_nid + 3))
-		return -1;
-	return pin_nid - base_nid;
-}
-
-/**
- * snd_hdac_sync_audio_rate - Set N/CTS based on the sample rate
- * @codec: HDA codec
- * @nid: the pin widget NID
- * @dev_id: device identifier
- * @rate: the sample rate to set
- *
- * This function is supposed to be used only by a HD-audio controller
- * driver that needs the interaction with i915 graphics.
- *
- * This function sets N/CTS value based on the given sample rate.
- * Returns zero for success, or a negative error code.
- */
-int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid,
-			     int dev_id, int rate)
-{
-	struct hdac_bus *bus = codec->bus;
-	struct i915_audio_component *acomp = bus->audio_component;
-	int port, pipe;
-
-	if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate)
-		return -ENODEV;
-	port = pin2port(codec, nid);
-	if (port < 0)
-		return -EINVAL;
-	pipe = dev_id;
-	return acomp->ops->sync_audio_rate(acomp->dev, port, pipe, rate);
-}
-EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate);
-
-/**
- * snd_hdac_acomp_get_eld - Get the audio state and ELD via component
- * @codec: HDA codec
- * @nid: the pin widget NID
- * @dev_id: device identifier
- * @audio_enabled: the pointer to store the current audio state
- * @buffer: the buffer pointer to store ELD bytes
- * @max_bytes: the max bytes to be stored on @buffer
- *
- * This function is supposed to be used only by a HD-audio controller
- * driver that needs the interaction with i915 graphics.
- *
- * This function queries the current state of the audio on the given
- * digital port and fetches the ELD bytes onto the given buffer.
- * It returns the number of bytes for the total ELD data, zero for
- * invalid ELD, or a negative error code.
- *
- * The return size is the total bytes required for the whole ELD bytes,
- * thus it may be over @max_bytes.  If it's over @max_bytes, it implies
- * that only a part of ELD bytes have been fetched.
- */
-int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id,
-			   bool *audio_enabled, char *buffer, int max_bytes)
-{
-	struct hdac_bus *bus = codec->bus;
-	struct i915_audio_component *acomp = bus->audio_component;
-	int port, pipe;
-
-	if (!acomp || !acomp->ops || !acomp->ops->get_eld)
-		return -ENODEV;
-
-	port = pin2port(codec, nid);
-	if (port < 0)
-		return -EINVAL;
-
-	pipe = dev_id;
-	return acomp->ops->get_eld(acomp->dev, port, pipe, audio_enabled,
-				   buffer, max_bytes);
-}
-EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld);
-
-static int hdac_component_master_bind(struct device *dev)
-{
-	struct i915_audio_component *acomp = hdac_acomp;
-	int ret;
-
-	ret = component_bind_all(dev, acomp);
-	if (ret < 0)
-		return ret;
-
-	if (WARN_ON(!(acomp->dev && acomp->ops && acomp->ops->get_power &&
-		      acomp->ops->put_power && acomp->ops->get_cdclk_freq))) {
-		ret = -EINVAL;
-		goto out_unbind;
-	}
-
-	/*
-	 * Atm, we don't support dynamic unbinding initiated by the child
-	 * component, so pin its containing module until we unbind.
-	 */
-	if (!try_module_get(acomp->ops->owner)) {
-		ret = -ENODEV;
-		goto out_unbind;
-	}
-
-	return 0;
-
-out_unbind:
-	component_unbind_all(dev, acomp);
-
-	return ret;
-}
-
-static void hdac_component_master_unbind(struct device *dev)
-{
-	struct i915_audio_component *acomp = hdac_acomp;
-
-	module_put(acomp->ops->owner);
-	component_unbind_all(dev, acomp);
-	WARN_ON(acomp->ops || acomp->dev);
-}
-
-static const struct component_master_ops hdac_component_master_ops = {
-	.bind = hdac_component_master_bind,
-	.unbind = hdac_component_master_unbind,
-};
-
-static int hdac_component_master_match(struct device *dev, void *data)
+static int i915_component_master_match(struct device *dev, void *data)
 {
-	/* i915 is the only supported component */
 	return !strcmp(dev->driver->name, "i915");
 }
 
-/**
- * snd_hdac_i915_register_notifier - Register i915 audio component ops
- * @aops: i915 audio component ops
- *
- * This function is supposed to be used only by a HD-audio controller
- * driver that needs the interaction with i915 graphics.
- *
- * This function sets the given ops to be called by the i915 graphics driver.
- *
- * Returns zero for success or a negative error code.
- */
-int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *aops)
-{
-	if (!hdac_acomp)
-		return -ENODEV;
-
-	hdac_acomp->audio_ops = aops;
-	return 0;
-}
-EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier);
-
 /* check whether intel graphics is present */
 static bool i915_gfx_present(void)
 {
@@ -345,6 +99,19 @@ static bool i915_gfx_present(void)
 	return pci_dev_present(ids);
 }
 
+static int i915_master_bind(struct device *dev,
+			    struct drm_audio_component *acomp)
+{
+	complete_all(&bind_complete);
+	/* clear audio_ops here as it was needed only for completion call */
+	acomp->audio_ops = NULL;
+	return 0;
+}
+
+static const struct drm_audio_component_audio_ops i915_init_ops = {
+	.master_bind = i915_master_bind
+};
+
 /**
  * snd_hdac_i915_init - Initialize i915 audio component
  * @bus: HDA core bus
@@ -359,83 +126,31 @@ static bool i915_gfx_present(void)
  */
 int snd_hdac_i915_init(struct hdac_bus *bus)
 {
-	struct component_match *match = NULL;
-	struct device *dev = bus->dev;
-	struct i915_audio_component *acomp;
-	int ret;
-
-	if (WARN_ON(hdac_acomp))
-		return -EBUSY;
+	struct drm_audio_component *acomp;
+	int err;
 
 	if (!i915_gfx_present())
 		return -ENODEV;
 
-	acomp = kzalloc(sizeof(*acomp), GFP_KERNEL);
-	if (!acomp)
-		return -ENOMEM;
-	bus->audio_component = acomp;
-	hdac_acomp = acomp;
-
-	component_match_add(dev, &match, hdac_component_master_match, bus);
-	ret = component_master_add_with_match(dev, &hdac_component_master_ops,
-					      match);
-	if (ret < 0)
-		goto out_err;
-
-	/*
-	 * Atm, we don't support deferring the component binding, so make sure
-	 * i915 is loaded and that the binding successfully completes.
-	 */
-	request_module("i915");
+	init_completion(&bind_complete);
 
+	err = snd_hdac_acomp_init(bus, &i915_init_ops,
+				  i915_component_master_match,
+				  sizeof(struct i915_audio_component) - sizeof(*acomp));
+	if (err < 0)
+		return err;
+	acomp = bus->audio_component;
+	if (!acomp)
+		return -ENODEV;
 	if (!acomp->ops) {
-		ret = -ENODEV;
-		goto out_master_del;
+		request_module("i915");
+		/* 10s timeout */
+		wait_for_completion_timeout(&bind_complete, 10 * 1000);
+	}
+	if (!acomp->ops) {
+		snd_hdac_acomp_exit(bus);
+		return -ENODEV;
 	}
-	dev_dbg(dev, "bound to i915 component master\n");
-
 	return 0;
-out_master_del:
-	component_master_del(dev, &hdac_component_master_ops);
-out_err:
-	kfree(acomp);
-	bus->audio_component = NULL;
-	hdac_acomp = NULL;
-	dev_info(dev, "failed to add i915 component master (%d)\n", ret);
-
-	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_i915_init);
-
-/**
- * snd_hdac_i915_exit - Finalize i915 audio component
- * @bus: HDA core bus
- *
- * This function is supposed to be used only by a HD-audio controller
- * driver that needs the interaction with i915 graphics.
- *
- * This function releases the i915 audio component that has been used.
- *
- * Returns zero for success or a negative error code.
- */
-int snd_hdac_i915_exit(struct hdac_bus *bus)
-{
-	struct device *dev = bus->dev;
-	struct i915_audio_component *acomp = bus->audio_component;
-
-	if (!acomp)
-		return 0;
-
-	WARN_ON(bus->i915_power_refcount);
-	if (bus->i915_power_refcount > 0 && acomp->ops)
-		acomp->ops->put_power(acomp->dev);
-
-	component_master_del(dev, &hdac_component_master_ops);
-
-	kfree(acomp);
-	bus->audio_component = NULL;
-	hdac_acomp = NULL;
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(snd_hdac_i915_exit);
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index e1472c7ab6c1..eee422390d8e 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -621,7 +621,7 @@ int snd_hdac_dsp_prepare(struct hdac_stream *azx_dev, unsigned int format,
 			 unsigned int byte_size, struct snd_dma_buffer *bufp)
 {
 	struct hdac_bus *bus = azx_dev->bus;
-	u32 *bdl;
+	__le32 *bdl;
 	int err;
 
 	snd_hdac_dsp_lock(azx_dev);
@@ -651,7 +651,7 @@ int snd_hdac_dsp_prepare(struct hdac_stream *azx_dev, unsigned int format,
 	snd_hdac_stream_writel(azx_dev, SD_BDLPU, 0);
 
 	azx_dev->frags = 0;
-	bdl = (u32 *)azx_dev->bdl.area;
+	bdl = (__le32 *)azx_dev->bdl.area;
 	err = setup_bdle(bus, bufp, azx_dev, &bdl, 0, byte_size, 0);
 	if (err < 0)
 		goto error;
diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c
index 7e21621e492a..2647309bc675 100644
--- a/sound/i2c/cs8427.c
+++ b/sound/i2c/cs8427.c
@@ -621,15 +621,3 @@ int snd_cs8427_iec958_pcm(struct snd_i2c_device *cs8427, unsigned int rate)
 }
 
 EXPORT_SYMBOL(snd_cs8427_iec958_pcm);
-
-static int __init alsa_cs8427_module_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_cs8427_module_exit(void)
-{
-}
-
-module_init(alsa_cs8427_module_init)
-module_exit(alsa_cs8427_module_exit)
diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c
index ef2a9afe9e19..c4a232f18a79 100644
--- a/sound/i2c/i2c.c
+++ b/sound/i2c/i2c.c
@@ -338,16 +338,3 @@ static int snd_i2c_bit_probeaddr(struct snd_i2c_bus *bus, unsigned short addr)
 	snd_i2c_bit_stop(bus);
 	return err;
 }
-
-
-static int __init alsa_i2c_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_i2c_exit(void)
-{
-}
-
-module_init(alsa_i2c_init)
-module_exit(alsa_i2c_exit)
diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c
index bf377dc192aa..7f2761a2e7c8 100644
--- a/sound/i2c/other/ak4xxx-adda.c
+++ b/sound/i2c/other/ak4xxx-adda.c
@@ -911,15 +911,3 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
 	return 0;
 }
 EXPORT_SYMBOL(snd_akm4xxx_build_controls);
-
-static int __init alsa_akm4xxx_module_init(void)
-{
-	return 0;
-}
-        
-static void __exit alsa_akm4xxx_module_exit(void)
-{
-}
-        
-module_init(alsa_akm4xxx_module_init)
-module_exit(alsa_akm4xxx_module_exit)
diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c
index 2d22310dce05..239c4822427f 100644
--- a/sound/i2c/tea6330t.c
+++ b/sound/i2c/tea6330t.c
@@ -368,19 +368,3 @@ int snd_tea6330t_update_mixer(struct snd_card *card,
 
 EXPORT_SYMBOL(snd_tea6330t_detect);
 EXPORT_SYMBOL(snd_tea6330t_update_mixer);
-
-/*
- *  INIT part
- */
-
-static int __init alsa_tea6330t_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_tea6330t_exit(void)
-{
-}
-
-module_init(alsa_tea6330t_init)
-module_exit(alsa_tea6330t_exit)
diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig
index 43b35a873d78..d7db1eeebc84 100644
--- a/sound/isa/Kconfig
+++ b/sound/isa/Kconfig
@@ -459,7 +459,7 @@ config SND_MSND_CLASSIC
 	  Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or
 	  Monterey (not for the Pinnacle or Fiji).
 
-	  See <file:Documentation/sound/oss/MultiSound> for important information
+	  See <file:Documentation/sound/cards/multisound.sh> for important information
 	  about this driver.  Note that it has been discontinued, but the
 	  Voyetra Turtle Beach knowledge base entry for it is still available
 	  at <http://www.turtlebeach.com/site/kb_ftp/790.asp>.
diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c
index 923201414469..fba6d22f7f4b 100644
--- a/sound/isa/ad1816a/ad1816a_lib.c
+++ b/sound/isa/ad1816a/ad1816a_lib.c
@@ -85,7 +85,8 @@ static void snd_ad1816a_write_mask(struct snd_ad1816a *chip, unsigned char reg,
 
 
 static unsigned char snd_ad1816a_get_format(struct snd_ad1816a *chip,
-					    unsigned int format, int channels)
+					    snd_pcm_format_t format,
+					    int channels)
 {
 	unsigned char retval = AD1816A_FMT_LINEAR_8;
 
diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c
index a826c138e7f5..3dfe7e592c25 100644
--- a/sound/isa/es1688/es1688.c
+++ b/sound/isa/es1688/es1688.c
@@ -260,7 +260,6 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard,
 	struct snd_card *card;
 	static unsigned int dev;
 	int error;
-	struct snd_es1688 *chip;
 
 	if (snd_es968_pnp_is_probed)
 		return -EBUSY;
@@ -276,7 +275,6 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard,
 			     sizeof(struct snd_es1688), &card);
 	if (error < 0)
 		return error;
-	chip = card->private_data;
 
 	error = snd_card_es968_pnp(card, dev, pcard, pid);
 	if (error < 0) {
diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c
index f9c0662e9a22..50cdce0e8946 100644
--- a/sound/isa/es1688/es1688_lib.c
+++ b/sound/isa/es1688/es1688_lib.c
@@ -1029,19 +1029,3 @@ EXPORT_SYMBOL(snd_es1688_mixer_write);
 EXPORT_SYMBOL(snd_es1688_create);
 EXPORT_SYMBOL(snd_es1688_pcm);
 EXPORT_SYMBOL(snd_es1688_mixer);
-
-/*
- *  INIT part
- */
-
-static int __init alsa_es1688_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_es1688_exit(void)
-{
-}
-
-module_init(alsa_es1688_init)
-module_exit(alsa_es1688_exit)
diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c
index 2a6960c3e2a4..0d103d6f805e 100644
--- a/sound/isa/es18xx.c
+++ b/sound/isa/es18xx.c
@@ -1024,6 +1024,7 @@ static int snd_es18xx_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
 			val = 3;
 		} else
 			retVal = snd_es18xx_mixer_bits(chip, 0x7a, 0x08, 0x00) != 0x00;
+		/* fall through */
  /* 4 source chips */
 	case 0x1868:
 	case 0x1878:
diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c
index b9994cc9f5fb..af9eea41379f 100644
--- a/sound/isa/galaxy/galaxy.c
+++ b/sound/isa/galaxy/galaxy.c
@@ -260,6 +260,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n)
 		break;
 	case 2:
 		irq[n] = 9;
+		/* Fall through */
 	case 9:
 		wss_config[n] |= WSS_CONFIG_IRQ_9;
 		break;
@@ -304,6 +305,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n)
 	case 1:
 		if (dma1[n] == 0)
 			break;
+		/* Fall through */
 	default:
 		dev_err(dev, "invalid capture DMA %d\n", dma2[n]);
 		return 0;
@@ -333,6 +335,7 @@ mpu:
 		break;
 	case 2:
 		mpu_irq[n] = 9;
+		/* Fall through */
 	case 9:
 		config[n] |= GALAXY_CONFIG_MPUIRQ_2;
 		break;
diff --git a/sound/isa/gus/gus_io.c b/sound/isa/gus/gus_io.c
index ca79878d8d8c..2fd32ef22c30 100644
--- a/sound/isa/gus/gus_io.c
+++ b/sound/isa/gus/gus_io.c
@@ -461,7 +461,7 @@ void snd_gf1_print_voice_registers(struct snd_gus_card * gus)
 			printk(KERN_INFO " -%i- GFA1 effect address         = 0x%x\n", voice, snd_gf1_i_read_addr(gus, 0x11, ctrl & 4));
 			printk(KERN_INFO " -%i- GFA1 effect volume          = 0x%x\n", voice, snd_gf1_i_read16(gus, 0x16));
 			printk(KERN_INFO " -%i- GFA1 effect volume final    = 0x%x\n", voice, snd_gf1_i_read16(gus, 0x1d));
-			printk(KERN_INFO " -%i- GFA1 effect acumulator      = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x14));
+			printk(KERN_INFO " -%i- GFA1 effect accumulator     = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x14));
 		}
 		if (mode & 0x20) {
 			printk(KERN_INFO " -%i- GFA1 left offset            = 0x%x (%i)\n", voice, snd_gf1_i_read16(gus, 0x13), snd_gf1_i_read16(gus, 0x13) >> 4);
diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c
index 3cf9b13c780a..3b8a0c880db5 100644
--- a/sound/isa/gus/gus_main.c
+++ b/sound/isa/gus/gus_main.c
@@ -465,19 +465,3 @@ EXPORT_SYMBOL(snd_gf1_mem_alloc);
 EXPORT_SYMBOL(snd_gf1_mem_xfree);
 EXPORT_SYMBOL(snd_gf1_mem_free);
 EXPORT_SYMBOL(snd_gf1_mem_lock);
-
-/*
- *  INIT part
- */
-
-static int __init alsa_gus_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_gus_exit(void)
-{
-}
-
-module_init(alsa_gus_init)
-module_exit(alsa_gus_exit)
diff --git a/sound/isa/gus/gus_reset.c b/sound/isa/gus/gus_reset.c
index 3d1fed0c2620..59b3f683d49b 100644
--- a/sound/isa/gus/gus_reset.c
+++ b/sound/isa/gus/gus_reset.c
@@ -292,7 +292,6 @@ void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice)
 {
 	unsigned long flags;
 	void (*private_free)(struct snd_gus_voice *voice);
-	void *private_data;
 
 	if (voice == NULL || !voice->use)
 		return;
@@ -300,7 +299,6 @@ void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice)
 	snd_gf1_clear_voices(gus, voice->number, voice->number);
 	spin_lock_irqsave(&gus->voice_alloc, flags);
 	private_free = voice->private_free;
-	private_data = voice->private_data;
 	voice->private_free = NULL;
 	voice->private_data = NULL;
 	if (voice->pcm)
diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c
index 569897f64fda..7c3203fe4869 100644
--- a/sound/isa/msnd/msnd.c
+++ b/sound/isa/msnd/msnd.c
@@ -54,7 +54,7 @@
 #define LOGNAME			"msnd"
 
 
-void snd_msnd_init_queue(void *base, int start, int size)
+void snd_msnd_init_queue(void __iomem *base, int start, int size)
 {
 	writew(PCTODSP_BASED(start), base + JQS_wStart);
 	writew(PCTODSP_OFFSET(size) - 1, base + JQS_wSize);
@@ -270,7 +270,7 @@ int snd_msnd_DARQ(struct snd_msnd *chip, int bank)
 		udelay(1);
 
 	if (chip->capturePeriods == 2) {
-		void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF +
+		void __iomem *pDAQ = chip->mappedbase + DARQ_DATA_BUFF +
 			     bank * DAQDS__size + DAQDS_wStart;
 		unsigned short offset = 0x3000 + chip->capturePeriodBytes;
 
@@ -309,7 +309,7 @@ int snd_msnd_DAPQ(struct snd_msnd *chip, int start)
 {
 	u16	DAPQ_tail;
 	int	protect = start, nbanks = 0;
-	void	*DAQD;
+	void	__iomem *DAQD;
 	static int play_banks_submitted;
 	/* unsigned long flags;
 	spin_lock_irqsave(&chip->lock, flags); not necessary */
@@ -370,7 +370,7 @@ static void snd_msnd_play_reset_queue(struct snd_msnd *chip,
 				      unsigned int pcm_count)
 {
 	int	n;
-	void	*pDAQ = chip->mappedbase + DAPQ_DATA_BUFF;
+	void	__iomem *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF;
 
 	chip->last_playbank = -1;
 	chip->playLimit = pcm_count * (pcm_periods - 1);
@@ -398,7 +398,7 @@ static void snd_msnd_capture_reset_queue(struct snd_msnd *chip,
 					 unsigned int pcm_count)
 {
 	int		n;
-	void		*pDAQ;
+	void		__iomem *pDAQ;
 	/* unsigned long	flags; */
 
 	/* snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); */
@@ -485,7 +485,7 @@ static int snd_msnd_playback_open(struct snd_pcm_substream *substream)
 	clear_bit(F_WRITING, &chip->flags);
 	snd_msnd_enable_irq(chip);
 
-	runtime->dma_area = chip->mappedbase;
+	runtime->dma_area = (__force void *)chip->mappedbase;
 	runtime->dma_bytes = 0x3000;
 
 	chip->playback_substream = substream;
@@ -508,7 +508,7 @@ static int snd_msnd_playback_hw_params(struct snd_pcm_substream *substream,
 {
 	int	i;
 	struct snd_msnd *chip = snd_pcm_substream_chip(substream);
-	void	*pDAQ =	chip->mappedbase + DAPQ_DATA_BUFF;
+	void	__iomem *pDAQ =	chip->mappedbase + DAPQ_DATA_BUFF;
 
 	chip->play_sample_size = snd_pcm_format_width(params_format(params));
 	chip->play_channels = params_channels(params);
@@ -589,7 +589,7 @@ static int snd_msnd_capture_open(struct snd_pcm_substream *substream)
 
 	set_bit(F_AUDIO_READ_INUSE, &chip->flags);
 	snd_msnd_enable_irq(chip);
-	runtime->dma_area = chip->mappedbase + 0x3000;
+	runtime->dma_area = (__force void *)chip->mappedbase + 0x3000;
 	runtime->dma_bytes = 0x3000;
 	memset(runtime->dma_area, 0, runtime->dma_bytes);
 	chip->capture_substream = substream;
@@ -654,7 +654,7 @@ static int snd_msnd_capture_hw_params(struct snd_pcm_substream *substream,
 {
 	int		i;
 	struct snd_msnd *chip = snd_pcm_substream_chip(substream);
-	void		*pDAQ = chip->mappedbase + DARQ_DATA_BUFF;
+	void		__iomem *pDAQ = chip->mappedbase + DARQ_DATA_BUFF;
 
 	chip->capture_sample_size = snd_pcm_format_width(params_format(params));
 	chip->capture_channels = params_channels(params);
diff --git a/sound/isa/msnd/msnd.h b/sound/isa/msnd/msnd.h
index 5f3c7dcd9f9d..80c718757eef 100644
--- a/sound/isa/msnd/msnd.h
+++ b/sound/isa/msnd/msnd.h
@@ -283,7 +283,7 @@ struct snd_msnd {
 
 };
 
-void snd_msnd_init_queue(void *base, int start, int size);
+void snd_msnd_init_queue(void __iomem *base, int start, int size);
 
 int snd_msnd_send_dsp_cmd(struct snd_msnd *chip, u8 cmd);
 int snd_msnd_send_word(struct snd_msnd *chip,
diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c
index 013d8d1170fe..42876b0cb68b 100644
--- a/sound/isa/msnd/msnd_midi.c
+++ b/sound/isa/msnd/msnd_midi.c
@@ -119,7 +119,7 @@ void snd_msndmidi_input_read(void *mpuv)
 {
 	unsigned long flags;
 	struct snd_msndmidi *mpu = mpuv;
-	void *pwMIDQData = mpu->dev->mappedbase + MIDQ_DATA_BUFF;
+	void __iomem *pwMIDQData = mpu->dev->mappedbase + MIDQ_DATA_BUFF;
 	u16 head, tail, size;
 
 	spin_lock_irqsave(&mpu->input_lock, flags);
diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c
index 6c584d9b6c42..11af9c40bc05 100644
--- a/sound/isa/msnd/msnd_pinnacle.c
+++ b/sound/isa/msnd/msnd_pinnacle.c
@@ -82,10 +82,10 @@
 
 static void set_default_audio_parameters(struct snd_msnd *chip)
 {
-	chip->play_sample_size = DEFSAMPLESIZE;
+	chip->play_sample_size = snd_pcm_format_width(DEFSAMPLESIZE);
 	chip->play_sample_rate = DEFSAMPLERATE;
 	chip->play_channels = DEFCHANNELS;
-	chip->capture_sample_size = DEFSAMPLESIZE;
+	chip->capture_sample_size = snd_pcm_format_width(DEFSAMPLESIZE);
 	chip->capture_sample_rate = DEFSAMPLERATE;
 	chip->capture_channels = DEFCHANNELS;
 }
@@ -169,7 +169,7 @@ static void snd_msnd_eval_dsp_msg(struct snd_msnd *chip, u16 wMessage)
 static irqreturn_t snd_msnd_interrupt(int irq, void *dev_id)
 {
 	struct snd_msnd *chip = dev_id;
-	void *pwDSPQData = chip->mappedbase + DSPQ_DATA_BUFF;
+	void __iomem *pwDSPQData = chip->mappedbase + DSPQ_DATA_BUFF;
 	u16 head, tail, size;
 
 	/* Send ack to DSP */
@@ -810,7 +810,7 @@ module_param(calibrate_signal, int, 0444);
 #ifndef MSND_CLASSIC
 module_param_array(digital, int, NULL, 0444);
 module_param_hw_array(cfg, long, ioport, NULL, 0444);
-module_param_array(reset, int, 0, 0444);
+module_param_array(reset, int, NULL, 0444);
 module_param_hw_array(mpu_io, long, ioport, NULL, 0444);
 module_param_hw_array(mpu_irq, int, irq, NULL, 0444);
 module_param_hw_array(ide_io0, long, ioport, NULL, 0444);
diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c
index 8894c7c18ad6..c6136c6b0214 100644
--- a/sound/isa/opti9xx/miro.c
+++ b/sound/isa/opti9xx/miro.c
@@ -176,10 +176,13 @@ static int aci_busy_wait(struct snd_miro_aci *aci)
 			switch (timeout-ACI_MINTIME) {
 			case 0 ... 9:
 				out /= 10;
+				/* fall through */
 			case 10 ... 19:
 				out /= 10;
+				/* fall through */
 			case 20 ... 30:
 				out /= 10;
+				/* fall through */
 			default:
 				set_current_state(TASK_UNINTERRUPTIBLE);
 				schedule_timeout(out);
@@ -834,6 +837,7 @@ static unsigned char snd_miro_read(struct snd_miro *chip,
 			retval = inb(chip->mc_base + 9);
 			break;
 		}
+		/* fall through */
 
 	case OPTi9XX_HW_82C929:
 		retval = inb(chip->mc_base + reg);
@@ -863,6 +867,7 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg,
 			outb(value, chip->mc_base + 9);
 			break;
 		}
+		/* fall through */
 
 	case OPTi9XX_HW_82C929:
 		outb(value, chip->mc_base + reg);
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index 505cd81e19fa..ac0ab6eb40f0 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -261,6 +261,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip,
 			retval = inb(chip->mc_base + 9);
 			break;
 		}
+		/* Fall through */
 
 	case OPTi9XX_HW_82C928:
 	case OPTi9XX_HW_82C929:
@@ -303,6 +304,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg,
 			outb(value, chip->mc_base + 9);
 			break;
 		}
+		/* Fall through */
 
 	case OPTi9XX_HW_82C928:
 	case OPTi9XX_HW_82C929:
@@ -350,6 +352,7 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip,
 		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc);
 		/* enable wave audio */
 		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02);
+		/* Fall through */
 
 	case OPTi9XX_HW_82C925:
 		/* enable WSS mode */
diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c
index c2e41d2762f7..d45a6b9d6437 100644
--- a/sound/isa/sb/emu8000_patch.c
+++ b/sound/isa/sb/emu8000_patch.c
@@ -165,11 +165,8 @@ snd_emu8000_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
 		return 0;
 
 	/* be sure loop points start < end */
-	if (sp->v.loopstart > sp->v.loopend) {
-		int tmp = sp->v.loopstart;
-		sp->v.loopstart = sp->v.loopend;
-		sp->v.loopend = tmp;
-	}
+	if (sp->v.loopstart > sp->v.loopend)
+		swap(sp->v.loopstart, sp->v.loopend);
 
 	/* compute true data size to be loaded */
 	truesize = sp->v.size;
diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c
index bc5af71d3bdb..f46f6ec3ea0c 100644
--- a/sound/isa/sb/emu8000_pcm.c
+++ b/sound/isa/sb/emu8000_pcm.c
@@ -470,7 +470,7 @@ static int emu8k_pcm_copy(struct snd_pcm_substream *subs,
 	/* convert to word unit */
 	pos = (pos << 1) + rec->loop_start[voice];
 	count <<= 1;
-	LOOP_WRITE(rec, pos, src, count, COPY_UESR);
+	LOOP_WRITE(rec, pos, src, count, COPY_USER);
 	return 0;
 }
 
diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c
index fa5780bb0c68..bf3db0d2ea12 100644
--- a/sound/isa/sb/sb16_csp.c
+++ b/sound/isa/sb/sb16_csp.c
@@ -60,18 +60,18 @@ MODULE_FIRMWARE("sb16/ima_adpcm_capture.csp");
  * RIFF data format
  */
 struct riff_header {
-	__u32 name;
-	__u32 len;
+	__le32 name;
+	__le32 len;
 };
 
 struct desc_header {
 	struct riff_header info;
-	__u16 func_nr;
-	__u16 VOC_type;
-	__u16 flags_play_rec;
-	__u16 flags_16bit_8bit;
-	__u16 flags_stereo_mono;
-	__u16 flags_rates;
+	__le16 func_nr;
+	__le16 VOC_type;
+	__le16 flags_play_rec;
+	__le16 flags_16bit_8bit;
+	__le16 flags_stereo_mono;
+	__le16 flags_rates;
 };
 
 /*
@@ -93,7 +93,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p,
 				struct snd_sb_csp_microcode __user * code);
 static int snd_sb_csp_unload(struct snd_sb_csp * p);
 static int snd_sb_csp_load_user(struct snd_sb_csp * p, const unsigned char __user *buf, int size, int load_flags);
-static int snd_sb_csp_autoload(struct snd_sb_csp * p, int pcm_sfmt, int play_rec_mode);
+static int snd_sb_csp_autoload(struct snd_sb_csp * p, snd_pcm_format_t pcm_sfmt, int play_rec_mode);
 static int snd_sb_csp_check_version(struct snd_sb_csp * p);
 
 static int snd_sb_csp_use(struct snd_sb_csp * p);
@@ -314,7 +314,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p,
 	unsigned short func_nr = 0;
 
 	struct riff_header file_h, item_h, code_h;
-	__u32 item_type;
+	__le32 item_type;
 	struct desc_header funcdesc_h;
 
 	unsigned long flags;
@@ -326,7 +326,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p,
 
 	if (copy_from_user(&file_h, data_ptr, sizeof(file_h)))
 		return -EFAULT;
-	if ((file_h.name != RIFF_HEADER) ||
+	if ((le32_to_cpu(file_h.name) != RIFF_HEADER) ||
 	    (le32_to_cpu(file_h.len) >= SNDRV_SB_CSP_MAX_MICROCODE_FILE_SIZE - sizeof(file_h))) {
 		snd_printd("%s: Invalid RIFF header\n", __func__);
 		return -EINVAL;
@@ -336,7 +336,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p,
 
 	if (copy_from_user(&item_type, data_ptr, sizeof(item_type)))
 		return -EFAULT;
-	if (item_type != CSP__HEADER) {
+	if (le32_to_cpu(item_type) != CSP__HEADER) {
 		snd_printd("%s: Invalid RIFF file type\n", __func__);
 		return -EINVAL;
 	}
@@ -346,12 +346,12 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p,
 		if (copy_from_user(&item_h, data_ptr, sizeof(item_h)))
 			return -EFAULT;
 		data_ptr += sizeof(item_h);
-		if (item_h.name != LIST_HEADER)
+		if (le32_to_cpu(item_h.name) != LIST_HEADER)
 			continue;
 
 		if (copy_from_user(&item_type, data_ptr, sizeof(item_type)))
 			 return -EFAULT;
-		switch (item_type) {
+		switch (le32_to_cpu(item_type)) {
 		case FUNC_HEADER:
 			if (copy_from_user(&funcdesc_h, data_ptr + sizeof(item_type), sizeof(funcdesc_h)))
 				return -EFAULT;
@@ -378,7 +378,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p,
 					return -EFAULT;
 
 				/* init microcode blocks */
-				if (code_h.name != INIT_HEADER)
+				if (le32_to_cpu(code_h.name) != INIT_HEADER)
 					break;
 				data_ptr += sizeof(code_h);
 				err = snd_sb_csp_load_user(p, data_ptr, le32_to_cpu(code_h.len),
@@ -391,7 +391,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p,
 			if (copy_from_user(&code_h, data_ptr, sizeof(code_h)))
 				return -EFAULT;
 
-			if (code_h.name != MAIN_HEADER) {
+			if (le32_to_cpu(code_h.name) != MAIN_HEADER) {
 				snd_printd("%s: Missing 'main' microcode\n", __func__);
 				return -EINVAL;
 			}
@@ -726,7 +726,7 @@ static int snd_sb_csp_firmware_load(struct snd_sb_csp *p, int index, int flags)
  * autoload hardware codec if necessary
  * return 0 if CSP is loaded and ready to run (p->running != 0)
  */
-static int snd_sb_csp_autoload(struct snd_sb_csp * p, int pcm_sfmt, int play_rec_mode)
+static int snd_sb_csp_autoload(struct snd_sb_csp * p, snd_pcm_format_t pcm_sfmt, int play_rec_mode)
 {
 	unsigned long flags;
 	int err = 0;
@@ -736,7 +736,7 @@ static int snd_sb_csp_autoload(struct snd_sb_csp * p, int pcm_sfmt, int play_rec
 		return -EBUSY;
 
 	/* autoload microcode only if requested hardware codec is not already loaded */
-	if (((1 << pcm_sfmt) & p->acc_format) && (play_rec_mode & p->mode)) {
+	if (((1U << (__force int)pcm_sfmt) & p->acc_format) && (play_rec_mode & p->mode)) {
 		p->running = SNDRV_SB_CSP_ST_AUTO;
 	} else {
 		switch (pcm_sfmt) {
@@ -1185,19 +1185,3 @@ static void info_read(struct snd_info_entry *entry, struct snd_info_buffer *buff
 /* */
 
 EXPORT_SYMBOL(snd_sb_csp_new);
-
-/*
- * INIT part
- */
-
-static int __init alsa_sb_csp_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_sb_csp_exit(void)
-{
-}
-
-module_init(alsa_sb_csp_init)
-module_exit(alsa_sb_csp_exit)
diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c
index 3e39ba220c39..37e6ce7b0b13 100644
--- a/sound/isa/sb/sb16_main.c
+++ b/sound/isa/sb/sb16_main.c
@@ -49,6 +49,9 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Routines for control of 16-bit SoundBlaster cards and clones");
 MODULE_LICENSE("GPL");
 
+#define runtime_format_bits(runtime) \
+	((unsigned int)pcm_format_to_bits((runtime)->format))
+
 #ifdef CONFIG_SND_SB16_CSP
 static void snd_sb16_csp_playback_prepare(struct snd_sb *chip, struct snd_pcm_runtime *runtime)
 {
@@ -58,7 +61,7 @@ static void snd_sb16_csp_playback_prepare(struct snd_sb *chip, struct snd_pcm_ru
 		if (csp->running & SNDRV_SB_CSP_ST_LOADED) {
 			/* manually loaded codec */
 			if ((csp->mode & SNDRV_SB_CSP_MODE_DSP_WRITE) &&
-			    ((1U << runtime->format) == csp->acc_format)) {
+			    (runtime_format_bits(runtime) == csp->acc_format)) {
 				/* Supported runtime PCM format for playback */
 				if (csp->ops.csp_use(csp) == 0) {
 					/* If CSP was successfully acquired */
@@ -66,7 +69,7 @@ static void snd_sb16_csp_playback_prepare(struct snd_sb *chip, struct snd_pcm_ru
 				}
 			} else if ((csp->mode & SNDRV_SB_CSP_MODE_QSOUND) && (csp->q_enabled)) {
 				/* QSound decoder is loaded and enabled */
-				if ((1 << runtime->format) & (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |
+				if (runtime_format_bits(runtime) & (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |
 							      SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE)) {
 					/* Only for simple PCM formats */
 					if (csp->ops.csp_use(csp) == 0) {
@@ -106,7 +109,7 @@ static void snd_sb16_csp_capture_prepare(struct snd_sb *chip, struct snd_pcm_run
 		if (csp->running & SNDRV_SB_CSP_ST_LOADED) {
 			/* manually loaded codec */
 			if ((csp->mode & SNDRV_SB_CSP_MODE_DSP_READ) &&
-			    ((1U << runtime->format) == csp->acc_format)) {
+			    (runtime_format_bits(runtime) == csp->acc_format)) {
 				/* Supported runtime PCM format for capture */
 				if (csp->ops.csp_use(csp) == 0) {
 					/* If CSP was successfully acquired */
@@ -897,19 +900,3 @@ EXPORT_SYMBOL(snd_sb16dsp_pcm);
 EXPORT_SYMBOL(snd_sb16dsp_get_pcm_ops);
 EXPORT_SYMBOL(snd_sb16dsp_configure);
 EXPORT_SYMBOL(snd_sb16dsp_interrupt);
-
-/*
- *  INIT part
- */
-
-static int __init alsa_sb16_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_sb16_exit(void)
-{
-}
-
-module_init(alsa_sb16_init)
-module_exit(alsa_sb16_exit)
diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c
index d45df5c54423..481797744b3c 100644
--- a/sound/isa/sb/sb8_main.c
+++ b/sound/isa/sb/sb8_main.c
@@ -381,7 +381,6 @@ static int snd_sb8_capture_trigger(struct snd_pcm_substream *substream,
 irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip)
 {
 	struct snd_pcm_substream *substream;
-	struct snd_pcm_runtime *runtime;
 
 	snd_sb_ack_8bit(chip);
 	switch (chip->mode) {
@@ -391,7 +390,6 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip)
 		/* fallthru */
 	case SB_MODE_PLAYBACK_8:
 		substream = chip->playback_substream;
-		runtime = substream->runtime;
 		if (chip->playback_format == SB_DSP_OUTPUT)
 		    	snd_sb8_playback_trigger(substream, SNDRV_PCM_TRIGGER_START);
 		snd_pcm_period_elapsed(substream);
@@ -402,7 +400,6 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip)
 		/* fallthru */
 	case SB_MODE_CAPTURE_8:
 		substream = chip->capture_substream;
-		runtime = substream->runtime;
 		if (chip->capture_format == SB_DSP_INPUT)
 		    	snd_sb8_capture_trigger(substream, SNDRV_PCM_TRIGGER_START);
 		snd_pcm_period_elapsed(substream);
@@ -624,19 +621,3 @@ EXPORT_SYMBOL(snd_sb8dsp_interrupt);
   /* sb8_midi.c */
 EXPORT_SYMBOL(snd_sb8dsp_midi_interrupt);
 EXPORT_SYMBOL(snd_sb8dsp_midi);
-
-/*
- *  INIT part
- */
-
-static int __init alsa_sb8_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_sb8_exit(void)
-{
-}
-
-module_init(alsa_sb8_init)
-module_exit(alsa_sb8_exit)
diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c
index 787a4ade4afd..90b254aaef74 100644
--- a/sound/isa/sb/sb_common.c
+++ b/sound/isa/sb/sb_common.c
@@ -305,19 +305,3 @@ EXPORT_SYMBOL(snd_sbmixer_add_ctl);
 EXPORT_SYMBOL(snd_sbmixer_suspend);
 EXPORT_SYMBOL(snd_sbmixer_resume);
 #endif
-
-/*
- *  INIT part
- */
-
-static int __init alsa_sb_common_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_sb_common_exit(void)
-{
-}
-
-module_init(alsa_sb_common_init)
-module_exit(alsa_sb_common_exit)
diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c
index 8a852042a066..32453f81b95a 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -541,7 +541,7 @@ static unsigned char snd_wss_get_rate(unsigned int rate)
 }
 
 static unsigned char snd_wss_get_format(struct snd_wss *chip,
-					int format,
+					snd_pcm_format_t format,
 					int channels)
 {
 	unsigned char rformat;
@@ -2279,19 +2279,3 @@ const struct snd_pcm_ops *snd_wss_get_pcm_ops(int direction)
 		&snd_wss_playback_ops : &snd_wss_capture_ops;
 }
 EXPORT_SYMBOL(snd_wss_get_pcm_ops);
-
-/*
- *  INIT part
- */
-
-static int __init alsa_wss_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_wss_exit(void)
-{
-}
-
-module_init(alsa_wss_init);
-module_exit(alsa_wss_exit);
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
index 9fb68b35de5a..3ec9391a4736 100644
--- a/sound/mips/sgio2audio.c
+++ b/sound/mips/sgio2audio.c
@@ -685,7 +685,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
 	.trigger =     snd_sgio2audio_pcm_trigger,
 	.pointer =     snd_sgio2audio_pcm_pointer,
 	.page =        snd_pcm_lib_get_vmalloc_page,
-	.mmap =        snd_pcm_lib_mmap_vmalloc,
 };
 
 static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
@@ -698,7 +697,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
 	.trigger =     snd_sgio2audio_pcm_trigger,
 	.pointer =     snd_sgio2audio_pcm_pointer,
 	.page =        snd_pcm_lib_get_vmalloc_page,
-	.mmap =        snd_pcm_lib_mmap_vmalloc,
 };
 
 static const struct snd_pcm_ops snd_sgio2audio_capture_ops = {
@@ -711,7 +709,6 @@ static const struct snd_pcm_ops snd_sgio2audio_capture_ops = {
 	.trigger =     snd_sgio2audio_pcm_trigger,
 	.pointer =     snd_sgio2audio_pcm_pointer,
 	.page =        snd_pcm_lib_get_vmalloc_page,
-	.mmap =        snd_pcm_lib_mmap_vmalloc,
 };
 
 /*
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 1ef7cdf1d3e8..f4459d1a9d67 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -2941,19 +2941,3 @@ int snd_ac97_tune_hardware(struct snd_ac97 *ac97,
 }
 
 EXPORT_SYMBOL(snd_ac97_tune_hardware);
-
-/*
- *  INIT part
- */
-
-static int __init alsa_ac97_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_ac97_exit(void)
-{
-}
-
-module_init(alsa_ac97_init)
-module_exit(alsa_ac97_exit)
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 39547e32e584..9f569379b77e 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -1484,12 +1484,9 @@ static struct snd_pcm_hardware snd_ali_capture =
 static void snd_ali_pcm_free_substream(struct snd_pcm_runtime *runtime)
 {
 	struct snd_ali_voice *pvoice = runtime->private_data;
-	struct snd_ali *codec;
 
-	if (pvoice) {
-		codec = pvoice->codec;
+	if (pvoice)
 		snd_ali_free_voice(pvoice->codec, pvoice);
-	}
 }
 
 static int snd_ali_open(struct snd_pcm_substream *substream, int rec,
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 64e0961f93ba..a31fe1550903 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -311,27 +311,29 @@ static void print_hwparams(struct snd_pcm_substream *substream,
 		snd_pcm_format_width(params_format(p)) / 8);
 }
 
+#define INVALID_FORMAT	(__force snd_pcm_format_t)(-1)
+
 static snd_pcm_format_t hpi_to_alsa_formats[] = {
-	-1,			/* INVALID */
+	INVALID_FORMAT,		/* INVALID */
 	SNDRV_PCM_FORMAT_U8,	/* HPI_FORMAT_PCM8_UNSIGNED        1 */
 	SNDRV_PCM_FORMAT_S16,	/* HPI_FORMAT_PCM16_SIGNED         2 */
-	-1,			/* HPI_FORMAT_MPEG_L1              3 */
+	INVALID_FORMAT,		/* HPI_FORMAT_MPEG_L1              3 */
 	SNDRV_PCM_FORMAT_MPEG,	/* HPI_FORMAT_MPEG_L2              4 */
 	SNDRV_PCM_FORMAT_MPEG,	/* HPI_FORMAT_MPEG_L3              5 */
-	-1,			/* HPI_FORMAT_DOLBY_AC2            6 */
-	-1,			/* HPI_FORMAT_DOLBY_AC3            7 */
+	INVALID_FORMAT,		/* HPI_FORMAT_DOLBY_AC2            6 */
+	INVALID_FORMAT,		/* HPI_FORMAT_DOLBY_AC3            7 */
 	SNDRV_PCM_FORMAT_S16_BE,/* HPI_FORMAT_PCM16_BIGENDIAN      8 */
-	-1,			/* HPI_FORMAT_AA_TAGIT1_HITS       9 */
-	-1,			/* HPI_FORMAT_AA_TAGIT1_INSERTS   10 */
+	INVALID_FORMAT,		/* HPI_FORMAT_AA_TAGIT1_HITS       9 */
+	INVALID_FORMAT,		/* HPI_FORMAT_AA_TAGIT1_INSERTS   10 */
 	SNDRV_PCM_FORMAT_S32,	/* HPI_FORMAT_PCM32_SIGNED        11 */
-	-1,			/* HPI_FORMAT_RAW_BITSTREAM       12 */
-	-1,			/* HPI_FORMAT_AA_TAGIT1_HITS_EX1  13 */
+	INVALID_FORMAT,		/* HPI_FORMAT_RAW_BITSTREAM       12 */
+	INVALID_FORMAT,		/* HPI_FORMAT_AA_TAGIT1_HITS_EX1  13 */
 	SNDRV_PCM_FORMAT_FLOAT,	/* HPI_FORMAT_PCM32_FLOAT         14 */
 #if 1
 	/* ALSA can't handle 3 byte sample size together with power-of-2
 	 *  constraint on buffer_bytes, so disable this format
 	 */
-	-1
+	INVALID_FORMAT
 #else
 	/* SNDRV_PCM_FORMAT_S24_3LE */ /* HPI_FORMAT_PCM24_SIGNED 15 */
 #endif
@@ -1023,7 +1025,7 @@ static u64 snd_card_asihpi_playback_formats(struct snd_card_asihpi *asihpi,
 					format, sample_rate, 128000, 0);
 		if (!err)
 			err = hpi_outstream_query_format(h_stream, &hpi_format);
-		if (!err && (hpi_to_alsa_formats[format] != -1))
+		if (!err && (hpi_to_alsa_formats[format] != INVALID_FORMAT))
 			formats |= pcm_format_to_bits(hpi_to_alsa_formats[format]);
 	}
 	return formats;
@@ -1205,7 +1207,7 @@ static u64 snd_card_asihpi_capture_formats(struct snd_card_asihpi *asihpi,
 					format, sample_rate, 128000, 0);
 		if (!err)
 			err = hpi_instream_query_format(h_stream, &hpi_format);
-		if (!err && (hpi_to_alsa_formats[format] != -1))
+		if (!err && (hpi_to_alsa_formats[format] != INVALID_FORMAT))
 			formats |= pcm_format_to_bits(hpi_to_alsa_formats[format]);
 	}
 	return formats;
diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c
index 8d5abfa4e24b..2864698436a5 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -635,7 +635,6 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao,
 	{
 		struct hpi_message hm;
 		struct hpi_response hr;
-		u32 max_streams;
 
 		HPI_DEBUG_LOG(VERBOSE, "init ADAPTER_GET_INFO\n");
 		memset(&hm, 0, sizeof(hm));
@@ -660,10 +659,6 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao,
 		pao->type = hr.u.ax.info.adapter_type;
 		pao->index = hr.u.ax.info.adapter_index;
 
-		max_streams =
-			hr.u.ax.info.num_outstreams +
-			hr.u.ax.info.num_instreams;
-
 		HPI_DEBUG_LOG(VERBOSE,
 			"got adapter info type %x index %d serial %d\n",
 			hr.u.ax.info.adapter_type, hr.u.ax.info.adapter_index,
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 7ae63d452bba..a1e4944dcfe8 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -207,10 +207,10 @@ struct atiixp;
  */
 
 struct atiixp_dma_desc {
-	u32 addr;	/* DMA buffer address */
+	__le32 addr;	/* DMA buffer address */
 	u16 status;	/* status bits */
 	u16 size;	/* size of the packet in dwords */
-	u32 next;	/* address of the next packet descriptor */
+	__le32 next;	/* address of the next packet descriptor */
 };
 
 /*
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index a586635664e0..dc1de860cedf 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -183,10 +183,10 @@ struct atiixp_modem;
  */
 
 struct atiixp_dma_desc {
-	u32 addr;	/* DMA buffer address */
+	__le32 addr;	/* DMA buffer address */
 	u16 status;	/* status bits */
 	u16 size;	/* size of the packet in dwords */
-	u32 next;	/* address of the next packet descriptor */
+	__le32 next;	/* address of the next packet descriptor */
 };
 
 /*
diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h
index bcc648bf6478..e3e31f07d766 100644
--- a/sound/pci/au88x0/au88x0.h
+++ b/sound/pci/au88x0/au88x0.h
@@ -241,7 +241,7 @@ static int vortex_core_init(vortex_t * card);
 static int vortex_core_shutdown(vortex_t * card);
 static void vortex_enable_int(vortex_t * card);
 static irqreturn_t vortex_interrupt(int irq, void *dev_id);
-static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v);
+static int vortex_alsafmt_aspfmt(snd_pcm_format_t alsafmt, vortex_t *v);
 
 /* Connection  stuff. */
 static void vortex_connect_default(vortex_t * vortex, int en);
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 4083c8b01619..2e5b460a847c 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -2770,7 +2770,7 @@ static int vortex_core_shutdown(vortex_t * vortex)
 
 /* Alsa support. */
 
-static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v)
+static int vortex_alsafmt_aspfmt(snd_pcm_format_t alsafmt, vortex_t *v)
 {
 	int fmt;
 
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index d8ade8771a32..ba971042f871 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -228,14 +228,14 @@ static int snd_bt87x_create_risc(struct snd_bt87x *chip, struct snd_pcm_substrea
 			       	 unsigned int periods, unsigned int period_bytes)
 {
 	unsigned int i, offset;
-	u32 *risc;
+	__le32 *risc;
 
 	if (chip->dma_risc.area == NULL) {
 		if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
 					PAGE_ALIGN(MAX_RISC_SIZE), &chip->dma_risc) < 0)
 			return -ENOMEM;
 	}
-	risc = (u32 *)chip->dma_risc.area;
+	risc = (__le32 *)chip->dma_risc.area;
 	offset = 0;
 	*risc++ = cpu_to_le32(RISC_SYNC | RISC_SYNC_FM1);
 	*risc++ = cpu_to_le32(0);
diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c
index abb01ce66983..8d0a3d357345 100644
--- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -73,13 +73,10 @@ static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry,
 {
 	struct proc_scb_info * scb_info  = entry->private_data;
 	struct dsp_scb_descriptor * scb = scb_info->scb_desc;
-	struct dsp_spos_instance * ins;
 	struct snd_cs46xx *chip = scb_info->chip;
 	int j,col;
 	void __iomem *dst = chip->region.idx[1].remap_addr + DSP_PARAMETER_BYTE_OFFSET;
 
-	ins = chip->dsp_spos_instance;
-
 	mutex_lock(&chip->spos_mutex);
 	snd_iprintf(buffer,"%04x %s:\n",scb->address,scb->scb_name);
 
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index de409cda50aa..4590086d9cd8 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -192,8 +192,6 @@ static void process_bm0_irq(struct cs5535audio *cs5535au)
 	bm_stat = cs_readb(cs5535au, ACC_BM0_STATUS);
 	spin_unlock(&cs5535au->reg_lock);
 	if (bm_stat & EOP) {
-		struct cs5535audio_dma *dma;
-		dma = cs5535au->playback_substream->runtime->private_data;
 		snd_pcm_period_elapsed(cs5535au->playback_substream);
 	} else {
 		dev_err(cs5535au->card->dev,
@@ -208,11 +206,8 @@ static void process_bm1_irq(struct cs5535audio *cs5535au)
 	spin_lock(&cs5535au->reg_lock);
 	bm_stat = cs_readb(cs5535au, ACC_BM1_STATUS);
 	spin_unlock(&cs5535au->reg_lock);
-	if (bm_stat & EOP) {
-		struct cs5535audio_dma *dma;
-		dma = cs5535au->capture_substream->runtime->private_data;
+	if (bm_stat & EOP)
 		snd_pcm_period_elapsed(cs5535au->capture_substream);
-	}
 }
 
 static irqreturn_t snd_cs5535audio_interrupt(int irq, void *dev_id)
diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h
index f4fcdf93f3c8..d84620a0c26c 100644
--- a/sound/pci/cs5535audio/cs5535audio.h
+++ b/sound/pci/cs5535audio/cs5535audio.h
@@ -67,9 +67,9 @@ struct cs5535audio_dma_ops {
 };
 
 struct cs5535audio_dma_desc {
-	u32 addr;
-	u16 size;
-	u16 ctlreserved;
+	__le32 addr;
+	__le16 size;
+	__le16 ctlreserved;
 };
 
 struct cs5535audio_dma {
diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c
index ee7065f6e162..326caec854e1 100644
--- a/sound/pci/cs5535audio/cs5535audio_pcm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pcm.c
@@ -158,8 +158,8 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au,
 	lastdesc->addr = cpu_to_le32((u32) dma->desc_buf.addr);
 	lastdesc->size = 0;
 	lastdesc->ctlreserved = cpu_to_le16(PRD_JMP);
-	jmpprd_addr = cpu_to_le32(lastdesc->addr +
-				  (sizeof(struct cs5535audio_dma_desc)*periods));
+	jmpprd_addr = (u32)dma->desc_buf.addr +
+		sizeof(struct cs5535audio_dma_desc) * periods;
 
 	dma->substream = substream;
 	dma->period_bytes = period_bytes;
diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c
index 8e6eb9d7984b..6a051a1c3724 100644
--- a/sound/pci/ctxfi/cthw20k1.c
+++ b/sound/pci/ctxfi/cthw20k1.c
@@ -1319,7 +1319,7 @@ static int hw_pll_init(struct hw *hw, unsigned int rsr)
 			break;
 
 		hw_write_20kx(hw, PLLCTL, pllctl);
-		mdelay(40);
+		msleep(40);
 	}
 	if (i >= 3) {
 		dev_alert(hw->card->dev, "PLL initialization failed!!!\n");
@@ -1407,7 +1407,7 @@ static int hw_reset_dac(struct hw *hw)
 	/* To be effective, need to reset the DAC twice. */
 	for (i = 0; i < 2;  i++) {
 		/* set gpio */
-		mdelay(100);
+		msleep(100);
 		gpioorg = (u16)hw_read_20kx(hw, GPIO);
 		gpioorg &= 0xfffd;
 		hw_write_20kx(hw, GPIO, gpioorg);
@@ -2030,7 +2030,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info)
 	hw_write_20kx(hw, GIE, 0);
 	/* Reset all SRC pending interrupts */
 	hw_write_20kx(hw, SRCIP, 0);
-	mdelay(30);
+	msleep(30);
 
 	/* Detect the card ID and configure GPIO accordingly. */
 	switch (hw->model) {
diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c
index b866d6b2c923..3c966fafc754 100644
--- a/sound/pci/ctxfi/cthw20k2.c
+++ b/sound/pci/ctxfi/cthw20k2.c
@@ -1316,12 +1316,12 @@ static int hw_pll_init(struct hw *hw, unsigned int rsr)
 	set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 4 : 147 - 4);
 	set_field(&pllctl, PLLCTL_RD, 48000 == rsr ? 1 - 1 : 10 - 1);
 	hw_write_20kx(hw, PLL_CTL, pllctl);
-	mdelay(40);
+	msleep(40);
 
 	pllctl = hw_read_20kx(hw, PLL_CTL);
 	set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 2 : 147 - 2);
 	hw_write_20kx(hw, PLL_CTL, pllctl);
-	mdelay(40);
+	msleep(40);
 
 	for (i = 0; i < 1000; i++) {
 		pllstat = hw_read_20kx(hw, PLL_STAT);
@@ -1584,7 +1584,7 @@ static void hw_dac_stop(struct hw *hw)
 	data = hw_read_20kx(hw, GPIO_DATA);
 	data &= 0xFFFFFFFD;
 	hw_write_20kx(hw, GPIO_DATA, data);
-	mdelay(10);
+	usleep_range(10000, 11000);
 }
 
 static void hw_dac_start(struct hw *hw)
@@ -1593,7 +1593,7 @@ static void hw_dac_start(struct hw *hw)
 	data = hw_read_20kx(hw, GPIO_DATA);
 	data |= 0x2;
 	hw_write_20kx(hw, GPIO_DATA, data);
-	mdelay(50);
+	msleep(50);
 }
 
 static void hw_dac_reset(struct hw *hw)
@@ -1864,11 +1864,11 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info)
 		hw_write_20kx(hw, GPIO_DATA, data);
 	}
 
-	mdelay(10);
+	usleep_range(10000, 11000);
 	/* Return the ADC to normal operation. */
 	data |= (0x1 << 15);
 	hw_write_20kx(hw, GPIO_DATA, data);
-	mdelay(50);
+	msleep(50);
 
 	/* I2C write to register offset 0x0B to set ADC LRCLK polarity */
 	/* invert bit, interface format to I2S, word length to 24-bit, */
diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c
index db710d0a609f..4777d50fbbf8 100644
--- a/sound/pci/ctxfi/ctmixer.c
+++ b/sound/pci/ctxfi/ctmixer.c
@@ -938,17 +938,18 @@ static int ct_mixer_topology_build(struct ct_mixer *mixer)
 	struct sum *sum;
 	struct amixer *amix_d, *amix_s;
 	enum CT_AMIXER_CTL i, j;
+	enum CT_SUM_CTL k;
 
 	/* Build topology from destination to source */
 
 	/* Set up Master mixer */
-	for (i = AMIXER_MASTER_F, j = SUM_IN_F;
-					i <= AMIXER_MASTER_S; i++, j++) {
+	for (i = AMIXER_MASTER_F, k = SUM_IN_F;
+					i <= AMIXER_MASTER_S; i++, k++) {
 		amix_d = mixer->amixers[i*CHN_NUM];
-		sum = mixer->sums[j*CHN_NUM];
+		sum = mixer->sums[k*CHN_NUM];
 		amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL);
 		amix_d = mixer->amixers[i*CHN_NUM+1];
-		sum = mixer->sums[j*CHN_NUM+1];
+		sum = mixer->sums[k*CHN_NUM+1];
 		amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL);
 	}
 
@@ -972,12 +973,12 @@ static int ct_mixer_topology_build(struct ct_mixer *mixer)
 	amix_d->ops->setup(amix_d, &amix_s->rsc, INIT_VOL, NULL);
 
 	/* Set up PCM-in mixer */
-	for (i = AMIXER_PCM_F, j = SUM_IN_F; i <= AMIXER_PCM_S; i++, j++) {
+	for (i = AMIXER_PCM_F, k = SUM_IN_F; i <= AMIXER_PCM_S; i++, k++) {
 		amix_d = mixer->amixers[i*CHN_NUM];
-		sum = mixer->sums[j*CHN_NUM];
+		sum = mixer->sums[k*CHN_NUM];
 		amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
 		amix_d = mixer->amixers[i*CHN_NUM+1];
-		sum = mixer->sums[j*CHN_NUM+1];
+		sum = mixer->sums[k*CHN_NUM+1];
 		amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
 	}
 
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 358ef7dcf410..907cf1a46712 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -713,6 +713,7 @@ static int pcm_prepare(struct snd_pcm_substream *substream)
 		break;
 	case SNDRV_PCM_FORMAT_S32_BE:
 		format.data_are_bigendian = 1;
+		/* fall through */
 	case SNDRV_PCM_FORMAT_S32_LE:
 		format.bits_per_sample = 32;
 		break;
@@ -764,6 +765,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 					pipe->last_counter = 0;
 					pipe->position = 0;
 					*pipe->dma_counter = 0;
+					/* fall through */
 				case PIPE_STATE_PAUSED:
 					pipe->state = PIPE_STATE_STARTED;
 					break;
diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h
index 44b390a667d5..be4d0489394a 100644
--- a/sound/pci/echoaudio/echoaudio.h
+++ b/sound/pci/echoaudio/echoaudio.h
@@ -294,7 +294,7 @@
 
 
 struct audiopipe {
-	volatile u32 *dma_counter;	/* Commpage register that contains
+	volatile __le32 *dma_counter;	/* Commpage register that contains
 					 * the current dma position
 					 * (lower 32 bits only)
 					 */
diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c
index 22c786b8a889..cc3c79387194 100644
--- a/sound/pci/echoaudio/echoaudio_3g.c
+++ b/sound/pci/echoaudio/echoaudio_3g.c
@@ -73,19 +73,21 @@ register. write_control_reg sends the new control register value to the DSP. */
 static int write_control_reg(struct echoaudio *chip, u32 ctl, u32 frq,
 			     char force)
 {
+	__le32 ctl_reg, frq_reg;
+
 	if (wait_handshake(chip))
 		return -EIO;
 
 	dev_dbg(chip->card->dev,
 		"WriteControlReg: Setting 0x%x, 0x%x\n", ctl, frq);
 
-	ctl = cpu_to_le32(ctl);
-	frq = cpu_to_le32(frq);
+	ctl_reg = cpu_to_le32(ctl);
+	frq_reg = cpu_to_le32(frq);
 
-	if (ctl != chip->comm_page->control_register ||
-	    frq != chip->comm_page->e3g_frq_register || force) {
-		chip->comm_page->e3g_frq_register = frq;
-		chip->comm_page->control_register = ctl;
+	if (ctl_reg != chip->comm_page->control_register ||
+	    frq_reg != chip->comm_page->e3g_frq_register || force) {
+		chip->comm_page->e3g_frq_register = frq_reg;
+		chip->comm_page->control_register = ctl_reg;
 		clear_handshake(chip);
 		return send_vector(chip, DSP_VC_WRITE_CONTROL_REG);
 	}
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index 15aae2fad8e4..b181752b8481 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -679,7 +679,7 @@ static int restore_dsp_rettings(struct echoaudio *chip)
 	/* Gina20/Darla20 only. Should be harmless for other cards. */
 	chip->comm_page->gd_clock_state = GD_CLOCK_UNDEF;
 	chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_UNDEF;
-	chip->comm_page->handshake = 0xffffffff;
+	chip->comm_page->handshake = cpu_to_le32(0xffffffff);
 
 	/* Restore output busses */
 	for (i = 0; i < num_busses_out(chip); i++) {
@@ -989,7 +989,7 @@ static int init_dsp_comm_page(struct echoaudio *chip)
 	/* Init the comm page */
 	chip->comm_page->comm_size =
 		cpu_to_le32(sizeof(struct comm_page));
-	chip->comm_page->handshake = 0xffffffff;
+	chip->comm_page->handshake = cpu_to_le32(0xffffffff);
 	chip->comm_page->midi_out_free_count =
 		cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE);
 	chip->comm_page->sample_rate = cpu_to_le32(44100);
@@ -1087,7 +1087,7 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe,
 	/* The counter register is where the DSP writes the 32 bit DMA
 	position for a pipe.  The DSP is constantly updating this value as
 	it moves data. The DMA counter is in units of bytes, not samples. */
-	pipe->dma_counter = &chip->comm_page->position[pipe_index];
+	pipe->dma_counter = (__le32 *)&chip->comm_page->position[pipe_index];
 	*pipe->dma_counter = 0;
 	return pipe_index;
 }
diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h
index cb7d75a0a503..aa9129519795 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.h
+++ b/sound/pci/echoaudio/echoaudio_dsp.h
@@ -627,8 +627,8 @@ sg_entry struct is read by the DSP, so all values must be little-endian. */
 #define MAX_SGLIST_ENTRIES 512
 
 struct sg_entry {
-	u32 addr;
-	u32 size;
+	__le32 addr;
+	__le32 size;
 };
 
 
@@ -643,18 +643,18 @@ struct sg_entry {
  ****************************************************************************/
 
 struct comm_page {		/*				Base	Length*/
-	u32 comm_size;		/* size of this object		0x000	4 */
-	u32 flags;		/* See Appendix A below		0x004	4 */
-	u32 unused;		/* Unused entry			0x008	4 */
-	u32 sample_rate;	/* Card sample rate in Hz	0x00c	4 */
-	u32 handshake;		/* DSP command handshake	0x010	4 */
-	u32 cmd_start;		/* Chs. to start mask		0x014	4 */
-	u32 cmd_stop;		/* Chs. to stop mask		0x018	4 */
-	u32 cmd_reset;		/* Chs. to reset mask		0x01c	4 */
-	u16 audio_format[DSP_MAXPIPES];	/* Chs. audio format	0x020	32*2 */
+	__le32 comm_size;	/* size of this object		0x000	4 */
+	__le32 flags;		/* See Appendix A below		0x004	4 */
+	__le32 unused;		/* Unused entry			0x008	4 */
+	__le32 sample_rate;	/* Card sample rate in Hz	0x00c	4 */
+	__le32 handshake;	/* DSP command handshake	0x010	4 */
+	__le32 cmd_start;	/* Chs. to start mask		0x014	4 */
+	__le32 cmd_stop;	/* Chs. to stop mask		0x018	4 */
+	__le32 cmd_reset;	/* Chs. to reset mask		0x01c	4 */
+	__le16 audio_format[DSP_MAXPIPES];	/* Chs. audio format	0x020	32*2 */
 	struct sg_entry sglist_addr[DSP_MAXPIPES];
 				/* Chs. Physical sglist addrs	0x060	32*8 */
-	u32 position[DSP_MAXPIPES];
+	__le32 position[DSP_MAXPIPES];
 				/* Positions for ea. ch.	0x160	32*4 */
 	s8 vu_meter[DSP_MAXPIPES];
 				/* VU meters			0x1e0	32*1 */
@@ -666,28 +666,28 @@ struct comm_page {		/*				Base	Length*/
 				/* Input gain			0x230	16*1 */
 	s8 monitors[MONITOR_ARRAY_SIZE];
 				/* Monitor map			0x240	0x180 */
-	u32 play_coeff[MAX_PLAY_TAPS];
+	__le32 play_coeff[MAX_PLAY_TAPS];
 			/* Gina/Darla play filters - obsolete	0x3c0	168*4 */
-	u32 rec_coeff[MAX_REC_TAPS];
+	__le32 rec_coeff[MAX_REC_TAPS];
 			/* Gina/Darla record filters - obsolete	0x660	192*4 */
-	u16 midi_input[MIDI_IN_BUFFER_SIZE];
+	__le16 midi_input[MIDI_IN_BUFFER_SIZE];
 			/* MIDI input data transfer buffer	0x960	256*2 */
 	u8 gd_clock_state;	/* Chg Gina/Darla clock state	0xb60	1 */
 	u8 gd_spdif_status;	/* Chg. Gina/Darla S/PDIF state	0xb61	1 */
 	u8 gd_resampler_state;	/* Should always be 3		0xb62	1 */
 	u8 filler2;		/*				0xb63	1 */
-	u32 nominal_level_mask;	/* -10 level enable mask	0xb64	4 */
-	u16 input_clock;	/* Chg. Input clock state	0xb68	2 */
-	u16 output_clock;	/* Chg. Output clock state	0xb6a	2 */
-	u32 status_clocks;	/* Current Input clock state	0xb6c	4 */
-	u32 ext_box_status;	/* External box status		0xb70	4 */
-	u32 cmd_add_buffer;	/* Pipes to add (obsolete)	0xb74	4 */
-	u32 midi_out_free_count;
+	__le32 nominal_level_mask;	/* -10 level enable mask	0xb64	4 */
+	__le16 input_clock;	/* Chg. Input clock state	0xb68	2 */
+	__le16 output_clock;	/* Chg. Output clock state	0xb6a	2 */
+	__le32 status_clocks;	/* Current Input clock state	0xb6c	4 */
+	__le32 ext_box_status;	/* External box status		0xb70	4 */
+	__le32 cmd_add_buffer;	/* Pipes to add (obsolete)	0xb74	4 */
+	__le32 midi_out_free_count;
 			/* # of bytes free in MIDI output FIFO	0xb78	4 */
-	u32 unused2;		/* Cyclic pipes			0xb7c	4 */
-	u32 control_register;
+	__le32 unused2;		/* Cyclic pipes			0xb7c	4 */
+	__le32 control_register;
 			/* Mona, Gina24, Layla24, 3G ctrl reg	0xb80	4 */
-	u32 e3g_frq_register;	/* 3G frequency register	0xb84	4 */
+	__le32 e3g_frq_register;	/* 3G frequency register	0xb84	4 */
 	u8 filler[24];		/* filler			0xb88	24*1 */
 	s8 vmixer[VMIXER_ARRAY_SIZE];
 				/* Vmixer levels		0xba0	64*1 */
diff --git a/sound/pci/echoaudio/echoaudio_gml.c b/sound/pci/echoaudio/echoaudio_gml.c
index 834b39e97db7..eea6fe530ab4 100644
--- a/sound/pci/echoaudio/echoaudio_gml.c
+++ b/sound/pci/echoaudio/echoaudio_gml.c
@@ -63,6 +63,8 @@ the control register.  write_control_reg sends the new control register
 value to the DSP. */
 static int write_control_reg(struct echoaudio *chip, u32 value, char force)
 {
+	__le32 reg_value;
+
 	/* Handle the digital input auto-mute */
 	if (chip->digital_in_automute)
 		value |= GML_DIGITAL_IN_AUTO_MUTE;
@@ -72,11 +74,11 @@ static int write_control_reg(struct echoaudio *chip, u32 value, char force)
 	dev_dbg(chip->card->dev, "write_control_reg: 0x%x\n", value);
 
 	/* Write the control register */
-	value = cpu_to_le32(value);
-	if (value != chip->comm_page->control_register || force) {
+	reg_value = cpu_to_le32(value);
+	if (reg_value != chip->comm_page->control_register || force) {
 		if (wait_handshake(chip))
 			return -EIO;
-		chip->comm_page->control_register = value;
+		chip->comm_page->control_register = reg_value;
 		clear_handshake(chip);
 		return send_vector(chip, DSP_VC_WRITE_CONTROL_REG);
 	}
diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c
index 0e069aeab86d..c32eb7053715 100644
--- a/sound/pci/emu10k1/emu10k1_patch.c
+++ b/sound/pci/emu10k1/emu10k1_patch.c
@@ -70,11 +70,8 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
 		loopend = sampleend;
 
 	/* be sure loop points start < end */
-	if (sp->v.loopstart >= sp->v.loopend) {
-		int tmp = sp->v.loopstart;
-		sp->v.loopstart = sp->v.loopend;
-		sp->v.loopend = tmp;
-	}
+	if (sp->v.loopstart >= sp->v.loopend)
+		swap(sp->v.loopstart, sp->v.loopend);
 
 	/* compute true data size to be loaded */
 	truesize = sp->v.size + BLANK_HEAD_SIZE;
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index de2ecbe95d6c..90713741c2dc 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -526,7 +526,7 @@ static int snd_emu10k1_gpr_poke(struct snd_emu10k1 *emu,
 		if (!test_bit(gpr, icode->gpr_valid))
 			continue;
 		if (in_kernel)
-			val = *(u32 *)&icode->gpr_map[gpr];
+			val = *(__force u32 *)&icode->gpr_map[gpr];
 		else if (get_user(val, &icode->gpr_map[gpr]))
 			return -EFAULT;
 		snd_emu10k1_ptr_write(emu, emu->gpr_base + gpr, 0, val);
@@ -560,8 +560,8 @@ static int snd_emu10k1_tram_poke(struct snd_emu10k1 *emu,
 		if (!test_bit(tram, icode->tram_valid))
 			continue;
 		if (in_kernel) {
-			val = *(u32 *)&icode->tram_data_map[tram];
-			addr = *(u32 *)&icode->tram_addr_map[tram];
+			val = *(__force u32 *)&icode->tram_data_map[tram];
+			addr = *(__force u32 *)&icode->tram_addr_map[tram];
 		} else {
 			if (get_user(val, &icode->tram_data_map[tram]) ||
 			    get_user(addr, &icode->tram_addr_map[tram]))
@@ -611,8 +611,8 @@ static int snd_emu10k1_code_poke(struct snd_emu10k1 *emu,
 		if (!test_bit(pc / 2, icode->code_valid))
 			continue;
 		if (in_kernel) {
-			lo = *(u32 *)&icode->code[pc + 0];
-			hi = *(u32 *)&icode->code[pc + 1];
+			lo = *(__force u32 *)&icode->code[pc + 0];
+			hi = *(__force u32 *)&icode->code[pc + 1];
 		} else {
 			if (get_user(lo, &icode->code[pc + 0]) ||
 			    get_user(hi, &icode->code[pc + 1]))
@@ -666,7 +666,7 @@ static unsigned int *copy_tlv(const unsigned int __user *_tlv, bool in_kernel)
 	if (!_tlv)
 		return NULL;
 	if (in_kernel)
-		memcpy(data, (void *)_tlv, sizeof(data));
+		memcpy(data, (__force void *)_tlv, sizeof(data));
 	else if (copy_from_user(data, _tlv, sizeof(data)))
 		return NULL;
 	if (data[1] >= MAX_TLV_SIZE)
@@ -676,7 +676,7 @@ static unsigned int *copy_tlv(const unsigned int __user *_tlv, bool in_kernel)
 		return NULL;
 	memcpy(tlv, data, sizeof(data));
 	if (in_kernel) {
-		memcpy(tlv + 2, (void *)(_tlv + 2),  data[1]);
+		memcpy(tlv + 2, (__force void *)(_tlv + 2),  data[1]);
 	} else if (copy_from_user(tlv + 2, _tlv + 2, data[1])) {
 		kfree(tlv);
 		return NULL;
@@ -693,7 +693,7 @@ static int copy_gctl(struct snd_emu10k1 *emu,
 
 	if (emu->support_tlv) {
 		if (in_kernel)
-			memcpy(gctl, (void *)&_gctl[idx], sizeof(*gctl));
+			memcpy(gctl, (__force void *)&_gctl[idx], sizeof(*gctl));
 		else if (copy_from_user(gctl, &_gctl[idx], sizeof(*gctl)))
 			return -EFAULT;
 		return 0;
@@ -701,7 +701,7 @@ static int copy_gctl(struct snd_emu10k1 *emu,
 
 	octl = (struct snd_emu10k1_fx8010_control_old_gpr __user *)_gctl;
 	if (in_kernel)
-		memcpy(gctl, (void *)&octl[idx], sizeof(*octl));
+		memcpy(gctl, (__force void *)&octl[idx], sizeof(*octl));
 	else if (copy_from_user(gctl, &octl[idx], sizeof(*octl)))
 		return -EFAULT;
 	gctl->tlv = NULL;
@@ -735,7 +735,7 @@ static int snd_emu10k1_verify_controls(struct snd_emu10k1 *emu,
 	for (i = 0, _id = icode->gpr_del_controls;
 	     i < icode->gpr_del_control_count; i++, _id++) {
 		if (in_kernel)
-			id = *(struct snd_ctl_elem_id *)_id;
+			id = *(__force struct snd_ctl_elem_id *)_id;
 		else if (copy_from_user(&id, _id, sizeof(id)))
 	     		return -EFAULT;
 		if (snd_emu10k1_look_for_ctl(emu, &id) == NULL)
@@ -833,7 +833,7 @@ static int snd_emu10k1_add_controls(struct snd_emu10k1 *emu,
 		knew.device = gctl->id.device;
 		knew.subdevice = gctl->id.subdevice;
 		knew.info = snd_emu10k1_gpr_ctl_info;
-		knew.tlv.p = copy_tlv(gctl->tlv, in_kernel);
+		knew.tlv.p = copy_tlv((__force const unsigned int __user *)gctl->tlv, in_kernel);
 		if (knew.tlv.p)
 			knew.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
 				SNDRV_CTL_ELEM_ACCESS_TLV_READ;
@@ -897,7 +897,7 @@ static int snd_emu10k1_del_controls(struct snd_emu10k1 *emu,
 	for (i = 0, _id = icode->gpr_del_controls;
 	     i < icode->gpr_del_control_count; i++, _id++) {
 		if (in_kernel)
-			id = *(struct snd_ctl_elem_id *)_id;
+			id = *(__force struct snd_ctl_elem_id *)_id;
 		else if (copy_from_user(&id, _id, sizeof(id)))
 			return -EFAULT;
 		down_write(&card->controls_rwsem);
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index 69f9b100bd24..9f2b6097f486 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -290,7 +290,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	unsigned int silent_page, tmp;
 	int voice, stereo, w_16;
-	unsigned char attn, send_amount[8];
+	unsigned char send_amount[8];
 	unsigned char send_routing[8];
 	unsigned long flags;
 	unsigned int pitch_target;
@@ -313,7 +313,6 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu,
 
 	/* volume parameters */
 	if (extra) {
-		attn = 0;
 		memset(send_routing, 0, sizeof(send_routing));
 		send_routing[0] = 0;
 		send_routing[1] = 1;
@@ -779,7 +778,7 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_START:
 		snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra);	/* do we need this? */
 		snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[0]);
-		/* follow thru */
+		/* fall through */
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 	case SNDRV_PCM_TRIGGER_RESUME:
 		if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE)
@@ -929,7 +928,7 @@ static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream,
 		}
 		snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra);
 
-		/* follow thru */
+		/* fall through */
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 	case SNDRV_PCM_TRIGGER_RESUME:
 		snd_emu10k1_playback_prepare_voice(emu, epcm->extra, 1, 1, NULL);
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 39f79a6b5283..727eb3da1fda 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -2392,7 +2392,7 @@ static int snd_audiopci_probe(struct pci_dev *pci,
 	static int dev;
 	struct snd_card *card;
 	struct ensoniq *ensoniq;
-	int err, pcm_devs[2];
+	int err;
 
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
@@ -2412,7 +2412,6 @@ static int snd_audiopci_probe(struct pci_dev *pci,
 	}
 	card->private_data = ensoniq;
 
-	pcm_devs[0] = 0; pcm_devs[1] = 1;
 #ifdef CHIP1370
 	if ((err = snd_ensoniq_1370_mixer(ensoniq)) < 0) {
 		snd_card_free(card);
diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c
index 1b48a8c19d28..8a7dbd1a7fbf 100644
--- a/sound/pci/hda/dell_wmi_helper.c
+++ b/sound/pci/hda/dell_wmi_helper.c
@@ -6,111 +6,18 @@
 #if IS_ENABLED(CONFIG_DELL_LAPTOP)
 #include <linux/dell-led.h>
 
-enum {
-	MICMUTE_LED_ON,
-	MICMUTE_LED_OFF,
-	MICMUTE_LED_FOLLOW_CAPTURE,
-	MICMUTE_LED_FOLLOW_MUTE,
-};
-
-static int dell_led_mode = MICMUTE_LED_FOLLOW_MUTE;
-static int dell_capture;
-static int dell_led_value;
 static int (*dell_micmute_led_set_func)(int);
-static void (*dell_old_cap_hook)(struct hda_codec *,
-			         struct snd_kcontrol *,
-				 struct snd_ctl_elem_value *);
-
-static void call_micmute_led_update(void)
-{
-	int val;
-
-	switch (dell_led_mode) {
-	case MICMUTE_LED_ON:
-		val = 1;
-		break;
-	case MICMUTE_LED_OFF:
-		val = 0;
-		break;
-	case MICMUTE_LED_FOLLOW_CAPTURE:
-		val = dell_capture;
-		break;
-	case MICMUTE_LED_FOLLOW_MUTE:
-	default:
-		val = !dell_capture;
-		break;
-	}
-
-	if (val == dell_led_value)
-		return;
-	dell_led_value = val;
-	dell_micmute_led_set_func(dell_led_value);
-}
-
-static void update_dell_wmi_micmute_led(struct hda_codec *codec,
-				        struct snd_kcontrol *kcontrol,
-					struct snd_ctl_elem_value *ucontrol)
-{
-	if (dell_old_cap_hook)
-		dell_old_cap_hook(codec, kcontrol, ucontrol);
-
-	if (!ucontrol || !dell_micmute_led_set_func)
-		return;
-	if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) {
-		/* TODO: How do I verify if it's a mono or stereo here? */
-		dell_capture = (ucontrol->value.integer.value[0] ||
-				ucontrol->value.integer.value[1]);
-		call_micmute_led_update();
-	}
-}
 
-static int dell_mic_mute_led_mode_info(struct snd_kcontrol *kcontrol,
-				       struct snd_ctl_elem_info *uinfo)
+static void dell_micmute_update(struct hda_codec *codec)
 {
-	static const char * const texts[] = {
-		"On", "Off", "Follow Capture", "Follow Mute",
-	};
-
-	return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts);
-}
+	struct hda_gen_spec *spec = codec->spec;
 
-static int dell_mic_mute_led_mode_get(struct snd_kcontrol *kcontrol,
-				      struct snd_ctl_elem_value *ucontrol)
-{
-	ucontrol->value.enumerated.item[0] = dell_led_mode;
-	return 0;
+	dell_micmute_led_set_func(spec->micmute_led.led_value);
 }
 
-static int dell_mic_mute_led_mode_put(struct snd_kcontrol *kcontrol,
-				      struct snd_ctl_elem_value *ucontrol)
-{
-	unsigned int mode;
-
-	mode = ucontrol->value.enumerated.item[0];
-	if (mode > MICMUTE_LED_FOLLOW_MUTE)
-		mode = MICMUTE_LED_FOLLOW_MUTE;
-	if (mode == dell_led_mode)
-		return 0;
-	dell_led_mode = mode;
-	call_micmute_led_update();
-	return 1;
-}
-
-static const struct snd_kcontrol_new dell_mic_mute_mode_ctls[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Mic Mute-LED Mode",
-		.info = dell_mic_mute_led_mode_info,
-		.get = dell_mic_mute_led_mode_get,
-		.put = dell_mic_mute_led_mode_put,
-	},
-	{}
-};
-
 static void alc_fixup_dell_wmi(struct hda_codec *codec,
 			       const struct hda_fixup *fix, int action)
 {
-	struct alc_spec *spec = codec->spec;
 	bool removefunc = false;
 
 	if (action == HDA_FIXUP_ACT_PROBE) {
@@ -121,25 +28,14 @@ static void alc_fixup_dell_wmi(struct hda_codec *codec,
 			return;
 		}
 
-		removefunc = true;
-		if (dell_micmute_led_set_func(false) >= 0) {
-			dell_led_value = 0;
-			if (spec->gen.num_adc_nids > 1 && !spec->gen.dyn_adc_switch)
-				codec_dbg(codec, "Skipping micmute LED control due to several ADCs");
-			else {
-				dell_old_cap_hook = spec->gen.cap_sync_hook;
-				spec->gen.cap_sync_hook = update_dell_wmi_micmute_led;
-				removefunc = false;
-				add_mixer(spec, dell_mic_mute_mode_ctls);
-			}
-		}
-
+		removefunc = (dell_micmute_led_set_func(false) < 0) ||
+			(snd_hda_gen_add_micmute_led(codec,
+						     dell_micmute_update) <= 0);
 	}
 
 	if (dell_micmute_led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) {
 		symbol_put(dell_micmute_led_set);
 		dell_micmute_led_set_func = NULL;
-		dell_old_cap_hook = NULL;
 	}
 }
 
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 20a171ac4bb2..0a5085537034 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -37,15 +37,8 @@
 #include "hda_jack.h"
 #include <sound/hda_hwdep.h>
 
-#ifdef CONFIG_PM
-#define codec_in_pm(codec)	atomic_read(&(codec)->core.in_pm)
-#define hda_codec_is_power_on(codec) \
-	(!pm_runtime_suspended(hda_codec_dev(codec)))
-#else
-#define codec_in_pm(codec)	0
-#define hda_codec_is_power_on(codec)	1
-#endif
-
+#define codec_in_pm(codec)		snd_hdac_is_in_pm(&codec->core)
+#define hda_codec_is_power_on(codec)	snd_hdac_is_power_on(&codec->core)
 #define codec_has_epss(codec) \
 	((codec)->core.power_caps & AC_PWRST_EPSS)
 #define codec_has_clkstop(codec) \
@@ -858,6 +851,39 @@ static void snd_hda_codec_dev_release(struct device *dev)
 	kfree(codec);
 }
 
+#define DEV_NAME_LEN 31
+
+static int snd_hda_codec_device_init(struct hda_bus *bus, struct snd_card *card,
+			unsigned int codec_addr, struct hda_codec **codecp)
+{
+	char name[DEV_NAME_LEN];
+	struct hda_codec *codec;
+	int err;
+
+	dev_dbg(card->dev, "%s: entry\n", __func__);
+
+	if (snd_BUG_ON(!bus))
+		return -EINVAL;
+	if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS))
+		return -EINVAL;
+
+	codec = kzalloc(sizeof(*codec), GFP_KERNEL);
+	if (!codec)
+		return -ENOMEM;
+
+	sprintf(name, "hdaudioC%dD%d", card->number, codec_addr);
+	err = snd_hdac_device_init(&codec->core, &bus->core, name, codec_addr);
+	if (err < 0) {
+		kfree(codec);
+		return err;
+	}
+
+	codec->core.type = HDA_DEV_LEGACY;
+	*codecp = codec;
+
+	return err;
+}
+
 /**
  * snd_hda_codec_new - create a HDA codec
  * @bus: the bus to assign
@@ -869,7 +895,19 @@ static void snd_hda_codec_dev_release(struct device *dev)
 int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card,
 		      unsigned int codec_addr, struct hda_codec **codecp)
 {
-	struct hda_codec *codec;
+	int ret;
+
+	ret = snd_hda_codec_device_init(bus, card, codec_addr, codecp);
+	if (ret < 0)
+		return ret;
+
+	return snd_hda_codec_device_new(bus, card, codec_addr, *codecp);
+}
+EXPORT_SYMBOL_GPL(snd_hda_codec_new);
+
+int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card,
+			unsigned int codec_addr, struct hda_codec *codec)
+{
 	char component[31];
 	hda_nid_t fg;
 	int err;
@@ -879,25 +917,14 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card,
 		.dev_free = snd_hda_codec_dev_free,
 	};
 
+	dev_dbg(card->dev, "%s: entry\n", __func__);
+
 	if (snd_BUG_ON(!bus))
 		return -EINVAL;
 	if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS))
 		return -EINVAL;
 
-	codec = kzalloc(sizeof(*codec), GFP_KERNEL);
-	if (!codec)
-		return -ENOMEM;
-
-	sprintf(component, "hdaudioC%dD%d", card->number, codec_addr);
-	err = snd_hdac_device_init(&codec->core, &bus->core, component,
-				   codec_addr);
-	if (err < 0) {
-		kfree(codec);
-		return err;
-	}
-
 	codec->core.dev.release = snd_hda_codec_dev_release;
-	codec->core.type = HDA_DEV_LEGACY;
 	codec->core.exec_verb = codec_exec_verb;
 
 	codec->bus = bus;
@@ -957,15 +984,13 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card,
 	if (err < 0)
 		goto error;
 
-	if (codecp)
-		*codecp = codec;
 	return 0;
 
  error:
 	put_device(hda_codec_dev(codec));
 	return err;
 }
-EXPORT_SYMBOL_GPL(snd_hda_codec_new);
+EXPORT_SYMBOL_GPL(snd_hda_codec_device_new);
 
 /**
  * snd_hda_codec_update_widgets - Refresh widget caps and pin defaults
@@ -2846,14 +2871,13 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec)
 {
 	unsigned int state;
 
-	atomic_inc(&codec->core.in_pm);
-
+	snd_hdac_enter_pm(&codec->core);
 	if (codec->patch_ops.suspend)
 		codec->patch_ops.suspend(codec);
 	hda_cleanup_all_streams(codec);
 	state = hda_set_power_state(codec, AC_PWRST_D3);
 	update_power_acct(codec, true);
-	atomic_dec(&codec->core.in_pm);
+	snd_hdac_leave_pm(&codec->core);
 	return state;
 }
 
@@ -2862,8 +2886,7 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec)
  */
 static void hda_call_codec_resume(struct hda_codec *codec)
 {
-	atomic_inc(&codec->core.in_pm);
-
+	snd_hdac_enter_pm(&codec->core);
 	if (codec->core.regmap)
 		regcache_mark_dirty(codec->core.regmap);
 
@@ -2886,7 +2909,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
 		hda_jackpoll_work(&codec->jackpoll_work.work);
 	else
 		snd_hda_jack_report_sync(codec);
-	atomic_dec(&codec->core.in_pm);
+	snd_hdac_leave_pm(&codec->core);
 }
 
 static int hda_codec_runtime_suspend(struct device *dev)
@@ -2992,6 +3015,7 @@ int snd_hda_codec_build_controls(struct hda_codec *codec)
 	sync_power_up_states(codec);
 	return 0;
 }
+EXPORT_SYMBOL_GPL(snd_hda_codec_build_controls);
 
 /*
  * PCM stuff
@@ -3197,6 +3221,7 @@ int snd_hda_codec_parse_pcms(struct hda_codec *codec)
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(snd_hda_codec_parse_pcms);
 
 /* assign all PCMs of the given codec */
 int snd_hda_codec_build_pcms(struct hda_codec *codec)
@@ -3252,8 +3277,8 @@ int snd_hda_add_new_ctls(struct hda_codec *codec,
 	for (; knew->name; knew++) {
 		struct snd_kcontrol *kctl;
 		int addr = 0, idx = 0;
-		if (knew->iface == -1)	/* skip this codec private value */
-			continue;
+		if (knew->iface == (__force snd_ctl_elem_iface_t)-1)
+			continue; /* skip this codec private value */
 		for (;;) {
 			kctl = snd_ctl_new1(knew, codec);
 			if (!kctl)
@@ -3843,7 +3868,7 @@ EXPORT_SYMBOL_GPL(snd_hda_correct_pin_ctl);
  * This function is a helper to set a pin ctl value more safely.
  * It corrects the pin ctl value via snd_hda_correct_pin_ctl(), stores the
  * value in pin target array via snd_hda_codec_set_pin_target(), then
- * actually writes the value via either snd_hda_codec_update_cache() or
+ * actually writes the value via either snd_hda_codec_write_cache() or
  * snd_hda_codec_write() depending on @cached flag.
  */
 int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin,
@@ -3852,7 +3877,7 @@ int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin,
 	val = snd_hda_correct_pin_ctl(codec, pin, val);
 	snd_hda_codec_set_pin_target(codec, pin, val);
 	if (cached)
-		return snd_hda_codec_update_cache(codec, pin, 0,
+		return snd_hda_codec_write_cache(codec, pin, 0,
 				AC_VERB_SET_PIN_WIDGET_CONTROL, val);
 	else
 		return snd_hda_codec_write(codec, pin, 0,
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index a8b1b31f161c..0d98bb9068b1 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -84,6 +84,7 @@ struct hda_bus {
  */
 typedef int (*hda_codec_patch_t)(struct hda_codec *);
 	
+#define HDA_CODEC_ID_SKIP_PROBE		0x00000001
 #define HDA_CODEC_ID_GENERIC_HDMI	0x00000101
 #define HDA_CODEC_ID_GENERIC		0x00000201
 
@@ -308,6 +309,8 @@ struct hda_codec {
  */
 int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card,
 		      unsigned int codec_addr, struct hda_codec **codecp);
+int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card,
+		      unsigned int codec_addr, struct hda_codec *codec);
 int snd_hda_codec_configure(struct hda_codec *codec);
 int snd_hda_codec_update_widgets(struct hda_codec *codec);
 
@@ -382,9 +385,6 @@ snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
 	return snd_hdac_regmap_write(&codec->core, nid, verb, parm);
 }
 
-#define snd_hda_codec_update_cache(codec, nid, flags, verb, parm) \
-	snd_hda_codec_write_cache(codec, nid, flags, verb, parm)
-
 /* the struct for codec->pin_configs */
 struct hda_pincfg {
 	hda_nid_t nid;
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index db773e219aaa..579984ecdec3 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -209,7 +209,7 @@ static void parse_user_hints(struct hda_codec *codec)
  */
 
 #define update_pin_ctl(codec, pin, val) \
-	snd_hda_codec_update_cache(codec, pin, 0, \
+	snd_hda_codec_write_cache(codec, pin, 0, \
 				   AC_VERB_SET_PIN_WIDGET_CONTROL, val)
 
 /* restore the pinctl based on the cached value */
@@ -898,7 +898,7 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path,
 		hda_nid_t nid = path->path[i];
 
 		if (enable && path->multi[i])
-			snd_hda_codec_update_cache(codec, nid, 0,
+			snd_hda_codec_write_cache(codec, nid, 0,
 					    AC_VERB_SET_CONNECT_SEL,
 					    path->idx[i]);
 		if (has_amp_in(codec, path, i))
@@ -930,7 +930,7 @@ static void set_pin_eapd(struct hda_codec *codec, hda_nid_t pin, bool enable)
 		return;
 	if (codec->inv_eapd)
 		enable = !enable;
-	snd_hda_codec_update_cache(codec, pin, 0,
+	snd_hda_codec_write_cache(codec, pin, 0,
 				   AC_VERB_SET_EAPD_BTLENABLE,
 				   enable ? 0x02 : 0x00);
 }
@@ -3900,6 +3900,142 @@ static int parse_mic_boost(struct hda_codec *codec)
 }
 
 /*
+ * mic mute LED hook helpers
+ */
+enum {
+	MICMUTE_LED_ON,
+	MICMUTE_LED_OFF,
+	MICMUTE_LED_FOLLOW_CAPTURE,
+	MICMUTE_LED_FOLLOW_MUTE,
+};
+
+static void call_micmute_led_update(struct hda_codec *codec)
+{
+	struct hda_gen_spec *spec = codec->spec;
+	unsigned int val;
+
+	switch (spec->micmute_led.led_mode) {
+	case MICMUTE_LED_ON:
+		val = 1;
+		break;
+	case MICMUTE_LED_OFF:
+		val = 0;
+		break;
+	case MICMUTE_LED_FOLLOW_CAPTURE:
+		val = !!spec->micmute_led.capture;
+		break;
+	case MICMUTE_LED_FOLLOW_MUTE:
+	default:
+		val = !spec->micmute_led.capture;
+		break;
+	}
+
+	if (val == spec->micmute_led.led_value)
+		return;
+	spec->micmute_led.led_value = val;
+	if (spec->micmute_led.update)
+		spec->micmute_led.update(codec);
+}
+
+static void update_micmute_led(struct hda_codec *codec,
+			       struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_gen_spec *spec = codec->spec;
+	unsigned int mask;
+
+	if (spec->micmute_led.old_hook)
+		spec->micmute_led.old_hook(codec, kcontrol, ucontrol);
+
+	if (!ucontrol)
+		return;
+	mask = 1U << snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+	if (!strcmp("Capture Switch", ucontrol->id.name)) {
+		/* TODO: How do I verify if it's a mono or stereo here? */
+		if (ucontrol->value.integer.value[0] ||
+		    ucontrol->value.integer.value[1])
+			spec->micmute_led.capture |= mask;
+		else
+			spec->micmute_led.capture &= ~mask;
+		call_micmute_led_update(codec);
+	}
+}
+
+static int micmute_led_mode_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	static const char * const texts[] = {
+		"On", "Off", "Follow Capture", "Follow Mute",
+	};
+
+	return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts);
+}
+
+static int micmute_led_mode_get(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;
+
+	ucontrol->value.enumerated.item[0] = spec->micmute_led.led_mode;
+	return 0;
+}
+
+static int micmute_led_mode_put(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;
+	unsigned int mode;
+
+	mode = ucontrol->value.enumerated.item[0];
+	if (mode > MICMUTE_LED_FOLLOW_MUTE)
+		mode = MICMUTE_LED_FOLLOW_MUTE;
+	if (mode == spec->micmute_led.led_mode)
+		return 0;
+	spec->micmute_led.led_mode = mode;
+	call_micmute_led_update(codec);
+	return 1;
+}
+
+static const struct snd_kcontrol_new micmute_led_mode_ctl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Mic Mute-LED Mode",
+	.info = micmute_led_mode_info,
+	.get = micmute_led_mode_get,
+	.put = micmute_led_mode_put,
+};
+
+/**
+ * snd_hda_gen_add_micmute_led - helper for setting up mic mute LED hook
+ * @codec: the HDA codec
+ * @hook: the callback for updating LED
+ *
+ * Called from the codec drivers for offering the mic mute LED controls.
+ * When established, it sets up cap_sync_hook and triggers the callback at
+ * each time when the capture mixer switch changes.  The callback is supposed
+ * to update the LED accordingly.
+ *
+ * Returns 0 if the hook is established or a negative error code.
+ */
+int snd_hda_gen_add_micmute_led(struct hda_codec *codec,
+				void (*hook)(struct hda_codec *))
+{
+	struct hda_gen_spec *spec = codec->spec;
+
+	spec->micmute_led.led_mode = MICMUTE_LED_FOLLOW_MUTE;
+	spec->micmute_led.capture = 0;
+	spec->micmute_led.led_value = 0;
+	spec->micmute_led.old_hook = spec->cap_sync_hook;
+	spec->micmute_led.update = hook;
+	spec->cap_sync_hook = update_micmute_led;
+	if (!snd_hda_gen_add_kctl(spec, NULL, &micmute_led_mode_ctl))
+		return -ENOMEM;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_hda_gen_add_micmute_led);
+
+/*
  * parse digital I/Os and set up NIDs in BIOS auto-parse mode
  */
 static void parse_digital(struct hda_codec *codec)
@@ -5837,7 +5973,7 @@ static void clear_unsol_on_unused_pins(struct hda_codec *codec)
 		hda_nid_t nid = pin->nid;
 		if (is_jack_detectable(codec, nid) &&
 		    !snd_hda_jack_tbl_get(codec, nid))
-			snd_hda_codec_update_cache(codec, nid, 0,
+			snd_hda_codec_write_cache(codec, nid, 0,
 					AC_VERB_SET_UNSOLICITED_ENABLE, 0);
 	}
 }
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 61772317de46..10123664fa61 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -86,6 +86,16 @@ struct badness_table {
 extern const struct badness_table hda_main_out_badness;
 extern const struct badness_table hda_extra_out_badness;
 
+struct hda_micmute_hook {
+	unsigned int led_mode;
+	unsigned int capture;
+	unsigned int led_value;
+	void (*update)(struct hda_codec *codec);
+	void (*old_hook)(struct hda_codec *codec,
+			 struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol);
+};
+
 struct hda_gen_spec {
 	char stream_name_analog[32];	/* analog PCM stream */
 	const struct hda_pcm_stream *stream_analog_playback;
@@ -276,6 +286,9 @@ struct hda_gen_spec {
 			      struct snd_kcontrol *kcontrol,
 			      struct snd_ctl_elem_value *ucontrol);
 
+	/* mic mute LED hook; called via cap_sync_hook */
+	struct hda_micmute_hook micmute_led;
+
 	/* PCM hooks */
 	void (*pcm_playback_hook)(struct hda_pcm_stream *hinfo,
 				  struct hda_codec *codec,
@@ -342,4 +355,7 @@ unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec,
 void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on);
 int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin);
 
+int snd_hda_gen_add_micmute_led(struct hda_codec *codec,
+				void (*hook)(struct hda_codec *));
+
 #endif /* __SOUND_HDA_GENERIC_H */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 1ae1850b3bfd..1b2ce304152a 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1319,15 +1319,16 @@ static const struct vga_switcheroo_client_ops azx_vs_ops = {
 static int register_vga_switcheroo(struct azx *chip)
 {
 	struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
+	struct pci_dev *p;
 	int err;
 
 	if (!hda->use_vga_switcheroo)
 		return 0;
-	/* FIXME: currently only handling DIS controller
-	 * is there any machine with two switchable HDMI audio controllers?
-	 */
-	err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops,
-						   VGA_SWITCHEROO_DIS);
+
+	p = get_bound_vga(chip->pci);
+	err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, p);
+	pci_dev_put(p);
+
 	if (err < 0)
 		return err;
 	hda->vga_switcheroo_registered = 1;
@@ -1429,7 +1430,7 @@ static struct pci_dev *get_bound_vga(struct pci_dev *pci)
 			p = pci_get_domain_bus_and_slot(pci_domain_nr(pci->bus),
 							pci->bus->number, 0);
 			if (p) {
-				if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA)
+				if ((p->class >> 16) == PCI_BASE_CLASS_DISPLAY)
 					return p;
 				pci_dev_put(p);
 			}
@@ -2207,7 +2208,7 @@ out_free:
  */
 static struct snd_pci_quirk power_save_blacklist[] = {
 	/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
-	SND_PCI_QUIRK(0x1849, 0x0c0c, "Asrock B85M-ITX", 0),
+	SND_PCI_QUIRK(0x1849, 0xc892, "Asrock B85M-ITX", 0),
 	/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
 	SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0),
 	/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
@@ -2535,7 +2536,8 @@ static const struct pci_device_id azx_ids[] = {
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB },
 	/* AMD Raven */
 	{ PCI_DEVICE(0x1022, 0x15e3),
-	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB },
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
+			 AZX_DCAPS_PM_RUNTIME },
 	/* ATI HDMI */
 	{ PCI_DEVICE(0x1002, 0x0002),
 	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 757857313426..fd476fb40e1b 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -148,7 +148,7 @@ static void ad_vmaster_eapd_hook(void *private_data, int enabled)
 		return;
 	if (codec->inv_eapd)
 		enabled = !enabled;
-	snd_hda_codec_update_cache(codec, spec->eapd_nid, 0,
+	snd_hda_codec_write_cache(codec, spec->eapd_nid, 0,
 				   AC_VERB_SET_EAPD_BTLENABLE,
 				   enabled ? 0x02 : 0x00);
 }
@@ -991,7 +991,7 @@ static void ad1884_vmaster_hp_gpio_hook(void *private_data, int enabled)
 
 	if (spec->eapd_nid)
 		ad_vmaster_eapd_hook(private_data, enabled);
-	snd_hda_codec_update_cache(codec, 0x01, 0,
+	snd_hda_codec_write_cache(codec, 0x01, 0,
 				   AC_VERB_SET_GPIO_DATA,
 				   enabled ? 0x00 : 0x02);
 }
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 321e95c409c1..0166a3d7cd55 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -897,7 +897,7 @@ struct ca0132_spec {
 	const struct hda_verb *base_init_verbs;
 	const struct hda_verb *base_exit_verbs;
 	const struct hda_verb *chip_init_verbs;
-	const struct hda_verb *sbz_init_verbs;
+	const struct hda_verb *desktop_init_verbs;
 	struct hda_verb *spec_init_verbs;
 	struct auto_pin_cfg autocfg;
 
@@ -965,9 +965,11 @@ struct ca0132_spec {
 	long cur_ctl_vals[TUNING_CTLS_COUNT];
 #endif
 	/*
-	 * Sound Blaster Z PCI region 2 iomem, used for input and output
-	 * switching, and other unknown commands.
+	 * The Recon3D, Sound Blaster Z, Sound Blaster ZxR, and Sound Blaster
+	 * AE-5 all use PCI region 2 to toggle GPIO and other currently unknown
+	 * things.
 	 */
+	bool use_pci_mmio;
 	void __iomem *mem_base;
 
 	/*
@@ -994,6 +996,7 @@ enum {
 	QUIRK_ALIENWARE_M17XR4,
 	QUIRK_SBZ,
 	QUIRK_R3DI,
+	QUIRK_R3D,
 };
 
 static const struct hda_pintbl alienware_pincfgs[] = {
@@ -1025,6 +1028,21 @@ static const struct hda_pintbl sbz_pincfgs[] = {
 	{}
 };
 
+/* Recon3D pin configs taken from Windows Driver */
+static const struct hda_pintbl r3d_pincfgs[] = {
+	{ 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */
+	{ 0x0c, 0x014510f0 }, /* SPDIF Out 1 */
+	{ 0x0d, 0x014510f0 }, /* Digital Out */
+	{ 0x0e, 0x01c520f0 }, /* SPDIF In */
+	{ 0x0f, 0x0221401f }, /* Port A -- BackPanel HP */
+	{ 0x10, 0x01016011 }, /* Port D -- Center/LFE or FP Hp */
+	{ 0x11, 0x01011014 }, /* Port B -- LineMicIn2 / Rear L/R */
+	{ 0x12, 0x02a090f0 }, /* Port C -- LineIn1 */
+	{ 0x13, 0x908700f0 }, /* What U Hear In*/
+	{ 0x18, 0x50d000f0 }, /* N/A */
+	{}
+};
+
 /* Recon3D integrated pin configs taken from Windows Driver */
 static const struct hda_pintbl r3di_pincfgs[] = {
 	{ 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */
@@ -1050,6 +1068,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
 	SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI),
 	SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI),
 	SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI),
+	SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D),
 	{}
 };
 
@@ -3073,6 +3092,24 @@ static bool dspload_wait_loaded(struct hda_codec *codec)
  */
 
 /*
+ * For cards with PCI-E region2 (Sound Blaster Z/ZxR, Recon3D, and AE-5)
+ * the mmio address 0x320 is used to set GPIO pins. The format for the data
+ * The first eight bits are just the number of the pin. So far, I've only seen
+ * this number go to 7.
+ */
+static void ca0132_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin,
+		bool enable)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned short gpio_data;
+
+	gpio_data = gpio_pin & 0xF;
+	gpio_data |= ((enable << 8) & 0x100);
+
+	writew(gpio_data, spec->mem_base + 0x320);
+}
+
+/*
  * Sets up the GPIO pins so that they are discoverable. If this isn't done,
  * the card shows as having no GPIO pins.
  */
@@ -3947,15 +3984,19 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		/*speaker out config*/
 		switch (spec->quirk) {
 		case QUIRK_SBZ:
-			writew(0x0007, spec->mem_base + 0x320);
-			writew(0x0104, spec->mem_base + 0x320);
-			writew(0x0101, spec->mem_base + 0x320);
+			ca0132_mmio_gpio_set(codec, 7, false);
+			ca0132_mmio_gpio_set(codec, 4, true);
+			ca0132_mmio_gpio_set(codec, 1, true);
 			chipio_set_control_param(codec, 0x0D, 0x18);
 			break;
 		case QUIRK_R3DI:
 			chipio_set_control_param(codec, 0x0D, 0x24);
 			r3di_gpio_out_set(codec, R3DI_LINE_OUT);
 			break;
+		case QUIRK_R3D:
+			chipio_set_control_param(codec, 0x0D, 0x24);
+			ca0132_mmio_gpio_set(codec, 1, true);
+			break;
 		}
 
 		/* disable headphone node */
@@ -3983,15 +4024,19 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		/* Headphone out config*/
 		switch (spec->quirk) {
 		case QUIRK_SBZ:
-			writew(0x0107, spec->mem_base + 0x320);
-			writew(0x0104, spec->mem_base + 0x320);
-			writew(0x0001, spec->mem_base + 0x320);
+			ca0132_mmio_gpio_set(codec, 7, true);
+			ca0132_mmio_gpio_set(codec, 4, true);
+			ca0132_mmio_gpio_set(codec, 1, false);
 			chipio_set_control_param(codec, 0x0D, 0x12);
 			break;
 		case QUIRK_R3DI:
 			chipio_set_control_param(codec, 0x0D, 0x21);
 			r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT);
 			break;
+		case QUIRK_R3D:
+			chipio_set_control_param(codec, 0x0D, 0x21);
+			ca0132_mmio_gpio_set(codec, 0x1, false);
+			break;
 		}
 
 		snd_hda_codec_write(codec, spec->out_pins[0], 0,
@@ -4025,15 +4070,19 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		/* Surround out config*/
 		switch (spec->quirk) {
 		case QUIRK_SBZ:
-			writew(0x0007, spec->mem_base + 0x320);
-			writew(0x0104, spec->mem_base + 0x320);
-			writew(0x0101, spec->mem_base + 0x320);
+			ca0132_mmio_gpio_set(codec, 7, false);
+			ca0132_mmio_gpio_set(codec, 4, true);
+			ca0132_mmio_gpio_set(codec, 1, true);
 			chipio_set_control_param(codec, 0x0D, 0x18);
 			break;
 		case QUIRK_R3DI:
 			chipio_set_control_param(codec, 0x0D, 0x24);
 			r3di_gpio_out_set(codec, R3DI_LINE_OUT);
 			break;
+		case QUIRK_R3D:
+			ca0132_mmio_gpio_set(codec, 1, true);
+			chipio_set_control_param(codec, 0x0D, 0x24);
+			break;
 		}
 		/* enable line out node */
 		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0,
@@ -4291,7 +4340,8 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 	case REAR_MIC:
 		switch (spec->quirk) {
 		case QUIRK_SBZ:
-			writew(0x0000, spec->mem_base + 0x320);
+		case QUIRK_R3D:
+			ca0132_mmio_gpio_set(codec, 0, false);
 			tmp = FLOAT_THREE;
 			break;
 		case QUIRK_R3DI:
@@ -4323,7 +4373,8 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 		ca0132_mic_boost_set(codec, 0);
 		switch (spec->quirk) {
 		case QUIRK_SBZ:
-			writew(0x0000, spec->mem_base + 0x320);
+		case QUIRK_R3D:
+			ca0132_mmio_gpio_set(codec, 0, false);
 			break;
 		case QUIRK_R3DI:
 			r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
@@ -4349,8 +4400,9 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 	case FRONT_MIC:
 		switch (spec->quirk) {
 		case QUIRK_SBZ:
-			writew(0x0100, spec->mem_base + 0x320);
-			writew(0x0005, spec->mem_base + 0x320);
+		case QUIRK_R3D:
+			ca0132_mmio_gpio_set(codec, 0, true);
+			ca0132_mmio_gpio_set(codec, 5, false);
 			tmp = FLOAT_THREE;
 			break;
 		case QUIRK_R3DI:
@@ -5516,8 +5568,7 @@ static int ca0132_alt_add_effect_slider(struct hda_codec *codec, hda_nid_t nid,
 
 	sprintf(namestr, "FX: %s %s Volume", pfx, dirstr[dir]);
 
-	knew.tlv.c = 0;
-	knew.tlv.p = 0;
+	knew.tlv.c = NULL;
 
 	switch (nid) {
 	case XBASS_XOVER:
@@ -5729,11 +5780,11 @@ static const struct snd_kcontrol_new ca0132_mixer[] = {
 };
 
 /*
- * SBZ specific control mixer. Removes auto-detect for mic, and adds surround
- * controls. Also sets both the Front Playback and Capture Volume controls to
- * alt so they set the DSP's decibel level.
+ * Desktop specific control mixer. Removes auto-detect for mic, and adds
+ * surround controls. Also sets both the Front Playback and Capture Volume
+ * controls to alt so they set the DSP's decibel level.
  */
-static const struct snd_kcontrol_new sbz_mixer[] = {
+static const struct snd_kcontrol_new desktop_mixer[] = {
 	CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT),
 	CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x04, 0, HDA_OUTPUT),
@@ -5804,8 +5855,8 @@ static int ca0132_build_controls(struct hda_codec *codec)
 	 */
 	num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT;
 	for (i = 0; i < num_fx; i++) {
-		/* SBZ breaks if Echo Cancellation is used */
-		if (spec->quirk == QUIRK_SBZ) {
+		/* SBZ and R3D break if Echo Cancellation is used. */
+		if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3D) {
 			if (i == (ECHO_CANCELLATION - IN_EFFECT_START_NID +
 						OUT_EFFECTS_COUNT))
 				continue;
@@ -6187,10 +6238,10 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec)
 }
 
 /*
- * Recon3Di r3di_setup_defaults sub functions.
+ * Recon3D r3d_setup_defaults sub functions.
  */
 
-static void r3di_dsp_scp_startup(struct hda_codec *codec)
+static void r3d_dsp_scp_startup(struct hda_codec *codec)
 {
 	unsigned int tmp;
 
@@ -6211,7 +6262,7 @@ static void r3di_dsp_scp_startup(struct hda_codec *codec)
 
 }
 
-static void r3di_dsp_initial_mic_setup(struct hda_codec *codec)
+static void r3d_dsp_initial_mic_setup(struct hda_codec *codec)
 {
 	unsigned int tmp;
 
@@ -6421,10 +6472,10 @@ static void ca0132_setup_defaults(struct hda_codec *codec)
 }
 
 /*
- * Setup default parameters for Recon3Di DSP.
+ * Setup default parameters for Recon3D/Recon3Di DSP.
  */
 
-static void r3di_setup_defaults(struct hda_codec *codec)
+static void r3d_setup_defaults(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
 	unsigned int tmp;
@@ -6434,9 +6485,9 @@ static void r3di_setup_defaults(struct hda_codec *codec)
 	if (spec->dsp_state != DSP_DOWNLOADED)
 		return;
 
-	r3di_dsp_scp_startup(codec);
+	r3d_dsp_scp_startup(codec);
 
-	r3di_dsp_initial_mic_setup(codec);
+	r3d_dsp_initial_mic_setup(codec);
 
 	/*remove DSP headroom*/
 	tmp = FLOAT_ZERO;
@@ -6450,7 +6501,8 @@ static void r3di_setup_defaults(struct hda_codec *codec)
 	/* Set speaker source? */
 	dspio_set_uint_param(codec, 0x32, 0x00, tmp);
 
-	r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED);
+	if (spec->quirk == QUIRK_R3DI)
+		r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED);
 
 	/* Setup effect defaults */
 	num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1;
@@ -6462,7 +6514,6 @@ static void r3di_setup_defaults(struct hda_codec *codec)
 					ca0132_effects[idx].def_vals[i]);
 		}
 	}
-
 }
 
 /*
@@ -6727,7 +6778,12 @@ static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
 
 static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
 {
-	ca0132_select_mic(codec);
+	struct ca0132_spec *spec = codec->spec;
+
+	if (spec->use_alt_functions)
+		ca0132_alt_select_in(codec);
+	else
+		ca0132_select_mic(codec);
 }
 
 static void ca0132_init_unsol(struct hda_codec *codec)
@@ -6798,8 +6854,8 @@ static struct hda_verb ca0132_init_verbs0[] = {
 	{}
 };
 
-/* Extra init verbs for SBZ */
-static struct hda_verb sbz_init_verbs[] = {
+/* Extra init verbs for desktop cards. */
+static struct hda_verb ca0132_init_verbs1[] = {
 	{0x15, 0x70D, 0x20},
 	{0x15, 0x70E, 0x19},
 	{0x15, 0x707, 0x00},
@@ -6891,16 +6947,12 @@ static void sbz_region2_exit(struct hda_codec *codec)
 		writeb(0x0, spec->mem_base + 0x100);
 	for (i = 0; i < 8; i++)
 		writeb(0xb3, spec->mem_base + 0x304);
-	/*
-	 * I believe these are GPIO, with the right most hex digit being the
-	 * gpio pin, and the second digit being on or off. We see this more in
-	 * the input/output select functions.
-	 */
-	writew(0x0000, spec->mem_base + 0x320);
-	writew(0x0001, spec->mem_base + 0x320);
-	writew(0x0104, spec->mem_base + 0x320);
-	writew(0x0005, spec->mem_base + 0x320);
-	writew(0x0007, spec->mem_base + 0x320);
+
+	ca0132_mmio_gpio_set(codec, 0, false);
+	ca0132_mmio_gpio_set(codec, 1, false);
+	ca0132_mmio_gpio_set(codec, 4, true);
+	ca0132_mmio_gpio_set(codec, 5, false);
+	ca0132_mmio_gpio_set(codec, 7, false);
 }
 
 static void sbz_set_pin_ctl_default(struct hda_codec *codec)
@@ -6916,7 +6968,7 @@ static void sbz_set_pin_ctl_default(struct hda_codec *codec)
 				AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00);
 }
 
-static void sbz_clear_unsolicited(struct hda_codec *codec)
+static void ca0132_clear_unsolicited(struct hda_codec *codec)
 {
 	hda_nid_t pins[7] = {0x0B, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13};
 	unsigned int i;
@@ -6969,21 +7021,22 @@ static void sbz_exit_chip(struct hda_codec *codec)
 
 	chipio_set_control_param(codec, 0x0D, 0x24);
 
-	sbz_clear_unsolicited(codec);
+	ca0132_clear_unsolicited(codec);
 	sbz_set_pin_ctl_default(codec);
 
 	snd_hda_codec_write(codec, 0x0B, 0,
 		AC_VERB_SET_EAPD_BTLENABLE, 0x00);
 
-	if (dspload_is_loaded(codec))
-		dsp_reset(codec);
-
-	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
-		VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE, 0x00);
-
 	sbz_region2_exit(codec);
 }
 
+static void r3d_exit_chip(struct hda_codec *codec)
+{
+	ca0132_clear_unsolicited(codec);
+	snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
+	snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5b);
+}
+
 static void ca0132_exit_chip(struct hda_codec *codec)
 {
 	/* put any chip cleanup stuffs here. */
@@ -7098,9 +7151,27 @@ static void sbz_pre_dsp_setup(struct hda_codec *codec)
 			AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44);
 }
 
-/*
- * Extra commands that don't really fit anywhere else.
- */
+static void r3d_pre_dsp_setup(struct hda_codec *codec)
+{
+
+	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfc);
+	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfd);
+	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfe);
+	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xff);
+
+	chipio_write(codec, 0x18b0a4, 0x000000c2);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x1E);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0x1C);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_DATA_WRITE, 0x5B);
+
+	snd_hda_codec_write(codec, 0x11, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44);
+}
+
 static void r3di_pre_dsp_setup(struct hda_codec *codec)
 {
 	chipio_write(codec, 0x18b0a4, 0x000000c2);
@@ -7125,13 +7196,12 @@ static void r3di_pre_dsp_setup(struct hda_codec *codec)
 			AC_VERB_SET_PIN_WIDGET_CONTROL, 0x04);
 }
 
-
 /*
  * These are sent before the DSP is downloaded. Not sure
  * what they do, or if they're necessary. Could possibly
  * be removed. Figure they're better to leave in.
  */
-static void sbz_region2_startup(struct hda_codec *codec)
+static void ca0132_mmio_init(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
 
@@ -7171,7 +7241,7 @@ static void ca0132_alt_init(struct hda_codec *codec)
 		ca0132_gpio_init(codec);
 		sbz_pre_dsp_setup(codec);
 		snd_hda_sequence_write(codec, spec->chip_init_verbs);
-		snd_hda_sequence_write(codec, spec->sbz_init_verbs);
+		snd_hda_sequence_write(codec, spec->desktop_init_verbs);
 		break;
 	case QUIRK_R3DI:
 		codec_dbg(codec, "R3DI alt_init");
@@ -7182,6 +7252,11 @@ static void ca0132_alt_init(struct hda_codec *codec)
 		snd_hda_sequence_write(codec, spec->chip_init_verbs);
 		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x6FF, 0xC4);
 		break;
+	case QUIRK_R3D:
+		r3d_pre_dsp_setup(codec);
+		snd_hda_sequence_write(codec, spec->chip_init_verbs);
+		snd_hda_sequence_write(codec, spec->desktop_init_verbs);
+		break;
 	}
 }
 
@@ -7218,8 +7293,8 @@ static int ca0132_init(struct hda_codec *codec)
 		spec->dsp_state = DSP_DOWNLOAD_INIT;
 	spec->curr_chip_addx = INVALID_CHIP_ADDRESS;
 
-	if (spec->quirk == QUIRK_SBZ)
-		sbz_region2_startup(codec);
+	if (spec->use_pci_mmio)
+		ca0132_mmio_init(codec);
 
 	snd_hda_power_up_pm(codec);
 
@@ -7236,14 +7311,13 @@ static int ca0132_init(struct hda_codec *codec)
 
 	ca0132_refresh_widget_caps(codec);
 
-	if (spec->quirk == QUIRK_SBZ)
-		writew(0x0107, spec->mem_base + 0x320);
-
 	switch (spec->quirk) {
 	case QUIRK_R3DI:
-		r3di_setup_defaults(codec);
+	case QUIRK_R3D:
+		r3d_setup_defaults(codec);
 		break;
 	case QUIRK_SBZ:
+		sbz_setup_defaults(codec);
 		break;
 	default:
 		ca0132_setup_defaults(codec);
@@ -7274,20 +7348,12 @@ static int ca0132_init(struct hda_codec *codec)
 		ca0132_gpio_setup(codec);
 
 	snd_hda_sequence_write(codec, spec->spec_init_verbs);
-	switch (spec->quirk) {
-	case QUIRK_SBZ:
-		sbz_setup_defaults(codec);
-		ca0132_alt_select_out(codec);
-		ca0132_alt_select_in(codec);
-		break;
-	case QUIRK_R3DI:
+	if (spec->use_alt_functions) {
 		ca0132_alt_select_out(codec);
 		ca0132_alt_select_in(codec);
-		break;
-	default:
+	} else {
 		ca0132_select_out(codec);
 		ca0132_select_mic(codec);
-		break;
 	}
 
 	snd_hda_jack_report_sync(codec);
@@ -7316,16 +7382,17 @@ static void ca0132_free(struct hda_codec *codec)
 	case QUIRK_SBZ:
 		sbz_exit_chip(codec);
 		break;
+	case QUIRK_R3D:
+		r3d_exit_chip(codec);
+		break;
 	case QUIRK_R3DI:
 		r3di_gpio_shutdown(codec);
-		snd_hda_sequence_write(codec, spec->base_exit_verbs);
-		ca0132_exit_chip(codec);
-		break;
-	default:
-		snd_hda_sequence_write(codec, spec->base_exit_verbs);
-		ca0132_exit_chip(codec);
 		break;
 	}
+
+	snd_hda_sequence_write(codec, spec->base_exit_verbs);
+	ca0132_exit_chip(codec);
+
 	snd_hda_power_down(codec);
 	if (spec->mem_base)
 		iounmap(spec->mem_base);
@@ -7386,8 +7453,15 @@ static void ca0132_config(struct hda_codec *codec)
 		spec->unsol_tag_amic1 = 0x11;
 		break;
 	case QUIRK_SBZ:
-		codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__);
-		snd_hda_apply_pincfgs(codec, sbz_pincfgs);
+	case QUIRK_R3D:
+		if (spec->quirk == QUIRK_SBZ) {
+			codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__);
+			snd_hda_apply_pincfgs(codec, sbz_pincfgs);
+		}
+		if (spec->quirk == QUIRK_R3D) {
+			codec_dbg(codec, "%s: QUIRK_R3D applied.\n", __func__);
+			snd_hda_apply_pincfgs(codec, r3d_pincfgs);
+		}
 
 		spec->num_outputs = 2;
 		spec->out_pins[0] = 0x0B; /* Line out */
@@ -7473,8 +7547,8 @@ static int ca0132_prepare_verbs(struct hda_codec *codec)
 	struct ca0132_spec *spec = codec->spec;
 
 	spec->chip_init_verbs = ca0132_init_verbs0;
-	if (spec->quirk == QUIRK_SBZ)
-		spec->sbz_init_verbs = sbz_init_verbs;
+	if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3D)
+		spec->desktop_init_verbs = ca0132_init_verbs1;
 	spec->spec_init_verbs = kcalloc(NUM_SPEC_VERBS,
 					sizeof(struct hda_verb),
 					GFP_KERNEL);
@@ -7530,25 +7604,19 @@ static int patch_ca0132(struct hda_codec *codec)
 	else
 		spec->quirk = QUIRK_NONE;
 
-	/* Setup BAR Region 2 for Sound Blaster Z */
-	if (spec->quirk == QUIRK_SBZ) {
-		spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20);
-		if (spec->mem_base == NULL) {
-			codec_warn(codec, "pci_iomap failed!");
-			codec_info(codec, "perhaps this is not an SBZ?");
-			spec->quirk = QUIRK_NONE;
-		}
-	}
-
 	spec->dsp_state = DSP_DOWNLOAD_INIT;
 	spec->num_mixers = 1;
 
 	/* Set which mixers each quirk uses. */
 	switch (spec->quirk) {
 	case QUIRK_SBZ:
-		spec->mixers[0] = sbz_mixer;
+		spec->mixers[0] = desktop_mixer;
 		snd_hda_codec_set_name(codec, "Sound Blaster Z");
 		break;
+	case QUIRK_R3D:
+		spec->mixers[0] = desktop_mixer;
+		snd_hda_codec_set_name(codec, "Recon3D");
+		break;
 	case QUIRK_R3DI:
 		spec->mixers[0] = r3di_mixer;
 		snd_hda_codec_set_name(codec, "Recon3Di");
@@ -7558,19 +7626,34 @@ static int patch_ca0132(struct hda_codec *codec)
 		break;
 	}
 
-	/* Setup whether or not to use alt functions/controls */
+	/* Setup whether or not to use alt functions/controls/pci_mmio */
 	switch (spec->quirk) {
 	case QUIRK_SBZ:
+	case QUIRK_R3D:
+		spec->use_alt_controls = true;
+		spec->use_alt_functions = true;
+		spec->use_pci_mmio = true;
+		break;
 	case QUIRK_R3DI:
 		spec->use_alt_controls = true;
 		spec->use_alt_functions = true;
+		spec->use_pci_mmio = false;
 		break;
 	default:
 		spec->use_alt_controls = false;
 		spec->use_alt_functions = false;
+		spec->use_pci_mmio = false;
 		break;
 	}
 
+	if (spec->use_pci_mmio) {
+		spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20);
+		if (spec->mem_base == NULL) {
+			codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE.");
+			spec->quirk = QUIRK_NONE;
+		}
+	}
+
 	spec->base_init_verbs = ca0132_base_init_verbs;
 	spec->base_exit_verbs = ca0132_base_exit_verbs;
 
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index d6e079f4ec09..a7f91be45194 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -1096,25 +1096,6 @@ static int cs421x_init(struct hda_codec *codec)
 	return 0;
 }
 
-static int cs421x_build_controls(struct hda_codec *codec)
-{
-	struct cs_spec *spec = codec->spec;
-	int err;
-
-	err = snd_hda_gen_build_controls(codec);
-	if (err < 0)
-		return err;
-
-	if (spec->gen.autocfg.speaker_outs &&
-	    spec->vendor_nid == CS4210_VENDOR_NID) {
-		err = snd_hda_ctl_add(codec, 0,
-			snd_ctl_new1(&cs421x_speaker_boost_ctl, codec));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 static void fix_volume_caps(struct hda_codec *codec, hda_nid_t dac)
 {
 	unsigned int caps;
@@ -1144,6 +1125,14 @@ static int cs421x_parse_auto_config(struct hda_codec *codec)
 		return err;
 
 	parse_cs421x_digital(codec);
+
+	if (spec->gen.autocfg.speaker_outs &&
+	    spec->vendor_nid == CS4210_VENDOR_NID) {
+		if (!snd_hda_gen_add_kctl(&spec->gen, NULL,
+					  &cs421x_speaker_boost_ctl))
+			return -ENOMEM;
+	}
+
 	return 0;
 }
 
@@ -1175,7 +1164,7 @@ static int cs421x_suspend(struct hda_codec *codec)
 #endif
 
 static const struct hda_codec_ops cs421x_patch_ops = {
-	.build_controls = cs421x_build_controls,
+	.build_controls = snd_hda_gen_build_controls,
 	.build_pcms = snd_hda_gen_build_pcms,
 	.init = cs421x_init,
 	.free = cs_free,
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index f641c20095f7..cfd4e4f97f8f 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -37,8 +37,6 @@
 struct conexant_spec {
 	struct hda_gen_spec gen;
 
-	unsigned int beep_amp;
-
 	/* extra EAPD pins */
 	unsigned int num_eapds;
 	hda_nid_t eapds[4];
@@ -62,65 +60,48 @@ struct conexant_spec {
 
 
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
-static inline void set_beep_amp(struct conexant_spec *spec, hda_nid_t nid,
-				int idx, int dir)
-{
-	spec->gen.beep_nid = nid;
-	spec->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir);
-}
-/* additional beep mixers; the actual parameters are overwritten at build */
+/* additional beep mixers; private_value will be overwritten */
 static const struct snd_kcontrol_new cxt_beep_mixer[] = {
 	HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0, 1, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_BEEP_MONO("Beep Playback Switch", 0, 1, 0, HDA_OUTPUT),
-	{ } /* end */
 };
 
-/* create beep controls if needed */
-static int add_beep_ctls(struct hda_codec *codec)
+static int set_beep_amp(struct conexant_spec *spec, hda_nid_t nid,
+			int idx, int dir)
 {
-	struct conexant_spec *spec = codec->spec;
-	int err;
+	struct snd_kcontrol_new *knew;
+	unsigned int beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir);
+	int i;
 
-	if (spec->beep_amp) {
-		const struct snd_kcontrol_new *knew;
-		for (knew = cxt_beep_mixer; knew->name; knew++) {
-			struct snd_kcontrol *kctl;
-			kctl = snd_ctl_new1(knew, codec);
-			if (!kctl)
-				return -ENOMEM;
-			kctl->private_value = spec->beep_amp;
-			err = snd_hda_ctl_add(codec, 0, kctl);
-			if (err < 0)
-				return err;
-		}
+	spec->gen.beep_nid = nid;
+	for (i = 0; i < ARRAY_SIZE(cxt_beep_mixer); i++) {
+		knew = snd_hda_gen_add_kctl(&spec->gen, NULL,
+					    &cxt_beep_mixer[i]);
+		if (!knew)
+			return -ENOMEM;
+		knew->private_value = beep_amp;
 	}
 	return 0;
 }
-#else
-#define set_beep_amp(spec, nid, idx, dir) /* NOP */
-#define add_beep_ctls(codec)	0
-#endif
-
-/*
- * Automatic parser for CX20641 & co
- */
 
-#ifdef CONFIG_SND_HDA_INPUT_BEEP
-static void cx_auto_parse_beep(struct hda_codec *codec)
+static int cx_auto_parse_beep(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
 	hda_nid_t nid;
 
 	for_each_hda_codec_node(nid, codec)
-		if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP) {
-			set_beep_amp(spec, nid, 0, HDA_OUTPUT);
-			break;
-		}
+		if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP)
+			return set_beep_amp(spec, nid, 0, HDA_OUTPUT);
+	return 0;
 }
 #else
-#define cx_auto_parse_beep(codec)
+#define cx_auto_parse_beep(codec)	0
 #endif
 
+/*
+ * Automatic parser for CX20641 & co
+ */
+
 /* parse EAPDs */
 static void cx_auto_parse_eapd(struct hda_codec *codec)
 {
@@ -179,21 +160,6 @@ static void cx_auto_vmaster_hook_mute_led(void *private_data, int enabled)
 			    enabled ? 0x00 : 0x02);
 }
 
-static int cx_auto_build_controls(struct hda_codec *codec)
-{
-	int err;
-
-	err = snd_hda_gen_build_controls(codec);
-	if (err < 0)
-		return err;
-
-	err = add_beep_ctls(codec);
-	if (err < 0)
-		return err;
-
-	return 0;
-}
-
 static int cx_auto_init(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
@@ -211,6 +177,7 @@ static void cx_auto_reboot_notify(struct hda_codec *codec)
 	struct conexant_spec *spec = codec->spec;
 
 	switch (codec->core.vendor_id) {
+	case 0x14f12008: /* CX8200 */
 	case 0x14f150f2: /* CX20722 */
 	case 0x14f150f4: /* CX20724 */
 		break;
@@ -218,13 +185,14 @@ static void cx_auto_reboot_notify(struct hda_codec *codec)
 		return;
 	}
 
-	/* Turn the CX20722 codec into D3 to avoid spurious noises
+	/* Turn the problematic codec into D3 to avoid spurious noises
 	   from the internal speaker during (and after) reboot */
 	cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false);
 
 	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
 	snd_hda_codec_write(codec, codec->core.afg, 0,
 			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+	msleep(10);
 }
 
 static void cx_auto_free(struct hda_codec *codec)
@@ -234,7 +202,7 @@ static void cx_auto_free(struct hda_codec *codec)
 }
 
 static const struct hda_codec_ops cx_auto_patch_ops = {
-	.build_controls = cx_auto_build_controls,
+	.build_controls = snd_hda_gen_build_controls,
 	.build_pcms = snd_hda_gen_build_pcms,
 	.init = cx_auto_init,
 	.reboot_notify = cx_auto_reboot_notify,
@@ -343,6 +311,7 @@ static void cxt_fixup_headphone_mic(struct hda_codec *codec,
 		snd_hdac_regmap_add_vendor_verb(&codec->core, 0x410);
 		break;
 	case HDA_FIXUP_ACT_PROBE:
+		WARN_ON(spec->gen.cap_sync_hook);
 		spec->gen.cap_sync_hook = cxt_update_headset_mode_hook;
 		spec->gen.automute_hook = cxt_update_headset_mode;
 		break;
@@ -374,7 +343,7 @@ static void cxt_fixup_headset_mic(struct hda_codec *codec,
  * control. */
 
 #define update_mic_pin(codec, nid, val)					\
-	snd_hda_codec_update_cache(codec, nid, 0,			\
+	snd_hda_codec_write_cache(codec, nid, 0,			\
 				   AC_VERB_SET_PIN_WIDGET_CONTROL, val)
 
 static const struct hda_input_mux olpc_xo_dc_bias = {
@@ -695,16 +664,12 @@ static void cxt_fixup_gpio_mute_hook(void *private_data, int enabled)
 }
 
 /* turn on/off mic-mute LED via GPIO per capture hook */
-static void cxt_fixup_gpio_mic_mute_hook(struct hda_codec *codec,
-					 struct snd_kcontrol *kcontrol,
-					 struct snd_ctl_elem_value *ucontrol)
+static void cxt_gpio_micmute_update(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
 
-	if (ucontrol)
-		cxt_update_gpio_led(codec, spec->gpio_mic_led_mask,
-				    ucontrol->value.integer.value[0] ||
-				    ucontrol->value.integer.value[1]);
+	cxt_update_gpio_led(codec, spec->gpio_mic_led_mask,
+			    spec->gen.micmute_led.led_value);
 }
 
 
@@ -721,11 +686,11 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec,
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook;
-		spec->gen.cap_sync_hook = cxt_fixup_gpio_mic_mute_hook;
 		spec->gpio_led = 0;
 		spec->mute_led_polarity = 0;
 		spec->gpio_mute_led_mask = 0x01;
 		spec->gpio_mic_led_mask = 0x02;
+		snd_hda_gen_add_micmute_led(codec, cxt_gpio_micmute_update);
 	}
 	snd_hda_add_verbs(codec, gpio_init);
 	if (spec->gpio_led)
@@ -1037,7 +1002,6 @@ static int patch_conexant_auto(struct hda_codec *codec)
 	codec->spec = spec;
 	codec->patch_ops = cx_auto_patch_ops;
 
-	cx_auto_parse_beep(codec);
 	cx_auto_parse_eapd(codec);
 	spec->gen.own_eapd_ctl = 1;
 	if (spec->dynamic_eapd)
@@ -1097,6 +1061,10 @@ static int patch_conexant_auto(struct hda_codec *codec)
 	if (err < 0)
 		goto error;
 
+	err = cx_auto_parse_beep(codec);
+	if (err < 0)
+		goto error;
+
 	/* Some laptops with Conexant chips show stalls in S3 resume,
 	 * which falls into the single-cmd mode.
 	 * Better to make reset, then.
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 8a49415aebac..cb587dce67a9 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -177,13 +177,13 @@ struct hdmi_spec {
 
 	/* i915/powerwell (Haswell+/Valleyview+) specific */
 	bool use_acomp_notifier; /* use i915 eld_notify callback for hotplug */
-	struct i915_audio_component_audio_ops i915_audio_ops;
+	struct drm_audio_component_audio_ops drm_audio_ops;
 
 	struct hdac_chmap chmap;
 	hda_nid_t vendor_nid;
 };
 
-#ifdef CONFIG_SND_HDA_I915
+#ifdef CONFIG_SND_HDA_COMPONENT
 static inline bool codec_has_acomp(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec = codec->spec;
@@ -339,13 +339,13 @@ static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol,
 	if (!per_pin) {
 		/* no pin is bound to the pcm */
 		uinfo->count = 0;
-		mutex_unlock(&spec->pcm_lock);
-		return 0;
+		goto unlock;
 	}
 	eld = &per_pin->sink_eld;
 	uinfo->count = eld->eld_valid ? eld->eld_size : 0;
-	mutex_unlock(&spec->pcm_lock);
 
+ unlock:
+	mutex_unlock(&spec->pcm_lock);
 	return 0;
 }
 
@@ -357,6 +357,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol,
 	struct hdmi_spec_per_pin *per_pin;
 	struct hdmi_eld *eld;
 	int pcm_idx;
+	int err = 0;
 
 	pcm_idx = kcontrol->private_value;
 	mutex_lock(&spec->pcm_lock);
@@ -365,16 +366,15 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol,
 		/* no pin is bound to the pcm */
 		memset(ucontrol->value.bytes.data, 0,
 		       ARRAY_SIZE(ucontrol->value.bytes.data));
-		mutex_unlock(&spec->pcm_lock);
-		return 0;
+		goto unlock;
 	}
-	eld = &per_pin->sink_eld;
 
+	eld = &per_pin->sink_eld;
 	if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data) ||
 	    eld->eld_size > ELD_MAX_SIZE) {
-		mutex_unlock(&spec->pcm_lock);
 		snd_BUG();
-		return -EINVAL;
+		err = -EINVAL;
+		goto unlock;
 	}
 
 	memset(ucontrol->value.bytes.data, 0,
@@ -382,9 +382,10 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol,
 	if (eld->eld_valid)
 		memcpy(ucontrol->value.bytes.data, eld->eld_buffer,
 		       eld->eld_size);
-	mutex_unlock(&spec->pcm_lock);
 
-	return 0;
+ unlock:
+	mutex_unlock(&spec->pcm_lock);
+	return err;
 }
 
 static const struct snd_kcontrol_new eld_bytes_ctl = {
@@ -1209,8 +1210,8 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 	pin_idx = hinfo_to_pin_index(codec, hinfo);
 	if (!spec->dyn_pcm_assign) {
 		if (snd_BUG_ON(pin_idx < 0)) {
-			mutex_unlock(&spec->pcm_lock);
-			return -EINVAL;
+			err = -EINVAL;
+			goto unlock;
 		}
 	} else {
 		/* no pin is assigned to the PCM
@@ -1218,16 +1219,13 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 		 */
 		if (pin_idx < 0) {
 			err = hdmi_pcm_open_no_pin(hinfo, codec, substream);
-			mutex_unlock(&spec->pcm_lock);
-			return err;
+			goto unlock;
 		}
 	}
 
 	err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx);
-	if (err < 0) {
-		mutex_unlock(&spec->pcm_lock);
-		return err;
-	}
+	if (err < 0)
+		goto unlock;
 
 	per_cvt = get_cvt(spec, cvt_idx);
 	/* Claim converter */
@@ -1264,12 +1262,11 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 			per_cvt->assigned = 0;
 			hinfo->nid = 0;
 			snd_hda_spdif_ctls_unassign(codec, pcm_idx);
-			mutex_unlock(&spec->pcm_lock);
-			return -ENODEV;
+			err = -ENODEV;
+			goto unlock;
 		}
 	}
 
-	mutex_unlock(&spec->pcm_lock);
 	/* Store the updated parameters */
 	runtime->hw.channels_min = hinfo->channels_min;
 	runtime->hw.channels_max = hinfo->channels_max;
@@ -1278,7 +1275,9 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 
 	snd_pcm_hw_constraint_step(substream->runtime, 0,
 				   SNDRV_PCM_HW_PARAM_CHANNELS, 2);
-	return 0;
+ unlock:
+	mutex_unlock(&spec->pcm_lock);
+	return err;
 }
 
 /*
@@ -1867,7 +1866,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	bool non_pcm;
 	int pinctl;
-	int err;
+	int err = 0;
 
 	mutex_lock(&spec->pcm_lock);
 	pin_idx = hinfo_to_pin_index(codec, hinfo);
@@ -1879,13 +1878,12 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 		pin_cvt_fixup(codec, NULL, cvt_nid);
 		snd_hda_codec_setup_stream(codec, cvt_nid,
 					stream_tag, 0, format);
-		mutex_unlock(&spec->pcm_lock);
-		return 0;
+		goto unlock;
 	}
 
 	if (snd_BUG_ON(pin_idx < 0)) {
-		mutex_unlock(&spec->pcm_lock);
-		return -EINVAL;
+		err = -EINVAL;
+		goto unlock;
 	}
 	per_pin = get_pin(spec, pin_idx);
 	pin_nid = per_pin->pin_nid;
@@ -1924,6 +1922,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 	/* snd_hda_set_dev_select() has been called before */
 	err = spec->ops.setup_stream(codec, cvt_nid, pin_nid,
 				 stream_tag, format);
+ unlock:
 	mutex_unlock(&spec->pcm_lock);
 	return err;
 }
@@ -1945,6 +1944,7 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
 	struct hdmi_spec_per_cvt *per_cvt;
 	struct hdmi_spec_per_pin *per_pin;
 	int pinctl;
+	int err = 0;
 
 	if (hinfo->nid) {
 		pcm_idx = hinfo_to_pcm_index(codec, hinfo);
@@ -1963,14 +1963,12 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
 		snd_hda_spdif_ctls_unassign(codec, pcm_idx);
 		clear_bit(pcm_idx, &spec->pcm_in_use);
 		pin_idx = hinfo_to_pin_index(codec, hinfo);
-		if (spec->dyn_pcm_assign && pin_idx < 0) {
-			mutex_unlock(&spec->pcm_lock);
-			return 0;
-		}
+		if (spec->dyn_pcm_assign && pin_idx < 0)
+			goto unlock;
 
 		if (snd_BUG_ON(pin_idx < 0)) {
-			mutex_unlock(&spec->pcm_lock);
-			return -EINVAL;
+			err = -EINVAL;
+			goto unlock;
 		}
 		per_pin = get_pin(spec, pin_idx);
 
@@ -1989,10 +1987,11 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
 		per_pin->setup = false;
 		per_pin->channels = 0;
 		mutex_unlock(&per_pin->lock);
+	unlock:
 		mutex_unlock(&spec->pcm_lock);
 	}
 
-	return 0;
+	return err;
 }
 
 static const struct hda_pcm_ops generic_ops = {
@@ -2288,7 +2287,7 @@ static void generic_hdmi_free(struct hda_codec *codec)
 	int pin_idx, pcm_idx;
 
 	if (codec_has_acomp(codec))
-		snd_hdac_i915_register_notifier(NULL);
+		snd_hdac_acomp_register_notifier(&codec->bus->core, NULL);
 
 	for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
 		struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
@@ -2471,6 +2470,38 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 	snd_hda_codec_set_power_to_all(codec, fg, power_state);
 }
 
+/* There is a fixed mapping between audio pin node and display port.
+ * on SNB, IVY, HSW, BSW, SKL, BXT, KBL:
+ * Pin Widget 5 - PORT B (port = 1 in i915 driver)
+ * Pin Widget 6 - PORT C (port = 2 in i915 driver)
+ * Pin Widget 7 - PORT D (port = 3 in i915 driver)
+ *
+ * on VLV, ILK:
+ * Pin Widget 4 - PORT B (port = 1 in i915 driver)
+ * Pin Widget 5 - PORT C (port = 2 in i915 driver)
+ * Pin Widget 6 - PORT D (port = 3 in i915 driver)
+ */
+static int intel_base_nid(struct hda_codec *codec)
+{
+	switch (codec->core.vendor_id) {
+	case 0x80860054: /* ILK */
+	case 0x80862804: /* ILK */
+	case 0x80862882: /* VLV */
+		return 4;
+	default:
+		return 5;
+	}
+}
+
+static int intel_pin2port(void *audio_ptr, int pin_nid)
+{
+	int base_nid = intel_base_nid(audio_ptr);
+
+	if (WARN_ON(pin_nid < base_nid || pin_nid >= base_nid + 3))
+		return -1;
+	return pin_nid - base_nid + 1; /* intel port is 1-based */
+}
+
 static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe)
 {
 	struct hda_codec *codec = audio_ptr;
@@ -2481,16 +2512,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe)
 	if (port < 1 || port > 3)
 		return;
 
-	switch (codec->core.vendor_id) {
-	case 0x80860054: /* ILK */
-	case 0x80862804: /* ILK */
-	case 0x80862882: /* VLV */
-		pin_nid = port + 0x03;
-		break;
-	default:
-		pin_nid = port + 0x04;
-		break;
-	}
+	pin_nid = port + intel_base_nid(codec) - 1; /* intel port is 1-based */
 
 	/* skip notification during system suspend (but not in runtime PM);
 	 * the state will be updated at resume
@@ -2498,7 +2520,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe)
 	if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0)
 		return;
 	/* ditto during suspend/resume process itself */
-	if (atomic_read(&(codec)->core.in_pm))
+	if (snd_hdac_is_in_pm(&codec->core))
 		return;
 
 	snd_hdac_i915_set_bclk(&codec->bus->core);
@@ -2511,14 +2533,16 @@ static void register_i915_notifier(struct hda_codec *codec)
 	struct hdmi_spec *spec = codec->spec;
 
 	spec->use_acomp_notifier = true;
-	spec->i915_audio_ops.audio_ptr = codec;
+	spec->drm_audio_ops.audio_ptr = codec;
 	/* intel_audio_codec_enable() or intel_audio_codec_disable()
 	 * will call pin_eld_notify with using audio_ptr pointer
 	 * We need make sure audio_ptr is really setup
 	 */
 	wmb();
-	spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify;
-	snd_hdac_i915_register_notifier(&spec->i915_audio_ops);
+	spec->drm_audio_ops.pin2port = intel_pin2port;
+	spec->drm_audio_ops.pin_eld_notify = intel_pin_eld_notify;
+	snd_hdac_acomp_register_notifier(&codec->bus->core,
+					&spec->drm_audio_ops);
 }
 
 /* setup_stream ops override for HSW+ */
@@ -2551,6 +2575,8 @@ static int alloc_intel_hdmi(struct hda_codec *codec)
 	/* requires i915 binding */
 	if (!codec->bus->core.audio_component) {
 		codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n");
+		/* set probe_id here to prevent generic fallback binding */
+		codec->probe_id = HDA_CODEC_ID_SKIP_PROBE;
 		return -ENODEV;
 	}
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f6af3e1c2b93..b20974ef1e13 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -43,11 +43,9 @@
 
 /* extra amp-initialization sequence types */
 enum {
+	ALC_INIT_UNDEFINED,
 	ALC_INIT_NONE,
 	ALC_INIT_DEFAULT,
-	ALC_INIT_GPIO1,
-	ALC_INIT_GPIO2,
-	ALC_INIT_GPIO3,
 };
 
 enum {
@@ -85,19 +83,20 @@ struct alc_spec {
 	struct hda_gen_spec gen; /* must be at head */
 
 	/* codec parameterization */
-	const struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
-	unsigned int num_mixers;
-	unsigned int beep_amp;	/* beep amp value, set via set_beep_amp() */
-
 	struct alc_customize_define cdefine;
 	unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */
 
+	/* GPIO bits */
+	unsigned int gpio_mask;
+	unsigned int gpio_dir;
+	unsigned int gpio_data;
+	bool gpio_write_delay;	/* add a delay before writing gpio_data */
+
 	/* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */
 	int mute_led_polarity;
 	hda_nid_t mute_led_nid;
 	hda_nid_t cap_mute_led_nid;
 
-	unsigned int gpio_led; /* used for alc269_fixup_hp_gpio_led() */
 	unsigned int gpio_mute_led_mask;
 	unsigned int gpio_mic_led_mask;
 
@@ -205,41 +204,87 @@ static void alc_process_coef_fw(struct hda_codec *codec,
 }
 
 /*
- * Append the given mixer and verb elements for the later use
- * The mixer array is referred in build_controls(), and init_verbs are
- * called in init().
+ * GPIO setup tables, used in initialization
  */
-static void add_mixer(struct alc_spec *spec, const struct snd_kcontrol_new *mix)
+
+/* Enable GPIO mask and set output */
+static void alc_setup_gpio(struct hda_codec *codec, unsigned int mask)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->gpio_mask |= mask;
+	spec->gpio_dir |= mask;
+	spec->gpio_data |= mask;
+}
+
+static void alc_write_gpio_data(struct hda_codec *codec)
 {
-	if (snd_BUG_ON(spec->num_mixers >= ARRAY_SIZE(spec->mixers)))
+	struct alc_spec *spec = codec->spec;
+
+	snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+			    spec->gpio_data);
+}
+
+static void alc_update_gpio_data(struct hda_codec *codec, unsigned int mask,
+				 bool on)
+{
+	struct alc_spec *spec = codec->spec;
+	unsigned int oldval = spec->gpio_data;
+
+	if (on)
+		spec->gpio_data |= mask;
+	else
+		spec->gpio_data &= ~mask;
+	if (oldval != spec->gpio_data)
+		alc_write_gpio_data(codec);
+}
+
+static void alc_write_gpio(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (!spec->gpio_mask)
 		return;
-	spec->mixers[spec->num_mixers++] = mix;
+
+	snd_hda_codec_write(codec, codec->core.afg, 0,
+			    AC_VERB_SET_GPIO_MASK, spec->gpio_mask);
+	snd_hda_codec_write(codec, codec->core.afg, 0,
+			    AC_VERB_SET_GPIO_DIRECTION, spec->gpio_dir);
+	if (spec->gpio_write_delay)
+		msleep(1);
+	alc_write_gpio_data(codec);
 }
 
-/*
- * GPIO setup tables, used in initialization
- */
-/* Enable GPIO mask and set output */
-static const struct hda_verb alc_gpio1_init_verbs[] = {
-	{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
-	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
-	{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
-	{ }
-};
+static void alc_fixup_gpio(struct hda_codec *codec, int action,
+			   unsigned int mask)
+{
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		alc_setup_gpio(codec, mask);
+}
 
-static const struct hda_verb alc_gpio2_init_verbs[] = {
-	{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DATA, 0x02},
-	{ }
-};
+static void alc_fixup_gpio1(struct hda_codec *codec,
+			    const struct hda_fixup *fix, int action)
+{
+	alc_fixup_gpio(codec, action, 0x01);
+}
 
-static const struct hda_verb alc_gpio3_init_verbs[] = {
-	{0x01, AC_VERB_SET_GPIO_MASK, 0x03},
-	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03},
-	{0x01, AC_VERB_SET_GPIO_DATA, 0x03},
-	{ }
-};
+static void alc_fixup_gpio2(struct hda_codec *codec,
+			    const struct hda_fixup *fix, int action)
+{
+	alc_fixup_gpio(codec, action, 0x02);
+}
+
+static void alc_fixup_gpio3(struct hda_codec *codec,
+			    const struct hda_fixup *fix, int action)
+{
+	alc_fixup_gpio(codec, action, 0x03);
+}
+
+static void alc_fixup_gpio4(struct hda_codec *codec,
+			    const struct hda_fixup *fix, int action)
+{
+	alc_fixup_gpio(codec, action, 0x04);
+}
 
 /*
  * Fix hardware PLL issue
@@ -447,16 +492,8 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
 {
 	alc_fill_eapd_coef(codec);
 	alc_auto_setup_eapd(codec, true);
+	alc_write_gpio(codec);
 	switch (type) {
-	case ALC_INIT_GPIO1:
-		snd_hda_sequence_write(codec, alc_gpio1_init_verbs);
-		break;
-	case ALC_INIT_GPIO2:
-		snd_hda_sequence_write(codec, alc_gpio2_init_verbs);
-		break;
-	case ALC_INIT_GPIO3:
-		snd_hda_sequence_write(codec, alc_gpio3_init_verbs);
-		break;
 	case ALC_INIT_DEFAULT:
 		switch (codec->core.vendor_id) {
 		case 0x10ec0260:
@@ -656,20 +693,22 @@ do_sku:
 	 * 7~6 : Reserved
 	*/
 	tmp = (ass & 0x38) >> 3;	/* external Amp control */
-	switch (tmp) {
-	case 1:
-		spec->init_amp = ALC_INIT_GPIO1;
-		break;
-	case 3:
-		spec->init_amp = ALC_INIT_GPIO2;
-		break;
-	case 7:
-		spec->init_amp = ALC_INIT_GPIO3;
-		break;
-	case 5:
-	default:
-		spec->init_amp = ALC_INIT_DEFAULT;
-		break;
+	if (spec->init_amp == ALC_INIT_UNDEFINED) {
+		switch (tmp) {
+		case 1:
+			alc_setup_gpio(codec, 0x01);
+			break;
+		case 3:
+			alc_setup_gpio(codec, 0x02);
+			break;
+		case 7:
+			alc_setup_gpio(codec, 0x03);
+			break;
+		case 5:
+		default:
+			spec->init_amp = ALC_INIT_DEFAULT;
+			break;
+		}
 	}
 
 	/* is laptop or Desktop and enable the function "Mute internal speaker
@@ -722,47 +761,14 @@ static void alc_fixup_inv_dmic(struct hda_codec *codec,
 }
 
 
-#ifdef CONFIG_SND_HDA_INPUT_BEEP
-/* additional beep mixers; the actual parameters are overwritten at build */
-static const struct snd_kcontrol_new alc_beep_mixer[] = {
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT),
-	HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_INPUT),
-	{ } /* end */
-};
-#endif
-
 static int alc_build_controls(struct hda_codec *codec)
 {
-	struct alc_spec *spec = codec->spec;
-	int i, err;
+	int err;
 
 	err = snd_hda_gen_build_controls(codec);
 	if (err < 0)
 		return err;
 
-	for (i = 0; i < spec->num_mixers; i++) {
-		err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
-		if (err < 0)
-			return err;
-	}
-
-#ifdef CONFIG_SND_HDA_INPUT_BEEP
-	/* create beep controls if needed */
-	if (spec->beep_amp) {
-		const struct snd_kcontrol_new *knew;
-		for (knew = alc_beep_mixer; knew->name; knew++) {
-			struct snd_kcontrol *kctl;
-			kctl = snd_ctl_new1(knew, codec);
-			if (!kctl)
-				return -ENOMEM;
-			kctl->private_value = spec->beep_amp;
-			err = snd_hda_ctl_add(codec, 0, kctl);
-			if (err < 0)
-				return err;
-		}
-	}
-#endif
-
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_BUILD);
 	return 0;
 }
@@ -973,8 +979,30 @@ static int alc_codec_rename_from_preset(struct hda_codec *codec)
  * Digital-beep handlers
  */
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
-#define set_beep_amp(spec, nid, idx, dir) \
-	((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir))
+
+/* additional beep mixers; private_value will be overwritten */
+static const struct snd_kcontrol_new alc_beep_mixer[] = {
+	HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT),
+	HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_INPUT),
+};
+
+/* set up and create beep controls */
+static int set_beep_amp(struct alc_spec *spec, hda_nid_t nid,
+			int idx, int dir)
+{
+	struct snd_kcontrol_new *knew;
+	unsigned int beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir);
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(alc_beep_mixer); i++) {
+		knew = snd_hda_gen_add_kctl(&spec->gen, NULL,
+					    &alc_beep_mixer[i]);
+		if (!knew)
+			return -ENOMEM;
+		knew->private_value = beep_amp;
+	}
+	return 0;
+}
 
 static const struct snd_pci_quirk beep_white_list[] = {
 	SND_PCI_QUIRK(0x1043, 0x103c, "ASUS", 1),
@@ -999,7 +1027,7 @@ static inline int has_cdefine_beep(struct hda_codec *codec)
 	return spec->cdefine.enable_pcbeep;
 }
 #else
-#define set_beep_amp(spec, nid, idx, dir) /* NOP */
+#define set_beep_amp(spec, nid, idx, dir)	0
 #define has_cdefine_beep(codec)		0
 #endif
 
@@ -1104,12 +1132,12 @@ static void alc880_fixup_vol_knob(struct hda_codec *codec,
 
 static const struct hda_fixup alc880_fixups[] = {
 	[ALC880_FIXUP_GPIO1] = {
-		.type = HDA_FIXUP_VERBS,
-		.v.verbs = alc_gpio1_init_verbs,
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_gpio1,
 	},
 	[ALC880_FIXUP_GPIO2] = {
-		.type = HDA_FIXUP_VERBS,
-		.v.verbs = alc_gpio2_init_verbs,
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_gpio2,
 	},
 	[ALC880_FIXUP_MEDION_RIM] = {
 		.type = HDA_FIXUP_VERBS,
@@ -1501,8 +1529,11 @@ static int patch_alc880(struct hda_codec *codec)
 	if (err < 0)
 		goto error;
 
-	if (!spec->gen.no_analog)
-		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+	if (!spec->gen.no_analog) {
+		err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+		if (err < 0)
+			goto error;
+	}
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
 
@@ -1544,8 +1575,8 @@ enum {
 static void alc260_gpio1_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
-			    spec->gen.hp_jack_present);
+
+	alc_update_gpio_data(codec, 0x01, spec->gen.hp_jack_present);
 }
 
 static void alc260_fixup_gpio1_toggle(struct hda_codec *codec,
@@ -1562,7 +1593,7 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec,
 		spec->gen.autocfg.hp_pins[0] = 0x0f; /* copy it for automute */
 		snd_hda_jack_detect_enable_callback(codec, 0x0f,
 						    snd_hda_gen_hp_automute);
-		snd_hda_add_verbs(codec, alc_gpio1_init_verbs);
+		alc_setup_gpio(codec, 0x01);
 	}
 }
 
@@ -1589,8 +1620,6 @@ static void alc260_fixup_kn1(struct hda_codec *codec,
 	switch (action) {
 	case HDA_FIXUP_ACT_PRE_PROBE:
 		snd_hda_apply_pincfgs(codec, pincfgs);
-		break;
-	case HDA_FIXUP_ACT_PROBE:
 		spec->init_amp = ALC_INIT_NONE;
 		break;
 	}
@@ -1600,7 +1629,7 @@ static void alc260_fixup_fsc_s7020(struct hda_codec *codec,
 				   const struct hda_fixup *fix, int action)
 {
 	struct alc_spec *spec = codec->spec;
-	if (action == HDA_FIXUP_ACT_PROBE)
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
 		spec->init_amp = ALC_INIT_NONE;
 }
 
@@ -1638,8 +1667,8 @@ static const struct hda_fixup alc260_fixups[] = {
 		},
 	},
 	[ALC260_FIXUP_GPIO1] = {
-		.type = HDA_FIXUP_VERBS,
-		.v.verbs = alc_gpio1_init_verbs,
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_gpio1,
 	},
 	[ALC260_FIXUP_GPIO1_TOGGLE] = {
 		.type = HDA_FIXUP_FUNC,
@@ -1751,8 +1780,11 @@ static int patch_alc260(struct hda_codec *codec)
 	if (err < 0)
 		goto error;
 
-	if (!spec->gen.no_analog)
-		set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
+	if (!spec->gen.no_analog) {
+		err = set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
+		if (err < 0)
+			goto error;
+	}
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
 
@@ -1824,47 +1856,14 @@ static void alc889_fixup_coef(struct hda_codec *codec,
 	alc_update_coef_idx(codec, 7, 0, 0x2030);
 }
 
-/* toggle speaker-output according to the hp-jack state */
-static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted)
-{
-	unsigned int gpiostate, gpiomask, gpiodir;
-
-	gpiostate = snd_hda_codec_read(codec, codec->core.afg, 0,
-				       AC_VERB_GET_GPIO_DATA, 0);
-
-	if (!muted)
-		gpiostate |= (1 << pin);
-	else
-		gpiostate &= ~(1 << pin);
-
-	gpiomask = snd_hda_codec_read(codec, codec->core.afg, 0,
-				      AC_VERB_GET_GPIO_MASK, 0);
-	gpiomask |= (1 << pin);
-
-	gpiodir = snd_hda_codec_read(codec, codec->core.afg, 0,
-				     AC_VERB_GET_GPIO_DIRECTION, 0);
-	gpiodir |= (1 << pin);
-
-
-	snd_hda_codec_write(codec, codec->core.afg, 0,
-			    AC_VERB_SET_GPIO_MASK, gpiomask);
-	snd_hda_codec_write(codec, codec->core.afg, 0,
-			    AC_VERB_SET_GPIO_DIRECTION, gpiodir);
-
-	msleep(1);
-
-	snd_hda_codec_write(codec, codec->core.afg, 0,
-			    AC_VERB_SET_GPIO_DATA, gpiostate);
-}
-
 /* set up GPIO at initialization */
 static void alc885_fixup_macpro_gpio(struct hda_codec *codec,
 				     const struct hda_fixup *fix, int action)
 {
-	if (action != HDA_FIXUP_ACT_INIT)
-		return;
-	alc882_gpio_mute(codec, 0, 0);
-	alc882_gpio_mute(codec, 1, 0);
+	struct alc_spec *spec = codec->spec;
+
+	spec->gpio_write_delay = true;
+	alc_fixup_gpio3(codec, fix, action);
 }
 
 /* Fix the connection of some pins for ALC889:
@@ -2143,20 +2142,20 @@ static const struct hda_fixup alc882_fixups[] = {
 		}
 	},
 	[ALC882_FIXUP_GPIO1] = {
-		.type = HDA_FIXUP_VERBS,
-		.v.verbs = alc_gpio1_init_verbs,
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_gpio1,
 	},
 	[ALC882_FIXUP_GPIO2] = {
-		.type = HDA_FIXUP_VERBS,
-		.v.verbs = alc_gpio2_init_verbs,
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_gpio2,
 	},
 	[ALC882_FIXUP_GPIO3] = {
-		.type = HDA_FIXUP_VERBS,
-		.v.verbs = alc_gpio3_init_verbs,
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_gpio3,
 	},
 	[ALC882_FIXUP_ASUS_W2JC] = {
-		.type = HDA_FIXUP_VERBS,
-		.v.verbs = alc_gpio1_init_verbs,
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_gpio1,
 		.chained = true,
 		.chain_id = ALC882_FIXUP_EAPD,
 	},
@@ -2376,12 +2375,37 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 };
 
 static const struct hda_model_fixup alc882_fixup_models[] = {
+	{.id = ALC882_FIXUP_ABIT_AW9D_MAX, .name = "abit-aw9d"},
+	{.id = ALC882_FIXUP_LENOVO_Y530, .name = "lenovo-y530"},
+	{.id = ALC882_FIXUP_ACER_ASPIRE_7736, .name = "acer-aspire-7736"},
+	{.id = ALC882_FIXUP_ASUS_W90V, .name = "asus-w90v"},
+	{.id = ALC889_FIXUP_CD, .name = "cd"},
+	{.id = ALC889_FIXUP_FRONT_HP_NO_PRESENCE, .name = "no-front-hp"},
+	{.id = ALC889_FIXUP_VAIO_TT, .name = "vaio-tt"},
+	{.id = ALC888_FIXUP_EEE1601, .name = "eee1601"},
+	{.id = ALC882_FIXUP_EAPD, .name = "alc882-eapd"},
+	{.id = ALC883_FIXUP_EAPD, .name = "alc883-eapd"},
+	{.id = ALC882_FIXUP_GPIO1, .name = "gpio1"},
+	{.id = ALC882_FIXUP_GPIO2, .name = "gpio2"},
+	{.id = ALC882_FIXUP_GPIO3, .name = "gpio3"},
+	{.id = ALC889_FIXUP_COEF, .name = "alc889-coef"},
+	{.id = ALC882_FIXUP_ASUS_W2JC, .name = "asus-w2jc"},
 	{.id = ALC882_FIXUP_ACER_ASPIRE_4930G, .name = "acer-aspire-4930g"},
 	{.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"},
 	{.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"},
+	{.id = ALC885_FIXUP_MACPRO_GPIO, .name = "macpro-gpio"},
+	{.id = ALC889_FIXUP_DAC_ROUTE, .name = "dac-route"},
+	{.id = ALC889_FIXUP_MBP_VREF, .name = "mbp-vref"},
+	{.id = ALC889_FIXUP_IMAC91_VREF, .name = "imac91-vref"},
+	{.id = ALC889_FIXUP_MBA11_VREF, .name = "mba11-vref"},
+	{.id = ALC889_FIXUP_MBA21_VREF, .name = "mba21-vref"},
+	{.id = ALC889_FIXUP_MP11_VREF, .name = "mp11-vref"},
+	{.id = ALC889_FIXUP_MP41_VREF, .name = "mp41-vref"},
 	{.id = ALC882_FIXUP_INV_DMIC, .name = "inv-dmic"},
 	{.id = ALC882_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"},
+	{.id = ALC887_FIXUP_ASUS_BASS, .name = "asus-bass"},
 	{.id = ALC1220_FIXUP_GB_DUAL_CODECS, .name = "dual-codecs"},
+	{.id = ALC1220_FIXUP_CLEVO_P950, .name = "clevo-p950"},
 	{}
 };
 
@@ -2435,8 +2459,11 @@ static int patch_alc882(struct hda_codec *codec)
 	if (err < 0)
 		goto error;
 
-	if (!spec->gen.no_analog && spec->gen.beep_nid)
-		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+	if (!spec->gen.no_analog && spec->gen.beep_nid) {
+		err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+		if (err < 0)
+			goto error;
+	}
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
 
@@ -2557,6 +2584,14 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = {
 
 static const struct hda_model_fixup alc262_fixup_models[] = {
 	{.id = ALC262_FIXUP_INV_DMIC, .name = "inv-dmic"},
+	{.id = ALC262_FIXUP_FSC_H270, .name = "fsc-h270"},
+	{.id = ALC262_FIXUP_FSC_S7110, .name = "fsc-s7110"},
+	{.id = ALC262_FIXUP_HP_Z200, .name = "hp-z200"},
+	{.id = ALC262_FIXUP_TYAN, .name = "tyan"},
+	{.id = ALC262_FIXUP_LENOVO_3000, .name = "lenovo-3000"},
+	{.id = ALC262_FIXUP_BENQ, .name = "benq"},
+	{.id = ALC262_FIXUP_BENQ_T31, .name = "benq-t31"},
+	{.id = ALC262_FIXUP_INTEL_BAYLEYBAY, .name = "bayleybay"},
 	{}
 };
 
@@ -2598,8 +2633,11 @@ static int patch_alc262(struct hda_codec *codec)
 	if (err < 0)
 		goto error;
 
-	if (!spec->gen.no_analog && spec->gen.beep_nid)
-		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+	if (!spec->gen.no_analog && spec->gen.beep_nid) {
+		err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+		if (err < 0)
+			goto error;
+	}
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
 
@@ -2645,7 +2683,6 @@ static const struct snd_kcontrol_new alc268_beep_mixer[] = {
 		.put = alc268_beep_switch_put,
 		.private_value = HDA_COMPOSE_AMP_VAL(0x0f, 3, 1, HDA_INPUT)
 	},
-	{ }
 };
 
 /* set PCBEEP vol = 0, mute connections */
@@ -2686,6 +2723,7 @@ static const struct hda_fixup alc268_fixups[] = {
 static const struct hda_model_fixup alc268_fixup_models[] = {
 	{.id = ALC268_FIXUP_INV_DMIC, .name = "inv-dmic"},
 	{.id = ALC268_FIXUP_HP_EAPD, .name = "hp-eapd"},
+	{.id = ALC268_FIXUP_SPDIF, .name = "spdif"},
 	{}
 };
 
@@ -2713,7 +2751,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 static int patch_alc268(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
-	int err;
+	int i, err;
 
 	/* ALC268 has no aa-loopback mixer */
 	err = alc_alloc_spec(codec, 0);
@@ -2735,7 +2773,13 @@ static int patch_alc268(struct hda_codec *codec)
 
 	if (err > 0 && !spec->gen.no_analog &&
 	    spec->gen.autocfg.speaker_pins[0] != 0x1d) {
-		add_mixer(spec, alc268_beep_mixer);
+		for (i = 0; i < ARRAY_SIZE(alc268_beep_mixer); i++) {
+			if (!snd_hda_gen_add_kctl(&spec->gen, NULL,
+						  &alc268_beep_mixer[i])) {
+				err = -ENOMEM;
+				goto error;
+			}
+		}
 		snd_hda_add_verbs(codec, alc268_beep_init_verbs);
 		if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
 			/* override the amp caps for beep generator */
@@ -3454,9 +3498,8 @@ static int alc269_resume(struct hda_codec *codec)
 	 * suspend, and won't restore the data after resume, so we restore it
 	 * in the driver.
 	 */
-	if (spec->gpio_led)
-		snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_DATA,
-			    spec->gpio_led);
+	if (spec->gpio_data)
+		alc_write_gpio_data(codec);
 
 	if (spec->has_alc5505_dsp)
 		alc5505_dsp_resume(codec);
@@ -3696,18 +3739,10 @@ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask,
 				bool enabled)
 {
 	struct alc_spec *spec = codec->spec;
-	unsigned int oldval = spec->gpio_led;
 
 	if (spec->mute_led_polarity)
 		enabled = !enabled;
-
-	if (enabled)
-		spec->gpio_led &= ~mask;
-	else
-		spec->gpio_led |= mask;
-	if (spec->gpio_led != oldval)
-		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
-				    spec->gpio_led);
+	alc_update_gpio_data(codec, mask, !enabled); /* muted -> LED on */
 }
 
 /* turn on/off mute LED via GPIO per vmaster hook */
@@ -3720,104 +3755,79 @@ static void alc_fixup_gpio_mute_hook(void *private_data, int enabled)
 }
 
 /* turn on/off mic-mute LED via GPIO per capture hook */
-static void alc_fixup_gpio_mic_mute_hook(struct hda_codec *codec,
-					 struct snd_kcontrol *kcontrol,
-					 struct snd_ctl_elem_value *ucontrol)
+static void alc_gpio_micmute_update(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (ucontrol)
-		alc_update_gpio_led(codec, spec->gpio_mic_led_mask,
-				    ucontrol->value.integer.value[0] ||
-				    ucontrol->value.integer.value[1]);
+	alc_update_gpio_led(codec, spec->gpio_mic_led_mask,
+			    spec->gen.micmute_led.led_value);
 }
 
-static void alc269_fixup_hp_gpio_led(struct hda_codec *codec,
-				const struct hda_fixup *fix, int action)
+/* setup mute and mic-mute GPIO bits, add hooks appropriately */
+static void alc_fixup_hp_gpio_led(struct hda_codec *codec,
+				  int action,
+				  unsigned int mute_mask,
+				  unsigned int micmute_mask)
 {
 	struct alc_spec *spec = codec->spec;
-	static const struct hda_verb gpio_init[] = {
-		{ 0x01, AC_VERB_SET_GPIO_MASK, 0x18 },
-		{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x18 },
-		{}
-	};
 
-	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+	alc_fixup_gpio(codec, action, mute_mask | micmute_mask);
+
+	if (action != HDA_FIXUP_ACT_PRE_PROBE)
+		return;
+	if (mute_mask) {
+		spec->gpio_mute_led_mask = mute_mask;
 		spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook;
-		spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook;
-		spec->gpio_led = 0;
-		spec->mute_led_polarity = 0;
-		spec->gpio_mute_led_mask = 0x08;
-		spec->gpio_mic_led_mask = 0x10;
-		snd_hda_add_verbs(codec, gpio_init);
+	}
+	if (micmute_mask) {
+		spec->gpio_mic_led_mask = micmute_mask;
+		snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update);
 	}
 }
 
-static void alc286_fixup_hp_gpio_led(struct hda_codec *codec,
+static void alc269_fixup_hp_gpio_led(struct hda_codec *codec,
 				const struct hda_fixup *fix, int action)
 {
-	struct alc_spec *spec = codec->spec;
-	static const struct hda_verb gpio_init[] = {
-		{ 0x01, AC_VERB_SET_GPIO_MASK, 0x22 },
-		{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x22 },
-		{}
-	};
+	alc_fixup_hp_gpio_led(codec, action, 0x08, 0x10);
+}
 
-	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook;
-		spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook;
-		spec->gpio_led = 0;
-		spec->mute_led_polarity = 0;
-		spec->gpio_mute_led_mask = 0x02;
-		spec->gpio_mic_led_mask = 0x20;
-		snd_hda_add_verbs(codec, gpio_init);
-	}
+static void alc286_fixup_hp_gpio_led(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	alc_fixup_hp_gpio_led(codec, action, 0x02, 0x20);
 }
 
 /* turn on/off mic-mute LED per capture hook */
-static void alc269_fixup_hp_cap_mic_mute_hook(struct hda_codec *codec,
-					       struct snd_kcontrol *kcontrol,
-					       struct snd_ctl_elem_value *ucontrol)
+static void alc_cap_micmute_update(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	unsigned int pinval, enable, disable;
+	unsigned int pinval;
 
+	if (!spec->cap_mute_led_nid)
+		return;
 	pinval = snd_hda_codec_get_pin_target(codec, spec->cap_mute_led_nid);
 	pinval &= ~AC_PINCTL_VREFEN;
-	enable  = pinval | AC_PINCTL_VREF_80;
-	disable = pinval | AC_PINCTL_VREF_HIZ;
-
-	if (!ucontrol)
-		return;
-
-	if (ucontrol->value.integer.value[0] ||
-	    ucontrol->value.integer.value[1])
-		pinval = disable;
+	if (spec->gen.micmute_led.led_value)
+		pinval |= AC_PINCTL_VREF_80;
 	else
-		pinval = enable;
-
-	if (spec->cap_mute_led_nid)
-		snd_hda_set_pin_ctl_cache(codec, spec->cap_mute_led_nid, pinval);
+		pinval |= AC_PINCTL_VREF_HIZ;
+	snd_hda_set_pin_ctl_cache(codec, spec->cap_mute_led_nid, pinval);
 }
 
 static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec,
 				const struct hda_fixup *fix, int action)
 {
 	struct alc_spec *spec = codec->spec;
-	static const struct hda_verb gpio_init[] = {
-		{ 0x01, AC_VERB_SET_GPIO_MASK, 0x08 },
-		{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x08 },
-		{}
-	};
 
+	alc_fixup_hp_gpio_led(codec, action, 0x08, 0);
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook;
-		spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook;
-		spec->gpio_led = 0;
-		spec->mute_led_polarity = 0;
-		spec->gpio_mute_led_mask = 0x08;
+		/* Like hp_gpio_mic1_led, but also needs GPIO4 low to
+		 * enable headphone amp
+		 */
+		spec->gpio_mask |= 0x10;
+		spec->gpio_dir |= 0x10;
 		spec->cap_mute_led_nid = 0x18;
-		snd_hda_add_verbs(codec, gpio_init);
+		snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update);
 		codec->power_filter = led_power_filter;
 	}
 }
@@ -3825,22 +3835,12 @@ static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec,
 static void alc280_fixup_hp_gpio4(struct hda_codec *codec,
 				   const struct hda_fixup *fix, int action)
 {
-	/* Like hp_gpio_mic1_led, but also needs GPIO4 low to enable headphone amp */
 	struct alc_spec *spec = codec->spec;
-	static const struct hda_verb gpio_init[] = {
-		{ 0x01, AC_VERB_SET_GPIO_MASK, 0x18 },
-		{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x18 },
-		{}
-	};
 
+	alc_fixup_hp_gpio_led(codec, action, 0x08, 0);
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook;
-		spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook;
-		spec->gpio_led = 0;
-		spec->mute_led_polarity = 0;
-		spec->gpio_mute_led_mask = 0x08;
 		spec->cap_mute_led_nid = 0x18;
-		snd_hda_add_verbs(codec, gpio_init);
+		snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update);
 		codec->power_filter = led_power_filter;
 	}
 }
@@ -3890,38 +3890,29 @@ static int alc_register_micmute_input_device(struct hda_codec *codec)
 	return 0;
 }
 
+/* GPIO1 = set according to SKU external amp
+ * GPIO2 = mic mute hotkey
+ * GPIO3 = mute LED
+ * GPIO4 = mic mute LED
+ */
 static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec,
 					     const struct hda_fixup *fix, int action)
 {
-	/* GPIO1 = set according to SKU external amp
-	   GPIO2 = mic mute hotkey
-	   GPIO3 = mute LED
-	   GPIO4 = mic mute LED */
-	static const struct hda_verb gpio_init[] = {
-		{ 0x01, AC_VERB_SET_GPIO_MASK, 0x1e },
-		{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x1a },
-		{ 0x01, AC_VERB_SET_GPIO_DATA, 0x02 },
-		{}
-	};
-
 	struct alc_spec *spec = codec->spec;
 
+	alc_fixup_hp_gpio_led(codec, action, 0x08, 0x10);
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->init_amp = ALC_INIT_DEFAULT;
 		if (alc_register_micmute_input_device(codec) != 0)
 			return;
 
-		snd_hda_add_verbs(codec, gpio_init);
+		spec->gpio_mask |= 0x06;
+		spec->gpio_dir |= 0x02;
+		spec->gpio_data |= 0x02;
 		snd_hda_codec_write_cache(codec, codec->core.afg, 0,
 					  AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x04);
 		snd_hda_jack_detect_enable_callback(codec, codec->core.afg,
 						    gpio2_mic_hotkey_event);
-
-		spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook;
-		spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook;
-		spec->gpio_led = 0;
-		spec->mute_led_polarity = 0;
-		spec->gpio_mute_led_mask = 0x08;
-		spec->gpio_mic_led_mask = 0x10;
 		return;
 	}
 
@@ -3929,40 +3920,28 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec,
 		return;
 
 	switch (action) {
-	case HDA_FIXUP_ACT_PROBE:
-		spec->init_amp = ALC_INIT_DEFAULT;
-		break;
 	case HDA_FIXUP_ACT_FREE:
 		input_unregister_device(spec->kb_dev);
 		spec->kb_dev = NULL;
 	}
 }
 
+/* Line2 = mic mute hotkey
+ * GPIO2 = mic mute LED
+ */
 static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec,
 					     const struct hda_fixup *fix, int action)
 {
-	/* Line2 = mic mute hotkey
-	   GPIO2 = mic mute LED */
-	static const struct hda_verb gpio_init[] = {
-		{ 0x01, AC_VERB_SET_GPIO_MASK, 0x04 },
-		{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 },
-		{}
-	};
-
 	struct alc_spec *spec = codec->spec;
 
+	alc_fixup_hp_gpio_led(codec, action, 0, 0x04);
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->init_amp = ALC_INIT_DEFAULT;
 		if (alc_register_micmute_input_device(codec) != 0)
 			return;
 
-		snd_hda_add_verbs(codec, gpio_init);
 		snd_hda_jack_detect_enable_callback(codec, 0x1b,
 						    gpio2_mic_hotkey_event);
-
-		spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook;
-		spec->gpio_led = 0;
-		spec->mute_led_polarity = 0;
-		spec->gpio_mic_led_mask = 0x04;
 		return;
 	}
 
@@ -3970,9 +3949,6 @@ static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec,
 		return;
 
 	switch (action) {
-	case HDA_FIXUP_ACT_PROBE:
-		spec->init_amp = ALC_INIT_DEFAULT;
-		break;
 	case HDA_FIXUP_ACT_FREE:
 		input_unregister_device(spec->kb_dev);
 		spec->kb_dev = NULL;
@@ -3988,14 +3964,10 @@ static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec,
 {
 	struct alc_spec *spec = codec->spec;
 
+	alc269_fixup_hp_mute_led_micx(codec, fix, action, 0x1a);
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook;
-		spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook;
-		spec->mute_led_polarity = 0;
-		spec->mute_led_nid = 0x1a;
 		spec->cap_mute_led_nid = 0x18;
-		spec->gen.vmaster_mute_enum = 1;
-		codec->power_filter = led_power_filter;
+		snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update);
 	}
 }
 
@@ -4843,6 +4815,7 @@ static void alc_probe_headset_mode(struct hda_codec *codec)
 			spec->headphone_mic_pin = cfg->inputs[i].pin;
 	}
 
+	WARN_ON(spec->gen.cap_sync_hook);
 	spec->gen.cap_sync_hook = alc_update_headset_mode_hook;
 	spec->gen.automute_hook = alc_update_headset_mode;
 	spec->gen.hp_automute_hook = alc_update_headset_jack_cb;
@@ -4934,13 +4907,10 @@ static void alc288_update_headset_jack_cb(struct hda_codec *codec,
 				       struct hda_jack_callback *jack)
 {
 	struct alc_spec *spec = codec->spec;
-	int present;
 
 	alc_update_headset_jack_cb(codec, jack);
 	/* Headset Mic enable or disable, only for Dell Dino */
-	present = spec->gen.hp_jack_present ? 0x40 : 0;
-	snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
-				present);
+	alc_update_gpio_data(codec, 0x40, spec->gen.hp_jack_present);
 }
 
 static void alc_fixup_headset_mode_dell_alc288(struct hda_codec *codec,
@@ -4949,6 +4919,9 @@ static void alc_fixup_headset_mode_dell_alc288(struct hda_codec *codec,
 	alc_fixup_headset_mode(codec, fix, action);
 	if (action == HDA_FIXUP_ACT_PROBE) {
 		struct alc_spec *spec = codec->spec;
+		/* toggled via hp_automute_hook */
+		spec->gpio_mask |= 0x40;
+		spec->gpio_dir |= 0x40;
 		spec->gen.hp_automute_hook = alc288_update_headset_jack_cb;
 	}
 }
@@ -4969,7 +4942,7 @@ static void alc_no_shutup(struct hda_codec *codec)
 static void alc_fixup_no_shutup(struct hda_codec *codec,
 				const struct hda_fixup *fix, int action)
 {
-	if (action == HDA_FIXUP_ACT_PROBE) {
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		struct alc_spec *spec = codec->spec;
 		spec->shutup = alc_no_shutup;
 	}
@@ -5051,10 +5024,9 @@ static void alc_fixup_dell_xps13(struct hda_codec *codec,
 		 * it causes a click noise at start up
 		 */
 		snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ);
+		spec->shutup = alc_shutup_dell_xps13;
 		break;
 	case HDA_FIXUP_ACT_PROBE:
-		spec->shutup = alc_shutup_dell_xps13;
-
 		/* Make the internal mic the default input source. */
 		for (i = 0; i < imux->num_items; i++) {
 			if (spec->gen.imux_pins[i] == 0x12) {
@@ -5231,13 +5203,6 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec,
 				    const struct hda_fixup *fix, int action)
 {
 	struct alc_spec *spec = codec->spec;
-	/* TX300 needs to set up GPIO2 for the speaker amp */
-	static const struct hda_verb gpio2_verbs[] = {
-		{ 0x01, AC_VERB_SET_GPIO_MASK, 0x04 },
-		{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 },
-		{ 0x01, AC_VERB_SET_GPIO_DATA, 0x04 },
-		{}
-	};
 	static const struct hda_pintbl dock_pins[] = {
 		{ 0x1b, 0x21114000 }, /* dock speaker pin */
 		{}
@@ -5245,13 +5210,18 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec,
 
 	switch (action) {
 	case HDA_FIXUP_ACT_PRE_PROBE:
-		snd_hda_add_verbs(codec, gpio2_verbs);
+		spec->init_amp = ALC_INIT_DEFAULT;
+		/* TX300 needs to set up GPIO2 for the speaker amp */
+		alc_setup_gpio(codec, 0x04);
 		snd_hda_apply_pincfgs(codec, dock_pins);
 		spec->gen.auto_mute_via_amp = 1;
 		spec->gen.automute_hook = asus_tx300_automute;
 		snd_hda_jack_detect_enable_callback(codec, 0x1b,
 						    snd_hda_gen_hp_automute);
 		break;
+	case HDA_FIXUP_ACT_PROBE:
+		spec->init_amp = ALC_INIT_DEFAULT;
+		break;
 	case HDA_FIXUP_ACT_BUILD:
 		/* this is a bit tricky; give more sane names for the main
 		 * (tablet) speaker and the dock speaker, respectively
@@ -5325,30 +5295,26 @@ static void alc280_fixup_hp_9480m(struct hda_codec *codec,
 				  int action)
 {
 	struct alc_spec *spec = codec->spec;
-	static const struct hda_verb gpio_init[] = {
-		{ 0x01, AC_VERB_SET_GPIO_MASK, 0x18 },
-		{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x18 },
-		{}
-	};
 
+	alc_fixup_hp_gpio_led(codec, action, 0x08, 0);
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		/* Set the hooks to turn the headphone amp on/off
-		 * as needed
-		 */
-		spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook;
+		/* amp at GPIO4; toggled via alc280_hp_gpio4_automute_hook() */
+		spec->gpio_mask |= 0x10;
+		spec->gpio_dir |= 0x10;
 		spec->gen.hp_automute_hook = alc280_hp_gpio4_automute_hook;
+	}
+}
 
-		/* The GPIOs are currently off */
-		spec->gpio_led = 0;
-
-		/* GPIO3 is connected to the output mute LED,
-		 * high is on, low is off
-		 */
-		spec->mute_led_polarity = 0;
-		spec->gpio_mute_led_mask = 0x08;
+static void alc275_fixup_gpio4_off(struct hda_codec *codec,
+				   const struct hda_fixup *fix,
+				   int action)
+{
+	struct alc_spec *spec = codec->spec;
 
-		/* Initialize GPIO configuration */
-		snd_hda_add_verbs(codec, gpio_init);
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->gpio_mask |= 0x04;
+		spec->gpio_dir |= 0x04;
+		/* set data bit low */
 	}
 }
 
@@ -5492,7 +5458,6 @@ enum {
 	ALC280_FIXUP_HP_9480M,
 	ALC288_FIXUP_DELL_HEADSET_MODE,
 	ALC288_FIXUP_DELL1_MIC_NO_PRESENCE,
-	ALC288_FIXUP_DELL_XPS_13_GPIO6,
 	ALC288_FIXUP_DELL_XPS_13,
 	ALC288_FIXUP_DISABLE_AAMIX,
 	ALC292_FIXUP_DELL_E7X,
@@ -5540,13 +5505,8 @@ static const struct hda_fixup alc269_fixups[] = {
 		}
 	},
 	[ALC275_FIXUP_SONY_VAIO_GPIO2] = {
-		.type = HDA_FIXUP_VERBS,
-		.v.verbs = (const struct hda_verb[]) {
-			{0x01, AC_VERB_SET_GPIO_MASK, 0x04},
-			{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04},
-			{0x01, AC_VERB_SET_GPIO_DATA, 0x00},
-			{ }
-		},
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc275_fixup_gpio4_off,
 		.chained = true,
 		.chain_id = ALC269_FIXUP_SONY_VAIO
 	},
@@ -6113,22 +6073,11 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC288_FIXUP_DELL_HEADSET_MODE
 	},
-	[ALC288_FIXUP_DELL_XPS_13_GPIO6] = {
-		.type = HDA_FIXUP_VERBS,
-		.v.verbs = (const struct hda_verb[]) {
-			{0x01, AC_VERB_SET_GPIO_MASK, 0x40},
-			{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x40},
-			{0x01, AC_VERB_SET_GPIO_DATA, 0x00},
-			{ }
-		},
-		.chained = true,
-		.chain_id = ALC288_FIXUP_DELL1_MIC_NO_PRESENCE
-	},
 	[ALC288_FIXUP_DISABLE_AAMIX] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_disable_aamix,
 		.chained = true,
-		.chain_id = ALC288_FIXUP_DELL_XPS_13_GPIO6
+		.chain_id = ALC288_FIXUP_DELL1_MIC_NO_PRESENCE
 	},
 	[ALC288_FIXUP_DELL_XPS_13] = {
 		.type = HDA_FIXUP_FUNC,
@@ -6291,14 +6240,9 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
 	},
 	[ALC256_FIXUP_ASUS_AIO_GPIO2] = {
-		.type = HDA_FIXUP_VERBS,
-		.v.verbs = (const struct hda_verb[]) {
-			/* Set up GPIO2 for the speaker amp */
-			{ 0x01, AC_VERB_SET_GPIO_MASK, 0x04 },
-			{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 },
-			{ 0x01, AC_VERB_SET_GPIO_DATA, 0x04 },
-			{}
-		},
+		.type = HDA_FIXUP_FUNC,
+		/* Set up GPIO2 for the speaker amp */
+		.v.func = alc_fixup_gpio4,
 	},
 	[ALC233_FIXUP_ASUS_MIC_NO_PRESENCE] = {
 		.type = HDA_FIXUP_PINS,
@@ -6530,6 +6474,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360),
 	SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -6713,13 +6658,95 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, .name = "hp-dock-gpio-mic1-led"},
 	{.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"},
 	{.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"},
+	{.id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, .name = "dell-headset3"},
+	{.id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, .name = "dell-headset4"},
 	{.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"},
 	{.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"},
 	{.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"},
 	{.id = ALC292_FIXUP_TPT440, .name = "tpt440"},
 	{.id = ALC292_FIXUP_TPT460, .name = "tpt460"},
+	{.id = ALC298_FIXUP_TPT470_DOCK, .name = "tpt470-dock"},
 	{.id = ALC233_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"},
 	{.id = ALC700_FIXUP_INTEL_REFERENCE, .name = "alc700-ref"},
+	{.id = ALC269_FIXUP_SONY_VAIO, .name = "vaio"},
+	{.id = ALC269_FIXUP_DELL_M101Z, .name = "dell-m101z"},
+	{.id = ALC269_FIXUP_ASUS_G73JW, .name = "asus-g73jw"},
+	{.id = ALC269_FIXUP_LENOVO_EAPD, .name = "lenovo-eapd"},
+	{.id = ALC275_FIXUP_SONY_HWEQ, .name = "sony-hweq"},
+	{.id = ALC269_FIXUP_PCM_44K, .name = "pcm44k"},
+	{.id = ALC269_FIXUP_LIFEBOOK, .name = "lifebook"},
+	{.id = ALC269_FIXUP_LIFEBOOK_EXTMIC, .name = "lifebook-extmic"},
+	{.id = ALC269_FIXUP_LIFEBOOK_HP_PIN, .name = "lifebook-hp-pin"},
+	{.id = ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC, .name = "lifebook-u7x7"},
+	{.id = ALC269VB_FIXUP_AMIC, .name = "alc269vb-amic"},
+	{.id = ALC269VB_FIXUP_DMIC, .name = "alc269vb-dmic"},
+	{.id = ALC269_FIXUP_HP_MUTE_LED_MIC1, .name = "hp-mute-led-mic1"},
+	{.id = ALC269_FIXUP_HP_MUTE_LED_MIC2, .name = "hp-mute-led-mic2"},
+	{.id = ALC269_FIXUP_HP_MUTE_LED_MIC3, .name = "hp-mute-led-mic3"},
+	{.id = ALC269_FIXUP_HP_GPIO_MIC1_LED, .name = "hp-gpio-mic1"},
+	{.id = ALC269_FIXUP_HP_LINE1_MIC1_LED, .name = "hp-line1-mic1"},
+	{.id = ALC269_FIXUP_NO_SHUTUP, .name = "noshutup"},
+	{.id = ALC286_FIXUP_SONY_MIC_NO_PRESENCE, .name = "sony-nomic"},
+	{.id = ALC269_FIXUP_ASPIRE_HEADSET_MIC, .name = "aspire-headset-mic"},
+	{.id = ALC269_FIXUP_ASUS_X101, .name = "asus-x101"},
+	{.id = ALC271_FIXUP_HP_GATE_MIC_JACK, .name = "acer-ao7xx"},
+	{.id = ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572, .name = "acer-aspire-e1"},
+	{.id = ALC269_FIXUP_ACER_AC700, .name = "acer-ac700"},
+	{.id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST, .name = "limit-mic-boost"},
+	{.id = ALC269VB_FIXUP_ASUS_ZENBOOK, .name = "asus-zenbook"},
+	{.id = ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A, .name = "asus-zenbook-ux31a"},
+	{.id = ALC269VB_FIXUP_ORDISSIMO_EVE2, .name = "ordissimo"},
+	{.id = ALC282_FIXUP_ASUS_TX300, .name = "asus-tx300"},
+	{.id = ALC283_FIXUP_INT_MIC, .name = "alc283-int-mic"},
+	{.id = ALC290_FIXUP_MONO_SPEAKERS_HSJACK, .name = "mono-speakers"},
+	{.id = ALC290_FIXUP_SUBWOOFER_HSJACK, .name = "alc290-subwoofer"},
+	{.id = ALC269_FIXUP_THINKPAD_ACPI, .name = "thinkpad"},
+	{.id = ALC269_FIXUP_DMIC_THINKPAD_ACPI, .name = "dmic-thinkpad"},
+	{.id = ALC255_FIXUP_ACER_MIC_NO_PRESENCE, .name = "alc255-acer"},
+	{.id = ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, .name = "alc255-asus"},
+	{.id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc255-dell1"},
+	{.id = ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "alc255-dell2"},
+	{.id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc293-dell1"},
+	{.id = ALC283_FIXUP_HEADSET_MIC, .name = "alc283-headset"},
+	{.id = ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, .name = "alc255-dell-mute"},
+	{.id = ALC282_FIXUP_ASPIRE_V5_PINS, .name = "aspire-v5"},
+	{.id = ALC280_FIXUP_HP_GPIO4, .name = "hp-gpio4"},
+	{.id = ALC286_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"},
+	{.id = ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, .name = "hp-gpio2-hotkey"},
+	{.id = ALC280_FIXUP_HP_DOCK_PINS, .name = "hp-dock-pins"},
+	{.id = ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, .name = "hp-dock-gpio-mic"},
+	{.id = ALC280_FIXUP_HP_9480M, .name = "hp-9480m"},
+	{.id = ALC288_FIXUP_DELL_HEADSET_MODE, .name = "alc288-dell-headset"},
+	{.id = ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc288-dell1"},
+	{.id = ALC288_FIXUP_DELL_XPS_13, .name = "alc288-dell-xps13"},
+	{.id = ALC292_FIXUP_DELL_E7X, .name = "dell-e7x"},
+	{.id = ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, .name = "alc293-dell"},
+	{.id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc298-dell1"},
+	{.id = ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, .name = "alc298-dell-aio"},
+	{.id = ALC275_FIXUP_DELL_XPS, .name = "alc275-dell-xps"},
+	{.id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, .name = "alc256-dell-xps13"},
+	{.id = ALC293_FIXUP_LENOVO_SPK_NOISE, .name = "lenovo-spk-noise"},
+	{.id = ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, .name = "lenovo-hotkey"},
+	{.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"},
+	{.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc255-dell1"},
+	{.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"},
+	{.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"},
+	{.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"},
+	{.id = ALC298_FIXUP_SPK_VOLUME, .name = "alc298-spk-volume"},
+	{.id = ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER, .name = "dell-inspiron-7559"},
+	{.id = ALC269_FIXUP_ATIV_BOOK_8, .name = "ativ-book"},
+	{.id = ALC221_FIXUP_HP_MIC_NO_PRESENCE, .name = "alc221-hp-mic"},
+	{.id = ALC256_FIXUP_ASUS_HEADSET_MODE, .name = "alc256-asus-headset"},
+	{.id = ALC256_FIXUP_ASUS_MIC, .name = "alc256-asus-mic"},
+	{.id = ALC256_FIXUP_ASUS_AIO_GPIO2, .name = "alc256-asus-aio"},
+	{.id = ALC233_FIXUP_ASUS_MIC_NO_PRESENCE, .name = "alc233-asus"},
+	{.id = ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE, .name = "alc233-eapd"},
+	{.id = ALC294_FIXUP_LENOVO_MIC_LOCATION, .name = "alc294-lenovo-mic"},
+	{.id = ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE, .name = "alc225-wyse"},
+	{.id = ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, .name = "alc274-dell-aio"},
+	{.id = ALC255_FIXUP_DUMMY_LINEOUT_VERB, .name = "alc255-dummy-lineout"},
+	{.id = ALC255_FIXUP_DELL_HEADSET_MIC, .name = "alc255-dell-headset"},
+	{.id = ALC295_FIXUP_HP_X360, .name = "alc295-hp-x360"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
@@ -6983,7 +7010,7 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x12, 0x90a60130},
 		{0x19, 0x03a11020},
 		{0x21, 0x0321101f}),
-	SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL_XPS_13_GPIO6,
+	SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE,
 		{0x12, 0x90a60120},
 		{0x14, 0x90170110},
 		{0x21, 0x0321101f}),
@@ -7140,18 +7167,6 @@ static int patch_alc269(struct hda_codec *codec)
 	spec->shutup = alc_default_shutup;
 	spec->init_hook = alc_default_init;
 
-	snd_hda_pick_fixup(codec, alc269_fixup_models,
-		       alc269_fixup_tbl, alc269_fixups);
-	snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups);
-	snd_hda_pick_fixup(codec, NULL,	alc269_fixup_vendor_tbl,
-			   alc269_fixups);
-	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
-
-	alc_auto_parse_customize_define(codec);
-
-	if (has_cdefine_beep(codec))
-		spec->gen.beep_nid = 0x01;
-
 	switch (codec->core.vendor_id) {
 	case 0x10ec0269:
 		spec->codec_variant = ALC269_TYPE_ALC269VA;
@@ -7271,13 +7286,28 @@ static int patch_alc269(struct hda_codec *codec)
 		spec->init_hook = alc5505_dsp_init;
 	}
 
+	snd_hda_pick_fixup(codec, alc269_fixup_models,
+		       alc269_fixup_tbl, alc269_fixups);
+	snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups);
+	snd_hda_pick_fixup(codec, NULL,	alc269_fixup_vendor_tbl,
+			   alc269_fixups);
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
+
+	alc_auto_parse_customize_define(codec);
+
+	if (has_cdefine_beep(codec))
+		spec->gen.beep_nid = 0x01;
+
 	/* automatic parse from the BIOS config */
 	err = alc269_parse_auto_config(codec);
 	if (err < 0)
 		goto error;
 
-	if (!spec->gen.no_analog && spec->gen.beep_nid && spec->gen.mixer_nid)
-		set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT);
+	if (!spec->gen.no_analog && spec->gen.beep_nid && spec->gen.mixer_nid) {
+		err = set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT);
+		if (err < 0)
+			goto error;
+	}
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
 
@@ -7406,8 +7436,11 @@ static int patch_alc861(struct hda_codec *codec)
 	if (err < 0)
 		goto error;
 
-	if (!spec->gen.no_analog)
-		set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
+	if (!spec->gen.no_analog) {
+		err = set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
+		if (err < 0)
+			goto error;
+	}
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
 
@@ -7447,16 +7480,21 @@ static void alc861vd_fixup_dallas(struct hda_codec *codec,
 	}
 }
 
+/* reset GPIO1 */
+static void alc660vd_fixup_asus_gpio1(struct hda_codec *codec,
+				      const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		spec->gpio_mask |= 0x02;
+	alc_fixup_gpio(codec, action, 0x01);
+}
+
 static const struct hda_fixup alc861vd_fixups[] = {
 	[ALC660VD_FIX_ASUS_GPIO1] = {
-		.type = HDA_FIXUP_VERBS,
-		.v.verbs = (const struct hda_verb[]) {
-			/* reset GPIO1 */
-			{0x01, AC_VERB_SET_GPIO_MASK, 0x03},
-			{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
-			{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
-			{ }
-		}
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc660vd_fixup_asus_gpio1,
 	},
 	[ALC861VD_FIX_DALLAS] = {
 		.type = HDA_FIXUP_FUNC,
@@ -7495,8 +7533,11 @@ static int patch_alc861vd(struct hda_codec *codec)
 	if (err < 0)
 		goto error;
 
-	if (!spec->gen.no_analog)
-		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+	if (!spec->gen.no_analog) {
+		err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+		if (err < 0)
+			goto error;
+	}
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
 
@@ -7577,7 +7618,7 @@ static unsigned int gpio_led_power_filter(struct hda_codec *codec,
 					  unsigned int power_state)
 {
 	struct alc_spec *spec = codec->spec;
-	if (nid == codec->core.afg && power_state == AC_PWRST_D3 && spec->gpio_led)
+	if (nid == codec->core.afg && power_state == AC_PWRST_D3 && spec->gpio_data)
 		return AC_PWRST_D0;
 	return power_state;
 }
@@ -7586,18 +7627,10 @@ static void alc662_fixup_led_gpio1(struct hda_codec *codec,
 				   const struct hda_fixup *fix, int action)
 {
 	struct alc_spec *spec = codec->spec;
-	static const struct hda_verb gpio_init[] = {
-		{ 0x01, AC_VERB_SET_GPIO_MASK, 0x01 },
-		{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01 },
-		{}
-	};
 
+	alc_fixup_hp_gpio_led(codec, action, 0x01, 0);
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook;
-		spec->gpio_led = 0;
 		spec->mute_led_polarity = 1;
-		spec->gpio_mute_led_mask = 0x01;
-		snd_hda_add_verbs(codec, gpio_init);
 		codec->power_filter = gpio_led_power_filter;
 	}
 }
@@ -8110,7 +8143,10 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 };
 
 static const struct hda_model_fixup alc662_fixup_models[] = {
+	{.id = ALC662_FIXUP_ASPIRE, .name = "aspire"},
+	{.id = ALC662_FIXUP_IDEAPAD, .name = "ideapad"},
 	{.id = ALC272_FIXUP_MARIO, .name = "mario"},
+	{.id = ALC662_FIXUP_HP_RP5800, .name = "hp-rp5800"},
 	{.id = ALC662_FIXUP_ASUS_MODE1, .name = "asus-mode1"},
 	{.id = ALC662_FIXUP_ASUS_MODE2, .name = "asus-mode2"},
 	{.id = ALC662_FIXUP_ASUS_MODE3, .name = "asus-mode3"},
@@ -8119,8 +8155,23 @@ static const struct hda_model_fixup alc662_fixup_models[] = {
 	{.id = ALC662_FIXUP_ASUS_MODE6, .name = "asus-mode6"},
 	{.id = ALC662_FIXUP_ASUS_MODE7, .name = "asus-mode7"},
 	{.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"},
+	{.id = ALC662_FIXUP_ZOTAC_Z68, .name = "zotac-z68"},
 	{.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"},
+	{.id = ALC662_FIXUP_DELL_MIC_NO_PRESENCE, .name = "alc662-headset-multi"},
 	{.id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE, .name = "dell-headset-multi"},
+	{.id = ALC662_FIXUP_HEADSET_MODE, .name = "alc662-headset"},
+	{.id = ALC668_FIXUP_HEADSET_MODE, .name = "alc668-headset"},
+	{.id = ALC662_FIXUP_BASS_16, .name = "bass16"},
+	{.id = ALC662_FIXUP_BASS_1A, .name = "bass1a"},
+	{.id = ALC668_FIXUP_AUTO_MUTE, .name = "automute"},
+	{.id = ALC668_FIXUP_DELL_XPS13, .name = "dell-xps13"},
+	{.id = ALC662_FIXUP_ASUS_Nx50, .name = "asus-nx50"},
+	{.id = ALC668_FIXUP_ASUS_Nx51, .name = "asus-nx51"},
+	{.id = ALC891_FIXUP_HEADSET_MODE, .name = "alc891-headset"},
+	{.id = ALC891_FIXUP_DELL_MIC_NO_PRESENCE, .name = "alc891-headset-multi"},
+	{.id = ALC662_FIXUP_ACER_VERITON, .name = "acer-veriton"},
+	{.id = ALC892_FIXUP_ASROCK_MOBO, .name = "asrock-mobo"},
+	{.id = ALC662_FIXUP_USI_HEADSET_MODE, .name = "usi-headset"},
 	{.id = ALC662_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"},
 	{}
 };
@@ -8214,18 +8265,20 @@ static int patch_alc662(struct hda_codec *codec)
 	if (!spec->gen.no_analog && spec->gen.beep_nid) {
 		switch (codec->core.vendor_id) {
 		case 0x10ec0662:
-			set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+			err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 			break;
 		case 0x10ec0272:
 		case 0x10ec0663:
 		case 0x10ec0665:
 		case 0x10ec0668:
-			set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
+			err = set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
 			break;
 		case 0x10ec0273:
-			set_beep_amp(spec, 0x0b, 0x03, HDA_INPUT);
+			err = set_beep_amp(spec, 0x0b, 0x03, HDA_INPUT);
 			break;
 		}
+		if (err < 0)
+			goto error;
 	}
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 63d15b545b33..046705b4691a 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -332,33 +332,15 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask,
 }
 
 /* hook for controlling mic-mute LED GPIO */
-static void stac_capture_led_hook(struct hda_codec *codec,
-				  struct snd_kcontrol *kcontrol,
-				  struct snd_ctl_elem_value *ucontrol)
+static void stac_capture_led_update(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	unsigned int mask;
-	bool cur_mute, prev_mute;
 
-	if (!kcontrol || !ucontrol)
-		return;
-
-	mask = 1U << snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-	prev_mute = !spec->mic_enabled;
-	if (ucontrol->value.integer.value[0] ||
-	    ucontrol->value.integer.value[1])
-		spec->mic_enabled |= mask;
+	if (spec->gen.micmute_led.led_value)
+		spec->gpio_data |= spec->mic_mute_led_gpio;
 	else
-		spec->mic_enabled &= ~mask;
-	cur_mute = !spec->mic_enabled;
-	if (cur_mute != prev_mute) {
-		if (cur_mute)
-			spec->gpio_data |= spec->mic_mute_led_gpio;
-		else
-			spec->gpio_data &= ~spec->mic_mute_led_gpio;
-		stac_gpio_set(codec, spec->gpio_mask,
-			      spec->gpio_dir, spec->gpio_data);
-	}
+		spec->gpio_data &= ~spec->mic_mute_led_gpio;
+	stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data);
 }
 
 static int stac_vrefout_set(struct hda_codec *codec,
@@ -4656,8 +4638,7 @@ static void stac_setup_gpio(struct hda_codec *codec)
 		spec->gpio_dir |= spec->mic_mute_led_gpio;
 		spec->mic_enabled = 0;
 		spec->gpio_data |= spec->mic_mute_led_gpio;
-
-		spec->gen.cap_sync_hook = stac_capture_led_hook;
+		snd_hda_gen_add_micmute_led(codec, stac_capture_led_update);
 	}
 }
 
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index fc30d1e8aa76..6b9617aee0e6 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -90,13 +90,6 @@ enum VIA_HDA_CODEC {
 struct via_spec {
 	struct hda_gen_spec gen;
 
-	/* codec parameterization */
-	const struct snd_kcontrol_new *mixers[6];
-	unsigned int num_mixers;
-
-	const struct hda_verb *init_verbs[5];
-	unsigned int num_iverbs;
-
 	/* HP mode source */
 	unsigned int dmic_enabled;
 	enum VIA_HDA_CODEC codec_type;
@@ -107,8 +100,6 @@ struct via_spec {
 	/* work to check hp jack state */
 	int hp_work_active;
 	int vt1708_jack_detect;
-
-	unsigned int beep_amp;
 };
 
 static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec);
@@ -262,69 +253,51 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static const struct snd_kcontrol_new via_pin_power_ctl_enum[] = {
-	{
+static const struct snd_kcontrol_new via_pin_power_ctl_enum = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Dynamic Power-Control",
 	.info = via_pin_power_ctl_info,
 	.get = via_pin_power_ctl_get,
 	.put = via_pin_power_ctl_put,
-	},
-	{} /* terminator */
 };
 
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
-static inline void set_beep_amp(struct via_spec *spec, hda_nid_t nid,
-				int idx, int dir)
-{
-	spec->gen.beep_nid = nid;
-	spec->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir);
-}
-
 /* additional beep mixers; the actual parameters are overwritten at build */
-static const struct snd_kcontrol_new cxt_beep_mixer[] = {
+static const struct snd_kcontrol_new via_beep_mixer[] = {
 	HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0, 1, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_BEEP_MONO("Beep Playback Switch", 0, 1, 0, HDA_OUTPUT),
-	{ } /* end */
 };
 
-/* create beep controls if needed */
-static int add_beep_ctls(struct hda_codec *codec)
+static int set_beep_amp(struct via_spec *spec, hda_nid_t nid,
+			int idx, int dir)
 {
-	struct via_spec *spec = codec->spec;
-	int err;
+	struct snd_kcontrol_new *knew;
+	unsigned int beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir);
+	int i;
 
-	if (spec->beep_amp) {
-		const struct snd_kcontrol_new *knew;
-		for (knew = cxt_beep_mixer; knew->name; knew++) {
-			struct snd_kcontrol *kctl;
-			kctl = snd_ctl_new1(knew, codec);
-			if (!kctl)
-				return -ENOMEM;
-			kctl->private_value = spec->beep_amp;
-			err = snd_hda_ctl_add(codec, 0, kctl);
-			if (err < 0)
-				return err;
-		}
+	spec->gen.beep_nid = nid;
+	for (i = 0; i < ARRAY_SIZE(via_beep_mixer); i++) {
+		knew = snd_hda_gen_add_kctl(&spec->gen, NULL,
+					    &via_beep_mixer[i]);
+		if (!knew)
+			return -ENOMEM;
+		knew->private_value = beep_amp;
 	}
 	return 0;
 }
 
-static void auto_parse_beep(struct hda_codec *codec)
+static int auto_parse_beep(struct hda_codec *codec)
 {
 	struct via_spec *spec = codec->spec;
 	hda_nid_t nid;
 
 	for_each_hda_codec_node(nid, codec)
-		if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP) {
-			set_beep_amp(spec, nid, 0, HDA_OUTPUT);
-			break;
-		}
+		if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP)
+			return set_beep_amp(spec, nid, 0, HDA_OUTPUT);
+	return 0;
 }
 #else
-#define set_beep_amp(spec, nid, idx, dir) /* NOP */
-#define add_beep_ctls(codec)	0
-#define auto_parse_beep(codec)
+#define auto_parse_beep(codec)	0
 #endif
 
 /* check AA path's mute status */
@@ -403,30 +376,6 @@ static void analog_low_current_mode(struct hda_codec *codec)
 	return __analog_low_current_mode(codec, false);
 }
 
-static int via_build_controls(struct hda_codec *codec)
-{
-	struct via_spec *spec = codec->spec;
-	int err, i;
-
-	err = snd_hda_gen_build_controls(codec);
-	if (err < 0)
-		return err;
-
-	err = add_beep_ctls(codec);
-	if (err < 0)
-		return err;
-
-	spec->mixers[spec->num_mixers++] = via_pin_power_ctl_enum;
-
-	for (i = 0; i < spec->num_mixers; i++) {
-		err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 static void via_playback_pcm_hook(struct hda_pcm_stream *hinfo,
 				  struct hda_codec *codec,
 				  struct snd_pcm_substream *substream,
@@ -481,7 +430,7 @@ static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid)
 static int via_init(struct hda_codec *codec);
 
 static const struct hda_codec_ops via_patch_ops = {
-	.build_controls = via_build_controls,
+	.build_controls = snd_hda_gen_build_controls,
 	.build_pcms = snd_hda_gen_build_pcms,
 	.init = via_init,
 	.free = via_free,
@@ -545,16 +494,13 @@ static int vt1708_jack_detect_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static const struct snd_kcontrol_new vt1708_jack_detect_ctl[] = {
-	{
+static const struct snd_kcontrol_new vt1708_jack_detect_ctl = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Jack Detect",
 	.count = 1,
 	.info = snd_ctl_boolean_mono_info,
 	.get = vt1708_jack_detect_get,
 	.put = vt1708_jack_detect_put,
-	},
-	{} /* terminator */
 };
 
 static const struct badness_table via_main_out_badness = {
@@ -586,12 +532,17 @@ static int via_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
-	auto_parse_beep(codec);
-
 	err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg);
 	if (err < 0)
 		return err;
 
+	err = auto_parse_beep(codec);
+	if (err < 0)
+		return err;
+
+	if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &via_pin_power_ctl_enum))
+		return -ENOMEM;
+
 	/* disable widget PM at start for compatibility */
 	codec->power_save_node = 0;
 	spec->gen.power_down_unused = 0;
@@ -600,12 +551,6 @@ static int via_parse_auto_config(struct hda_codec *codec)
 
 static int via_init(struct hda_codec *codec)
 {
-	struct via_spec *spec = codec->spec;
-	int i;
-
-	for (i = 0; i < spec->num_iverbs; i++)
-		snd_hda_sequence_write(codec, spec->init_verbs[i]);
-
 	/* init power states */
 	__analog_low_current_mode(codec, true);
 
@@ -623,7 +568,7 @@ static int vt1708_build_controls(struct hda_codec *codec)
 	int err;
 	int old_interval = codec->jackpoll_interval;
 	codec->jackpoll_interval = msecs_to_jiffies(100);
-	err = via_build_controls(codec);
+	err = snd_hda_gen_build_controls(codec);
 	codec->jackpoll_interval = old_interval;
 	return err;
 }
@@ -684,22 +629,29 @@ static int patch_vt1708(struct hda_codec *codec)
 	vt1708_set_pinconfig_connect(codec, VT1708_HP_PIN_NID);
 	vt1708_set_pinconfig_connect(codec, VT1708_CD_PIN_NID);
 
+	err = snd_hda_add_verbs(codec, vt1708_init_verbs);
+	if (err < 0)
+		goto error;
+
 	/* automatic parse from the BIOS config */
 	err = via_parse_auto_config(codec);
-	if (err < 0) {
-		via_free(codec);
-		return err;
-	}
+	if (err < 0)
+		goto error;
 
 	/* add jack detect on/off control */
-	spec->mixers[spec->num_mixers++] = vt1708_jack_detect_ctl;
-
-	spec->init_verbs[spec->num_iverbs++] = vt1708_init_verbs;
+	if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1708_jack_detect_ctl)) {
+		err = -ENOMEM;
+		goto error;
+	}
 
 	/* clear jackpoll_interval again; it's set dynamically */
 	codec->jackpoll_interval = 0;
 
 	return 0;
+
+ error:
+	via_free(codec);
+	return err;
 }
 
 static int patch_vt1709(struct hda_codec *codec)
@@ -715,12 +667,14 @@ static int patch_vt1709(struct hda_codec *codec)
 	spec->gen.mixer_nid = 0x18;
 
 	err = via_parse_auto_config(codec);
-	if (err < 0) {
-		via_free(codec);
-		return err;
-	}
+	if (err < 0)
+		goto error;
 
 	return 0;
+
+ error:
+	via_free(codec);
+	return err;
 }
 
 static int patch_vt1708S(struct hda_codec *codec);
@@ -741,12 +695,14 @@ static int patch_vt1708B(struct hda_codec *codec)
 
 	/* automatic parse from the BIOS config */
 	err = via_parse_auto_config(codec);
-	if (err < 0) {
-		via_free(codec);
-		return err;
-	}
+	if (err < 0)
+		goto error;
 
 	return 0;
+
+ error:
+	via_free(codec);
+	return err;
 }
 
 /* Patch for VT1708S */
@@ -791,16 +747,20 @@ static int patch_vt1708S(struct hda_codec *codec)
 	if (codec->core.vendor_id == 0x11064397)
 		snd_hda_codec_set_name(codec, "VT1705");
 
+	err = snd_hda_add_verbs(codec, vt1708S_init_verbs);
+	if (err < 0)
+		goto error;
+
 	/* automatic parse from the BIOS config */
 	err = via_parse_auto_config(codec);
-	if (err < 0) {
-		via_free(codec);
-		return err;
-	}
-
-	spec->init_verbs[spec->num_iverbs++] = vt1708S_init_verbs;
+	if (err < 0)
+		goto error;
 
 	return 0;
+
+ error:
+	via_free(codec);
+	return err;
 }
 
 /* Patch for VT1702 */
@@ -832,16 +792,20 @@ static int patch_vt1702(struct hda_codec *codec)
 				  (0x5 << AC_AMPCAP_STEP_SIZE_SHIFT) |
 				  (1 << AC_AMPCAP_MUTE_SHIFT));
 
+	err = snd_hda_add_verbs(codec, vt1702_init_verbs);
+	if (err < 0)
+		goto error;
+
 	/* automatic parse from the BIOS config */
 	err = via_parse_auto_config(codec);
-	if (err < 0) {
-		via_free(codec);
-		return err;
-	}
-
-	spec->init_verbs[spec->num_iverbs++] = vt1702_init_verbs;
+	if (err < 0)
+		goto error;
 
 	return 0;
+
+ error:
+	via_free(codec);
+	return err;
 }
 
 /* Patch for VT1718S */
@@ -904,16 +868,20 @@ static int patch_vt1718S(struct hda_codec *codec)
 	override_mic_boost(codec, 0x29, 0, 3, 40);
 	add_secret_dac_path(codec);
 
+	err = snd_hda_add_verbs(codec, vt1718S_init_verbs);
+	if (err < 0)
+		goto error;
+
 	/* automatic parse from the BIOS config */
 	err = via_parse_auto_config(codec);
-	if (err < 0) {
-		via_free(codec);
-		return err;
-	}
-
-	spec->init_verbs[spec->num_iverbs++] = vt1718S_init_verbs;
+	if (err < 0)
+		goto error;
 
 	return 0;
+
+ error:
+	via_free(codec);
+	return err;
 }
 
 /* Patch for VT1716S */
@@ -955,9 +923,9 @@ static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static const struct snd_kcontrol_new vt1716s_dmic_mixer[] = {
-	HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT),
-	{
+static const struct snd_kcontrol_new vt1716s_dmic_mixer_vol =
+	HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT);
+static const struct snd_kcontrol_new vt1716s_dmic_mixer_sw = {
 	 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	 .name = "Digital Mic Capture Switch",
 	 .subdevice = HDA_SUBDEV_NID_FLAG | 0x26,
@@ -965,16 +933,12 @@ static const struct snd_kcontrol_new vt1716s_dmic_mixer[] = {
 	 .info = vt1716s_dmic_info,
 	 .get = vt1716s_dmic_get,
 	 .put = vt1716s_dmic_put,
-	 },
-	{}			/* end */
 };
 
 
 /* mono-out mixer elements */
-static const struct snd_kcontrol_new vt1716S_mono_out_mixer[] = {
-	HDA_CODEC_MUTE("Mono Playback Switch", 0x2a, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
+static const struct snd_kcontrol_new vt1716S_mono_out_mixer =
+	HDA_CODEC_MUTE("Mono Playback Switch", 0x2a, 0x0, HDA_OUTPUT);
 
 static const struct hda_verb vt1716S_init_verbs[] = {
 	/* Enable Boost Volume backdoor */
@@ -1000,19 +964,27 @@ static int patch_vt1716S(struct hda_codec *codec)
 	override_mic_boost(codec, 0x1a, 0, 3, 40);
 	override_mic_boost(codec, 0x1e, 0, 3, 40);
 
+	err = snd_hda_add_verbs(codec, vt1716S_init_verbs);
+	if (err < 0)
+		goto error;
+
 	/* automatic parse from the BIOS config */
 	err = via_parse_auto_config(codec);
-	if (err < 0) {
-		via_free(codec);
-		return err;
-	}
-
-	spec->init_verbs[spec->num_iverbs++]  = vt1716S_init_verbs;
+	if (err < 0)
+		goto error;
 
-	spec->mixers[spec->num_mixers++] = vt1716s_dmic_mixer;
-	spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer;
+	if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1716s_dmic_mixer_vol) ||
+	    !snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1716s_dmic_mixer_sw) ||
+	    !snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1716S_mono_out_mixer)) {
+		err = -ENOMEM;
+		goto error;
+	}
 
 	return 0;
+
+ error:
+	via_free(codec);
+	return err;
 }
 
 /* for vt2002P */
@@ -1107,19 +1079,23 @@ static int patch_vt2002P(struct hda_codec *codec)
 	snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
-	/* automatic parse from the BIOS config */
-	err = via_parse_auto_config(codec);
-	if (err < 0) {
-		via_free(codec);
-		return err;
-	}
-
 	if (spec->codec_type == VT1802)
-		spec->init_verbs[spec->num_iverbs++] = vt1802_init_verbs;
+		err = snd_hda_add_verbs(codec, vt1802_init_verbs);
 	else
-		spec->init_verbs[spec->num_iverbs++] = vt2002P_init_verbs;
+		err = snd_hda_add_verbs(codec, vt2002P_init_verbs);
+	if (err < 0)
+		goto error;
+
+	/* automatic parse from the BIOS config */
+	err = via_parse_auto_config(codec);
+	if (err < 0)
+		goto error;
 
 	return 0;
+
+ error:
+	via_free(codec);
+	return err;
 }
 
 /* for vt1812 */
@@ -1148,16 +1124,20 @@ static int patch_vt1812(struct hda_codec *codec)
 	override_mic_boost(codec, 0x29, 0, 3, 40);
 	add_secret_dac_path(codec);
 
+	err = snd_hda_add_verbs(codec, vt1812_init_verbs);
+	if (err < 0)
+		goto error;
+
 	/* automatic parse from the BIOS config */
 	err = via_parse_auto_config(codec);
-	if (err < 0) {
-		via_free(codec);
-		return err;
-	}
-
-	spec->init_verbs[spec->num_iverbs++]  = vt1812_init_verbs;
+	if (err < 0)
+		goto error;
 
 	return 0;
+
+ error:
+	via_free(codec);
+	return err;
 }
 
 /* patch for vt3476 */
@@ -1185,16 +1165,20 @@ static int patch_vt3476(struct hda_codec *codec)
 	spec->gen.mixer_nid = 0x3f;
 	add_secret_dac_path(codec);
 
+	err = snd_hda_add_verbs(codec, vt3476_init_verbs);
+	if (err < 0)
+		goto error;
+
 	/* automatic parse from the BIOS config */
 	err = via_parse_auto_config(codec);
-	if (err < 0) {
-		via_free(codec);
-		return err;
-	}
-
-	spec->init_verbs[spec->num_iverbs++] = vt3476_init_verbs;
+	if (err < 0)
+		goto error;
 
 	return 0;
+
+ error:
+	via_free(codec);
+	return err;
 }
 
 /*
diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c
index 65bb3ac6af4c..97f49b751e6e 100644
--- a/sound/pci/hda/thinkpad_helper.c
+++ b/sound/pci/hda/thinkpad_helper.c
@@ -27,17 +27,11 @@ static void update_tpacpi_mute_led(void *private_data, int enabled)
 		led_set_func(TPACPI_LED_MUTE, !enabled);
 }
 
-static void update_tpacpi_micmute_led(struct hda_codec *codec,
-				      struct snd_kcontrol *kcontrol,
-				      struct snd_ctl_elem_value *ucontrol)
+static void update_tpacpi_micmute(struct hda_codec *codec)
 {
-	if (!ucontrol || !led_set_func)
-		return;
-	if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) {
-		/* TODO: How do I verify if it's a mono or stereo here? */
-		bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1];
-		led_set_func(TPACPI_LED_MICMUTE, !val);
-	}
+	struct hda_gen_spec *spec = codec->spec;
+
+	led_set_func(TPACPI_LED_MICMUTE, spec->micmute_led.led_value);
 }
 
 static void hda_fixup_thinkpad_acpi(struct hda_codec *codec,
@@ -63,15 +57,10 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec,
 			spec->vmaster_mute.hook = update_tpacpi_mute_led;
 			removefunc = false;
 		}
-		if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) {
-			if (spec->num_adc_nids > 1 && !spec->dyn_adc_switch)
-				codec_dbg(codec,
-					  "Skipping micmute LED control due to several ADCs");
-			else {
-				spec->cap_sync_hook = update_tpacpi_micmute_led;
-				removefunc = false;
-			}
-		}
+		if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0 &&
+		    snd_hda_gen_add_micmute_led(codec,
+						update_tpacpi_micmute) > 0)
+			removefunc = false;
 	}
 
 	if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) {
diff --git a/sound/pci/ice1712/ak4xxx.c b/sound/pci/ice1712/ak4xxx.c
index 179ef7a5f0d1..a553897a4c4f 100644
--- a/sound/pci/ice1712/ak4xxx.c
+++ b/sound/pci/ice1712/ak4xxx.c
@@ -179,18 +179,6 @@ int snd_ice1712_akm4xxx_build_controls(struct snd_ice1712 *ice)
 	return 0;
 }
 
-static int __init alsa_ice1712_akm4xxx_module_init(void)
-{
-	return 0;
-}
-        
-static void __exit alsa_ice1712_akm4xxx_module_exit(void)
-{
-}
-        
-module_init(alsa_ice1712_akm4xxx_module_init)
-module_exit(alsa_ice1712_akm4xxx_module_exit)
-
 EXPORT_SYMBOL(snd_ice1712_akm4xxx_init);
 EXPORT_SYMBOL(snd_ice1712_akm4xxx_free);
 EXPORT_SYMBOL(snd_ice1712_akm4xxx_build_controls);
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index d7366ade5a25..c97b5528e4b8 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -314,26 +314,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] = {
 
 /* --------------- */
 
-/*
- * Logarithmic volume values for WM87*6
- * Computed as 20 * Log10(255 / x)
- */
-static const unsigned char wm_vol[256] = {
-	127, 48, 42, 39, 36, 34, 33, 31, 30, 29, 28, 27, 27, 26, 25, 25, 24, 24, 23,
-	23, 22, 22, 21, 21, 21, 20, 20, 20, 19, 19, 19, 18, 18, 18, 18, 17, 17, 17,
-	17, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 14, 14, 14, 13, 13, 13,
-	13, 13, 13, 13, 12, 12, 12, 12, 12, 12, 12, 11, 11, 11, 11, 11, 11, 11, 11,
-	11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 8, 8,
-	8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 6, 6, 6,
-	6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
-	5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3,
-	3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
-	2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 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, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
-	0, 0
-};
-
-#define WM_VOL_MAX	(sizeof(wm_vol) - 1)
+#define WM_VOL_MAX	255
 #define WM_VOL_MUTE	0x8000
 
 
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 4c24346340f4..5ee468d1aefe 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -351,7 +351,7 @@ enum {
 struct ichdev {
 	unsigned int ichd;			/* ich device number */
 	unsigned long reg_offset;		/* offset to bmaddr */
-	u32 *bdbar;				/* CPU address (32bit) */
+	__le32 *bdbar;				/* CPU address (32bit) */
 	unsigned int bdbar_addr;		/* PCI bus address (32bit) */
 	struct snd_pcm_substream *substream;
 	unsigned int physbuf;			/* physical address (32bit) */
@@ -677,7 +677,7 @@ static void snd_intel8x0_ali_codec_write(struct snd_ac97 *ac97, unsigned short r
 static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ichdev) 
 {
 	int idx;
-	u32 *bdbar = ichdev->bdbar;
+	__le32 *bdbar = ichdev->bdbar;
 	unsigned long port = ichdev->reg_offset;
 
 	iputdword(chip, port + ICH_REG_OFF_BDBAR, ichdev->bdbar_addr);
@@ -3143,7 +3143,7 @@ static int snd_intel8x0_create(struct snd_card *card,
 	int_sta_masks = 0;
 	for (i = 0; i < chip->bdbars_count; i++) {
 		ichdev = &chip->ichd[i];
-		ichdev->bdbar = ((u32 *)chip->bdbars.area) +
+		ichdev->bdbar = ((__le32 *)chip->bdbars.area) +
 			(i * ICH_MAX_FRAGS * 2);
 		ichdev->bdbar_addr = chip->bdbars.addr +
 			(i * sizeof(u32) * ICH_MAX_FRAGS * 2);
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 3a4769a97d29..943a726b1c1b 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -168,7 +168,7 @@ enum { ALID_MDMIN, ALID_MDMOUT, ALID_MDMLAST = ALID_MDMOUT };
 struct ichdev {
 	unsigned int ichd;			/* ich device number */
 	unsigned long reg_offset;		/* offset to bmaddr */
-	u32 *bdbar;				/* CPU address (32bit) */
+	__le32 *bdbar;				/* CPU address (32bit) */
 	unsigned int bdbar_addr;		/* PCI bus address (32bit) */
 	struct snd_pcm_substream *substream;
 	unsigned int physbuf;			/* physical address (32bit) */
@@ -395,7 +395,7 @@ static unsigned short snd_intel8x0m_codec_read(struct snd_ac97 *ac97,
 static void snd_intel8x0m_setup_periods(struct intel8x0m *chip, struct ichdev *ichdev)
 {
 	int idx;
-	u32 *bdbar = ichdev->bdbar;
+	__le32 *bdbar = ichdev->bdbar;
 	unsigned long port = ichdev->reg_offset;
 
 	iputdword(chip, port + ICH_REG_OFF_BDBAR, ichdev->bdbar_addr);
@@ -1217,7 +1217,7 @@ static int snd_intel8x0m_create(struct snd_card *card,
 	int_sta_masks = 0;
 	for (i = 0; i < chip->bdbars_count; i++) {
 		ichdev = &chip->ichd[i];
-		ichdev->bdbar = ((u32 *)chip->bdbars.area) + (i * ICH_MAX_FRAGS * 2);
+		ichdev->bdbar = ((__le32 *)chip->bdbars.area) + (i * ICH_MAX_FRAGS * 2);
 		ichdev->bdbar_addr = chip->bdbars.addr + (i * sizeof(u32) * ICH_MAX_FRAGS * 2);
 		int_sta_masks |= ichdev->int_sta_mask;
 	}
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 4206ba44d8bb..4e189a93f475 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -1326,7 +1326,7 @@ static int snd_korg1212_copy_to(struct snd_pcm_substream *substream,
 		}
 #endif
 		if (in_kernel)
-			memcpy((void *)dst, src, size);
+			memcpy((__force void *)dst, src, size);
 		else if (copy_to_user(dst, src, size))
 			return -EFAULT;
 		src++;
@@ -1365,7 +1365,7 @@ static int snd_korg1212_copy_from(struct snd_pcm_substream *substream,
 		}
 #endif
 		if (in_kernel)
-			memcpy((void *)dst, src, size);
+			memcpy(dst, (__force void *)src, size);
 		else if (copy_from_user(dst, src, size))
 			return -EFAULT;
 		dst++;
diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c
index 9ff600084973..254f24366892 100644
--- a/sound/pci/lola/lola.c
+++ b/sound/pci/lola/lola.c
@@ -369,9 +369,9 @@ static int setup_corb_rirb(struct lola *chip)
 		return err;
 
 	chip->corb.addr = chip->rb.addr;
-	chip->corb.buf = (u32 *)chip->rb.area;
+	chip->corb.buf = (__le32 *)chip->rb.area;
 	chip->rirb.addr = chip->rb.addr + 2048;
-	chip->rirb.buf = (u32 *)(chip->rb.area + 2048);
+	chip->rirb.buf = (__le32 *)(chip->rb.area + 2048);
 
 	/* disable ringbuffer DMAs */
 	lola_writeb(chip, BAR0, RIRBCTL, 0);
diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h
index f0b100059efd..bd852fed8bb6 100644
--- a/sound/pci/lola/lola.h
+++ b/sound/pci/lola/lola.h
@@ -220,7 +220,7 @@ struct lola_bar {
 
 /* CORB/RIRB */
 struct lola_rb {
-	u32 *buf;		/* CORB/RIRB buffer, 8 byte per each entry */
+	__le32 *buf;		/* CORB/RIRB buffer, 8 byte per each entry */
 	dma_addr_t addr;	/* physical address of CORB/RIRB buffer */
 	unsigned short rp, wp;	/* read/write pointers */
 	int cmds;		/* number of pending requests */
@@ -275,7 +275,7 @@ struct lola_mixer_array {
 struct lola_mixer_widget {
 	unsigned int nid;
 	unsigned int caps;
-	struct lola_mixer_array __user *array;
+	struct lola_mixer_array __iomem *array;
 	struct lola_mixer_array *array_saved;
 	unsigned int src_stream_outs;
 	unsigned int src_phys_ins;
diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c
index 310b26a756c9..e70276c3ea20 100644
--- a/sound/pci/lola/lola_pcm.c
+++ b/sound/pci/lola/lola_pcm.c
@@ -316,10 +316,10 @@ static int lola_pcm_hw_free(struct snd_pcm_substream *substream)
  * set up a BDL entry
  */
 static int setup_bdle(struct snd_pcm_substream *substream,
-		      struct lola_stream *str, u32 **bdlp,
+		      struct lola_stream *str, __le32 **bdlp,
 		      int ofs, int size)
 {
-	u32 *bdl = *bdlp;
+	__le32 *bdl = *bdlp;
 
 	while (size > 0) {
 		dma_addr_t addr;
@@ -355,14 +355,14 @@ static int lola_setup_periods(struct lola *chip, struct lola_pcm *pcm,
 			      struct snd_pcm_substream *substream,
 			      struct lola_stream *str)
 {
-	u32 *bdl;
+	__le32 *bdl;
 	int i, ofs, periods, period_bytes;
 
 	period_bytes = str->period_bytes;
 	periods = str->bufsize / period_bytes;
 
 	/* program the initial BDL entries */
-	bdl = (u32 *)(pcm->bdl.area + LOLA_BDL_ENTRY_SIZE * str->index);
+	bdl = (__le32 *)(pcm->bdl.area + LOLA_BDL_ENTRY_SIZE * str->index);
 	ofs = 0;
 	str->frags = 0;
 	for (i = 0; i < periods; i++) {
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 224e942f556d..62962178a9d7 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -2103,7 +2103,7 @@ static const u16 minisrc_lpf[MINISRC_LPF_LEN] = {
 static void snd_m3_assp_init(struct snd_m3 *chip)
 {
 	unsigned int i;
-	const u16 *data;
+	const __le16 *data;
 
 	/* zero kernel data */
 	for (i = 0; i < (REV_B_DATA_MEMORY_UNIT_LENGTH * NUM_UNITS_KERNEL_DATA) / 2; i++)
@@ -2121,7 +2121,7 @@ static void snd_m3_assp_init(struct snd_m3 *chip)
 			  KDATA_DMA_XFER0);
 
 	/* write kernel into code memory.. */
-	data = (const u16 *)chip->assp_kernel_image->data;
+	data = (const __le16 *)chip->assp_kernel_image->data;
 	for (i = 0 ; i * 2 < chip->assp_kernel_image->size; i++) {
 		snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, 
 				  REV_B_CODE_MEMORY_BEGIN + i,
@@ -2134,7 +2134,7 @@ static void snd_m3_assp_init(struct snd_m3 *chip)
 	 * drop it there.  It seems that the minisrc doesn't
 	 * need vectors, so we won't bother with them..
 	 */
-	data = (const u16 *)chip->assp_minisrc_image->data;
+	data = (const __le16 *)chip->assp_minisrc_image->data;
 	for (i = 0; i * 2 < chip->assp_minisrc_image->size; i++) {
 		snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, 
 				  0x400 + i, le16_to_cpu(data[i]));
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index a74f1ad7e7b8..9cd297a42f24 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -182,6 +182,7 @@ static int mixart_set_clock(struct mixart_mgr *mgr,
 	case PIPE_RUNNING:
 		if(rate != 0)
 			break;
+		/* fall through */
 	default:
 		if(rate == 0)
 			return 0; /* nothing to do */
diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c
index 8bf2ce32d4a8..71776bfe0485 100644
--- a/sound/pci/mixart/mixart_core.c
+++ b/sound/pci/mixart/mixart_core.c
@@ -107,7 +107,7 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp,
 #ifndef __BIG_ENDIAN
 	size /= 4; /* u32 size */
 	for(i=0; i < size; i++) {
-		((u32*)resp->data)[i] = be32_to_cpu(((u32*)resp->data)[i]);
+		((u32*)resp->data)[i] = be32_to_cpu(((__be32*)resp->data)[i]);
 	}
 #endif
 
@@ -519,7 +519,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id)
 					/* Traces are text: the swapped msg_data has to be swapped back ! */
 					int i;
 					for(i=0; i<(resp.size/4); i++) {
-						(mixart_msg_data)[i] = cpu_to_be32((mixart_msg_data)[i]);
+						((__be32*)mixart_msg_data)[i] = cpu_to_be32((mixart_msg_data)[i]);
 					}
 #endif
 					((char*)mixart_msg_data)[resp.size - 1] = 0;
@@ -540,7 +540,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id)
 				dev_err(&mgr->pci->dev,
 					"canceled notification %x !\n", msg);
 			}
-			/* no break, continue ! */
+			/* fall through */
 		case MSG_TYPE_ANSWER:
 			/* answer or notification to a message we are waiting for*/
 			mutex_lock(&mgr->msg_lock);
diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c
index 5bfd3ac80db5..bc92758de82c 100644
--- a/sound/pci/mixart/mixart_hwdep.c
+++ b/sound/pci/mixart/mixart_hwdep.c
@@ -73,30 +73,30 @@ static int mixart_wait_nice_for_register_value(struct mixart_mgr *mgr,
  */
 struct snd_mixart_elf32_ehdr {
 	u8      e_ident[16];
-	u16     e_type;
-	u16     e_machine;
-	u32     e_version;
-	u32     e_entry;
-	u32     e_phoff;
-	u32     e_shoff;
-	u32     e_flags;
-	u16     e_ehsize;
-	u16     e_phentsize;
-	u16     e_phnum;
-	u16     e_shentsize;
-	u16     e_shnum;
-	u16     e_shstrndx;
+	__be16  e_type;
+	__be16  e_machine;
+	__be32  e_version;
+	__be32  e_entry;
+	__be32  e_phoff;
+	__be32  e_shoff;
+	__be32  e_flags;
+	__be16  e_ehsize;
+	__be16  e_phentsize;
+	__be16  e_phnum;
+	__be16  e_shentsize;
+	__be16  e_shnum;
+	__be16  e_shstrndx;
 };
 
 struct snd_mixart_elf32_phdr {
-	u32     p_type;
-	u32     p_offset;
-	u32     p_vaddr;
-	u32     p_paddr;
-	u32     p_filesz;
-	u32     p_memsz;
-	u32     p_flags;
-	u32     p_align;
+	__be32  p_type;
+	__be32  p_offset;
+	__be32  p_vaddr;
+	__be32  p_paddr;
+	__be32  p_filesz;
+	__be32  p_memsz;
+	__be32  p_flags;
+	__be32  p_align;
 };
 
 static int mixart_load_elf(struct mixart_mgr *mgr, const struct firmware *dsp )
diff --git a/sound/pci/mixart/mixart_hwdep.h b/sound/pci/mixart/mixart_hwdep.h
index 812e288ef2e7..2794cd385b8e 100644
--- a/sound/pci/mixart/mixart_hwdep.h
+++ b/sound/pci/mixart/mixart_hwdep.h
@@ -26,19 +26,19 @@
 #include <sound/hwdep.h>
 
 #ifndef readl_be
-#define readl_be(x) be32_to_cpu(__raw_readl(x))
+#define readl_be(x) be32_to_cpu((__force __be32)__raw_readl(x))
 #endif
 
 #ifndef writel_be
-#define writel_be(data,addr) __raw_writel(cpu_to_be32(data),addr)
+#define writel_be(data,addr) __raw_writel((__force u32)cpu_to_be32(data),addr)
 #endif
 
 #ifndef readl_le
-#define readl_le(x) le32_to_cpu(__raw_readl(x))
+#define readl_le(x) le32_to_cpu((__force __le32)__raw_readl(x))
 #endif
 
 #ifndef writel_le
-#define writel_le(data,addr) __raw_writel(cpu_to_le32(data),addr)
+#define writel_le(data,addr) __raw_writel((__force u32)cpu_to_le32(data),addr)
 #endif
 
 #define MIXART_MEM(mgr,x)	((mgr)->mem[0].virt + (x))
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 44f3b48d47b1..23017e3bc76c 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -470,10 +470,10 @@ struct snd_riptide {
 };
 
 struct sgd {			/* scatter gather desriptor */
-	u32 dwNextLink;
-	u32 dwSegPtrPhys;
-	u32 dwSegLen;
-	u32 dwStat_Ctl;
+	__le32 dwNextLink;
+	__le32 dwSegPtrPhys;
+	__le32 dwSegLen;
+	__le32 dwStat_Ctl;
 };
 
 struct pcmhw {			/* pcm descriptor */
@@ -1017,7 +1017,7 @@ getsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int *rate)
 static int
 setsampleformat(struct cmdif *cif,
 		unsigned char mixer, unsigned char id,
-		unsigned char channels, unsigned char format)
+		unsigned char channels, snd_pcm_format_t format)
 {
 	unsigned char w, ch, sig, order;
 
diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
index 7fbdb703bfcd..7218f38b59db 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -1433,14 +1433,12 @@ static int snd_sonicvibes_midi(struct sonicvibes *sonic,
 {
 	struct snd_mpu401 * mpu = rmidi->private_data;
 	struct snd_card *card = sonic->card;
-	struct snd_rawmidi_str *dir;
 	unsigned int idx;
 	int err;
 
 	mpu->private_data = sonic;
 	mpu->open_input = snd_sonicvibes_midi_input_open;
 	mpu->close_input = snd_sonicvibes_midi_input_close;
-	dir = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
 	for (idx = 0; idx < ARRAY_SIZE(snd_sonicvibes_midi_controls); idx++)
 		if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_sonicvibes_midi_controls[idx], sonic))) < 0)
 			return err;
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index cedf13b64803..2f18b1cdc2cd 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -123,7 +123,7 @@ static int snd_trident_probe(struct pci_dev *pci,
 	} else {
 		strcpy(card->shortname, "Trident ");
 	}
-	strcat(card->shortname, card->driver);
+	strcat(card->shortname, str);
 	sprintf(card->longname, "%s PCI Audio at 0x%lx, irq %d",
 		card->shortname, trident->port, trident->irq);
 
diff --git a/sound/pci/trident/trident.h b/sound/pci/trident/trident.h
index 9624e5937719..2d62c1921255 100644
--- a/sound/pci/trident/trident.h
+++ b/sound/pci/trident/trident.h
@@ -264,7 +264,7 @@ struct snd_trident_memblk_arg {
 };
 
 struct snd_trident_tlb {
-	unsigned int * entries;		/* 16k-aligned TLB table */
+	__le32 *entries;		/* 16k-aligned TLB table */
 	dma_addr_t entries_dmaaddr;	/* 16k-aligned PCI address to TLB table */
 	unsigned long * shadow_entries;	/* shadow entries with virtual addresses */
 	struct snd_dma_buffer buffer;
diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index 49c64fae3466..5523e193d556 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -3359,7 +3359,7 @@ static int snd_trident_tlb_alloc(struct snd_trident *trident)
 		dev_err(trident->card->dev, "unable to allocate TLB buffer\n");
 		return -ENOMEM;
 	}
-	trident->tlb.entries = (unsigned int*)ALIGN((unsigned long)trident->tlb.buffer.area, SNDRV_TRIDENT_MAX_PAGES * 4);
+	trident->tlb.entries = (__le32 *)ALIGN((unsigned long)trident->tlb.buffer.area, SNDRV_TRIDENT_MAX_PAGES * 4);
 	trident->tlb.entries_dmaaddr = ALIGN(trident->tlb.buffer.addr, SNDRV_TRIDENT_MAX_PAGES * 4);
 	/* allocate shadow TLB page table (virtual addresses) */
 	trident->tlb.shadow_entries =
diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c
index d4298af6d3ee..c0d0bf44f365 100644
--- a/sound/pci/vx222/vx222_ops.c
+++ b/sound/pci/vx222/vx222_ops.c
@@ -275,7 +275,7 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime,
 		length >>= 2; /* in 32bit words */
 		/* Transfer using pseudo-dma. */
 		for (; length > 0; length--) {
-			outl(cpu_to_le32(*addr), port);
+			outl(*addr, port);
 			addr++;
 		}
 		addr = (u32 *)runtime->dma_area;
@@ -285,7 +285,7 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime,
 	count >>= 2; /* in 32bit words */
 	/* Transfer using pseudo-dma. */
 	for (; count > 0; count--) {
-		outl(cpu_to_le32(*addr), port);
+		outl(*addr, port);
 		addr++;
 	}
 
@@ -313,7 +313,7 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime,
 		length >>= 2; /* in 32bit words */
 		/* Transfer using pseudo-dma. */
 		for (; length > 0; length--)
-			*addr++ = le32_to_cpu(inl(port));
+			*addr++ = inl(port);
 		addr = (u32 *)runtime->dma_area;
 		pipe->hw_ptr = 0;
 	}
@@ -321,7 +321,7 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime,
 	count >>= 2; /* in 32bit words */
 	/* Transfer using pseudo-dma. */
 	for (; count > 0; count--)
-		*addr++ = le32_to_cpu(inl(port));
+		*addr++ = inl(port);
 
 	vx2_release_pseudo_dma(chip);
 }
diff --git a/sound/pci/ymfpci/ymfpci.h b/sound/pci/ymfpci/ymfpci.h
index aa9bb065f385..e2fa7e360d79 100644
--- a/sound/pci/ymfpci/ymfpci.h
+++ b/sound/pci/ymfpci/ymfpci.h
@@ -185,50 +185,50 @@
  */
 
 struct snd_ymfpci_playback_bank {
-	u32 format;
-	u32 loop_default;
-	u32 base;			/* 32-bit address */
-	u32 loop_start;			/* 32-bit offset */
-	u32 loop_end;			/* 32-bit offset */
-	u32 loop_frac;			/* 8-bit fraction - loop_start */
-	u32 delta_end;			/* pitch delta end */
-	u32 lpfK_end;
-	u32 eg_gain_end;
-	u32 left_gain_end;
-	u32 right_gain_end;
-	u32 eff1_gain_end;
-	u32 eff2_gain_end;
-	u32 eff3_gain_end;
-	u32 lpfQ;
-	u32 status;
-	u32 num_of_frames;
-	u32 loop_count;
-	u32 start;
-	u32 start_frac;
-	u32 delta;
-	u32 lpfK;
-	u32 eg_gain;
-	u32 left_gain;
-	u32 right_gain;
-	u32 eff1_gain;
-	u32 eff2_gain;
-	u32 eff3_gain;
-	u32 lpfD1;
-	u32 lpfD2;
+	__le32 format;
+	__le32 loop_default;
+	__le32 base;			/* 32-bit address */
+	__le32 loop_start;		/* 32-bit offset */
+	__le32 loop_end;		/* 32-bit offset */
+	__le32 loop_frac;		/* 8-bit fraction - loop_start */
+	__le32 delta_end;		/* pitch delta end */
+	__le32 lpfK_end;
+	__le32 eg_gain_end;
+	__le32 left_gain_end;
+	__le32 right_gain_end;
+	__le32 eff1_gain_end;
+	__le32 eff2_gain_end;
+	__le32 eff3_gain_end;
+	__le32 lpfQ;
+	__le32 status;
+	__le32 num_of_frames;
+	__le32 loop_count;
+	__le32 start;
+	__le32 start_frac;
+	__le32 delta;
+	__le32 lpfK;
+	__le32 eg_gain;
+	__le32 left_gain;
+	__le32 right_gain;
+	__le32 eff1_gain;
+	__le32 eff2_gain;
+	__le32 eff3_gain;
+	__le32 lpfD1;
+	__le32 lpfD2;
  };
 
 struct snd_ymfpci_capture_bank {
-	u32 base;			/* 32-bit address */
-	u32 loop_end;			/* 32-bit offset */
-	u32 start;			/* 32-bit offset */
-	u32 num_of_loops;		/* counter */
+	__le32 base;			/* 32-bit address */
+	__le32 loop_end;		/* 32-bit offset */
+	__le32 start;			/* 32-bit offset */
+	__le32 num_of_loops;		/* counter */
 };
 
 struct snd_ymfpci_effect_bank {
-	u32 base;			/* 32-bit address */
-	u32 loop_end;			/* 32-bit offset */
-	u32 start;			/* 32-bit offset */
-	u32 temp;
+	__le32 base;			/* 32-bit address */
+	__le32 loop_end;		/* 32-bit offset */
+	__le32 start;			/* 32-bit offset */
+	__le32 temp;
 };
 
 struct snd_ymfpci_pcm;
@@ -316,7 +316,7 @@ struct snd_ymfpci {
 	dma_addr_t work_base_addr;
 	struct snd_dma_buffer ac3_tmp_base;
 
-	u32 *ctrl_playback;
+	__le32 *ctrl_playback;
 	struct snd_ymfpci_playback_bank *bank_playback[YDSXG_PLAYBACK_VOICES][2];
 	struct snd_ymfpci_capture_bank *bank_capture[YDSXG_CAPTURE_VOICES][2];
 	struct snd_ymfpci_effect_bank *bank_effect[YDSXG_EFFECT_VOICES][2];
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index 6f81396aadc9..a4926fb03991 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -336,7 +336,7 @@ static void snd_ymfpci_pcm_interrupt(struct snd_ymfpci *chip, struct snd_ymfpci_
 			unsigned int subs = ypcm->substream->number;
 			unsigned int next_bank = 1 - chip->active_bank;
 			struct snd_ymfpci_playback_bank *bank;
-			u32 volume;
+			__le32 volume;
 			
 			bank = &voice->bank[next_bank];
 			volume = cpu_to_le32(chip->pcm_mixer[subs].left << 15);
@@ -505,7 +505,7 @@ static void snd_ymfpci_pcm_init_voice(struct snd_ymfpci_pcm *ypcm, unsigned int
 	u32 lpfK = snd_ymfpci_calc_lpfK(runtime->rate);
 	struct snd_ymfpci_playback_bank *bank;
 	unsigned int nbank;
-	u32 vol_left, vol_right;
+	__le32 vol_left, vol_right;
 	u8 use_left, use_right;
 	unsigned long flags;
 
@@ -2135,7 +2135,7 @@ static int snd_ymfpci_memalloc(struct snd_ymfpci *chip)
 
 	chip->bank_base_playback = ptr;
 	chip->bank_base_playback_addr = ptr_addr;
-	chip->ctrl_playback = (u32 *)ptr;
+	chip->ctrl_playback = (__le32 *)ptr;
 	chip->ctrl_playback[0] = cpu_to_le32(YDSXG_PLAYBACK_VOICES);
 	ptr += ALIGN(playback_ctrl_size, 0x100);
 	ptr_addr += ALIGN(playback_ctrl_size, 0x100);
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
index 4a2354b5ae00..98a6863e933c 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
@@ -276,7 +276,6 @@ static const struct snd_pcm_ops pdacf_pcm_capture_ops = {
 	.trigger =	pdacf_pcm_trigger,
 	.pointer =	pdacf_pcm_capture_pointer,
 	.page =		snd_pcm_lib_get_vmalloc_page,
-	.mmap =		snd_pcm_lib_mmap_vmalloc,
 };
 
 
diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c
index 8cde40226355..4c4ef1fec69f 100644
--- a/sound/pcmcia/vx/vxp_ops.c
+++ b/sound/pcmcia/vx/vxp_ops.c
@@ -375,7 +375,7 @@ static void vxp_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime,
 		length >>= 1; /* in 16bit words */
 		/* Transfer using pseudo-dma. */
 		for (; length > 0; length--) {
-			outw(cpu_to_le16(*addr), port);
+			outw(*addr, port);
 			addr++;
 		}
 		addr = (unsigned short *)runtime->dma_area;
@@ -385,7 +385,7 @@ static void vxp_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime,
 	count >>= 1; /* in 16bit words */
 	/* Transfer using pseudo-dma. */
 	for (; count > 0; count--) {
-		outw(cpu_to_le16(*addr), port);
+		outw(*addr, port);
 		addr++;
 	}
 	vx_release_pseudo_dma(chip);
@@ -417,7 +417,7 @@ static void vxp_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime,
 		length >>= 1; /* in 16bit words */
 		/* Transfer using pseudo-dma. */
 		for (; length > 0; length--)
-			*addr++ = le16_to_cpu(inw(port));
+			*addr++ = inw(port);
 		addr = (unsigned short *)runtime->dma_area;
 		pipe->hw_ptr = 0;
 	}
@@ -425,12 +425,12 @@ static void vxp_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime,
 	count >>= 1; /* in 16bit words */
 	/* Transfer using pseudo-dma. */
 	for (; count > 1; count--)
-		*addr++ = le16_to_cpu(inw(port));
+		*addr++ = inw(port);
 	/* Disable DMA */
 	pchip->regDIALOG &= ~VXP_DLG_DMAREAD_SEL_MASK;
 	vx_outb(chip, DIALOG, pchip->regDIALOG);
 	/* Read the last word (16 bits) */
-	*addr = le16_to_cpu(inw(port));
+	*addr = inw(port);
 	/* Disable 16-bit accesses */
 	pchip->regDIALOG &= ~VXP_DLG_DMA16_SEL_MASK;
 	vx_outb(chip, DIALOG, pchip->regDIALOG);
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 41af6b9cc350..1cf11cf51e1d 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -57,6 +57,7 @@ source "sound/soc/kirkwood/Kconfig"
 source "sound/soc/img/Kconfig"
 source "sound/soc/intel/Kconfig"
 source "sound/soc/mediatek/Kconfig"
+source "sound/soc/meson/Kconfig"
 source "sound/soc/mxs/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/qcom/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 06389a5385d7..62a5f87c3cfc 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -38,6 +38,7 @@ obj-$(CONFIG_SND_SOC)	+= jz4740/
 obj-$(CONFIG_SND_SOC)	+= img/
 obj-$(CONFIG_SND_SOC)	+= intel/
 obj-$(CONFIG_SND_SOC)	+= mediatek/
+obj-$(CONFIG_SND_SOC)	+= meson/
 obj-$(CONFIG_SND_SOC)	+= mxs/
 obj-$(CONFIG_SND_SOC)	+= nuc900/
 obj-$(CONFIG_SND_SOC)	+= omap/
diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig
index 6cbf9cf4d1a4..58c1dcb4d255 100644
--- a/sound/soc/amd/Kconfig
+++ b/sound/soc/amd/Kconfig
@@ -8,6 +8,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH
 	select SND_SOC_DA7219
 	select SND_SOC_MAX98357A
 	select SND_SOC_ADAU7002
+	select REGULATOR
 	depends on SND_SOC_AMD_ACP && I2C
 	help
 	 This option enables machine driver for DA7219 and MAX9835.
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index ccddc6650b9c..8e3275a96a82 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -32,6 +32,8 @@
 #include <linux/clk.h>
 #include <linux/gpio.h>
 #include <linux/module.h>
+#include <linux/regulator/machine.h>
+#include <linux/regulator/driver.h>
 #include <linux/i2c.h>
 #include <linux/input.h>
 #include <linux/acpi.h>
@@ -148,7 +150,8 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream)
 	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
 				   &constraints_rates);
 
-	machine->i2s_instance = I2S_BT_INSTANCE;
+	machine->i2s_instance = I2S_SP_INSTANCE;
+	machine->capture_channel = CAP_CHANNEL1;
 	return da7219_clk_enable(substream);
 }
 
@@ -163,7 +166,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
-	machine->i2s_instance = I2S_SP_INSTANCE;
+	machine->i2s_instance = I2S_BT_INSTANCE;
 	return da7219_clk_enable(substream);
 }
 
@@ -172,13 +175,24 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream)
 	da7219_clk_disable();
 }
 
-static int cz_dmic_startup(struct snd_pcm_substream *substream)
+static int cz_dmic0_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+	machine->i2s_instance = I2S_BT_INSTANCE;
+	return da7219_clk_enable(substream);
+}
+
+static int cz_dmic1_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
 	machine->i2s_instance = I2S_SP_INSTANCE;
+	machine->capture_channel = CAP_CHANNEL0;
 	return da7219_clk_enable(substream);
 }
 
@@ -197,23 +211,39 @@ static const struct snd_soc_ops cz_max_play_ops = {
 	.shutdown = cz_max_shutdown,
 };
 
-static const struct snd_soc_ops cz_dmic_cap_ops = {
-	.startup = cz_dmic_startup,
+static const struct snd_soc_ops cz_dmic0_cap_ops = {
+	.startup = cz_dmic0_startup,
+	.shutdown = cz_dmic_shutdown,
+};
+
+static const struct snd_soc_ops cz_dmic1_cap_ops = {
+	.startup = cz_dmic1_startup,
 	.shutdown = cz_dmic_shutdown,
 };
 
 static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 	{
-		.name = "amd-da7219-play-cap",
-		.stream_name = "Playback and Capture",
+		.name = "amd-da7219-play",
+		.stream_name = "Playback",
 		.platform_name = "acp_audio_dma.0.auto",
-		.cpu_dai_name = "designware-i2s.3.auto",
+		.cpu_dai_name = "designware-i2s.1.auto",
 		.codec_dai_name = "da7219-hifi",
 		.codec_name = "i2c-DLGS7219:00",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.init = cz_da7219_init,
 		.dpcm_playback = 1,
+		.ops = &cz_da7219_cap_ops,
+	},
+	{
+		.name = "amd-da7219-cap",
+		.stream_name = "Capture",
+		.platform_name = "acp_audio_dma.0.auto",
+		.cpu_dai_name = "designware-i2s.2.auto",
+		.codec_dai_name = "da7219-hifi",
+		.codec_name = "i2c-DLGS7219:00",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
 		.ops = &cz_da7219_cap_ops,
 	},
@@ -221,7 +251,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 		.name = "amd-max98357-play",
 		.stream_name = "HiFi Playback",
 		.platform_name = "acp_audio_dma.0.auto",
-		.cpu_dai_name = "designware-i2s.1.auto",
+		.cpu_dai_name = "designware-i2s.3.auto",
 		.codec_dai_name = "HiFi",
 		.codec_name = "MX98357A:00",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
@@ -230,8 +260,22 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 		.ops = &cz_max_play_ops,
 	},
 	{
-		.name = "dmic",
-		.stream_name = "DMIC Capture",
+		/* C panel DMIC */
+		.name = "dmic0",
+		.stream_name = "DMIC0 Capture",
+		.platform_name = "acp_audio_dma.0.auto",
+		.cpu_dai_name = "designware-i2s.3.auto",
+		.codec_dai_name = "adau7002-hifi",
+		.codec_name = "ADAU7002:00",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+		.dpcm_capture = 1,
+		.ops = &cz_dmic0_cap_ops,
+	},
+	{
+		/* A/B panel DMIC */
+		.name = "dmic1",
+		.stream_name = "DMIC1 Capture",
 		.platform_name = "acp_audio_dma.0.auto",
 		.cpu_dai_name = "designware-i2s.2.auto",
 		.codec_dai_name = "adau7002-hifi",
@@ -239,7 +283,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
-		.ops = &cz_dmic_cap_ops,
+		.ops = &cz_dmic1_cap_ops,
 	},
 };
 
@@ -278,11 +322,52 @@ static struct snd_soc_card cz_card = {
 	.num_controls = ARRAY_SIZE(cz_mc_controls),
 };
 
+static struct regulator_consumer_supply acp_da7219_supplies[] = {
+	REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"),
+	REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"),
+	REGULATOR_SUPPLY("VDDIO", "i2c-DLGS7219:00"),
+	REGULATOR_SUPPLY("IOVDD", "ADAU7002:00"),
+};
+
+static struct regulator_init_data acp_da7219_data = {
+	.constraints = {
+		.always_on = 1,
+	},
+	.num_consumer_supplies = ARRAY_SIZE(acp_da7219_supplies),
+	.consumer_supplies = acp_da7219_supplies,
+};
+
+static struct regulator_config acp_da7219_cfg = {
+	.init_data = &acp_da7219_data,
+};
+
+static struct regulator_ops acp_da7219_ops = {
+};
+
+static struct regulator_desc acp_da7219_desc = {
+	.name = "reg-fixed-1.8V",
+	.type = REGULATOR_VOLTAGE,
+	.owner = THIS_MODULE,
+	.ops = &acp_da7219_ops,
+	.fixed_uV = 1800000, /* 1.8V */
+	.n_voltages = 1,
+};
+
 static int cz_probe(struct platform_device *pdev)
 {
 	int ret;
 	struct snd_soc_card *card;
 	struct acp_platform_info *machine;
+	struct regulator_dev *rdev;
+
+	acp_da7219_cfg.dev = &pdev->dev;
+	rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc,
+				       &acp_da7219_cfg);
+	if (IS_ERR(rdev)) {
+		dev_err(&pdev->dev, "Failed to register regulator: %d\n",
+			(int)PTR_ERR(rdev));
+		return -EINVAL;
+	}
 
 	machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info),
 			       GFP_KERNEL);
diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c
index 77203841c535..e359938e3d7e 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -224,13 +224,11 @@ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio,
 			switch (asic_type) {
 			case CHIP_STONEY:
 				dmadscr[i].xfer_val |=
-				BIT(22) |
 				(ACP_DMA_ATTR_SHARED_MEM_TO_DAGB_GARLIC << 16) |
 				(size / 2);
 				break;
 			default:
 				dmadscr[i].xfer_val |=
-				BIT(22) |
 				(ACP_DMA_ATTR_SHAREDMEM_TO_DAGB_ONION << 16) |
 				(size / 2);
 			}
@@ -322,22 +320,87 @@ static void config_acp_dma(void __iomem *acp_mmio,
 			   struct audio_substream_data *rtd,
 			   u32 asic_type)
 {
+	u16 ch_acp_sysmem, ch_acp_i2s;
+
 	acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages,
 		       rtd->pte_offset);
+
+	if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) {
+		ch_acp_sysmem = rtd->ch1;
+		ch_acp_i2s = rtd->ch2;
+	} else {
+		ch_acp_i2s = rtd->ch1;
+		ch_acp_sysmem = rtd->ch2;
+	}
 	/* Configure System memory <-> ACP SRAM DMA descriptors */
 	set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size,
 				       rtd->direction, rtd->pte_offset,
-				       rtd->ch1, rtd->sram_bank,
+				       ch_acp_sysmem, rtd->sram_bank,
 				       rtd->dma_dscr_idx_1, asic_type);
 	/* Configure ACP SRAM <-> I2S DMA descriptors */
 	set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size,
 				       rtd->direction, rtd->sram_bank,
-				       rtd->destination, rtd->ch2,
+				       rtd->destination, ch_acp_i2s,
 				       rtd->dma_dscr_idx_2, asic_type);
 }
 
+static void acp_dma_cap_channel_enable(void __iomem *acp_mmio,
+				       u16 cap_channel)
+{
+	u32 val, ch_reg, imr_reg, res_reg;
+
+	switch (cap_channel) {
+	case CAP_CHANNEL1:
+		ch_reg = mmACP_I2SMICSP_RER1;
+		res_reg = mmACP_I2SMICSP_RCR1;
+		imr_reg = mmACP_I2SMICSP_IMR1;
+		break;
+	case CAP_CHANNEL0:
+	default:
+		ch_reg = mmACP_I2SMICSP_RER0;
+		res_reg = mmACP_I2SMICSP_RCR0;
+		imr_reg = mmACP_I2SMICSP_IMR0;
+		break;
+	}
+	val = acp_reg_read(acp_mmio,
+			   mmACP_I2S_16BIT_RESOLUTION_EN);
+	if (val & ACP_I2S_MIC_16BIT_RESOLUTION_EN) {
+		acp_reg_write(0x0, acp_mmio, ch_reg);
+		/* Set 16bit resolution on capture */
+		acp_reg_write(0x2, acp_mmio, res_reg);
+	}
+	val = acp_reg_read(acp_mmio, imr_reg);
+	val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK;
+	val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK;
+	acp_reg_write(val, acp_mmio, imr_reg);
+	acp_reg_write(0x1, acp_mmio, ch_reg);
+}
+
+static void acp_dma_cap_channel_disable(void __iomem *acp_mmio,
+					u16 cap_channel)
+{
+	u32 val, ch_reg, imr_reg;
+
+	switch (cap_channel) {
+	case CAP_CHANNEL1:
+		imr_reg = mmACP_I2SMICSP_IMR1;
+		ch_reg = mmACP_I2SMICSP_RER1;
+		break;
+	case CAP_CHANNEL0:
+	default:
+		imr_reg = mmACP_I2SMICSP_IMR0;
+		ch_reg = mmACP_I2SMICSP_RER0;
+		break;
+	}
+	val = acp_reg_read(acp_mmio, imr_reg);
+	val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK;
+	val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK;
+	acp_reg_write(val, acp_mmio, imr_reg);
+	acp_reg_write(0x0, acp_mmio, ch_reg);
+}
+
 /* Start a given DMA channel transfer */
-static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num)
+static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num, bool is_circular)
 {
 	u32 dma_ctrl;
 
@@ -356,10 +419,8 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num)
 
 	switch (ch_num) {
 	case ACP_TO_I2S_DMA_CH_NUM:
-	case ACP_TO_SYSRAM_CH_NUM:
 	case I2S_TO_ACP_DMA_CH_NUM:
 	case ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM:
-	case ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM:
 	case I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM:
 		dma_ctrl |= ACP_DMA_CNTL_0__DMAChIOCEn_MASK;
 		break;
@@ -368,8 +429,11 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num)
 		break;
 	}
 
-	/* circular for both DMA channel */
-	dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK;
+	/* enable for ACP to SRAM DMA channel */
+	if (is_circular == true)
+		dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK;
+	else
+		dma_ctrl &= ~ACP_DMA_CNTL_0__Circular_DMA_En_MASK;
 
 	acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 + ch_num);
 }
@@ -613,6 +677,7 @@ static int acp_deinit(void __iomem *acp_mmio)
 /* ACP DMA irq handler routine for playback, capture usecases */
 static irqreturn_t dma_irq_handler(int irq, void *arg)
 {
+	u16 dscr_idx;
 	u32 intr_flag, ext_intr_status;
 	struct audio_drv_data *irq_data;
 	void __iomem *acp_mmio;
@@ -644,32 +709,39 @@ static irqreturn_t dma_irq_handler(int irq, void *arg)
 
 	if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) {
 		valid_irq = true;
+		if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_14) ==
+				CAPTURE_START_DMA_DESCR_CH15)
+			dscr_idx = CAPTURE_END_DMA_DESCR_CH14;
+		else
+			dscr_idx = CAPTURE_START_DMA_DESCR_CH14;
+		config_acp_dma_channel(acp_mmio, ACP_TO_SYSRAM_CH_NUM, dscr_idx,
+				       1, 0);
+		acp_dma_start(acp_mmio, ACP_TO_SYSRAM_CH_NUM, false);
+
 		snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream);
 		acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16,
 			      acp_mmio, mmACP_EXTERNAL_INTR_STAT);
 	}
 
-	if ((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) != 0) {
-		valid_irq = true;
-		acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) << 16,
-			      acp_mmio, mmACP_EXTERNAL_INTR_STAT);
-	}
-
 	if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) {
 		valid_irq = true;
+		if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_10) ==
+			CAPTURE_START_DMA_DESCR_CH11)
+			dscr_idx = CAPTURE_END_DMA_DESCR_CH10;
+		else
+			dscr_idx = CAPTURE_START_DMA_DESCR_CH10;
+		config_acp_dma_channel(acp_mmio,
+				       ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM,
+				       dscr_idx, 1, 0);
+		acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM,
+			      false);
+
 		snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream);
 		acp_reg_write((intr_flag &
 			      BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16,
 			      acp_mmio, mmACP_EXTERNAL_INTR_STAT);
 	}
 
-	if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) {
-		valid_irq = true;
-		acp_reg_write((intr_flag &
-			      BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) << 16,
-			      acp_mmio, mmACP_EXTERNAL_INTR_STAT);
-	}
-
 	if (valid_irq)
 		return IRQ_HANDLED;
 	else
@@ -773,7 +845,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
 	if (WARN_ON(!rtd))
 		return -EINVAL;
 
-	rtd->i2s_instance = pinfo->i2s_instance;
+	if (pinfo) {
+		rtd->i2s_instance = pinfo->i2s_instance;
+		rtd->capture_channel = pinfo->capture_channel;
+	}
 	if (adata->asic_type == CHIP_STONEY) {
 		val = acp_reg_read(adata->acp_mmio,
 				   mmACP_I2S_16BIT_RESOLUTION_EN);
@@ -841,8 +916,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
 		switch (rtd->i2s_instance) {
 		case I2S_BT_INSTANCE:
 			rtd->pte_offset = ACP_ST_BT_CAPTURE_PTE_OFFSET;
-			rtd->ch1 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM;
-			rtd->ch2 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM;
+			rtd->ch1 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM;
+			rtd->ch2 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM;
 			rtd->sram_bank = ACP_SRAM_BANK_4_ADDRESS;
 			rtd->destination = FROM_BLUETOOTH;
 			rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10;
@@ -851,13 +926,14 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
 					mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH;
 			rtd->byte_cnt_low_reg_offset =
 					mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW;
+			rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_11;
 			adata->capture_i2sbt_stream = substream;
 			break;
 		case I2S_SP_INSTANCE:
 		default:
 			rtd->pte_offset = ACP_CAPTURE_PTE_OFFSET;
-			rtd->ch1 = ACP_TO_SYSRAM_CH_NUM;
-			rtd->ch2 = I2S_TO_ACP_DMA_CH_NUM;
+			rtd->ch1 = I2S_TO_ACP_DMA_CH_NUM;
+			rtd->ch2 = ACP_TO_SYSRAM_CH_NUM;
 			switch (adata->asic_type) {
 			case CHIP_STONEY:
 				rtd->pte_offset = ACP_ST_CAPTURE_PTE_OFFSET;
@@ -874,6 +950,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
 					mmACP_I2S_RECEIVED_BYTE_CNT_HIGH;
 			rtd->byte_cnt_low_reg_offset =
 					mmACP_I2S_RECEIVED_BYTE_CNT_LOW;
+			rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_15;
 			adata->capture_i2ssp_stream = substream;
 		}
 	}
@@ -927,6 +1004,8 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream)
 	u32 buffersize;
 	u32 pos = 0;
 	u64 bytescount = 0;
+	u16 dscr;
+	u32 period_bytes, delay;
 
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct audio_substream_data *rtd = runtime->private_data;
@@ -934,12 +1013,25 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream)
 	if (!rtd)
 		return -EINVAL;
 
-	buffersize = frames_to_bytes(runtime, runtime->buffer_size);
-	bytescount = acp_get_byte_count(rtd);
-
-	if (bytescount > rtd->bytescount)
-		bytescount -= rtd->bytescount;
-	pos = do_div(bytescount, buffersize);
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		period_bytes = frames_to_bytes(runtime, runtime->period_size);
+		dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr);
+		if (dscr == rtd->dma_dscr_idx_1)
+			pos = period_bytes;
+		else
+			pos = 0;
+		bytescount = acp_get_byte_count(rtd);
+		if (bytescount > rtd->bytescount)
+			bytescount -= rtd->bytescount;
+		delay = do_div(bytescount, period_bytes);
+		runtime->delay = bytes_to_frames(runtime, delay);
+	} else {
+		buffersize = frames_to_bytes(runtime, runtime->buffer_size);
+		bytescount = acp_get_byte_count(rtd);
+		if (bytescount > rtd->bytescount)
+			bytescount -= rtd->bytescount;
+		pos = do_div(bytescount, buffersize);
+	}
 	return bytes_to_frames(runtime, pos);
 }
 
@@ -953,16 +1045,24 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct audio_substream_data *rtd = runtime->private_data;
+	u16 ch_acp_sysmem, ch_acp_i2s;
 
 	if (!rtd)
 		return -EINVAL;
 
+	if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) {
+		ch_acp_sysmem = rtd->ch1;
+		ch_acp_i2s = rtd->ch2;
+	} else {
+		ch_acp_i2s = rtd->ch1;
+		ch_acp_sysmem = rtd->ch2;
+	}
 	config_acp_dma_channel(rtd->acp_mmio,
-			       rtd->ch1,
+			       ch_acp_sysmem,
 			       rtd->dma_dscr_idx_1,
 			       NUM_DSCRS_PER_CHANNEL, 0);
 	config_acp_dma_channel(rtd->acp_mmio,
-			       rtd->ch2,
+			       ch_acp_i2s,
 			       rtd->dma_dscr_idx_2,
 			       NUM_DSCRS_PER_CHANNEL, 0);
 	return 0;
@@ -971,7 +1071,6 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream)
 static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd)
 {
 	int ret;
-	u64 bytescount = 0;
 
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct audio_substream_data *rtd = runtime->private_data;
@@ -982,37 +1081,32 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 	case SNDRV_PCM_TRIGGER_RESUME:
-		bytescount = acp_get_byte_count(rtd);
-		if (rtd->bytescount == 0)
-			rtd->bytescount = bytescount;
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			acp_dma_start(rtd->acp_mmio, rtd->ch1);
-			acp_dma_start(rtd->acp_mmio, rtd->ch2);
+		rtd->bytescount = acp_get_byte_count(rtd);
+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+			if (rtd->capture_channel == CAP_CHANNEL0) {
+				acp_dma_cap_channel_disable(rtd->acp_mmio,
+							    CAP_CHANNEL1);
+				acp_dma_cap_channel_enable(rtd->acp_mmio,
+							   CAP_CHANNEL0);
+			}
+			if (rtd->capture_channel == CAP_CHANNEL1) {
+				acp_dma_cap_channel_disable(rtd->acp_mmio,
+							    CAP_CHANNEL0);
+				acp_dma_cap_channel_enable(rtd->acp_mmio,
+							   CAP_CHANNEL1);
+			}
+			acp_dma_start(rtd->acp_mmio, rtd->ch1, true);
 		} else {
-			acp_dma_start(rtd->acp_mmio, rtd->ch2);
-			acp_dma_start(rtd->acp_mmio, rtd->ch1);
+			acp_dma_start(rtd->acp_mmio, rtd->ch1, true);
+			acp_dma_start(rtd->acp_mmio, rtd->ch2, true);
 		}
 		ret = 0;
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
-		/* For playback, non circular dma should be stopped first
-		 * i.e Sysram to acp dma transfer channel(rtd->ch1) should be
-		 * stopped before stopping cirular dma which is acp sram to i2s
-		 * fifo dma transfer channel(rtd->ch2). Where as in Capture
-		 * scenario, i2s fifo to acp sram dma channel(rtd->ch2) stopped
-		 * first before stopping acp sram to sysram which is circular
-		 * dma(rtd->ch1).
-		 */
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			acp_dma_stop(rtd->acp_mmio, rtd->ch1);
-			ret =  acp_dma_stop(rtd->acp_mmio, rtd->ch2);
-		} else {
-			acp_dma_stop(rtd->acp_mmio, rtd->ch2);
-			ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1);
-		}
-		rtd->bytescount = 0;
+		acp_dma_stop(rtd->acp_mmio, rtd->ch2);
+		ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1);
 		break;
 	default:
 		ret = -EINVAL;
diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h
index 9cd3e96c84d4..be3963e8f4fa 100644
--- a/sound/soc/amd/acp.h
+++ b/sound/soc/amd/acp.h
@@ -55,6 +55,8 @@
 
 #define I2S_SP_INSTANCE                 0x01
 #define I2S_BT_INSTANCE                 0x02
+#define CAP_CHANNEL0			0x00
+#define CAP_CHANNEL1			0x01
 
 #define ACP_TILE_ON_MASK                0x03
 #define ACP_TILE_OFF_MASK               0x02
@@ -72,16 +74,16 @@
 #define ACP_TO_I2S_DMA_CH_NUM 13
 
 /* Capture DMA channels */
-#define ACP_TO_SYSRAM_CH_NUM 14
-#define I2S_TO_ACP_DMA_CH_NUM 15
+#define I2S_TO_ACP_DMA_CH_NUM 14
+#define ACP_TO_SYSRAM_CH_NUM 15
 
 /* Playback DMA Channels for I2S BT instance */
 #define SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM  8
 #define ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM 9
 
 /* Capture DMA Channels for I2S BT Instance */
-#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 10
-#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 11
+#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 10
+#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 11
 
 #define NUM_DSCRS_PER_CHANNEL 2
 
@@ -125,6 +127,7 @@ struct audio_substream_data {
 	unsigned int order;
 	u16 num_of_pages;
 	u16 i2s_instance;
+	u16 capture_channel;
 	u16 direction;
 	u16 ch1;
 	u16 ch2;
@@ -135,6 +138,7 @@ struct audio_substream_data {
 	u32 sram_bank;
 	u32 byte_cnt_high_reg_offset;
 	u32 byte_cnt_low_reg_offset;
+	u32 dma_curr_dscr;
 	uint64_t size;
 	u64 bytescount;
 	void __iomem *acp_mmio;
@@ -155,6 +159,7 @@ struct audio_drv_data {
  */
 struct acp_platform_info {
 	u16 i2s_instance;
+	u16 capture_channel;
 };
 
 union acp_dma_count {
diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c
index 5d3b5af9fd92..d88c1d995036 100644
--- a/sound/soc/atmel/atmel-i2s.c
+++ b/sound/soc/atmel/atmel-i2s.c
@@ -206,7 +206,6 @@ struct atmel_i2s_dev {
 	struct regmap				*regmap;
 	struct clk				*pclk;
 	struct clk				*gclk;
-	struct clk				*aclk;
 	struct snd_dmaengine_dai_dma_data	playback;
 	struct snd_dmaengine_dai_dma_data	capture;
 	unsigned int				fmt;
@@ -303,7 +302,7 @@ static int atmel_i2s_get_gck_param(struct atmel_i2s_dev *dev, int fs)
 {
 	int i, best;
 
-	if (!dev->gclk || !dev->aclk) {
+	if (!dev->gclk) {
 		dev_err(dev->dev, "cannot generate the I2S Master Clock\n");
 		return -EINVAL;
 	}
@@ -421,7 +420,7 @@ static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev,
 					  bool enabled)
 {
 	unsigned int mr, mr_mask;
-	unsigned long aclk_rate;
+	unsigned long gclk_rate;
 	int ret;
 
 	mr = 0;
@@ -445,35 +444,18 @@ static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev,
 		/* Disable/unprepare the PMC generated clock. */
 		clk_disable_unprepare(dev->gclk);
 
-		/* Disable/unprepare the PLL audio clock. */
-		clk_disable_unprepare(dev->aclk);
 		return 0;
 	}
 
 	if (!dev->gck_param)
 		return -EINVAL;
 
-	aclk_rate = dev->gck_param->mck * (dev->gck_param->imckdiv + 1);
+	gclk_rate = dev->gck_param->mck * (dev->gck_param->imckdiv + 1);
 
-	/* Fist change the PLL audio clock frequency ... */
-	ret = clk_set_rate(dev->aclk, aclk_rate);
+	ret = clk_set_rate(dev->gclk, gclk_rate);
 	if (ret)
 		return ret;
 
-	/*
-	 * ... then set the PMC generated clock rate to the very same frequency
-	 * to set the gclk parent to aclk.
-	 */
-	ret = clk_set_rate(dev->gclk, aclk_rate);
-	if (ret)
-		return ret;
-
-	/* Prepare and enable the PLL audio clock first ... */
-	ret = clk_prepare_enable(dev->aclk);
-	if (ret)
-		return ret;
-
-	/* ... then prepare and enable the PMC generated clock. */
 	ret = clk_prepare_enable(dev->gclk);
 	if (ret)
 		return ret;
@@ -668,28 +650,14 @@ static int atmel_i2s_probe(struct platform_device *pdev)
 		return err;
 	}
 
-	/* Get audio clocks to generate the I2S Master Clock (I2S_MCK) */
-	dev->aclk = devm_clk_get(&pdev->dev, "aclk");
+	/* Get audio clock to generate the I2S Master Clock (I2S_MCK) */
 	dev->gclk = devm_clk_get(&pdev->dev, "gclk");
-	if (IS_ERR(dev->aclk) && IS_ERR(dev->gclk)) {
-		if (PTR_ERR(dev->aclk) == -EPROBE_DEFER ||
-		    PTR_ERR(dev->gclk) == -EPROBE_DEFER)
+	if (IS_ERR(dev->gclk)) {
+		if (PTR_ERR(dev->gclk) == -EPROBE_DEFER)
 			return -EPROBE_DEFER;
 		/* Master Mode not supported */
-		dev->aclk = NULL;
 		dev->gclk = NULL;
-	} else if (IS_ERR(dev->gclk)) {
-		err = PTR_ERR(dev->gclk);
-		dev_err(&pdev->dev,
-			"failed to get the PMC generated clock: %d\n", err);
-		return err;
-	} else if (IS_ERR(dev->aclk)) {
-		err = PTR_ERR(dev->aclk);
-		dev_err(&pdev->dev,
-			"failed to get the PLL audio clock: %d\n", err);
-		return err;
 	}
-
 	dev->dev = &pdev->dev;
 	dev->regmap = regmap;
 	platform_set_drvdata(pdev, dev);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 63cf62e9c9aa..efb095dbcd71 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -74,12 +74,12 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_DA7219 if I2C
 	select SND_SOC_DA732X if I2C
 	select SND_SOC_DA9055 if I2C
-	select SND_SOC_DIO2125
 	select SND_SOC_DMIC if GPIOLIB
 	select SND_SOC_ES8316 if I2C
 	select SND_SOC_ES8328_SPI if SPI_MASTER
 	select SND_SOC_ES8328_I2C if I2C
 	select SND_SOC_ES7134
+	select SND_SOC_ES7241
 	select SND_SOC_GTM601
 	select SND_SOC_HDAC_HDMI
 	select SND_SOC_ICS43432
@@ -141,8 +141,10 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_RT5668 if I2C
 	select SND_SOC_RT5670 if I2C
 	select SND_SOC_RT5677 if I2C && SPI_MASTER
+	select SND_SOC_RT5682 if I2C
 	select SND_SOC_SGTL5000 if I2C
 	select SND_SOC_SI476X if MFD_SI476X_CORE
+	select SND_SOC_SIMPLE_AMPLIFIER
 	select SND_SOC_SIRF_AUDIO_CODEC
 	select SND_SOC_SPDIF
 	select SND_SOC_SSM2305
@@ -572,10 +574,6 @@ config SND_SOC_DA732X
 config SND_SOC_DA9055
 	tristate
 
-config SND_SOC_DIO2125
-	tristate "Dioo DIO2125 Amplifier"
-	select GPIOLIB
-
 config SND_SOC_DMIC
 	tristate
 
@@ -588,6 +586,9 @@ config SND_SOC_HDMI_CODEC
 config SND_SOC_ES7134
        tristate "Everest Semi ES7134 CODEC"
 
+config SND_SOC_ES7241
+       tristate "Everest Semi ES7241 CODEC"
+
 config SND_SOC_ES8316
 	tristate "Everest Semi ES8316 CODEC"
 	depends on I2C
@@ -778,6 +779,7 @@ config SND_SOC_RL6231
 	default y if SND_SOC_RT5668=y
 	default y if SND_SOC_RT5670=y
 	default y if SND_SOC_RT5677=y
+	default y if SND_SOC_RT5682=y
 	default y if SND_SOC_RT1305=y
 	default m if SND_SOC_RT5514=m
 	default m if SND_SOC_RT5616=m
@@ -791,6 +793,7 @@ config SND_SOC_RL6231
 	default m if SND_SOC_RT5668=m
 	default m if SND_SOC_RT5670=m
 	default m if SND_SOC_RT5677=m
+	default m if SND_SOC_RT5682=m
 	default m if SND_SOC_RT1305=m
 
 config SND_SOC_RL6347A
@@ -871,6 +874,9 @@ config SND_SOC_RT5677_SPI
 	tristate
 	default SND_SOC_RT5677 && SPI
 
+config SND_SOC_RT5682
+	tristate
+
 #Freescale sgtl5000 codec
 config SND_SOC_SGTL5000
 	tristate "Freescale SGTL5000 CODEC"
@@ -891,6 +897,10 @@ config SND_SOC_SIGMADSP_REGMAP
 	tristate
 	select SND_SOC_SIGMADSP
 
+config SND_SOC_SIMPLE_AMPLIFIER
+	tristate "Simple Audio Amplifier"
+	select GPIOLIB
+
 config SND_SOC_SIRF_AUDIO_CODEC
 	tristate "SiRF SoC internal audio codec"
 	select REGMAP_MMIO
@@ -953,8 +963,11 @@ config SND_SOC_TAS5086
 	depends on I2C
 
 config SND_SOC_TAS571X
-	tristate "Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 power amplifiers"
+	tristate "Texas Instruments TAS571x power amplifiers"
 	depends on I2C
+	help
+	  Enable support for Texas Instruments TAS5707, TAS5711, TAS5717,
+	  TAS5719 and TAS5721 power amplifiers
 
 config SND_SOC_TAS5720
 	tristate "Texas Instruments TAS5720 Mono Audio amplifier"
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index e023fdf85221..7ae7c85e8219 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -71,6 +71,7 @@ snd-soc-da732x-objs := da732x.o
 snd-soc-da9055-objs := da9055.o
 snd-soc-dmic-objs := dmic.o
 snd-soc-es7134-objs := es7134.o
+snd-soc-es7241-objs := es7241.o
 snd-soc-es8316-objs := es8316.o
 snd-soc-es8328-objs := es8328.o
 snd-soc-es8328-i2c-objs := es8328-i2c.o
@@ -146,6 +147,7 @@ snd-soc-rt5668-objs := rt5668.o
 snd-soc-rt5670-objs := rt5670.o
 snd-soc-rt5677-objs := rt5677.o
 snd-soc-rt5677-spi-objs := rt5677-spi.o
+snd-soc-rt5682-objs := rt5682.o
 snd-soc-sgtl5000-objs := sgtl5000.o
 snd-soc-alc5623-objs := alc5623.o
 snd-soc-alc5632-objs := alc5632.o
@@ -249,9 +251,9 @@ snd-soc-wm9713-objs := wm9713.o
 snd-soc-wm-hubs-objs := wm_hubs.o
 snd-soc-zx-aud96p22-objs := zx_aud96p22.o
 # Amp
-snd-soc-dio2125-objs := dio2125.o
 snd-soc-max9877-objs := max9877.o
 snd-soc-max98504-objs := max98504.o
+snd-soc-simple-amplifier-objs := simple-amplifier.o
 snd-soc-tpa6130a2-objs := tpa6130a2.o
 snd-soc-tas2552-objs := tas2552.o
 
@@ -329,6 +331,7 @@ obj-$(CONFIG_SND_SOC_DA732X)	+= snd-soc-da732x.o
 obj-$(CONFIG_SND_SOC_DA9055)	+= snd-soc-da9055.o
 obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
 obj-$(CONFIG_SND_SOC_ES7134)	+= snd-soc-es7134.o
+obj-$(CONFIG_SND_SOC_ES7241)	+= snd-soc-es7241.o
 obj-$(CONFIG_SND_SOC_ES8316)    += snd-soc-es8316.o
 obj-$(CONFIG_SND_SOC_ES8328)	+= snd-soc-es8328.o
 obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
@@ -405,6 +408,7 @@ obj-$(CONFIG_SND_SOC_RT5668)	+= snd-soc-rt5668.o
 obj-$(CONFIG_SND_SOC_RT5670)	+= snd-soc-rt5670.o
 obj-$(CONFIG_SND_SOC_RT5677)	+= snd-soc-rt5677.o
 obj-$(CONFIG_SND_SOC_RT5677_SPI)	+= snd-soc-rt5677-spi.o
+obj-$(CONFIG_SND_SOC_RT5682)	+= snd-soc-rt5682.o
 obj-$(CONFIG_SND_SOC_SGTL5000)  += snd-soc-sgtl5000.o
 obj-$(CONFIG_SND_SOC_SIGMADSP)	+= snd-soc-sigmadsp.o
 obj-$(CONFIG_SND_SOC_SIGMADSP_I2C)	+= snd-soc-sigmadsp-i2c.o
@@ -507,7 +511,7 @@ obj-$(CONFIG_SND_SOC_WM_HUBS)	+= snd-soc-wm-hubs.o
 obj-$(CONFIG_SND_SOC_ZX_AUD96P22) += snd-soc-zx-aud96p22.o
 
 # Amp
-obj-$(CONFIG_SND_SOC_DIO2125)	+= snd-soc-dio2125.o
 obj-$(CONFIG_SND_SOC_MAX9877)	+= snd-soc-max9877.o
 obj-$(CONFIG_SND_SOC_MAX98504)	+= snd-soc-max98504.o
+obj-$(CONFIG_SND_SOC_SIMPLE_AMPLIFIER)	+= snd-soc-simple-amplifier.o
 obj-$(CONFIG_SND_SOC_TPA6130A2)	+= snd-soc-tpa6130a2.o
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index ae41edd1c406..57169b8ff14e 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -299,6 +299,7 @@ static const struct snd_soc_dapm_route adau17x1_dsp_dapm_routes[] = {
 
 	{ "DSP", NULL, "Left Decimator" },
 	{ "DSP", NULL, "Right Decimator" },
+	{ "DSP", NULL, "Playback" },
 };
 
 static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = {
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index db21ecbe0762..8b9ca7e7a682 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -648,6 +648,7 @@ static int adav80x_set_pll(struct snd_soc_component *component, int pll_id,
 			pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV;
 			break;
 		}
+		/* fall through */
 	default:
 		return -EINVAL;
 	}
diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c
index 31ec0ba2e639..299ada4dfaa0 100644
--- a/sound/soc/codecs/ak4458.c
+++ b/sound/soc/codecs/ak4458.c
@@ -558,7 +558,7 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev)
 }
 #endif /* CONFIG_PM */
 
-struct snd_soc_component_driver soc_codec_dev_ak4458 = {
+static const struct snd_soc_component_driver soc_codec_dev_ak4458 = {
 	.probe			= ak4458_probe,
 	.remove			= ak4458_remove,
 	.controls		= ak4458_snd_controls,
diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c
index b7ee13406d93..2fa83a1a84cf 100644
--- a/sound/soc/codecs/ak4554.c
+++ b/sound/soc/codecs/ak4554.c
@@ -1,13 +1,8 @@
-/*
- * ak4554.c
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+// ak4554.c
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
 
 #include <linux/module.h>
 #include <sound/soc.h>
@@ -97,6 +92,6 @@ static struct platform_driver ak4554_driver = {
 };
 module_platform_driver(ak4554_driver);
 
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_DESCRIPTION("SoC AK4554 driver");
 MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c
index 8523ff9351cf..c1181a20714d 100644
--- a/sound/soc/codecs/ak4613.c
+++ b/sound/soc/codecs/ak4613.c
@@ -1,18 +1,14 @@
-/*
- * ak4613.c  --  Asahi Kasei ALSA Soc Audio driver
- *
- * Copyright (C) 2015 Renesas Electronics Corporation
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * Based on ak4642.c by Kuninori Morimoto
- * Based on wm8731.c by Richard Purdie
- * Based on ak4535.c by Richard Purdie
- * Based on wm8753.c by Liam Girdwood
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ak4613.c  --  Asahi Kasei ALSA Soc Audio driver
+//
+// Copyright (C) 2015 Renesas Electronics Corporation
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// Based on ak4642.c by Kuninori Morimoto
+// Based on wm8731.c by Richard Purdie
+// Based on ak4535.c by Richard Purdie
+// Based on wm8753.c by Liam Girdwood
 
 #include <linux/clk.h>
 #include <linux/delay.h>
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 605055964529..353237025514 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -1,17 +1,13 @@
-/*
- * ak4642.c  --  AK4642/AK4643 ALSA Soc Audio driver
- *
- * Copyright (C) 2009 Renesas Solutions Corp.
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * Based on wm8731.c by Richard Purdie
- * Based on ak4535.c by Richard Purdie
- * Based on wm8753.c by Liam Girdwood
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ak4642.c  --  AK4642/AK4643 ALSA Soc Audio driver
+//
+// Copyright (C) 2009 Renesas Solutions Corp.
+// Kuninori Morimoto <morimoto.kuninori@renesas.com>
+//
+// Based on wm8731.c by Richard Purdie
+// Based on ak4535.c by Richard Purdie
+// Based on wm8753.c by Liam Girdwood
 
 /* ** CAUTION **
  *
@@ -709,4 +705,4 @@ module_i2c_driver(ak4642_i2c_driver);
 
 MODULE_DESCRIPTION("Soc AK4642 driver");
 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c
index f4ed5cc40661..448bb90c9c8e 100644
--- a/sound/soc/codecs/ak5558.c
+++ b/sound/soc/codecs/ak5558.c
@@ -322,13 +322,13 @@ static int __maybe_unused ak5558_runtime_resume(struct device *dev)
 	return regcache_sync(ak5558->regmap);
 }
 
-const struct dev_pm_ops ak5558_pm = {
+static const struct dev_pm_ops ak5558_pm = {
 	SET_RUNTIME_PM_OPS(ak5558_runtime_suspend, ak5558_runtime_resume, NULL)
 	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
 				pm_runtime_force_resume)
 };
 
-struct snd_soc_component_driver soc_codec_dev_ak5558 = {
+static const struct snd_soc_component_driver soc_codec_dev_ak5558 = {
 	.probe			= ak5558_probe,
 	.remove			= ak5558_remove,
 	.controls		= ak5558_snd_controls,
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 2a7a4168c072..3c266eeb89bf 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -219,7 +219,7 @@ static bool cs4270_reg_is_volatile(struct device *dev, unsigned int reg)
 {
 	/* Unreadable registers are considered volatile */
 	if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG))
-		return 1;
+		return true;
 
 	return reg == CS4270_CHIPID;
 }
diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index 196e9c343aeb..45e50fe3bf25 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -235,6 +235,9 @@ ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE),
 
 ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
+
+WM_ADSP_FW_CONTROL("DSP2", 1),
+WM_ADSP_FW_CONTROL("DSP3", 2),
 };
 
 ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
@@ -1283,6 +1286,12 @@ static int cs47l24_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1);
+	if (ret != 0)
+		dev_warn(&pdev->dev,
+			 "Failed to set compressed IRQ as a wake source: %d\n",
+			 ret);
+
 	arizona_init_common(arizona);
 
 	ret = arizona_init_vol_limit(arizona);
@@ -1306,6 +1315,7 @@ static int cs47l24_probe(struct platform_device *pdev)
 err_spk_irqs:
 	arizona_free_spk_irqs(arizona);
 err_dsp_irq:
+	arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
 	arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, cs47l24);
 
 	return ret;
@@ -1323,6 +1333,7 @@ static int cs47l24_remove(struct platform_device *pdev)
 
 	arizona_free_spk_irqs(arizona);
 
+	arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
 	arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, cs47l24);
 
 	return 0;
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 07dd33b09596..ab174b5114dc 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -362,8 +362,27 @@ static int cx20442_component_probe(struct snd_soc_component *component)
 		return -ENOMEM;
 
 	cx20442->por = regulator_get(component->dev, "POR");
-	if (IS_ERR(cx20442->por))
-		dev_warn(component->dev, "failed to get the regulator");
+	if (IS_ERR(cx20442->por)) {
+		int err = PTR_ERR(cx20442->por);
+
+		dev_warn(component->dev, "failed to get POR supply (%d)", err);
+		/*
+		 * When running on a non-dt platform and requested regulator
+		 * is not available, regulator_get() never returns
+		 * -EPROBE_DEFER as it is not able to justify if the regulator
+		 * may still appear later.  On the other hand, the board can
+		 * still set full constraints flag at late_initcall in order
+		 * to instruct regulator_get() to return a dummy one if
+		 * sufficient.  Hence, if we get -ENODEV here, let's convert
+		 * it to -EPROBE_DEFER and wait for the board to decide or
+		 * let Deferred Probe infrastructure handle this error.
+		 */
+		if (err == -ENODEV)
+			err = -EPROBE_DEFER;
+		kfree(cx20442);
+		return err;
+	}
+
 	cx20442->tty = NULL;
 
 	snd_soc_component_set_drvdata(component, cx20442);
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index a664111b7184..e172913d04a4 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -1,19 +1,14 @@
-/*
- * DA7210 ALSA Soc codec driver
- *
- * Copyright (c) 2009 Dialog Semiconductor
- * Written by David Chen <Dajun.chen@diasemi.com>
- *
- * Copyright (C) 2009 Renesas Solutions Corp.
- * Cleanups by Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// DA7210 ALSA Soc codec driver
+//
+// Copyright (c) 2009 Dialog Semiconductor
+// Written by David Chen <Dajun.chen@diasemi.com>
+//
+// Copyright (C) 2009 Renesas Solutions Corp.
+// Cleanups by Kuninori Morimoto <morimoto.kuninori@renesas.com>
+//
+// Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S
 
 #include <linux/delay.h>
 #include <linux/i2c.h>
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 54cb5f24969f..92d006a5283e 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -1140,9 +1140,9 @@ static bool da7213_volatile_register(struct device *dev, unsigned int reg)
 	case DA7213_ALC_OFFSET_AUTO_M_R:
 	case DA7213_ALC_OFFSET_AUTO_U_R:
 	case DA7213_ALC_CIC_OP_LVL_DATA:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c
index a49ab751a036..2c7d5088e6f2 100644
--- a/sound/soc/codecs/da7219-aad.c
+++ b/sound/soc/codecs/da7219-aad.c
@@ -59,6 +59,7 @@ static void da7219_aad_btn_det_work(struct work_struct *work)
 		container_of(work, struct da7219_aad_priv, btn_det_work);
 	struct snd_soc_component *component = da7219_aad->component;
 	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+	struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component);
 	u8 statusa, micbias_ctrl;
 	bool micbias_up = false;
 	int retries = 0;
@@ -86,6 +87,8 @@ static void da7219_aad_btn_det_work(struct work_struct *work)
 	if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES)
 		dev_warn(component->dev, "Mic bias status check timed out");
 
+	da7219->micbias_on_event = true;
+
 	/*
 	 * Mic bias pulse required to enable mic, must be done before enabling
 	 * button detection to prevent erroneous button readings.
@@ -439,6 +442,8 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data)
 			snd_soc_component_update_bits(component, DA7219_ACCDET_CONFIG_1,
 					    DA7219_BUTTON_CONFIG_MASK, 0);
 
+			da7219->micbias_on_event = false;
+
 			/* Disable mic bias */
 			snd_soc_dapm_disable_pin(dapm, "Mic Bias");
 			snd_soc_dapm_sync(dapm);
diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c
index 980a6a8bf56d..e46e9f4bc994 100644
--- a/sound/soc/codecs/da7219.c
+++ b/sound/soc/codecs/da7219.c
@@ -768,6 +768,30 @@ static const struct snd_kcontrol_new da7219_st_out_filtr_mix_controls[] = {
  * DAPM Events
  */
 
+static int da7219_mic_pga_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
+	struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		if (da7219->micbias_on_event) {
+			/*
+			 * Delay only for first capture after bias enabled to
+			 * avoid possible DC offset related noise.
+			 */
+			da7219->micbias_on_event = false;
+			msleep(da7219->mic_pga_delay);
+		}
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
 static int da7219_dai_event(struct snd_soc_dapm_widget *w,
 			    struct snd_kcontrol *kcontrol, int event)
 {
@@ -937,12 +961,12 @@ static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("MIC"),
 
 	/* Input PGAs */
-	SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL,
-			 DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT,
-			 NULL, 0),
-	SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL,
-			 DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT,
-			 NULL, 0),
+	SND_SOC_DAPM_PGA_E("Mic PGA", DA7219_MIC_1_CTRL,
+			   DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT,
+			   NULL, 0, da7219_mic_pga_event, SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_PGA_E("Mixin PGA", DA7219_MIXIN_L_CTRL,
+			   DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT,
+			   NULL, 0, da7219_settling_event, SND_SOC_DAPM_POST_PMU),
 
 	/* Input Filters */
 	SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT,
@@ -1847,6 +1871,14 @@ static void da7219_handle_pdata(struct snd_soc_component *component)
 
 		snd_soc_component_write(component, DA7219_MICBIAS_CTRL, micbias_lvl);
 
+		/*
+		 * Calculate delay required to compensate for DC offset in
+		 * Mic PGA, based on Mic Bias voltage.
+		 */
+		da7219->mic_pga_delay =  DA7219_MIC_PGA_BASE_DELAY +
+					(pdata->micbias_lvl *
+					 DA7219_MIC_PGA_OFFSET_DELAY);
+
 		/* Mic */
 		switch (pdata->mic_amp_in_sel) {
 		case DA7219_MIC_AMP_IN_SEL_DIFF:
@@ -2143,9 +2175,9 @@ static bool da7219_volatile_register(struct device *dev, unsigned int reg)
 	case DA7219_ACCDET_IRQ_EVENT_B:
 	case DA7219_ACCDET_CONFIG_8:
 	case DA7219_SYSTEM_STATUS:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h
index 1b00023e33cd..3a006862f0e7 100644
--- a/sound/soc/codecs/da7219.h
+++ b/sound/soc/codecs/da7219.h
@@ -781,8 +781,10 @@
 #define DA7219_SYS_STAT_CHECK_DELAY	50
 
 /* Power up/down Delays */
-#define DA7219_SETTLING_DELAY	40
-#define DA7219_MIN_GAIN_DELAY	30
+#define DA7219_SETTLING_DELAY		40
+#define DA7219_MIN_GAIN_DELAY		30
+#define DA7219_MIC_PGA_BASE_DELAY	100
+#define DA7219_MIC_PGA_OFFSET_DELAY	40
 
 enum da7219_clk_src {
 	DA7219_CLKSRC_MCLK = 0,
@@ -828,6 +830,8 @@ struct da7219_priv {
 
 	bool master;
 	bool alc_en;
+	bool micbias_on_event;
+	unsigned int mic_pga_delay;
 	u8 gain_ramp_ctrl;
 };
 
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index afdf90c78884..f6a7bf9560e7 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -1041,9 +1041,9 @@ static bool da9055_volatile_register(struct device *dev,
 	case DA9055_HP_R_GAIN_STATUS:
 	case DA9055_LINE_GAIN_STATUS:
 	case DA9055_ALC_CIC_OP_LVL_DATA:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c
index 58515bb1a303..6d7bca7b78ca 100644
--- a/sound/soc/codecs/es7134.c
+++ b/sound/soc/codecs/es7134.c
@@ -17,6 +17,7 @@
  * in the file called COPYING.
  */
 
+#include <linux/of_platform.h>
 #include <linux/module.h>
 #include <sound/soc.h>
 
@@ -24,6 +25,82 @@
  * The everest 7134 is a very simple DA converter with no register
  */
 
+struct es7134_clock_mode {
+	unsigned int rate_min;
+	unsigned int rate_max;
+	unsigned int *mclk_fs;
+	unsigned int mclk_fs_num;
+};
+
+struct es7134_chip {
+	struct snd_soc_dai_driver *dai_drv;
+	const struct es7134_clock_mode *modes;
+	unsigned int mode_num;
+	const struct snd_soc_dapm_widget *extra_widgets;
+	unsigned int extra_widget_num;
+	const struct snd_soc_dapm_route *extra_routes;
+	unsigned int extra_route_num;
+};
+
+struct es7134_data {
+	unsigned int mclk;
+	const struct es7134_chip *chip;
+};
+
+static int es7134_check_mclk(struct snd_soc_dai *dai,
+			     struct es7134_data *priv,
+			     unsigned int rate)
+{
+	unsigned int mfs = priv->mclk / rate;
+	int i, j;
+
+	for (i = 0; i < priv->chip->mode_num; i++) {
+		const struct es7134_clock_mode *mode = &priv->chip->modes[i];
+
+		if (rate < mode->rate_min || rate > mode->rate_max)
+			continue;
+
+		for (j = 0; j < mode->mclk_fs_num; j++) {
+			if (mode->mclk_fs[j] == mfs)
+				return 0;
+		}
+
+		dev_err(dai->dev, "unsupported mclk_fs %u for rate %u\n",
+			mfs, rate);
+		return -EINVAL;
+	}
+
+	/* should not happen */
+	dev_err(dai->dev, "unsupported rate: %u\n", rate);
+	return -EINVAL;
+}
+
+static int es7134_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct es7134_data *priv = snd_soc_dai_get_drvdata(dai);
+
+	/* mclk has not been provided, assume it is OK */
+	if (!priv->mclk)
+		return 0;
+
+	return es7134_check_mclk(dai, priv, params_rate(params));
+}
+
+static int es7134_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+			     unsigned int freq, int dir)
+{
+	struct es7134_data *priv = snd_soc_dai_get_drvdata(dai);
+
+	if (dir == SND_SOC_CLOCK_IN && clk_id == 0) {
+		priv->mclk = freq;
+		return 0;
+	}
+
+	return -ENOTSUPP;
+}
+
 static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 {
 	fmt &= (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK |
@@ -38,8 +115,38 @@ static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	return 0;
 }
 
+static int es7134_component_probe(struct snd_soc_component *c)
+{
+	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(c);
+	struct es7134_data *priv = snd_soc_component_get_drvdata(c);
+	const struct es7134_chip *chip = priv->chip;
+	int ret;
+
+	if (chip->extra_widget_num) {
+		ret = snd_soc_dapm_new_controls(dapm, chip->extra_widgets,
+						chip->extra_widget_num);
+		if (ret) {
+			dev_err(c->dev, "failed to add extra widgets\n");
+			return ret;
+		}
+	}
+
+	if (chip->extra_route_num) {
+		ret = snd_soc_dapm_add_routes(dapm, chip->extra_routes,
+					      chip->extra_route_num);
+		if (ret) {
+			dev_err(c->dev, "failed to add extra routes\n");
+			return ret;
+		}
+	}
+
+	return 0;
+}
+
 static const struct snd_soc_dai_ops es7134_dai_ops = {
 	.set_fmt	= es7134_set_fmt,
+	.hw_params	= es7134_hw_params,
+	.set_sysclk	= es7134_set_sysclk,
 };
 
 static struct snd_soc_dai_driver es7134_dai = {
@@ -48,7 +155,11 @@ static struct snd_soc_dai_driver es7134_dai = {
 		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000,
+		.rates = (SNDRV_PCM_RATE_8000_48000 |
+			  SNDRV_PCM_RATE_88200      |
+			  SNDRV_PCM_RATE_96000      |
+			  SNDRV_PCM_RATE_176400     |
+			  SNDRV_PCM_RATE_192000),
 		.formats = (SNDRV_PCM_FMTBIT_S16_LE  |
 			    SNDRV_PCM_FMTBIT_S18_3LE |
 			    SNDRV_PCM_FMTBIT_S20_3LE |
@@ -58,18 +169,56 @@ static struct snd_soc_dai_driver es7134_dai = {
 	.ops = &es7134_dai_ops,
 };
 
+static const struct es7134_clock_mode es7134_modes[] = {
+	{
+		/* Single speed mode */
+		.rate_min = 8000,
+		.rate_max = 50000,
+		.mclk_fs = (unsigned int[]) { 256, 384, 512, 768, 1024 },
+		.mclk_fs_num = 5,
+	}, {
+		/* Double speed mode */
+		.rate_min = 84000,
+		.rate_max = 100000,
+		.mclk_fs = (unsigned int[]) { 128, 192, 256, 384, 512 },
+		.mclk_fs_num = 5,
+	}, {
+		/* Quad speed mode */
+		.rate_min = 167000,
+		.rate_max = 192000,
+		.mclk_fs = (unsigned int[]) { 128, 192, 256 },
+		.mclk_fs_num = 3,
+	},
+};
+
+/* Digital I/O are also supplied by VDD on the es7134 */
+static const struct snd_soc_dapm_route es7134_extra_routes[] = {
+	{ "Playback", NULL, "VDD", }
+};
+
+static const struct es7134_chip es7134_chip = {
+	.dai_drv = &es7134_dai,
+	.modes = es7134_modes,
+	.mode_num = ARRAY_SIZE(es7134_modes),
+	.extra_routes = es7134_extra_routes,
+	.extra_route_num = ARRAY_SIZE(es7134_extra_routes),
+};
+
 static const struct snd_soc_dapm_widget es7134_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("AOUTL"),
 	SND_SOC_DAPM_OUTPUT("AOUTR"),
 	SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("VDD", 0, 0),
 };
 
 static const struct snd_soc_dapm_route es7134_dapm_routes[] = {
 	{ "AOUTL", NULL, "DAC" },
 	{ "AOUTR", NULL, "DAC" },
+	{ "DAC", NULL, "VDD" },
 };
 
 static const struct snd_soc_component_driver es7134_component_driver = {
+	.probe			= es7134_component_probe,
 	.dapm_widgets		= es7134_dapm_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(es7134_dapm_widgets),
 	.dapm_routes		= es7134_dapm_routes,
@@ -80,17 +229,87 @@ static const struct snd_soc_component_driver es7134_component_driver = {
 	.non_legacy_dai_naming	= 1,
 };
 
+static struct snd_soc_dai_driver es7154_dai = {
+	.name = "es7154-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = (SNDRV_PCM_RATE_8000_48000 |
+			  SNDRV_PCM_RATE_88200      |
+			  SNDRV_PCM_RATE_96000),
+		.formats = (SNDRV_PCM_FMTBIT_S16_LE  |
+			    SNDRV_PCM_FMTBIT_S18_3LE |
+			    SNDRV_PCM_FMTBIT_S20_3LE |
+			    SNDRV_PCM_FMTBIT_S24_3LE |
+			    SNDRV_PCM_FMTBIT_S24_LE),
+	},
+	.ops = &es7134_dai_ops,
+};
+
+static const struct es7134_clock_mode es7154_modes[] = {
+	{
+		/* Single speed mode */
+		.rate_min = 8000,
+		.rate_max = 50000,
+		.mclk_fs = (unsigned int[]) { 32, 64, 128, 192, 256,
+					      384, 512, 768, 1024 },
+		.mclk_fs_num = 9,
+	}, {
+		/* Double speed mode */
+		.rate_min = 84000,
+		.rate_max = 100000,
+		.mclk_fs = (unsigned int[]) { 128, 192, 256, 384, 512,
+					      768, 1024},
+		.mclk_fs_num = 7,
+	}
+};
+
+/* Es7154 has a separate supply for digital I/O  */
+static const struct snd_soc_dapm_widget es7154_extra_widgets[] = {
+	SND_SOC_DAPM_REGULATOR_SUPPLY("PVDD", 0, 0),
+};
+
+static const struct snd_soc_dapm_route es7154_extra_routes[] = {
+	{ "Playback", NULL, "PVDD", }
+};
+
+static const struct es7134_chip es7154_chip = {
+	.dai_drv = &es7154_dai,
+	.modes = es7154_modes,
+	.mode_num = ARRAY_SIZE(es7154_modes),
+	.extra_routes = es7154_extra_routes,
+	.extra_route_num = ARRAY_SIZE(es7154_extra_routes),
+	.extra_widgets = es7154_extra_widgets,
+	.extra_widget_num = ARRAY_SIZE(es7154_extra_widgets),
+};
+
 static int es7134_probe(struct platform_device *pdev)
 {
+	struct device *dev = &pdev->dev;
+	struct es7134_data *priv;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, priv);
+
+	priv->chip = of_device_get_match_data(dev);
+	if (!priv->chip) {
+		dev_err(dev, "failed to match device\n");
+		return -ENODEV;
+	}
+
 	return devm_snd_soc_register_component(&pdev->dev,
 				      &es7134_component_driver,
-				      &es7134_dai, 1);
+				      priv->chip->dai_drv, 1);
 }
 
 #ifdef CONFIG_OF
 static const struct of_device_id es7134_ids[] = {
-	{ .compatible = "everest,es7134", },
-	{ .compatible = "everest,es7144", },
+	{ .compatible = "everest,es7134", .data = &es7134_chip },
+	{ .compatible = "everest,es7144", .data = &es7134_chip },
+	{ .compatible = "everest,es7154", .data = &es7154_chip },
 	{ }
 };
 MODULE_DEVICE_TABLE(of, es7134_ids);
diff --git a/sound/soc/codecs/es7241.c b/sound/soc/codecs/es7241.c
new file mode 100644
index 000000000000..87991bd4acef
--- /dev/null
+++ b/sound/soc/codecs/es7241.c
@@ -0,0 +1,322 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/gpio/consumer.h>
+#include <linux/of_platform.h>
+#include <linux/module.h>
+#include <sound/soc.h>
+
+struct es7241_clock_mode {
+	unsigned int rate_min;
+	unsigned int rate_max;
+	unsigned int *slv_mfs;
+	unsigned int slv_mfs_num;
+	unsigned int mst_mfs;
+	unsigned int mst_m0:1;
+	unsigned int mst_m1:1;
+};
+
+struct es7241_chip {
+	const struct es7241_clock_mode *modes;
+	unsigned int mode_num;
+};
+
+struct es7241_data {
+	struct gpio_desc *reset;
+	struct gpio_desc *m0;
+	struct gpio_desc *m1;
+	unsigned int fmt;
+	unsigned int mclk;
+	bool is_slave;
+	const struct es7241_chip *chip;
+};
+
+static void es7241_set_mode(struct es7241_data *priv,  int m0, int m1)
+{
+	/* put the device in reset */
+	gpiod_set_value_cansleep(priv->reset, 0);
+
+	/* set the mode */
+	gpiod_set_value_cansleep(priv->m0, m0);
+	gpiod_set_value_cansleep(priv->m1, m1);
+
+	/* take the device out of reset - datasheet does not specify a delay */
+	gpiod_set_value_cansleep(priv->reset, 1);
+}
+
+static int es7241_set_slave_mode(struct es7241_data *priv,
+				 const struct es7241_clock_mode *mode,
+				 unsigned int mfs)
+{
+	int j;
+
+	if (!mfs)
+		goto out_ok;
+
+	for (j = 0; j < mode->slv_mfs_num; j++) {
+		if (mode->slv_mfs[j] == mfs)
+			goto out_ok;
+	}
+
+	return -EINVAL;
+
+out_ok:
+	es7241_set_mode(priv, 1, 1);
+	return 0;
+}
+
+static int es7241_set_master_mode(struct es7241_data *priv,
+				  const struct es7241_clock_mode *mode,
+				  unsigned int mfs)
+{
+	/*
+	 * We can't really set clock ratio, if the mclk/lrclk is different
+	 * from what we provide, then error out
+	 */
+	if (mfs && mfs != mode->mst_mfs)
+		return -EINVAL;
+
+	es7241_set_mode(priv, mode->mst_m0, mode->mst_m1);
+
+	return 0;
+}
+
+static int es7241_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct es7241_data *priv = snd_soc_dai_get_drvdata(dai);
+	unsigned int rate = params_rate(params);
+	unsigned int mfs = priv->mclk / rate;
+	int i;
+
+	for (i = 0; i < priv->chip->mode_num; i++) {
+		const struct es7241_clock_mode *mode = &priv->chip->modes[i];
+
+		if (rate < mode->rate_min || rate >= mode->rate_max)
+			continue;
+
+		if (priv->is_slave)
+			return es7241_set_slave_mode(priv, mode, mfs);
+		else
+			return es7241_set_master_mode(priv, mode, mfs);
+	}
+
+	/* should not happen */
+	dev_err(dai->dev, "unsupported rate: %u\n", rate);
+	return -EINVAL;
+}
+
+static int es7241_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+			     unsigned int freq, int dir)
+{
+	struct es7241_data *priv = snd_soc_dai_get_drvdata(dai);
+
+	if (dir == SND_SOC_CLOCK_IN && clk_id == 0) {
+		priv->mclk = freq;
+		return 0;
+	}
+
+	return -ENOTSUPP;
+}
+
+static int es7241_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct es7241_data *priv = snd_soc_dai_get_drvdata(dai);
+
+	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
+		dev_err(dai->dev, "Unsupported dai clock inversion\n");
+		return -EINVAL;
+	}
+
+	if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != priv->fmt) {
+		dev_err(dai->dev, "Invalid dai format\n");
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		priv->is_slave = true;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		priv->is_slave = false;
+		break;
+
+	default:
+		dev_err(dai->dev, "Unsupported clock configuration\n");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops es7241_dai_ops = {
+	.set_fmt	= es7241_set_fmt,
+	.hw_params	= es7241_hw_params,
+	.set_sysclk	= es7241_set_sysclk,
+};
+
+static struct snd_soc_dai_driver es7241_dai = {
+	.name = "es7241-hifi",
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = (SNDRV_PCM_FMTBIT_S16_LE  |
+			    SNDRV_PCM_FMTBIT_S24_3LE |
+			    SNDRV_PCM_FMTBIT_S24_LE),
+	},
+	.ops = &es7241_dai_ops,
+};
+
+static const struct es7241_clock_mode es7241_modes[] = {
+	{
+		/* Single speed mode */
+		.rate_min = 8000,
+		.rate_max = 50000,
+		.slv_mfs = (unsigned int[]) { 256, 384, 512, 768, 1024 },
+		.slv_mfs_num = 5,
+		.mst_mfs = 256,
+		.mst_m0 = 0,
+		.mst_m1 = 0,
+	}, {
+		/* Double speed mode */
+		.rate_min = 50000,
+		.rate_max = 100000,
+		.slv_mfs = (unsigned int[]) { 128, 192 },
+		.slv_mfs_num = 2,
+		.mst_mfs = 128,
+		.mst_m0 = 1,
+		.mst_m1 = 0,
+	}, {
+		/* Quad speed mode */
+		.rate_min = 100000,
+		.rate_max = 200000,
+		.slv_mfs = (unsigned int[]) { 64 },
+		.slv_mfs_num = 1,
+		.mst_mfs = 64,
+		.mst_m0 = 0,
+		.mst_m1 = 1,
+	},
+};
+
+static const struct es7241_chip es7241_chip = {
+	.modes = es7241_modes,
+	.mode_num = ARRAY_SIZE(es7241_modes),
+};
+
+static const struct snd_soc_dapm_widget es7241_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("AINL"),
+	SND_SOC_DAPM_INPUT("AINR"),
+	SND_SOC_DAPM_DAC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("VDDP", 0, 0),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("VDDD", 0, 0),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("VDDA", 0, 0),
+};
+
+static const struct snd_soc_dapm_route es7241_dapm_routes[] = {
+	{ "ADC", NULL, "AINL", },
+	{ "ADC", NULL, "AINR", },
+	{ "ADC", NULL, "VDDA", },
+	{ "Capture", NULL, "VDDP", },
+	{ "Capture", NULL, "VDDD", },
+};
+
+static const struct snd_soc_component_driver es7241_component_driver = {
+	.dapm_widgets		= es7241_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(es7241_dapm_widgets),
+	.dapm_routes		= es7241_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(es7241_dapm_routes),
+	.idle_bias_on		= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
+static void es7241_parse_fmt(struct device *dev, struct es7241_data *priv)
+{
+	bool is_leftj;
+
+	/*
+	 * The format is given by a pull resistor on the SDOUT pin:
+	 * pull-up for i2s, pull-down for left justified.
+	 */
+	is_leftj = of_property_read_bool(dev->of_node,
+					 "everest,sdout-pull-down");
+	if (is_leftj)
+		priv->fmt = SND_SOC_DAIFMT_LEFT_J;
+	else
+		priv->fmt = SND_SOC_DAIFMT_I2S;
+}
+
+static int es7241_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct es7241_data *priv;
+	int err;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, priv);
+
+	priv->chip = of_device_get_match_data(dev);
+	if (!priv->chip) {
+		dev_err(dev, "failed to match device\n");
+		return -ENODEV;
+	}
+
+	es7241_parse_fmt(dev, priv);
+
+	priv->reset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW);
+	if (IS_ERR(priv->reset)) {
+		err = PTR_ERR(priv->reset);
+		if (err != -EPROBE_DEFER)
+			dev_err(dev, "Failed to get 'reset' gpio: %d", err);
+		return err;
+	}
+
+	priv->m0 = devm_gpiod_get_optional(dev, "m0", GPIOD_OUT_LOW);
+	if (IS_ERR(priv->m0)) {
+		err = PTR_ERR(priv->m0);
+		if (err != -EPROBE_DEFER)
+			dev_err(dev, "Failed to get 'm0' gpio: %d", err);
+		return err;
+	}
+
+	priv->m1 = devm_gpiod_get_optional(dev, "m1", GPIOD_OUT_LOW);
+	if (IS_ERR(priv->m1)) {
+		err = PTR_ERR(priv->m1);
+		if (err != -EPROBE_DEFER)
+			dev_err(dev, "Failed to get 'm1' gpio: %d", err);
+		return err;
+	}
+
+	return devm_snd_soc_register_component(&pdev->dev,
+				      &es7241_component_driver,
+				      &es7241_dai, 1);
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id es7241_ids[] = {
+	{ .compatible = "everest,es7241", .data = &es7241_chip },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, es7241_ids);
+#endif
+
+static struct platform_driver es7241_driver = {
+	.driver = {
+		.name = "es7241",
+		.of_match_table = of_match_ptr(es7241_ids),
+	},
+	.probe = es7241_probe,
+};
+
+module_platform_driver(es7241_driver);
+
+MODULE_DESCRIPTION("ASoC ES7241 audio codec driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 84f7a7a36e4b..7b8533abf637 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -85,7 +85,7 @@ struct hdac_hdmi_pin {
 	bool mst_capable;
 	struct hdac_hdmi_port *ports;
 	int num_ports;
-	struct hdac_ext_device *edev;
+	struct hdac_device *hdev;
 };
 
 struct hdac_hdmi_port {
@@ -126,6 +126,9 @@ struct hdac_hdmi_drv_data {
 };
 
 struct hdac_hdmi_priv {
+	struct hdac_device *hdev;
+	struct snd_soc_component *component;
+	struct snd_card *card;
 	struct hdac_hdmi_dai_port_map dai_map[HDA_MAX_CVTS];
 	struct list_head pin_list;
 	struct list_head cvt_list;
@@ -139,7 +142,7 @@ struct hdac_hdmi_priv {
 	struct snd_soc_dai_driver *dai_drv;
 };
 
-#define hdev_to_hdmi_priv(_hdev) ((to_ehdac_device(_hdev))->private_data)
+#define hdev_to_hdmi_priv(_hdev) dev_get_drvdata(&(_hdev)->dev)
 
 static struct hdac_hdmi_pcm *
 hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi,
@@ -158,7 +161,7 @@ hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi,
 static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm,
 		struct hdac_hdmi_port *port, bool is_connect)
 {
-	struct hdac_ext_device *edev = port->pin->edev;
+	struct hdac_device *hdev = port->pin->hdev;
 
 	if (is_connect)
 		snd_soc_dapm_enable_pin(port->dapm, port->jack_pin);
@@ -172,7 +175,7 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm,
 		 * ports.
 		 */
 		if (pcm->jack_event == 0) {
-			dev_dbg(&edev->hdev.dev,
+			dev_dbg(&hdev->dev,
 					"jack report for pcm=%d\n",
 					pcm->pcm_id);
 			snd_soc_jack_report(pcm->jack, SND_JACK_AVOUT,
@@ -198,19 +201,18 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm,
 /*
  * Get the no devices that can be connected to a port on the Pin widget.
  */
-static int hdac_hdmi_get_port_len(struct hdac_ext_device *edev, hda_nid_t nid)
+static int hdac_hdmi_get_port_len(struct hdac_device *hdev, hda_nid_t nid)
 {
 	unsigned int caps;
 	unsigned int type, param;
 
-	caps = get_wcaps(&edev->hdev, nid);
+	caps = get_wcaps(hdev, nid);
 	type = get_wcaps_type(caps);
 
 	if (!(caps & AC_WCAP_DIGITAL) || (type != AC_WID_PIN))
 		return 0;
 
-	param = snd_hdac_read_parm_uncached(&edev->hdev, nid,
-					AC_PAR_DEVLIST_LEN);
+	param = snd_hdac_read_parm_uncached(hdev, nid, AC_PAR_DEVLIST_LEN);
 	if (param == -1)
 		return param;
 
@@ -222,10 +224,10 @@ static int hdac_hdmi_get_port_len(struct hdac_ext_device *edev, hda_nid_t nid)
  * id selected on the pin. Return 0 means the first port entry
  * is selected or MST is not supported.
  */
-static int hdac_hdmi_port_select_get(struct hdac_ext_device *edev,
+static int hdac_hdmi_port_select_get(struct hdac_device *hdev,
 					struct hdac_hdmi_port *port)
 {
-	return snd_hdac_codec_read(&edev->hdev, port->pin->nid,
+	return snd_hdac_codec_read(hdev, port->pin->nid,
 				0, AC_VERB_GET_DEVICE_SEL, 0);
 }
 
@@ -233,7 +235,7 @@ static int hdac_hdmi_port_select_get(struct hdac_ext_device *edev,
  * Sets the selected port entry for the configuring Pin widget verb.
  * returns error if port set is not equal to port get otherwise success
  */
-static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev,
+static int hdac_hdmi_port_select_set(struct hdac_device *hdev,
 					struct hdac_hdmi_port *port)
 {
 	int num_ports;
@@ -242,8 +244,7 @@ static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev,
 		return 0;
 
 	/* AC_PAR_DEVLIST_LEN is 0 based. */
-	num_ports = hdac_hdmi_get_port_len(edev, port->pin->nid);
-
+	num_ports = hdac_hdmi_get_port_len(hdev, port->pin->nid);
 	if (num_ports < 0)
 		return -EIO;
 	/*
@@ -253,13 +254,13 @@ static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev,
 	if (num_ports + 1  < port->id)
 		return 0;
 
-	snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0,
+	snd_hdac_codec_write(hdev, port->pin->nid, 0,
 			AC_VERB_SET_DEVICE_SEL, port->id);
 
-	if (port->id != hdac_hdmi_port_select_get(edev, port))
+	if (port->id != hdac_hdmi_port_select_get(hdev, port))
 		return -EIO;
 
-	dev_dbg(&edev->hdev.dev, "Selected the port=%d\n", port->id);
+	dev_dbg(&hdev->dev, "Selected the port=%d\n", port->id);
 
 	return 0;
 }
@@ -277,13 +278,6 @@ static struct hdac_hdmi_pcm *get_hdmi_pcm_from_id(struct hdac_hdmi_priv *hdmi,
 	return NULL;
 }
 
-static inline struct hdac_ext_device *to_hda_ext_device(struct device *dev)
-{
-	struct hdac_device *hdev = dev_to_hdac_dev(dev);
-
-	return to_ehdac_device(hdev);
-}
-
 static unsigned int sad_format(const u8 *sad)
 {
 	return ((sad[0] >> 0x3) & 0x1f);
@@ -324,15 +318,13 @@ format_constraint:
 }
 
 static void
-hdac_hdmi_set_dip_index(struct hdac_ext_device *edev, hda_nid_t pin_nid,
+hdac_hdmi_set_dip_index(struct hdac_device *hdev, hda_nid_t pin_nid,
 				int packet_index, int byte_index)
 {
 	int val;
 
 	val = (packet_index << 5) | (byte_index & 0x1f);
-
-	snd_hdac_codec_write(&edev->hdev, pin_nid, 0,
-				AC_VERB_SET_HDMI_DIP_INDEX, val);
+	snd_hdac_codec_write(hdev, pin_nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val);
 }
 
 struct dp_audio_infoframe {
@@ -347,14 +339,14 @@ struct dp_audio_infoframe {
 	u8 LFEPBL01_LSV36_DM_INH7;
 };
 
-static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *edev,
+static int hdac_hdmi_setup_audio_infoframe(struct hdac_device *hdev,
 		   struct hdac_hdmi_pcm *pcm, struct hdac_hdmi_port *port)
 {
 	uint8_t buffer[HDMI_INFOFRAME_HEADER_SIZE + HDMI_AUDIO_INFOFRAME_SIZE];
 	struct hdmi_audio_infoframe frame;
 	struct hdac_hdmi_pin *pin = port->pin;
 	struct dp_audio_infoframe dp_ai;
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	struct hdac_hdmi_cvt *cvt = pcm->cvt;
 	u8 *dip;
 	int ret;
@@ -363,11 +355,11 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *edev,
 	u8 conn_type;
 	int channels, ca;
 
-	ca = snd_hdac_channel_allocation(&edev->hdev, port->eld.info.spk_alloc,
+	ca = snd_hdac_channel_allocation(hdev, port->eld.info.spk_alloc,
 			pcm->channels, pcm->chmap_set, true, pcm->chmap);
 
 	channels = snd_hdac_get_active_channels(ca);
-	hdmi->chmap.ops.set_channel_count(&edev->hdev, cvt->nid, channels);
+	hdmi->chmap.ops.set_channel_count(hdev, cvt->nid, channels);
 
 	snd_hdac_setup_channel_mapping(&hdmi->chmap, pin->nid, false, ca,
 				pcm->channels, pcm->chmap, pcm->chmap_set);
@@ -400,32 +392,31 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *edev,
 		break;
 
 	default:
-		dev_err(&edev->hdev.dev, "Invalid connection type: %d\n",
-						conn_type);
+		dev_err(&hdev->dev, "Invalid connection type: %d\n", conn_type);
 		return -EIO;
 	}
 
 	/* stop infoframe transmission */
-	hdac_hdmi_set_dip_index(edev, pin->nid, 0x0, 0x0);
-	snd_hdac_codec_write(&edev->hdev, pin->nid, 0,
+	hdac_hdmi_set_dip_index(hdev, pin->nid, 0x0, 0x0);
+	snd_hdac_codec_write(hdev, pin->nid, 0,
 			AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_DISABLE);
 
 
 	/*  Fill infoframe. Index auto-incremented */
-	hdac_hdmi_set_dip_index(edev, pin->nid, 0x0, 0x0);
+	hdac_hdmi_set_dip_index(hdev, pin->nid, 0x0, 0x0);
 	if (conn_type == DRM_ELD_CONN_TYPE_HDMI) {
 		for (i = 0; i < sizeof(buffer); i++)
-			snd_hdac_codec_write(&edev->hdev, pin->nid, 0,
+			snd_hdac_codec_write(hdev, pin->nid, 0,
 				AC_VERB_SET_HDMI_DIP_DATA, buffer[i]);
 	} else {
 		for (i = 0; i < sizeof(dp_ai); i++)
-			snd_hdac_codec_write(&edev->hdev, pin->nid, 0,
+			snd_hdac_codec_write(hdev, pin->nid, 0,
 				AC_VERB_SET_HDMI_DIP_DATA, dip[i]);
 	}
 
 	/* Start infoframe */
-	hdac_hdmi_set_dip_index(edev, pin->nid, 0x0, 0x0);
-	snd_hdac_codec_write(&edev->hdev, pin->nid, 0,
+	hdac_hdmi_set_dip_index(hdev, pin->nid, 0x0, 0x0);
+	snd_hdac_codec_write(hdev, pin->nid, 0,
 			AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_BEST);
 
 	return 0;
@@ -435,12 +426,12 @@ static int hdac_hdmi_set_tdm_slot(struct snd_soc_dai *dai,
 		unsigned int tx_mask, unsigned int rx_mask,
 		int slots, int slot_width)
 {
-	struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai);
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai);
+	struct hdac_device *hdev = hdmi->hdev;
 	struct hdac_hdmi_dai_port_map *dai_map;
 	struct hdac_hdmi_pcm *pcm;
 
-	dev_dbg(&edev->hdev.dev, "%s: strm_tag: %d\n", __func__, tx_mask);
+	dev_dbg(&hdev->dev, "%s: strm_tag: %d\n", __func__, tx_mask);
 
 	dai_map = &hdmi->dai_map[dai->id];
 
@@ -455,8 +446,8 @@ static int hdac_hdmi_set_tdm_slot(struct snd_soc_dai *dai,
 static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *hparams, struct snd_soc_dai *dai)
 {
-	struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai);
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai);
+	struct hdac_device *hdev = hdmi->hdev;
 	struct hdac_hdmi_dai_port_map *dai_map;
 	struct hdac_hdmi_port *port;
 	struct hdac_hdmi_pcm *pcm;
@@ -469,7 +460,7 @@ static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream,
 		return -ENODEV;
 
 	if ((!port->eld.monitor_present) || (!port->eld.eld_valid)) {
-		dev_err(&edev->hdev.dev,
+		dev_err(&hdev->dev,
 			"device is not configured for this pin:port%d:%d\n",
 					port->pin->nid, port->id);
 		return -ENODEV;
@@ -489,28 +480,28 @@ static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int hdac_hdmi_query_port_connlist(struct hdac_ext_device *edev,
+static int hdac_hdmi_query_port_connlist(struct hdac_device *hdev,
 					struct hdac_hdmi_pin *pin,
 					struct hdac_hdmi_port *port)
 {
-	if (!(get_wcaps(&edev->hdev, pin->nid) & AC_WCAP_CONN_LIST)) {
-		dev_warn(&edev->hdev.dev,
+	if (!(get_wcaps(hdev, pin->nid) & AC_WCAP_CONN_LIST)) {
+		dev_warn(&hdev->dev,
 			"HDMI: pin %d wcaps %#x does not support connection list\n",
-			pin->nid, get_wcaps(&edev->hdev, pin->nid));
+			pin->nid, get_wcaps(hdev, pin->nid));
 		return -EINVAL;
 	}
 
-	if (hdac_hdmi_port_select_set(edev, port) < 0)
+	if (hdac_hdmi_port_select_set(hdev, port) < 0)
 		return -EIO;
 
-	port->num_mux_nids = snd_hdac_get_connections(&edev->hdev, pin->nid,
+	port->num_mux_nids = snd_hdac_get_connections(hdev, pin->nid,
 			port->mux_nids, HDA_MAX_CONNECTIONS);
 	if (port->num_mux_nids == 0)
-		dev_warn(&edev->hdev.dev,
+		dev_warn(&hdev->dev,
 			"No connections found for pin:port %d:%d\n",
 						pin->nid, port->id);
 
-	dev_dbg(&edev->hdev.dev, "num_mux_nids %d for pin:port %d:%d\n",
+	dev_dbg(&hdev->dev, "num_mux_nids %d for pin:port %d:%d\n",
 			port->num_mux_nids, pin->nid, port->id);
 
 	return port->num_mux_nids;
@@ -526,7 +517,7 @@ static int hdac_hdmi_query_port_connlist(struct hdac_ext_device *edev,
  * connected.
  */
 static struct hdac_hdmi_port *hdac_hdmi_get_port_from_cvt(
-			struct hdac_ext_device *edev,
+			struct hdac_device *hdev,
 			struct hdac_hdmi_priv *hdmi,
 			struct hdac_hdmi_cvt *cvt)
 {
@@ -541,7 +532,7 @@ static struct hdac_hdmi_port *hdac_hdmi_get_port_from_cvt(
 
 			list_for_each_entry(port, &pcm->port_list, head) {
 				mutex_lock(&pcm->lock);
-				ret = hdac_hdmi_query_port_connlist(edev,
+				ret = hdac_hdmi_query_port_connlist(hdev,
 							port->pin, port);
 				mutex_unlock(&pcm->lock);
 				if (ret < 0)
@@ -568,8 +559,8 @@ static struct hdac_hdmi_port *hdac_hdmi_get_port_from_cvt(
 static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream,
 			struct snd_soc_dai *dai)
 {
-	struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai);
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai);
+	struct hdac_device *hdev = hdmi->hdev;
 	struct hdac_hdmi_dai_port_map *dai_map;
 	struct hdac_hdmi_cvt *cvt;
 	struct hdac_hdmi_port *port;
@@ -578,7 +569,7 @@ static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream,
 	dai_map = &hdmi->dai_map[dai->id];
 
 	cvt = dai_map->cvt;
-	port = hdac_hdmi_get_port_from_cvt(edev, hdmi, cvt);
+	port = hdac_hdmi_get_port_from_cvt(hdev, hdmi, cvt);
 
 	/*
 	 * To make PA and other userland happy.
@@ -589,7 +580,7 @@ static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream,
 	if ((!port->eld.monitor_present) ||
 			(!port->eld.eld_valid)) {
 
-		dev_warn(&edev->hdev.dev,
+		dev_warn(&hdev->dev,
 			"Failed: present?:%d ELD valid?:%d pin:port: %d:%d\n",
 			port->eld.monitor_present, port->eld.eld_valid,
 			port->pin->nid, port->id);
@@ -611,8 +602,7 @@ static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream,
 static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
-	struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai);
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai);
 	struct hdac_hdmi_dai_port_map *dai_map;
 	struct hdac_hdmi_pcm *pcm;
 
@@ -695,10 +685,10 @@ static void hdac_hdmi_fill_route(struct snd_soc_dapm_route *route,
 	route->connected = handler;
 }
 
-static struct hdac_hdmi_pcm *hdac_hdmi_get_pcm(struct hdac_ext_device *edev,
+static struct hdac_hdmi_pcm *hdac_hdmi_get_pcm(struct hdac_device *hdev,
 					struct hdac_hdmi_port *port)
 {
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	struct hdac_hdmi_pcm *pcm = NULL;
 	struct hdac_hdmi_port *p;
 
@@ -715,33 +705,32 @@ static struct hdac_hdmi_pcm *hdac_hdmi_get_pcm(struct hdac_ext_device *edev,
 	return NULL;
 }
 
-static void hdac_hdmi_set_power_state(struct hdac_ext_device *edev,
+static void hdac_hdmi_set_power_state(struct hdac_device *hdev,
 			     hda_nid_t nid, unsigned int pwr_state)
 {
 	int count;
 	unsigned int state;
 
-	if (get_wcaps(&edev->hdev, nid) & AC_WCAP_POWER) {
-		if (!snd_hdac_check_power_state(&edev->hdev, nid, pwr_state)) {
+	if (get_wcaps(hdev, nid) & AC_WCAP_POWER) {
+		if (!snd_hdac_check_power_state(hdev, nid, pwr_state)) {
 			for (count = 0; count < 10; count++) {
-				snd_hdac_codec_read(&edev->hdev, nid, 0,
+				snd_hdac_codec_read(hdev, nid, 0,
 						AC_VERB_SET_POWER_STATE,
 						pwr_state);
-				state = snd_hdac_sync_power_state(&edev->hdev,
+				state = snd_hdac_sync_power_state(hdev,
 						nid, pwr_state);
 				if (!(state & AC_PWRST_ERROR))
 					break;
 			}
 		}
-
 	}
 }
 
-static void hdac_hdmi_set_amp(struct hdac_ext_device *edev,
+static void hdac_hdmi_set_amp(struct hdac_device *hdev,
 				   hda_nid_t nid, int val)
 {
-	if (get_wcaps(&edev->hdev, nid) & AC_WCAP_OUT_AMP)
-		snd_hdac_codec_write(&edev->hdev, nid, 0,
+	if (get_wcaps(hdev, nid) & AC_WCAP_OUT_AMP)
+		snd_hdac_codec_write(hdev, nid, 0,
 					AC_VERB_SET_AMP_GAIN_MUTE, val);
 }
 
@@ -750,40 +739,40 @@ static int hdac_hdmi_pin_output_widget_event(struct snd_soc_dapm_widget *w,
 					struct snd_kcontrol *kc, int event)
 {
 	struct hdac_hdmi_port *port = w->priv;
-	struct hdac_ext_device *edev = to_hda_ext_device(w->dapm->dev);
+	struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev);
 	struct hdac_hdmi_pcm *pcm;
 
-	dev_dbg(&edev->hdev.dev, "%s: widget: %s event: %x\n",
+	dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n",
 			__func__, w->name, event);
 
-	pcm = hdac_hdmi_get_pcm(edev, port);
+	pcm = hdac_hdmi_get_pcm(hdev, port);
 	if (!pcm)
 		return -EIO;
 
 	/* set the device if pin is mst_capable */
-	if (hdac_hdmi_port_select_set(edev, port) < 0)
+	if (hdac_hdmi_port_select_set(hdev, port) < 0)
 		return -EIO;
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
-		hdac_hdmi_set_power_state(edev, port->pin->nid, AC_PWRST_D0);
+		hdac_hdmi_set_power_state(hdev, port->pin->nid, AC_PWRST_D0);
 
 		/* Enable out path for this pin widget */
-		snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0,
+		snd_hdac_codec_write(hdev, port->pin->nid, 0,
 				AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
 
-		hdac_hdmi_set_amp(edev, port->pin->nid, AMP_OUT_UNMUTE);
+		hdac_hdmi_set_amp(hdev, port->pin->nid, AMP_OUT_UNMUTE);
 
-		return hdac_hdmi_setup_audio_infoframe(edev, pcm, port);
+		return hdac_hdmi_setup_audio_infoframe(hdev, pcm, port);
 
 	case SND_SOC_DAPM_POST_PMD:
-		hdac_hdmi_set_amp(edev, port->pin->nid, AMP_OUT_MUTE);
+		hdac_hdmi_set_amp(hdev, port->pin->nid, AMP_OUT_MUTE);
 
 		/* Disable out path for this pin widget */
-		snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0,
+		snd_hdac_codec_write(hdev, port->pin->nid, 0,
 				AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
 
-		hdac_hdmi_set_power_state(edev, port->pin->nid, AC_PWRST_D3);
+		hdac_hdmi_set_power_state(hdev, port->pin->nid, AC_PWRST_D3);
 		break;
 
 	}
@@ -795,11 +784,11 @@ static int hdac_hdmi_cvt_output_widget_event(struct snd_soc_dapm_widget *w,
 					struct snd_kcontrol *kc, int event)
 {
 	struct hdac_hdmi_cvt *cvt = w->priv;
-	struct hdac_ext_device *edev = to_hda_ext_device(w->dapm->dev);
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev);
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	struct hdac_hdmi_pcm *pcm;
 
-	dev_dbg(&edev->hdev.dev, "%s: widget: %s event: %x\n",
+	dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n",
 			__func__, w->name, event);
 
 	pcm = hdac_hdmi_get_pcm_from_cvt(hdmi, cvt);
@@ -808,29 +797,29 @@ static int hdac_hdmi_cvt_output_widget_event(struct snd_soc_dapm_widget *w,
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
-		hdac_hdmi_set_power_state(edev, cvt->nid, AC_PWRST_D0);
+		hdac_hdmi_set_power_state(hdev, cvt->nid, AC_PWRST_D0);
 
 		/* Enable transmission */
-		snd_hdac_codec_write(&edev->hdev, cvt->nid, 0,
+		snd_hdac_codec_write(hdev, cvt->nid, 0,
 			AC_VERB_SET_DIGI_CONVERT_1, 1);
 
 		/* Category Code (CC) to zero */
-		snd_hdac_codec_write(&edev->hdev, cvt->nid, 0,
+		snd_hdac_codec_write(hdev, cvt->nid, 0,
 			AC_VERB_SET_DIGI_CONVERT_2, 0);
 
-		snd_hdac_codec_write(&edev->hdev, cvt->nid, 0,
+		snd_hdac_codec_write(hdev, cvt->nid, 0,
 				AC_VERB_SET_CHANNEL_STREAMID, pcm->stream_tag);
-		snd_hdac_codec_write(&edev->hdev, cvt->nid, 0,
+		snd_hdac_codec_write(hdev, cvt->nid, 0,
 				AC_VERB_SET_STREAM_FORMAT, pcm->format);
 		break;
 
 	case SND_SOC_DAPM_POST_PMD:
-		snd_hdac_codec_write(&edev->hdev, cvt->nid, 0,
+		snd_hdac_codec_write(hdev, cvt->nid, 0,
 				AC_VERB_SET_CHANNEL_STREAMID, 0);
-		snd_hdac_codec_write(&edev->hdev, cvt->nid, 0,
+		snd_hdac_codec_write(hdev, cvt->nid, 0,
 				AC_VERB_SET_STREAM_FORMAT, 0);
 
-		hdac_hdmi_set_power_state(edev, cvt->nid, AC_PWRST_D3);
+		hdac_hdmi_set_power_state(hdev, cvt->nid, AC_PWRST_D3);
 		break;
 
 	}
@@ -842,10 +831,10 @@ static int hdac_hdmi_pin_mux_widget_event(struct snd_soc_dapm_widget *w,
 					struct snd_kcontrol *kc, int event)
 {
 	struct hdac_hdmi_port *port = w->priv;
-	struct hdac_ext_device *edev = to_hda_ext_device(w->dapm->dev);
+	struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev);
 	int mux_idx;
 
-	dev_dbg(&edev->hdev.dev, "%s: widget: %s event: %x\n",
+	dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n",
 			__func__, w->name, event);
 
 	if (!kc)
@@ -854,11 +843,11 @@ static int hdac_hdmi_pin_mux_widget_event(struct snd_soc_dapm_widget *w,
 	mux_idx = dapm_kcontrol_get_value(kc);
 
 	/* set the device if pin is mst_capable */
-	if (hdac_hdmi_port_select_set(edev, port) < 0)
+	if (hdac_hdmi_port_select_set(hdev, port) < 0)
 		return -EIO;
 
 	if (mux_idx > 0) {
-		snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0,
+		snd_hdac_codec_write(hdev, port->pin->nid, 0,
 			AC_VERB_SET_CONNECT_SEL, (mux_idx - 1));
 	}
 
@@ -877,8 +866,8 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol,
 	struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kcontrol);
 	struct snd_soc_dapm_context *dapm = w->dapm;
 	struct hdac_hdmi_port *port = w->priv;
-	struct hdac_ext_device *edev = to_hda_ext_device(dapm->dev);
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_device *hdev = dev_to_hdac_dev(dapm->dev);
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	struct hdac_hdmi_pcm *pcm = NULL;
 	const char *cvt_name =  e->texts[ucontrol->value.enumerated.item[0]];
 
@@ -931,12 +920,12 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol,
  * care of selecting the right one and leaving all other inputs selected to
  * "NONE"
  */
-static int hdac_hdmi_create_pin_port_muxs(struct hdac_ext_device *edev,
+static int hdac_hdmi_create_pin_port_muxs(struct hdac_device *hdev,
 				struct hdac_hdmi_port *port,
 				struct snd_soc_dapm_widget *widget,
 				const char *widget_name)
 {
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	struct hdac_hdmi_pin *pin = port->pin;
 	struct snd_kcontrol_new *kc;
 	struct hdac_hdmi_cvt *cvt;
@@ -948,17 +937,17 @@ static int hdac_hdmi_create_pin_port_muxs(struct hdac_ext_device *edev,
 	int i = 0;
 	int num_items = hdmi->num_cvt + 1;
 
-	kc = devm_kzalloc(&edev->hdev.dev, sizeof(*kc), GFP_KERNEL);
+	kc = devm_kzalloc(&hdev->dev, sizeof(*kc), GFP_KERNEL);
 	if (!kc)
 		return -ENOMEM;
 
-	se = devm_kzalloc(&edev->hdev.dev, sizeof(*se), GFP_KERNEL);
+	se = devm_kzalloc(&hdev->dev, sizeof(*se), GFP_KERNEL);
 	if (!se)
 		return -ENOMEM;
 
 	snprintf(kc_name, NAME_SIZE, "Pin %d port %d Input",
 						pin->nid, port->id);
-	kc->name = devm_kstrdup(&edev->hdev.dev, kc_name, GFP_KERNEL);
+	kc->name = devm_kstrdup(&hdev->dev, kc_name, GFP_KERNEL);
 	if (!kc->name)
 		return -ENOMEM;
 
@@ -976,35 +965,35 @@ static int hdac_hdmi_create_pin_port_muxs(struct hdac_ext_device *edev,
 	se->mask = roundup_pow_of_two(se->items) - 1;
 
 	sprintf(mux_items, "NONE");
-	items[i] = devm_kstrdup(&edev->hdev.dev, mux_items, GFP_KERNEL);
+	items[i] = devm_kstrdup(&hdev->dev, mux_items, GFP_KERNEL);
 	if (!items[i])
 		return -ENOMEM;
 
 	list_for_each_entry(cvt, &hdmi->cvt_list, head) {
 		i++;
 		sprintf(mux_items, "cvt %d", cvt->nid);
-		items[i] = devm_kstrdup(&edev->hdev.dev, mux_items, GFP_KERNEL);
+		items[i] = devm_kstrdup(&hdev->dev, mux_items, GFP_KERNEL);
 		if (!items[i])
 			return -ENOMEM;
 	}
 
-	se->texts = devm_kmemdup(&edev->hdev.dev, items,
+	se->texts = devm_kmemdup(&hdev->dev, items,
 			(num_items  * sizeof(char *)), GFP_KERNEL);
 	if (!se->texts)
 		return -ENOMEM;
 
-	return hdac_hdmi_fill_widget_info(&edev->hdev.dev, widget,
+	return hdac_hdmi_fill_widget_info(&hdev->dev, widget,
 			snd_soc_dapm_mux, port, widget_name, NULL, kc, 1,
 			hdac_hdmi_pin_mux_widget_event,
 			SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_REG);
 }
 
 /* Add cvt <- input <- mux route map */
-static void hdac_hdmi_add_pinmux_cvt_route(struct hdac_ext_device *edev,
+static void hdac_hdmi_add_pinmux_cvt_route(struct hdac_device *hdev,
 			struct snd_soc_dapm_widget *widgets,
 			struct snd_soc_dapm_route *route, int rindex)
 {
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	const struct snd_kcontrol_new *kc;
 	struct soc_enum *se;
 	int mux_index = hdmi->num_cvt + hdmi->num_ports;
@@ -1046,8 +1035,8 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm)
 {
 	struct snd_soc_dapm_widget *widgets;
 	struct snd_soc_dapm_route *route;
-	struct hdac_ext_device *edev = to_hda_ext_device(dapm->dev);
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_device *hdev = dev_to_hdac_dev(dapm->dev);
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	struct snd_soc_dai_driver *dai_drv = hdmi->dai_drv;
 	char widget_name[NAME_SIZE];
 	struct hdac_hdmi_cvt *cvt;
@@ -1099,7 +1088,7 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm)
 		for (j = 0; j < pin->num_ports; j++) {
 			sprintf(widget_name, "Pin%d-Port%d Mux",
 				pin->nid, pin->ports[j].id);
-			ret = hdac_hdmi_create_pin_port_muxs(edev,
+			ret = hdac_hdmi_create_pin_port_muxs(hdev,
 						&pin->ports[j], &widgets[i],
 						widget_name);
 			if (ret < 0)
@@ -1134,7 +1123,7 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm)
 		}
 	}
 
-	hdac_hdmi_add_pinmux_cvt_route(edev, widgets, route, i);
+	hdac_hdmi_add_pinmux_cvt_route(hdev, widgets, route, i);
 
 	snd_soc_dapm_new_controls(dapm, widgets,
 		((2 * hdmi->num_ports) + hdmi->num_cvt));
@@ -1146,9 +1135,9 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm)
 
 }
 
-static int hdac_hdmi_init_dai_map(struct hdac_ext_device *edev)
+static int hdac_hdmi_init_dai_map(struct hdac_device *hdev)
 {
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	struct hdac_hdmi_dai_port_map *dai_map;
 	struct hdac_hdmi_cvt *cvt;
 	int dai_id = 0;
@@ -1164,7 +1153,7 @@ static int hdac_hdmi_init_dai_map(struct hdac_ext_device *edev)
 		dai_id++;
 
 		if (dai_id == HDA_MAX_CVTS) {
-			dev_warn(&edev->hdev.dev,
+			dev_warn(&hdev->dev,
 				"Max dais supported: %d\n", dai_id);
 			break;
 		}
@@ -1173,9 +1162,9 @@ static int hdac_hdmi_init_dai_map(struct hdac_ext_device *edev)
 	return 0;
 }
 
-static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid)
+static int hdac_hdmi_add_cvt(struct hdac_device *hdev, hda_nid_t nid)
 {
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	struct hdac_hdmi_cvt *cvt;
 	char name[NAME_SIZE];
 
@@ -1190,10 +1179,10 @@ static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid)
 	list_add_tail(&cvt->head, &hdmi->cvt_list);
 	hdmi->num_cvt++;
 
-	return hdac_hdmi_query_cvt_params(&edev->hdev, cvt);
+	return hdac_hdmi_query_cvt_params(hdev, cvt);
 }
 
-static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev,
+static int hdac_hdmi_parse_eld(struct hdac_device *hdev,
 			struct hdac_hdmi_port *port)
 {
 	unsigned int ver, mnl;
@@ -1202,7 +1191,7 @@ static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev,
 						>> DRM_ELD_VER_SHIFT;
 
 	if (ver != ELD_VER_CEA_861D && ver != ELD_VER_PARTIAL) {
-		dev_err(&edev->hdev.dev, "HDMI: Unknown ELD version %d\n", ver);
+		dev_err(&hdev->dev, "HDMI: Unknown ELD version %d\n", ver);
 		return -EINVAL;
 	}
 
@@ -1210,7 +1199,7 @@ static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev,
 		DRM_ELD_MNL_MASK) >> DRM_ELD_MNL_SHIFT;
 
 	if (mnl > ELD_MAX_MNL) {
-		dev_err(&edev->hdev.dev, "HDMI: MNL Invalid %d\n", mnl);
+		dev_err(&hdev->dev, "HDMI: MNL Invalid %d\n", mnl);
 		return -EINVAL;
 	}
 
@@ -1222,8 +1211,8 @@ static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev,
 static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin,
 				    struct hdac_hdmi_port *port)
 {
-	struct hdac_ext_device *edev = pin->edev;
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_device *hdev = pin->hdev;
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	struct hdac_hdmi_pcm *pcm;
 	int size = 0;
 	int port_id = -1;
@@ -1241,14 +1230,14 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin,
 	if (pin->mst_capable)
 		port_id = port->id;
 
-	size = snd_hdac_acomp_get_eld(&edev->hdev, pin->nid, port_id,
+	size = snd_hdac_acomp_get_eld(hdev, pin->nid, port_id,
 				&port->eld.monitor_present,
 				port->eld.eld_buffer,
 				ELD_MAX_SIZE);
 
 	if (size > 0) {
 		size = min(size, ELD_MAX_SIZE);
-		if (hdac_hdmi_parse_eld(edev, port) < 0)
+		if (hdac_hdmi_parse_eld(hdev, port) < 0)
 			size = -EINVAL;
 	}
 
@@ -1260,11 +1249,11 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin,
 		port->eld.eld_size = 0;
 	}
 
-	pcm = hdac_hdmi_get_pcm(edev, port);
+	pcm = hdac_hdmi_get_pcm(hdev, port);
 
 	if (!port->eld.monitor_present || !port->eld.eld_valid) {
 
-		dev_err(&edev->hdev.dev, "%s: disconnect for pin:port %d:%d\n",
+		dev_err(&hdev->dev, "%s: disconnect for pin:port %d:%d\n",
 						__func__, pin->nid, port->id);
 
 		/*
@@ -1316,9 +1305,9 @@ static int hdac_hdmi_add_ports(struct hdac_hdmi_priv *hdmi,
 	return 0;
 }
 
-static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid)
+static int hdac_hdmi_add_pin(struct hdac_device *hdev, hda_nid_t nid)
 {
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	struct hdac_hdmi_pin *pin;
 	int ret;
 
@@ -1328,7 +1317,7 @@ static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid)
 
 	pin->nid = nid;
 	pin->mst_capable = false;
-	pin->edev = edev;
+	pin->hdev = hdev;
 	ret = hdac_hdmi_add_ports(hdmi, pin);
 	if (ret < 0)
 		return ret;
@@ -1459,15 +1448,14 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev,
  * Parse all nodes and store the cvt/pin nids in array
  * Add one time initialization for pin and cvt widgets
  */
-static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev,
+static int hdac_hdmi_parse_and_map_nid(struct hdac_device *hdev,
 		struct snd_soc_dai_driver **dais, int *num_dais)
 {
 	hda_nid_t nid;
 	int i, num_nodes;
 	struct hdac_hdmi_cvt *temp_cvt, *cvt_next;
 	struct hdac_hdmi_pin *temp_pin, *pin_next;
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
-	struct hdac_device *hdev = &edev->hdev;
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	int ret;
 
 	hdac_hdmi_skl_enable_all_pins(hdev);
@@ -1492,13 +1480,13 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev,
 		switch (type) {
 
 		case AC_WID_AUD_OUT:
-			ret = hdac_hdmi_add_cvt(edev, nid);
+			ret = hdac_hdmi_add_cvt(hdev, nid);
 			if (ret < 0)
 				goto free_widgets;
 			break;
 
 		case AC_WID_PIN:
-			ret = hdac_hdmi_add_pin(edev, nid);
+			ret = hdac_hdmi_add_pin(hdev, nid);
 			if (ret < 0)
 				goto free_widgets;
 			break;
@@ -1518,7 +1506,7 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev,
 	}
 
 	*num_dais = hdmi->num_cvt;
-	ret = hdac_hdmi_init_dai_map(edev);
+	ret = hdac_hdmi_init_dai_map(hdev);
 	if (ret < 0)
 		goto free_widgets;
 
@@ -1542,19 +1530,24 @@ free_widgets:
 	return ret;
 }
 
+static int hdac_hdmi_pin2port(void *aptr, int pin)
+{
+	return pin - 4; /* map NID 0x05 -> port #1 */
+}
+
 static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe)
 {
-	struct hdac_ext_device *edev = aptr;
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_device *hdev = aptr;
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	struct hdac_hdmi_pin *pin = NULL;
 	struct hdac_hdmi_port *hport = NULL;
-	struct snd_soc_component *component = edev->scodec;
+	struct snd_soc_component *component = hdmi->component;
 	int i;
 
 	/* Don't know how this mapping is derived */
 	hda_nid_t pin_nid = port + 0x04;
 
-	dev_dbg(&edev->hdev.dev, "%s: for pin:%d port=%d\n", __func__,
+	dev_dbg(&hdev->dev, "%s: for pin:%d port=%d\n", __func__,
 							pin_nid, pipe);
 
 	/*
@@ -1567,7 +1560,7 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe)
 			SNDRV_CTL_POWER_D0)
 		return;
 
-	if (atomic_read(&edev->hdev.in_pm))
+	if (atomic_read(&hdev->in_pm))
 		return;
 
 	list_for_each_entry(pin, &hdmi->pin_list, head) {
@@ -1595,7 +1588,8 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe)
 
 }
 
-static struct i915_audio_component_audio_ops aops = {
+static struct drm_audio_component_audio_ops aops = {
+	.pin2port	= hdac_hdmi_pin2port,
 	.pin_eld_notify	= hdac_hdmi_eld_notify_cb,
 };
 
@@ -1614,15 +1608,15 @@ static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card,
 
 /* create jack pin kcontrols */
 static int create_fill_jack_kcontrols(struct snd_soc_card *card,
-				    struct hdac_ext_device *edev)
+				    struct hdac_device *hdev)
 {
 	struct hdac_hdmi_pin *pin;
 	struct snd_kcontrol_new *kc;
 	char kc_name[NAME_SIZE], xname[NAME_SIZE];
 	char *name;
 	int i = 0, j;
-	struct snd_soc_component *component = edev->scodec;
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
+	struct snd_soc_component *component = hdmi->component;
 
 	kc = devm_kcalloc(component->dev, hdmi->num_ports,
 				sizeof(*kc), GFP_KERNEL);
@@ -1659,8 +1653,8 @@ static int create_fill_jack_kcontrols(struct snd_soc_card *card,
 int hdac_hdmi_jack_port_init(struct snd_soc_component *component,
 			struct snd_soc_dapm_context *dapm)
 {
-	struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component);
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component);
+	struct hdac_device *hdev = hdmi->hdev;
 	struct hdac_hdmi_pin *pin;
 	struct snd_soc_dapm_widget *widgets;
 	struct snd_soc_dapm_route *route;
@@ -1715,7 +1709,7 @@ int hdac_hdmi_jack_port_init(struct snd_soc_component *component,
 		return ret;
 
 	/* Add Jack Pin switch Kcontrol */
-	ret = create_fill_jack_kcontrols(dapm->card, edev);
+	ret = create_fill_jack_kcontrols(dapm->card, hdev);
 
 	if (ret < 0)
 		return ret;
@@ -1735,8 +1729,8 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device,
 				struct snd_soc_jack *jack)
 {
 	struct snd_soc_component *component = dai->component;
-	struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component);
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component);
+	struct hdac_device *hdev = hdmi->hdev;
 	struct hdac_hdmi_pcm *pcm;
 	struct snd_pcm *snd_pcm;
 	int err;
@@ -1758,7 +1752,7 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device,
 	if (snd_pcm) {
 		err = snd_hdac_add_chmap_ctls(snd_pcm, device, &hdmi->chmap);
 		if (err < 0) {
-			dev_err(&edev->hdev.dev,
+			dev_err(&hdev->dev,
 				"chmap control add failed with err: %d for pcm: %d\n",
 				err, device);
 			kfree(pcm);
@@ -1772,7 +1766,7 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device,
 }
 EXPORT_SYMBOL_GPL(hdac_hdmi_jack_init);
 
-static void hdac_hdmi_present_sense_all_pins(struct hdac_ext_device *edev,
+static void hdac_hdmi_present_sense_all_pins(struct hdac_device *hdev,
 			struct hdac_hdmi_priv *hdmi, bool detect_pin_caps)
 {
 	int i;
@@ -1781,7 +1775,7 @@ static void hdac_hdmi_present_sense_all_pins(struct hdac_ext_device *edev,
 	list_for_each_entry(pin, &hdmi->pin_list, head) {
 		if (detect_pin_caps) {
 
-			if (hdac_hdmi_get_port_len(edev, pin->nid)  == 0)
+			if (hdac_hdmi_get_port_len(hdev, pin->nid)  == 0)
 				pin->mst_capable = false;
 			else
 				pin->mst_capable = true;
@@ -1798,68 +1792,67 @@ static void hdac_hdmi_present_sense_all_pins(struct hdac_ext_device *edev,
 
 static int hdmi_codec_probe(struct snd_soc_component *component)
 {
-	struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component);
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component);
+	struct hdac_device *hdev = hdmi->hdev;
 	struct snd_soc_dapm_context *dapm =
 		snd_soc_component_get_dapm(component);
 	struct hdac_ext_link *hlink = NULL;
 	int ret;
 
-	edev->scodec = component;
+	hdmi->component = component;
 
 	/*
 	 * hold the ref while we probe, also no need to drop the ref on
 	 * exit, we call pm_runtime_suspend() so that will do for us
 	 */
-	hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdev.dev));
+	hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
 	if (!hlink) {
-		dev_err(&edev->hdev.dev, "hdac link not found\n");
+		dev_err(&hdev->dev, "hdac link not found\n");
 		return -EIO;
 	}
 
-	snd_hdac_ext_bus_link_get(edev->ebus, hlink);
+	snd_hdac_ext_bus_link_get(hdev->bus, hlink);
 
 	ret = create_fill_widget_route_map(dapm);
 	if (ret < 0)
 		return ret;
 
-	aops.audio_ptr = edev;
-	ret = snd_hdac_i915_register_notifier(&aops);
+	aops.audio_ptr = hdev;
+	ret = snd_hdac_acomp_register_notifier(hdev->bus, &aops);
 	if (ret < 0) {
-		dev_err(&edev->hdev.dev, "notifier register failed: err: %d\n",
-				ret);
+		dev_err(&hdev->dev, "notifier register failed: err: %d\n", ret);
 		return ret;
 	}
 
-	hdac_hdmi_present_sense_all_pins(edev, hdmi, true);
+	hdac_hdmi_present_sense_all_pins(hdev, hdmi, true);
 	/* Imp: Store the card pointer in hda_codec */
-	edev->card = dapm->card->snd_card;
+	hdmi->card = dapm->card->snd_card;
 
 	/*
 	 * hdac_device core already sets the state to active and calls
 	 * get_noresume. So enable runtime and set the device to suspend.
 	 */
-	pm_runtime_enable(&edev->hdev.dev);
-	pm_runtime_put(&edev->hdev.dev);
-	pm_runtime_suspend(&edev->hdev.dev);
+	pm_runtime_enable(&hdev->dev);
+	pm_runtime_put(&hdev->dev);
+	pm_runtime_suspend(&hdev->dev);
 
 	return 0;
 }
 
 static void hdmi_codec_remove(struct snd_soc_component *component)
 {
-	struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component);
+	struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component);
+	struct hdac_device *hdev = hdmi->hdev;
 
-	pm_runtime_disable(&edev->hdev.dev);
+	pm_runtime_disable(&hdev->dev);
 }
 
 #ifdef CONFIG_PM
 static int hdmi_codec_prepare(struct device *dev)
 {
-	struct hdac_ext_device *edev = to_hda_ext_device(dev);
-	struct hdac_device *hdev = &edev->hdev;
+	struct hdac_device *hdev = dev_to_hdac_dev(dev);
 
-	pm_runtime_get_sync(&edev->hdev.dev);
+	pm_runtime_get_sync(&hdev->dev);
 
 	/*
 	 * Power down afg.
@@ -1876,16 +1869,15 @@ static int hdmi_codec_prepare(struct device *dev)
 
 static void hdmi_codec_complete(struct device *dev)
 {
-	struct hdac_ext_device *edev = to_hda_ext_device(dev);
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
-	struct hdac_device *hdev = &edev->hdev;
+	struct hdac_device *hdev = dev_to_hdac_dev(dev);
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 
 	/* Power up afg */
 	snd_hdac_codec_read(hdev, hdev->afg, 0,	AC_VERB_SET_POWER_STATE,
 							AC_PWRST_D0);
 
-	hdac_hdmi_skl_enable_all_pins(&edev->hdev);
-	hdac_hdmi_skl_enable_dp12(&edev->hdev);
+	hdac_hdmi_skl_enable_all_pins(hdev);
+	hdac_hdmi_skl_enable_dp12(hdev);
 
 	/*
 	 * As the ELD notify callback request is not entertained while the
@@ -1893,9 +1885,9 @@ static void hdmi_codec_complete(struct device *dev)
 	 * all pins here. pin capablity change is not support, so use the
 	 * already set pin caps.
 	 */
-	hdac_hdmi_present_sense_all_pins(edev, hdmi, false);
+	hdac_hdmi_present_sense_all_pins(hdev, hdmi, false);
 
-	pm_runtime_put_sync(&edev->hdev.dev);
+	pm_runtime_put_sync(&hdev->dev);
 }
 #else
 #define hdmi_codec_prepare NULL
@@ -1922,7 +1914,6 @@ static void hdac_hdmi_get_chmap(struct hdac_device *hdev, int pcm_idx,
 static void hdac_hdmi_set_chmap(struct hdac_device *hdev, int pcm_idx,
 				unsigned char *chmap, int prepared)
 {
-	struct hdac_ext_device *edev = to_ehdac_device(hdev);
 	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx);
 	struct hdac_hdmi_port *port;
@@ -1938,7 +1929,7 @@ static void hdac_hdmi_set_chmap(struct hdac_device *hdev, int pcm_idx,
 	memcpy(pcm->chmap, chmap, ARRAY_SIZE(pcm->chmap));
 	list_for_each_entry(port, &pcm->port_list, head)
 		if (prepared)
-			hdac_hdmi_setup_audio_infoframe(edev, pcm, port);
+			hdac_hdmi_setup_audio_infoframe(hdev, pcm, port);
 	mutex_unlock(&pcm->lock);
 }
 
@@ -1987,10 +1978,9 @@ static struct hdac_hdmi_drv_data intel_drv_data  = {
 	.vendor_nid = INTEL_VENDOR_NID,
 };
 
-static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
+static int hdac_hdmi_dev_probe(struct hdac_device *hdev)
 {
-	struct hdac_device *hdev = &edev->hdev;
-	struct hdac_hdmi_priv *hdmi_priv;
+	struct hdac_hdmi_priv *hdmi_priv = NULL;
 	struct snd_soc_dai_driver *hdmi_dais = NULL;
 	struct hdac_ext_link *hlink = NULL;
 	int num_dais = 0;
@@ -1999,24 +1989,24 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
 	const struct hda_device_id *hdac_id = hdac_get_device_id(hdev, hdrv);
 
 	/* hold the ref while we probe */
-	hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdev.dev));
+	hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
 	if (!hlink) {
-		dev_err(&edev->hdev.dev, "hdac link not found\n");
+		dev_err(&hdev->dev, "hdac link not found\n");
 		return -EIO;
 	}
 
-	snd_hdac_ext_bus_link_get(edev->ebus, hlink);
+	snd_hdac_ext_bus_link_get(hdev->bus, hlink);
 
 	hdmi_priv = devm_kzalloc(&hdev->dev, sizeof(*hdmi_priv), GFP_KERNEL);
 	if (hdmi_priv == NULL)
 		return -ENOMEM;
 
-	edev->private_data = hdmi_priv;
 	snd_hdac_register_chmap_ops(hdev, &hdmi_priv->chmap);
 	hdmi_priv->chmap.ops.get_chmap = hdac_hdmi_get_chmap;
 	hdmi_priv->chmap.ops.set_chmap = hdac_hdmi_set_chmap;
 	hdmi_priv->chmap.ops.is_pcm_attached = is_hdac_hdmi_pcm_attached;
 	hdmi_priv->chmap.ops.get_spk_alloc = hdac_hdmi_get_spk_alloc;
+	hdmi_priv->hdev = hdev;
 
 	if (!hdac_id)
 		return -ENODEV;
@@ -2027,7 +2017,7 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
 	else
 		hdmi_priv->drv_data = &intel_drv_data;
 
-	dev_set_drvdata(&hdev->dev, edev);
+	dev_set_drvdata(&hdev->dev, hdmi_priv);
 
 	INIT_LIST_HEAD(&hdmi_priv->pin_list);
 	INIT_LIST_HEAD(&hdmi_priv->cvt_list);
@@ -2038,15 +2028,15 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
 	 * Turned off in the runtime_suspend during the first explicit
 	 * pm_runtime_suspend call.
 	 */
-	ret = snd_hdac_display_power(edev->hdev.bus, true);
+	ret = snd_hdac_display_power(hdev->bus, true);
 	if (ret < 0) {
-		dev_err(&edev->hdev.dev,
+		dev_err(&hdev->dev,
 			"Cannot turn on display power on i915 err: %d\n",
 			ret);
 		return ret;
 	}
 
-	ret = hdac_hdmi_parse_and_map_nid(edev, &hdmi_dais, &num_dais);
+	ret = hdac_hdmi_parse_and_map_nid(hdev, &hdmi_dais, &num_dais);
 	if (ret < 0) {
 		dev_err(&hdev->dev,
 			"Failed in parse and map nid with err: %d\n", ret);
@@ -2058,14 +2048,14 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
 	ret = devm_snd_soc_register_component(&hdev->dev, &hdmi_hda_codec,
 					hdmi_dais, num_dais);
 
-	snd_hdac_ext_bus_link_put(edev->ebus, hlink);
+	snd_hdac_ext_bus_link_put(hdev->bus, hlink);
 
 	return ret;
 }
 
-static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev)
+static int hdac_hdmi_dev_remove(struct hdac_device *hdev)
 {
-	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev);
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
 	struct hdac_hdmi_pin *pin, *pin_next;
 	struct hdac_hdmi_cvt *cvt, *cvt_next;
 	struct hdac_hdmi_pcm *pcm, *pcm_next;
@@ -2103,12 +2093,79 @@ static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev)
 }
 
 #ifdef CONFIG_PM
+/*
+ * Power management sequences
+ * ==========================
+ *
+ * The following explains the PM handling of HDAC HDMI with its parent
+ * device SKL and display power usage
+ *
+ * Probe
+ * -----
+ * In SKL probe,
+ * 1. skl_probe_work() powers up the display (refcount++ -> 1)
+ * 2. enumerates the codecs on the link
+ * 3. powers down the display  (refcount-- -> 0)
+ *
+ * In HDAC HDMI probe,
+ * 1. hdac_hdmi_dev_probe() powers up the display (refcount++ -> 1)
+ * 2. probe the codec
+ * 3. put the HDAC HDMI device to runtime suspend
+ * 4. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
+ *
+ * Once children are runtime suspended, SKL device also goes to runtime
+ * suspend
+ *
+ * HDMI Playback
+ * -------------
+ * Open HDMI device,
+ * 1. skl_runtime_resume() invoked
+ * 2. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1)
+ *
+ * Close HDMI device,
+ * 1. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
+ * 2. skl_runtime_suspend() invoked
+ *
+ * S0/S3 Cycle with playback in progress
+ * -------------------------------------
+ * When the device is opened for playback, the device is runtime active
+ * already and the display refcount is 1 as explained above.
+ *
+ * Entering to S3,
+ * 1. hdmi_codec_prepare() invoke the runtime resume of codec which just
+ *    increments the PM runtime usage count of the codec since the device
+ *    is in use already
+ * 2. skl_suspend() powers down the display (refcount-- -> 0)
+ *
+ * Wakeup from S3,
+ * 1. skl_resume() powers up the display (refcount++ -> 1)
+ * 2. hdmi_codec_complete() invokes the runtime suspend of codec which just
+ *    decrements the PM runtime usage count of the codec since the device
+ *    is in use already
+ *
+ * Once playback is stopped, the display refcount is set to 0 as explained
+ * above in the HDMI playback sequence. The PM handlings are designed in
+ * such way that to balance the refcount of display power when the codec
+ * device put to S3 while playback is going on.
+ *
+ * S0/S3 Cycle without playback in progress
+ * ----------------------------------------
+ * Entering to S3,
+ * 1. hdmi_codec_prepare() invoke the runtime resume of codec
+ * 2. skl_runtime_resume() invoked
+ * 3. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1)
+ * 4. skl_suspend() powers down the display (refcount-- -> 0)
+ *
+ * Wakeup from S3,
+ * 1. skl_resume() powers up the display (refcount++ -> 1)
+ * 2. hdmi_codec_complete() invokes the runtime suspend of codec
+ * 3. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
+ * 4. skl_runtime_suspend() invoked
+ */
 static int hdac_hdmi_runtime_suspend(struct device *dev)
 {
-	struct hdac_ext_device *edev = to_hda_ext_device(dev);
-	struct hdac_device *hdev = &edev->hdev;
+	struct hdac_device *hdev = dev_to_hdac_dev(dev);
 	struct hdac_bus *bus = hdev->bus;
-	struct hdac_ext_bus *ebus = hbus_to_ebus(bus);
 	struct hdac_ext_link *hlink = NULL;
 	int err;
 
@@ -2129,27 +2186,25 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
 							AC_PWRST_D3);
 	err = snd_hdac_display_power(bus, false);
 	if (err < 0) {
-		dev_err(bus->dev, "Cannot turn on display power on i915\n");
+		dev_err(dev, "Cannot turn on display power on i915\n");
 		return err;
 	}
 
-	hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev));
+	hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev));
 	if (!hlink) {
 		dev_err(dev, "hdac link not found\n");
 		return -EIO;
 	}
 
-	snd_hdac_ext_bus_link_put(ebus, hlink);
+	snd_hdac_ext_bus_link_put(bus, hlink);
 
 	return 0;
 }
 
 static int hdac_hdmi_runtime_resume(struct device *dev)
 {
-	struct hdac_ext_device *edev = to_hda_ext_device(dev);
-	struct hdac_device *hdev = &edev->hdev;
+	struct hdac_device *hdev = dev_to_hdac_dev(dev);
 	struct hdac_bus *bus = hdev->bus;
-	struct hdac_ext_bus *ebus = hbus_to_ebus(bus);
 	struct hdac_ext_link *hlink = NULL;
 	int err;
 
@@ -2159,22 +2214,22 @@ static int hdac_hdmi_runtime_resume(struct device *dev)
 	if (!bus)
 		return 0;
 
-	hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev));
+	hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev));
 	if (!hlink) {
 		dev_err(dev, "hdac link not found\n");
 		return -EIO;
 	}
 
-	snd_hdac_ext_bus_link_get(ebus, hlink);
+	snd_hdac_ext_bus_link_get(bus, hlink);
 
 	err = snd_hdac_display_power(bus, true);
 	if (err < 0) {
-		dev_err(bus->dev, "Cannot turn on display power on i915\n");
+		dev_err(dev, "Cannot turn on display power on i915\n");
 		return err;
 	}
 
-	hdac_hdmi_skl_enable_all_pins(&edev->hdev);
-	hdac_hdmi_skl_enable_dp12(&edev->hdev);
+	hdac_hdmi_skl_enable_all_pins(hdev);
+	hdac_hdmi_skl_enable_dp12(hdev);
 
 	/* Power up afg */
 	snd_hdac_codec_read(hdev, hdev->afg, 0,	AC_VERB_SET_POWER_STATE,
@@ -2206,14 +2261,12 @@ static const struct hda_device_id hdmi_list[] = {
 
 MODULE_DEVICE_TABLE(hdaudio, hdmi_list);
 
-static struct hdac_ext_driver hdmi_driver = {
-	. hdac = {
-		.driver = {
-			.name   = "HDMI HDA Codec",
-			.pm = &hdac_hdmi_pm,
-		},
-		.id_table       = hdmi_list,
+static struct hdac_driver hdmi_driver = {
+	.driver = {
+		.name   = "HDMI HDA Codec",
+		.pm = &hdac_hdmi_pm,
 	},
+	.id_table       = hdmi_list,
 	.probe          = hdac_hdmi_dev_probe,
 	.remove         = hdac_hdmi_dev_remove,
 };
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 38e4a8515709..d00734d31e04 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -291,10 +291,6 @@ static const struct snd_soc_dapm_widget hdmi_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("TX"),
 };
 
-static const struct snd_soc_dapm_route hdmi_routes[] = {
-	{ "TX", NULL, "Playback" },
-};
-
 enum {
 	DAI_ID_I2S = 0,
 	DAI_ID_SPDIF,
@@ -689,9 +685,23 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd,
 	return snd_ctl_add(rtd->card->snd_card, kctl);
 }
 
+static int hdmi_dai_probe(struct snd_soc_dai *dai)
+{
+	struct snd_soc_dapm_context *dapm;
+	struct snd_soc_dapm_route route = {
+		.sink = "TX",
+		.source = dai->driver->playback.stream_name,
+	};
+
+	dapm = snd_soc_component_get_dapm(dai->component);
+
+	return snd_soc_dapm_add_routes(dapm, &route, 1);
+}
+
 static const struct snd_soc_dai_driver hdmi_i2s_dai = {
 	.name = "i2s-hifi",
 	.id = DAI_ID_I2S,
+	.probe = hdmi_dai_probe,
 	.playback = {
 		.stream_name = "I2S Playback",
 		.channels_min = 2,
@@ -707,6 +717,7 @@ static const struct snd_soc_dai_driver hdmi_i2s_dai = {
 static const struct snd_soc_dai_driver hdmi_spdif_dai = {
 	.name = "spdif-hifi",
 	.id = DAI_ID_SPDIF,
+	.probe = hdmi_dai_probe,
 	.playback = {
 		.stream_name = "SPDIF Playback",
 		.channels_min = 2,
@@ -733,8 +744,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component,
 static const struct snd_soc_component_driver hdmi_driver = {
 	.dapm_widgets		= hdmi_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(hdmi_widgets),
-	.dapm_routes		= hdmi_routes,
-	.num_dapm_routes	= ARRAY_SIZE(hdmi_routes),
 	.of_xlate_dai_id	= hdmi_of_xlate_dai_id,
 	.idle_bias_on		= 1,
 	.use_pmdown_time	= 1,
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index a92586106932..92b7125ea169 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -488,6 +488,7 @@ static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv,
 static bool max98373_readable_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
+	case MAX98373_R2000_SW_RESET:
 	case MAX98373_R2001_INT_RAW1 ... MAX98373_R200C_INT_EN3:
 	case MAX98373_R2010_IRQ_CTRL:
 	case MAX98373_R2014_THERM_WARN_THRESH
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index 74d7f52c7e73..6e6134589588 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -52,9 +52,9 @@ static bool max9850_volatile_register(struct device *dev, unsigned int reg)
 	switch (reg) {
 	case MAX9850_STATUSA:
 	case MAX9850_STATUSB:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c
index 17104f8dc1a9..e3c8cd17daf2 100644
--- a/sound/soc/codecs/nau8540.c
+++ b/sound/soc/codecs/nau8540.c
@@ -362,11 +362,8 @@ static const struct snd_soc_dapm_route nau8540_dapm_routes[] = {
 
 static int nau8540_clock_check(struct nau8540 *nau8540, int rate, int osr)
 {
-	int osrate;
-
 	if (osr >= ARRAY_SIZE(osr_adc_sel))
 		return -EINVAL;
-	osrate = osr_adc_sel[osr].osr;
 
 	if (rate * osr > CLK_ADC_MAX) {
 		dev_err(nau8540->dev, "exceed the maximum frequency of CLK_ADC\n");
diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c
index 6bd14453f06e..468d5143e2c4 100644
--- a/sound/soc/codecs/nau8824.c
+++ b/sound/soc/codecs/nau8824.c
@@ -1274,7 +1274,7 @@ static int nau8824_calc_fll_param(unsigned int fll_in,
 	fvco_max = 0;
 	fvco_sel = ARRAY_SIZE(mclk_src_scaling);
 	for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) {
-		fvco = 256 * fs * 2 * mclk_src_scaling[i].param;
+		fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param;
 		if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX &&
 			fvco_max < fvco) {
 			fvco_max = fvco;
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
index dc6ea4987b7d..b9fed99d8b5e 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -2016,7 +2016,7 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs,
 	fvco_max = 0;
 	fvco_sel = ARRAY_SIZE(mclk_src_scaling);
 	for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) {
-		fvco = 256 * fs * 2 * mclk_src_scaling[i].param;
+		fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param;
 		if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX &&
 			fvco_max < fvco) {
 			fvco_max = fvco;
diff --git a/sound/soc/codecs/pcm1789.c b/sound/soc/codecs/pcm1789.c
index 21f15219b3ad..8df6447c76a6 100644
--- a/sound/soc/codecs/pcm1789.c
+++ b/sound/soc/codecs/pcm1789.c
@@ -262,8 +262,7 @@ int pcm1789_common_exit(struct device *dev)
 {
 	struct pcm1789_private *priv = dev_get_drvdata(dev);
 
-	if (&priv->work)
-		flush_work(&priv->work);
+	flush_work(&priv->work);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c
index 88fde70b1e9e..690c26e7389e 100644
--- a/sound/soc/codecs/pcm186x.c
+++ b/sound/soc/codecs/pcm186x.c
@@ -265,7 +265,7 @@ static int pcm186x_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_component *component = dai->component;
 	struct pcm186x_priv *priv = snd_soc_component_get_drvdata(component);
 	unsigned int rate = params_rate(params);
-	unsigned int format = params_format(params);
+	snd_pcm_format_t format = params_format(params);
 	unsigned int width = params_width(params);
 	unsigned int channels = params_channels(params);
 	unsigned int div_lrck;
diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c
index f4c8c45f4010..c4452efc7970 100644
--- a/sound/soc/codecs/rt1305.c
+++ b/sound/soc/codecs/rt1305.c
@@ -1066,7 +1066,7 @@ static void rt1305_calibrate(struct rt1305_priv *rt1305)
 	pr_debug("Left_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl);
 	pr_info("Left channel %d.%dohm\n", (r0ohm/10), (r0ohm%10));
 
-	r0l = 562949953421312;
+	r0l = 562949953421312ULL;
 	if (rhl != 0)
 		do_div(r0l, rhl);
 	pr_debug("Left_r0 = 0x%llx\n", r0l);
@@ -1083,7 +1083,7 @@ static void rt1305_calibrate(struct rt1305_priv *rt1305)
 	pr_debug("Right_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl);
 	pr_info("Right channel %d.%dohm\n", (r0ohm/10), (r0ohm%10));
 
-	r0r = 562949953421312;
+	r0r = 562949953421312ULL;
 	if (rhl != 0)
 		do_div(r0r, rhl);
 	pr_debug("Right_r0 = 0x%llx\n", r0r);
@@ -1150,17 +1150,11 @@ static int rt1305_i2c_probe(struct i2c_client *i2c,
 	rt1305_reset(rt1305->regmap);
 	rt1305_calibrate(rt1305);
 
-	return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt1305,
+	return devm_snd_soc_register_component(&i2c->dev,
+			&soc_component_dev_rt1305,
 			rt1305_dai, ARRAY_SIZE(rt1305_dai));
 }
 
-static int rt1305_i2c_remove(struct i2c_client *i2c)
-{
-	snd_soc_unregister_component(&i2c->dev);
-
-	return 0;
-}
-
 static void rt1305_i2c_shutdown(struct i2c_client *client)
 {
 	struct rt1305_priv *rt1305 = i2c_get_clientdata(client);
@@ -1180,7 +1174,6 @@ static struct i2c_driver rt1305_i2c_driver = {
 #endif
 	},
 	.probe = rt1305_i2c_probe,
-	.remove   = rt1305_i2c_remove,
 	.shutdown = rt1305_i2c_shutdown,
 	.id_table = rt1305_i2c_id,
 };
diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c
index 18686ffb0cd5..6478d10c4f4a 100644
--- a/sound/soc/codecs/rt5514-spi.c
+++ b/sound/soc/codecs/rt5514-spi.c
@@ -268,7 +268,6 @@ static const struct snd_pcm_ops rt5514_spi_pcm_ops = {
 	.hw_params	= rt5514_spi_hw_params,
 	.hw_free	= rt5514_spi_hw_free,
 	.pointer	= rt5514_spi_pcm_pointer,
-	.mmap		= snd_pcm_lib_mmap_vmalloc,
 	.page		= snd_pcm_lib_get_vmalloc_page,
 };
 
diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 1570b91bf018..dca82dd6e3bf 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = {
 	{RT5514_ANA_CTRL_LDO10,		0x00028604},
 	{RT5514_ANA_CTRL_ADCFED,	0x00000800},
 	{RT5514_ASRC_IN_CTRL1,		0x00000003},
-	{RT5514_DOWNFILTER0_CTRL3,	0x10000362},
-	{RT5514_DOWNFILTER1_CTRL3,	0x10000362},
+	{RT5514_DOWNFILTER0_CTRL3,	0x10000352},
+	{RT5514_DOWNFILTER1_CTRL3,	0x10000352},
 };
 
 static const struct reg_default rt5514_reg[] = {
@@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = {
 	{RT5514_ASRC_IN_CTRL1,		0x00000003},
 	{RT5514_DOWNFILTER0_CTRL1,	0x00020c2f},
 	{RT5514_DOWNFILTER0_CTRL2,	0x00020c2f},
-	{RT5514_DOWNFILTER0_CTRL3,	0x10000362},
+	{RT5514_DOWNFILTER0_CTRL3,	0x10000352},
 	{RT5514_DOWNFILTER1_CTRL1,	0x00020c2f},
 	{RT5514_DOWNFILTER1_CTRL2,	0x00020c2f},
-	{RT5514_DOWNFILTER1_CTRL3,	0x10000362},
+	{RT5514_DOWNFILTER1_CTRL3,	0x10000352},
 	{RT5514_ANA_CTRL_LDO10,		0x00028604},
 	{RT5514_ANA_CTRL_LDO18_16,	0x02000345},
 	{RT5514_ANA_CTRL_ADC12,		0x0000a2a8},
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index cf6dce69eb2a..865f49ac38dd 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -105,9 +105,9 @@ static bool rt5631_volatile_register(struct device *dev, unsigned int reg)
 	case RT5631_INDEX_ADD:
 	case RT5631_INDEX_DATA:
 	case RT5631_EQ_CTRL:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
@@ -164,9 +164,9 @@ static bool rt5631_readable_register(struct device *dev, unsigned int reg)
 	case RT5631_VENDOR_ID:
 	case RT5631_VENDOR_ID1:
 	case RT5631_VENDOR_ID2:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
@@ -229,10 +229,10 @@ static SOC_ENUM_SINGLE_DECL(rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG,
 static const struct snd_kcontrol_new rt5631_snd_controls[] = {
 	/* MIC */
 	SOC_ENUM("MIC1 Mode Control",  rt5631_mic1_mode_enum),
-	SOC_SINGLE_TLV("MIC1 Boost", RT5631_MIC_CTRL_2,
+	SOC_SINGLE_TLV("MIC1 Boost Volume", RT5631_MIC_CTRL_2,
 		RT5631_MIC1_BOOST_SHIFT, 8, 0, mic_bst_tlv),
 	SOC_ENUM("MIC2 Mode Control", rt5631_mic2_mode_enum),
-	SOC_SINGLE_TLV("MIC2 Boost", RT5631_MIC_CTRL_2,
+	SOC_SINGLE_TLV("MIC2 Boost Volume", RT5631_MIC_CTRL_2,
 		RT5631_MIC2_BOOST_SHIFT, 8, 0, mic_bst_tlv),
 	/* MONO IN */
 	SOC_ENUM("MONOIN Mode Control", rt5631_monoin_mode_enum),
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 8bf8d360c25f..27770143ae8f 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -1665,6 +1665,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id)
 			break;
 		case RT5640_IF_113:
 			ret |= RT5640_U_IF1;
+			/* fall through */
 		case RT5640_IF_312:
 		case RT5640_IF_213:
 			ret |= RT5640_U_IF2;
@@ -1680,6 +1681,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id)
 			break;
 		case RT5640_IF_223:
 			ret |= RT5640_U_IF1;
+			/* fall through */
 		case RT5640_IF_123:
 		case RT5640_IF_321:
 			ret |= RT5640_U_IF2;
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 6b5669f3e85d..985852fd9723 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -331,11 +331,13 @@ static const struct snd_kcontrol_new rt5651_snd_controls[] = {
 	SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5651_DAC2_DIG_VOL,
 			RT5651_L_VOL_SFT, RT5651_R_VOL_SFT,
 			175, 0, dac_vol_tlv),
-	/* IN1/IN2 Control */
+	/* IN1/IN2/IN3 Control */
 	SOC_SINGLE_TLV("IN1 Boost", RT5651_IN1_IN2,
 		RT5651_BST_SFT1, 8, 0, bst_tlv),
 	SOC_SINGLE_TLV("IN2 Boost", RT5651_IN1_IN2,
 		RT5651_BST_SFT2, 8, 0, bst_tlv),
+	SOC_SINGLE_TLV("IN3 Boost", RT5651_IN3,
+		RT5651_BST_SFT1, 8, 0, bst_tlv),
 	/* INL/INR Volume Control */
 	SOC_DOUBLE_TLV("IN Capture Volume", RT5651_INL1_INR1_VOL,
 			RT5651_INL_VOL_SFT, RT5651_INR_VOL_SFT,
@@ -1581,6 +1583,24 @@ static void rt5651_disable_micbias1_for_ovcd(struct snd_soc_component *component
 	snd_soc_dapm_mutex_unlock(dapm);
 }
 
+static void rt5651_enable_micbias1_ovcd_irq(struct snd_soc_component *component)
+{
+	struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
+
+	snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2,
+		RT5651_IRQ_MB1_OC_MASK, RT5651_IRQ_MB1_OC_NOR);
+	rt5651->ovcd_irq_enabled = true;
+}
+
+static void rt5651_disable_micbias1_ovcd_irq(struct snd_soc_component *component)
+{
+	struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
+
+	snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2,
+		RT5651_IRQ_MB1_OC_MASK, RT5651_IRQ_MB1_OC_BP);
+	rt5651->ovcd_irq_enabled = false;
+}
+
 static void rt5651_clear_micbias1_ovcd(struct snd_soc_component *component)
 {
 	snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2,
@@ -1622,10 +1642,80 @@ static bool rt5651_jack_inserted(struct snd_soc_component *component)
 	return val == 0;
 }
 
-/* Jack detect timings */
+/* Jack detect and button-press timings */
 #define JACK_SETTLE_TIME	100 /* milli seconds */
 #define JACK_DETECT_COUNT	5
 #define JACK_DETECT_MAXCOUNT	20  /* Aprox. 2 seconds worth of tries */
+#define JACK_UNPLUG_TIME	80  /* milli seconds */
+#define BP_POLL_TIME		10  /* milli seconds */
+#define BP_POLL_MAXCOUNT	200 /* assume something is wrong after this */
+#define BP_THRESHOLD		3
+
+static void rt5651_start_button_press_work(struct snd_soc_component *component)
+{
+	struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
+
+	rt5651->poll_count = 0;
+	rt5651->press_count = 0;
+	rt5651->release_count = 0;
+	rt5651->pressed = false;
+	rt5651->press_reported = false;
+	rt5651_clear_micbias1_ovcd(component);
+	schedule_delayed_work(&rt5651->bp_work, msecs_to_jiffies(BP_POLL_TIME));
+}
+
+static void rt5651_button_press_work(struct work_struct *work)
+{
+	struct rt5651_priv *rt5651 =
+		container_of(work, struct rt5651_priv, bp_work.work);
+	struct snd_soc_component *component = rt5651->component;
+
+	/* Check the jack was not removed underneath us */
+	if (!rt5651_jack_inserted(component))
+		return;
+
+	if (rt5651_micbias1_ovcd(component)) {
+		rt5651->release_count = 0;
+		rt5651->press_count++;
+		/* Remember till after JACK_UNPLUG_TIME wait */
+		if (rt5651->press_count >= BP_THRESHOLD)
+			rt5651->pressed = true;
+		rt5651_clear_micbias1_ovcd(component);
+	} else {
+		rt5651->press_count = 0;
+		rt5651->release_count++;
+	}
+
+	/*
+	 * The pins get temporarily shorted on jack unplug, so we poll for
+	 * at least JACK_UNPLUG_TIME milli-seconds before reporting a press.
+	 */
+	rt5651->poll_count++;
+	if (rt5651->poll_count < (JACK_UNPLUG_TIME / BP_POLL_TIME)) {
+		schedule_delayed_work(&rt5651->bp_work,
+				      msecs_to_jiffies(BP_POLL_TIME));
+		return;
+	}
+
+	if (rt5651->pressed && !rt5651->press_reported) {
+		dev_dbg(component->dev, "headset button press\n");
+		snd_soc_jack_report(rt5651->hp_jack, SND_JACK_BTN_0,
+				    SND_JACK_BTN_0);
+		rt5651->press_reported = true;
+	}
+
+	if (rt5651->release_count >= BP_THRESHOLD) {
+		if (rt5651->press_reported) {
+			dev_dbg(component->dev, "headset button release\n");
+			snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0);
+		}
+		/* Re-enable OVCD IRQ to detect next press */
+		rt5651_enable_micbias1_ovcd_irq(component);
+		return; /* Stop polling */
+	}
+
+	schedule_delayed_work(&rt5651->bp_work, msecs_to_jiffies(BP_POLL_TIME));
+}
 
 static int rt5651_detect_headset(struct snd_soc_component *component)
 {
@@ -1676,15 +1766,58 @@ static void rt5651_jack_detect_work(struct work_struct *work)
 {
 	struct rt5651_priv *rt5651 =
 		container_of(work, struct rt5651_priv, jack_detect_work);
+	struct snd_soc_component *component = rt5651->component;
 	int report = 0;
 
-	if (rt5651_jack_inserted(rt5651->component)) {
-		rt5651_enable_micbias1_for_ovcd(rt5651->component);
-		report = rt5651_detect_headset(rt5651->component);
-		rt5651_disable_micbias1_for_ovcd(rt5651->component);
+	if (!rt5651_jack_inserted(component)) {
+		/* Jack removed, or spurious IRQ? */
+		if (rt5651->hp_jack->status & SND_JACK_HEADPHONE) {
+			if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) {
+				cancel_delayed_work_sync(&rt5651->bp_work);
+				rt5651_disable_micbias1_ovcd_irq(component);
+				rt5651_disable_micbias1_for_ovcd(component);
+			}
+			snd_soc_jack_report(rt5651->hp_jack, 0,
+					    SND_JACK_HEADSET | SND_JACK_BTN_0);
+			dev_dbg(component->dev, "jack unplugged\n");
+		}
+	} else if (!(rt5651->hp_jack->status & SND_JACK_HEADPHONE)) {
+		/* Jack inserted */
+		WARN_ON(rt5651->ovcd_irq_enabled);
+		rt5651_enable_micbias1_for_ovcd(component);
+		report = rt5651_detect_headset(component);
+		if (report == SND_JACK_HEADSET) {
+			/* Enable ovcd IRQ for button press detect. */
+			rt5651_enable_micbias1_ovcd_irq(component);
+		} else {
+			/* No more need for overcurrent detect. */
+			rt5651_disable_micbias1_for_ovcd(component);
+		}
+		dev_dbg(component->dev, "detect report %#02x\n", report);
+		snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET);
+	} else if (rt5651->ovcd_irq_enabled && rt5651_micbias1_ovcd(component)) {
+		dev_dbg(component->dev, "OVCD IRQ\n");
+
+		/*
+		 * The ovcd IRQ keeps firing while the button is pressed, so
+		 * we disable it and start polling the button until released.
+		 *
+		 * The disable will make the IRQ pin 0 again and since we get
+		 * IRQs on both edges (so as to detect both jack plugin and
+		 * unplug) this means we will immediately get another IRQ.
+		 * The ovcd_irq_enabled check above makes the 2ND IRQ a NOP.
+		 */
+		rt5651_disable_micbias1_ovcd_irq(component);
+		rt5651_start_button_press_work(component);
+
+		/*
+		 * If the jack-detect IRQ flag goes high (unplug) after our
+		 * above rt5651_jack_inserted() check and before we have
+		 * disabled the OVCD IRQ, the IRQ pin will stay high and as
+		 * we react to edges, we miss the unplug event -> recheck.
+		 */
+		queue_work(system_long_wq, &rt5651->jack_detect_work);
 	}
-
-	snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET);
 }
 
 static irqreturn_t rt5651_irq(int irq, void *data)
@@ -1696,14 +1829,18 @@ static irqreturn_t rt5651_irq(int irq, void *data)
 	return IRQ_HANDLED;
 }
 
-static int rt5651_set_jack(struct snd_soc_component *component,
-			   struct snd_soc_jack *hp_jack, void *data)
+static void rt5651_cancel_work(void *data)
 {
-	struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
-	int ret;
+	struct rt5651_priv *rt5651 = data;
 
-	if (!rt5651->irq)
-		return -EINVAL;
+	cancel_work_sync(&rt5651->jack_detect_work);
+	cancel_delayed_work_sync(&rt5651->bp_work);
+}
+
+static void rt5651_enable_jack_detect(struct snd_soc_component *component,
+				      struct snd_soc_jack *hp_jack)
+{
+	struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
 
 	/* IRQ output on GPIO1 */
 	snd_soc_component_update_bits(component, RT5651_GPIO_CTRL1,
@@ -1730,10 +1867,10 @@ static int rt5651_set_jack(struct snd_soc_component *component,
 			RT5651_JD2_IRQ_EN, RT5651_JD2_IRQ_EN);
 		break;
 	case RT5651_JD_NULL:
-		return 0;
+		return;
 	default:
 		dev_err(component->dev, "Currently only JD1_1 / JD1_2 / JD2 are supported\n");
-		return -EINVAL;
+		return;
 	}
 
 	/* Enable jack detect power */
@@ -1767,19 +1904,39 @@ static int rt5651_set_jack(struct snd_soc_component *component,
 		RT5651_MB1_OC_STKY_MASK, RT5651_MB1_OC_STKY_EN);
 
 	rt5651->hp_jack = hp_jack;
-
-	ret = devm_request_threaded_irq(component->dev, rt5651->irq, NULL,
-					rt5651_irq,
-					IRQF_TRIGGER_RISING |
-					IRQF_TRIGGER_FALLING |
-					IRQF_ONESHOT, "rt5651", rt5651);
-	if (ret) {
-		dev_err(component->dev, "Failed to reguest IRQ: %d\n", ret);
-		return ret;
+	if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) {
+		rt5651_enable_micbias1_for_ovcd(component);
+		rt5651_enable_micbias1_ovcd_irq(component);
 	}
 
+	enable_irq(rt5651->irq);
 	/* sync initial jack state */
 	queue_work(system_power_efficient_wq, &rt5651->jack_detect_work);
+}
+
+static void rt5651_disable_jack_detect(struct snd_soc_component *component)
+{
+	struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
+
+	disable_irq(rt5651->irq);
+	rt5651_cancel_work(rt5651);
+
+	if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) {
+		rt5651_disable_micbias1_ovcd_irq(component);
+		rt5651_disable_micbias1_for_ovcd(component);
+		snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0);
+	}
+
+	rt5651->hp_jack = NULL;
+}
+
+static int rt5651_set_jack(struct snd_soc_component *component,
+			   struct snd_soc_jack *jack, void *data)
+{
+	if (jack)
+		rt5651_enable_jack_detect(component, jack);
+	else
+		rt5651_disable_jack_detect(component);
 
 	return 0;
 }
@@ -2034,8 +2191,26 @@ static int rt5651_i2c_probe(struct i2c_client *i2c,
 	rt5651->irq = i2c->irq;
 	rt5651->hp_mute = 1;
 
+	INIT_DELAYED_WORK(&rt5651->bp_work, rt5651_button_press_work);
 	INIT_WORK(&rt5651->jack_detect_work, rt5651_jack_detect_work);
 
+	/* Make sure work is stopped on probe-error / remove */
+	ret = devm_add_action_or_reset(&i2c->dev, rt5651_cancel_work, rt5651);
+	if (ret)
+		return ret;
+
+	ret = devm_request_irq(&i2c->dev, rt5651->irq, rt5651_irq,
+			       IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
+			       | IRQF_ONESHOT, "rt5651", rt5651);
+	if (ret == 0) {
+		/* Gets re-enabled by rt5651_set_jack() */
+		disable_irq(rt5651->irq);
+	} else {
+		dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n",
+			 rt5651->irq, ret);
+		rt5651->irq = -ENXIO;
+	}
+
 	ret = devm_snd_soc_register_component(&i2c->dev,
 				&soc_component_dev_rt5651,
 				rt5651_dai, ARRAY_SIZE(rt5651_dai));
@@ -2043,15 +2218,6 @@ static int rt5651_i2c_probe(struct i2c_client *i2c,
 	return ret;
 }
 
-static int rt5651_i2c_remove(struct i2c_client *i2c)
-{
-	struct rt5651_priv *rt5651 = i2c_get_clientdata(i2c);
-
-	cancel_work_sync(&rt5651->jack_detect_work);
-
-	return 0;
-}
-
 static struct i2c_driver rt5651_i2c_driver = {
 	.driver = {
 		.name = "rt5651",
@@ -2059,7 +2225,6 @@ static struct i2c_driver rt5651_i2c_driver = {
 		.of_match_table = of_match_ptr(rt5651_of_match),
 	},
 	.probe = rt5651_i2c_probe,
-	.remove   = rt5651_i2c_remove,
 	.id_table = rt5651_i2c_id,
 };
 module_i2c_driver(rt5651_i2c_driver);
diff --git a/sound/soc/codecs/rt5651.h b/sound/soc/codecs/rt5651.h
index 3a0968c53fde..ac6de6fb5414 100644
--- a/sound/soc/codecs/rt5651.h
+++ b/sound/soc/codecs/rt5651.h
@@ -2071,8 +2071,16 @@ struct rt5651_pll_code {
 struct rt5651_priv {
 	struct snd_soc_component *component;
 	struct regmap *regmap;
+	/* Jack and button detect data */
 	struct snd_soc_jack *hp_jack;
 	struct work_struct jack_detect_work;
+	struct delayed_work bp_work;
+	bool ovcd_irq_enabled;
+	bool pressed;
+	bool press_reported;
+	int press_count;
+	int release_count;
+	int poll_count;
 	unsigned int jd_src;
 	unsigned int ovcd_th;
 	unsigned int ovcd_sf;
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 8a0181a2db08..9b7a1833d331 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -4417,6 +4417,7 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 		break;
 	case 25:
 		slot_width_25 = 0x8080;
+		/* fall through */
 	case 24:
 		val |= (2 << 8);
 		break;
@@ -5007,7 +5008,7 @@ static const struct regmap_config rt5677_regmap = {
 };
 
 static const struct of_device_id rt5677_of_match[] = {
-	{ .compatible = "realtek,rt5677", RT5677 },
+	{ .compatible = "realtek,rt5677", .data = (const void *)RT5677 },
 	{ }
 };
 MODULE_DEVICE_TABLE(of, rt5677_of_match);
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
new file mode 100644
index 000000000000..640d400ca013
--- /dev/null
+++ b/sound/soc/codecs/rt5682.c
@@ -0,0 +1,2681 @@
+/*
+ * rt5682.c  --  RT5682 ALSA SoC audio component driver
+ *
+ * Copyright 2018 Realtek Semiconductor Corp.
+ * Author: Bard Liao <bardliao@realtek.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <linux/acpi.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+#include <linux/regulator/consumer.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/jack.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/rt5682.h>
+
+#include "rl6231.h"
+#include "rt5682.h"
+
+#define RT5682_NUM_SUPPLIES 3
+
+static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = {
+	"AVDD",
+	"MICVDD",
+	"VBAT",
+};
+
+struct rt5682_priv {
+	struct snd_soc_component *component;
+	struct rt5682_platform_data pdata;
+	struct regmap *regmap;
+	struct snd_soc_jack *hs_jack;
+	struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES];
+	struct delayed_work jack_detect_work;
+	struct delayed_work jd_check_work;
+	struct mutex calibrate_mutex;
+
+	int sysclk;
+	int sysclk_src;
+	int lrck[RT5682_AIFS];
+	int bclk[RT5682_AIFS];
+	int master[RT5682_AIFS];
+
+	int pll_src;
+	int pll_in;
+	int pll_out;
+
+	int jack_type;
+};
+
+static const struct reg_sequence patch_list[] = {
+	{0x01c1, 0x1000},
+};
+
+static const struct reg_default rt5682_reg[] = {
+	{0x0002, 0x8080},
+	{0x0003, 0x8000},
+	{0x0005, 0x0000},
+	{0x0006, 0x0000},
+	{0x0008, 0x800f},
+	{0x000b, 0x0000},
+	{0x0010, 0x4040},
+	{0x0011, 0x0000},
+	{0x0012, 0x1404},
+	{0x0013, 0x1000},
+	{0x0014, 0xa00a},
+	{0x0015, 0x0404},
+	{0x0016, 0x0404},
+	{0x0019, 0xafaf},
+	{0x001c, 0x2f2f},
+	{0x001f, 0x0000},
+	{0x0022, 0x5757},
+	{0x0023, 0x0039},
+	{0x0024, 0x000b},
+	{0x0026, 0xc0c4},
+	{0x0029, 0x8080},
+	{0x002a, 0xa0a0},
+	{0x002b, 0x0300},
+	{0x0030, 0x0000},
+	{0x003c, 0x0080},
+	{0x0044, 0x0c0c},
+	{0x0049, 0x0000},
+	{0x0061, 0x0000},
+	{0x0062, 0x0000},
+	{0x0063, 0x003f},
+	{0x0064, 0x0000},
+	{0x0065, 0x0000},
+	{0x0066, 0x0030},
+	{0x0067, 0x0000},
+	{0x006b, 0x0000},
+	{0x006c, 0x0000},
+	{0x006d, 0x2200},
+	{0x006e, 0x0a10},
+	{0x0070, 0x8000},
+	{0x0071, 0x8000},
+	{0x0073, 0x0000},
+	{0x0074, 0x0000},
+	{0x0075, 0x0002},
+	{0x0076, 0x0001},
+	{0x0079, 0x0000},
+	{0x007a, 0x0000},
+	{0x007b, 0x0000},
+	{0x007c, 0x0100},
+	{0x007e, 0x0000},
+	{0x0080, 0x0000},
+	{0x0081, 0x0000},
+	{0x0082, 0x0000},
+	{0x0083, 0x0000},
+	{0x0084, 0x0000},
+	{0x0085, 0x0000},
+	{0x0086, 0x0005},
+	{0x0087, 0x0000},
+	{0x0088, 0x0000},
+	{0x008c, 0x0003},
+	{0x008d, 0x0000},
+	{0x008e, 0x0060},
+	{0x008f, 0x1000},
+	{0x0091, 0x0c26},
+	{0x0092, 0x0073},
+	{0x0093, 0x0000},
+	{0x0094, 0x0080},
+	{0x0098, 0x0000},
+	{0x009a, 0x0000},
+	{0x009b, 0x0000},
+	{0x009c, 0x0000},
+	{0x009d, 0x0000},
+	{0x009e, 0x100c},
+	{0x009f, 0x0000},
+	{0x00a0, 0x0000},
+	{0x00a3, 0x0002},
+	{0x00a4, 0x0001},
+	{0x00ae, 0x2040},
+	{0x00af, 0x0000},
+	{0x00b6, 0x0000},
+	{0x00b7, 0x0000},
+	{0x00b8, 0x0000},
+	{0x00b9, 0x0002},
+	{0x00be, 0x0000},
+	{0x00c0, 0x0160},
+	{0x00c1, 0x82a0},
+	{0x00c2, 0x0000},
+	{0x00d0, 0x0000},
+	{0x00d1, 0x2244},
+	{0x00d2, 0x3300},
+	{0x00d3, 0x2200},
+	{0x00d4, 0x0000},
+	{0x00d9, 0x0009},
+	{0x00da, 0x0000},
+	{0x00db, 0x0000},
+	{0x00dc, 0x00c0},
+	{0x00dd, 0x2220},
+	{0x00de, 0x3131},
+	{0x00df, 0x3131},
+	{0x00e0, 0x3131},
+	{0x00e2, 0x0000},
+	{0x00e3, 0x4000},
+	{0x00e4, 0x0aa0},
+	{0x00e5, 0x3131},
+	{0x00e6, 0x3131},
+	{0x00e7, 0x3131},
+	{0x00e8, 0x3131},
+	{0x00ea, 0xb320},
+	{0x00eb, 0x0000},
+	{0x00f0, 0x0000},
+	{0x00f1, 0x00d0},
+	{0x00f2, 0x00d0},
+	{0x00f6, 0x0000},
+	{0x00fa, 0x0000},
+	{0x00fb, 0x0000},
+	{0x00fc, 0x0000},
+	{0x00fd, 0x0000},
+	{0x00fe, 0x10ec},
+	{0x00ff, 0x6530},
+	{0x0100, 0xa0a0},
+	{0x010b, 0x0000},
+	{0x010c, 0xae00},
+	{0x010d, 0xaaa0},
+	{0x010e, 0x8aa2},
+	{0x010f, 0x02a2},
+	{0x0110, 0xc000},
+	{0x0111, 0x04a2},
+	{0x0112, 0x2800},
+	{0x0113, 0x0000},
+	{0x0117, 0x0100},
+	{0x0125, 0x0410},
+	{0x0132, 0x6026},
+	{0x0136, 0x5555},
+	{0x0138, 0x3700},
+	{0x013a, 0x2000},
+	{0x013b, 0x2000},
+	{0x013c, 0x2005},
+	{0x013f, 0x0000},
+	{0x0142, 0x0000},
+	{0x0145, 0x0002},
+	{0x0146, 0x0000},
+	{0x0147, 0x0000},
+	{0x0148, 0x0000},
+	{0x0149, 0x0000},
+	{0x0150, 0x79a1},
+	{0x0151, 0x0000},
+	{0x0160, 0x4ec0},
+	{0x0161, 0x0080},
+	{0x0162, 0x0200},
+	{0x0163, 0x0800},
+	{0x0164, 0x0000},
+	{0x0165, 0x0000},
+	{0x0166, 0x0000},
+	{0x0167, 0x000f},
+	{0x0168, 0x000f},
+	{0x0169, 0x0021},
+	{0x0190, 0x413d},
+	{0x0194, 0x0000},
+	{0x0195, 0x0000},
+	{0x0197, 0x0022},
+	{0x0198, 0x0000},
+	{0x0199, 0x0000},
+	{0x01af, 0x0000},
+	{0x01b0, 0x0400},
+	{0x01b1, 0x0000},
+	{0x01b2, 0x0000},
+	{0x01b3, 0x0000},
+	{0x01b4, 0x0000},
+	{0x01b5, 0x0000},
+	{0x01b6, 0x01c3},
+	{0x01b7, 0x02a0},
+	{0x01b8, 0x03e9},
+	{0x01b9, 0x1389},
+	{0x01ba, 0xc351},
+	{0x01bb, 0x0009},
+	{0x01bc, 0x0018},
+	{0x01bd, 0x002a},
+	{0x01be, 0x004c},
+	{0x01bf, 0x0097},
+	{0x01c0, 0x433d},
+	{0x01c2, 0x0000},
+	{0x01c3, 0x0000},
+	{0x01c4, 0x0000},
+	{0x01c5, 0x0000},
+	{0x01c6, 0x0000},
+	{0x01c7, 0x0000},
+	{0x01c8, 0x40af},
+	{0x01c9, 0x0702},
+	{0x01ca, 0x0000},
+	{0x01cb, 0x0000},
+	{0x01cc, 0x5757},
+	{0x01cd, 0x5757},
+	{0x01ce, 0x5757},
+	{0x01cf, 0x5757},
+	{0x01d0, 0x5757},
+	{0x01d1, 0x5757},
+	{0x01d2, 0x5757},
+	{0x01d3, 0x5757},
+	{0x01d4, 0x5757},
+	{0x01d5, 0x5757},
+	{0x01d6, 0x0000},
+	{0x01d7, 0x0008},
+	{0x01d8, 0x0029},
+	{0x01d9, 0x3333},
+	{0x01da, 0x0000},
+	{0x01db, 0x0004},
+	{0x01dc, 0x0000},
+	{0x01de, 0x7c00},
+	{0x01df, 0x0320},
+	{0x01e0, 0x06a1},
+	{0x01e1, 0x0000},
+	{0x01e2, 0x0000},
+	{0x01e3, 0x0000},
+	{0x01e4, 0x0000},
+	{0x01e6, 0x0001},
+	{0x01e7, 0x0000},
+	{0x01e8, 0x0000},
+	{0x01ea, 0x0000},
+	{0x01eb, 0x0000},
+	{0x01ec, 0x0000},
+	{0x01ed, 0x0000},
+	{0x01ee, 0x0000},
+	{0x01ef, 0x0000},
+	{0x01f0, 0x0000},
+	{0x01f1, 0x0000},
+	{0x01f2, 0x0000},
+	{0x01f3, 0x0000},
+	{0x01f4, 0x0000},
+	{0x0210, 0x6297},
+	{0x0211, 0xa005},
+	{0x0212, 0x824c},
+	{0x0213, 0xf7ff},
+	{0x0214, 0xf24c},
+	{0x0215, 0x0102},
+	{0x0216, 0x00a3},
+	{0x0217, 0x0048},
+	{0x0218, 0xa2c0},
+	{0x0219, 0x0400},
+	{0x021a, 0x00c8},
+	{0x021b, 0x00c0},
+	{0x021c, 0x0000},
+	{0x0250, 0x4500},
+	{0x0251, 0x40b3},
+	{0x0252, 0x0000},
+	{0x0253, 0x0000},
+	{0x0254, 0x0000},
+	{0x0255, 0x0000},
+	{0x0256, 0x0000},
+	{0x0257, 0x0000},
+	{0x0258, 0x0000},
+	{0x0259, 0x0000},
+	{0x025a, 0x0005},
+	{0x0270, 0x0000},
+	{0x02ff, 0x0110},
+	{0x0300, 0x001f},
+	{0x0301, 0x032c},
+	{0x0302, 0x5f21},
+	{0x0303, 0x4000},
+	{0x0304, 0x4000},
+	{0x0305, 0x06d5},
+	{0x0306, 0x8000},
+	{0x0307, 0x0700},
+	{0x0310, 0x4560},
+	{0x0311, 0xa4a8},
+	{0x0312, 0x7418},
+	{0x0313, 0x0000},
+	{0x0314, 0x0006},
+	{0x0315, 0xffff},
+	{0x0316, 0xc400},
+	{0x0317, 0x0000},
+	{0x03c0, 0x7e00},
+	{0x03c1, 0x8000},
+	{0x03c2, 0x8000},
+	{0x03c3, 0x8000},
+	{0x03c4, 0x8000},
+	{0x03c5, 0x8000},
+	{0x03c6, 0x8000},
+	{0x03c7, 0x8000},
+	{0x03c8, 0x8000},
+	{0x03c9, 0x8000},
+	{0x03ca, 0x8000},
+	{0x03cb, 0x8000},
+	{0x03cc, 0x8000},
+	{0x03d0, 0x0000},
+	{0x03d1, 0x0000},
+	{0x03d2, 0x0000},
+	{0x03d3, 0x0000},
+	{0x03d4, 0x2000},
+	{0x03d5, 0x2000},
+	{0x03d6, 0x0000},
+	{0x03d7, 0x0000},
+	{0x03d8, 0x2000},
+	{0x03d9, 0x2000},
+	{0x03da, 0x2000},
+	{0x03db, 0x2000},
+	{0x03dc, 0x0000},
+	{0x03dd, 0x0000},
+	{0x03de, 0x0000},
+	{0x03df, 0x2000},
+	{0x03e0, 0x0000},
+	{0x03e1, 0x0000},
+	{0x03e2, 0x0000},
+	{0x03e3, 0x0000},
+	{0x03e4, 0x0000},
+	{0x03e5, 0x0000},
+	{0x03e6, 0x0000},
+	{0x03e7, 0x0000},
+	{0x03e8, 0x0000},
+	{0x03e9, 0x0000},
+	{0x03ea, 0x0000},
+	{0x03eb, 0x0000},
+	{0x03ec, 0x0000},
+	{0x03ed, 0x0000},
+	{0x03ee, 0x0000},
+	{0x03ef, 0x0000},
+	{0x03f0, 0x0800},
+	{0x03f1, 0x0800},
+	{0x03f2, 0x0800},
+	{0x03f3, 0x0800},
+};
+
+static bool rt5682_volatile_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case RT5682_RESET:
+	case RT5682_CBJ_CTRL_2:
+	case RT5682_INT_ST_1:
+	case RT5682_4BTN_IL_CMD_1:
+	case RT5682_AJD1_CTRL:
+	case RT5682_HP_CALIB_CTRL_1:
+	case RT5682_DEVICE_ID:
+	case RT5682_I2C_MODE:
+	case RT5682_HP_CALIB_CTRL_10:
+	case RT5682_EFUSE_CTRL_2:
+	case RT5682_JD_TOP_VC_VTRL:
+	case RT5682_HP_IMP_SENS_CTRL_19:
+	case RT5682_IL_CMD_1:
+	case RT5682_SAR_IL_CMD_2:
+	case RT5682_SAR_IL_CMD_4:
+	case RT5682_SAR_IL_CMD_10:
+	case RT5682_SAR_IL_CMD_11:
+	case RT5682_EFUSE_CTRL_6...RT5682_EFUSE_CTRL_11:
+	case RT5682_HP_CALIB_STA_1...RT5682_HP_CALIB_STA_11:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool rt5682_readable_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case RT5682_RESET:
+	case RT5682_VERSION_ID:
+	case RT5682_VENDOR_ID:
+	case RT5682_DEVICE_ID:
+	case RT5682_HP_CTRL_1:
+	case RT5682_HP_CTRL_2:
+	case RT5682_HPL_GAIN:
+	case RT5682_HPR_GAIN:
+	case RT5682_I2C_CTRL:
+	case RT5682_CBJ_BST_CTRL:
+	case RT5682_CBJ_CTRL_1:
+	case RT5682_CBJ_CTRL_2:
+	case RT5682_CBJ_CTRL_3:
+	case RT5682_CBJ_CTRL_4:
+	case RT5682_CBJ_CTRL_5:
+	case RT5682_CBJ_CTRL_6:
+	case RT5682_CBJ_CTRL_7:
+	case RT5682_DAC1_DIG_VOL:
+	case RT5682_STO1_ADC_DIG_VOL:
+	case RT5682_STO1_ADC_BOOST:
+	case RT5682_HP_IMP_GAIN_1:
+	case RT5682_HP_IMP_GAIN_2:
+	case RT5682_SIDETONE_CTRL:
+	case RT5682_STO1_ADC_MIXER:
+	case RT5682_AD_DA_MIXER:
+	case RT5682_STO1_DAC_MIXER:
+	case RT5682_A_DAC1_MUX:
+	case RT5682_DIG_INF2_DATA:
+	case RT5682_REC_MIXER:
+	case RT5682_CAL_REC:
+	case RT5682_ALC_BACK_GAIN:
+	case RT5682_PWR_DIG_1:
+	case RT5682_PWR_DIG_2:
+	case RT5682_PWR_ANLG_1:
+	case RT5682_PWR_ANLG_2:
+	case RT5682_PWR_ANLG_3:
+	case RT5682_PWR_MIXER:
+	case RT5682_PWR_VOL:
+	case RT5682_CLK_DET:
+	case RT5682_RESET_LPF_CTRL:
+	case RT5682_RESET_HPF_CTRL:
+	case RT5682_DMIC_CTRL_1:
+	case RT5682_I2S1_SDP:
+	case RT5682_I2S2_SDP:
+	case RT5682_ADDA_CLK_1:
+	case RT5682_ADDA_CLK_2:
+	case RT5682_I2S1_F_DIV_CTRL_1:
+	case RT5682_I2S1_F_DIV_CTRL_2:
+	case RT5682_TDM_CTRL:
+	case RT5682_TDM_ADDA_CTRL_1:
+	case RT5682_TDM_ADDA_CTRL_2:
+	case RT5682_DATA_SEL_CTRL_1:
+	case RT5682_TDM_TCON_CTRL:
+	case RT5682_GLB_CLK:
+	case RT5682_PLL_CTRL_1:
+	case RT5682_PLL_CTRL_2:
+	case RT5682_PLL_TRACK_1:
+	case RT5682_PLL_TRACK_2:
+	case RT5682_PLL_TRACK_3:
+	case RT5682_PLL_TRACK_4:
+	case RT5682_PLL_TRACK_5:
+	case RT5682_PLL_TRACK_6:
+	case RT5682_PLL_TRACK_11:
+	case RT5682_SDW_REF_CLK:
+	case RT5682_DEPOP_1:
+	case RT5682_DEPOP_2:
+	case RT5682_HP_CHARGE_PUMP_1:
+	case RT5682_HP_CHARGE_PUMP_2:
+	case RT5682_MICBIAS_1:
+	case RT5682_MICBIAS_2:
+	case RT5682_PLL_TRACK_12:
+	case RT5682_PLL_TRACK_14:
+	case RT5682_PLL2_CTRL_1:
+	case RT5682_PLL2_CTRL_2:
+	case RT5682_PLL2_CTRL_3:
+	case RT5682_PLL2_CTRL_4:
+	case RT5682_RC_CLK_CTRL:
+	case RT5682_I2S_M_CLK_CTRL_1:
+	case RT5682_I2S2_F_DIV_CTRL_1:
+	case RT5682_I2S2_F_DIV_CTRL_2:
+	case RT5682_EQ_CTRL_1:
+	case RT5682_EQ_CTRL_2:
+	case RT5682_IRQ_CTRL_1:
+	case RT5682_IRQ_CTRL_2:
+	case RT5682_IRQ_CTRL_3:
+	case RT5682_IRQ_CTRL_4:
+	case RT5682_INT_ST_1:
+	case RT5682_GPIO_CTRL_1:
+	case RT5682_GPIO_CTRL_2:
+	case RT5682_GPIO_CTRL_3:
+	case RT5682_HP_AMP_DET_CTRL_1:
+	case RT5682_HP_AMP_DET_CTRL_2:
+	case RT5682_MID_HP_AMP_DET:
+	case RT5682_LOW_HP_AMP_DET:
+	case RT5682_DELAY_BUF_CTRL:
+	case RT5682_SV_ZCD_1:
+	case RT5682_SV_ZCD_2:
+	case RT5682_IL_CMD_1:
+	case RT5682_IL_CMD_2:
+	case RT5682_IL_CMD_3:
+	case RT5682_IL_CMD_4:
+	case RT5682_IL_CMD_5:
+	case RT5682_IL_CMD_6:
+	case RT5682_4BTN_IL_CMD_1:
+	case RT5682_4BTN_IL_CMD_2:
+	case RT5682_4BTN_IL_CMD_3:
+	case RT5682_4BTN_IL_CMD_4:
+	case RT5682_4BTN_IL_CMD_5:
+	case RT5682_4BTN_IL_CMD_6:
+	case RT5682_4BTN_IL_CMD_7:
+	case RT5682_ADC_STO1_HP_CTRL_1:
+	case RT5682_ADC_STO1_HP_CTRL_2:
+	case RT5682_AJD1_CTRL:
+	case RT5682_JD1_THD:
+	case RT5682_JD2_THD:
+	case RT5682_JD_CTRL_1:
+	case RT5682_DUMMY_1:
+	case RT5682_DUMMY_2:
+	case RT5682_DUMMY_3:
+	case RT5682_DAC_ADC_DIG_VOL1:
+	case RT5682_BIAS_CUR_CTRL_2:
+	case RT5682_BIAS_CUR_CTRL_3:
+	case RT5682_BIAS_CUR_CTRL_4:
+	case RT5682_BIAS_CUR_CTRL_5:
+	case RT5682_BIAS_CUR_CTRL_6:
+	case RT5682_BIAS_CUR_CTRL_7:
+	case RT5682_BIAS_CUR_CTRL_8:
+	case RT5682_BIAS_CUR_CTRL_9:
+	case RT5682_BIAS_CUR_CTRL_10:
+	case RT5682_VREF_REC_OP_FB_CAP_CTRL:
+	case RT5682_CHARGE_PUMP_1:
+	case RT5682_DIG_IN_CTRL_1:
+	case RT5682_PAD_DRIVING_CTRL:
+	case RT5682_SOFT_RAMP_DEPOP:
+	case RT5682_CHOP_DAC:
+	case RT5682_CHOP_ADC:
+	case RT5682_CALIB_ADC_CTRL:
+	case RT5682_VOL_TEST:
+	case RT5682_SPKVDD_DET_STA:
+	case RT5682_TEST_MODE_CTRL_1:
+	case RT5682_TEST_MODE_CTRL_2:
+	case RT5682_TEST_MODE_CTRL_3:
+	case RT5682_TEST_MODE_CTRL_4:
+	case RT5682_TEST_MODE_CTRL_5:
+	case RT5682_PLL1_INTERNAL:
+	case RT5682_PLL2_INTERNAL:
+	case RT5682_STO_NG2_CTRL_1:
+	case RT5682_STO_NG2_CTRL_2:
+	case RT5682_STO_NG2_CTRL_3:
+	case RT5682_STO_NG2_CTRL_4:
+	case RT5682_STO_NG2_CTRL_5:
+	case RT5682_STO_NG2_CTRL_6:
+	case RT5682_STO_NG2_CTRL_7:
+	case RT5682_STO_NG2_CTRL_8:
+	case RT5682_STO_NG2_CTRL_9:
+	case RT5682_STO_NG2_CTRL_10:
+	case RT5682_STO1_DAC_SIL_DET:
+	case RT5682_SIL_PSV_CTRL1:
+	case RT5682_SIL_PSV_CTRL2:
+	case RT5682_SIL_PSV_CTRL3:
+	case RT5682_SIL_PSV_CTRL4:
+	case RT5682_SIL_PSV_CTRL5:
+	case RT5682_HP_IMP_SENS_CTRL_01:
+	case RT5682_HP_IMP_SENS_CTRL_02:
+	case RT5682_HP_IMP_SENS_CTRL_03:
+	case RT5682_HP_IMP_SENS_CTRL_04:
+	case RT5682_HP_IMP_SENS_CTRL_05:
+	case RT5682_HP_IMP_SENS_CTRL_06:
+	case RT5682_HP_IMP_SENS_CTRL_07:
+	case RT5682_HP_IMP_SENS_CTRL_08:
+	case RT5682_HP_IMP_SENS_CTRL_09:
+	case RT5682_HP_IMP_SENS_CTRL_10:
+	case RT5682_HP_IMP_SENS_CTRL_11:
+	case RT5682_HP_IMP_SENS_CTRL_12:
+	case RT5682_HP_IMP_SENS_CTRL_13:
+	case RT5682_HP_IMP_SENS_CTRL_14:
+	case RT5682_HP_IMP_SENS_CTRL_15:
+	case RT5682_HP_IMP_SENS_CTRL_16:
+	case RT5682_HP_IMP_SENS_CTRL_17:
+	case RT5682_HP_IMP_SENS_CTRL_18:
+	case RT5682_HP_IMP_SENS_CTRL_19:
+	case RT5682_HP_IMP_SENS_CTRL_20:
+	case RT5682_HP_IMP_SENS_CTRL_21:
+	case RT5682_HP_IMP_SENS_CTRL_22:
+	case RT5682_HP_IMP_SENS_CTRL_23:
+	case RT5682_HP_IMP_SENS_CTRL_24:
+	case RT5682_HP_IMP_SENS_CTRL_25:
+	case RT5682_HP_IMP_SENS_CTRL_26:
+	case RT5682_HP_IMP_SENS_CTRL_27:
+	case RT5682_HP_IMP_SENS_CTRL_28:
+	case RT5682_HP_IMP_SENS_CTRL_29:
+	case RT5682_HP_IMP_SENS_CTRL_30:
+	case RT5682_HP_IMP_SENS_CTRL_31:
+	case RT5682_HP_IMP_SENS_CTRL_32:
+	case RT5682_HP_IMP_SENS_CTRL_33:
+	case RT5682_HP_IMP_SENS_CTRL_34:
+	case RT5682_HP_IMP_SENS_CTRL_35:
+	case RT5682_HP_IMP_SENS_CTRL_36:
+	case RT5682_HP_IMP_SENS_CTRL_37:
+	case RT5682_HP_IMP_SENS_CTRL_38:
+	case RT5682_HP_IMP_SENS_CTRL_39:
+	case RT5682_HP_IMP_SENS_CTRL_40:
+	case RT5682_HP_IMP_SENS_CTRL_41:
+	case RT5682_HP_IMP_SENS_CTRL_42:
+	case RT5682_HP_IMP_SENS_CTRL_43:
+	case RT5682_HP_LOGIC_CTRL_1:
+	case RT5682_HP_LOGIC_CTRL_2:
+	case RT5682_HP_LOGIC_CTRL_3:
+	case RT5682_HP_CALIB_CTRL_1:
+	case RT5682_HP_CALIB_CTRL_2:
+	case RT5682_HP_CALIB_CTRL_3:
+	case RT5682_HP_CALIB_CTRL_4:
+	case RT5682_HP_CALIB_CTRL_5:
+	case RT5682_HP_CALIB_CTRL_6:
+	case RT5682_HP_CALIB_CTRL_7:
+	case RT5682_HP_CALIB_CTRL_9:
+	case RT5682_HP_CALIB_CTRL_10:
+	case RT5682_HP_CALIB_CTRL_11:
+	case RT5682_HP_CALIB_STA_1:
+	case RT5682_HP_CALIB_STA_2:
+	case RT5682_HP_CALIB_STA_3:
+	case RT5682_HP_CALIB_STA_4:
+	case RT5682_HP_CALIB_STA_5:
+	case RT5682_HP_CALIB_STA_6:
+	case RT5682_HP_CALIB_STA_7:
+	case RT5682_HP_CALIB_STA_8:
+	case RT5682_HP_CALIB_STA_9:
+	case RT5682_HP_CALIB_STA_10:
+	case RT5682_HP_CALIB_STA_11:
+	case RT5682_SAR_IL_CMD_1:
+	case RT5682_SAR_IL_CMD_2:
+	case RT5682_SAR_IL_CMD_3:
+	case RT5682_SAR_IL_CMD_4:
+	case RT5682_SAR_IL_CMD_5:
+	case RT5682_SAR_IL_CMD_6:
+	case RT5682_SAR_IL_CMD_7:
+	case RT5682_SAR_IL_CMD_8:
+	case RT5682_SAR_IL_CMD_9:
+	case RT5682_SAR_IL_CMD_10:
+	case RT5682_SAR_IL_CMD_11:
+	case RT5682_SAR_IL_CMD_12:
+	case RT5682_SAR_IL_CMD_13:
+	case RT5682_EFUSE_CTRL_1:
+	case RT5682_EFUSE_CTRL_2:
+	case RT5682_EFUSE_CTRL_3:
+	case RT5682_EFUSE_CTRL_4:
+	case RT5682_EFUSE_CTRL_5:
+	case RT5682_EFUSE_CTRL_6:
+	case RT5682_EFUSE_CTRL_7:
+	case RT5682_EFUSE_CTRL_8:
+	case RT5682_EFUSE_CTRL_9:
+	case RT5682_EFUSE_CTRL_10:
+	case RT5682_EFUSE_CTRL_11:
+	case RT5682_JD_TOP_VC_VTRL:
+	case RT5682_DRC1_CTRL_0:
+	case RT5682_DRC1_CTRL_1:
+	case RT5682_DRC1_CTRL_2:
+	case RT5682_DRC1_CTRL_3:
+	case RT5682_DRC1_CTRL_4:
+	case RT5682_DRC1_CTRL_5:
+	case RT5682_DRC1_CTRL_6:
+	case RT5682_DRC1_HARD_LMT_CTRL_1:
+	case RT5682_DRC1_HARD_LMT_CTRL_2:
+	case RT5682_DRC1_PRIV_1:
+	case RT5682_DRC1_PRIV_2:
+	case RT5682_DRC1_PRIV_3:
+	case RT5682_DRC1_PRIV_4:
+	case RT5682_DRC1_PRIV_5:
+	case RT5682_DRC1_PRIV_6:
+	case RT5682_DRC1_PRIV_7:
+	case RT5682_DRC1_PRIV_8:
+	case RT5682_EQ_AUTO_RCV_CTRL1:
+	case RT5682_EQ_AUTO_RCV_CTRL2:
+	case RT5682_EQ_AUTO_RCV_CTRL3:
+	case RT5682_EQ_AUTO_RCV_CTRL4:
+	case RT5682_EQ_AUTO_RCV_CTRL5:
+	case RT5682_EQ_AUTO_RCV_CTRL6:
+	case RT5682_EQ_AUTO_RCV_CTRL7:
+	case RT5682_EQ_AUTO_RCV_CTRL8:
+	case RT5682_EQ_AUTO_RCV_CTRL9:
+	case RT5682_EQ_AUTO_RCV_CTRL10:
+	case RT5682_EQ_AUTO_RCV_CTRL11:
+	case RT5682_EQ_AUTO_RCV_CTRL12:
+	case RT5682_EQ_AUTO_RCV_CTRL13:
+	case RT5682_ADC_L_EQ_LPF1_A1:
+	case RT5682_R_EQ_LPF1_A1:
+	case RT5682_L_EQ_LPF1_H0:
+	case RT5682_R_EQ_LPF1_H0:
+	case RT5682_L_EQ_BPF1_A1:
+	case RT5682_R_EQ_BPF1_A1:
+	case RT5682_L_EQ_BPF1_A2:
+	case RT5682_R_EQ_BPF1_A2:
+	case RT5682_L_EQ_BPF1_H0:
+	case RT5682_R_EQ_BPF1_H0:
+	case RT5682_L_EQ_BPF2_A1:
+	case RT5682_R_EQ_BPF2_A1:
+	case RT5682_L_EQ_BPF2_A2:
+	case RT5682_R_EQ_BPF2_A2:
+	case RT5682_L_EQ_BPF2_H0:
+	case RT5682_R_EQ_BPF2_H0:
+	case RT5682_L_EQ_BPF3_A1:
+	case RT5682_R_EQ_BPF3_A1:
+	case RT5682_L_EQ_BPF3_A2:
+	case RT5682_R_EQ_BPF3_A2:
+	case RT5682_L_EQ_BPF3_H0:
+	case RT5682_R_EQ_BPF3_H0:
+	case RT5682_L_EQ_BPF4_A1:
+	case RT5682_R_EQ_BPF4_A1:
+	case RT5682_L_EQ_BPF4_A2:
+	case RT5682_R_EQ_BPF4_A2:
+	case RT5682_L_EQ_BPF4_H0:
+	case RT5682_R_EQ_BPF4_H0:
+	case RT5682_L_EQ_HPF1_A1:
+	case RT5682_R_EQ_HPF1_A1:
+	case RT5682_L_EQ_HPF1_H0:
+	case RT5682_R_EQ_HPF1_H0:
+	case RT5682_L_EQ_PRE_VOL:
+	case RT5682_R_EQ_PRE_VOL:
+	case RT5682_L_EQ_POST_VOL:
+	case RT5682_R_EQ_POST_VOL:
+	case RT5682_I2C_MODE:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0);
+static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
+
+/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
+static const DECLARE_TLV_DB_RANGE(bst_tlv,
+	0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+	1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
+	2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0),
+	3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0),
+	6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0),
+	7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0),
+	8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0)
+);
+
+/* Interface data select */
+static const char * const rt5682_data_select[] = {
+	"L/R", "R/L", "L/L", "R/R"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if2_adc_enum,
+	RT5682_DIG_INF2_DATA, RT5682_IF2_ADC_SEL_SFT, rt5682_data_select);
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if1_01_adc_enum,
+	RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC1_SEL_SFT, rt5682_data_select);
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if1_23_adc_enum,
+	RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC2_SEL_SFT, rt5682_data_select);
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if1_45_adc_enum,
+	RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC3_SEL_SFT, rt5682_data_select);
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if1_67_adc_enum,
+	RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC4_SEL_SFT, rt5682_data_select);
+
+static const struct snd_kcontrol_new rt5682_if2_adc_swap_mux =
+	SOC_DAPM_ENUM("IF2 ADC Swap Mux", rt5682_if2_adc_enum);
+
+static const struct snd_kcontrol_new rt5682_if1_01_adc_swap_mux =
+	SOC_DAPM_ENUM("IF1 01 ADC Swap Mux", rt5682_if1_01_adc_enum);
+
+static const struct snd_kcontrol_new rt5682_if1_23_adc_swap_mux =
+	SOC_DAPM_ENUM("IF1 23 ADC Swap Mux", rt5682_if1_23_adc_enum);
+
+static const struct snd_kcontrol_new rt5682_if1_45_adc_swap_mux =
+	SOC_DAPM_ENUM("IF1 45 ADC Swap Mux", rt5682_if1_45_adc_enum);
+
+static const struct snd_kcontrol_new rt5682_if1_67_adc_swap_mux =
+	SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682_if1_67_adc_enum);
+
+static void rt5682_reset(struct regmap *regmap)
+{
+	regmap_write(regmap, RT5682_RESET, 0);
+	regmap_write(regmap, RT5682_I2C_MODE, 1);
+}
+/**
+ * rt5682_sel_asrc_clk_src - select ASRC clock source for a set of filters
+ * @component: SoC audio component device.
+ * @filter_mask: mask of filters.
+ * @clk_src: clock source
+ *
+ * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5682 can
+ * only support standard 32fs or 64fs i2s format, ASRC should be enabled to
+ * support special i2s clock format such as Intel's 100fs(100 * sampling rate).
+ * ASRC function will track i2s clock and generate a corresponding system clock
+ * for codec. This function provides an API to select the clock source for a
+ * set of filters specified by the mask. And the component driver will turn on
+ * ASRC for these filters if ASRC is selected as their clock source.
+ */
+int rt5682_sel_asrc_clk_src(struct snd_soc_component *component,
+		unsigned int filter_mask, unsigned int clk_src)
+{
+
+	switch (clk_src) {
+	case RT5682_CLK_SEL_SYS:
+	case RT5682_CLK_SEL_I2S1_ASRC:
+	case RT5682_CLK_SEL_I2S2_ASRC:
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	if (filter_mask & RT5682_DA_STEREO1_FILTER) {
+		snd_soc_component_update_bits(component, RT5682_PLL_TRACK_2,
+			RT5682_FILTER_CLK_SEL_MASK,
+			clk_src << RT5682_FILTER_CLK_SEL_SFT);
+	}
+
+	if (filter_mask & RT5682_AD_STEREO1_FILTER) {
+		snd_soc_component_update_bits(component, RT5682_PLL_TRACK_3,
+			RT5682_FILTER_CLK_SEL_MASK,
+			clk_src << RT5682_FILTER_CLK_SEL_SFT);
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(rt5682_sel_asrc_clk_src);
+
+static int rt5682_button_detect(struct snd_soc_component *component)
+{
+	int btn_type, val;
+
+	val = snd_soc_component_read32(component, RT5682_4BTN_IL_CMD_1);
+	btn_type = val & 0xfff0;
+	snd_soc_component_write(component, RT5682_4BTN_IL_CMD_1, val);
+	pr_debug("%s btn_type=%x\n", __func__, btn_type);
+	snd_soc_component_update_bits(component,
+		RT5682_SAR_IL_CMD_2, 0x10, 0x10);
+
+	return btn_type;
+}
+
+static void rt5682_enable_push_button_irq(struct snd_soc_component *component,
+		bool enable)
+{
+	if (enable) {
+		snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1,
+			RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_EN);
+		snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_13,
+			RT5682_SAR_SOUR_MASK, RT5682_SAR_SOUR_BTN);
+		snd_soc_component_write(component, RT5682_IL_CMD_1, 0x0040);
+		snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2,
+			RT5682_4BTN_IL_MASK | RT5682_4BTN_IL_RST_MASK,
+			RT5682_4BTN_IL_EN | RT5682_4BTN_IL_NOR);
+		snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3,
+			RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_EN);
+	} else {
+		snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3,
+			RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_DIS);
+		snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1,
+			RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_DIS);
+		snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2,
+			RT5682_4BTN_IL_MASK, RT5682_4BTN_IL_DIS);
+		snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2,
+			RT5682_4BTN_IL_RST_MASK, RT5682_4BTN_IL_RST);
+		snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_13,
+			RT5682_SAR_SOUR_MASK, RT5682_SAR_SOUR_TYPE);
+	}
+}
+
+/**
+ * rt5682_headset_detect - Detect headset.
+ * @component: SoC audio component device.
+ * @jack_insert: Jack insert or not.
+ *
+ * Detect whether is headset or not when jack inserted.
+ *
+ * Returns detect status.
+ */
+static int rt5682_headset_detect(struct snd_soc_component *component,
+		int jack_insert)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	unsigned int val, count;
+
+	if (jack_insert) {
+		snd_soc_dapm_force_enable_pin(dapm, "CBJ Power");
+		snd_soc_dapm_sync(dapm);
+		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
+			RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH);
+
+		count = 0;
+		val = snd_soc_component_read32(component, RT5682_CBJ_CTRL_2)
+			& RT5682_JACK_TYPE_MASK;
+		while (val == 0 && count < 50) {
+			usleep_range(10000, 15000);
+			val = snd_soc_component_read32(component,
+				RT5682_CBJ_CTRL_2) & RT5682_JACK_TYPE_MASK;
+			count++;
+		}
+
+		switch (val) {
+		case 0x1:
+		case 0x2:
+			rt5682->jack_type = SND_JACK_HEADSET;
+			rt5682_enable_push_button_irq(component, true);
+			break;
+		default:
+			rt5682->jack_type = SND_JACK_HEADPHONE;
+		}
+
+	} else {
+		rt5682_enable_push_button_irq(component, false);
+		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
+			RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW);
+		snd_soc_dapm_disable_pin(dapm, "CBJ Power");
+		snd_soc_dapm_sync(dapm);
+
+		rt5682->jack_type = 0;
+	}
+
+	dev_dbg(component->dev, "jack_type = %d\n", rt5682->jack_type);
+	return rt5682->jack_type;
+}
+
+static irqreturn_t rt5682_irq(int irq, void *data)
+{
+	struct rt5682_priv *rt5682 = data;
+
+	mod_delayed_work(system_power_efficient_wq,
+			&rt5682->jack_detect_work, msecs_to_jiffies(250));
+
+	return IRQ_HANDLED;
+}
+
+static void rt5682_jd_check_handler(struct work_struct *work)
+{
+	struct rt5682_priv *rt5682 = container_of(work, struct rt5682_priv,
+		jd_check_work.work);
+
+	if (snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL)
+		& RT5682_JDH_RS_MASK) {
+		/* jack out */
+		rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0);
+
+		snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type,
+				SND_JACK_HEADSET |
+				SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+				SND_JACK_BTN_2 | SND_JACK_BTN_3);
+	} else {
+		schedule_delayed_work(&rt5682->jd_check_work, 500);
+	}
+}
+
+static int rt5682_set_jack_detect(struct snd_soc_component *component,
+	struct snd_soc_jack *hs_jack, void *data)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	switch (rt5682->pdata.jd_src) {
+	case RT5682_JD1:
+		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2,
+			RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL);
+		snd_soc_component_write(component, RT5682_CBJ_CTRL_1, 0xd042);
+		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_3,
+			RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN);
+		snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1,
+			RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN);
+		regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+			RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ);
+		regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+				RT5682_POW_IRQ | RT5682_POW_JDH |
+				RT5682_POW_ANA, RT5682_POW_IRQ |
+				RT5682_POW_JDH | RT5682_POW_ANA);
+		regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2,
+			RT5682_PWR_JDH | RT5682_PWR_JDL,
+			RT5682_PWR_JDH | RT5682_PWR_JDL);
+		regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+			RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK,
+			RT5682_JD1_EN | RT5682_JD1_POL_NOR);
+		mod_delayed_work(system_power_efficient_wq,
+			   &rt5682->jack_detect_work, msecs_to_jiffies(250));
+		break;
+
+	case RT5682_JD_NULL:
+		regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+			RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
+		regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+				RT5682_POW_JDH | RT5682_POW_JDL, 0);
+		break;
+
+	default:
+		dev_warn(component->dev, "Wrong JD source\n");
+		break;
+	}
+
+	rt5682->hs_jack = hs_jack;
+
+	return 0;
+}
+
+static void rt5682_jack_detect_handler(struct work_struct *work)
+{
+	struct rt5682_priv *rt5682 =
+		container_of(work, struct rt5682_priv, jack_detect_work.work);
+	int val, btn_type;
+
+	while (!rt5682->component)
+		usleep_range(10000, 15000);
+
+	while (!rt5682->component->card->instantiated)
+		usleep_range(10000, 15000);
+
+	mutex_lock(&rt5682->calibrate_mutex);
+
+	val = snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL)
+		& RT5682_JDH_RS_MASK;
+	if (!val) {
+		/* jack in */
+		if (rt5682->jack_type == 0) {
+			/* jack was out, report jack type */
+			rt5682->jack_type =
+				rt5682_headset_detect(rt5682->component, 1);
+		} else {
+			/* jack is already in, report button event */
+			rt5682->jack_type = SND_JACK_HEADSET;
+			btn_type = rt5682_button_detect(rt5682->component);
+			/**
+			 * rt5682 can report three kinds of button behavior,
+			 * one click, double click and hold. However,
+			 * currently we will report button pressed/released
+			 * event. So all the three button behaviors are
+			 * treated as button pressed.
+			 */
+			switch (btn_type) {
+			case 0x8000:
+			case 0x4000:
+			case 0x2000:
+				rt5682->jack_type |= SND_JACK_BTN_0;
+				break;
+			case 0x1000:
+			case 0x0800:
+			case 0x0400:
+				rt5682->jack_type |= SND_JACK_BTN_1;
+				break;
+			case 0x0200:
+			case 0x0100:
+			case 0x0080:
+				rt5682->jack_type |= SND_JACK_BTN_2;
+				break;
+			case 0x0040:
+			case 0x0020:
+			case 0x0010:
+				rt5682->jack_type |= SND_JACK_BTN_3;
+				break;
+			case 0x0000: /* unpressed */
+				break;
+			default:
+				btn_type = 0;
+				dev_err(rt5682->component->dev,
+					"Unexpected button code 0x%04x\n",
+					btn_type);
+				break;
+			}
+		}
+	} else {
+		/* jack out */
+		rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0);
+	}
+
+	snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type,
+			SND_JACK_HEADSET |
+			    SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+			    SND_JACK_BTN_2 | SND_JACK_BTN_3);
+
+	if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+		SND_JACK_BTN_2 | SND_JACK_BTN_3))
+		schedule_delayed_work(&rt5682->jd_check_work, 0);
+	else
+		cancel_delayed_work_sync(&rt5682->jd_check_work);
+
+	mutex_unlock(&rt5682->calibrate_mutex);
+}
+
+static const struct snd_kcontrol_new rt5682_snd_controls[] = {
+	/* Headphone Output Volume */
+	SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5682_HPL_GAIN,
+		RT5682_HPR_GAIN, RT5682_G_HP_SFT, 15, 1, hp_vol_tlv),
+
+	/* DAC Digital Volume */
+	SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL,
+		RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 175, 0, dac_vol_tlv),
+
+	/* IN Boost Volume */
+	SOC_SINGLE_TLV("CBJ Boost Volume", RT5682_CBJ_BST_CTRL,
+		RT5682_BST_CBJ_SFT, 8, 0, bst_tlv),
+
+	/* ADC Digital Volume Control */
+	SOC_DOUBLE("STO1 ADC Capture Switch", RT5682_STO1_ADC_DIG_VOL,
+		RT5682_L_MUTE_SFT, RT5682_R_MUTE_SFT, 1, 1),
+	SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5682_STO1_ADC_DIG_VOL,
+		RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 127, 0, adc_vol_tlv),
+
+	/* ADC Boost Volume Control */
+	SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682_STO1_ADC_BOOST,
+		RT5682_STO1_ADC_L_BST_SFT, RT5682_STO1_ADC_R_BST_SFT,
+		3, 0, adc_bst_tlv),
+};
+
+
+static int rt5682_div_sel(struct rt5682_priv *rt5682,
+			  int target, const int div[], int size)
+{
+	int i;
+
+	if (rt5682->sysclk < target) {
+		pr_err("sysclk rate %d is too low\n",
+			rt5682->sysclk);
+		return 0;
+	}
+
+	for (i = 0; i < size - 1; i++) {
+		pr_info("div[%d]=%d\n", i, div[i]);
+		if (target * div[i] == rt5682->sysclk)
+			return i;
+		if (target * div[i + 1] > rt5682->sysclk) {
+			pr_err("can't find div for sysclk %d\n",
+				rt5682->sysclk);
+			return i;
+		}
+	}
+
+	if (target * div[i] < rt5682->sysclk)
+		pr_err("sysclk rate %d is too high\n",
+			rt5682->sysclk);
+
+	return size - 1;
+
+}
+
+/**
+ * set_dmic_clk - Set parameter of dmic.
+ *
+ * @w: DAPM widget.
+ * @kcontrol: The kcontrol of this widget.
+ * @event: Event id.
+ *
+ * Choose dmic clock between 1MHz and 3MHz.
+ * It is better for clock to approximate 3MHz.
+ */
+static int set_dmic_clk(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	int idx = -EINVAL;
+	static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128};
+
+	idx = rt5682_div_sel(rt5682, 1500000, div, ARRAY_SIZE(div));
+
+	snd_soc_component_update_bits(component, RT5682_DMIC_CTRL_1,
+		RT5682_DMIC_CLK_MASK, idx << RT5682_DMIC_CLK_SFT);
+
+	return 0;
+}
+
+static int set_filter_clk(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	int ref, val, reg, sft, mask, idx = -EINVAL;
+	static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48};
+	static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48};
+
+	val = snd_soc_component_read32(component, RT5682_GPIO_CTRL_1) &
+		RT5682_GP4_PIN_MASK;
+	if (w->shift == RT5682_PWR_ADC_S1F_BIT &&
+		val == RT5682_GP4_PIN_ADCDAT2)
+		ref = 256 * rt5682->lrck[RT5682_AIF2];
+	else
+		ref = 256 * rt5682->lrck[RT5682_AIF1];
+
+	idx = rt5682_div_sel(rt5682, ref, div_f, ARRAY_SIZE(div_f));
+
+	if (w->shift == RT5682_PWR_ADC_S1F_BIT) {
+		reg = RT5682_PLL_TRACK_3;
+		sft = RT5682_ADC_OSR_SFT;
+		mask = RT5682_ADC_OSR_MASK;
+	} else {
+		reg = RT5682_PLL_TRACK_2;
+		sft = RT5682_DAC_OSR_SFT;
+		mask = RT5682_DAC_OSR_MASK;
+	}
+
+	snd_soc_component_update_bits(component, reg,
+		RT5682_FILTER_CLK_DIV_MASK, idx << RT5682_FILTER_CLK_DIV_SFT);
+
+	/* select over sample rate */
+	for (idx = 0; idx < ARRAY_SIZE(div_o); idx++) {
+		if (rt5682->sysclk <= 12288000 * div_o[idx])
+			break;
+	}
+
+	snd_soc_component_update_bits(component, RT5682_ADDA_CLK_1,
+		mask, idx << sft);
+
+	return 0;
+}
+
+static int is_sys_clk_from_pll1(struct snd_soc_dapm_widget *w,
+			 struct snd_soc_dapm_widget *sink)
+{
+	unsigned int val;
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	val = snd_soc_component_read32(component, RT5682_GLB_CLK);
+	val &= RT5682_SCLK_SRC_MASK;
+	if (val == RT5682_SCLK_SRC_PLL1)
+		return 1;
+	else
+		return 0;
+}
+
+static int is_using_asrc(struct snd_soc_dapm_widget *w,
+			 struct snd_soc_dapm_widget *sink)
+{
+	unsigned int reg, shift, val;
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	switch (w->shift) {
+	case RT5682_ADC_STO1_ASRC_SFT:
+		reg = RT5682_PLL_TRACK_3;
+		shift = RT5682_FILTER_CLK_SEL_SFT;
+		break;
+	case RT5682_DAC_STO1_ASRC_SFT:
+		reg = RT5682_PLL_TRACK_2;
+		shift = RT5682_FILTER_CLK_SEL_SFT;
+		break;
+	default:
+		return 0;
+	}
+
+	val = (snd_soc_component_read32(component, reg) >> shift) & 0xf;
+	switch (val) {
+	case RT5682_CLK_SEL_I2S1_ASRC:
+	case RT5682_CLK_SEL_I2S2_ASRC:
+		return 1;
+	default:
+		return 0;
+	}
+
+}
+
+/* Digital Mixer */
+static const struct snd_kcontrol_new rt5682_sto1_adc_l_mix[] = {
+	SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER,
+			RT5682_M_STO1_ADC_L1_SFT, 1, 1),
+	SOC_DAPM_SINGLE("ADC2 Switch", RT5682_STO1_ADC_MIXER,
+			RT5682_M_STO1_ADC_L2_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_sto1_adc_r_mix[] = {
+	SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER,
+			RT5682_M_STO1_ADC_R1_SFT, 1, 1),
+	SOC_DAPM_SINGLE("ADC2 Switch", RT5682_STO1_ADC_MIXER,
+			RT5682_M_STO1_ADC_R2_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_dac_l_mix[] = {
+	SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682_AD_DA_MIXER,
+			RT5682_M_ADCMIX_L_SFT, 1, 1),
+	SOC_DAPM_SINGLE("DAC1 Switch", RT5682_AD_DA_MIXER,
+			RT5682_M_DAC1_L_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_dac_r_mix[] = {
+	SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682_AD_DA_MIXER,
+			RT5682_M_ADCMIX_R_SFT, 1, 1),
+	SOC_DAPM_SINGLE("DAC1 Switch", RT5682_AD_DA_MIXER,
+			RT5682_M_DAC1_R_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_sto1_dac_l_mix[] = {
+	SOC_DAPM_SINGLE("DAC L1 Switch", RT5682_STO1_DAC_MIXER,
+			RT5682_M_DAC_L1_STO_L_SFT, 1, 1),
+	SOC_DAPM_SINGLE("DAC R1 Switch", RT5682_STO1_DAC_MIXER,
+			RT5682_M_DAC_R1_STO_L_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_sto1_dac_r_mix[] = {
+	SOC_DAPM_SINGLE("DAC L1 Switch", RT5682_STO1_DAC_MIXER,
+			RT5682_M_DAC_L1_STO_R_SFT, 1, 1),
+	SOC_DAPM_SINGLE("DAC R1 Switch", RT5682_STO1_DAC_MIXER,
+			RT5682_M_DAC_R1_STO_R_SFT, 1, 1),
+};
+
+/* Analog Input Mixer */
+static const struct snd_kcontrol_new rt5682_rec1_l_mix[] = {
+	SOC_DAPM_SINGLE("CBJ Switch", RT5682_REC_MIXER,
+			RT5682_M_CBJ_RM1_L_SFT, 1, 1),
+};
+
+/* STO1 ADC1 Source */
+/* MX-26 [13] [5] */
+static const char * const rt5682_sto1_adc1_src[] = {
+	"DAC MIX", "ADC"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_sto1_adc1l_enum, RT5682_STO1_ADC_MIXER,
+	RT5682_STO1_ADC1L_SRC_SFT, rt5682_sto1_adc1_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adc1l_mux =
+	SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682_sto1_adc1l_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_sto1_adc1r_enum, RT5682_STO1_ADC_MIXER,
+	RT5682_STO1_ADC1R_SRC_SFT, rt5682_sto1_adc1_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adc1r_mux =
+	SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682_sto1_adc1r_enum);
+
+/* STO1 ADC Source */
+/* MX-26 [11:10] [3:2] */
+static const char * const rt5682_sto1_adc_src[] = {
+	"ADC1 L", "ADC1 R"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_sto1_adcl_enum, RT5682_STO1_ADC_MIXER,
+	RT5682_STO1_ADCL_SRC_SFT, rt5682_sto1_adc_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adcl_mux =
+	SOC_DAPM_ENUM("Stereo1 ADCL Source", rt5682_sto1_adcl_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_sto1_adcr_enum, RT5682_STO1_ADC_MIXER,
+	RT5682_STO1_ADCR_SRC_SFT, rt5682_sto1_adc_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adcr_mux =
+	SOC_DAPM_ENUM("Stereo1 ADCR Source", rt5682_sto1_adcr_enum);
+
+/* STO1 ADC2 Source */
+/* MX-26 [12] [4] */
+static const char * const rt5682_sto1_adc2_src[] = {
+	"DAC MIX", "DMIC"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_sto1_adc2l_enum, RT5682_STO1_ADC_MIXER,
+	RT5682_STO1_ADC2L_SRC_SFT, rt5682_sto1_adc2_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adc2l_mux =
+	SOC_DAPM_ENUM("Stereo1 ADC2L Source", rt5682_sto1_adc2l_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_sto1_adc2r_enum, RT5682_STO1_ADC_MIXER,
+	RT5682_STO1_ADC2R_SRC_SFT, rt5682_sto1_adc2_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adc2r_mux =
+	SOC_DAPM_ENUM("Stereo1 ADC2R Source", rt5682_sto1_adc2r_enum);
+
+/* MX-79 [6:4] I2S1 ADC data location */
+static const unsigned int rt5682_if1_adc_slot_values[] = {
+	0,
+	2,
+	4,
+	6,
+};
+
+static const char * const rt5682_if1_adc_slot_src[] = {
+	"Slot 0", "Slot 2", "Slot 4", "Slot 6"
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_if1_adc_slot_enum,
+	RT5682_TDM_CTRL, RT5682_TDM_ADC_LCA_SFT, RT5682_TDM_ADC_LCA_MASK,
+	rt5682_if1_adc_slot_src, rt5682_if1_adc_slot_values);
+
+static const struct snd_kcontrol_new rt5682_if1_adc_slot_mux =
+	SOC_DAPM_ENUM("IF1 ADC Slot location", rt5682_if1_adc_slot_enum);
+
+/* Analog DAC L1 Source, Analog DAC R1 Source*/
+/* MX-2B [4], MX-2B [0]*/
+static const char * const rt5682_alg_dac1_src[] = {
+	"Stereo1 DAC Mixer", "DAC1"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_alg_dac_l1_enum, RT5682_A_DAC1_MUX,
+	RT5682_A_DACL1_SFT, rt5682_alg_dac1_src);
+
+static const struct snd_kcontrol_new rt5682_alg_dac_l1_mux =
+	SOC_DAPM_ENUM("Analog DAC L1 Source", rt5682_alg_dac_l1_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_alg_dac_r1_enum, RT5682_A_DAC1_MUX,
+	RT5682_A_DACR1_SFT, rt5682_alg_dac1_src);
+
+static const struct snd_kcontrol_new rt5682_alg_dac_r1_mux =
+	SOC_DAPM_ENUM("Analog DAC R1 Source", rt5682_alg_dac_r1_enum);
+
+/* Out Switch */
+static const struct snd_kcontrol_new hpol_switch =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1,
+					RT5682_L_MUTE_SFT, 1, 1);
+static const struct snd_kcontrol_new hpor_switch =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1,
+					RT5682_R_MUTE_SFT, 1, 1);
+
+static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		snd_soc_component_write(component,
+			RT5682_HP_LOGIC_CTRL_2, 0x0012);
+		snd_soc_component_write(component,
+			RT5682_HP_CTRL_2, 0x6000);
+		snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1,
+			RT5682_NG2_EN_MASK, RT5682_NG2_EN);
+		snd_soc_component_update_bits(component,
+			RT5682_DEPOP_1, 0x60, 0x60);
+		break;
+
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_component_update_bits(component,
+			RT5682_DEPOP_1, 0x60, 0x0);
+		snd_soc_component_write(component,
+			RT5682_HP_CTRL_2, 0x0000);
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+
+}
+
+static int set_dmic_power(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		/*Add delay to avoid pop noise*/
+		msleep(150);
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static int rt5655_set_verf(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		switch (w->shift) {
+		case RT5682_PWR_VREF1_BIT:
+			snd_soc_component_update_bits(component,
+				RT5682_PWR_ANLG_1, RT5682_PWR_FV1, 0);
+			break;
+
+		case RT5682_PWR_VREF2_BIT:
+			snd_soc_component_update_bits(component,
+				RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0);
+			break;
+
+		default:
+			break;
+		}
+		break;
+
+	case SND_SOC_DAPM_POST_PMU:
+		usleep_range(15000, 20000);
+		switch (w->shift) {
+		case RT5682_PWR_VREF1_BIT:
+			snd_soc_component_update_bits(component,
+				RT5682_PWR_ANLG_1, RT5682_PWR_FV1,
+				RT5682_PWR_FV1);
+			break;
+
+		case RT5682_PWR_VREF2_BIT:
+			snd_soc_component_update_bits(component,
+				RT5682_PWR_ANLG_1, RT5682_PWR_FV2,
+				RT5682_PWR_FV2);
+			break;
+
+		default:
+			break;
+		}
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static const unsigned int rt5682_adcdat_pin_values[] = {
+	1,
+	3,
+};
+
+static const char * const rt5682_adcdat_pin_select[] = {
+	"ADCDAT1",
+	"ADCDAT2",
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_adcdat_pin_enum,
+	RT5682_GPIO_CTRL_1, RT5682_GP4_PIN_SFT, RT5682_GP4_PIN_MASK,
+	rt5682_adcdat_pin_select, rt5682_adcdat_pin_values);
+
+static const struct snd_kcontrol_new rt5682_adcdat_pin_ctrl =
+	SOC_DAPM_ENUM("ADCDAT", rt5682_adcdat_pin_enum);
+
+static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
+	SND_SOC_DAPM_SUPPLY("LDO2", RT5682_PWR_ANLG_3, RT5682_PWR_LDO2_BIT,
+		0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL1", RT5682_PWR_ANLG_3, RT5682_PWR_PLL_BIT,
+		0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL2B", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2B_BIT,
+		0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL2F", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2F_BIT,
+		0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0,
+		rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0,
+		rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+
+	/* ASRC */
+	SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1,
+		RT5682_DAC_STO1_ASRC_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5682_PLL_TRACK_1,
+		RT5682_ADC_STO1_ASRC_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY_S("AD ASRC", 1, RT5682_PLL_TRACK_1,
+		RT5682_AD_ASRC_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY_S("DA ASRC", 1, RT5682_PLL_TRACK_1,
+		RT5682_DA_ASRC_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY_S("DMIC ASRC", 1, RT5682_PLL_TRACK_1,
+		RT5682_DMIC_ASRC_SFT, 0, NULL, 0),
+
+	/* Input Side */
+	SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5682_PWR_ANLG_2, RT5682_PWR_MB1_BIT,
+		0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5682_PWR_ANLG_2, RT5682_PWR_MB2_BIT,
+		0, NULL, 0),
+
+	/* Input Lines */
+	SND_SOC_DAPM_INPUT("DMIC L1"),
+	SND_SOC_DAPM_INPUT("DMIC R1"),
+
+	SND_SOC_DAPM_INPUT("IN1P"),
+
+	SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0,
+		set_dmic_clk, SND_SOC_DAPM_PRE_PMU),
+	SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5682_DMIC_CTRL_1,
+		RT5682_DMIC_1_EN_SFT, 0, set_dmic_power, SND_SOC_DAPM_POST_PMU),
+
+	/* Boost */
+	SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM,
+		0, 0, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("CBJ Power", RT5682_PWR_ANLG_3,
+		RT5682_PWR_CBJ_BIT, 0, NULL, 0),
+
+	/* REC Mixer */
+	SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5682_rec1_l_mix,
+		ARRAY_SIZE(rt5682_rec1_l_mix)),
+	SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5682_PWR_ANLG_2,
+		RT5682_PWR_RM1_L_BIT, 0, NULL, 0),
+
+	/* ADCs */
+	SND_SOC_DAPM_ADC("ADC1 L", NULL, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_ADC("ADC1 R", NULL, SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_SUPPLY("ADC1 L Power", RT5682_PWR_DIG_1,
+		RT5682_PWR_ADC_L1_BIT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC1 R Power", RT5682_PWR_DIG_1,
+		RT5682_PWR_ADC_R1_BIT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC1 clock", RT5682_CHOP_ADC,
+		RT5682_CKGEN_ADC1_SFT, 0, NULL, 0),
+
+	/* ADC Mux */
+	SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0,
+		&rt5682_sto1_adc1l_mux),
+	SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0,
+		&rt5682_sto1_adc1r_mux),
+	SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0,
+		&rt5682_sto1_adc2l_mux),
+	SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0,
+		&rt5682_sto1_adc2r_mux),
+	SND_SOC_DAPM_MUX("Stereo1 ADC L Mux", SND_SOC_NOPM, 0, 0,
+		&rt5682_sto1_adcl_mux),
+	SND_SOC_DAPM_MUX("Stereo1 ADC R Mux", SND_SOC_NOPM, 0, 0,
+		&rt5682_sto1_adcr_mux),
+	SND_SOC_DAPM_MUX("IF1_ADC Mux", SND_SOC_NOPM, 0, 0,
+		&rt5682_if1_adc_slot_mux),
+
+	/* ADC Mixer */
+	SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5682_PWR_DIG_2,
+		RT5682_PWR_ADC_S1F_BIT, 0, set_filter_clk,
+		SND_SOC_DAPM_PRE_PMU),
+	SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", RT5682_STO1_ADC_DIG_VOL,
+		RT5682_L_MUTE_SFT, 1, rt5682_sto1_adc_l_mix,
+		ARRAY_SIZE(rt5682_sto1_adc_l_mix)),
+	SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682_STO1_ADC_DIG_VOL,
+		RT5682_R_MUTE_SFT, 1, rt5682_sto1_adc_r_mix,
+		ARRAY_SIZE(rt5682_sto1_adc_r_mix)),
+	SND_SOC_DAPM_SUPPLY("BTN Detection Mode", RT5682_SAR_IL_CMD_1,
+		14, 1, NULL, 0),
+
+	/* ADC PGA */
+	SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Digital Interface */
+	SND_SOC_DAPM_SUPPLY("I2S1", RT5682_PWR_DIG_1, RT5682_PWR_I2S1_BIT,
+		0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("I2S2", RT5682_PWR_DIG_1, RT5682_PWR_I2S2_BIT,
+		0, NULL, 0),
+	SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Digital Interface Select */
+	SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+			&rt5682_if1_01_adc_swap_mux),
+	SND_SOC_DAPM_MUX("IF1 23 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+			&rt5682_if1_23_adc_swap_mux),
+	SND_SOC_DAPM_MUX("IF1 45 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+			&rt5682_if1_45_adc_swap_mux),
+	SND_SOC_DAPM_MUX("IF1 67 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+			&rt5682_if1_67_adc_swap_mux),
+	SND_SOC_DAPM_MUX("IF2 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+			&rt5682_if2_adc_swap_mux),
+
+	SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0,
+			&rt5682_adcdat_pin_ctrl),
+
+	/* Audio Interface */
+	SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0,
+		RT5682_I2S1_SDP, RT5682_SEL_ADCDAT_SFT, 1),
+	SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0,
+		RT5682_I2S2_SDP, RT5682_I2S2_PIN_CFG_SFT, 1),
+	SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+
+	/* Output Side */
+	/* DAC mixer before sound effect  */
+	SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0,
+		rt5682_dac_l_mix, ARRAY_SIZE(rt5682_dac_l_mix)),
+	SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0,
+		rt5682_dac_r_mix, ARRAY_SIZE(rt5682_dac_r_mix)),
+
+	/* DAC channel Mux */
+	SND_SOC_DAPM_MUX("DAC L1 Source", SND_SOC_NOPM, 0, 0,
+		&rt5682_alg_dac_l1_mux),
+	SND_SOC_DAPM_MUX("DAC R1 Source", SND_SOC_NOPM, 0, 0,
+		&rt5682_alg_dac_r1_mux),
+
+	/* DAC Mixer */
+	SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5682_PWR_DIG_2,
+		RT5682_PWR_DAC_S1F_BIT, 0, set_filter_clk,
+		SND_SOC_DAPM_PRE_PMU),
+	SND_SOC_DAPM_MIXER("Stereo1 DAC MIXL", SND_SOC_NOPM, 0, 0,
+		rt5682_sto1_dac_l_mix, ARRAY_SIZE(rt5682_sto1_dac_l_mix)),
+	SND_SOC_DAPM_MIXER("Stereo1 DAC MIXR", SND_SOC_NOPM, 0, 0,
+		rt5682_sto1_dac_r_mix, ARRAY_SIZE(rt5682_sto1_dac_r_mix)),
+
+	/* DACs */
+	SND_SOC_DAPM_DAC("DAC L1", NULL, RT5682_PWR_DIG_1,
+		RT5682_PWR_DAC_L1_BIT, 0),
+	SND_SOC_DAPM_DAC("DAC R1", NULL, RT5682_PWR_DIG_1,
+		RT5682_PWR_DAC_R1_BIT, 0),
+	SND_SOC_DAPM_SUPPLY_S("DAC 1 Clock", 3, RT5682_CHOP_DAC,
+		RT5682_CKGEN_DAC1_SFT, 0, NULL, 0),
+
+	/* HPO */
+	SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5682_hp_event,
+		SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU),
+
+	SND_SOC_DAPM_SUPPLY("HP Amp L", RT5682_PWR_ANLG_1,
+		RT5682_PWR_HA_L_BIT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1,
+		RT5682_PWR_HA_R_BIT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1,
+		RT5682_PUMP_EN_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1,
+		RT5682_CAPLESS_EN_SFT, 0, NULL, 0),
+
+	SND_SOC_DAPM_SWITCH("HPOL Playback", SND_SOC_NOPM, 0, 0,
+		&hpol_switch),
+	SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0,
+		&hpor_switch),
+
+	/* CLK DET */
+	SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682_CLK_DET,
+		RT5682_SYS_CLK_DET_SFT,	0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("CLKDET PLL1", RT5682_CLK_DET,
+		RT5682_PLL1_CLK_DET_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("CLKDET PLL2", RT5682_CLK_DET,
+		RT5682_PLL2_CLK_DET_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("CLKDET", RT5682_CLK_DET,
+		RT5682_POW_CLK_DET_SFT, 0, NULL, 0),
+
+	/* Output Lines */
+	SND_SOC_DAPM_OUTPUT("HPOL"),
+	SND_SOC_DAPM_OUTPUT("HPOR"),
+
+};
+
+static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
+	/*PLL*/
+	{"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1},
+	{"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1},
+
+	/*ASRC*/
+	{"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc},
+	{"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc},
+	{"ADC STO1 ASRC", NULL, "AD ASRC"},
+	{"ADC STO1 ASRC", NULL, "CLKDET"},
+	{"DAC STO1 ASRC", NULL, "DA ASRC"},
+	{"DAC STO1 ASRC", NULL, "CLKDET"},
+
+	/*Vref*/
+	{"MICBIAS1", NULL, "Vref1"},
+	{"MICBIAS1", NULL, "Vref2"},
+	{"MICBIAS2", NULL, "Vref1"},
+	{"MICBIAS2", NULL, "Vref2"},
+
+	{"CLKDET SYS", NULL, "CLKDET"},
+
+	{"IN1P", NULL, "LDO2"},
+
+	{"BST1 CBJ", NULL, "IN1P"},
+	{"BST1 CBJ", NULL, "CBJ Power"},
+	{"CBJ Power", NULL, "Vref2"},
+
+	{"RECMIX1L", "CBJ Switch", "BST1 CBJ"},
+	{"RECMIX1L", NULL, "RECMIX1L Power"},
+
+	{"ADC1 L", NULL, "RECMIX1L"},
+	{"ADC1 L", NULL, "ADC1 L Power"},
+	{"ADC1 L", NULL, "ADC1 clock"},
+
+	{"DMIC L1", NULL, "DMIC CLK"},
+	{"DMIC L1", NULL, "DMIC1 Power"},
+	{"DMIC R1", NULL, "DMIC CLK"},
+	{"DMIC R1", NULL, "DMIC1 Power"},
+	{"DMIC CLK", NULL, "DMIC ASRC"},
+
+	{"Stereo1 ADC L Mux", "ADC1 L", "ADC1 L"},
+	{"Stereo1 ADC L Mux", "ADC1 R", "ADC1 R"},
+	{"Stereo1 ADC R Mux", "ADC1 L", "ADC1 L"},
+	{"Stereo1 ADC R Mux", "ADC1 R", "ADC1 R"},
+
+	{"Stereo1 ADC L1 Mux", "ADC", "Stereo1 ADC L Mux"},
+	{"Stereo1 ADC L1 Mux", "DAC MIX", "Stereo1 DAC MIXL"},
+	{"Stereo1 ADC L2 Mux", "DMIC", "DMIC L1"},
+	{"Stereo1 ADC L2 Mux", "DAC MIX", "Stereo1 DAC MIXL"},
+
+	{"Stereo1 ADC R1 Mux", "ADC", "Stereo1 ADC R Mux"},
+	{"Stereo1 ADC R1 Mux", "DAC MIX", "Stereo1 DAC MIXR"},
+	{"Stereo1 ADC R2 Mux", "DMIC", "DMIC R1"},
+	{"Stereo1 ADC R2 Mux", "DAC MIX", "Stereo1 DAC MIXR"},
+
+	{"Stereo1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux"},
+	{"Stereo1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux"},
+	{"Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter"},
+
+	{"Stereo1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux"},
+	{"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"},
+	{"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"},
+
+	{"ADC Stereo1 Filter", NULL, "BTN Detection Mode"},
+
+	{"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"},
+	{"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"},
+
+	{"IF1 01 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+	{"IF1 01 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+	{"IF1 01 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+	{"IF1 01 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+	{"IF1 23 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+	{"IF1 23 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+	{"IF1 23 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+	{"IF1 23 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+	{"IF1 45 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+	{"IF1 45 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+	{"IF1 45 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+	{"IF1 45 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+	{"IF1 67 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+	{"IF1 67 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+	{"IF1 67 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+	{"IF1 67 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+
+	{"IF1_ADC Mux", "Slot 0", "IF1 01 ADC Swap Mux"},
+	{"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"},
+	{"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"},
+	{"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"},
+	{"IF1_ADC Mux", NULL, "I2S1"},
+	{"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"},
+	{"AIF1TX", NULL, "ADCDAT Mux"},
+	{"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+	{"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+	{"IF2 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+	{"IF2 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+	{"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"},
+	{"AIF2TX", NULL, "ADCDAT Mux"},
+
+	{"IF1 DAC1 L", NULL, "AIF1RX"},
+	{"IF1 DAC1 L", NULL, "I2S1"},
+	{"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"},
+	{"IF1 DAC1 R", NULL, "AIF1RX"},
+	{"IF1 DAC1 R", NULL, "I2S1"},
+	{"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"},
+
+	{"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"},
+	{"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"},
+	{"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"},
+	{"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"},
+
+	{"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"},
+	{"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"},
+
+	{"Stereo1 DAC MIXR", "DAC R1 Switch", "DAC1 MIXR"},
+	{"Stereo1 DAC MIXR", "DAC L1 Switch", "DAC1 MIXL"},
+
+	{"DAC L1 Source", "DAC1", "DAC1 MIXL"},
+	{"DAC L1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXL"},
+	{"DAC R1 Source", "DAC1", "DAC1 MIXR"},
+	{"DAC R1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXR"},
+
+	{"DAC L1", NULL, "DAC L1 Source"},
+	{"DAC R1", NULL, "DAC R1 Source"},
+
+	{"DAC L1", NULL, "DAC 1 Clock"},
+	{"DAC R1", NULL, "DAC 1 Clock"},
+
+	{"HP Amp", NULL, "DAC L1"},
+	{"HP Amp", NULL, "DAC R1"},
+	{"HP Amp", NULL, "HP Amp L"},
+	{"HP Amp", NULL, "HP Amp R"},
+	{"HP Amp", NULL, "Capless"},
+	{"HP Amp", NULL, "Charge Pump"},
+	{"HP Amp", NULL, "CLKDET SYS"},
+	{"HP Amp", NULL, "CBJ Power"},
+	{"HP Amp", NULL, "Vref2"},
+	{"HPOL Playback", "Switch", "HP Amp"},
+	{"HPOR Playback", "Switch", "HP Amp"},
+	{"HPOL", NULL, "HPOL Playback"},
+	{"HPOR", NULL, "HPOR Playback"},
+};
+
+static int rt5682_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+			unsigned int rx_mask, int slots, int slot_width)
+{
+	struct snd_soc_component *component = dai->component;
+	unsigned int cl, val = 0;
+
+	if (tx_mask || rx_mask)
+		snd_soc_component_update_bits(component, RT5682_TDM_ADDA_CTRL_2,
+			RT5682_TDM_EN, RT5682_TDM_EN);
+	else
+		snd_soc_component_update_bits(component, RT5682_TDM_ADDA_CTRL_2,
+			RT5682_TDM_EN, 0);
+
+	switch (slots) {
+	case 4:
+		val |= RT5682_TDM_TX_CH_4;
+		val |= RT5682_TDM_RX_CH_4;
+		break;
+	case 6:
+		val |= RT5682_TDM_TX_CH_6;
+		val |= RT5682_TDM_RX_CH_6;
+		break;
+	case 8:
+		val |= RT5682_TDM_TX_CH_8;
+		val |= RT5682_TDM_RX_CH_8;
+		break;
+	case 2:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component, RT5682_TDM_CTRL,
+		RT5682_TDM_TX_CH_MASK | RT5682_TDM_RX_CH_MASK, val);
+
+	switch (slot_width) {
+	case 8:
+		if (tx_mask || rx_mask)
+			return -EINVAL;
+		cl = RT5682_I2S1_TX_CHL_8 | RT5682_I2S1_RX_CHL_8;
+		break;
+	case 16:
+		val = RT5682_TDM_CL_16;
+		cl = RT5682_I2S1_TX_CHL_16 | RT5682_I2S1_RX_CHL_16;
+		break;
+	case 20:
+		val = RT5682_TDM_CL_20;
+		cl = RT5682_I2S1_TX_CHL_20 | RT5682_I2S1_RX_CHL_20;
+		break;
+	case 24:
+		val = RT5682_TDM_CL_24;
+		cl = RT5682_I2S1_TX_CHL_24 | RT5682_I2S1_RX_CHL_24;
+		break;
+	case 32:
+		val = RT5682_TDM_CL_32;
+		cl = RT5682_I2S1_TX_CHL_32 | RT5682_I2S1_RX_CHL_32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL,
+		RT5682_TDM_CL_MASK, val);
+	snd_soc_component_update_bits(component, RT5682_I2S1_SDP,
+		RT5682_I2S1_TX_CHL_MASK | RT5682_I2S1_RX_CHL_MASK, cl);
+
+	return 0;
+}
+
+
+static int rt5682_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	unsigned int len_1 = 0, len_2 = 0;
+	int pre_div, frame_size;
+
+	rt5682->lrck[dai->id] = params_rate(params);
+	pre_div = rl6231_get_clk_info(rt5682->sysclk, rt5682->lrck[dai->id]);
+
+	frame_size = snd_soc_params_to_frame_size(params);
+	if (frame_size < 0) {
+		dev_err(component->dev, "Unsupported frame size: %d\n",
+			frame_size);
+		return -EINVAL;
+	}
+
+	dev_dbg(dai->dev, "lrck is %dHz and pre_div is %d for iis %d\n",
+				rt5682->lrck[dai->id], pre_div, dai->id);
+
+	switch (params_width(params)) {
+	case 16:
+		break;
+	case 20:
+		len_1 |= RT5682_I2S1_DL_20;
+		len_2 |= RT5682_I2S2_DL_20;
+		break;
+	case 24:
+		len_1 |= RT5682_I2S1_DL_24;
+		len_2 |= RT5682_I2S2_DL_24;
+		break;
+	case 32:
+		len_1 |= RT5682_I2S1_DL_32;
+		len_2 |= RT5682_I2S2_DL_24;
+		break;
+	case 8:
+		len_1 |= RT5682_I2S2_DL_8;
+		len_2 |= RT5682_I2S2_DL_8;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (dai->id) {
+	case RT5682_AIF1:
+		snd_soc_component_update_bits(component, RT5682_I2S1_SDP,
+			RT5682_I2S1_DL_MASK, len_1);
+		if (rt5682->master[RT5682_AIF1]) {
+			snd_soc_component_update_bits(component,
+				RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK,
+				pre_div << RT5682_I2S_M_DIV_SFT);
+		}
+		if (params_channels(params) == 1) /* mono mode */
+			snd_soc_component_update_bits(component,
+				RT5682_I2S1_SDP, RT5682_I2S1_MONO_MASK,
+				RT5682_I2S1_MONO_EN);
+		else
+			snd_soc_component_update_bits(component,
+				RT5682_I2S1_SDP, RT5682_I2S1_MONO_MASK,
+				RT5682_I2S1_MONO_DIS);
+		break;
+	case RT5682_AIF2:
+		snd_soc_component_update_bits(component, RT5682_I2S2_SDP,
+			RT5682_I2S2_DL_MASK, len_2);
+		if (rt5682->master[RT5682_AIF2]) {
+			snd_soc_component_update_bits(component,
+				RT5682_I2S_M_CLK_CTRL_1, RT5682_I2S2_M_PD_MASK,
+				pre_div << RT5682_I2S2_M_PD_SFT);
+		}
+		if (params_channels(params) == 1) /* mono mode */
+			snd_soc_component_update_bits(component,
+				RT5682_I2S2_SDP, RT5682_I2S2_MONO_MASK,
+				RT5682_I2S2_MONO_EN);
+		else
+			snd_soc_component_update_bits(component,
+				RT5682_I2S2_SDP, RT5682_I2S2_MONO_MASK,
+				RT5682_I2S2_MONO_DIS);
+		break;
+	default:
+		dev_err(component->dev, "Invalid dai->id: %d\n", dai->id);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int rt5682_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_component *component = dai->component;
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	unsigned int reg_val = 0, tdm_ctrl = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		rt5682->master[dai->id] = 1;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		rt5682->master[dai->id] = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		reg_val |= RT5682_I2S_BP_INV;
+		tdm_ctrl |= RT5682_TDM_S_BP_INV;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		if (dai->id == RT5682_AIF1)
+			tdm_ctrl |= RT5682_TDM_S_LP_INV | RT5682_TDM_M_BP_INV;
+		else
+			return -EINVAL;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		if (dai->id == RT5682_AIF1)
+			tdm_ctrl |= RT5682_TDM_S_BP_INV | RT5682_TDM_S_LP_INV |
+				    RT5682_TDM_M_BP_INV | RT5682_TDM_M_LP_INV;
+		else
+			return -EINVAL;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		reg_val |= RT5682_I2S_DF_LEFT;
+		tdm_ctrl |= RT5682_TDM_DF_LEFT;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		reg_val |= RT5682_I2S_DF_PCM_A;
+		tdm_ctrl |= RT5682_TDM_DF_PCM_A;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		reg_val |= RT5682_I2S_DF_PCM_B;
+		tdm_ctrl |= RT5682_TDM_DF_PCM_B;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (dai->id) {
+	case RT5682_AIF1:
+		snd_soc_component_update_bits(component, RT5682_I2S1_SDP,
+			RT5682_I2S_DF_MASK, reg_val);
+		snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL,
+			RT5682_TDM_MS_MASK | RT5682_TDM_S_BP_MASK |
+			RT5682_TDM_DF_MASK | RT5682_TDM_M_BP_MASK |
+			RT5682_TDM_M_LP_MASK | RT5682_TDM_S_LP_MASK,
+			tdm_ctrl | rt5682->master[dai->id]);
+		break;
+	case RT5682_AIF2:
+		if (rt5682->master[dai->id] == 0)
+			reg_val |= RT5682_I2S2_MS_S;
+		snd_soc_component_update_bits(component, RT5682_I2S2_SDP,
+			RT5682_I2S2_MS_MASK | RT5682_I2S_BP_MASK |
+			RT5682_I2S_DF_MASK, reg_val);
+		break;
+	default:
+		dev_err(component->dev, "Invalid dai->id: %d\n", dai->id);
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static int rt5682_set_component_sysclk(struct snd_soc_component *component,
+		int clk_id, int source, unsigned int freq, int dir)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	unsigned int reg_val = 0, src = 0;
+
+	if (freq == rt5682->sysclk && clk_id == rt5682->sysclk_src)
+		return 0;
+
+	switch (clk_id) {
+	case RT5682_SCLK_S_MCLK:
+		reg_val |= RT5682_SCLK_SRC_MCLK;
+		src = RT5682_CLK_SRC_MCLK;
+		break;
+	case RT5682_SCLK_S_PLL1:
+		reg_val |= RT5682_SCLK_SRC_PLL1;
+		src = RT5682_CLK_SRC_PLL1;
+		break;
+	case RT5682_SCLK_S_PLL2:
+		reg_val |= RT5682_SCLK_SRC_PLL2;
+		src = RT5682_CLK_SRC_PLL2;
+		break;
+	case RT5682_SCLK_S_RCCLK:
+		reg_val |= RT5682_SCLK_SRC_RCCLK;
+		src = RT5682_CLK_SRC_RCCLK;
+		break;
+	default:
+		dev_err(component->dev, "Invalid clock id (%d)\n", clk_id);
+		return -EINVAL;
+	}
+	snd_soc_component_update_bits(component, RT5682_GLB_CLK,
+		RT5682_SCLK_SRC_MASK, reg_val);
+
+	if (rt5682->master[RT5682_AIF2]) {
+		snd_soc_component_update_bits(component,
+			RT5682_I2S_M_CLK_CTRL_1, RT5682_I2S2_SRC_MASK,
+			src << RT5682_I2S2_SRC_SFT);
+	}
+
+	rt5682->sysclk = freq;
+	rt5682->sysclk_src = clk_id;
+
+	dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n",
+		freq, clk_id);
+
+	return 0;
+}
+
+static int rt5682_set_component_pll(struct snd_soc_component *component,
+		int pll_id, int source, unsigned int freq_in,
+		unsigned int freq_out)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	struct rl6231_pll_code pll_code;
+	int ret;
+
+	if (source == rt5682->pll_src && freq_in == rt5682->pll_in &&
+	    freq_out == rt5682->pll_out)
+		return 0;
+
+	if (!freq_in || !freq_out) {
+		dev_dbg(component->dev, "PLL disabled\n");
+
+		rt5682->pll_in = 0;
+		rt5682->pll_out = 0;
+		snd_soc_component_update_bits(component, RT5682_GLB_CLK,
+			RT5682_SCLK_SRC_MASK, RT5682_SCLK_SRC_MCLK);
+		return 0;
+	}
+
+	switch (source) {
+	case RT5682_PLL1_S_MCLK:
+		snd_soc_component_update_bits(component, RT5682_GLB_CLK,
+			RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_MCLK);
+		break;
+	case RT5682_PLL1_S_BCLK1:
+		snd_soc_component_update_bits(component, RT5682_GLB_CLK,
+				RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_BCLK1);
+		break;
+	default:
+		dev_err(component->dev, "Unknown PLL Source %d\n", source);
+		return -EINVAL;
+	}
+
+	ret = rl6231_pll_calc(freq_in, freq_out, &pll_code);
+	if (ret < 0) {
+		dev_err(component->dev, "Unsupport input clock %d\n", freq_in);
+		return ret;
+	}
+
+	dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n",
+		pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code),
+		pll_code.n_code, pll_code.k_code);
+
+	snd_soc_component_write(component, RT5682_PLL_CTRL_1,
+		pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code);
+	snd_soc_component_write(component, RT5682_PLL_CTRL_2,
+		(pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT |
+		pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST);
+
+	rt5682->pll_in = freq_in;
+	rt5682->pll_out = freq_out;
+	rt5682->pll_src = source;
+
+	return 0;
+}
+
+static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+{
+	struct snd_soc_component *component = dai->component;
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	rt5682->bclk[dai->id] = ratio;
+
+	switch (ratio) {
+	case 64:
+		snd_soc_component_update_bits(component, RT5682_ADDA_CLK_2,
+			RT5682_I2S2_BCLK_MS2_MASK,
+			RT5682_I2S2_BCLK_MS2_64);
+		break;
+	case 32:
+		snd_soc_component_update_bits(component, RT5682_ADDA_CLK_2,
+			RT5682_I2S2_BCLK_MS2_MASK,
+			RT5682_I2S2_BCLK_MS2_32);
+		break;
+	default:
+		dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int rt5682_set_bias_level(struct snd_soc_component *component,
+			enum snd_soc_bias_level level)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1,
+			RT5682_PWR_MB | RT5682_PWR_BG,
+			RT5682_PWR_MB | RT5682_PWR_BG);
+		regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1,
+			RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO,
+			RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1,
+			RT5682_PWR_MB, RT5682_PWR_MB);
+		regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1,
+			RT5682_DIG_GATE_CTRL, RT5682_DIG_GATE_CTRL);
+		break;
+	case SND_SOC_BIAS_OFF:
+		regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1,
+			RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO, 0);
+		regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1,
+			RT5682_PWR_MB | RT5682_PWR_BG, 0);
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int rt5682_probe(struct snd_soc_component *component)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	rt5682->component = component;
+
+	return 0;
+}
+
+static void rt5682_remove(struct snd_soc_component *component)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	rt5682_reset(rt5682->regmap);
+}
+
+#ifdef CONFIG_PM
+static int rt5682_suspend(struct snd_soc_component *component)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	regcache_cache_only(rt5682->regmap, true);
+	regcache_mark_dirty(rt5682->regmap);
+	return 0;
+}
+
+static int rt5682_resume(struct snd_soc_component *component)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	regcache_cache_only(rt5682->regmap, false);
+	regcache_sync(rt5682->regmap);
+
+	return 0;
+}
+#else
+#define rt5682_suspend NULL
+#define rt5682_resume NULL
+#endif
+
+#define RT5682_STEREO_RATES SNDRV_PCM_RATE_8000_192000
+#define RT5682_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
+
+static const struct snd_soc_dai_ops rt5682_aif1_dai_ops = {
+	.hw_params = rt5682_hw_params,
+	.set_fmt = rt5682_set_dai_fmt,
+	.set_tdm_slot = rt5682_set_tdm_slot,
+};
+
+static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = {
+	.hw_params = rt5682_hw_params,
+	.set_fmt = rt5682_set_dai_fmt,
+	.set_bclk_ratio = rt5682_set_bclk_ratio,
+};
+
+static struct snd_soc_dai_driver rt5682_dai[] = {
+	{
+		.name = "rt5682-aif1",
+		.id = RT5682_AIF1,
+		.playback = {
+			.stream_name = "AIF1 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = RT5682_STEREO_RATES,
+			.formats = RT5682_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIF1 Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = RT5682_STEREO_RATES,
+			.formats = RT5682_FORMATS,
+		},
+		.ops = &rt5682_aif1_dai_ops,
+	},
+	{
+		.name = "rt5682-aif2",
+		.id = RT5682_AIF2,
+		.capture = {
+			.stream_name = "AIF2 Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = RT5682_STEREO_RATES,
+			.formats = RT5682_FORMATS,
+		},
+		.ops = &rt5682_aif2_dai_ops,
+	},
+};
+
+static const struct snd_soc_component_driver soc_component_dev_rt5682 = {
+	.probe = rt5682_probe,
+	.remove = rt5682_remove,
+	.suspend = rt5682_suspend,
+	.resume = rt5682_resume,
+	.set_bias_level = rt5682_set_bias_level,
+	.controls = rt5682_snd_controls,
+	.num_controls = ARRAY_SIZE(rt5682_snd_controls),
+	.dapm_widgets = rt5682_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(rt5682_dapm_widgets),
+	.dapm_routes = rt5682_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(rt5682_dapm_routes),
+	.set_sysclk = rt5682_set_component_sysclk,
+	.set_pll = rt5682_set_component_pll,
+	.set_jack = rt5682_set_jack_detect,
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
+static const struct regmap_config rt5682_regmap = {
+	.reg_bits = 16,
+	.val_bits = 16,
+	.max_register = RT5682_I2C_MODE,
+	.volatile_reg = rt5682_volatile_register,
+	.readable_reg = rt5682_readable_register,
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = rt5682_reg,
+	.num_reg_defaults = ARRAY_SIZE(rt5682_reg),
+	.use_single_rw = true,
+};
+
+static const struct i2c_device_id rt5682_i2c_id[] = {
+	{"rt5682", 0},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, rt5682_i2c_id);
+
+static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev)
+{
+
+	device_property_read_u32(dev, "realtek,dmic1-data-pin",
+		&rt5682->pdata.dmic1_data_pin);
+	device_property_read_u32(dev, "realtek,dmic1-clk-pin",
+		&rt5682->pdata.dmic1_clk_pin);
+	device_property_read_u32(dev, "realtek,jd-src",
+		&rt5682->pdata.jd_src);
+
+	rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node,
+		"realtek,ldo1-en-gpios", 0);
+
+	return 0;
+}
+
+static void rt5682_calibrate(struct rt5682_priv *rt5682)
+{
+	int value, count;
+
+	mutex_lock(&rt5682->calibrate_mutex);
+
+	rt5682_reset(rt5682->regmap);
+	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf);
+	usleep_range(15000, 20000);
+	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf);
+	regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380);
+	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8001);
+	regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000);
+	regmap_write(rt5682->regmap, RT5682_STO1_DAC_MIXER, 0x2080);
+	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x4040);
+	regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0069);
+	regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000);
+	regmap_write(rt5682->regmap, RT5682_HP_CTRL_2, 0x6000);
+	regmap_write(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, 0x0f26);
+	regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7f05);
+	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c);
+	regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d);
+	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_9, 0x000f);
+	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8d01);
+	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321);
+	regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004);
+	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00);
+	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1);
+	regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311);
+	regmap_write(rt5682->regmap, RT5682_RESET_HPF_CTRL, 0x0000);
+	regmap_write(rt5682->regmap, RT5682_ADC_STO1_HP_CTRL_1, 0x3320);
+
+	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00);
+
+	for (count = 0; count < 60; count++) {
+		regmap_read(rt5682->regmap, RT5682_HP_CALIB_STA_1, &value);
+		if (!(value & 0x8000))
+			break;
+
+		usleep_range(10000, 10005);
+	}
+
+	if (count >= 60)
+		pr_err("HP Calibration Failure\n");
+
+	/* restore settings */
+	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4);
+	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000);
+
+	mutex_unlock(&rt5682->calibrate_mutex);
+
+}
+
+static int rt5682_i2c_probe(struct i2c_client *i2c,
+		    const struct i2c_device_id *id)
+{
+	struct rt5682_platform_data *pdata = dev_get_platdata(&i2c->dev);
+	struct rt5682_priv *rt5682;
+	int i, ret;
+	unsigned int val;
+
+	rt5682 = devm_kzalloc(&i2c->dev, sizeof(struct rt5682_priv),
+		GFP_KERNEL);
+
+	if (rt5682 == NULL)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, rt5682);
+
+	if (pdata)
+		rt5682->pdata = *pdata;
+	else
+		rt5682_parse_dt(rt5682, &i2c->dev);
+
+	rt5682->regmap = devm_regmap_init_i2c(i2c, &rt5682_regmap);
+	if (IS_ERR(rt5682->regmap)) {
+		ret = PTR_ERR(rt5682->regmap);
+		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+			ret);
+		return ret;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(rt5682->supplies); i++)
+		rt5682->supplies[i].supply = rt5682_supply_names[i];
+
+	ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(rt5682->supplies),
+				      rt5682->supplies);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret);
+		return ret;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(rt5682->supplies),
+				    rt5682->supplies);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret);
+		return ret;
+	}
+
+	if (gpio_is_valid(rt5682->pdata.ldo1_en)) {
+		if (devm_gpio_request_one(&i2c->dev, rt5682->pdata.ldo1_en,
+					  GPIOF_OUT_INIT_HIGH, "rt5682"))
+			dev_err(&i2c->dev, "Fail gpio_request gpio_ldo\n");
+	}
+
+	/* Sleep for 300 ms miniumum */
+	usleep_range(300000, 350000);
+
+	regmap_write(rt5682->regmap, RT5682_I2C_MODE, 0x1);
+	usleep_range(10000, 15000);
+
+	regmap_read(rt5682->regmap, RT5682_DEVICE_ID, &val);
+	if (val != DEVICE_ID) {
+		pr_err("Device with ID register %x is not rt5682\n", val);
+		return -ENODEV;
+	}
+
+	rt5682_reset(rt5682->regmap);
+
+	rt5682_calibrate(rt5682);
+
+	ret = regmap_register_patch(rt5682->regmap, patch_list,
+				    ARRAY_SIZE(patch_list));
+	if (ret != 0)
+		dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret);
+
+	regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0000);
+
+	/* DMIC pin*/
+	if (rt5682->pdata.dmic1_data_pin != RT5682_DMIC1_NULL) {
+		switch (rt5682->pdata.dmic1_data_pin) {
+		case RT5682_DMIC1_DATA_GPIO2: /* share with LRCK2 */
+			regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1,
+				RT5682_DMIC_1_DP_MASK, RT5682_DMIC_1_DP_GPIO2);
+			regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+				RT5682_GP2_PIN_MASK, RT5682_GP2_PIN_DMIC_SDA);
+			break;
+
+		case RT5682_DMIC1_DATA_GPIO5: /* share with DACDAT1 */
+			regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1,
+				RT5682_DMIC_1_DP_MASK, RT5682_DMIC_1_DP_GPIO5);
+			regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+				RT5682_GP5_PIN_MASK, RT5682_GP5_PIN_DMIC_SDA);
+			break;
+
+		default:
+			dev_warn(&i2c->dev, "invalid DMIC_DAT pin\n");
+			break;
+		}
+
+		switch (rt5682->pdata.dmic1_clk_pin) {
+		case RT5682_DMIC1_CLK_GPIO1: /* share with IRQ */
+			regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+				RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_DMIC_CLK);
+			break;
+
+		case RT5682_DMIC1_CLK_GPIO3: /* share with BCLK2 */
+			regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+				RT5682_GP3_PIN_MASK, RT5682_GP3_PIN_DMIC_CLK);
+			break;
+
+		default:
+			dev_warn(&i2c->dev, "invalid DMIC_CLK pin\n");
+			break;
+		}
+	}
+
+	regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1,
+			RT5682_LDO1_DVO_MASK | RT5682_HP_DRIVER_MASK,
+			RT5682_LDO1_DVO_12 | RT5682_HP_DRIVER_5X);
+	regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380);
+	regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+			RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK,
+			RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1);
+	regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000);
+
+	INIT_DELAYED_WORK(&rt5682->jack_detect_work,
+				rt5682_jack_detect_handler);
+	INIT_DELAYED_WORK(&rt5682->jd_check_work,
+				rt5682_jd_check_handler);
+
+	mutex_init(&rt5682->calibrate_mutex);
+
+	if (i2c->irq) {
+		ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL,
+			rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
+			| IRQF_ONESHOT, "rt5682", rt5682);
+		if (ret)
+			dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret);
+
+	}
+
+	return devm_snd_soc_register_component(&i2c->dev,
+			&soc_component_dev_rt5682,
+			rt5682_dai, ARRAY_SIZE(rt5682_dai));
+}
+
+static void rt5682_i2c_shutdown(struct i2c_client *client)
+{
+	struct rt5682_priv *rt5682 = i2c_get_clientdata(client);
+
+	rt5682_reset(rt5682->regmap);
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id rt5682_of_match[] = {
+	{.compatible = "realtek,rt5682i"},
+	{},
+};
+MODULE_DEVICE_TABLE(of, rt5682_of_match);
+#endif
+
+#ifdef CONFIG_ACPI
+static const struct acpi_device_id rt5682_acpi_match[] = {
+	{"10EC5682", 0,},
+	{},
+};
+MODULE_DEVICE_TABLE(acpi, rt5682_acpi_match);
+#endif
+
+static struct i2c_driver rt5682_i2c_driver = {
+	.driver = {
+		.name = "rt5682",
+		.of_match_table = of_match_ptr(rt5682_of_match),
+		.acpi_match_table = ACPI_PTR(rt5682_acpi_match),
+	},
+	.probe = rt5682_i2c_probe,
+	.shutdown = rt5682_i2c_shutdown,
+	.id_table = rt5682_i2c_id,
+};
+module_i2c_driver(rt5682_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC RT5682 driver");
+MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h
new file mode 100644
index 000000000000..8068140ebe3f
--- /dev/null
+++ b/sound/soc/codecs/rt5682.h
@@ -0,0 +1,1324 @@
+/*
+ * rt5682.h  --  RT5682/RT5658 ALSA SoC audio driver
+ *
+ * Copyright 2018 Realtek Microelectronics
+ * Author: Bard Liao <bardliao@realtek.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 __RT5682_H__
+#define __RT5682_H__
+
+#include <sound/rt5682.h>
+
+#define DEVICE_ID 0x6530
+
+/* Info */
+#define RT5682_RESET				0x0000
+#define RT5682_VERSION_ID			0x00fd
+#define RT5682_VENDOR_ID			0x00fe
+#define RT5682_DEVICE_ID			0x00ff
+/*  I/O - Output */
+#define RT5682_HP_CTRL_1			0x0002
+#define RT5682_HP_CTRL_2			0x0003
+#define RT5682_HPL_GAIN				0x0005
+#define RT5682_HPR_GAIN				0x0006
+
+#define RT5682_I2C_CTRL				0x0008
+
+/* I/O - Input */
+#define RT5682_CBJ_BST_CTRL			0x000b
+#define RT5682_CBJ_CTRL_1			0x0010
+#define RT5682_CBJ_CTRL_2			0x0011
+#define RT5682_CBJ_CTRL_3			0x0012
+#define RT5682_CBJ_CTRL_4			0x0013
+#define RT5682_CBJ_CTRL_5			0x0014
+#define RT5682_CBJ_CTRL_6			0x0015
+#define RT5682_CBJ_CTRL_7			0x0016
+/* I/O - ADC/DAC/DMIC */
+#define RT5682_DAC1_DIG_VOL			0x0019
+#define RT5682_STO1_ADC_DIG_VOL			0x001c
+#define RT5682_STO1_ADC_BOOST			0x001f
+#define RT5682_HP_IMP_GAIN_1			0x0022
+#define RT5682_HP_IMP_GAIN_2			0x0023
+/* Mixer - D-D */
+#define RT5682_SIDETONE_CTRL			0x0024
+#define RT5682_STO1_ADC_MIXER			0x0026
+#define RT5682_AD_DA_MIXER			0x0029
+#define RT5682_STO1_DAC_MIXER			0x002a
+#define RT5682_A_DAC1_MUX			0x002b
+#define RT5682_DIG_INF2_DATA			0x0030
+/* Mixer - ADC */
+#define RT5682_REC_MIXER			0x003c
+#define RT5682_CAL_REC				0x0044
+#define RT5682_ALC_BACK_GAIN			0x0049
+/* Power */
+#define RT5682_PWR_DIG_1			0x0061
+#define RT5682_PWR_DIG_2			0x0062
+#define RT5682_PWR_ANLG_1			0x0063
+#define RT5682_PWR_ANLG_2			0x0064
+#define RT5682_PWR_ANLG_3			0x0065
+#define RT5682_PWR_MIXER			0x0066
+#define RT5682_PWR_VOL				0x0067
+/* Clock Detect */
+#define RT5682_CLK_DET				0x006b
+/* Filter Auto Reset */
+#define RT5682_RESET_LPF_CTRL			0x006c
+#define RT5682_RESET_HPF_CTRL			0x006d
+/* DMIC */
+#define RT5682_DMIC_CTRL_1			0x006e
+/* Format - ADC/DAC */
+#define RT5682_I2S1_SDP				0x0070
+#define RT5682_I2S2_SDP				0x0071
+#define RT5682_ADDA_CLK_1			0x0073
+#define RT5682_ADDA_CLK_2			0x0074
+#define RT5682_I2S1_F_DIV_CTRL_1		0x0075
+#define RT5682_I2S1_F_DIV_CTRL_2		0x0076
+/* Format - TDM Control */
+#define RT5682_TDM_CTRL				0x0079
+#define RT5682_TDM_ADDA_CTRL_1			0x007a
+#define RT5682_TDM_ADDA_CTRL_2			0x007b
+#define RT5682_DATA_SEL_CTRL_1			0x007c
+#define RT5682_TDM_TCON_CTRL			0x007e
+/* Function - Analog */
+#define RT5682_GLB_CLK				0x0080
+#define RT5682_PLL_CTRL_1			0x0081
+#define RT5682_PLL_CTRL_2			0x0082
+#define RT5682_PLL_TRACK_1			0x0083
+#define RT5682_PLL_TRACK_2			0x0084
+#define RT5682_PLL_TRACK_3			0x0085
+#define RT5682_PLL_TRACK_4			0x0086
+#define RT5682_PLL_TRACK_5			0x0087
+#define RT5682_PLL_TRACK_6			0x0088
+#define RT5682_PLL_TRACK_11			0x008c
+#define RT5682_SDW_REF_CLK			0x008d
+#define RT5682_DEPOP_1				0x008e
+#define RT5682_DEPOP_2				0x008f
+#define RT5682_HP_CHARGE_PUMP_1			0x0091
+#define RT5682_HP_CHARGE_PUMP_2			0x0092
+#define RT5682_MICBIAS_1			0x0093
+#define RT5682_MICBIAS_2			0x0094
+#define RT5682_PLL_TRACK_12			0x0098
+#define RT5682_PLL_TRACK_14			0x009a
+#define RT5682_PLL2_CTRL_1			0x009b
+#define RT5682_PLL2_CTRL_2			0x009c
+#define RT5682_PLL2_CTRL_3			0x009d
+#define RT5682_PLL2_CTRL_4			0x009e
+#define RT5682_RC_CLK_CTRL			0x009f
+#define RT5682_I2S_M_CLK_CTRL_1			0x00a0
+#define RT5682_I2S2_F_DIV_CTRL_1		0x00a3
+#define RT5682_I2S2_F_DIV_CTRL_2		0x00a4
+/* Function - Digital */
+#define RT5682_EQ_CTRL_1			0x00ae
+#define RT5682_EQ_CTRL_2			0x00af
+#define RT5682_IRQ_CTRL_1			0x00b6
+#define RT5682_IRQ_CTRL_2			0x00b7
+#define RT5682_IRQ_CTRL_3			0x00b8
+#define RT5682_IRQ_CTRL_4			0x00b9
+#define RT5682_INT_ST_1				0x00be
+#define RT5682_GPIO_CTRL_1			0x00c0
+#define RT5682_GPIO_CTRL_2			0x00c1
+#define RT5682_GPIO_CTRL_3			0x00c2
+#define RT5682_HP_AMP_DET_CTRL_1		0x00d0
+#define RT5682_HP_AMP_DET_CTRL_2		0x00d1
+#define RT5682_MID_HP_AMP_DET			0x00d2
+#define RT5682_LOW_HP_AMP_DET			0x00d3
+#define RT5682_DELAY_BUF_CTRL			0x00d4
+#define RT5682_SV_ZCD_1				0x00d9
+#define RT5682_SV_ZCD_2				0x00da
+#define RT5682_IL_CMD_1				0x00db
+#define RT5682_IL_CMD_2				0x00dc
+#define RT5682_IL_CMD_3				0x00dd
+#define RT5682_IL_CMD_4				0x00de
+#define RT5682_IL_CMD_5				0x00df
+#define RT5682_IL_CMD_6				0x00e0
+#define RT5682_4BTN_IL_CMD_1			0x00e2
+#define RT5682_4BTN_IL_CMD_2			0x00e3
+#define RT5682_4BTN_IL_CMD_3			0x00e4
+#define RT5682_4BTN_IL_CMD_4			0x00e5
+#define RT5682_4BTN_IL_CMD_5			0x00e6
+#define RT5682_4BTN_IL_CMD_6			0x00e7
+#define RT5682_4BTN_IL_CMD_7			0x00e8
+
+#define RT5682_ADC_STO1_HP_CTRL_1		0x00ea
+#define RT5682_ADC_STO1_HP_CTRL_2		0x00eb
+#define RT5682_AJD1_CTRL			0x00f0
+#define RT5682_JD1_THD				0x00f1
+#define RT5682_JD2_THD				0x00f2
+#define RT5682_JD_CTRL_1			0x00f6
+/* General Control */
+#define RT5682_DUMMY_1				0x00fa
+#define RT5682_DUMMY_2				0x00fb
+#define RT5682_DUMMY_3				0x00fc
+
+#define RT5682_DAC_ADC_DIG_VOL1			0x0100
+#define RT5682_BIAS_CUR_CTRL_2			0x010b
+#define RT5682_BIAS_CUR_CTRL_3			0x010c
+#define RT5682_BIAS_CUR_CTRL_4			0x010d
+#define RT5682_BIAS_CUR_CTRL_5			0x010e
+#define RT5682_BIAS_CUR_CTRL_6			0x010f
+#define RT5682_BIAS_CUR_CTRL_7			0x0110
+#define RT5682_BIAS_CUR_CTRL_8			0x0111
+#define RT5682_BIAS_CUR_CTRL_9			0x0112
+#define RT5682_BIAS_CUR_CTRL_10			0x0113
+#define RT5682_VREF_REC_OP_FB_CAP_CTRL		0x0117
+#define RT5682_CHARGE_PUMP_1			0x0125
+#define RT5682_DIG_IN_CTRL_1			0x0132
+#define RT5682_PAD_DRIVING_CTRL			0x0136
+#define RT5682_SOFT_RAMP_DEPOP			0x0138
+#define RT5682_CHOP_DAC				0x013a
+#define RT5682_CHOP_ADC				0x013b
+#define RT5682_CALIB_ADC_CTRL			0x013c
+#define RT5682_VOL_TEST				0x013f
+#define RT5682_SPKVDD_DET_STA			0x0142
+#define RT5682_TEST_MODE_CTRL_1			0x0145
+#define RT5682_TEST_MODE_CTRL_2			0x0146
+#define RT5682_TEST_MODE_CTRL_3			0x0147
+#define RT5682_TEST_MODE_CTRL_4			0x0148
+#define RT5682_TEST_MODE_CTRL_5			0x0149
+#define RT5682_PLL1_INTERNAL			0x0150
+#define RT5682_PLL2_INTERNAL			0x0151
+#define RT5682_STO_NG2_CTRL_1			0x0160
+#define RT5682_STO_NG2_CTRL_2			0x0161
+#define RT5682_STO_NG2_CTRL_3			0x0162
+#define RT5682_STO_NG2_CTRL_4			0x0163
+#define RT5682_STO_NG2_CTRL_5			0x0164
+#define RT5682_STO_NG2_CTRL_6			0x0165
+#define RT5682_STO_NG2_CTRL_7			0x0166
+#define RT5682_STO_NG2_CTRL_8			0x0167
+#define RT5682_STO_NG2_CTRL_9			0x0168
+#define RT5682_STO_NG2_CTRL_10			0x0169
+#define RT5682_STO1_DAC_SIL_DET			0x0190
+#define RT5682_SIL_PSV_CTRL1			0x0194
+#define RT5682_SIL_PSV_CTRL2			0x0195
+#define RT5682_SIL_PSV_CTRL3			0x0197
+#define RT5682_SIL_PSV_CTRL4			0x0198
+#define RT5682_SIL_PSV_CTRL5			0x0199
+#define RT5682_HP_IMP_SENS_CTRL_01		0x01af
+#define RT5682_HP_IMP_SENS_CTRL_02		0x01b0
+#define RT5682_HP_IMP_SENS_CTRL_03		0x01b1
+#define RT5682_HP_IMP_SENS_CTRL_04		0x01b2
+#define RT5682_HP_IMP_SENS_CTRL_05		0x01b3
+#define RT5682_HP_IMP_SENS_CTRL_06		0x01b4
+#define RT5682_HP_IMP_SENS_CTRL_07		0x01b5
+#define RT5682_HP_IMP_SENS_CTRL_08		0x01b6
+#define RT5682_HP_IMP_SENS_CTRL_09		0x01b7
+#define RT5682_HP_IMP_SENS_CTRL_10		0x01b8
+#define RT5682_HP_IMP_SENS_CTRL_11		0x01b9
+#define RT5682_HP_IMP_SENS_CTRL_12		0x01ba
+#define RT5682_HP_IMP_SENS_CTRL_13		0x01bb
+#define RT5682_HP_IMP_SENS_CTRL_14		0x01bc
+#define RT5682_HP_IMP_SENS_CTRL_15		0x01bd
+#define RT5682_HP_IMP_SENS_CTRL_16		0x01be
+#define RT5682_HP_IMP_SENS_CTRL_17		0x01bf
+#define RT5682_HP_IMP_SENS_CTRL_18		0x01c0
+#define RT5682_HP_IMP_SENS_CTRL_19		0x01c1
+#define RT5682_HP_IMP_SENS_CTRL_20		0x01c2
+#define RT5682_HP_IMP_SENS_CTRL_21		0x01c3
+#define RT5682_HP_IMP_SENS_CTRL_22		0x01c4
+#define RT5682_HP_IMP_SENS_CTRL_23		0x01c5
+#define RT5682_HP_IMP_SENS_CTRL_24		0x01c6
+#define RT5682_HP_IMP_SENS_CTRL_25		0x01c7
+#define RT5682_HP_IMP_SENS_CTRL_26		0x01c8
+#define RT5682_HP_IMP_SENS_CTRL_27		0x01c9
+#define RT5682_HP_IMP_SENS_CTRL_28		0x01ca
+#define RT5682_HP_IMP_SENS_CTRL_29		0x01cb
+#define RT5682_HP_IMP_SENS_CTRL_30		0x01cc
+#define RT5682_HP_IMP_SENS_CTRL_31		0x01cd
+#define RT5682_HP_IMP_SENS_CTRL_32		0x01ce
+#define RT5682_HP_IMP_SENS_CTRL_33		0x01cf
+#define RT5682_HP_IMP_SENS_CTRL_34		0x01d0
+#define RT5682_HP_IMP_SENS_CTRL_35		0x01d1
+#define RT5682_HP_IMP_SENS_CTRL_36		0x01d2
+#define RT5682_HP_IMP_SENS_CTRL_37		0x01d3
+#define RT5682_HP_IMP_SENS_CTRL_38		0x01d4
+#define RT5682_HP_IMP_SENS_CTRL_39		0x01d5
+#define RT5682_HP_IMP_SENS_CTRL_40		0x01d6
+#define RT5682_HP_IMP_SENS_CTRL_41		0x01d7
+#define RT5682_HP_IMP_SENS_CTRL_42		0x01d8
+#define RT5682_HP_IMP_SENS_CTRL_43		0x01d9
+#define RT5682_HP_LOGIC_CTRL_1			0x01da
+#define RT5682_HP_LOGIC_CTRL_2			0x01db
+#define RT5682_HP_LOGIC_CTRL_3			0x01dc
+#define RT5682_HP_CALIB_CTRL_1			0x01de
+#define RT5682_HP_CALIB_CTRL_2			0x01df
+#define RT5682_HP_CALIB_CTRL_3			0x01e0
+#define RT5682_HP_CALIB_CTRL_4			0x01e1
+#define RT5682_HP_CALIB_CTRL_5			0x01e2
+#define RT5682_HP_CALIB_CTRL_6			0x01e3
+#define RT5682_HP_CALIB_CTRL_7			0x01e4
+#define RT5682_HP_CALIB_CTRL_9			0x01e6
+#define RT5682_HP_CALIB_CTRL_10			0x01e7
+#define RT5682_HP_CALIB_CTRL_11			0x01e8
+#define RT5682_HP_CALIB_STA_1			0x01ea
+#define RT5682_HP_CALIB_STA_2			0x01eb
+#define RT5682_HP_CALIB_STA_3			0x01ec
+#define RT5682_HP_CALIB_STA_4			0x01ed
+#define RT5682_HP_CALIB_STA_5			0x01ee
+#define RT5682_HP_CALIB_STA_6			0x01ef
+#define RT5682_HP_CALIB_STA_7			0x01f0
+#define RT5682_HP_CALIB_STA_8			0x01f1
+#define RT5682_HP_CALIB_STA_9			0x01f2
+#define RT5682_HP_CALIB_STA_10			0x01f3
+#define RT5682_HP_CALIB_STA_11			0x01f4
+#define RT5682_SAR_IL_CMD_1			0x0210
+#define RT5682_SAR_IL_CMD_2			0x0211
+#define RT5682_SAR_IL_CMD_3			0x0212
+#define RT5682_SAR_IL_CMD_4			0x0213
+#define RT5682_SAR_IL_CMD_5			0x0214
+#define RT5682_SAR_IL_CMD_6			0x0215
+#define RT5682_SAR_IL_CMD_7			0x0216
+#define RT5682_SAR_IL_CMD_8			0x0217
+#define RT5682_SAR_IL_CMD_9			0x0218
+#define RT5682_SAR_IL_CMD_10			0x0219
+#define RT5682_SAR_IL_CMD_11			0x021a
+#define RT5682_SAR_IL_CMD_12			0x021b
+#define RT5682_SAR_IL_CMD_13			0x021c
+#define RT5682_EFUSE_CTRL_1			0x0250
+#define RT5682_EFUSE_CTRL_2			0x0251
+#define RT5682_EFUSE_CTRL_3			0x0252
+#define RT5682_EFUSE_CTRL_4			0x0253
+#define RT5682_EFUSE_CTRL_5			0x0254
+#define RT5682_EFUSE_CTRL_6			0x0255
+#define RT5682_EFUSE_CTRL_7			0x0256
+#define RT5682_EFUSE_CTRL_8			0x0257
+#define RT5682_EFUSE_CTRL_9			0x0258
+#define RT5682_EFUSE_CTRL_10			0x0259
+#define RT5682_EFUSE_CTRL_11			0x025a
+#define RT5682_JD_TOP_VC_VTRL			0x0270
+#define RT5682_DRC1_CTRL_0			0x02ff
+#define RT5682_DRC1_CTRL_1			0x0300
+#define RT5682_DRC1_CTRL_2			0x0301
+#define RT5682_DRC1_CTRL_3			0x0302
+#define RT5682_DRC1_CTRL_4			0x0303
+#define RT5682_DRC1_CTRL_5			0x0304
+#define RT5682_DRC1_CTRL_6			0x0305
+#define RT5682_DRC1_HARD_LMT_CTRL_1		0x0306
+#define RT5682_DRC1_HARD_LMT_CTRL_2		0x0307
+#define RT5682_DRC1_PRIV_1			0x0310
+#define RT5682_DRC1_PRIV_2			0x0311
+#define RT5682_DRC1_PRIV_3			0x0312
+#define RT5682_DRC1_PRIV_4			0x0313
+#define RT5682_DRC1_PRIV_5			0x0314
+#define RT5682_DRC1_PRIV_6			0x0315
+#define RT5682_DRC1_PRIV_7			0x0316
+#define RT5682_DRC1_PRIV_8			0x0317
+#define RT5682_EQ_AUTO_RCV_CTRL1		0x03c0
+#define RT5682_EQ_AUTO_RCV_CTRL2		0x03c1
+#define RT5682_EQ_AUTO_RCV_CTRL3		0x03c2
+#define RT5682_EQ_AUTO_RCV_CTRL4		0x03c3
+#define RT5682_EQ_AUTO_RCV_CTRL5		0x03c4
+#define RT5682_EQ_AUTO_RCV_CTRL6		0x03c5
+#define RT5682_EQ_AUTO_RCV_CTRL7		0x03c6
+#define RT5682_EQ_AUTO_RCV_CTRL8		0x03c7
+#define RT5682_EQ_AUTO_RCV_CTRL9		0x03c8
+#define RT5682_EQ_AUTO_RCV_CTRL10		0x03c9
+#define RT5682_EQ_AUTO_RCV_CTRL11		0x03ca
+#define RT5682_EQ_AUTO_RCV_CTRL12		0x03cb
+#define RT5682_EQ_AUTO_RCV_CTRL13		0x03cc
+#define RT5682_ADC_L_EQ_LPF1_A1			0x03d0
+#define RT5682_R_EQ_LPF1_A1			0x03d1
+#define RT5682_L_EQ_LPF1_H0			0x03d2
+#define RT5682_R_EQ_LPF1_H0			0x03d3
+#define RT5682_L_EQ_BPF1_A1			0x03d4
+#define RT5682_R_EQ_BPF1_A1			0x03d5
+#define RT5682_L_EQ_BPF1_A2			0x03d6
+#define RT5682_R_EQ_BPF1_A2			0x03d7
+#define RT5682_L_EQ_BPF1_H0			0x03d8
+#define RT5682_R_EQ_BPF1_H0			0x03d9
+#define RT5682_L_EQ_BPF2_A1			0x03da
+#define RT5682_R_EQ_BPF2_A1			0x03db
+#define RT5682_L_EQ_BPF2_A2			0x03dc
+#define RT5682_R_EQ_BPF2_A2			0x03dd
+#define RT5682_L_EQ_BPF2_H0			0x03de
+#define RT5682_R_EQ_BPF2_H0			0x03df
+#define RT5682_L_EQ_BPF3_A1			0x03e0
+#define RT5682_R_EQ_BPF3_A1			0x03e1
+#define RT5682_L_EQ_BPF3_A2			0x03e2
+#define RT5682_R_EQ_BPF3_A2			0x03e3
+#define RT5682_L_EQ_BPF3_H0			0x03e4
+#define RT5682_R_EQ_BPF3_H0			0x03e5
+#define RT5682_L_EQ_BPF4_A1			0x03e6
+#define RT5682_R_EQ_BPF4_A1			0x03e7
+#define RT5682_L_EQ_BPF4_A2			0x03e8
+#define RT5682_R_EQ_BPF4_A2			0x03e9
+#define RT5682_L_EQ_BPF4_H0			0x03ea
+#define RT5682_R_EQ_BPF4_H0			0x03eb
+#define RT5682_L_EQ_HPF1_A1			0x03ec
+#define RT5682_R_EQ_HPF1_A1			0x03ed
+#define RT5682_L_EQ_HPF1_H0			0x03ee
+#define RT5682_R_EQ_HPF1_H0			0x03ef
+#define RT5682_L_EQ_PRE_VOL			0x03f0
+#define RT5682_R_EQ_PRE_VOL			0x03f1
+#define RT5682_L_EQ_POST_VOL			0x03f2
+#define RT5682_R_EQ_POST_VOL			0x03f3
+#define RT5682_I2C_MODE				0xffff
+
+
+/* global definition */
+#define RT5682_L_MUTE				(0x1 << 15)
+#define RT5682_L_MUTE_SFT			15
+#define RT5682_VOL_L_MUTE			(0x1 << 14)
+#define RT5682_VOL_L_SFT			14
+#define RT5682_R_MUTE				(0x1 << 7)
+#define RT5682_R_MUTE_SFT			7
+#define RT5682_VOL_R_MUTE			(0x1 << 6)
+#define RT5682_VOL_R_SFT			6
+#define RT5682_L_VOL_MASK			(0x3f << 8)
+#define RT5682_L_VOL_SFT			8
+#define RT5682_R_VOL_MASK			(0x3f)
+#define RT5682_R_VOL_SFT			0
+
+/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/
+#define RT5682_G_HP				(0xf << 8)
+#define RT5682_G_HP_SFT				8
+#define RT5682_G_STO_DA_DMIX			(0xf)
+#define RT5682_G_STO_DA_SFT			0
+
+/* CBJ Control (0x000b) */
+#define RT5682_BST_CBJ_MASK			(0xf << 8)
+#define RT5682_BST_CBJ_SFT			8
+
+/* Embeeded Jack and Type Detection Control 1 (0x0010) */
+#define RT5682_EMB_JD_EN			(0x1 << 15)
+#define RT5682_EMB_JD_EN_SFT			15
+#define RT5682_EMB_JD_RST			(0x1 << 14)
+#define RT5682_JD_MODE				(0x1 << 13)
+#define RT5682_JD_MODE_SFT			13
+#define RT5682_DET_TYPE				(0x1 << 12)
+#define RT5682_DET_TYPE_SFT			12
+#define RT5682_POLA_EXT_JD_MASK			(0x1 << 11)
+#define RT5682_POLA_EXT_JD_LOW			(0x1 << 11)
+#define RT5682_POLA_EXT_JD_HIGH			(0x0 << 11)
+#define RT5682_EXT_JD_DIG			(0x1 << 9)
+#define RT5682_POL_FAST_OFF_MASK		(0x1 << 8)
+#define RT5682_POL_FAST_OFF_HIGH		(0x1 << 8)
+#define RT5682_POL_FAST_OFF_LOW			(0x0 << 8)
+#define RT5682_FAST_OFF_MASK			(0x1 << 7)
+#define RT5682_FAST_OFF_EN			(0x1 << 7)
+#define RT5682_FAST_OFF_DIS			(0x0 << 7)
+#define RT5682_VREF_POW_MASK			(0x1 << 6)
+#define RT5682_VREF_POW_FSM			(0x0 << 6)
+#define RT5682_VREF_POW_REG			(0x1 << 6)
+#define RT5682_MB1_PATH_MASK			(0x1 << 5)
+#define RT5682_CTRL_MB1_REG			(0x1 << 5)
+#define RT5682_CTRL_MB1_FSM			(0x0 << 5)
+#define RT5682_MB2_PATH_MASK			(0x1 << 4)
+#define RT5682_CTRL_MB2_REG			(0x1 << 4)
+#define RT5682_CTRL_MB2_FSM			(0x0 << 4)
+#define RT5682_TRIG_JD_MASK			(0x1 << 3)
+#define RT5682_TRIG_JD_HIGH			(0x1 << 3)
+#define RT5682_TRIG_JD_LOW			(0x0 << 3)
+#define RT5682_MIC_CAP_MASK			(0x1 << 1)
+#define RT5682_MIC_CAP_HS			(0x1 << 1)
+#define RT5682_MIC_CAP_HP			(0x0 << 1)
+#define RT5682_MIC_CAP_SRC_MASK			(0x1)
+#define RT5682_MIC_CAP_SRC_REG			(0x1)
+#define RT5682_MIC_CAP_SRC_ANA			(0x0)
+
+/* Embeeded Jack and Type Detection Control 2 (0x0011) */
+#define RT5682_EXT_JD_SRC			(0x7 << 4)
+#define RT5682_EXT_JD_SRC_SFT			4
+#define RT5682_EXT_JD_SRC_GPIO_JD1		(0x0 << 4)
+#define RT5682_EXT_JD_SRC_GPIO_JD2		(0x1 << 4)
+#define RT5682_EXT_JD_SRC_JDH			(0x2 << 4)
+#define RT5682_EXT_JD_SRC_JDL			(0x3 << 4)
+#define RT5682_EXT_JD_SRC_MANUAL		(0x4 << 4)
+#define RT5682_JACK_TYPE_MASK			(0x3)
+
+/* Combo Jack and Type Detection Control 3 (0x0012) */
+#define RT5682_CBJ_IN_BUF_EN			(0x1 << 7)
+
+/* Combo Jack and Type Detection Control 4 (0x0013) */
+#define RT5682_SEL_SHT_MID_TON_MASK		(0x3 << 12)
+#define RT5682_SEL_SHT_MID_TON_2		(0x0 << 12)
+#define RT5682_SEL_SHT_MID_TON_3		(0x1 << 12)
+#define RT5682_CBJ_JD_TEST_MASK			(0x1 << 6)
+#define RT5682_CBJ_JD_TEST_NORM			(0x0 << 6)
+#define RT5682_CBJ_JD_TEST_MODE			(0x1 << 6)
+
+/* DAC1 Digital Volume (0x0019) */
+#define RT5682_DAC_L1_VOL_MASK			(0xff << 8)
+#define RT5682_DAC_L1_VOL_SFT			8
+#define RT5682_DAC_R1_VOL_MASK			(0xff)
+#define RT5682_DAC_R1_VOL_SFT			0
+
+/* ADC Digital Volume Control (0x001c) */
+#define RT5682_ADC_L_VOL_MASK			(0x7f << 8)
+#define RT5682_ADC_L_VOL_SFT			8
+#define RT5682_ADC_R_VOL_MASK			(0x7f)
+#define RT5682_ADC_R_VOL_SFT			0
+
+/* Stereo1 ADC Boost Gain Control (0x001f) */
+#define RT5682_STO1_ADC_L_BST_MASK		(0x3 << 14)
+#define RT5682_STO1_ADC_L_BST_SFT		14
+#define RT5682_STO1_ADC_R_BST_MASK		(0x3 << 12)
+#define RT5682_STO1_ADC_R_BST_SFT		12
+
+/* Sidetone Control (0x0024) */
+#define RT5682_ST_SRC_SEL			(0x1 << 8)
+#define RT5682_ST_SRC_SFT			8
+#define RT5682_ST_EN_MASK			(0x1 << 6)
+#define RT5682_ST_DIS				(0x0 << 6)
+#define RT5682_ST_EN				(0x1 << 6)
+#define RT5682_ST_EN_SFT			6
+
+/* Stereo1 ADC Mixer Control (0x0026) */
+#define RT5682_M_STO1_ADC_L1			(0x1 << 15)
+#define RT5682_M_STO1_ADC_L1_SFT		15
+#define RT5682_M_STO1_ADC_L2			(0x1 << 14)
+#define RT5682_M_STO1_ADC_L2_SFT		14
+#define RT5682_STO1_ADC1L_SRC_MASK		(0x1 << 13)
+#define RT5682_STO1_ADC1L_SRC_SFT		13
+#define RT5682_STO1_ADC1_SRC_ADC		(0x1 << 13)
+#define RT5682_STO1_ADC1_SRC_DACMIX		(0x0 << 13)
+#define RT5682_STO1_ADC2L_SRC_MASK		(0x1 << 12)
+#define RT5682_STO1_ADC2L_SRC_SFT		12
+#define RT5682_STO1_ADCL_SRC_MASK		(0x3 << 10)
+#define RT5682_STO1_ADCL_SRC_SFT		10
+#define RT5682_STO1_DD_L_SRC_MASK		(0x1 << 9)
+#define RT5682_STO1_DD_L_SRC_SFT		9
+#define RT5682_STO1_DMIC_SRC_MASK		(0x1 << 8)
+#define RT5682_STO1_DMIC_SRC_SFT		8
+#define RT5682_STO1_DMIC_SRC_DMIC2		(0x1 << 8)
+#define RT5682_STO1_DMIC_SRC_DMIC1		(0x0 << 8)
+#define RT5682_M_STO1_ADC_R1			(0x1 << 7)
+#define RT5682_M_STO1_ADC_R1_SFT		7
+#define RT5682_M_STO1_ADC_R2			(0x1 << 6)
+#define RT5682_M_STO1_ADC_R2_SFT		6
+#define RT5682_STO1_ADC1R_SRC_MASK		(0x1 << 5)
+#define RT5682_STO1_ADC1R_SRC_SFT		5
+#define RT5682_STO1_ADC2R_SRC_MASK		(0x1 << 4)
+#define RT5682_STO1_ADC2R_SRC_SFT		4
+#define RT5682_STO1_ADCR_SRC_MASK		(0x3 << 2)
+#define RT5682_STO1_ADCR_SRC_SFT		2
+
+/* ADC Mixer to DAC Mixer Control (0x0029) */
+#define RT5682_M_ADCMIX_L			(0x1 << 15)
+#define RT5682_M_ADCMIX_L_SFT			15
+#define RT5682_M_DAC1_L				(0x1 << 14)
+#define RT5682_M_DAC1_L_SFT			14
+#define RT5682_DAC1_R_SEL_MASK			(0x1 << 10)
+#define RT5682_DAC1_R_SEL_SFT			10
+#define RT5682_DAC1_L_SEL_MASK			(0x1 << 8)
+#define RT5682_DAC1_L_SEL_SFT			8
+#define RT5682_M_ADCMIX_R			(0x1 << 7)
+#define RT5682_M_ADCMIX_R_SFT			7
+#define RT5682_M_DAC1_R				(0x1 << 6)
+#define RT5682_M_DAC1_R_SFT			6
+
+/* Stereo1 DAC Mixer Control (0x002a) */
+#define RT5682_M_DAC_L1_STO_L			(0x1 << 15)
+#define RT5682_M_DAC_L1_STO_L_SFT		15
+#define RT5682_G_DAC_L1_STO_L_MASK		(0x1 << 14)
+#define RT5682_G_DAC_L1_STO_L_SFT		14
+#define RT5682_M_DAC_R1_STO_L			(0x1 << 13)
+#define RT5682_M_DAC_R1_STO_L_SFT		13
+#define RT5682_G_DAC_R1_STO_L_MASK		(0x1 << 12)
+#define RT5682_G_DAC_R1_STO_L_SFT		12
+#define RT5682_M_DAC_L1_STO_R			(0x1 << 7)
+#define RT5682_M_DAC_L1_STO_R_SFT		7
+#define RT5682_G_DAC_L1_STO_R_MASK		(0x1 << 6)
+#define RT5682_G_DAC_L1_STO_R_SFT		6
+#define RT5682_M_DAC_R1_STO_R			(0x1 << 5)
+#define RT5682_M_DAC_R1_STO_R_SFT		5
+#define RT5682_G_DAC_R1_STO_R_MASK		(0x1 << 4)
+#define RT5682_G_DAC_R1_STO_R_SFT		4
+
+/* Analog DAC1 Input Source Control (0x002b) */
+#define RT5682_M_ST_STO_L			(0x1 << 9)
+#define RT5682_M_ST_STO_L_SFT			9
+#define RT5682_M_ST_STO_R			(0x1 << 8)
+#define RT5682_M_ST_STO_R_SFT			8
+#define RT5682_DAC_L1_SRC_MASK			(0x3 << 4)
+#define RT5682_A_DACL1_SFT			4
+#define RT5682_DAC_R1_SRC_MASK			(0x3)
+#define RT5682_A_DACR1_SFT			0
+
+/* Digital Interface Data Control (0x0030) */
+#define RT5682_IF2_ADC_SEL_MASK			(0x3 << 0)
+#define RT5682_IF2_ADC_SEL_SFT			0
+
+/* REC Left Mixer Control 2 (0x003c) */
+#define RT5682_G_CBJ_RM1_L			(0x7 << 10)
+#define RT5682_G_CBJ_RM1_L_SFT			10
+#define RT5682_M_CBJ_RM1_L			(0x1 << 7)
+#define RT5682_M_CBJ_RM1_L_SFT			7
+
+/* Power Management for Digital 1 (0x0061) */
+#define RT5682_PWR_I2S1				(0x1 << 15)
+#define RT5682_PWR_I2S1_BIT			15
+#define RT5682_PWR_I2S2				(0x1 << 14)
+#define RT5682_PWR_I2S2_BIT			14
+#define RT5682_PWR_DAC_L1			(0x1 << 11)
+#define RT5682_PWR_DAC_L1_BIT			11
+#define RT5682_PWR_DAC_R1			(0x1 << 10)
+#define RT5682_PWR_DAC_R1_BIT			10
+#define RT5682_PWR_LDO				(0x1 << 8)
+#define RT5682_PWR_LDO_BIT			8
+#define RT5682_PWR_ADC_L1			(0x1 << 4)
+#define RT5682_PWR_ADC_L1_BIT			4
+#define RT5682_PWR_ADC_R1			(0x1 << 3)
+#define RT5682_PWR_ADC_R1_BIT			3
+#define RT5682_DIG_GATE_CTRL			(0x1 << 0)
+#define RT5682_DIG_GATE_CTRL_SFT		0
+
+
+/* Power Management for Digital 2 (0x0062) */
+#define RT5682_PWR_ADC_S1F			(0x1 << 15)
+#define RT5682_PWR_ADC_S1F_BIT			15
+#define RT5682_PWR_DAC_S1F			(0x1 << 10)
+#define RT5682_PWR_DAC_S1F_BIT			10
+
+/* Power Management for Analog 1 (0x0063) */
+#define RT5682_PWR_VREF1			(0x1 << 15)
+#define RT5682_PWR_VREF1_BIT			15
+#define RT5682_PWR_FV1				(0x1 << 14)
+#define RT5682_PWR_FV1_BIT			14
+#define RT5682_PWR_VREF2			(0x1 << 13)
+#define RT5682_PWR_VREF2_BIT			13
+#define RT5682_PWR_FV2				(0x1 << 12)
+#define RT5682_PWR_FV2_BIT			12
+#define RT5682_LDO1_DBG_MASK			(0x3 << 10)
+#define RT5682_PWR_MB				(0x1 << 9)
+#define RT5682_PWR_MB_BIT			9
+#define RT5682_PWR_BG				(0x1 << 7)
+#define RT5682_PWR_BG_BIT			7
+#define RT5682_LDO1_BYPASS_MASK			(0x1 << 6)
+#define RT5682_LDO1_BYPASS			(0x1 << 6)
+#define RT5682_LDO1_NOT_BYPASS			(0x0 << 6)
+#define RT5682_PWR_MA_BIT			6
+#define RT5682_LDO1_DVO_MASK			(0x3 << 4)
+#define RT5682_LDO1_DVO_09			(0x0 << 4)
+#define RT5682_LDO1_DVO_10			(0x1 << 4)
+#define RT5682_LDO1_DVO_12			(0x2 << 4)
+#define RT5682_LDO1_DVO_14			(0x3 << 4)
+#define RT5682_HP_DRIVER_MASK			(0x3 << 2)
+#define RT5682_HP_DRIVER_1X			(0x0 << 2)
+#define RT5682_HP_DRIVER_3X			(0x1 << 2)
+#define RT5682_HP_DRIVER_5X			(0x3 << 2)
+#define RT5682_PWR_HA_L				(0x1 << 1)
+#define RT5682_PWR_HA_L_BIT			1
+#define RT5682_PWR_HA_R				(0x1 << 0)
+#define RT5682_PWR_HA_R_BIT			0
+
+/* Power Management for Analog 2 (0x0064) */
+#define RT5682_PWR_MB1				(0x1 << 11)
+#define RT5682_PWR_MB1_PWR_DOWN			(0x0 << 11)
+#define RT5682_PWR_MB1_BIT			11
+#define RT5682_PWR_MB2				(0x1 << 10)
+#define RT5682_PWR_MB2_PWR_DOWN			(0x0 << 10)
+#define RT5682_PWR_MB2_BIT			10
+#define RT5682_PWR_JDH				(0x1 << 3)
+#define RT5682_PWR_JDH_BIT			3
+#define RT5682_PWR_JDL				(0x1 << 2)
+#define RT5682_PWR_JDL_BIT			2
+#define RT5682_PWR_RM1_L			(0x1 << 1)
+#define RT5682_PWR_RM1_L_BIT			1
+
+/* Power Management for Analog 3 (0x0065) */
+#define RT5682_PWR_CBJ				(0x1 << 9)
+#define RT5682_PWR_CBJ_BIT			9
+#define RT5682_PWR_PLL				(0x1 << 6)
+#define RT5682_PWR_PLL_BIT			6
+#define RT5682_PWR_PLL2B			(0x1 << 5)
+#define RT5682_PWR_PLL2B_BIT			5
+#define RT5682_PWR_PLL2F			(0x1 << 4)
+#define RT5682_PWR_PLL2F_BIT			4
+#define RT5682_PWR_LDO2				(0x1 << 2)
+#define RT5682_PWR_LDO2_BIT			2
+#define RT5682_PWR_DET_SPKVDD			(0x1 << 1)
+#define RT5682_PWR_DET_SPKVDD_BIT		1
+
+/* Power Management for Mixer (0x0066) */
+#define RT5682_PWR_STO1_DAC_L			(0x1 << 5)
+#define RT5682_PWR_STO1_DAC_L_BIT		5
+#define RT5682_PWR_STO1_DAC_R			(0x1 << 4)
+#define RT5682_PWR_STO1_DAC_R_BIT		4
+
+/* MCLK and System Clock Detection Control (0x006b) */
+#define RT5682_SYS_CLK_DET			(0x1 << 15)
+#define RT5682_SYS_CLK_DET_SFT			15
+#define RT5682_PLL1_CLK_DET			(0x1 << 14)
+#define RT5682_PLL1_CLK_DET_SFT			14
+#define RT5682_PLL2_CLK_DET			(0x1 << 13)
+#define RT5682_PLL2_CLK_DET_SFT			13
+#define RT5682_POW_CLK_DET2_SFT			8
+#define RT5682_POW_CLK_DET_SFT			0
+
+/* Digital Microphone Control 1 (0x006e) */
+#define RT5682_DMIC_1_EN_MASK			(0x1 << 15)
+#define RT5682_DMIC_1_EN_SFT			15
+#define RT5682_DMIC_1_DIS			(0x0 << 15)
+#define RT5682_DMIC_1_EN			(0x1 << 15)
+#define RT5682_DMIC_1_DP_MASK			(0x3 << 4)
+#define RT5682_DMIC_1_DP_SFT			4
+#define RT5682_DMIC_1_DP_GPIO2			(0x0 << 4)
+#define RT5682_DMIC_1_DP_GPIO5			(0x1 << 4)
+#define RT5682_DMIC_CLK_MASK			(0xf << 0)
+#define RT5682_DMIC_CLK_SFT			0
+
+/* I2S1 Audio Serial Data Port Control (0x0070) */
+#define RT5682_SEL_ADCDAT_MASK			(0x1 << 15)
+#define RT5682_SEL_ADCDAT_OUT			(0x0 << 15)
+#define RT5682_SEL_ADCDAT_IN			(0x1 << 15)
+#define RT5682_SEL_ADCDAT_SFT			15
+#define RT5682_I2S1_TX_CHL_MASK			(0x7 << 12)
+#define RT5682_I2S1_TX_CHL_SFT			12
+#define RT5682_I2S1_TX_CHL_16			(0x0 << 12)
+#define RT5682_I2S1_TX_CHL_20			(0x1 << 12)
+#define RT5682_I2S1_TX_CHL_24			(0x2 << 12)
+#define RT5682_I2S1_TX_CHL_32			(0x3 << 12)
+#define RT5682_I2S1_TX_CHL_8			(0x4 << 12)
+#define RT5682_I2S1_RX_CHL_MASK			(0x7 << 8)
+#define RT5682_I2S1_RX_CHL_SFT			8
+#define RT5682_I2S1_RX_CHL_16			(0x0 << 8)
+#define RT5682_I2S1_RX_CHL_20			(0x1 << 8)
+#define RT5682_I2S1_RX_CHL_24			(0x2 << 8)
+#define RT5682_I2S1_RX_CHL_32			(0x3 << 8)
+#define RT5682_I2S1_RX_CHL_8			(0x4 << 8)
+#define RT5682_I2S1_MONO_MASK			(0x1 << 7)
+#define RT5682_I2S1_MONO_EN			(0x1 << 7)
+#define RT5682_I2S1_MONO_DIS			(0x0 << 7)
+#define RT5682_I2S2_MONO_MASK			(0x1 << 6)
+#define RT5682_I2S2_MONO_EN			(0x1 << 6)
+#define RT5682_I2S2_MONO_DIS			(0x0 << 6)
+#define RT5682_I2S1_DL_MASK			(0x7 << 4)
+#define RT5682_I2S1_DL_SFT			4
+#define RT5682_I2S1_DL_16			(0x0 << 4)
+#define RT5682_I2S1_DL_20			(0x1 << 4)
+#define RT5682_I2S1_DL_24			(0x2 << 4)
+#define RT5682_I2S1_DL_32			(0x3 << 4)
+#define RT5682_I2S1_DL_8			(0x4 << 4)
+
+/* I2S1/2 Audio Serial Data Port Control (0x0070)(0x0071) */
+#define RT5682_I2S2_MS_MASK			(0x1 << 15)
+#define RT5682_I2S2_MS_SFT			15
+#define RT5682_I2S2_MS_M			(0x0 << 15)
+#define RT5682_I2S2_MS_S			(0x1 << 15)
+#define RT5682_I2S2_PIN_CFG_MASK		(0x1 << 14)
+#define RT5682_I2S2_PIN_CFG_SFT			14
+#define RT5682_I2S2_CLK_SEL_MASK		(0x1 << 11)
+#define RT5682_I2S2_CLK_SEL_SFT			11
+#define RT5682_I2S2_OUT_MASK			(0x1 << 9)
+#define RT5682_I2S2_OUT_SFT			9
+#define RT5682_I2S2_OUT_UM			(0x0 << 9)
+#define RT5682_I2S2_OUT_M			(0x1 << 9)
+#define RT5682_I2S_BP_MASK			(0x1 << 8)
+#define RT5682_I2S_BP_SFT			8
+#define RT5682_I2S_BP_NOR			(0x0 << 8)
+#define RT5682_I2S_BP_INV			(0x1 << 8)
+#define RT5682_I2S2_MONO_EN			(0x1 << 6)
+#define RT5682_I2S2_MONO_DIS			(0x0 << 6)
+#define RT5682_I2S2_DL_MASK			(0x3 << 4)
+#define RT5682_I2S2_DL_SFT			4
+#define RT5682_I2S2_DL_16			(0x0 << 4)
+#define RT5682_I2S2_DL_20			(0x1 << 4)
+#define RT5682_I2S2_DL_24			(0x2 << 4)
+#define RT5682_I2S2_DL_8			(0x3 << 4)
+#define RT5682_I2S_DF_MASK			(0x7)
+#define RT5682_I2S_DF_SFT			0
+#define RT5682_I2S_DF_I2S			(0x0)
+#define RT5682_I2S_DF_LEFT			(0x1)
+#define RT5682_I2S_DF_PCM_A			(0x2)
+#define RT5682_I2S_DF_PCM_B			(0x3)
+#define RT5682_I2S_DF_PCM_A_N			(0x6)
+#define RT5682_I2S_DF_PCM_B_N			(0x7)
+
+/* ADC/DAC Clock Control 1 (0x0073) */
+#define RT5682_ADC_OSR_MASK			(0xf << 12)
+#define RT5682_ADC_OSR_SFT			12
+#define RT5682_ADC_OSR_D_1			(0x0 << 12)
+#define RT5682_ADC_OSR_D_2			(0x1 << 12)
+#define RT5682_ADC_OSR_D_4			(0x2 << 12)
+#define RT5682_ADC_OSR_D_6			(0x3 << 12)
+#define RT5682_ADC_OSR_D_8			(0x4 << 12)
+#define RT5682_ADC_OSR_D_12			(0x5 << 12)
+#define RT5682_ADC_OSR_D_16			(0x6 << 12)
+#define RT5682_ADC_OSR_D_24			(0x7 << 12)
+#define RT5682_ADC_OSR_D_32			(0x8 << 12)
+#define RT5682_ADC_OSR_D_48			(0x9 << 12)
+#define RT5682_I2S_M_DIV_MASK			(0xf << 12)
+#define RT5682_I2S_M_DIV_SFT			8
+#define RT5682_I2S_M_D_1			(0x0 << 8)
+#define RT5682_I2S_M_D_2			(0x1 << 8)
+#define RT5682_I2S_M_D_3			(0x2 << 8)
+#define RT5682_I2S_M_D_4			(0x3 << 8)
+#define RT5682_I2S_M_D_6			(0x4 << 8)
+#define RT5682_I2S_M_D_8			(0x5 << 8)
+#define RT5682_I2S_M_D_12			(0x6 << 8)
+#define RT5682_I2S_M_D_16			(0x7 << 8)
+#define RT5682_I2S_M_D_24			(0x8 << 8)
+#define RT5682_I2S_M_D_32			(0x9 << 8)
+#define RT5682_I2S_M_D_48			(0x10 << 8)
+#define RT5682_I2S_CLK_SRC_MASK			(0x7 << 4)
+#define RT5682_I2S_CLK_SRC_SFT			4
+#define RT5682_I2S_CLK_SRC_MCLK			(0x0 << 4)
+#define RT5682_I2S_CLK_SRC_PLL1			(0x1 << 4)
+#define RT5682_I2S_CLK_SRC_PLL2			(0x2 << 4)
+#define RT5682_I2S_CLK_SRC_SDW			(0x3 << 4)
+#define RT5682_I2S_CLK_SRC_RCCLK		(0x4 << 4) /* 25M */
+#define RT5682_DAC_OSR_MASK			(0xf << 0)
+#define RT5682_DAC_OSR_SFT			0
+#define RT5682_DAC_OSR_D_1			(0x0 << 0)
+#define RT5682_DAC_OSR_D_2			(0x1 << 0)
+#define RT5682_DAC_OSR_D_4			(0x2 << 0)
+#define RT5682_DAC_OSR_D_6			(0x3 << 0)
+#define RT5682_DAC_OSR_D_8			(0x4 << 0)
+#define RT5682_DAC_OSR_D_12			(0x5 << 0)
+#define RT5682_DAC_OSR_D_16			(0x6 << 0)
+#define RT5682_DAC_OSR_D_24			(0x7 << 0)
+#define RT5682_DAC_OSR_D_32			(0x8 << 0)
+#define RT5682_DAC_OSR_D_48			(0x9 << 0)
+
+/* ADC/DAC Clock Control 2 (0x0074) */
+#define RT5682_I2S2_BCLK_MS2_MASK		(0x1 << 11)
+#define RT5682_I2S2_BCLK_MS2_SFT		11
+#define RT5682_I2S2_BCLK_MS2_32			(0x0 << 11)
+#define RT5682_I2S2_BCLK_MS2_64			(0x1 << 11)
+
+
+/* TDM control 1 (0x0079) */
+#define RT5682_TDM_TX_CH_MASK			(0x3 << 12)
+#define RT5682_TDM_TX_CH_2			(0x0 << 12)
+#define RT5682_TDM_TX_CH_4			(0x1 << 12)
+#define RT5682_TDM_TX_CH_6			(0x2 << 12)
+#define RT5682_TDM_TX_CH_8			(0x3 << 12)
+#define RT5682_TDM_RX_CH_MASK			(0x3 << 8)
+#define RT5682_TDM_RX_CH_2			(0x0 << 8)
+#define RT5682_TDM_RX_CH_4			(0x1 << 8)
+#define RT5682_TDM_RX_CH_6			(0x2 << 8)
+#define RT5682_TDM_RX_CH_8			(0x3 << 8)
+#define RT5682_TDM_ADC_LCA_MASK			(0xf << 4)
+#define RT5682_TDM_ADC_LCA_SFT			4
+#define RT5682_TDM_ADC_DL_SFT			0
+
+/* TDM control 2 (0x007a) */
+#define RT5682_IF1_ADC1_SEL_SFT			14
+#define RT5682_IF1_ADC2_SEL_SFT			12
+#define RT5682_IF1_ADC3_SEL_SFT			10
+#define RT5682_IF1_ADC4_SEL_SFT			8
+#define RT5682_TDM_ADC_SEL_SFT			4
+
+/* TDM control 3 (0x007b) */
+#define RT5682_TDM_EN				(0x1 << 7)
+
+/* TDM/I2S control (0x007e) */
+#define RT5682_TDM_S_BP_MASK			(0x1 << 15)
+#define RT5682_TDM_S_BP_SFT			15
+#define RT5682_TDM_S_BP_NOR			(0x0 << 15)
+#define RT5682_TDM_S_BP_INV			(0x1 << 15)
+#define RT5682_TDM_S_LP_MASK			(0x1 << 14)
+#define RT5682_TDM_S_LP_SFT			14
+#define RT5682_TDM_S_LP_NOR			(0x0 << 14)
+#define RT5682_TDM_S_LP_INV			(0x1 << 14)
+#define RT5682_TDM_DF_MASK			(0x7 << 11)
+#define RT5682_TDM_DF_SFT			11
+#define RT5682_TDM_DF_I2S			(0x0 << 11)
+#define RT5682_TDM_DF_LEFT			(0x1 << 11)
+#define RT5682_TDM_DF_PCM_A			(0x2 << 11)
+#define RT5682_TDM_DF_PCM_B			(0x3 << 11)
+#define RT5682_TDM_DF_PCM_A_N			(0x6 << 11)
+#define RT5682_TDM_DF_PCM_B_N			(0x7 << 11)
+#define RT5682_TDM_CL_MASK			(0x3 << 4)
+#define RT5682_TDM_CL_16			(0x0 << 4)
+#define RT5682_TDM_CL_20			(0x1 << 4)
+#define RT5682_TDM_CL_24			(0x2 << 4)
+#define RT5682_TDM_CL_32			(0x3 << 4)
+#define RT5682_TDM_M_BP_MASK			(0x1 << 2)
+#define RT5682_TDM_M_BP_SFT			2
+#define RT5682_TDM_M_BP_NOR			(0x0 << 2)
+#define RT5682_TDM_M_BP_INV			(0x1 << 2)
+#define RT5682_TDM_M_LP_MASK			(0x1 << 1)
+#define RT5682_TDM_M_LP_SFT			1
+#define RT5682_TDM_M_LP_NOR			(0x0 << 1)
+#define RT5682_TDM_M_LP_INV			(0x1 << 1)
+#define RT5682_TDM_MS_MASK			(0x1 << 0)
+#define RT5682_TDM_MS_SFT			0
+#define RT5682_TDM_MS_M				(0x0 << 0)
+#define RT5682_TDM_MS_S				(0x1 << 0)
+
+/* Global Clock Control (0x0080) */
+#define RT5682_SCLK_SRC_MASK			(0x7 << 13)
+#define RT5682_SCLK_SRC_SFT			13
+#define RT5682_SCLK_SRC_MCLK			(0x0 << 13)
+#define RT5682_SCLK_SRC_PLL1			(0x1 << 13)
+#define RT5682_SCLK_SRC_PLL2			(0x2 << 13)
+#define RT5682_SCLK_SRC_SDW			(0x3 << 13)
+#define RT5682_SCLK_SRC_RCCLK			(0x4 << 13)
+#define RT5682_PLL1_SRC_MASK			(0x3 << 10)
+#define RT5682_PLL1_SRC_SFT			10
+#define RT5682_PLL1_SRC_MCLK			(0x0 << 10)
+#define RT5682_PLL1_SRC_BCLK1			(0x1 << 10)
+#define RT5682_PLL1_SRC_SDW			(0x2 << 10)
+#define RT5682_PLL1_SRC_RC			(0x3 << 10)
+#define RT5682_PLL2_SRC_MASK			(0x3 << 8)
+#define RT5682_PLL2_SRC_SFT			8
+#define RT5682_PLL2_SRC_MCLK			(0x0 << 8)
+#define RT5682_PLL2_SRC_BCLK1			(0x1 << 8)
+#define RT5682_PLL2_SRC_SDW			(0x2 << 8)
+#define RT5682_PLL2_SRC_RC			(0x3 << 8)
+
+
+
+#define RT5682_PLL_INP_MAX			40000000
+#define RT5682_PLL_INP_MIN			256000
+/* PLL M/N/K Code Control 1 (0x0081) */
+#define RT5682_PLL_N_MAX			0x001ff
+#define RT5682_PLL_N_MASK			(RT5682_PLL_N_MAX << 7)
+#define RT5682_PLL_N_SFT			7
+#define RT5682_PLL_K_MAX			0x001f
+#define RT5682_PLL_K_MASK			(RT5682_PLL_K_MAX)
+#define RT5682_PLL_K_SFT			0
+
+/* PLL M/N/K Code Control 2 (0x0082) */
+#define RT5682_PLL_M_MAX			0x00f
+#define RT5682_PLL_M_MASK			(RT5682_PLL_M_MAX << 12)
+#define RT5682_PLL_M_SFT			12
+#define RT5682_PLL_M_BP				(0x1 << 11)
+#define RT5682_PLL_M_BP_SFT			11
+#define RT5682_PLL_K_BP				(0x1 << 10)
+#define RT5682_PLL_K_BP_SFT			10
+#define RT5682_PLL_RST				(0x1 << 1)
+
+/* PLL tracking mode 1 (0x0083) */
+#define RT5682_DA_ASRC_MASK			(0x1 << 13)
+#define RT5682_DA_ASRC_SFT			13
+#define RT5682_DAC_STO1_ASRC_MASK		(0x1 << 12)
+#define RT5682_DAC_STO1_ASRC_SFT		12
+#define RT5682_AD_ASRC_MASK			(0x1 << 8)
+#define RT5682_AD_ASRC_SFT			8
+#define RT5682_AD_ASRC_SEL_MASK			(0x1 << 4)
+#define RT5682_AD_ASRC_SEL_SFT			4
+#define RT5682_DMIC_ASRC_MASK			(0x1 << 3)
+#define RT5682_DMIC_ASRC_SFT			3
+#define RT5682_ADC_STO1_ASRC_MASK		(0x1 << 2)
+#define RT5682_ADC_STO1_ASRC_SFT		2
+#define RT5682_DA_ASRC_SEL_MASK			(0x1 << 0)
+#define RT5682_DA_ASRC_SEL_SFT			0
+
+/* PLL tracking mode 2 3 (0x0084)(0x0085)*/
+#define RT5682_FILTER_CLK_SEL_MASK		(0x7 << 12)
+#define RT5682_FILTER_CLK_SEL_SFT		12
+#define RT5682_FILTER_CLK_DIV_MASK		(0xf << 8)
+#define RT5682_FILTER_CLK_DIV_SFT		8
+
+/* ASRC Control 4 (0x0086) */
+#define RT5682_ASRCIN_FTK_N1_MASK		(0x3 << 14)
+#define RT5682_ASRCIN_FTK_N1_SFT		14
+#define RT5682_ASRCIN_FTK_N2_MASK		(0x3 << 12)
+#define RT5682_ASRCIN_FTK_N2_SFT		12
+#define RT5682_ASRCIN_FTK_M1_MASK		(0x7 << 8)
+#define RT5682_ASRCIN_FTK_M1_SFT		8
+#define RT5682_ASRCIN_FTK_M2_MASK		(0x7 << 4)
+#define RT5682_ASRCIN_FTK_M2_SFT		4
+
+/* SoundWire reference clk (0x008d) */
+#define RT5682_PLL2_OUT_MASK			(0x1 << 8)
+#define RT5682_PLL2_OUT_98M			(0x0 << 8)
+#define RT5682_PLL2_OUT_49M			(0x1 << 8)
+#define RT5682_SDW_REF_2_MASK			(0xf << 4)
+#define RT5682_SDW_REF_2_SFT			4
+#define RT5682_SDW_REF_2_48K			(0x0 << 4)
+#define RT5682_SDW_REF_2_96K			(0x1 << 4)
+#define RT5682_SDW_REF_2_192K			(0x2 << 4)
+#define RT5682_SDW_REF_2_32K			(0x3 << 4)
+#define RT5682_SDW_REF_2_24K			(0x4 << 4)
+#define RT5682_SDW_REF_2_16K			(0x5 << 4)
+#define RT5682_SDW_REF_2_12K			(0x6 << 4)
+#define RT5682_SDW_REF_2_8K			(0x7 << 4)
+#define RT5682_SDW_REF_2_44K			(0x8 << 4)
+#define RT5682_SDW_REF_2_88K			(0x9 << 4)
+#define RT5682_SDW_REF_2_176K			(0xa << 4)
+#define RT5682_SDW_REF_2_353K			(0xb << 4)
+#define RT5682_SDW_REF_2_22K			(0xc << 4)
+#define RT5682_SDW_REF_2_384K			(0xd << 4)
+#define RT5682_SDW_REF_2_11K			(0xe << 4)
+#define RT5682_SDW_REF_1_MASK			(0xf << 0)
+#define RT5682_SDW_REF_1_SFT			0
+#define RT5682_SDW_REF_1_48K			(0x0 << 0)
+#define RT5682_SDW_REF_1_96K			(0x1 << 0)
+#define RT5682_SDW_REF_1_192K			(0x2 << 0)
+#define RT5682_SDW_REF_1_32K			(0x3 << 0)
+#define RT5682_SDW_REF_1_24K			(0x4 << 0)
+#define RT5682_SDW_REF_1_16K			(0x5 << 0)
+#define RT5682_SDW_REF_1_12K			(0x6 << 0)
+#define RT5682_SDW_REF_1_8K			(0x7 << 0)
+#define RT5682_SDW_REF_1_44K			(0x8 << 0)
+#define RT5682_SDW_REF_1_88K			(0x9 << 0)
+#define RT5682_SDW_REF_1_176K			(0xa << 0)
+#define RT5682_SDW_REF_1_353K			(0xb << 0)
+#define RT5682_SDW_REF_1_22K			(0xc << 0)
+#define RT5682_SDW_REF_1_384K			(0xd << 0)
+#define RT5682_SDW_REF_1_11K			(0xe << 0)
+
+/* Depop Mode Control 1 (0x008e) */
+#define RT5682_PUMP_EN				(0x1 << 3)
+#define RT5682_PUMP_EN_SFT				3
+#define RT5682_CAPLESS_EN			(0x1 << 0)
+#define RT5682_CAPLESS_EN_SFT			0
+
+/* Depop Mode Control 2 (0x8f) */
+#define RT5682_RAMP_MASK			(0x1 << 12)
+#define RT5682_RAMP_SFT				12
+#define RT5682_RAMP_DIS				(0x0 << 12)
+#define RT5682_RAMP_EN				(0x1 << 12)
+#define RT5682_BPS_MASK				(0x1 << 11)
+#define RT5682_BPS_SFT				11
+#define RT5682_BPS_DIS				(0x0 << 11)
+#define RT5682_BPS_EN				(0x1 << 11)
+#define RT5682_FAST_UPDN_MASK			(0x1 << 10)
+#define RT5682_FAST_UPDN_SFT			10
+#define RT5682_FAST_UPDN_DIS			(0x0 << 10)
+#define RT5682_FAST_UPDN_EN			(0x1 << 10)
+#define RT5682_VLO_MASK				(0x1 << 7)
+#define RT5682_VLO_SFT				7
+#define RT5682_VLO_3V				(0x0 << 7)
+#define RT5682_VLO_33V				(0x1 << 7)
+
+/* HPOUT charge pump 1 (0x0091) */
+#define RT5682_OSW_L_MASK			(0x1 << 11)
+#define RT5682_OSW_L_SFT			11
+#define RT5682_OSW_L_DIS			(0x0 << 11)
+#define RT5682_OSW_L_EN				(0x1 << 11)
+#define RT5682_OSW_R_MASK			(0x1 << 10)
+#define RT5682_OSW_R_SFT			10
+#define RT5682_OSW_R_DIS			(0x0 << 10)
+#define RT5682_OSW_R_EN				(0x1 << 10)
+#define RT5682_PM_HP_MASK			(0x3 << 8)
+#define RT5682_PM_HP_SFT			8
+#define RT5682_PM_HP_LV				(0x0 << 8)
+#define RT5682_PM_HP_MV				(0x1 << 8)
+#define RT5682_PM_HP_HV				(0x2 << 8)
+#define RT5682_IB_HP_MASK			(0x3 << 6)
+#define RT5682_IB_HP_SFT			6
+#define RT5682_IB_HP_125IL			(0x0 << 6)
+#define RT5682_IB_HP_25IL			(0x1 << 6)
+#define RT5682_IB_HP_5IL			(0x2 << 6)
+#define RT5682_IB_HP_1IL			(0x3 << 6)
+
+/* Micbias Control1 (0x93) */
+#define RT5682_MIC1_OV_MASK			(0x3 << 14)
+#define RT5682_MIC1_OV_SFT			14
+#define RT5682_MIC1_OV_2V7			(0x0 << 14)
+#define RT5682_MIC1_OV_2V4			(0x1 << 14)
+#define RT5682_MIC1_OV_2V25			(0x3 << 14)
+#define RT5682_MIC1_OV_1V8			(0x4 << 14)
+#define RT5682_MIC1_CLK_MASK			(0x1 << 13)
+#define RT5682_MIC1_CLK_SFT			13
+#define RT5682_MIC1_CLK_DIS			(0x0 << 13)
+#define RT5682_MIC1_CLK_EN			(0x1 << 13)
+#define RT5682_MIC1_OVCD_MASK			(0x1 << 12)
+#define RT5682_MIC1_OVCD_SFT			12
+#define RT5682_MIC1_OVCD_DIS			(0x0 << 12)
+#define RT5682_MIC1_OVCD_EN			(0x1 << 12)
+#define RT5682_MIC1_OVTH_MASK			(0x3 << 10)
+#define RT5682_MIC1_OVTH_SFT			10
+#define RT5682_MIC1_OVTH_768UA			(0x0 << 10)
+#define RT5682_MIC1_OVTH_960UA			(0x1 << 10)
+#define RT5682_MIC1_OVTH_1152UA			(0x2 << 10)
+#define RT5682_MIC1_OVTH_1960UA			(0x3 << 10)
+#define RT5682_MIC2_OV_MASK			(0x3 << 8)
+#define RT5682_MIC2_OV_SFT			8
+#define RT5682_MIC2_OV_2V7			(0x0 << 8)
+#define RT5682_MIC2_OV_2V4			(0x1 << 8)
+#define RT5682_MIC2_OV_2V25			(0x3 << 8)
+#define RT5682_MIC2_OV_1V8			(0x4 << 8)
+#define RT5682_MIC2_CLK_MASK			(0x1 << 7)
+#define RT5682_MIC2_CLK_SFT			7
+#define RT5682_MIC2_CLK_DIS			(0x0 << 7)
+#define RT5682_MIC2_CLK_EN			(0x1 << 7)
+#define RT5682_MIC2_OVTH_MASK			(0x3 << 4)
+#define RT5682_MIC2_OVTH_SFT			4
+#define RT5682_MIC2_OVTH_768UA			(0x0 << 4)
+#define RT5682_MIC2_OVTH_960UA			(0x1 << 4)
+#define RT5682_MIC2_OVTH_1152UA			(0x2 << 4)
+#define RT5682_MIC2_OVTH_1960UA			(0x3 << 4)
+#define RT5682_PWR_MB_MASK			(0x1 << 3)
+#define RT5682_PWR_MB_SFT			3
+#define RT5682_PWR_MB_PD			(0x0 << 3)
+#define RT5682_PWR_MB_PU			(0x1 << 3)
+
+/* Micbias Control2 (0x0094) */
+#define RT5682_PWR_CLK25M_MASK			(0x1 << 9)
+#define RT5682_PWR_CLK25M_SFT			9
+#define RT5682_PWR_CLK25M_PD			(0x0 << 9)
+#define RT5682_PWR_CLK25M_PU			(0x1 << 9)
+#define RT5682_PWR_CLK1M_MASK			(0x1 << 8)
+#define RT5682_PWR_CLK1M_SFT			8
+#define RT5682_PWR_CLK1M_PD			(0x0 << 8)
+#define RT5682_PWR_CLK1M_PU			(0x1 << 8)
+
+/* RC Clock Control (0x009f) */
+#define RT5682_POW_IRQ				(0x1 << 15)
+#define RT5682_POW_JDH				(0x1 << 14)
+#define RT5682_POW_JDL				(0x1 << 13)
+#define RT5682_POW_ANA				(0x1 << 12)
+
+/* I2S Master Mode Clock Control 1 (0x00a0) */
+#define RT5682_CLK_SRC_MCLK			(0x0)
+#define RT5682_CLK_SRC_PLL1			(0x1)
+#define RT5682_CLK_SRC_PLL2			(0x2)
+#define RT5682_CLK_SRC_SDW			(0x3)
+#define RT5682_CLK_SRC_RCCLK			(0x4)
+#define RT5682_I2S_PD_1				(0x0)
+#define RT5682_I2S_PD_2				(0x1)
+#define RT5682_I2S_PD_3				(0x2)
+#define RT5682_I2S_PD_4				(0x3)
+#define RT5682_I2S_PD_6				(0x4)
+#define RT5682_I2S_PD_8				(0x5)
+#define RT5682_I2S_PD_12			(0x6)
+#define RT5682_I2S_PD_16			(0x7)
+#define RT5682_I2S_PD_24			(0x8)
+#define RT5682_I2S_PD_32			(0x9)
+#define RT5682_I2S_PD_48			(0xa)
+#define RT5682_I2S2_SRC_MASK			(0x3 << 4)
+#define RT5682_I2S2_SRC_SFT			4
+#define RT5682_I2S2_M_PD_MASK			(0xf << 0)
+#define RT5682_I2S2_M_PD_SFT			0
+
+/* IRQ Control 1 (0x00b6) */
+#define RT5682_JD1_PULSE_EN_MASK		(0x1 << 10)
+#define RT5682_JD1_PULSE_EN_SFT			10
+#define RT5682_JD1_PULSE_DIS			(0x0 << 10)
+#define RT5682_JD1_PULSE_EN			(0x1 << 10)
+
+/* IRQ Control 2 (0x00b7) */
+#define RT5682_JD1_EN_MASK			(0x1 << 15)
+#define RT5682_JD1_EN_SFT			15
+#define RT5682_JD1_DIS				(0x0 << 15)
+#define RT5682_JD1_EN				(0x1 << 15)
+#define RT5682_JD1_POL_MASK			(0x1 << 13)
+#define RT5682_JD1_POL_NOR			(0x0 << 13)
+#define RT5682_JD1_POL_INV			(0x1 << 13)
+
+/* IRQ Control 3 (0x00b8) */
+#define RT5682_IL_IRQ_MASK			(0x1 << 7)
+#define RT5682_IL_IRQ_DIS			(0x0 << 7)
+#define RT5682_IL_IRQ_EN			(0x1 << 7)
+
+/* GPIO Control 1 (0x00c0) */
+#define RT5682_GP1_PIN_MASK			(0x3 << 14)
+#define RT5682_GP1_PIN_SFT			14
+#define RT5682_GP1_PIN_GPIO1			(0x0 << 14)
+#define RT5682_GP1_PIN_IRQ			(0x1 << 14)
+#define RT5682_GP1_PIN_DMIC_CLK			(0x2 << 14)
+#define RT5682_GP2_PIN_MASK			(0x3 << 12)
+#define RT5682_GP2_PIN_SFT			12
+#define RT5682_GP2_PIN_GPIO2			(0x0 << 12)
+#define RT5682_GP2_PIN_LRCK2			(0x1 << 12)
+#define RT5682_GP2_PIN_DMIC_SDA			(0x2 << 12)
+#define RT5682_GP3_PIN_MASK			(0x3 << 10)
+#define RT5682_GP3_PIN_SFT			10
+#define RT5682_GP3_PIN_GPIO3			(0x0 << 10)
+#define RT5682_GP3_PIN_BCLK2			(0x1 << 10)
+#define RT5682_GP3_PIN_DMIC_CLK			(0x2 << 10)
+#define RT5682_GP4_PIN_MASK			(0x3 << 8)
+#define RT5682_GP4_PIN_SFT			8
+#define RT5682_GP4_PIN_GPIO4			(0x0 << 8)
+#define RT5682_GP4_PIN_ADCDAT1			(0x1 << 8)
+#define RT5682_GP4_PIN_DMIC_CLK			(0x2 << 8)
+#define RT5682_GP4_PIN_ADCDAT2			(0x3 << 8)
+#define RT5682_GP5_PIN_MASK			(0x3 << 6)
+#define RT5682_GP5_PIN_SFT			6
+#define RT5682_GP5_PIN_GPIO5			(0x0 << 6)
+#define RT5682_GP5_PIN_DACDAT1			(0x1 << 6)
+#define RT5682_GP5_PIN_DMIC_SDA			(0x2 << 6)
+#define RT5682_GP6_PIN_MASK			(0x1 << 5)
+#define RT5682_GP6_PIN_SFT			5
+#define RT5682_GP6_PIN_GPIO6			(0x0 << 5)
+#define RT5682_GP6_PIN_LRCK1			(0x1 << 5)
+
+/* GPIO Control 2 (0x00c1)*/
+#define RT5682_GP1_PF_MASK			(0x1 << 15)
+#define RT5682_GP1_PF_IN			(0x0 << 15)
+#define RT5682_GP1_PF_OUT			(0x1 << 15)
+#define RT5682_GP1_OUT_MASK			(0x1 << 14)
+#define RT5682_GP1_OUT_L			(0x0 << 14)
+#define RT5682_GP1_OUT_H			(0x1 << 14)
+#define RT5682_GP2_PF_MASK			(0x1 << 13)
+#define RT5682_GP2_PF_IN			(0x0 << 13)
+#define RT5682_GP2_PF_OUT			(0x1 << 13)
+#define RT5682_GP2_OUT_MASK			(0x1 << 12)
+#define RT5682_GP2_OUT_L			(0x0 << 12)
+#define RT5682_GP2_OUT_H			(0x1 << 12)
+#define RT5682_GP3_PF_MASK			(0x1 << 11)
+#define RT5682_GP3_PF_IN			(0x0 << 11)
+#define RT5682_GP3_PF_OUT			(0x1 << 11)
+#define RT5682_GP3_OUT_MASK			(0x1 << 10)
+#define RT5682_GP3_OUT_L			(0x0 << 10)
+#define RT5682_GP3_OUT_H			(0x1 << 10)
+#define RT5682_GP4_PF_MASK			(0x1 << 9)
+#define RT5682_GP4_PF_IN			(0x0 << 9)
+#define RT5682_GP4_PF_OUT			(0x1 << 9)
+#define RT5682_GP4_OUT_MASK			(0x1 << 8)
+#define RT5682_GP4_OUT_L			(0x0 << 8)
+#define RT5682_GP4_OUT_H			(0x1 << 8)
+#define RT5682_GP5_PF_MASK			(0x1 << 7)
+#define RT5682_GP5_PF_IN			(0x0 << 7)
+#define RT5682_GP5_PF_OUT			(0x1 << 7)
+#define RT5682_GP5_OUT_MASK			(0x1 << 6)
+#define RT5682_GP5_OUT_L			(0x0 << 6)
+#define RT5682_GP5_OUT_H			(0x1 << 6)
+#define RT5682_GP6_PF_MASK			(0x1 << 5)
+#define RT5682_GP6_PF_IN			(0x0 << 5)
+#define RT5682_GP6_PF_OUT			(0x1 << 5)
+#define RT5682_GP6_OUT_MASK			(0x1 << 4)
+#define RT5682_GP6_OUT_L			(0x0 << 4)
+#define RT5682_GP6_OUT_H			(0x1 << 4)
+
+
+/* GPIO Status (0x00c2) */
+#define RT5682_GP6_STA				(0x1 << 6)
+#define RT5682_GP5_STA				(0x1 << 5)
+#define RT5682_GP4_STA				(0x1 << 4)
+#define RT5682_GP3_STA				(0x1 << 3)
+#define RT5682_GP2_STA				(0x1 << 2)
+#define RT5682_GP1_STA				(0x1 << 1)
+
+/* Soft volume and zero cross control 1 (0x00d9) */
+#define RT5682_SV_MASK				(0x1 << 15)
+#define RT5682_SV_SFT				15
+#define RT5682_SV_DIS				(0x0 << 15)
+#define RT5682_SV_EN				(0x1 << 15)
+#define RT5682_ZCD_MASK				(0x1 << 10)
+#define RT5682_ZCD_SFT				10
+#define RT5682_ZCD_PD				(0x0 << 10)
+#define RT5682_ZCD_PU				(0x1 << 10)
+#define RT5682_SV_DLY_MASK			(0xf)
+#define RT5682_SV_DLY_SFT			0
+
+/* Soft volume and zero cross control 2 (0x00da) */
+#define RT5682_ZCD_BST1_CBJ_MASK		(0x1 << 7)
+#define RT5682_ZCD_BST1_CBJ_SFT			7
+#define RT5682_ZCD_BST1_CBJ_DIS			(0x0 << 7)
+#define RT5682_ZCD_BST1_CBJ_EN			(0x1 << 7)
+#define RT5682_ZCD_RECMIX_MASK			(0x1)
+#define RT5682_ZCD_RECMIX_SFT			0
+#define RT5682_ZCD_RECMIX_DIS			(0x0)
+#define RT5682_ZCD_RECMIX_EN			(0x1)
+
+/* 4 Button Inline Command Control 2 (0x00e3) */
+#define RT5682_4BTN_IL_MASK			(0x1 << 15)
+#define RT5682_4BTN_IL_EN			(0x1 << 15)
+#define RT5682_4BTN_IL_DIS			(0x0 << 15)
+#define RT5682_4BTN_IL_RST_MASK			(0x1 << 14)
+#define RT5682_4BTN_IL_NOR			(0x1 << 14)
+#define RT5682_4BTN_IL_RST			(0x0 << 14)
+
+/* Analog JD Control (0x00f0) */
+#define RT5682_JDH_RS_MASK			(0x1 << 4)
+#define RT5682_JDH_NO_PLUG			(0x1 << 4)
+#define RT5682_JDH_PLUG				(0x0 << 4)
+
+/* Chopper and Clock control for DAC (0x013a)*/
+#define RT5682_CKXEN_DAC1_MASK			(0x1 << 13)
+#define RT5682_CKXEN_DAC1_SFT			13
+#define RT5682_CKGEN_DAC1_MASK			(0x1 << 12)
+#define RT5682_CKGEN_DAC1_SFT			12
+
+/* Chopper and Clock control for ADC (0x013b)*/
+#define RT5682_CKXEN_ADC1_MASK			(0x1 << 13)
+#define RT5682_CKXEN_ADC1_SFT			13
+#define RT5682_CKGEN_ADC1_MASK			(0x1 << 12)
+#define RT5682_CKGEN_ADC1_SFT			12
+
+/* Volume test (0x013f)*/
+#define RT5682_SEL_CLK_VOL_MASK			(0x1 << 15)
+#define RT5682_SEL_CLK_VOL_EN			(0x1 << 15)
+#define RT5682_SEL_CLK_VOL_DIS			(0x0 << 15)
+
+/* Test Mode Control 1 (0x0145) */
+#define RT5682_AD2DA_LB_MASK			(0x1 << 10)
+#define RT5682_AD2DA_LB_SFT			10
+
+/* Stereo Noise Gate Control 1 (0x0160) */
+#define RT5682_NG2_EN_MASK			(0x1 << 15)
+#define RT5682_NG2_EN				(0x1 << 15)
+#define RT5682_NG2_DIS				(0x0 << 15)
+
+/* Stereo1 DAC Silence Detection Control (0x0190) */
+#define RT5682_DEB_STO_DAC_MASK			(0x7 << 4)
+#define RT5682_DEB_80_MS			(0x0 << 4)
+
+/* SAR ADC Inline Command Control 1 (0x0210) */
+#define RT5682_SAR_BUTT_DET_MASK		(0x1 << 15)
+#define RT5682_SAR_BUTT_DET_EN			(0x1 << 15)
+#define RT5682_SAR_BUTT_DET_DIS			(0x0 << 15)
+#define RT5682_SAR_BUTDET_MODE_MASK		(0x1 << 14)
+#define RT5682_SAR_BUTDET_POW_SAV		(0x1 << 14)
+#define RT5682_SAR_BUTDET_POW_NORM		(0x0 << 14)
+#define RT5682_SAR_BUTDET_RST_MASK		(0x1 << 13)
+#define RT5682_SAR_BUTDET_RST_NORMAL		(0x1 << 13)
+#define RT5682_SAR_BUTDET_RST			(0x0 << 13)
+#define RT5682_SAR_POW_MASK			(0x1 << 12)
+#define RT5682_SAR_POW_EN			(0x1 << 12)
+#define RT5682_SAR_POW_DIS			(0x0 << 12)
+#define RT5682_SAR_RST_MASK			(0x1 << 11)
+#define RT5682_SAR_RST_NORMAL			(0x1 << 11)
+#define RT5682_SAR_RST				(0x0 << 11)
+#define RT5682_SAR_BYPASS_MASK			(0x1 << 10)
+#define RT5682_SAR_BYPASS_EN			(0x1 << 10)
+#define RT5682_SAR_BYPASS_DIS			(0x0 << 10)
+#define RT5682_SAR_SEL_MB1_MASK			(0x1 << 9)
+#define RT5682_SAR_SEL_MB1_SEL			(0x1 << 9)
+#define RT5682_SAR_SEL_MB1_NOSEL		(0x0 << 9)
+#define RT5682_SAR_SEL_MB2_MASK			(0x1 << 8)
+#define RT5682_SAR_SEL_MB2_SEL			(0x1 << 8)
+#define RT5682_SAR_SEL_MB2_NOSEL		(0x0 << 8)
+#define RT5682_SAR_SEL_MODE_MASK		(0x1 << 7)
+#define RT5682_SAR_SEL_MODE_CMP			(0x1 << 7)
+#define RT5682_SAR_SEL_MODE_ADC			(0x0 << 7)
+#define RT5682_SAR_SEL_MB1_MB2_MASK		(0x1 << 5)
+#define RT5682_SAR_SEL_MB1_MB2_AUTO		(0x1 << 5)
+#define RT5682_SAR_SEL_MB1_MB2_MANU		(0x0 << 5)
+#define RT5682_SAR_SEL_SIGNAL_MASK		(0x1 << 4)
+#define RT5682_SAR_SEL_SIGNAL_AUTO		(0x1 << 4)
+#define RT5682_SAR_SEL_SIGNAL_MANU		(0x0 << 4)
+
+/* SAR ADC Inline Command Control 13 (0x021c) */
+#define RT5682_SAR_SOUR_MASK			(0x3f)
+#define RT5682_SAR_SOUR_BTN			(0x3f)
+#define RT5682_SAR_SOUR_TYPE			(0x0)
+
+
+/* System Clock Source */
+enum {
+	RT5682_SCLK_S_MCLK,
+	RT5682_SCLK_S_PLL1,
+	RT5682_SCLK_S_PLL2,
+	RT5682_SCLK_S_RCCLK,
+};
+
+/* PLL Source */
+enum {
+	RT5682_PLL1_S_MCLK,
+	RT5682_PLL1_S_BCLK1,
+	RT5682_PLL1_S_RCCLK,
+};
+
+enum {
+	RT5682_AIF1,
+	RT5682_AIF2,
+	RT5682_AIFS
+};
+
+/* filter mask */
+enum {
+	RT5682_DA_STEREO1_FILTER = 0x1,
+	RT5682_AD_STEREO1_FILTER = (0x1 << 1),
+};
+
+enum {
+	RT5682_CLK_SEL_SYS,
+	RT5682_CLK_SEL_I2S1_ASRC,
+	RT5682_CLK_SEL_I2S2_ASRC,
+};
+
+int rt5682_sel_asrc_clk_src(struct snd_soc_component *component,
+		unsigned int filter_mask, unsigned int clk_src);
+
+#endif /* __RT5682_H__ */
diff --git a/sound/soc/codecs/dio2125.c b/sound/soc/codecs/simple-amplifier.c
index 09451cd44f9b..85524acf3e9c 100644
--- a/sound/soc/codecs/dio2125.c
+++ b/sound/soc/codecs/simple-amplifier.c
@@ -21,9 +21,9 @@
 #include <linux/module.h>
 #include <sound/soc.h>
 
-#define DRV_NAME "dio2125"
+#define DRV_NAME "simple-amplifier"
 
-struct dio2125 {
+struct simple_amp {
 	struct gpio_desc *gpiod_enable;
 };
 
@@ -31,7 +31,7 @@ static int drv_event(struct snd_soc_dapm_widget *w,
 		     struct snd_kcontrol *control, int event)
 {
 	struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm);
-	struct dio2125 *priv = snd_soc_component_get_drvdata(c);
+	struct simple_amp *priv = snd_soc_component_get_drvdata(c);
 	int val;
 
 	switch (event) {
@@ -51,7 +51,7 @@ static int drv_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget simple_amp_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("INL"),
 	SND_SOC_DAPM_INPUT("INR"),
 	SND_SOC_DAPM_OUT_DRV_E("DRV", SND_SOC_NOPM, 0, 0, NULL, 0, drv_event,
@@ -60,24 +60,24 @@ static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("OUTR"),
 };
 
-static const struct snd_soc_dapm_route dio2125_dapm_routes[] = {
+static const struct snd_soc_dapm_route simple_amp_dapm_routes[] = {
 	{ "DRV", NULL, "INL" },
 	{ "DRV", NULL, "INR" },
 	{ "OUTL", NULL, "DRV" },
 	{ "OUTR", NULL, "DRV" },
 };
 
-static const struct snd_soc_component_driver dio2125_component_driver = {
-	.dapm_widgets		= dio2125_dapm_widgets,
-	.num_dapm_widgets	= ARRAY_SIZE(dio2125_dapm_widgets),
-	.dapm_routes		= dio2125_dapm_routes,
-	.num_dapm_routes	= ARRAY_SIZE(dio2125_dapm_routes),
+static const struct snd_soc_component_driver simple_amp_component_driver = {
+	.dapm_widgets		= simple_amp_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(simple_amp_dapm_widgets),
+	.dapm_routes		= simple_amp_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(simple_amp_dapm_routes),
 };
 
-static int dio2125_probe(struct platform_device *pdev)
+static int simple_amp_probe(struct platform_device *pdev)
 {
 	struct device *dev = &pdev->dev;
-	struct dio2125 *priv;
+	struct simple_amp *priv;
 	int err;
 
 	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -93,28 +93,30 @@ static int dio2125_probe(struct platform_device *pdev)
 		return err;
 	}
 
-	return devm_snd_soc_register_component(dev, &dio2125_component_driver,
+	return devm_snd_soc_register_component(dev,
+					       &simple_amp_component_driver,
 					       NULL, 0);
 }
 
 #ifdef CONFIG_OF
-static const struct of_device_id dio2125_ids[] = {
+static const struct of_device_id simple_amp_ids[] = {
 	{ .compatible = "dioo,dio2125", },
+	{ .compatible = "simple-audio-amplifier", },
 	{ }
 };
-MODULE_DEVICE_TABLE(of, dio2125_ids);
+MODULE_DEVICE_TABLE(of, simple_amp_ids);
 #endif
 
-static struct platform_driver dio2125_driver = {
+static struct platform_driver simple_amp_driver = {
 	.driver = {
 		.name = DRV_NAME,
-		.of_match_table = of_match_ptr(dio2125_ids),
+		.of_match_table = of_match_ptr(simple_amp_ids),
 	},
-	.probe = dio2125_probe,
+	.probe = simple_amp_probe,
 };
 
-module_platform_driver(dio2125_driver);
+module_platform_driver(simple_amp_driver);
 
-MODULE_DESCRIPTION("ASoC DIO2125 output driver");
+MODULE_DESCRIPTION("ASoC Simple Audio Amplifier driver");
 MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c
index 52f34c94ec25..ca2dfe12344e 100644
--- a/sound/soc/codecs/tas571x.c
+++ b/sound/soc/codecs/tas571x.c
@@ -7,6 +7,9 @@
  * TAS5721 support:
  * Copyright (C) 2016 Petr Kulhavy, Barix AG <petr@barix.com>
  *
+ * TAS5707 support:
+ * Copyright (C) 2018 Jerome Brunet, Baylibre SAS <jbrunet@baylibre.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
@@ -444,6 +447,111 @@ static const struct tas571x_chip tas5711_chip = {
 	.vol_reg_size			= 1,
 };
 
+static const struct regmap_range tas5707_volatile_regs_range[] = {
+	regmap_reg_range(TAS571X_CLK_CTRL_REG,  TAS571X_ERR_STATUS_REG),
+	regmap_reg_range(TAS571X_OSC_TRIM_REG,  TAS571X_OSC_TRIM_REG),
+	regmap_reg_range(TAS5707_CH1_BQ0_REG, TAS5707_CH2_BQ6_REG),
+};
+
+static const struct regmap_access_table tas5707_volatile_regs = {
+	.yes_ranges =	tas5707_volatile_regs_range,
+	.n_yes_ranges =	ARRAY_SIZE(tas5707_volatile_regs_range),
+
+};
+
+static const DECLARE_TLV_DB_SCALE(tas5707_volume_tlv, -7900, 50, 1);
+
+static const char * const tas5707_volume_slew_step_txt[] = {
+	"256", "512", "1024", "2048",
+};
+
+static const unsigned int tas5707_volume_slew_step_values[] = {
+	3, 0, 1, 2,
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(tas5707_volume_slew_step_enum,
+				  TAS571X_VOL_CFG_REG, 0, 0x3,
+				  tas5707_volume_slew_step_txt,
+				  tas5707_volume_slew_step_values);
+
+static const struct snd_kcontrol_new tas5707_controls[] = {
+	SOC_SINGLE_TLV("Master Volume",
+		       TAS571X_MVOL_REG,
+		       0, 0xff, 1, tas5707_volume_tlv),
+	SOC_DOUBLE_R_TLV("Speaker Volume",
+			 TAS571X_CH1_VOL_REG,
+			 TAS571X_CH2_VOL_REG,
+			 0, 0xff, 1, tas5707_volume_tlv),
+	SOC_DOUBLE("Speaker Switch",
+		   TAS571X_SOFT_MUTE_REG,
+		   TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT,
+		   1, 1),
+
+	SOC_ENUM("Slew Rate Steps", tas5707_volume_slew_step_enum),
+
+	BIQUAD_COEFS("CH1 - Biquad 0", TAS5707_CH1_BQ0_REG),
+	BIQUAD_COEFS("CH1 - Biquad 1", TAS5707_CH1_BQ1_REG),
+	BIQUAD_COEFS("CH1 - Biquad 2", TAS5707_CH1_BQ2_REG),
+	BIQUAD_COEFS("CH1 - Biquad 3", TAS5707_CH1_BQ3_REG),
+	BIQUAD_COEFS("CH1 - Biquad 4", TAS5707_CH1_BQ4_REG),
+	BIQUAD_COEFS("CH1 - Biquad 5", TAS5707_CH1_BQ5_REG),
+	BIQUAD_COEFS("CH1 - Biquad 6", TAS5707_CH1_BQ6_REG),
+
+	BIQUAD_COEFS("CH2 - Biquad 0", TAS5707_CH2_BQ0_REG),
+	BIQUAD_COEFS("CH2 - Biquad 1", TAS5707_CH2_BQ1_REG),
+	BIQUAD_COEFS("CH2 - Biquad 2", TAS5707_CH2_BQ2_REG),
+	BIQUAD_COEFS("CH2 - Biquad 3", TAS5707_CH2_BQ3_REG),
+	BIQUAD_COEFS("CH2 - Biquad 4", TAS5707_CH2_BQ4_REG),
+	BIQUAD_COEFS("CH2 - Biquad 5", TAS5707_CH2_BQ5_REG),
+	BIQUAD_COEFS("CH2 - Biquad 6", TAS5707_CH2_BQ6_REG),
+};
+
+static const struct reg_default tas5707_reg_defaults[] = {
+	{TAS571X_CLK_CTRL_REG,		0x6c},
+	{TAS571X_DEV_ID_REG,		0x70},
+	{TAS571X_ERR_STATUS_REG,	0x00},
+	{TAS571X_SYS_CTRL_1_REG,	0xa0},
+	{TAS571X_SDI_REG,		0x05},
+	{TAS571X_SYS_CTRL_2_REG,	0x40},
+	{TAS571X_SOFT_MUTE_REG,		0x00},
+	{TAS571X_MVOL_REG,		0xff},
+	{TAS571X_CH1_VOL_REG,		0x30},
+	{TAS571X_CH2_VOL_REG,		0x30},
+	{TAS571X_VOL_CFG_REG,		0x91},
+	{TAS571X_MODULATION_LIMIT_REG,	0x02},
+	{TAS571X_IC_DELAY_CH1_REG,	0xac},
+	{TAS571X_IC_DELAY_CH2_REG,	0x54},
+	{TAS571X_IC_DELAY_CH3_REG,	0xac},
+	{TAS571X_IC_DELAY_CH4_REG,	0x54},
+	{TAS571X_START_STOP_PERIOD_REG,	0x0f},
+	{TAS571X_OSC_TRIM_REG,		0x82},
+	{TAS571X_BKND_ERR_REG,		0x02},
+	{TAS571X_INPUT_MUX_REG,		0x17772},
+	{TAS571X_PWM_MUX_REG,		0x1021345},
+};
+
+static const struct regmap_config tas5707_regmap_config = {
+	.reg_bits			= 8,
+	.val_bits			= 32,
+	.max_register			= 0xff,
+	.reg_read			= tas571x_reg_read,
+	.reg_write			= tas571x_reg_write,
+	.reg_defaults			= tas5707_reg_defaults,
+	.num_reg_defaults		= ARRAY_SIZE(tas5707_reg_defaults),
+	.cache_type			= REGCACHE_RBTREE,
+	.wr_table			= &tas571x_write_regs,
+	.volatile_table			= &tas5707_volatile_regs,
+};
+
+static const struct tas571x_chip tas5707_chip = {
+	.supply_names			= tas5711_supply_names,
+	.num_supply_names		= ARRAY_SIZE(tas5711_supply_names),
+	.controls			= tas5707_controls,
+	.num_controls			= ARRAY_SIZE(tas5707_controls),
+	.regmap_config			= &tas5707_regmap_config,
+	.vol_reg_size			= 1,
+};
+
 static const char *const tas5717_supply_names[] = {
 	"AVDD",
 	"DVDD",
@@ -775,6 +883,7 @@ static int tas571x_i2c_remove(struct i2c_client *client)
 }
 
 static const struct of_device_id tas571x_of_match[] = {
+	{ .compatible = "ti,tas5707", .data = &tas5707_chip, },
 	{ .compatible = "ti,tas5711", .data = &tas5711_chip, },
 	{ .compatible = "ti,tas5717", .data = &tas5717_chip, },
 	{ .compatible = "ti,tas5719", .data = &tas5717_chip, },
@@ -784,6 +893,7 @@ static const struct of_device_id tas571x_of_match[] = {
 MODULE_DEVICE_TABLE(of, tas571x_of_match);
 
 static const struct i2c_device_id tas571x_i2c_id[] = {
+	{ "tas5707", (kernel_ulong_t) &tas5707_chip },
 	{ "tas5711", (kernel_ulong_t) &tas5711_chip },
 	{ "tas5717", (kernel_ulong_t) &tas5717_chip },
 	{ "tas5719", (kernel_ulong_t) &tas5717_chip },
diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h
index c45677bc26ad..bd23e89cfe79 100644
--- a/sound/soc/codecs/tas571x.h
+++ b/sound/soc/codecs/tas571x.h
@@ -53,6 +53,22 @@
 #define TAS571X_PWM_MUX_REG		0x25
 
 /* 20-byte biquad registers */
+#define TAS5707_CH1_BQ0_REG		0x29
+#define TAS5707_CH1_BQ1_REG		0x2a
+#define TAS5707_CH1_BQ2_REG		0x2b
+#define TAS5707_CH1_BQ3_REG		0x2c
+#define TAS5707_CH1_BQ4_REG		0x2d
+#define TAS5707_CH1_BQ5_REG		0x2e
+#define TAS5707_CH1_BQ6_REG		0x2f
+
+#define TAS5707_CH2_BQ0_REG		0x30
+#define TAS5707_CH2_BQ1_REG		0x31
+#define TAS5707_CH2_BQ2_REG		0x32
+#define TAS5707_CH2_BQ3_REG		0x33
+#define TAS5707_CH2_BQ4_REG		0x34
+#define TAS5707_CH2_BQ5_REG		0x35
+#define TAS5707_CH2_BQ6_REG		0x36
+
 #define TAS5717_CH1_BQ0_REG		0x26
 #define TAS5717_CH1_BQ1_REG		0x27
 #define TAS5717_CH1_BQ2_REG		0x28
diff --git a/sound/soc/codecs/tda7419.c b/sound/soc/codecs/tda7419.c
index 225c210ac38f..7f3b79c5a563 100644
--- a/sound/soc/codecs/tda7419.c
+++ b/sound/soc/codecs/tda7419.c
@@ -142,9 +142,9 @@ struct tda7419_vol_control {
 static inline bool tda7419_vol_is_stereo(struct tda7419_vol_control *tvc)
 {
 	if (tvc->reg == tvc->rreg)
-		return 0;
+		return false;
 
-	return 1;
+	return true;
 }
 
 static int tda7419_vol_info(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c
index d18ff17719cc..7396a6e5277e 100644
--- a/sound/soc/codecs/tscs42xx.c
+++ b/sound/soc/codecs/tscs42xx.c
@@ -625,25 +625,34 @@ static int bytes_info_ext(struct snd_kcontrol *kcontrol,
 
 static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
 	/* Volumes */
-	SOC_DOUBLE_R_TLV("Headphone Playback Volume", R_HPVOLL, R_HPVOLR,
+	SOC_DOUBLE_R_TLV("Headphone Volume", R_HPVOLL, R_HPVOLR,
 			FB_HPVOLL, 0x7F, 0, hpvol_scale),
-	SOC_DOUBLE_R_TLV("Speaker Playback Volume", R_SPKVOLL, R_SPKVOLR,
+	SOC_DOUBLE_R_TLV("Speaker Volume", R_SPKVOLL, R_SPKVOLR,
 			FB_SPKVOLL, 0x7F, 0, spkvol_scale),
-	SOC_DOUBLE_R_TLV("Master Playback Volume", R_DACVOLL, R_DACVOLR,
+	SOC_DOUBLE_R_TLV("Master Volume", R_DACVOLL, R_DACVOLR,
 			FB_DACVOLL, 0xFF, 0, dacvol_scale),
-	SOC_DOUBLE_R_TLV("PCM Capture Volume", R_ADCVOLL, R_ADCVOLR,
+	SOC_DOUBLE_R_TLV("PCM Volume", R_ADCVOLL, R_ADCVOLR,
 			FB_ADCVOLL, 0xFF, 0, adcvol_scale),
-	SOC_DOUBLE_R_TLV("Master Capture Volume", R_INVOLL, R_INVOLR,
+	SOC_DOUBLE_R_TLV("Input Volume", R_INVOLL, R_INVOLR,
 			FB_INVOLL, 0x3F, 0, invol_scale),
 
 	/* INSEL */
-	SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", R_INSELL, R_INSELR,
+	SOC_DOUBLE_R_TLV("Mic Boost Volume", R_INSELL, R_INSELR,
 			FB_INSELL_MICBSTL, FV_INSELL_MICBSTL_30DB,
 			0, mic_boost_scale),
 
 	/* Input Channel Map */
 	SOC_ENUM("Input Channel Map", ch_map_select_enum),
 
+	/* Mic Bias */
+	SOC_SINGLE("Mic Bias Boost Switch", 0x71, 0x07, 1, 0),
+
+	/* Headphone Auto Switching */
+	SOC_SINGLE("Headphone Auto Switching Switch",
+			R_CTL, FB_CTL_HPSWEN, 1, 0),
+	SOC_SINGLE("Headphone Detect Polarity Toggle Switch",
+			R_CTL, FB_CTL_HPSWPOL, 1, 0),
+
 	/* Coefficient Ram */
 	COEFF_RAM_CTL("Cascade1L BiQuad1", BIQUAD_SIZE, 0x00),
 	COEFF_RAM_CTL("Cascade1L BiQuad2", BIQUAD_SIZE, 0x05),
@@ -733,9 +742,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
 		R_CLECTL, FB_CLECTL_LIMIT_EN, 1, 0),
 	SOC_SINGLE("Comp Switch",
 		R_CLECTL, FB_CLECTL_COMP_EN, 1, 0),
-	SOC_SINGLE_TLV("CLE Make-Up Gain Playback Volume",
+	SOC_SINGLE_TLV("CLE Make-Up Gain Volume",
 		R_MUGAIN, FB_MUGAIN_CLEMUG, 0x1f, 0, mugain_scale),
-	SOC_SINGLE_TLV("Comp Thresh Playback Volume",
+	SOC_SINGLE_TLV("Comp Thresh Volume",
 		R_COMPTH, FB_COMPTH, 0xff, 0, compth_scale),
 	SOC_ENUM("Comp Ratio", compressor_ratio_enum),
 	SND_SOC_BYTES("Comp Atk Time", R_CATKTCL, 2),
@@ -766,9 +775,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
 
 	SOC_SINGLE("MBC1 Phase Invert Switch",
 		R_DACMBCMUG1, FB_DACMBCMUG1_PHASE, 1, 0),
-	SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Playback Volume",
+	SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Volume",
 		R_DACMBCMUG1, FB_DACMBCMUG1_MUGAIN, 0x1f, 0, mugain_scale),
-	SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Playback Volume",
+	SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Volume",
 		R_DACMBCTHR1, FB_DACMBCTHR1_THRESH, 0xff, 0, compth_scale),
 	SOC_ENUM("DAC MBC1 Comp Ratio",
 		dac_mbc1_compressor_ratio_enum),
@@ -778,9 +787,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
 
 	SOC_SINGLE("MBC2 Phase Invert Switch",
 		R_DACMBCMUG2, FB_DACMBCMUG2_PHASE, 1, 0),
-	SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Playback Volume",
+	SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Volume",
 		R_DACMBCMUG2, FB_DACMBCMUG2_MUGAIN, 0x1f, 0, mugain_scale),
-	SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Playback Volume",
+	SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Volume",
 		R_DACMBCTHR2, FB_DACMBCTHR2_THRESH, 0xff, 0, compth_scale),
 	SOC_ENUM("DAC MBC2 Comp Ratio",
 		dac_mbc2_compressor_ratio_enum),
@@ -790,9 +799,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
 
 	SOC_SINGLE("MBC3 Phase Invert Switch",
 		R_DACMBCMUG3, FB_DACMBCMUG3_PHASE, 1, 0),
-	SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Playback Volume",
+	SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Volume",
 		R_DACMBCMUG3, FB_DACMBCMUG3_MUGAIN, 0x1f, 0, mugain_scale),
-	SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Playback Volume",
+	SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Volume",
 		R_DACMBCTHR3, FB_DACMBCTHR3_THRESH, 0xff, 0, compth_scale),
 	SOC_ENUM("DAC MBC3 Comp Ratio",
 		dac_mbc3_compressor_ratio_enum),
diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h
index 814c8f3c4a68..6b3a21081635 100644
--- a/sound/soc/codecs/tscs42xx.h
+++ b/sound/soc/codecs/tscs42xx.h
@@ -34,6 +34,7 @@ enum {
 #define R_DACSR         0x19
 #define R_PWRM1         0x1A
 #define R_PWRM2         0x1B
+#define R_CTL		0x1C
 #define R_CONFIG0       0x1F
 #define R_CONFIG1       0x20
 #define R_DMICCTL       0x24
@@ -1110,6 +1111,13 @@ enum {
 #define RV_PWRM2_VREF_DISABLE \
 	 RV(FV_PWRM2_VREF_DISABLE, FB_PWRM2_VREF)
 
+/******************************
+ *      R_CTL (0x1C)          *
+ ******************************/
+
+/* Fiel Offsets */
+#define FB_CTL_HPSWEN                        7
+#define FB_CTL_HPSWPOL                       6
 
 /******************************
  *      R_CONFIG0 (0x1F)      *
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index bfd1abd72253..94675da514c8 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -148,7 +148,7 @@ static bool twl6040_can_write_to_chip(struct snd_soc_component *component,
 	case TWL6040_REG_HFRCTL:
 		return priv->dl2_unmuted;
 	default:
-		return 1;
+		return true;
 	}
 }
 
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index 3663b9fd4d65..deff65161504 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1180,6 +1180,9 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L,
 SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT,
 	   WM2200_SPK1R_MUTE_SHIFT, 1, 1),
 SOC_ENUM("RxANC Src", wm2200_rxanc_input_sel),
+
+WM_ADSP_FW_CONTROL("DSP1", 0),
+WM_ADSP_FW_CONTROL("DSP2", 1),
 };
 
 WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE);
@@ -1553,15 +1556,10 @@ static const struct snd_soc_dapm_route wm2200_dapm_routes[] = {
 static int wm2200_probe(struct snd_soc_component *component)
 {
 	struct wm2200_priv *wm2200 = snd_soc_component_get_drvdata(component);
-	int ret;
 
 	wm2200->component = component;
 
-	ret = snd_soc_add_component_controls(component, wm_adsp_fw_controls, 2);
-	if (ret != 0)
-		return ret;
-
-	return ret;
+	return 0;
 }
 
 static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c
index e239f4bf2460..9e987cf07450 100644
--- a/sound/soc/codecs/wm5100-tables.c
+++ b/sound/soc/codecs/wm5100-tables.c
@@ -30,7 +30,7 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg)
 	case WM5100_OUTPUT_STATUS_2:
 	case WM5100_INPUT_ENABLES_STATUS:
 	case WM5100_MIC_DETECT_3:
-		return 1;
+		return true;
 	default:
 		if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) ||
 		    (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) ||
@@ -41,9 +41,9 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg)
 		    (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) ||
 		    (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) ||
 		    (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511))
-			return 1;
+			return true;
 		else
-			return 0;
+			return false;
 	}
 }
 
@@ -798,7 +798,7 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg)
 	case WM5100_DSP3_CONTROL_28:
 	case WM5100_DSP3_CONTROL_29:
 	case WM5100_DSP3_CONTROL_30:
-		return 1;
+		return true;
 	default:
 		if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) ||
 		    (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) ||
@@ -809,9 +809,9 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg)
 		    (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) ||
 		    (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) ||
 		    (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511))
-			return 1;
+			return true;
 		else
-			return 0;
+			return false;
 	}
 }
 
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 1ac83388d1b8..7e817e1877c2 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -985,6 +985,8 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE),
+
+WM_ADSP_FW_CONTROL("DSP1", 0),
 };
 
 ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
@@ -2094,6 +2096,12 @@ static int wm5102_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1);
+	if (ret != 0)
+		dev_warn(&pdev->dev,
+			 "Failed to set compressed IRQ as a wake source: %d\n",
+			 ret);
+
 	arizona_init_common(arizona);
 
 	ret = arizona_init_vol_limit(arizona);
@@ -2117,6 +2125,7 @@ static int wm5102_probe(struct platform_device *pdev)
 err_spk_irqs:
 	arizona_free_spk_irqs(arizona);
 err_dsp_irq:
+	arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
 	arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5102);
 
 	return ret;
@@ -2133,6 +2142,7 @@ static int wm5102_remove(struct platform_device *pdev)
 
 	arizona_free_spk_irqs(arizona);
 
+	arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
 	arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5102);
 
 	return 0;
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index fb9835dcd836..b0789a03d699 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -927,6 +927,11 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE),
+
+WM_ADSP_FW_CONTROL("DSP1", 0),
+WM_ADSP_FW_CONTROL("DSP2", 1),
+WM_ADSP_FW_CONTROL("DSP3", 2),
+WM_ADSP_FW_CONTROL("DSP4", 3),
 };
 
 ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
@@ -2455,6 +2460,12 @@ static int wm5110_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1);
+	if (ret != 0)
+		dev_warn(&pdev->dev,
+			 "Failed to set compressed IRQ as a wake source: %d\n",
+			 ret);
+
 	arizona_init_common(arizona);
 
 	ret = arizona_init_vol_limit(arizona);
@@ -2478,6 +2489,7 @@ static int wm5110_probe(struct platform_device *pdev)
 err_spk_irqs:
 	arizona_free_spk_irqs(arizona);
 err_dsp_irq:
+	arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
 	arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5110);
 
 	return ret;
@@ -2496,6 +2508,7 @@ static int wm5110_remove(struct platform_device *pdev)
 
 	arizona_free_spk_irqs(arizona);
 
+	arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
 	arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5110);
 
 	return 0;
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 7b8b6ef2f632..6cb3c153ba19 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -251,10 +251,10 @@ static bool wm8903_volatile_register(struct device *dev, unsigned int reg)
 	case WM8903_DC_SERVO_READBACK_2:
 	case WM8903_DC_SERVO_READBACK_3:
 	case WM8903_DC_SERVO_READBACK_4:
-		return 1;
+		return true;
 
 	default:
-		return 0;
+		return false;
 	}
 }
 
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 9037a35b931d..1965635ec07c 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1455,6 +1455,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
 		aif1 |= 0x3 | WM8904_AIF_LRCLK_INV;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_A:
 		aif1 |= 0x3;
 		break;
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index ba44e3d6c1e0..cd204f79647d 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -686,6 +686,7 @@ static int wm8955_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
 		aif |= WM8955_LRP;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_A:
 		aif |= 0x3;
 		break;
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index c30f5aa392c6..8dc1f3d6a988 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -839,6 +839,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
 			iface |= 0x000c;
 			break;
 		}
+		/* fall through */
 	default:
 		dev_err(component->dev, "unsupported width %d\n",
 			params_width(params));
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index f70f563d59f3..68b4cadc308f 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -653,6 +653,7 @@ static int wm8961_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 
 	case SND_SOC_DAIFMT_DSP_B:
 		aif |= WM8961_LRP;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_A:
 		aif |= 3;
 		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index a11e9d6bf950..efd8910b1ff7 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2649,6 +2649,7 @@ static int wm8962_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
 		aif0 |= WM8962_LRCLK_INV | 3;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_A:
 		aif0 |= 3;
 
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 62200117444b..6e52c6a8bab3 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -522,7 +522,7 @@ static inline int get_coeff(int mclk, int rate)
 /* The set of rates we can generate from the above for each SYSCLK */
 
 static const unsigned int rates_12288[] = {
-	8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000,
+	8000, 12000, 16000, 24000, 32000, 48000, 96000,
 };
 
 static const struct snd_pcm_hw_constraint_list constraints_12288 = {
@@ -540,7 +540,7 @@ static const struct snd_pcm_hw_constraint_list constraints_112896 = {
 };
 
 static const unsigned int rates_12[] = {
-	8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000,
+	8000, 11025, 12000, 16000, 22050, 24000, 32000, 41100, 48000,
 	48000, 88235, 96000,
 };
 
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 411b9eee88c2..457bc437ce54 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -40,9 +40,9 @@ static bool wm8990_volatile_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
 	case WM8990_RESET:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 7fdfdf3f6e67..14f1b0c0d286 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2432,6 +2432,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
 			snd_soc_component_update_bits(component, WM8994_POWER_MANAGEMENT_2,
 					    WM8994_OPCLK_ENA, 0);
 		}
+		break;
 
 	default:
 		return -EINVAL;
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 60e227832331..68c99fe37097 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1465,6 +1465,7 @@ static int wm8995_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
 		aif |= WM8995_AIF1_LRCLK_INV;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_A:
 		aif |= (0x3 << WM8995_AIF1_FMT_SHIFT);
 		break;
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index d9d206046f8c..91711f8958c5 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1498,9 +1498,9 @@ static bool wm8996_readable_register(struct device *dev, unsigned int reg)
 	case WM8996_RIGHT_PDM_SPEAKER:
 	case WM8996_PDM_SPEAKER_MUTE_SEQUENCE:
 	case WM8996_PDM_SPEAKER_VOLUME:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
@@ -1522,9 +1522,9 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg)
 	case WM8996_MIC_DETECT_3:
 	case WM8996_HEADPHONE_DETECT_1:
 	case WM8996_HEADPHONE_DETECT_2:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
@@ -1858,6 +1858,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
 	case 24576000:
 		ratediv = WM8996_SYSCLK_DIV;
 		wm8996->sysclk /= 2;
+		/* fall through */
 	case 11289600:
 	case 12288000:
 		snd_soc_component_update_bits(component, WM8996_AIF_RATE,
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 5a0ea7b3c149..399255d1f78a 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -933,6 +933,7 @@ static int wm9081_set_dai_fmt(struct snd_soc_dai *dai,
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
 		aif2 |= WM9081_AIF_LRCLK_INV;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_A:
 		aif2 |= 0x3;
 		break;
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 2fcdd84021a5..f61656070225 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -10,6 +10,7 @@
  * published by the Free Software Foundation.
  */
 
+#include <linux/ctype.h>
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/init.h>
@@ -35,15 +36,15 @@
 #include "wm_adsp.h"
 
 #define adsp_crit(_dsp, fmt, ...) \
-	dev_crit(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+	dev_crit(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
 #define adsp_err(_dsp, fmt, ...) \
-	dev_err(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+	dev_err(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
 #define adsp_warn(_dsp, fmt, ...) \
-	dev_warn(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+	dev_warn(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
 #define adsp_info(_dsp, fmt, ...) \
-	dev_info(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+	dev_info(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
 #define adsp_dbg(_dsp, fmt, ...) \
-	dev_dbg(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+	dev_dbg(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
 
 #define ADSP1_CONTROL_1                   0x00
 #define ADSP1_CONTROL_2                   0x02
@@ -418,7 +419,7 @@ static const struct wm_adsp_fw_caps ctrl_caps[] = {
 	{
 		.id = SND_AUDIOCODEC_BESPOKE,
 		.desc = {
-			.max_ch = 1,
+			.max_ch = 8,
 			.sample_rates = { 16000 },
 			.num_sample_rates = 1,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,
@@ -608,7 +609,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp,
 				  struct snd_soc_component *component)
 {
 	struct dentry *root = NULL;
-	char *root_name;
 	int i;
 
 	if (!component->debugfs_root) {
@@ -616,13 +616,7 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp,
 		goto err;
 	}
 
-	root_name = kmalloc(PAGE_SIZE, GFP_KERNEL);
-	if (!root_name)
-		goto err;
-
-	snprintf(root_name, PAGE_SIZE, "dsp%d", dsp->num);
-	root = debugfs_create_dir(root_name, component->debugfs_root);
-	kfree(root_name);
+	root = debugfs_create_dir(dsp->name, component->debugfs_root);
 
 	if (!root)
 		goto err;
@@ -684,8 +678,8 @@ static inline void wm_adsp_debugfs_clear(struct wm_adsp *dsp)
 }
 #endif
 
-static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol,
-			  struct snd_ctl_elem_value *ucontrol)
+int wm_adsp_fw_get(struct snd_kcontrol *kcontrol,
+		   struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
@@ -695,9 +689,10 @@ static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol,
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(wm_adsp_fw_get);
 
-static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol,
-			  struct snd_ctl_elem_value *ucontrol)
+int wm_adsp_fw_put(struct snd_kcontrol *kcontrol,
+		   struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
@@ -721,8 +716,9 @@ static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol,
 
 	return ret;
 }
+EXPORT_SYMBOL_GPL(wm_adsp_fw_put);
 
-static const struct soc_enum wm_adsp_fw_enum[] = {
+const struct soc_enum wm_adsp_fw_enum[] = {
 	SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
 	SOC_ENUM_SINGLE(0, 1, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
 	SOC_ENUM_SINGLE(0, 2, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
@@ -731,24 +727,7 @@ static const struct soc_enum wm_adsp_fw_enum[] = {
 	SOC_ENUM_SINGLE(0, 5, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
 	SOC_ENUM_SINGLE(0, 6, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
 };
-
-const struct snd_kcontrol_new wm_adsp_fw_controls[] = {
-	SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0],
-		     wm_adsp_fw_get, wm_adsp_fw_put),
-	SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1],
-		     wm_adsp_fw_get, wm_adsp_fw_put),
-	SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2],
-		     wm_adsp_fw_get, wm_adsp_fw_put),
-	SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3],
-		     wm_adsp_fw_get, wm_adsp_fw_put),
-	SOC_ENUM_EXT("DSP5 Firmware", wm_adsp_fw_enum[4],
-		     wm_adsp_fw_get, wm_adsp_fw_put),
-	SOC_ENUM_EXT("DSP6 Firmware", wm_adsp_fw_enum[5],
-		     wm_adsp_fw_get, wm_adsp_fw_put),
-	SOC_ENUM_EXT("DSP7 Firmware", wm_adsp_fw_enum[6],
-		     wm_adsp_fw_get, wm_adsp_fw_put),
-};
-EXPORT_SYMBOL_GPL(wm_adsp_fw_controls);
+EXPORT_SYMBOL_GPL(wm_adsp_fw_enum);
 
 static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp,
 							int type)
@@ -1330,12 +1309,12 @@ static int wm_adsp_create_control(struct wm_adsp *dsp,
 	switch (dsp->fw_ver) {
 	case 0:
 	case 1:
-		snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "DSP%d %s %x",
-			 dsp->num, region_name, alg_region->alg);
+		snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s %x",
+			 dsp->name, region_name, alg_region->alg);
 		break;
 	default:
 		ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
-				"DSP%d%c %.12s %x", dsp->num, *region_name,
+				"%s%c %.12s %x", dsp->name, *region_name,
 				wm_adsp_fw_text[dsp->fw], alg_region->alg);
 
 		/* Truncate the subname from the start if it is too long */
@@ -1343,6 +1322,9 @@ static int wm_adsp_create_control(struct wm_adsp *dsp,
 			int avail = SNDRV_CTL_ELEM_ID_NAME_MAXLEN - ret - 2;
 			int skip = 0;
 
+			if (dsp->component->name_prefix)
+				avail -= strlen(dsp->component->name_prefix) + 1;
+
 			if (subname_len > avail)
 				skip = subname_len - avail;
 
@@ -1604,6 +1586,15 @@ static int wm_adsp_parse_coeff(struct wm_adsp *dsp,
 			if (ret)
 				return -EINVAL;
 			break;
+		case WMFW_CTL_TYPE_HOST_BUFFER:
+			ret = wm_adsp_check_coeff_flags(dsp, &coeff_blk,
+						WMFW_CTL_FLAG_SYS |
+						WMFW_CTL_FLAG_VOLATILE |
+						WMFW_CTL_FLAG_READABLE,
+						0);
+			if (ret)
+				return -EINVAL;
+			break;
 		default:
 			adsp_err(dsp, "Unknown control type: %d\n",
 				 coeff_blk.ctl_type);
@@ -1651,7 +1642,7 @@ static int wm_adsp_load(struct wm_adsp *dsp)
 	if (file == NULL)
 		return -ENOMEM;
 
-	snprintf(file, PAGE_SIZE, "%s-dsp%d-%s.wmfw", dsp->part, dsp->num,
+	snprintf(file, PAGE_SIZE, "%s-%s-%s.wmfw", dsp->part, dsp->fwf_name,
 		 wm_adsp_fw[dsp->fw].file);
 	file[PAGE_SIZE - 1] = '\0';
 
@@ -1871,9 +1862,11 @@ static void wm_adsp_ctl_fixup_base(struct wm_adsp *dsp,
 }
 
 static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs,
+			       const struct wm_adsp_region *mem,
 			       unsigned int pos, unsigned int len)
 {
 	void *alg;
+	unsigned int reg;
 	int ret;
 	__be32 val;
 
@@ -1888,7 +1881,9 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs,
 	}
 
 	/* Read the terminator first to validate the length */
-	ret = regmap_raw_read(dsp->regmap, pos + len, &val, sizeof(val));
+	reg = wm_adsp_region_to_reg(mem, pos + len);
+
+	ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val));
 	if (ret != 0) {
 		adsp_err(dsp, "Failed to read algorithm list end: %d\n",
 			ret);
@@ -1897,13 +1892,18 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs,
 
 	if (be32_to_cpu(val) != 0xbedead)
 		adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbedead\n",
-			  pos + len, be32_to_cpu(val));
+			  reg, be32_to_cpu(val));
+
+	/* Convert length from DSP words to bytes */
+	len *= sizeof(u32);
 
-	alg = kcalloc(len, 2, GFP_KERNEL | GFP_DMA);
+	alg = kzalloc(len, GFP_KERNEL | GFP_DMA);
 	if (!alg)
 		return ERR_PTR(-ENOMEM);
 
-	ret = regmap_raw_read(dsp->regmap, pos, alg, len * 2);
+	reg = wm_adsp_region_to_reg(mem, pos);
+
+	ret = regmap_raw_read(dsp->regmap, reg, alg, len);
 	if (ret != 0) {
 		adsp_err(dsp, "Failed to read algorithm list: %d\n", ret);
 		kfree(alg);
@@ -2002,10 +2002,11 @@ static int wm_adsp1_setup_algs(struct wm_adsp *dsp)
 	if (IS_ERR(alg_region))
 		return PTR_ERR(alg_region);
 
-	pos = sizeof(adsp1_id) / 2;
-	len = (sizeof(*adsp1_alg) * n_algs) / 2;
+	/* Calculate offset and length in DSP words */
+	pos = sizeof(adsp1_id) / sizeof(u32);
+	len = (sizeof(*adsp1_alg) * n_algs) / sizeof(u32);
 
-	adsp1_alg = wm_adsp_read_algs(dsp, n_algs, mem->base + pos, len);
+	adsp1_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len);
 	if (IS_ERR(adsp1_alg))
 		return PTR_ERR(adsp1_alg);
 
@@ -2113,10 +2114,11 @@ static int wm_adsp2_setup_algs(struct wm_adsp *dsp)
 	if (IS_ERR(alg_region))
 		return PTR_ERR(alg_region);
 
-	pos = sizeof(adsp2_id) / 2;
-	len = (sizeof(*adsp2_alg) * n_algs) / 2;
+	/* Calculate offset and length in DSP words */
+	pos = sizeof(adsp2_id) / sizeof(u32);
+	len = (sizeof(*adsp2_alg) * n_algs) / sizeof(u32);
 
-	adsp2_alg = wm_adsp_read_algs(dsp, n_algs, mem->base + pos, len);
+	adsp2_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len);
 	if (IS_ERR(adsp2_alg))
 		return PTR_ERR(adsp2_alg);
 
@@ -2218,7 +2220,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
 	if (file == NULL)
 		return -ENOMEM;
 
-	snprintf(file, PAGE_SIZE, "%s-dsp%d-%s.bin", dsp->part, dsp->num,
+	snprintf(file, PAGE_SIZE, "%s-%s-%s.bin", dsp->part, dsp->fwf_name,
 		 wm_adsp_fw[dsp->fw].file);
 	file[PAGE_SIZE - 1] = '\0';
 
@@ -2390,8 +2392,38 @@ out:
 	return ret;
 }
 
+static int wm_adsp_create_name(struct wm_adsp *dsp)
+{
+	char *p;
+
+	if (!dsp->name) {
+		dsp->name = devm_kasprintf(dsp->dev, GFP_KERNEL, "DSP%d",
+					   dsp->num);
+		if (!dsp->name)
+			return -ENOMEM;
+	}
+
+	if (!dsp->fwf_name) {
+		p = devm_kstrdup(dsp->dev, dsp->name, GFP_KERNEL);
+		if (!p)
+			return -ENOMEM;
+
+		dsp->fwf_name = p;
+		for (; *p != 0; ++p)
+			*p = tolower(*p);
+	}
+
+	return 0;
+}
+
 int wm_adsp1_init(struct wm_adsp *dsp)
 {
+	int ret;
+
+	ret = wm_adsp_create_name(dsp);
+	if (ret)
+		return ret;
+
 	INIT_LIST_HEAD(&dsp->alg_regions);
 
 	mutex_init(&dsp->pwr_lock);
@@ -2642,7 +2674,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
-	struct wm_adsp *dsp = snd_soc_component_get_drvdata(component);
+	struct wm_adsp *dsps = snd_soc_component_get_drvdata(component);
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct wm_adsp *dsp = &dsps[mc->shift - 1];
 
 	ucontrol->value.integer.value[0] = dsp->preloaded;
 
@@ -2654,13 +2689,14 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
-	struct wm_adsp *dsp = snd_soc_component_get_drvdata(component);
+	struct wm_adsp *dsps = snd_soc_component_get_drvdata(component);
 	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
+	struct wm_adsp *dsp = &dsps[mc->shift - 1];
 	char preload[32];
 
-	snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift);
+	snprintf(preload, ARRAY_SIZE(preload), "%s Preload", dsp->name);
 
 	dsp->preloaded = ucontrol->value.integer.value[0];
 
@@ -2671,6 +2707,8 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol,
 
 	snd_soc_dapm_sync(dapm);
 
+	flush_work(&dsp->boot_work);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put);
@@ -2853,17 +2891,14 @@ int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *comp
 {
 	char preload[32];
 
-	snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", dsp->num);
-
+	snprintf(preload, ARRAY_SIZE(preload), "%s Preload", dsp->name);
 	snd_soc_component_disable_pin(component, preload);
 
 	wm_adsp2_init_debugfs(dsp, component);
 
 	dsp->component = component;
 
-	return snd_soc_add_component_controls(component,
-					  &wm_adsp_fw_controls[dsp->num - 1],
-					  1);
+	return 0;
 }
 EXPORT_SYMBOL_GPL(wm_adsp2_component_probe);
 
@@ -2879,6 +2914,10 @@ int wm_adsp2_init(struct wm_adsp *dsp)
 {
 	int ret;
 
+	ret = wm_adsp_create_name(dsp);
+	if (ret)
+		return ret;
+
 	switch (dsp->rev) {
 	case 0:
 		/*
@@ -3186,7 +3225,7 @@ static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf,
 				       buf->host_buf_ptr + field_offset, data);
 }
 
-static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf)
+static int wm_adsp_legacy_host_buf_addr(struct wm_adsp_compr_buf *buf)
 {
 	struct wm_adsp_alg_region *alg_region;
 	struct wm_adsp *dsp = buf->dsp;
@@ -3225,6 +3264,61 @@ static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf)
 	return 0;
 }
 
+static struct wm_coeff_ctl *
+wm_adsp_find_host_buffer_ctrl(struct wm_adsp_compr_buf *buf)
+{
+	struct wm_adsp *dsp = buf->dsp;
+	struct wm_coeff_ctl *ctl;
+
+	list_for_each_entry(ctl, &dsp->ctl_list, list) {
+		if (ctl->type != WMFW_CTL_TYPE_HOST_BUFFER)
+			continue;
+
+		if (!ctl->enabled)
+			continue;
+
+		return ctl;
+	}
+
+	return NULL;
+}
+
+static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf)
+{
+	struct wm_adsp *dsp = buf->dsp;
+	struct wm_coeff_ctl *ctl;
+	unsigned int reg;
+	u32 val;
+	int i, ret;
+
+	ctl = wm_adsp_find_host_buffer_ctrl(buf);
+	if (!ctl)
+		return wm_adsp_legacy_host_buf_addr(buf);
+
+	ret = wm_coeff_base_reg(ctl, &reg);
+	if (ret)
+		return ret;
+
+	for (i = 0; i < 5; ++i) {
+		ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val));
+		if (ret < 0)
+			return ret;
+
+		if (val)
+			break;
+
+		usleep_range(1000, 2000);
+	}
+
+	if (!val)
+		return -EIO;
+
+	buf->host_buf_ptr = be32_to_cpu(val);
+	adsp_dbg(dsp, "host_buf_ptr=%x\n", buf->host_buf_ptr);
+
+	return 0;
+}
+
 static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf)
 {
 	const struct wm_adsp_fw_caps *caps = wm_adsp_fw[buf->dsp->fw].caps;
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index bc6d359f0533..4b8778b0b06c 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -57,6 +57,8 @@ struct wm_adsp_compr_buf;
 
 struct wm_adsp {
 	const char *part;
+	const char *name;
+	const char *fwf_name;
 	int rev;
 	int num;
 	int type;
@@ -121,7 +123,11 @@ struct wm_adsp {
 	.reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp2_event, \
 	.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD }
 
-extern const struct snd_kcontrol_new wm_adsp_fw_controls[];
+#define WM_ADSP_FW_CONTROL(dspname, num) \
+	SOC_ENUM_EXT(dspname " Firmware", wm_adsp_fw_enum[num], \
+		     wm_adsp_fw_get, wm_adsp_fw_put)
+
+extern const struct soc_enum wm_adsp_fw_enum[];
 
 int wm_adsp1_init(struct wm_adsp *dsp);
 int wm_adsp2_init(struct wm_adsp *dsp);
@@ -144,6 +150,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol);
 int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol);
+int wm_adsp_fw_get(struct snd_kcontrol *kcontrol,
+		   struct snd_ctl_elem_value *ucontrol);
+int wm_adsp_fw_put(struct snd_kcontrol *kcontrol,
+		   struct snd_ctl_elem_value *ucontrol);
 
 int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream);
 int wm_adsp_compr_free(struct snd_compr_stream *stream);
diff --git a/sound/soc/codecs/wmfw.h b/sound/soc/codecs/wmfw.h
index ec78b9da020f..0c3f50acb8b1 100644
--- a/sound/soc/codecs/wmfw.h
+++ b/sound/soc/codecs/wmfw.h
@@ -29,6 +29,7 @@
 /* Non-ALSA coefficient types start at 0x1000 */
 #define WMFW_CTL_TYPE_ACKED       0x1000 /* acked control */
 #define WMFW_CTL_TYPE_HOSTEVENT   0x1001 /* event control */
+#define WMFW_CTL_TYPE_HOST_BUFFER 0x1002 /* host buffer pointer */
 
 struct wmfw_header {
 	char magic[4];
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 807040bb3921..a3206e65e5e5 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -340,6 +340,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		 * rate is lowered.
 		 */
 		inv_fs = true;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_A:
 		dev->mode = MOD_DSP_A;
 		break;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 47c0c821d325..f70db8412c7c 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -320,12 +320,8 @@ static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data)
 		handled_mask |= XUNDRN;
 
 		substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK];
-		if (substream) {
-			snd_pcm_stream_lock_irq(substream);
-			if (snd_pcm_running(substream))
-				snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-			snd_pcm_stream_unlock_irq(substream);
-		}
+		if (substream)
+			snd_pcm_stop_xrun(substream);
 	}
 
 	if (!handled_mask)
@@ -355,12 +351,8 @@ static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data)
 		handled_mask |= ROVRN;
 
 		substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE];
-		if (substream) {
-			snd_pcm_stream_lock_irq(substream);
-			if (snd_pcm_running(substream))
-				snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-			snd_pcm_stream_unlock_irq(substream);
-		}
+		if (substream)
+			snd_pcm_stop_xrun(substream);
 	}
 
 	if (!handled_mask)
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 4a6750aa3637..44433b20435c 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -1,14 +1,10 @@
-/*
- * Freescale Generic ASoC Sound Card driver with ASRC
- *
- * Copyright (C) 2014 Freescale Semiconductor, Inc.
- *
- * Author: Nicolin Chen <nicoleotsuka@gmail.com>
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale Generic ASoC Sound Card driver with ASRC
+//
+// Copyright (C) 2014 Freescale Semiconductor, Inc.
+//
+// Author: Nicolin Chen <nicoleotsuka@gmail.com>
 
 #include <linux/clk.h>
 #include <linux/i2c.h>
@@ -199,7 +195,7 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 	snd_mask_none(mask);
-	snd_mask_set(mask, (__force int)priv->asrc_format);
+	snd_mask_set_format(mask, priv->asrc_format);
 
 	return 0;
 }
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index adfb8135d739..528e8b108422 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -1,14 +1,10 @@
-/*
- * Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver
- *
- * Copyright (C) 2014 Freescale Semiconductor, Inc.
- *
- * Author: Nicolin Chen <nicoleotsuka@gmail.com>
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver
+//
+// Copyright (C) 2014 Freescale Semiconductor, Inc.
+//
+// Author: Nicolin Chen <nicoleotsuka@gmail.com>
 
 #include <linux/clk.h>
 #include <linux/delay.h>
diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h
index d558dd5499a5..c60075112570 100644
--- a/sound/soc/fsl/fsl_asrc.h
+++ b/sound/soc/fsl/fsl_asrc.h
@@ -1,13 +1,10 @@
+/* SPDX-License-Identifier: GPL-2.0 */
 /*
  * fsl_asrc.h - Freescale ASRC ALSA SoC header file
  *
  * Copyright (C) 2014 Freescale Semiconductor, Inc.
  *
  * Author: Nicolin Chen <nicoleotsuka@gmail.com>
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
  */
 
 #ifndef _FSL_ASRC_H
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 565e16d8fe85..1033ac6631b0 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -1,14 +1,10 @@
-/*
- * Freescale ASRC ALSA SoC Platform (DMA) driver
- *
- * Copyright (C) 2014 Freescale Semiconductor, Inc.
- *
- * Author: Nicolin Chen <nicoleotsuka@gmail.com>
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale ASRC ALSA SoC Platform (DMA) driver
+//
+// Copyright (C) 2014 Freescale Semiconductor, Inc.
+//
+// Author: Nicolin Chen <nicoleotsuka@gmail.com>
 
 #include <linux/dma-mapping.h>
 #include <linux/module.h>
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 8f43110373b8..c1d1d06783e5 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -249,6 +249,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
 		break;
 	case ESAI_HCKT_EXTAL:
 		ecr |= ESAI_ECR_ETI;
+		/* fall through */
 	case ESAI_HCKR_EXTAL:
 		ecr |= ESAI_ECR_ERI;
 		break;
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 9b59d87b61bf..740b90df44bb 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1118,7 +1118,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
 
 	for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) {
 		for (txclk_df = 1; txclk_df <= 128; txclk_df++) {
-			rate_ideal = rate[index] * txclk_df * 64;
+			rate_ideal = rate[index] * txclk_df * 64ULL;
 			if (round)
 				rate_actual = clk_round_rate(clk, rate_ideal);
 			else
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
index 7592b0406370..7f0fa4b52223 100644
--- a/sound/soc/fsl/fsl_utils.c
+++ b/sound/soc/fsl/fsl_utils.c
@@ -1,14 +1,10 @@
-/**
- * Freescale ALSA SoC Machine driver utility
- *
- * Author: Timur Tabi <timur@freescale.com>
- *
- * Copyright 2010 Freescale Semiconductor, Inc.
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2.  This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale ALSA SoC Machine driver utility
+//
+// Author: Timur Tabi <timur@freescale.com>
+//
+// Copyright 2010 Freescale Semiconductor, Inc.
 
 #include <linux/module.h>
 #include <linux/of_address.h>
diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h
index 1687b66ef18e..c5dc2a14b492 100644
--- a/sound/soc/fsl/fsl_utils.h
+++ b/sound/soc/fsl/fsl_utils.h
@@ -1,13 +1,10 @@
-/**
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
  * Freescale ALSA SoC Machine driver utility
  *
  * Author: Timur Tabi <timur@freescale.com>
  *
  * Copyright 2010 Freescale Semiconductor, Inc.
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2.  This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
  */
 
 #ifndef _FSL_UTILS_H
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index b99e0b5e00e9..c29200cf755a 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -1,14 +1,7 @@
-/*
- * Copyright 2012 Freescale Semiconductor, Inc.
- * Copyright 2012 Linaro Ltd.
- *
- * The code contained herein is licensed under the GNU General Public
- * License. You may obtain a copy of the GNU General Public License
- * Version 2 or later at the following locations:
- *
- * http://www.opensource.org/licenses/gpl-license.html
- * http://www.gnu.org/copyleft/gpl.html
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// Copyright 2012 Freescale Semiconductor, Inc.
+// Copyright 2012 Linaro Ltd.
 
 #include <linux/module.h>
 #include <linux/of.h>
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index d93bacacbd5b..2094d2c8919f 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -1,15 +1,12 @@
-/*
- * ASoC audio graph sound card support
- *
- * Copyright (C) 2016 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * based on ${LINUX}/sound/soc/generic/simple-card.c
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ASoC audio graph sound card support
+//
+// Copyright (C) 2016 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// based on ${LINUX}/sound/soc/generic/simple-card.c
+
 #include <linux/clk.h>
 #include <linux/device.h>
 #include <linux/gpio.h>
@@ -21,7 +18,6 @@
 #include <linux/of_graph.h>
 #include <linux/platform_device.h>
 #include <linux/string.h>
-#include <sound/jack.h>
 #include <sound/simple_card_utils.h>
 
 struct graph_card_data {
@@ -32,6 +28,8 @@ struct graph_card_data {
 		unsigned int mclk_fs;
 	} *dai_props;
 	unsigned int mclk_fs;
+	struct asoc_simple_jack hp_jack;
+	struct asoc_simple_jack mic_jack;
 	struct snd_soc_dai_link *dai_link;
 	struct gpio_desc *pa_gpio;
 };
@@ -278,6 +276,22 @@ static int asoc_graph_get_dais_count(struct device *dev)
 	return count;
 }
 
+static int asoc_graph_soc_card_probe(struct snd_soc_card *card)
+{
+	struct graph_card_data *priv = snd_soc_card_get_drvdata(card);
+	int ret;
+
+	ret = asoc_simple_card_init_hp(card, &priv->hp_jack, NULL);
+	if (ret < 0)
+		return ret;
+
+	ret = asoc_simple_card_init_mic(card, &priv->mic_jack, NULL);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
 static int asoc_graph_card_probe(struct platform_device *pdev)
 {
 	struct graph_card_data *priv;
@@ -319,6 +333,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
 	card->num_links	= num;
 	card->dapm_widgets = asoc_graph_card_dapm_widgets;
 	card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets);
+	card->probe	= asoc_graph_soc_card_probe;
 
 	ret = asoc_graph_card_parse_of(priv);
 	if (ret < 0) {
diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c
index 095ef6426d42..92882e392d6c 100644
--- a/sound/soc/generic/audio-graph-scu-card.c
+++ b/sound/soc/generic/audio-graph-scu-card.c
@@ -1,17 +1,14 @@
-/*
- * ASoC audio graph SCU sound card support
- *
- * Copyright (C) 2017 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * based on
- *	${LINUX}/sound/soc/generic/simple-scu-card.c
- *	${LINUX}/sound/soc/generic/audio-graph-card.c
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ASoC audio graph SCU sound card support
+//
+// Copyright (C) 2017 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// based on
+//	${LINUX}/sound/soc/generic/simple-scu-card.c
+//	${LINUX}/sound/soc/generic/audio-graph-card.c
+
 #include <linux/clk.h>
 #include <linux/device.h>
 #include <linux/gpio.h>
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index 3751a07de6aa..d3f3f0fec74c 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -1,16 +1,17 @@
-/*
- * simple-card-utils.c
- *
- * Copyright (c) 2016 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// simple-card-utils.c
+//
+// Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #include <linux/clk.h>
+#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
 #include <linux/module.h>
 #include <linux/of.h>
+#include <linux/of_gpio.h>
 #include <linux/of_graph.h>
+#include <sound/jack.h>
 #include <sound/simple_card_utils.h>
 
 void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data,
@@ -419,6 +420,61 @@ int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card,
 }
 EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_widgets);
 
+int asoc_simple_card_init_jack(struct snd_soc_card *card,
+			       struct asoc_simple_jack *sjack,
+			       int is_hp, char *prefix)
+{
+	struct device *dev = card->dev;
+	enum of_gpio_flags flags;
+	char prop[128];
+	char *pin_name;
+	char *gpio_name;
+	int mask;
+	int det;
+
+	if (!prefix)
+		prefix = "";
+
+	sjack->gpio.gpio = -ENOENT;
+
+	if (is_hp) {
+		snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix);
+		pin_name	= "Headphones";
+		gpio_name	= "Headphone detection";
+		mask		= SND_JACK_HEADPHONE;
+	} else {
+		snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix);
+		pin_name	= "Mic Jack";
+		gpio_name	= "Mic detection";
+		mask		= SND_JACK_MICROPHONE;
+	}
+
+	det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags);
+	if (det == -EPROBE_DEFER)
+		return -EPROBE_DEFER;
+
+	if (gpio_is_valid(det)) {
+		sjack->pin.pin		= pin_name;
+		sjack->pin.mask		= mask;
+
+		sjack->gpio.name	= gpio_name;
+		sjack->gpio.report	= mask;
+		sjack->gpio.gpio	= det;
+		sjack->gpio.invert	= !!(flags & OF_GPIO_ACTIVE_LOW);
+		sjack->gpio.debounce_time = 150;
+
+		snd_soc_card_jack_new(card, pin_name, mask,
+				      &sjack->jack,
+				      &sjack->pin, 1);
+
+		snd_soc_jack_add_gpios(&sjack->jack, 1,
+				       &sjack->gpio);
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(asoc_simple_card_init_jack);
+
 /* Module information */
 MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
 MODULE_DESCRIPTION("ALSA SoC Simple Card Utils");
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 8b374af86a6e..64bf3560c1d1 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -1,32 +1,20 @@
-/*
- * ASoC simple sound card support
- *
- * Copyright (C) 2012 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ASoC simple sound card support
+//
+// Copyright (C) 2012 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #include <linux/clk.h>
 #include <linux/device.h>
-#include <linux/gpio.h>
 #include <linux/module.h>
 #include <linux/of.h>
-#include <linux/of_gpio.h>
 #include <linux/platform_device.h>
 #include <linux/string.h>
-#include <sound/jack.h>
 #include <sound/simple_card.h>
 #include <sound/soc-dai.h>
 #include <sound/soc.h>
 
-struct asoc_simple_jack {
-	struct snd_soc_jack jack;
-	struct snd_soc_jack_pin pin;
-	struct snd_soc_jack_gpio gpio;
-};
-
 struct simple_card_data {
 	struct snd_soc_card snd_card;
 	struct simple_dai_props {
@@ -49,61 +37,6 @@ struct simple_card_data {
 #define CELL	"#sound-dai-cells"
 #define PREFIX	"simple-audio-card,"
 
-#define asoc_simple_card_init_hp(card, sjack, prefix)\
-	asoc_simple_card_init_jack(card, sjack, 1, prefix)
-#define asoc_simple_card_init_mic(card, sjack, prefix)\
-	asoc_simple_card_init_jack(card, sjack, 0, prefix)
-static int asoc_simple_card_init_jack(struct snd_soc_card *card,
-				      struct asoc_simple_jack *sjack,
-				      int is_hp, char *prefix)
-{
-	struct device *dev = card->dev;
-	enum of_gpio_flags flags;
-	char prop[128];
-	char *pin_name;
-	char *gpio_name;
-	int mask;
-	int det;
-
-	sjack->gpio.gpio = -ENOENT;
-
-	if (is_hp) {
-		snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix);
-		pin_name	= "Headphones";
-		gpio_name	= "Headphone detection";
-		mask		= SND_JACK_HEADPHONE;
-	} else {
-		snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix);
-		pin_name	= "Mic Jack";
-		gpio_name	= "Mic detection";
-		mask		= SND_JACK_MICROPHONE;
-	}
-
-	det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags);
-	if (det == -EPROBE_DEFER)
-		return -EPROBE_DEFER;
-
-	if (gpio_is_valid(det)) {
-		sjack->pin.pin		= pin_name;
-		sjack->pin.mask		= mask;
-
-		sjack->gpio.name	= gpio_name;
-		sjack->gpio.report	= mask;
-		sjack->gpio.gpio	= det;
-		sjack->gpio.invert	= !!(flags & OF_GPIO_ACTIVE_LOW);
-		sjack->gpio.debounce_time = 150;
-
-		snd_soc_card_jack_new(card, pin_name, mask,
-				      &sjack->jack,
-				      &sjack->pin, 1);
-
-		snd_soc_jack_add_gpios(&sjack->jack, 1,
-				       &sjack->gpio);
-	}
-
-	return 0;
-}
-
 static int asoc_simple_card_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -213,14 +146,6 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
 	if (ret < 0)
 		return ret;
 
-	ret = asoc_simple_card_init_hp(rtd->card, &priv->hp_jack, PREFIX);
-	if (ret < 0)
-		return ret;
-
-	ret = asoc_simple_card_init_mic(rtd->card, &priv->mic_jack, PREFIX);
-	if (ret < 0)
-		return ret;
-
 	return 0;
 }
 
@@ -414,6 +339,22 @@ card_parse_end:
 	return ret;
 }
 
+static int asoc_simple_soc_card_probe(struct snd_soc_card *card)
+{
+	struct simple_card_data *priv = snd_soc_card_get_drvdata(card);
+	int ret;
+
+	ret = asoc_simple_card_init_hp(card, &priv->hp_jack, PREFIX);
+	if (ret < 0)
+		return ret;
+
+	ret = asoc_simple_card_init_mic(card, &priv->mic_jack, PREFIX);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
 static int asoc_simple_card_probe(struct platform_device *pdev)
 {
 	struct simple_card_data *priv;
@@ -449,6 +390,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	card->dev		= dev;
 	card->dai_link		= priv->dai_link;
 	card->num_links		= num;
+	card->probe		= asoc_simple_soc_card_probe;
 
 	if (np && of_device_is_available(np)) {
 
diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c
index 487716559deb..16a83bc51e0e 100644
--- a/sound/soc/generic/simple-scu-card.c
+++ b/sound/soc/generic/simple-scu-card.c
@@ -1,15 +1,12 @@
-/*
- * ASoC simple SCU sound card support
- *
- * Copyright (C) 2015 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * based on ${LINUX}/sound/soc/generic/simple-card.c
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ASoC simple SCU sound card support
+//
+// Copyright (C) 2015 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// based on ${LINUX}/sound/soc/generic/simple-card.c
+
 #include <linux/clk.h>
 #include <linux/device.h>
 #include <linux/module.h>
diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c
index 6a8b253c58d2..5455d6e0ab53 100644
--- a/sound/soc/intel/atom/sst/sst_drv_interface.c
+++ b/sound/soc/intel/atom/sst/sst_drv_interface.c
@@ -266,17 +266,15 @@ static int sst_cdev_ack(struct device *dev, unsigned int str_id,
 	stream->cumm_bytes += bytes;
 	dev_dbg(dev, "bytes copied %d inc by %ld\n", stream->cumm_bytes, bytes);
 
-	memcpy_fromio(&fw_tstamp,
-		((void *)(ctx->mailbox + ctx->tstamp)
-		+(str_id * sizeof(fw_tstamp))),
-		sizeof(fw_tstamp));
+	addr =  ((void __iomem *)(ctx->mailbox + ctx->tstamp)) +
+		(str_id * sizeof(fw_tstamp));
+
+	memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp));
 
 	fw_tstamp.bytes_copied = stream->cumm_bytes;
 	dev_dbg(dev, "bytes sent to fw %llu inc by %ld\n",
 			fw_tstamp.bytes_copied, bytes);
 
-	addr =  ((void *)(ctx->mailbox + ctx->tstamp)) +
-			(str_id * sizeof(fw_tstamp));
 	offset =  offsetof(struct snd_sst_tstamp, bytes_copied);
 	sst_shim_write(addr, offset, fw_tstamp.bytes_copied);
 	return 0;
@@ -360,11 +358,12 @@ static int sst_cdev_tstamp(struct device *dev, unsigned int str_id,
 	struct snd_sst_tstamp fw_tstamp = {0,};
 	struct stream_info *stream;
 	struct intel_sst_drv *ctx = dev_get_drvdata(dev);
+	void __iomem *addr;
+
+	addr = (void __iomem *)(ctx->mailbox + ctx->tstamp) +
+		(str_id * sizeof(fw_tstamp));
 
-	memcpy_fromio(&fw_tstamp,
-		((void *)(ctx->mailbox + ctx->tstamp)
-		+(str_id * sizeof(fw_tstamp))),
-		sizeof(fw_tstamp));
+	memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp));
 
 	stream = get_stream_info(ctx, str_id);
 	if (!stream)
@@ -530,6 +529,7 @@ static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info)
 	struct snd_sst_tstamp fw_tstamp;
 	unsigned int str_id;
 	struct intel_sst_drv *ctx = dev_get_drvdata(dev);
+	void __iomem *addr;
 
 	str_id = info->str_id;
 	stream = get_stream_info(ctx, str_id);
@@ -540,10 +540,11 @@ static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info)
 		return -EINVAL;
 	substream = stream->pcm_substream;
 
-	memcpy_fromio(&fw_tstamp,
-		((void *)(ctx->mailbox + ctx->tstamp)
-			+ (str_id * sizeof(fw_tstamp))),
-		sizeof(fw_tstamp));
+	addr = (void __iomem *)(ctx->mailbox + ctx->tstamp) +
+		(str_id * sizeof(fw_tstamp));
+
+	memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp));
+
 	return sst_calc_tstamp(ctx, info, substream, &fw_tstamp);
 }
 
diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c
index a686eef2cf7f..27413ebae956 100644
--- a/sound/soc/intel/atom/sst/sst_loader.c
+++ b/sound/soc/intel/atom/sst/sst_loader.c
@@ -44,15 +44,15 @@ void memcpy32_toio(void __iomem *dst, const void *src, int count)
 	/* __iowrite32_copy uses 32-bit count values so divide by 4 for
 	 * right count in words
 	 */
-	__iowrite32_copy(dst, src, count/4);
+	__iowrite32_copy(dst, src, count / 4);
 }
 
 void memcpy32_fromio(void *dst, const void __iomem *src, int count)
 {
-	/* __iowrite32_copy uses 32-bit count values so divide by 4 for
+	/* __ioread32_copy uses 32-bit count values so divide by 4 for
 	 * right count in words
 	 */
-	__iowrite32_copy(dst, src, count/4);
+	__ioread32_copy(dst, src, count / 4);
 }
 
 /**
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 24797482a3d2..cccda87f4b34 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -281,6 +281,20 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH
 	  Say Y if you have such a device.
 	  If unsure select "N".
 
+config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
+	tristate "GLK with RT5682 and MAX98357A in I2S Mode"
+	depends on MFD_INTEL_LPSS && I2C && ACPI
+	select SND_SOC_RT5682
+	select SND_SOC_MAX98357A
+	select SND_SOC_DMIC
+	select SND_SOC_HDAC_HDMI
+	select SND_HDA_DSP_LOADER
+	help
+	   This adds support for ASoC machine driver for Geminilake platforms
+	   with RT5682 + MAX98357A I2S audio codec.
+	   Say Y if you have such a device.
+	   If unsure select "N".
+
 endif ## SND_SOC_INTEL_SKYLAKE
 
 endif ## SND_SOC_INTEL_MACH
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 92b5507291af..87ef8b4058e5 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -6,6 +6,7 @@ snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o
 snd-soc-sst-broadwell-objs := broadwell.o
 snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o
 snd-soc-sst-bxt-rt298-objs := bxt_rt298.o
+snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o
 snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
 snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o
 snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
@@ -27,6 +28,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o
 obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o
+obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o
 obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
 obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o
diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c
index 6ea360f33575..efcfd906c856 100644
--- a/sound/soc/intel/boards/bdw-rt5677.c
+++ b/sound/soc/intel/boards/bdw-rt5677.c
@@ -154,9 +154,7 @@ static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
 	channels->min = channels->max = 2;
 
 	/* set SSP0 to 16 bit */
-	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
-				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
-				    SNDRV_PCM_FORMAT_S16_LE);
+	params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
 	return 0;
 }
 
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index 40eb979d5ac1..6f052fc8d1e2 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -160,7 +160,7 @@ static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	/* set SSP to 24 bit */
 	snd_mask_none(fmt);
-	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 
 	return 0;
 }
@@ -324,8 +324,22 @@ static const struct snd_pcm_hw_constraint_list constraints_16000 = {
 	.list  = rates_16000,
 };
 
+static const unsigned int ch_mono[] = {
+	1,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_refcap = {
+	.count = ARRAY_SIZE(ch_mono),
+	.list  = ch_mono,
+};
+
 static int broxton_refcap_startup(struct snd_pcm_substream *substream)
 {
+	substream->runtime->hw.channels_max = 1;
+	snd_pcm_hw_constraint_list(substream->runtime, 0,
+				   SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_refcap);
+
 	return snd_pcm_hw_constraint_list(substream->runtime, 0,
 			SNDRV_PCM_HW_PARAM_RATE,
 			&constraints_16000);
@@ -586,7 +600,7 @@ static int broxton_audio_probe(struct platform_device *pdev)
 static struct platform_driver broxton_audio = {
 	.probe = broxton_audio_probe,
 	.driver = {
-		.name = "bxt_da7219_max98357a_i2s",
+		.name = "bxt_da7219_max98357a",
 		.pm = &snd_soc_pm_ops,
 	},
 };
@@ -599,4 +613,4 @@ MODULE_AUTHOR("Rohit Ainapure <rohit.m.ainapure@intel.com>");
 MODULE_AUTHOR("Harsha Priya <harshapriya.n@intel.com>");
 MODULE_AUTHOR("Conrad Cooke <conrad.cooke@intel.com>");
 MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:bxt_da7219_max98357a_i2s");
+MODULE_ALIAS("platform:bxt_da7219_max98357a");
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index b68c289558a8..27308337ab12 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -221,7 +221,7 @@ static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	/* set SSP5 to 24 bit */
 	snd_mask_none(fmt);
-	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 33065ba294a9..d32844f94d74 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -404,7 +404,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 		},
 		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
 					BYT_RT5640_JD_SRC_JD1_IN4P |
-					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_TH_1500UA |
 					BYT_RT5640_OVCD_SF_0P75 |
 					BYT_RT5640_SSP0_AIF1 |
 					BYT_RT5640_MCLK_EN),
@@ -464,12 +464,38 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_MCLK_EN),
 	},
 	{
+		/* Chuwi Vi10 (CWI505) */
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"),
+			DMI_MATCH(DMI_BOARD_NAME, "BYT-PF02"),
+			DMI_MATCH(DMI_SYS_VENDOR, "ilife"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "S165"),
+		},
+		.driver_data = (void *)(BYT_RT5640_IN1_MAP |
+					BYT_RT5640_JD_SRC_JD2_IN4N |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_DIFF_MIC |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
+	{
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"),
 			DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"),
 		},
 		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP),
 	},
+	{	/* Connect Tablet 9 */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Connect"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "Tablet 9"),
+		},
+		.driver_data = (void *)(BYTCR_INPUT_DEFAULTS |
+					BYT_RT5640_MONO_SPEAKER |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{
 		.matches = {
 			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Dell Inc."),
@@ -536,6 +562,19 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_SSP0_AIF1 |
 					BYT_RT5640_MCLK_EN),
 	},
+	{	/* Lenovo Miix 2 8 */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "20326"),
+			DMI_EXACT_MATCH(DMI_BOARD_NAME, "Hiking"),
+		},
+		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
+					BYT_RT5640_JD_SRC_JD2_IN4N |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_MONO_SPEAKER |
+					BYT_RT5640_MCLK_EN),
+	},
 	{	/* MSI S100 tablet */
 		.matches = {
 			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."),
@@ -549,6 +588,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_DIFF_MIC |
 					BYT_RT5640_MCLK_EN),
 	},
+	{	/* Nuvison/TMax TM800W560 */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TMAX"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TM800W560L"),
+		},
+		.driver_data = (void *)(BYT_RT5640_IN1_MAP |
+					BYT_RT5640_JD_SRC_JD2_IN4N |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_JD_NOT_INV |
+					BYT_RT5640_DIFF_MIC |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{	/* Pipo W4 */
 		.matches = {
 			DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"),
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 987720e203f9..f8a68bdb3885 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -26,8 +26,12 @@
 #include <linux/clk.h>
 #include <linux/device.h>
 #include <linux/dmi.h>
+#include <linux/input.h>
+#include <linux/gpio/consumer.h>
+#include <linux/gpio/machine.h>
 #include <linux/slab.h>
 #include <asm/cpu_device_id.h>
+#include <asm/intel-family.h>
 #include <asm/platform_sst_audio.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -42,8 +46,6 @@ enum {
 	BYT_RT5651_IN1_MAP,
 	BYT_RT5651_IN2_MAP,
 	BYT_RT5651_IN1_IN2_MAP,
-	BYT_RT5651_IN1_HS_IN3_MAP,
-	BYT_RT5651_IN2_HS_IN3_MAP,
 };
 
 enum {
@@ -76,21 +78,26 @@ enum {
 #define BYT_RT5651_SSP2_AIF2		BIT(19) /* default is using AIF1  */
 #define BYT_RT5651_SSP0_AIF1		BIT(20)
 #define BYT_RT5651_SSP0_AIF2		BIT(21)
+#define BYT_RT5651_HP_LR_SWAPPED	BIT(22)
+#define BYT_RT5651_MONO_SPEAKER		BIT(23)
+
+#define BYT_RT5651_DEFAULT_QUIRKS	(BYT_RT5651_MCLK_EN | \
+					 BYT_RT5651_JD1_1   | \
+					 BYT_RT5651_OVCD_TH_2000UA | \
+					 BYT_RT5651_OVCD_SF_0P75)
 
 /* jack-detect-source + dmic-en + ovcd-th + -sf + terminating empty entry */
 #define MAX_NO_PROPS 5
 
 struct byt_rt5651_private {
 	struct clk *mclk;
+	struct gpio_desc *ext_amp_gpio;
 	struct snd_soc_jack jack;
 };
 
 /* Default: jack-detect on JD1_1, internal mic on in2, headsetmic on in3 */
-static unsigned long byt_rt5651_quirk = BYT_RT5651_MCLK_EN |
-					BYT_RT5651_JD1_1 |
-					BYT_RT5651_OVCD_TH_2000UA |
-					BYT_RT5651_OVCD_SF_0P75 |
-					BYT_RT5651_IN2_HS_IN3_MAP;
+static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS |
+					BYT_RT5651_IN2_MAP;
 
 static void log_quirks(struct device *dev)
 {
@@ -100,10 +107,8 @@ static void log_quirks(struct device *dev)
 		dev_info(dev, "quirk IN1_MAP enabled");
 	if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP)
 		dev_info(dev, "quirk IN2_MAP enabled");
-	if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_HS_IN3_MAP)
-		dev_info(dev, "quirk IN1_HS_IN3_MAP enabled");
-	if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_HS_IN3_MAP)
-		dev_info(dev, "quirk IN2_HS_IN3_MAP enabled");
+	if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_IN2_MAP)
+		dev_info(dev, "quirk IN1_IN2_MAP enabled");
 	if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) {
 		dev_info(dev, "quirk realtek,jack-detect-source %ld\n",
 			 BYT_RT5651_JDSRC(byt_rt5651_quirk));
@@ -124,6 +129,8 @@ static void log_quirks(struct device *dev)
 		dev_info(dev, "quirk SSP0_AIF1 enabled\n");
 	if (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2)
 		dev_info(dev, "quirk SSP0_AIF2 enabled\n");
+	if (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER)
+		dev_info(dev, "quirk MONO_SPEAKER enabled\n");
 }
 
 #define BYT_CODEC_DAI1	"rt5651-aif1"
@@ -211,6 +218,20 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int rt5651_ext_amp_power_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_card *card = w->dapm->card;
+	struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card);
+
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		gpiod_set_value_cansleep(priv->ext_amp_gpio, 1);
+	else
+		gpiod_set_value_cansleep(priv->ext_amp_gpio, 0);
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = {
 	SND_SOC_DAPM_HP("Headphone", NULL),
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
@@ -220,7 +241,9 @@ static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
 			    platform_clock_control, SND_SOC_DAPM_PRE_PMU |
 			    SND_SOC_DAPM_POST_PMD),
-
+	SND_SOC_DAPM_SUPPLY("Ext Amp Power", SND_SOC_NOPM, 0, 0,
+			    rt5651_ext_amp_power_event,
+			    SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 };
 
 static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = {
@@ -228,6 +251,7 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = {
 	{"Headset Mic", NULL, "Platform Clock"},
 	{"Internal Mic", NULL, "Platform Clock"},
 	{"Speaker", NULL, "Platform Clock"},
+	{"Speaker", NULL, "Ext Amp Power"},
 	{"Line In", NULL, "Platform Clock"},
 
 	{"Headset Mic", NULL, "micbias1"}, /* lowercase for rt5651 */
@@ -241,38 +265,26 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = {
 };
 
 static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic_map[] = {
-	{"IN2P", NULL, "Headset Mic"},
 	{"DMIC L1", NULL, "Internal Mic"},
 	{"DMIC R1", NULL, "Internal Mic"},
+	{"IN3P", NULL, "Headset Mic"},
 };
 
 static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = {
 	{"Internal Mic", NULL, "micbias1"},
 	{"IN1P", NULL, "Internal Mic"},
-	{"IN2P", NULL, "Headset Mic"},
+	{"IN3P", NULL, "Headset Mic"},
 };
 
 static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_map[] = {
 	{"Internal Mic", NULL, "micbias1"},
-	{"IN1P", NULL, "Headset Mic"},
-	{"IN2P", NULL, "Internal Mic"},
-};
-
-static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = {
-	{"Internal Mic", NULL, "micbias1"},
-	{"IN1P", NULL, "Internal Mic"},
 	{"IN2P", NULL, "Internal Mic"},
 	{"IN3P", NULL, "Headset Mic"},
 };
 
-static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_hs_in3_map[] = {
+static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = {
 	{"Internal Mic", NULL, "micbias1"},
 	{"IN1P", NULL, "Internal Mic"},
-	{"IN3P", NULL, "Headset Mic"},
-};
-
-static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_hs_in3_map[] = {
-	{"Internal Mic", NULL, "micbias1"},
 	{"IN2P", NULL, "Internal Mic"},
 	{"IN3P", NULL, "Headset Mic"},
 };
@@ -357,46 +369,72 @@ static int byt_rt5651_quirk_cb(const struct dmi_system_id *id)
 
 static const struct dmi_system_id byt_rt5651_quirk_table[] = {
 	{
+		/* Chuwi Hi8 Pro (CWI513) */
 		.callback = byt_rt5651_quirk_cb,
 		.matches = {
-			DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"),
-			DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"),
+			DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "X1D3_C806N"),
 		},
-		.driver_data = (void *)(BYT_RT5651_IN1_HS_IN3_MAP),
+		.driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
+					BYT_RT5651_IN2_MAP |
+					BYT_RT5651_HP_LR_SWAPPED |
+					BYT_RT5651_MONO_SPEAKER),
 	},
 	{
+		/* Chuwi Vi8 Plus (CWI519) */
 		.callback = byt_rt5651_quirk_cb,
 		.matches = {
-			DMI_MATCH(DMI_SYS_VENDOR, "ADI"),
-			DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"),
+			DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"),
 		},
-		.driver_data = (void *)(BYT_RT5651_MCLK_EN |
-					BYT_RT5651_IN1_HS_IN3_MAP),
+		.driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
+					BYT_RT5651_IN2_MAP |
+					BYT_RT5651_HP_LR_SWAPPED |
+					BYT_RT5651_MONO_SPEAKER),
+	},
+	{
+		/* I.T.Works TW701, Ployer Momo7w and Trekstor ST70416-6
+		 * (these all use the same mainboard) */
+		.callback = byt_rt5651_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_BIOS_VENDOR, "INSYDE Corp."),
+			/* Partial match for all of itWORKS.G.WI71C.JGBMRBA,
+			 * TREK.G.WI71C.JGBMRBA0x and MOMO.G.WI71C.MABMRBA02 */
+			DMI_MATCH(DMI_BIOS_VERSION, ".G.WI71C."),
+		},
+		.driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
+					BYT_RT5651_IN2_MAP |
+					BYT_RT5651_SSP0_AIF1 |
+					BYT_RT5651_MONO_SPEAKER),
 	},
 	{
+		/* KIANO SlimNote 14.2 */
 		.callback = byt_rt5651_quirk_cb,
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "KIANO"),
 			DMI_MATCH(DMI_PRODUCT_NAME, "KIANO SlimNote 14.2"),
 		},
-		.driver_data = (void *)(BYT_RT5651_MCLK_EN |
-					BYT_RT5651_JD1_1 |
-					BYT_RT5651_OVCD_TH_2000UA |
-					BYT_RT5651_OVCD_SF_0P75 |
+		.driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
 					BYT_RT5651_IN1_IN2_MAP),
 	},
 	{
-		/* Chuwi Vi8 Plus (CWI519) */
+		/* Minnowboard Max B3 */
 		.callback = byt_rt5651_quirk_cb,
 		.matches = {
-			DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"),
-			DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"),
+			DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"),
+		},
+		.driver_data = (void *)(BYT_RT5651_IN1_MAP),
+	},
+	{
+		/* Minnowboard Turbot */
+		.callback = byt_rt5651_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "ADI"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"),
 		},
 		.driver_data = (void *)(BYT_RT5651_MCLK_EN |
-					BYT_RT5651_JD1_1 |
-					BYT_RT5651_OVCD_TH_2000UA |
-					BYT_RT5651_OVCD_SF_0P75 |
-					BYT_RT5651_IN2_HS_IN3_MAP),
+					BYT_RT5651_IN1_MAP),
 	},
 	{
 		/* VIOS LTH17 */
@@ -405,11 +443,24 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = {
 			DMI_MATCH(DMI_SYS_VENDOR, "VIOS"),
 			DMI_MATCH(DMI_PRODUCT_NAME, "LTH17"),
 		},
-		.driver_data = (void *)(BYT_RT5651_MCLK_EN |
+		.driver_data = (void *)(BYT_RT5651_IN1_IN2_MAP |
 					BYT_RT5651_JD1_1 |
 					BYT_RT5651_OVCD_TH_2000UA |
 					BYT_RT5651_OVCD_SF_1P0 |
-					BYT_RT5651_IN1_IN2_MAP),
+					BYT_RT5651_MCLK_EN),
+	},
+	{
+		/* Yours Y8W81 (and others using the same mainboard) */
+		.callback = byt_rt5651_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_BIOS_VENDOR, "INSYDE Corp."),
+			/* Partial match for all devs with a W86C mainboard */
+			DMI_MATCH(DMI_BIOS_VERSION, ".F.W86C."),
+		},
+		.driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
+					BYT_RT5651_IN2_MAP |
+					BYT_RT5651_SSP0_AIF1 |
+					BYT_RT5651_MONO_SPEAKER),
 	},
 	{}
 };
@@ -418,15 +469,10 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = {
  * Note this MUST be called before snd_soc_register_card(), so that the props
  * are in place before the codec component driver's probe function parses them.
  */
-static int byt_rt5651_add_codec_device_props(const char *i2c_dev_name)
+static int byt_rt5651_add_codec_device_props(struct device *i2c_dev)
 {
 	struct property_entry props[MAX_NO_PROPS] = {};
-	struct device *i2c_dev;
-	int ret, cnt = 0;
-
-	i2c_dev = bus_find_device_by_name(&i2c_bus_type, NULL, i2c_dev_name);
-	if (!i2c_dev)
-		return -EPROBE_DEFER;
+	int cnt = 0;
 
 	props[cnt++] = PROPERTY_ENTRY_U32("realtek,jack-detect-source",
 				BYT_RT5651_JDSRC(byt_rt5651_quirk));
@@ -440,10 +486,7 @@ static int byt_rt5651_add_codec_device_props(const char *i2c_dev_name)
 	if (byt_rt5651_quirk & BYT_RT5651_DMIC_EN)
 		props[cnt++] = PROPERTY_ENTRY_BOOL("realtek,dmic-en");
 
-	ret = device_add_properties(i2c_dev, props);
-	put_device(i2c_dev);
-
-	return ret;
+	return device_add_properties(i2c_dev, props);
 }
 
 static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
@@ -475,14 +518,6 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
 		custom_map = byt_rt5651_intmic_in1_in2_map;
 		num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_in2_map);
 		break;
-	case BYT_RT5651_IN1_HS_IN3_MAP:
-		custom_map = byt_rt5651_intmic_in1_hs_in3_map;
-		num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_hs_in3_map);
-		break;
-	case BYT_RT5651_IN2_HS_IN3_MAP:
-		custom_map = byt_rt5651_intmic_in2_hs_in3_map;
-		num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_hs_in3_map);
-		break;
 	default:
 		custom_map = byt_rt5651_intmic_dmic_map;
 		num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map);
@@ -546,13 +581,17 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
 
 	if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) {
 		ret = snd_soc_card_jack_new(runtime->card, "Headset",
-				    SND_JACK_HEADSET, &priv->jack,
-				    bytcr_jack_pins, ARRAY_SIZE(bytcr_jack_pins));
+				    SND_JACK_HEADSET | SND_JACK_BTN_0,
+				    &priv->jack, bytcr_jack_pins,
+				    ARRAY_SIZE(bytcr_jack_pins));
 		if (ret) {
 			dev_err(runtime->dev, "jack creation failed %d\n", ret);
 			return ret;
 		}
 
+		snd_jack_set_key(priv->jack.jack, SND_JACK_BTN_0,
+				 KEY_PLAYPAUSE);
+
 		ret = snd_soc_component_set_jack(codec, &priv->jack, NULL);
 		if (ret)
 			return ret;
@@ -691,6 +730,48 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = {
 };
 
 /* SoC card */
+static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN];
+static char byt_rt5651_codec_aif_name[12]; /*  = "rt5651-aif[1|2]" */
+static char byt_rt5651_cpu_dai_name[10]; /*  = "ssp[0|2]-port" */
+static char byt_rt5651_long_name[50]; /* = "bytcr-rt5651-*-spk-*-mic[-swapped-hp]" */
+
+static int byt_rt5651_suspend(struct snd_soc_card *card)
+{
+	struct snd_soc_component *component;
+
+	if (!BYT_RT5651_JDSRC(byt_rt5651_quirk))
+		return 0;
+
+	list_for_each_entry(component, &card->component_dev_list, card_list) {
+		if (!strcmp(component->name, byt_rt5651_codec_name)) {
+			dev_dbg(component->dev, "disabling jack detect before suspend\n");
+			snd_soc_component_set_jack(component, NULL, NULL);
+			break;
+		}
+	}
+
+	return 0;
+}
+
+static int byt_rt5651_resume(struct snd_soc_card *card)
+{
+	struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_component *component;
+
+	if (!BYT_RT5651_JDSRC(byt_rt5651_quirk))
+		return 0;
+
+	list_for_each_entry(component, &card->component_dev_list, card_list) {
+		if (!strcmp(component->name, byt_rt5651_codec_name)) {
+			dev_dbg(component->dev, "re-enabling jack detect after resume\n");
+			snd_soc_component_set_jack(component, &priv->jack, NULL);
+			break;
+		}
+	}
+
+	return 0;
+}
+
 static struct snd_soc_card byt_rt5651_card = {
 	.name = "bytcr-rt5651",
 	.owner = THIS_MODULE,
@@ -701,23 +782,86 @@ static struct snd_soc_card byt_rt5651_card = {
 	.dapm_routes = byt_rt5651_audio_map,
 	.num_dapm_routes = ARRAY_SIZE(byt_rt5651_audio_map),
 	.fully_routed = true,
+	.suspend_pre = byt_rt5651_suspend,
+	.resume_post = byt_rt5651_resume,
 };
 
-static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN];
-static char byt_rt5651_codec_aif_name[12]; /*  = "rt5651-aif[1|2]" */
-static char byt_rt5651_cpu_dai_name[10]; /*  = "ssp[0|2]-port" */
-static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-spk-*-mic" */
+static const struct x86_cpu_id baytrail_cpu_ids[] = {
+	{ X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_SILVERMONT1 }, /* Valleyview */
+	{}
+};
+
+static const struct x86_cpu_id cherrytrail_cpu_ids[] = {
+	{ X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_AIRMONT },     /* Braswell */
+	{}
+};
+
+static const struct acpi_gpio_params first_gpio = { 0, 0, false };
+static const struct acpi_gpio_params second_gpio = { 1, 0, false };
+
+static const struct acpi_gpio_mapping byt_rt5651_amp_en_first[] = {
+	{ "ext-amp-enable-gpios", &first_gpio, 1 },
+	{ },
+};
 
-static bool is_valleyview(void)
+static const struct acpi_gpio_mapping byt_rt5651_amp_en_second[] = {
+	{ "ext-amp-enable-gpios", &second_gpio, 1 },
+	{ },
+};
+
+/*
+ * Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other
+ * boards may  have I2cSerialBusV2, GpioInt, GpioIo instead. We want the
+ * GpioIo one for the ext-amp-enable-gpio and both count for the index in
+ * acpi_gpio_params index.  So we have 2 different mappings and the code
+ * below figures out which one to use.
+ */
+struct byt_rt5651_acpi_resource_data {
+	int gpio_count;
+	int gpio_int_idx;
+};
+
+static int snd_byt_rt5651_acpi_resource(struct acpi_resource *ares, void *arg)
 {
-	static const struct x86_cpu_id cpu_ids[] = {
-		{ X86_VENDOR_INTEL, 6, 55 }, /* Valleyview, Bay Trail */
-		{}
-	};
-
-	if (!x86_match_cpu(cpu_ids))
-		return false;
-	return true;
+	struct byt_rt5651_acpi_resource_data *data = arg;
+
+	if (ares->type != ACPI_RESOURCE_TYPE_GPIO)
+		return 0;
+
+	if (ares->data.gpio.connection_type == ACPI_RESOURCE_GPIO_TYPE_INT)
+		data->gpio_int_idx = data->gpio_count;
+
+	data->gpio_count++;
+	return 0;
+}
+
+static void snd_byt_rt5651_mc_add_amp_en_gpio_mapping(struct device *codec)
+{
+	struct byt_rt5651_acpi_resource_data data = { 0, -1 };
+	LIST_HEAD(resources);
+	int ret;
+
+	ret = acpi_dev_get_resources(ACPI_COMPANION(codec), &resources,
+				     snd_byt_rt5651_acpi_resource, &data);
+	if (ret < 0) {
+		dev_warn(codec, "Failed to get ACPI resources, not adding external amplifier GPIO mapping\n");
+		return;
+	}
+
+	/* All info we need is gathered during the walk */
+	acpi_dev_free_resource_list(&resources);
+
+	switch (data.gpio_int_idx) {
+	case 0:
+		devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_second);
+		break;
+	case 1:
+		devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_first);
+		break;
+	default:
+		dev_warn(codec, "Unknown GpioInt index %d, not adding external amplifier GPIO mapping\n",
+			 data.gpio_int_idx);
+	}
 }
 
 struct acpi_chan_package {   /* ACPICA seems to require 64 bit integers */
@@ -727,13 +871,12 @@ struct acpi_chan_package {   /* ACPICA seems to require 64 bit integers */
 
 static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 {
-	const char * const intmic_name[] =
-		{ "dmic", "in1", "in2", "in12", "in1", "in2" };
-	const char * const hsmic_name[] =
-		{  "in2", "in2", "in1",  "in3", "in3", "in3" };
+	const char * const mic_name[] = { "dmic", "in1", "in2", "in12" };
 	struct byt_rt5651_private *priv;
 	struct snd_soc_acpi_mach *mach;
+	struct device *codec_dev;
 	const char *i2c_name = NULL;
+	const char *hp_swapped;
 	bool is_bytcr = false;
 	int ret_val = 0;
 	int dai_index = 0;
@@ -767,11 +910,16 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 		"%s%s", "i2c-", i2c_name);
 	byt_rt5651_dais[dai_index].codec_name = byt_rt5651_codec_name;
 
+	codec_dev = bus_find_device_by_name(&i2c_bus_type, NULL,
+					    byt_rt5651_codec_name);
+	if (!codec_dev)
+		return -EPROBE_DEFER;
+
 	/*
 	 * swap SSP0 if bytcr is detected
 	 * (will be overridden if DMI quirk is detected)
 	 */
-	if (is_valleyview()) {
+	if (x86_match_cpu(baytrail_cpu_ids)) {
 		struct sst_platform_info *p_info = mach->pdata;
 		const struct sst_res_info *res_info = p_info->res_info;
 
@@ -830,9 +978,37 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 	dmi_check_system(byt_rt5651_quirk_table);
 
 	/* Must be called before register_card, also see declaration comment. */
-	ret_val = byt_rt5651_add_codec_device_props(byt_rt5651_codec_name);
-	if (ret_val)
+	ret_val = byt_rt5651_add_codec_device_props(codec_dev);
+	if (ret_val) {
+		put_device(codec_dev);
 		return ret_val;
+	}
+
+	/* Cherry Trail devices use an external amplifier enable gpio */
+	if (x86_match_cpu(cherrytrail_cpu_ids)) {
+		snd_byt_rt5651_mc_add_amp_en_gpio_mapping(codec_dev);
+		priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child(
+						&pdev->dev, "ext-amp-enable", 0,
+						codec_dev->fwnode,
+						GPIOD_OUT_LOW, "speaker-amp");
+		if (IS_ERR(priv->ext_amp_gpio)) {
+			ret_val = PTR_ERR(priv->ext_amp_gpio);
+			switch (ret_val) {
+			case -ENOENT:
+				priv->ext_amp_gpio = NULL;
+				break;
+			default:
+				dev_err(&pdev->dev, "Failed to get ext-amp-enable GPIO: %d\n",
+					ret_val);
+				/* fall through */
+			case -EPROBE_DEFER:
+				put_device(codec_dev);
+				return ret_val;
+			}
+		}
+	}
+
+	put_device(codec_dev);
 
 	log_quirks(&pdev->dev);
 
@@ -876,10 +1052,16 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 		}
 	}
 
+	if (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED)
+		hp_swapped = "-hp-swapped";
+	else
+		hp_swapped = "";
+
 	snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name),
-		 "bytcr-rt5651-%s-intmic-%s-hsmic",
-		 intmic_name[BYT_RT5651_MAP(byt_rt5651_quirk)],
-		 hsmic_name[BYT_RT5651_MAP(byt_rt5651_quirk)]);
+		 "bytcr-rt5651-%s-spk-%s-mic%s",
+		 (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ?
+			"mono" : "stereo",
+		 mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], hp_swapped);
 	byt_rt5651_card.long_name = byt_rt5651_long_name;
 
 	ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_rt5651_card);
diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c
new file mode 100644
index 000000000000..c4b94e2617c5
--- /dev/null
+++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c
@@ -0,0 +1,643 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2018 Intel Corporation.
+
+/*
+ * Intel Geminilake I2S Machine Driver with MAX98357A & RT5682 Codecs
+ *
+ * Modified from:
+ *   Intel Apollolake I2S Machine driver
+ */
+
+#include <linux/input.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../skylake/skl.h"
+#include "../../codecs/rt5682.h"
+#include "../../codecs/hdac_hdmi.h"
+
+/* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */
+#define GLK_PLAT_CLK_FREQ 19200000
+#define RT5682_PLL_FREQ (48000 * 512)
+#define GLK_REALTEK_CODEC_DAI "rt5682-aif1"
+#define GLK_MAXIM_CODEC_DAI "HiFi"
+#define MAXIM_DEV0_NAME "MX98357A:00"
+#define DUAL_CHANNEL 2
+#define QUAD_CHANNEL 4
+#define NAME_SIZE 32
+
+static struct snd_soc_jack geminilake_hdmi[3];
+
+struct glk_hdmi_pcm {
+	struct list_head head;
+	struct snd_soc_dai *codec_dai;
+	int device;
+};
+
+struct glk_card_private {
+	struct snd_soc_jack geminilake_headset;
+	struct list_head hdmi_pcm_list;
+};
+
+enum {
+	GLK_DPCM_AUDIO_PB = 0,
+	GLK_DPCM_AUDIO_CP,
+	GLK_DPCM_AUDIO_HS_PB,
+	GLK_DPCM_AUDIO_ECHO_REF_CP,
+	GLK_DPCM_AUDIO_REF_CP,
+	GLK_DPCM_AUDIO_DMIC_CP,
+	GLK_DPCM_AUDIO_HDMI1_PB,
+	GLK_DPCM_AUDIO_HDMI2_PB,
+	GLK_DPCM_AUDIO_HDMI3_PB,
+};
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+					struct snd_kcontrol *k, int  event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct snd_soc_card *card = dapm->card;
+	struct snd_soc_dai *codec_dai;
+	int ret = 0;
+
+	codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI);
+	if (!codec_dai) {
+		dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n");
+		return -EIO;
+	}
+
+	if (SND_SOC_DAPM_EVENT_OFF(event)) {
+		ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
+		if (ret)
+			dev_err(card->dev, "failed to stop sysclk: %d\n", ret);
+	} else if (SND_SOC_DAPM_EVENT_ON(event)) {
+		ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK,
+					GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ);
+		if (ret < 0) {
+			dev_err(card->dev, "can't set codec pll: %d\n", ret);
+			return ret;
+		}
+	}
+
+	if (ret)
+		dev_err(card->dev, "failed to start internal clk: %d\n", ret);
+
+	return ret;
+}
+
+static const struct snd_kcontrol_new geminilake_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Spk"),
+};
+
+static const struct snd_soc_dapm_widget geminilake_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_SPK("Spk", NULL),
+	SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+	SND_SOC_DAPM_SPK("HDMI1", NULL),
+	SND_SOC_DAPM_SPK("HDMI2", NULL),
+	SND_SOC_DAPM_SPK("HDMI3", NULL),
+	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+			platform_clock_control, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route geminilake_map[] = {
+	/* HP jack connectors - unknown if we have jack detection */
+	{ "Headphone Jack", NULL, "Platform Clock" },
+	{ "Headphone Jack", NULL, "HPOL" },
+	{ "Headphone Jack", NULL, "HPOR" },
+
+	/* speaker */
+	{ "Spk", NULL, "Speaker" },
+
+	/* other jacks */
+	{ "Headset Mic", NULL, "Platform Clock" },
+	{ "IN1P", NULL, "Headset Mic" },
+
+	/* digital mics */
+	{ "DMic", NULL, "SoC DMIC" },
+
+	/* CODEC BE connections */
+	{ "HiFi Playback", NULL, "ssp1 Tx" },
+	{ "ssp1 Tx", NULL, "codec0_out" },
+
+	{ "AIF1 Playback", NULL, "ssp2 Tx" },
+	{ "ssp2 Tx", NULL, "codec1_out" },
+
+	{ "codec0_in", NULL, "ssp2 Rx" },
+	{ "ssp2 Rx", NULL, "AIF1 Capture" },
+
+	{ "HDMI1", NULL, "hif5-0 Output" },
+	{ "HDMI2", NULL, "hif6-0 Output" },
+	{ "HDMI2", NULL, "hif7-0 Output" },
+
+	{ "hifi3", NULL, "iDisp3 Tx" },
+	{ "iDisp3 Tx", NULL, "iDisp3_out" },
+	{ "hifi2", NULL, "iDisp2 Tx" },
+	{ "iDisp2 Tx", NULL, "iDisp2_out" },
+	{ "hifi1", NULL, "iDisp1 Tx" },
+	{ "iDisp1 Tx", NULL, "iDisp1_out" },
+
+	/* DMIC */
+	{ "dmic01_hifi", NULL, "DMIC01 Rx" },
+	{ "DMIC01 Rx", NULL, "DMIC AIF" },
+};
+
+static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_CHANNELS);
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	/* The ADSP will convert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = DUAL_CHANNEL;
+
+	/* set SSP to 24 bit */
+	snd_mask_none(fmt);
+	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+
+	return 0;
+}
+
+static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_jack *jack;
+	int ret;
+
+	/* Configure sysclk for codec */
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1,
+					RT5682_PLL_FREQ, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret);
+
+	/*
+	 * Headset buttons map to the google Reference headset.
+	 * These can be configured by userspace.
+	 */
+	ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
+			SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+			SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT,
+			&ctx->geminilake_headset, NULL, 0);
+	if (ret) {
+		dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
+		return ret;
+	}
+
+	jack = &ctx->geminilake_headset;
+
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+	ret = snd_soc_component_set_jack(component, jack, NULL);
+
+	if (ret) {
+		dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret);
+		return ret;
+	}
+
+	return ret;
+};
+
+static int geminilake_rt5682_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;
+
+	/* Set valid bitmask & configuration for I2S in 24 bit */
+	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, 24);
+	if (ret < 0) {
+		dev_err(rtd->dev, "set TDM slot err:%d\n", ret);
+		return ret;
+	}
+
+	return ret;
+}
+
+static struct snd_soc_ops geminilake_rt5682_ops = {
+	.hw_params = geminilake_rt5682_hw_params,
+};
+
+static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct glk_hdmi_pcm *pcm;
+
+	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
+
+	pcm->device = GLK_DPCM_AUDIO_HDMI1_PB + dai->id;
+	pcm->codec_dai = dai;
+
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
+}
+
+static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_component *component = rtd->cpu_dai->component;
+	struct snd_soc_dapm_context *dapm;
+	int ret;
+
+	dapm = snd_soc_component_get_dapm(component);
+	ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
+	if (ret) {
+		dev_err(rtd->dev, "Ref Cap ignore suspend failed %d\n", ret);
+		return ret;
+	}
+
+	return ret;
+}
+
+static const unsigned int rates[] = {
+	48000,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_rates = {
+	.count = ARRAY_SIZE(rates),
+	.list  = rates,
+	.mask = 0,
+};
+
+static const unsigned int channels[] = {
+	DUAL_CHANNEL,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_channels = {
+	.count = ARRAY_SIZE(channels),
+	.list = channels,
+	.mask = 0,
+};
+
+static unsigned int channels_quad[] = {
+	QUAD_CHANNEL,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_channels_quad = {
+	.count = ARRAY_SIZE(channels_quad),
+	.list = channels_quad,
+	.mask = 0,
+};
+
+static int geminilake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
+		struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *channels = hw_param_interval(params,
+				SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/*
+	 * set BE channel constraint as user FE channels
+	 */
+	channels->min = channels->max = 4;
+
+	return 0;
+}
+
+static int geminilake_dmic_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+			&constraints_channels_quad);
+
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+}
+
+static const struct snd_soc_ops geminilake_dmic_ops = {
+	.startup = geminilake_dmic_startup,
+};
+
+static const unsigned int rates_16000[] = {
+	16000,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_16000 = {
+	.count = ARRAY_SIZE(rates_16000),
+	.list  = rates_16000,
+};
+
+static int geminilake_refcap_startup(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+				SNDRV_PCM_HW_PARAM_RATE,
+				&constraints_16000);
+};
+
+static const struct snd_soc_ops geminilake_refcap_ops = {
+	.startup = geminilake_refcap_startup,
+};
+
+/* geminilake digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link geminilake_dais[] = {
+	/* Front End DAI links */
+	[GLK_DPCM_AUDIO_PB] = {
+		.name = "Glk Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:0e.0",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.nonatomic = 1,
+		.init = geminilake_rt5682_fe_init,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	[GLK_DPCM_AUDIO_CP] = {
+		.name = "Glk Audio Capture Port",
+		.stream_name = "Audio Record",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:0e.0",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.nonatomic = 1,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+	},
+	[GLK_DPCM_AUDIO_HS_PB] = {
+		.name = "Glk Audio Headset Playback",
+		.stream_name = "Headset Audio",
+		.cpu_dai_name = "System Pin2",
+		.platform_name = "0000:00:0e.0",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.dpcm_playback = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[GLK_DPCM_AUDIO_ECHO_REF_CP] = {
+		.name = "Glk Audio Echo Reference cap",
+		.stream_name = "Echoreference Capture",
+		.cpu_dai_name = "Echoref Pin",
+		.platform_name = "0000:00:0e.0",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.init = NULL,
+		.capture_only = 1,
+		.nonatomic = 1,
+	},
+	[GLK_DPCM_AUDIO_REF_CP] = {
+		.name = "Glk Audio Reference cap",
+		.stream_name = "Refcap",
+		.cpu_dai_name = "Reference Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:0e.0",
+		.init = NULL,
+		.dpcm_capture = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+		.ops = &geminilake_refcap_ops,
+	},
+	[GLK_DPCM_AUDIO_DMIC_CP] = {
+		.name = "Glk Audio DMIC cap",
+		.stream_name = "dmiccap",
+		.cpu_dai_name = "DMIC Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:0e.0",
+		.init = NULL,
+		.dpcm_capture = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+		.ops = &geminilake_dmic_ops,
+	},
+	[GLK_DPCM_AUDIO_HDMI1_PB] = {
+		.name = "Glk HDMI Port1",
+		.stream_name = "Hdmi1",
+		.cpu_dai_name = "HDMI1 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:0e.0",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[GLK_DPCM_AUDIO_HDMI2_PB] =	{
+		.name = "Glk HDMI Port2",
+		.stream_name = "Hdmi2",
+		.cpu_dai_name = "HDMI2 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:0e.0",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[GLK_DPCM_AUDIO_HDMI3_PB] =	{
+		.name = "Glk HDMI Port3",
+		.stream_name = "Hdmi3",
+		.cpu_dai_name = "HDMI3 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:0e.0",
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+		.init = NULL,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	/* Back End DAI links */
+	{
+		/* SSP1 - Codec */
+		.name = "SSP1-Codec",
+		.id = 0,
+		.cpu_dai_name = "SSP1 Pin",
+		.platform_name = "0000:00:0e.0",
+		.no_pcm = 1,
+		.codec_name = MAXIM_DEV0_NAME,
+		.codec_dai_name = GLK_MAXIM_CODEC_DAI,
+		.dai_fmt = SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = geminilake_ssp_fixup,
+		.dpcm_playback = 1,
+	},
+	{
+		/* SSP2 - Codec */
+		.name = "SSP2-Codec",
+		.id = 1,
+		.cpu_dai_name = "SSP2 Pin",
+		.platform_name = "0000:00:0e.0",
+		.no_pcm = 1,
+		.codec_name = "i2c-10EC5682:00",
+		.codec_dai_name = GLK_REALTEK_CODEC_DAI,
+		.init = geminilake_rt5682_codec_init,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = geminilake_ssp_fixup,
+		.ops = &geminilake_rt5682_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	{
+		.name = "dmic01",
+		.id = 2,
+		.cpu_dai_name = "DMIC01 Pin",
+		.codec_name = "dmic-codec",
+		.codec_dai_name = "dmic-hifi",
+		.platform_name = "0000:00:0e.0",
+		.ignore_suspend = 1,
+		.be_hw_params_fixup = geminilake_dmic_fixup,
+		.dpcm_capture = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp1",
+		.id = 3,
+		.cpu_dai_name = "iDisp1 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi1",
+		.platform_name = "0000:00:0e.0",
+		.init = geminilake_hdmi_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp2",
+		.id = 4,
+		.cpu_dai_name = "iDisp2 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi2",
+		.platform_name = "0000:00:0e.0",
+		.init = geminilake_hdmi_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp3",
+		.id = 5,
+		.cpu_dai_name = "iDisp3 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi3",
+		.platform_name = "0000:00:0e.0",
+		.init = geminilake_hdmi_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+};
+
+static int glk_card_late_probe(struct snd_soc_card *card)
+{
+	struct glk_card_private *ctx = snd_soc_card_get_drvdata(card);
+	struct snd_soc_component *component = NULL;
+	char jack_name[NAME_SIZE];
+	struct glk_hdmi_pcm *pcm;
+	int err = 0;
+	int i = 0;
+
+	list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+		component = pcm->codec_dai->component;
+		snprintf(jack_name, sizeof(jack_name),
+			"HDMI/DP, pcm=%d Jack", pcm->device);
+		err = snd_soc_card_jack_new(card, jack_name,
+					SND_JACK_AVOUT, &geminilake_hdmi[i],
+					NULL, 0);
+
+		if (err)
+			return err;
+
+		err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
+						&geminilake_hdmi[i]);
+		if (err < 0)
+			return err;
+
+		i++;
+	}
+
+	if (!component)
+		return -EINVAL;
+
+	return hdac_hdmi_jack_port_init(component, &card->dapm);
+}
+
+/* geminilake audio machine driver for SPT + RT5682 */
+static struct snd_soc_card glk_audio_card_rt5682_m98357a = {
+	.name = "glkrt5682max",
+	.owner = THIS_MODULE,
+	.dai_link = geminilake_dais,
+	.num_links = ARRAY_SIZE(geminilake_dais),
+	.controls = geminilake_controls,
+	.num_controls = ARRAY_SIZE(geminilake_controls),
+	.dapm_widgets = geminilake_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(geminilake_widgets),
+	.dapm_routes = geminilake_map,
+	.num_dapm_routes = ARRAY_SIZE(geminilake_map),
+	.fully_routed = true,
+	.late_probe = glk_card_late_probe,
+};
+
+static int geminilake_audio_probe(struct platform_device *pdev)
+{
+	struct glk_card_private *ctx;
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+	if (!ctx)
+		return -ENOMEM;
+
+	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
+	glk_audio_card_rt5682_m98357a.dev = &pdev->dev;
+	snd_soc_card_set_drvdata(&glk_audio_card_rt5682_m98357a, ctx);
+
+	return devm_snd_soc_register_card(&pdev->dev,
+					&glk_audio_card_rt5682_m98357a);
+}
+
+static const struct platform_device_id glk_board_ids[] = {
+	{
+		.name = "glk_rt5682_max98357a",
+		.driver_data =
+			(kernel_ulong_t)&glk_audio_card_rt5682_m98357a,
+	},
+	{ }
+};
+
+static struct platform_driver geminilake_audio = {
+	.probe = geminilake_audio_probe,
+	.driver = {
+		.name = "glk_rt5682_max98357a",
+		.pm = &snd_soc_pm_ops,
+	},
+	.id_table = glk_board_ids,
+};
+module_platform_driver(geminilake_audio)
+
+/* Module information */
+MODULE_DESCRIPTION("Geminilake Audio Machine driver-RT5682 & MAX98357A in I2S mode");
+MODULE_AUTHOR("Naveen Manohar <naveen.m@intel.com>");
+MODULE_AUTHOR("Harsha Priya <harshapriya.n@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:glk_rt5682_max98357a");
diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c
index 94294c27d1db..38f6ab74709d 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98357a.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c
@@ -152,7 +152,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	/* set SSP to 24 bit */
 	snd_mask_none(fmt);
-	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 
 	return 0;
 }
@@ -380,6 +380,7 @@ static struct snd_soc_dai_link kabylake_dais[] = {
 		.trigger = {
 			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
 		.dpcm_capture = 1,
+		.ops = &kabylake_da7219_fe_ops,
 	},
 	[KBL_DPCM_AUDIO_DMIC_CP] = {
 		.name = "Kbl Audio DMIC cap",
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 3a61252fe450..21a6490746a6 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -434,14 +434,14 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 		rate->min = rate->max = 48000;
 		channels->min = channels->max = 2;
 		snd_mask_none(fmt);
-		snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 	}
 	/*
 	 * The speaker on the SSP0 supports S16_LE and not S24_LE.
 	 * thus changing the mask here
 	 */
 	if (!strcmp(be_dai_link->name, "SSP0-Codec"))
-		snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE);
+		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 92f5fb2ae0a3..a892b37eab7c 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -307,7 +307,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 		rate->min = rate->max = 48000;
 		channels->min = channels->max = 2;
 		snd_mask_none(fmt);
-		snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 	} else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) {
 		if (params_channels(params) == 2 ||
 				DMIC_CH(dmic_constraints) == 2)
@@ -320,7 +320,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	 * thus changing the mask here
 	 */
 	if (!strcmp(be_dai_link->name, "SSP0-Codec"))
-		snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE);
+		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
index 3ff6646cfa21..d31482b8c9bb 100644
--- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c
+++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
@@ -157,7 +157,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	/* set SSP0 to 24 bit */
 	snd_mask_none(fmt);
-	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
index b0610bba3cfa..e877bb60beb1 100644
--- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
+++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
@@ -346,7 +346,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	/* set SSP0 to 24 bit */
 	snd_mask_none(fmt);
-	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 	return 0;
 }
 
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index 38a1495c29cf..0e1818dd4cc6 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -229,7 +229,7 @@ static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	/* set SSP0 to 24 bit */
 	snd_mask_none(fmt);
-	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 	return 0;
 }
 
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index 7379d8830c39..915a34cdc8ac 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -3,7 +3,11 @@ snd-soc-sst-dsp-objs := sst-dsp.o
 snd-soc-sst-acpi-objs := sst-acpi.o
 snd-soc-sst-ipc-objs := sst-ipc.o
 snd-soc-sst-firmware-objs := sst-firmware.o
-snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o soc-acpi-intel-hsw-bdw-match.o
+snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o \
+	soc-acpi-intel-hsw-bdw-match.o \
+	soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \
+	soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \
+	soc-acpi-intel-cnl-match.o
 
 obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
 obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o
diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
new file mode 100644
index 000000000000..f39386e540d3
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
@@ -0,0 +1,59 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-bxt-match.c - tables and support for BXT ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+
+static struct snd_soc_acpi_codecs bxt_codecs = {
+	.num_codecs = 1,
+	.codecs = {"MX98357A"}
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = {
+	{
+		.id = "INT343A",
+		.drv_name = "bxt_alc298s_i2s",
+		.fw_filename = "intel/dsp_fw_bxtn.bin",
+	},
+	{
+		.id = "DLGS7219",
+		.drv_name = "bxt_da7219_max98357a",
+		.fw_filename = "intel/dsp_fw_bxtn.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &bxt_codecs,
+		.sof_fw_filename = "intel/sof-apl.ri",
+		.sof_tplg_filename = "intel/sof-apl-da7219.tplg",
+		.asoc_plat_name = "0000:00:0e.0",
+	},
+	{
+		.id = "104C5122",
+		.drv_name = "bxt-pcm512x",
+		.sof_fw_filename = "intel/sof-apl.ri",
+		.sof_tplg_filename = "intel/sof-apl-pcm512x.tplg",
+		.asoc_plat_name = "0000:00:0e.0",
+	},
+	{
+		.id = "1AEC8804",
+		.drv_name = "bxt-wm8804",
+		.sof_fw_filename = "intel/sof-apl.ri",
+		.sof_tplg_filename = "intel/sof-apl-wm8804.tplg",
+		.asoc_plat_name = "0000:00:0e.0",
+	},
+	{
+		.id = "INT34C3",
+		.drv_name = "bxt_tdf8532",
+		.sof_fw_filename = "intel/sof-apl.ri",
+		.sof_tplg_filename = "intel/sof-apl-tdf8532.tplg",
+		.asoc_plat_name = "0000:00:0e.0",
+	},
+	{},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_bxt_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
index bfe1ca68a542..4daa8a4f0c0c 100644
--- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
@@ -59,8 +59,8 @@ static struct snd_soc_acpi_mach byt_thinkpad_10 = {
 	.drv_name = "cht-bsw-rt5672",
 	.fw_filename = "intel/fw_sst_0f28.bin",
 	.board = "cht-bsw",
-	.sof_fw_filename = "intel/reef-byt.ri",
-	.sof_tplg_filename = "intel/reef-byt-rt5670.tplg",
+	.sof_fw_filename = "intel/sof-byt.ri",
+	.sof_tplg_filename = "intel/sof-byt-rt5670.tplg",
 	.asoc_plat_name = "sst-mfld-platform",
 };
 
@@ -98,8 +98,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "bytcr_rt5640",
 		.machine_quirk = byt_quirk,
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-rt5640.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-rt5640.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -107,8 +107,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "bytcr_rt5640",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "bytcr_rt5640",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-rt5640.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-rt5640.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -116,8 +116,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "bytcr_rt5640",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "bytcr_rt5640",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-rt5640.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-rt5640.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -125,8 +125,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "bytcr_rt5651",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "bytcr_rt5651",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-rt5651.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-rt5651.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -134,8 +134,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "bytcht_da7213",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "bytcht_da7213",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-da7213.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-da7213.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -143,8 +143,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "bytcht_da7213",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "bytcht_da7213",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-da7213.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-da7213.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	/* some Baytrail platforms rely on RT5645, use CHT machine driver */
@@ -153,8 +153,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "cht-bsw-rt5645",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-rt5645.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-rt5645.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -162,8 +162,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "cht-bsw-rt5645",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-rt5645.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-rt5645.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	/* use CHT driver to Baytrail Chromebooks */
@@ -172,8 +172,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "cht-bsw-max98090",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-max98090.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-max98090.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH)
diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c
index ad1eb2d644be..91bb99b69601 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c
@@ -44,8 +44,8 @@ static struct snd_soc_acpi_mach cht_surface_mach = {
 	.drv_name = "cht-bsw-rt5645",
 	.fw_filename = "intel/fw_sst_22a8.bin",
 	.board = "cht-bsw",
-	.sof_fw_filename = "intel/reef-cht.ri",
-	.sof_tplg_filename = "intel/reef-cht-rt5645.tplg",
+	.sof_fw_filename = "intel/sof-cht.ri",
+	.sof_tplg_filename = "intel/sof-cht-rt5645.tplg",
 	.asoc_plat_name = "sst-mfld-platform",
 };
 
@@ -68,8 +68,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "cht-bsw-rt5672",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5670.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5670.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -77,8 +77,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "cht-bsw-rt5672",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5670.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5670.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -86,8 +86,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "cht-bsw-rt5645",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5645.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5645.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -95,8 +95,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "cht-bsw-rt5645",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5645.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5645.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -104,8 +104,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "cht-bsw-rt5645",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5645.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5645.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -113,8 +113,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "cht-bsw-max98090",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-max98090.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-max98090.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -122,8 +122,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "cht-bsw-nau8824",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-nau8824.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-nau8824.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -131,8 +131,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "bytcht_da7213",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "bytcht_da7213",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-da7213.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-da7213.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -140,8 +140,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "bytcht_da7213",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "bytcht_da7213",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-da7213.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-da7213.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -149,8 +149,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "bytcht_es8316",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "bytcht_es8316",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-es8316.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-es8316.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	/* some CHT-T platforms rely on RT5640, use Baytrail machine driver */
@@ -160,8 +160,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "bytcr_rt5640",
 		.machine_quirk = cht_quirk,
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5640.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5640.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -169,8 +169,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "bytcr_rt5640",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "bytcr_rt5640",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5640.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5640.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	/* some CHT-T platforms rely on RT5651, use Baytrail machine driver */
@@ -179,8 +179,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "bytcr_rt5651",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "bytcr_rt5651",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5651.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5651.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH)
diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
new file mode 100644
index 000000000000..ec8e28e7b937
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
@@ -0,0 +1,32 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-cnl-match.c - tables and support for CNL ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+#include "../skylake/skl.h"
+
+static struct skl_machine_pdata cnl_pdata = {
+	.use_tplg_pcm = true,
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = {
+	{
+		.id = "INT34C2",
+		.drv_name = "cnl_rt274",
+		.fw_filename = "intel/dsp_fw_cnl.bin",
+		.pdata = &cnl_pdata,
+		.sof_fw_filename = "intel/sof-cnl.ri",
+		.sof_tplg_filename = "intel/sof-cnl-rt274.tplg",
+		.asoc_plat_name = "0000:00:1f.3",
+	},
+	{},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cnl_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
new file mode 100644
index 000000000000..305875af71ca
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
@@ -0,0 +1,41 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-glk-match.c - tables and support for GLK ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+
+static struct snd_soc_acpi_codecs glk_codecs = {
+	.num_codecs = 1,
+	.codecs = {"MX98357A"}
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = {
+	{
+		.id = "INT343A",
+		.drv_name = "glk_alc298s_i2s",
+		.fw_filename = "intel/dsp_fw_glk.bin",
+		.sof_fw_filename = "intel/sof-glk.ri",
+		.sof_tplg_filename = "intel/sof-glk-alc298.tplg",
+		.asoc_plat_name = "0000:00:0e.0",
+	},
+	{
+		.id = "DLGS7219",
+		.drv_name = "glk_da7219_max98357a",
+		.fw_filename = "intel/dsp_fw_glk.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &glk_codecs,
+		.sof_fw_filename = "intel/sof-glk.ri",
+		.sof_tplg_filename = "intel/sof-glk-da7219.tplg",
+		.asoc_plat_name = "0000:00:0e.0",
+	},
+	{},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_glk_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
index e0e8c8c27528..494a0ea9b029 100644
--- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
@@ -23,8 +23,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_haswell_machines[] = {
 		.id = "INT33CA",
 		.drv_name = "haswell-audio",
 		.fw_filename = "intel/IntcSST1.bin",
-		.sof_fw_filename = "intel/reef-hsw.ri",
-		.sof_tplg_filename = "intel/reef-hsw.tplg",
+		.sof_fw_filename = "intel/sof-hsw.ri",
+		.sof_tplg_filename = "intel/sof-hsw.tplg",
 		.asoc_plat_name = "haswell-pcm-audio",
 	},
 	{}
@@ -36,24 +36,24 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = {
 		.id = "INT343A",
 		.drv_name = "broadwell-audio",
 		.fw_filename =  "intel/IntcSST2.bin",
-		.sof_fw_filename = "intel/reef-bdw.ri",
-		.sof_tplg_filename = "intel/reef-bdw-rt286.tplg",
+		.sof_fw_filename = "intel/sof-bdw.ri",
+		.sof_tplg_filename = "intel/sof-bdw-rt286.tplg",
 		.asoc_plat_name = "haswell-pcm-audio",
 	},
 	{
 		.id = "RT5677CE",
 		.drv_name = "bdw-rt5677",
 		.fw_filename =  "intel/IntcSST2.bin",
-		.sof_fw_filename = "intel/reef-bdw.ri",
-		.sof_tplg_filename = "intel/reef-bdw-rt286.tplg",
+		.sof_fw_filename = "intel/sof-bdw.ri",
+		.sof_tplg_filename = "intel/sof-bdw-rt5677.tplg",
 		.asoc_plat_name = "haswell-pcm-audio",
 	},
 	{
 		.id = "INT33CA",
 		.drv_name = "haswell-audio",
 		.fw_filename = "intel/IntcSST2.bin",
-		.sof_fw_filename = "intel/reef-bdw.ri",
-		.sof_tplg_filename = "intel/reef-bdw-rt5640.tplg",
+		.sof_fw_filename = "intel/sof-bdw.ri",
+		.sof_tplg_filename = "intel/sof-bdw-rt5640.tplg",
 		.asoc_plat_name = "haswell-pcm-audio",
 	},
 	{}
diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
new file mode 100644
index 000000000000..0ee173ca437d
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
@@ -0,0 +1,91 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-kbl-match.c - tables and support for KBL ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+#include "../skylake/skl.h"
+
+static struct skl_machine_pdata skl_dmic_data;
+
+static struct snd_soc_acpi_codecs kbl_codecs = {
+	.num_codecs = 1,
+	.codecs = {"10508825"}
+};
+
+static struct snd_soc_acpi_codecs kbl_poppy_codecs = {
+	.num_codecs = 1,
+	.codecs = {"10EC5663"}
+};
+
+static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = {
+	.num_codecs = 2,
+	.codecs = {"10EC5663", "10EC5514"}
+};
+
+static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = {
+	.num_codecs = 1,
+	.codecs = {"MX98357A"}
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = {
+	{
+		.id = "INT343A",
+		.drv_name = "kbl_alc286s_i2s",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+	},
+	{
+		.id = "INT343B",
+		.drv_name = "kbl_n88l25_s4567",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &kbl_codecs,
+		.pdata = &skl_dmic_data,
+	},
+	{
+		.id = "MX98357A",
+		.drv_name = "kbl_n88l25_m98357a",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &kbl_codecs,
+		.pdata = &skl_dmic_data,
+	},
+	{
+		.id = "MX98927",
+		.drv_name = "kbl_r5514_5663_max",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &kbl_5663_5514_codecs,
+		.pdata = &skl_dmic_data,
+	},
+	{
+		.id = "MX98927",
+		.drv_name = "kbl_rt5663_m98927",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &kbl_poppy_codecs,
+		.pdata = &skl_dmic_data,
+	},
+	{
+		.id = "10EC5663",
+		.drv_name = "kbl_rt5663",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+	},
+	{
+		.id = "DLGS7219",
+		.drv_name = "kbl_da7219_max98357a",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &kbl_7219_98357_codecs,
+		.pdata = &skl_dmic_data,
+	},
+	{},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-skl-match.c b/sound/soc/intel/common/soc-acpi-intel-skl-match.c
new file mode 100644
index 000000000000..0c9c0edd35b3
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-skl-match.c
@@ -0,0 +1,47 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-skl-match.c - tables and support for SKL ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+#include "../skylake/skl.h"
+
+static struct skl_machine_pdata skl_dmic_data;
+
+static struct snd_soc_acpi_codecs skl_codecs = {
+	.num_codecs = 1,
+	.codecs = {"10508825"}
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[] = {
+	{
+		.id = "INT343A",
+		.drv_name = "skl_alc286s_i2s",
+		.fw_filename = "intel/dsp_fw_release.bin",
+	},
+	{
+		.id = "INT343B",
+		.drv_name = "skl_n88l25_s4567",
+		.fw_filename = "intel/dsp_fw_release.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &skl_codecs,
+		.pdata = &skl_dmic_data,
+	},
+	{
+		.id = "MX98357A",
+		.drv_name = "skl_n88l25_m98357a",
+		.fw_filename = "intel/dsp_fw_release.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &skl_codecs,
+		.pdata = &skl_dmic_data,
+	},
+	{},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_skl_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c
index 657afc02f1c4..11041aedea31 100644
--- a/sound/soc/intel/common/sst-firmware.c
+++ b/sound/soc/intel/common/sst-firmware.c
@@ -270,7 +270,7 @@ void sst_dsp_dma_put_channel(struct sst_dsp *dsp)
 }
 EXPORT_SYMBOL_GPL(sst_dsp_dma_put_channel);
 
-int sst_dma_new(struct sst_dsp *sst)
+static int sst_dma_new(struct sst_dsp *sst)
 {
 	struct sst_pdata *sst_pdata = sst->pdata;
 	struct sst_dma *dma;
@@ -320,9 +320,8 @@ err_dma_dev:
 	devm_kfree(sst->dev, dma);
 	return ret;
 }
-EXPORT_SYMBOL(sst_dma_new);
 
-void sst_dma_free(struct sst_dma *dma)
+static void sst_dma_free(struct sst_dma *dma)
 {
 
 	if (dma == NULL)
@@ -335,7 +334,6 @@ void sst_dma_free(struct sst_dma *dma)
 		dw_remove(dma->chip);
 
 }
-EXPORT_SYMBOL(sst_dma_free);
 
 /* create new generic firmware object */
 struct sst_fw *sst_fw_new(struct sst_dsp *dsp, 
diff --git a/sound/soc/intel/haswell/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c
index b2bec36d074c..a28220e67cdf 100644
--- a/sound/soc/intel/haswell/sst-haswell-dsp.c
+++ b/sound/soc/intel/haswell/sst-haswell-dsp.c
@@ -93,29 +93,31 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
 	struct sst_module_template template;
 	int count, ret;
 	void __iomem *ram;
+	int type = le16_to_cpu(module->type);
+	int entry_point = le32_to_cpu(module->entry_point);
 
 	/* TODO: allowed module types need to be configurable */
-	if (module->type != SST_HSW_MODULE_BASE_FW
-		&& module->type != SST_HSW_MODULE_PCM_SYSTEM
-		&& module->type != SST_HSW_MODULE_PCM
-		&& module->type != SST_HSW_MODULE_PCM_REFERENCE
-		&& module->type != SST_HSW_MODULE_PCM_CAPTURE
-		&& module->type != SST_HSW_MODULE_WAVES
-		&& module->type != SST_HSW_MODULE_LPAL)
+	if (type != SST_HSW_MODULE_BASE_FW &&
+	    type != SST_HSW_MODULE_PCM_SYSTEM &&
+	    type != SST_HSW_MODULE_PCM &&
+	    type != SST_HSW_MODULE_PCM_REFERENCE &&
+	    type != SST_HSW_MODULE_PCM_CAPTURE &&
+	    type != SST_HSW_MODULE_WAVES &&
+	    type != SST_HSW_MODULE_LPAL)
 		return 0;
 
 	dev_dbg(dsp->dev, "new module sign 0x%s size 0x%x blocks 0x%x type 0x%x\n",
 		module->signature, module->mod_size,
-		module->blocks, module->type);
-	dev_dbg(dsp->dev, " entrypoint 0x%x\n", module->entry_point);
+		module->blocks, type);
+	dev_dbg(dsp->dev, " entrypoint 0x%x\n", entry_point);
 	dev_dbg(dsp->dev, " persistent 0x%x scratch 0x%x\n",
 		module->info.persistent_size, module->info.scratch_size);
 
 	memset(&template, 0, sizeof(template));
-	template.id = module->type;
-	template.entry = module->entry_point - 4;
-	template.persistent_size = module->info.persistent_size;
-	template.scratch_size = module->info.scratch_size;
+	template.id = type;
+	template.entry = entry_point - 4;
+	template.persistent_size = le32_to_cpu(module->info.persistent_size);
+	template.scratch_size = le32_to_cpu(module->info.scratch_size);
 
 	mod = sst_module_new(fw, &template, NULL);
 	if (mod == NULL)
@@ -123,26 +125,26 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
 
 	block = (void *)module + sizeof(*module);
 
-	for (count = 0; count < module->blocks; count++) {
+	for (count = 0; count < le32_to_cpu(module->blocks); count++) {
 
-		if (block->size <= 0) {
+		if (le32_to_cpu(block->size) <= 0) {
 			dev_err(dsp->dev,
 				"error: block %d size invalid\n", count);
 			sst_module_free(mod);
 			return -EINVAL;
 		}
 
-		switch (block->type) {
+		switch (le32_to_cpu(block->type)) {
 		case SST_HSW_IRAM:
 			ram = dsp->addr.lpe;
-			mod->offset =
-				block->ram_offset + dsp->addr.iram_offset;
+			mod->offset = le32_to_cpu(block->ram_offset) +
+				dsp->addr.iram_offset;
 			mod->type = SST_MEM_IRAM;
 			break;
 		case SST_HSW_DRAM:
 		case SST_HSW_REGS:
 			ram = dsp->addr.lpe;
-			mod->offset = block->ram_offset;
+			mod->offset = le32_to_cpu(block->ram_offset);
 			mod->type = SST_MEM_DRAM;
 			break;
 		default:
@@ -152,7 +154,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
 			return -EINVAL;
 		}
 
-		mod->size = block->size;
+		mod->size = le32_to_cpu(block->size);
 		mod->data = (void *)block + sizeof(*block);
 		mod->data_offset = mod->data - fw->dma_buf;
 
@@ -169,7 +171,8 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
 			return ret;
 		}
 
-		block = (void *)block + sizeof(*block) + block->size;
+		block = (void *)block + sizeof(*block) +
+			le32_to_cpu(block->size);
 	}
 	mod->state = SST_MODULE_STATE_LOADED;
 
@@ -188,7 +191,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw)
 
 	/* verify FW */
 	if ((strncmp(header->signature, SST_HSW_FW_SIGN, 4) != 0) ||
-		(sst_fw->size != header->file_size + sizeof(*header))) {
+	    (sst_fw->size !=
+	     le32_to_cpu(header->file_size) + sizeof(*header))) {
 		dev_err(dsp->dev, "error: invalid fw sign/filesize mismatch\n");
 		return -EINVAL;
 	}
@@ -199,7 +203,7 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw)
 
 	/* parse each module */
 	module = (void *)sst_fw->dma_buf + sizeof(*header);
-	for (count = 0; count < header->modules; count++) {
+	for (count = 0; count < le32_to_cpu(header->modules); count++) {
 
 		/* module */
 		ret = hsw_parse_module(dsp, sst_fw, module);
@@ -207,7 +211,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw)
 			dev_err(dsp->dev, "error: invalid module %d\n", count);
 			return ret;
 		}
-		module = (void *)module + sizeof(*module) + module->mod_size;
+		module = (void *)module + sizeof(*module) +
+			le32_to_cpu(module->mod_size);
 	}
 
 	return 0;
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index d5f9c30eba32..8bfb8b0fa3d5 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -33,8 +33,7 @@
 static int skl_alloc_dma_buf(struct device *dev,
 		struct snd_dma_buffer *dmab, size_t size)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = dev_get_drvdata(dev);
 
 	if (!bus)
 		return -ENODEV;
@@ -44,8 +43,7 @@ static int skl_alloc_dma_buf(struct device *dev,
 
 static int skl_free_dma_buf(struct device *dev, struct snd_dma_buffer *dmab)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = dev_get_drvdata(dev);
 
 	if (!bus)
 		return -ENODEV;
@@ -89,8 +87,7 @@ void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable)
 static int skl_dsp_setup_spib(struct device *dev, unsigned int size,
 				int stream_tag, int enable)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = dev_get_drvdata(dev);
 	struct hdac_stream *stream = snd_hdac_get_stream(bus,
 			SNDRV_PCM_STREAM_PLAYBACK, stream_tag);
 	struct hdac_ext_stream *estream;
@@ -100,10 +97,10 @@ static int skl_dsp_setup_spib(struct device *dev, unsigned int size,
 
 	estream = stream_to_hdac_ext_stream(stream);
 	/* enable/disable SPIB for this hdac stream */
-	snd_hdac_ext_stream_spbcap_enable(ebus, enable, stream->index);
+	snd_hdac_ext_stream_spbcap_enable(bus, enable, stream->index);
 
 	/* set the spib value */
-	snd_hdac_ext_stream_set_spib(ebus, estream, size);
+	snd_hdac_ext_stream_set_spib(bus, estream, size);
 
 	return 0;
 }
@@ -111,8 +108,7 @@ static int skl_dsp_setup_spib(struct device *dev, unsigned int size,
 static int skl_dsp_prepare(struct device *dev, unsigned int format,
 			unsigned int size, struct snd_dma_buffer *dmab)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = dev_get_drvdata(dev);
 	struct hdac_ext_stream *estream;
 	struct hdac_stream *stream;
 	struct snd_pcm_substream substream;
@@ -124,7 +120,7 @@ static int skl_dsp_prepare(struct device *dev, unsigned int format,
 	memset(&substream, 0, sizeof(substream));
 	substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
 
-	estream = snd_hdac_ext_stream_assign(ebus, &substream,
+	estream = snd_hdac_ext_stream_assign(bus, &substream,
 					HDAC_EXT_STREAM_TYPE_HOST);
 	if (!estream)
 		return -ENODEV;
@@ -143,9 +139,8 @@ static int skl_dsp_prepare(struct device *dev, unsigned int format,
 
 static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+	struct hdac_bus *bus = dev_get_drvdata(dev);
 	struct hdac_stream *stream;
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
 
 	if (!bus)
 		return -ENODEV;
@@ -163,10 +158,9 @@ static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag)
 static int skl_dsp_cleanup(struct device *dev,
 		struct snd_dma_buffer *dmab, int stream_tag)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+	struct hdac_bus *bus = dev_get_drvdata(dev);
 	struct hdac_stream *stream;
 	struct hdac_ext_stream *estream;
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
 
 	if (!bus)
 		return -ENODEV;
@@ -270,8 +264,7 @@ const struct skl_dsp_ops *skl_get_dsp_ops(int pci_id)
 int skl_init_dsp(struct skl *skl)
 {
 	void __iomem *mmio_base;
-	struct hdac_ext_bus *ebus = &skl->ebus;
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = skl_to_bus(skl);
 	struct skl_dsp_loader_ops loader_ops;
 	int irq = bus->irq;
 	const struct skl_dsp_ops *ops;
@@ -279,8 +272,8 @@ int skl_init_dsp(struct skl *skl)
 	int ret;
 
 	/* enable ppcap interrupt */
-	snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true);
-	snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true);
+	snd_hdac_ext_bus_ppcap_enable(bus, true);
+	snd_hdac_ext_bus_ppcap_int_enable(bus, true);
 
 	/* read the BAR of the ADSP MMIO */
 	mmio_base = pci_ioremap_bar(skl->pci, 4);
@@ -335,12 +328,11 @@ unmap_mmio:
 
 int skl_free_dsp(struct skl *skl)
 {
-	struct hdac_ext_bus *ebus = &skl->ebus;
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = skl_to_bus(skl);
 	struct skl_sst *ctx = skl->skl_sst;
 
 	/* disable  ppcap interrupt */
-	snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, false);
+	snd_hdac_ext_bus_ppcap_int_enable(bus, false);
 
 	ctx->dsp_ops->cleanup(bus->dev, ctx);
 
@@ -383,10 +375,11 @@ int skl_suspend_late_dsp(struct skl *skl)
 int skl_suspend_dsp(struct skl *skl)
 {
 	struct skl_sst *ctx = skl->skl_sst;
+	struct hdac_bus *bus = skl_to_bus(skl);
 	int ret;
 
 	/* if ppcap is not supported return 0 */
-	if (!skl->ebus.bus.ppcap)
+	if (!bus->ppcap)
 		return 0;
 
 	ret = skl_dsp_sleep(ctx->dsp);
@@ -394,8 +387,8 @@ int skl_suspend_dsp(struct skl *skl)
 		return ret;
 
 	/* disable ppcap interrupt */
-	snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, false);
-	snd_hdac_ext_bus_ppcap_enable(&skl->ebus, false);
+	snd_hdac_ext_bus_ppcap_int_enable(bus, false);
+	snd_hdac_ext_bus_ppcap_enable(bus, false);
 
 	return 0;
 }
@@ -403,15 +396,16 @@ int skl_suspend_dsp(struct skl *skl)
 int skl_resume_dsp(struct skl *skl)
 {
 	struct skl_sst *ctx = skl->skl_sst;
+	struct hdac_bus *bus = skl_to_bus(skl);
 	int ret;
 
 	/* if ppcap is not supported return 0 */
-	if (!skl->ebus.bus.ppcap)
+	if (!bus->ppcap)
 		return 0;
 
 	/* enable ppcap interrupt */
-	snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true);
-	snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true);
+	snd_hdac_ext_bus_ppcap_enable(bus, true);
+	snd_hdac_ext_bus_ppcap_int_enable(bus, true);
 
 	/* check if DSP 1st boot is done */
 	if (skl->skl_sst->is_first_boot == true)
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index b9b140275be0..01a050cf8775 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -141,7 +141,7 @@ struct nhlt_specific_cfg
 {
 	struct nhlt_fmt *fmt;
 	struct nhlt_endpoint *epnt;
-	struct hdac_bus *bus = ebus_to_hbus(&skl->ebus);
+	struct hdac_bus *bus = skl_to_bus(skl);
 	struct device *dev = bus->dev;
 	struct nhlt_specific_cfg *sp_config;
 	struct nhlt_acpi_table *nhlt = skl->nhlt;
@@ -228,7 +228,7 @@ static void skl_nhlt_trim_space(char *trim)
 int skl_nhlt_update_topology_bin(struct skl *skl)
 {
 	struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt;
-	struct hdac_bus *bus = ebus_to_hbus(&skl->ebus);
+	struct hdac_bus *bus = skl_to_bus(skl);
 	struct device *dev = bus->dev;
 
 	dev_dbg(dev, "oem_id %.6s, oem_table_id %8s oem_revision %d\n",
@@ -248,8 +248,8 @@ static ssize_t skl_nhlt_platform_id_show(struct device *dev,
 			struct device_attribute *attr, char *buf)
 {
 	struct pci_dev *pci = to_pci_dev(dev);
-	struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
-	struct skl *skl = ebus_to_skl(ebus);
+	struct hdac_bus *bus = pci_get_drvdata(pci);
+	struct skl *skl = bus_to_skl(bus);
 	struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt;
 	char platform_id[32];
 
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index afa86b9e4dcf..823e39103edd 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -67,16 +67,15 @@ struct hdac_ext_stream *get_hdac_ext_stream(struct snd_pcm_substream *substream)
 	return substream->runtime->private_data;
 }
 
-static struct hdac_ext_bus *get_bus_ctx(struct snd_pcm_substream *substream)
+static struct hdac_bus *get_bus_ctx(struct snd_pcm_substream *substream)
 {
 	struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
 	struct hdac_stream *hstream = hdac_stream(stream);
 	struct hdac_bus *bus = hstream->bus;
-
-	return hbus_to_ebus(bus);
+	return bus;
 }
 
-static int skl_substream_alloc_pages(struct hdac_ext_bus *ebus,
+static int skl_substream_alloc_pages(struct hdac_bus *bus,
 				 struct snd_pcm_substream *substream,
 				 size_t size)
 {
@@ -95,7 +94,7 @@ static int skl_substream_free_pages(struct hdac_bus *bus,
 	return snd_pcm_lib_free_pages(substream);
 }
 
-static void skl_set_pcm_constrains(struct hdac_ext_bus *ebus,
+static void skl_set_pcm_constrains(struct hdac_bus *bus,
 				 struct snd_pcm_runtime *runtime)
 {
 	snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
@@ -105,9 +104,9 @@ static void skl_set_pcm_constrains(struct hdac_ext_bus *ebus,
 				     20, 178000000);
 }
 
-static enum hdac_ext_stream_type skl_get_host_stream_type(struct hdac_ext_bus *ebus)
+static enum hdac_ext_stream_type skl_get_host_stream_type(struct hdac_bus *bus)
 {
-	if ((ebus_to_hbus(ebus))->ppcap)
+	if (bus->ppcap)
 		return HDAC_EXT_STREAM_TYPE_HOST;
 	else
 		return HDAC_EXT_STREAM_TYPE_COUPLED;
@@ -123,9 +122,9 @@ static enum hdac_ext_stream_type skl_get_host_stream_type(struct hdac_ext_bus *e
 static void skl_set_suspend_active(struct snd_pcm_substream *substream,
 					 struct snd_soc_dai *dai, bool enable)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev);
+	struct hdac_bus *bus = dev_get_drvdata(dai->dev);
 	struct snd_soc_dapm_widget *w;
-	struct skl *skl = ebus_to_skl(ebus);
+	struct skl *skl = bus_to_skl(bus);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		w = dai->playback_widget;
@@ -140,8 +139,7 @@ static void skl_set_suspend_active(struct snd_pcm_substream *substream,
 
 int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = dev_get_drvdata(dev);
 	unsigned int format_val;
 	struct hdac_stream *hstream;
 	struct hdac_ext_stream *stream;
@@ -153,7 +151,7 @@ int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params)
 		return -EINVAL;
 
 	stream = stream_to_hdac_ext_stream(hstream);
-	snd_hdac_ext_stream_decouple(ebus, stream, true);
+	snd_hdac_ext_stream_decouple(bus, stream, true);
 
 	format_val = snd_hdac_calc_stream_format(params->s_freq,
 			params->ch, params->format, params->host_bps, 0);
@@ -177,8 +175,7 @@ int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params)
 
 int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = dev_get_drvdata(dev);
 	unsigned int format_val;
 	struct hdac_stream *hstream;
 	struct hdac_ext_stream *stream;
@@ -190,7 +187,7 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params)
 		return -EINVAL;
 
 	stream = stream_to_hdac_ext_stream(hstream);
-	snd_hdac_ext_stream_decouple(ebus, stream, true);
+	snd_hdac_ext_stream_decouple(bus, stream, true);
 	format_val = snd_hdac_calc_stream_format(params->s_freq, params->ch,
 					params->format, params->link_bps, 0);
 
@@ -201,7 +198,7 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params)
 
 	snd_hdac_ext_link_stream_setup(stream, format_val);
 
-	list_for_each_entry(link, &ebus->hlink_list, list) {
+	list_for_each_entry(link, &bus->hlink_list, list) {
 		if (link->index == params->link_index)
 			snd_hdac_ext_link_set_stream_id(link,
 					hstream->stream_tag);
@@ -215,7 +212,7 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params)
 static int skl_pcm_open(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev);
+	struct hdac_bus *bus = dev_get_drvdata(dai->dev);
 	struct hdac_ext_stream *stream;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct skl_dma_params *dma_params;
@@ -224,12 +221,12 @@ static int skl_pcm_open(struct snd_pcm_substream *substream,
 
 	dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
 
-	stream = snd_hdac_ext_stream_assign(ebus, substream,
-					skl_get_host_stream_type(ebus));
+	stream = snd_hdac_ext_stream_assign(bus, substream,
+					skl_get_host_stream_type(bus));
 	if (stream == NULL)
 		return -EBUSY;
 
-	skl_set_pcm_constrains(ebus, runtime);
+	skl_set_pcm_constrains(bus, runtime);
 
 	/*
 	 * disable WALLCLOCK timestamps for capture streams
@@ -301,7 +298,7 @@ static int skl_pcm_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev);
+	struct hdac_bus *bus = dev_get_drvdata(dai->dev);
 	struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct skl_pipe_params p_params = {0};
@@ -309,7 +306,7 @@ static int skl_pcm_hw_params(struct snd_pcm_substream *substream,
 	int ret, dma_id;
 
 	dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
-	ret = skl_substream_alloc_pages(ebus, substream,
+	ret = skl_substream_alloc_pages(bus, substream,
 					  params_buffer_bytes(params));
 	if (ret < 0)
 		return ret;
@@ -343,14 +340,14 @@ static void skl_pcm_close(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
 	struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev);
+	struct hdac_bus *bus = dev_get_drvdata(dai->dev);
 	struct skl_dma_params *dma_params = NULL;
-	struct skl *skl = ebus_to_skl(ebus);
+	struct skl *skl = bus_to_skl(bus);
 	struct skl_module_cfg *mconfig;
 
 	dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
 
-	snd_hdac_ext_stream_release(stream, skl_get_host_stream_type(ebus));
+	snd_hdac_ext_stream_release(stream, skl_get_host_stream_type(bus));
 
 	dma_params = snd_soc_dai_get_dma_data(dai, substream);
 	/*
@@ -380,7 +377,7 @@ static void skl_pcm_close(struct snd_pcm_substream *substream,
 static int skl_pcm_hw_free(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev);
+	struct hdac_bus *bus = dev_get_drvdata(dai->dev);
 	struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
 	struct skl *skl = get_skl_ctx(dai->dev);
 	struct skl_module_cfg *mconfig;
@@ -400,7 +397,7 @@ static int skl_pcm_hw_free(struct snd_pcm_substream *substream,
 	snd_hdac_stream_cleanup(hdac_stream(stream));
 	hdac_stream(stream)->prepared = 0;
 
-	return skl_substream_free_pages(ebus_to_hbus(ebus), substream);
+	return skl_substream_free_pages(bus, substream);
 }
 
 static int skl_be_hw_params(struct snd_pcm_substream *substream,
@@ -420,8 +417,7 @@ static int skl_be_hw_params(struct snd_pcm_substream *substream,
 static int skl_decoupled_trigger(struct snd_pcm_substream *substream,
 		int cmd)
 {
-	struct hdac_ext_bus *ebus = get_bus_ctx(substream);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = get_bus_ctx(substream);
 	struct hdac_ext_stream *stream;
 	int start;
 	unsigned long cookie;
@@ -470,7 +466,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 	struct skl *skl = get_skl_ctx(dai->dev);
 	struct skl_sst *ctx = skl->skl_sst;
 	struct skl_module_cfg *mconfig;
-	struct hdac_ext_bus *ebus = get_bus_ctx(substream);
+	struct hdac_bus *bus = get_bus_ctx(substream);
 	struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
 	struct snd_soc_dapm_widget *w;
 	int ret;
@@ -492,9 +488,9 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 			 * dpib & lpib position to resume before starting the
 			 * DMA
 			 */
-			snd_hdac_ext_stream_drsm_enable(ebus, true,
+			snd_hdac_ext_stream_drsm_enable(bus, true,
 						hdac_stream(stream)->index);
-			snd_hdac_ext_stream_set_dpibr(ebus, stream,
+			snd_hdac_ext_stream_set_dpibr(bus, stream,
 							stream->lpib);
 			snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
 		}
@@ -528,14 +524,14 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 		ret = skl_decoupled_trigger(substream, cmd);
 		if ((cmd == SNDRV_PCM_TRIGGER_SUSPEND) && !w->ignore_suspend) {
 			/* save the dpib and lpib positions */
-			stream->dpib = readl(ebus->bus.remap_addr +
+			stream->dpib = readl(bus->remap_addr +
 					AZX_REG_VS_SDXDPIB_XBASE +
 					(AZX_REG_VS_SDXDPIB_XINTERVAL *
 					hdac_stream(stream)->index));
 
 			stream->lpib = snd_hdac_stream_get_pos_lpib(
 							hdac_stream(stream));
-			snd_hdac_ext_stream_decouple(ebus, stream, false);
+			snd_hdac_ext_stream_decouple(bus, stream, false);
 		}
 		break;
 
@@ -546,11 +542,12 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 	return 0;
 }
 
+
 static int skl_link_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev);
+	struct hdac_bus *bus = dev_get_drvdata(dai->dev);
 	struct hdac_ext_stream *link_dev;
 	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
@@ -558,14 +555,14 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream,
 	struct hdac_ext_link *link;
 	int stream_tag;
 
-	link_dev = snd_hdac_ext_stream_assign(ebus, substream,
+	link_dev = snd_hdac_ext_stream_assign(bus, substream,
 					HDAC_EXT_STREAM_TYPE_LINK);
 	if (!link_dev)
 		return -EBUSY;
 
 	snd_soc_dai_set_dma_data(dai, substream, (void *)link_dev);
 
-	link = snd_hdac_ext_bus_get_link(ebus, codec_dai->component->name);
+	link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name);
 	if (!link)
 		return -EINVAL;
 
@@ -610,7 +607,7 @@ static int skl_link_pcm_trigger(struct snd_pcm_substream *substream,
 {
 	struct hdac_ext_stream *link_dev =
 				snd_soc_dai_get_dma_data(dai, substream);
-	struct hdac_ext_bus *ebus = get_bus_ctx(substream);
+	struct hdac_bus *bus = get_bus_ctx(substream);
 	struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
 
 	dev_dbg(dai->dev, "In %s cmd=%d\n", __func__, cmd);
@@ -626,7 +623,7 @@ static int skl_link_pcm_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_STOP:
 		snd_hdac_ext_link_stream_clear(link_dev);
 		if (cmd == SNDRV_PCM_TRIGGER_SUSPEND)
-			snd_hdac_ext_stream_decouple(ebus, stream, false);
+			snd_hdac_ext_stream_decouple(bus, stream, false);
 		break;
 
 	default:
@@ -638,7 +635,7 @@ static int skl_link_pcm_trigger(struct snd_pcm_substream *substream,
 static int skl_link_hw_free(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev);
+	struct hdac_bus *bus = dev_get_drvdata(dai->dev);
 	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
 	struct hdac_ext_stream *link_dev =
 				snd_soc_dai_get_dma_data(dai, substream);
@@ -648,7 +645,7 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream,
 
 	link_dev->link_prepared = 0;
 
-	link = snd_hdac_ext_bus_get_link(ebus, rtd->codec_dai->component->name);
+	link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name);
 	if (!link)
 		return -EINVAL;
 
@@ -1017,10 +1014,11 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
 },
 };
 
-int skl_dai_load(struct snd_soc_component *cmp,
-		 struct snd_soc_dai_driver *pcm_dai)
+int skl_dai_load(struct snd_soc_component *cmp, int index,
+			struct snd_soc_dai_driver *dai_drv,
+			struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai)
 {
-	pcm_dai->ops = &skl_pcm_dai_ops;
+	dai_drv->ops = &skl_pcm_dai_ops;
 
 	return 0;
 }
@@ -1041,8 +1039,7 @@ static int skl_platform_open(struct snd_pcm_substream *substream)
 static int skl_coupled_trigger(struct snd_pcm_substream *substream,
 					int cmd)
 {
-	struct hdac_ext_bus *ebus = get_bus_ctx(substream);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = get_bus_ctx(substream);
 	struct hdac_ext_stream *stream;
 	struct snd_pcm_substream *s;
 	bool start;
@@ -1115,9 +1112,9 @@ static int skl_coupled_trigger(struct snd_pcm_substream *substream,
 static int skl_platform_pcm_trigger(struct snd_pcm_substream *substream,
 					int cmd)
 {
-	struct hdac_ext_bus *ebus = get_bus_ctx(substream);
+	struct hdac_bus *bus = get_bus_ctx(substream);
 
-	if (!(ebus_to_hbus(ebus))->ppcap)
+	if (!bus->ppcap)
 		return skl_coupled_trigger(substream, cmd);
 
 	return 0;
@@ -1127,7 +1124,7 @@ static snd_pcm_uframes_t skl_platform_pcm_pointer
 			(struct snd_pcm_substream *substream)
 {
 	struct hdac_ext_stream *hstream = get_hdac_ext_stream(substream);
-	struct hdac_ext_bus *ebus = get_bus_ctx(substream);
+	struct hdac_bus *bus = get_bus_ctx(substream);
 	unsigned int pos;
 
 	/*
@@ -1152,12 +1149,12 @@ static snd_pcm_uframes_t skl_platform_pcm_pointer
 	 */
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		pos = readl(ebus->bus.remap_addr + AZX_REG_VS_SDXDPIB_XBASE +
+		pos = readl(bus->remap_addr + AZX_REG_VS_SDXDPIB_XBASE +
 				(AZX_REG_VS_SDXDPIB_XINTERVAL *
 				hdac_stream(hstream)->index));
 	} else {
 		udelay(20);
-		readl(ebus->bus.remap_addr +
+		readl(bus->remap_addr +
 				AZX_REG_VS_SDXDPIB_XBASE +
 				(AZX_REG_VS_SDXDPIB_XINTERVAL *
 				 hdac_stream(hstream)->index));
@@ -1242,11 +1239,11 @@ static void skl_pcm_free(struct snd_pcm *pcm)
 static int skl_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dai *dai = rtd->cpu_dai;
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev);
+	struct hdac_bus *bus = dev_get_drvdata(dai->dev);
 	struct snd_pcm *pcm = rtd->pcm;
 	unsigned int size;
 	int retval = 0;
-	struct skl *skl = ebus_to_skl(ebus);
+	struct skl *skl = bus_to_skl(bus);
 
 	if (dai->driver->playback.channels_min ||
 		dai->driver->capture.channels_min) {
@@ -1356,19 +1353,19 @@ static int skl_populate_modules(struct skl *skl)
 
 static int skl_platform_soc_probe(struct snd_soc_component *component)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(component->dev);
-	struct skl *skl = ebus_to_skl(ebus);
+	struct hdac_bus *bus = dev_get_drvdata(component->dev);
+	struct skl *skl = bus_to_skl(bus);
 	const struct skl_dsp_ops *ops;
 	int ret;
 
 	pm_runtime_get_sync(component->dev);
-	if ((ebus_to_hbus(ebus))->ppcap) {
+	if (bus->ppcap) {
 		skl->component = component;
 
 		/* init debugfs */
 		skl->debugfs = skl_debugfs_init(skl);
 
-		ret = skl_tplg_init(component, ebus);
+		ret = skl_tplg_init(component, bus);
 		if (ret < 0) {
 			dev_err(component->dev, "Failed to init topology!\n");
 			return ret;
@@ -1425,10 +1422,10 @@ static const struct snd_soc_component_driver skl_component  = {
 int skl_platform_register(struct device *dev)
 {
 	int ret;
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
-	struct skl *skl = ebus_to_skl(ebus);
 	struct snd_soc_dai_driver *dais;
 	int num_dais = ARRAY_SIZE(skl_platform_dai);
+	struct hdac_bus *bus = dev_get_drvdata(dev);
+	struct skl *skl = bus_to_skl(bus);
 
 	INIT_LIST_HEAD(&skl->ppl_list);
 	INIT_LIST_HEAD(&skl->bind_list);
@@ -1464,8 +1461,8 @@ err:
 
 int skl_platform_unregister(struct device *dev)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
-	struct skl *skl = ebus_to_skl(ebus);
+	struct hdac_bus *bus = dev_get_drvdata(dev);
+	struct skl *skl = bus_to_skl(bus);
 	struct skl_module_deferred_bind *modules, *tmp;
 
 	if (!list_empty(&skl->bind_list)) {
diff --git a/sound/soc/intel/skylake/skl-sst-cldma.c b/sound/soc/intel/skylake/skl-sst-cldma.c
index d2b1d60fec02..5bc0d38da7e3 100644
--- a/sound/soc/intel/skylake/skl-sst-cldma.c
+++ b/sound/soc/intel/skylake/skl-sst-cldma.c
@@ -83,9 +83,9 @@ static void skl_cldma_stream_clear(struct sst_dsp  *ctx)
 /* Code loader helper APIs */
 static void skl_cldma_setup_bdle(struct sst_dsp *ctx,
 		struct snd_dma_buffer *dmab_data,
-		u32 **bdlp, int size, int with_ioc)
+		__le32 **bdlp, int size, int with_ioc)
 {
-	u32 *bdl = *bdlp;
+	__le32 *bdl = *bdlp;
 
 	ctx->cl_dev.frags = 0;
 	while (size > 0) {
@@ -330,7 +330,7 @@ void skl_cldma_process_intr(struct sst_dsp *ctx)
 int skl_cldma_prepare(struct sst_dsp *ctx)
 {
 	int ret;
-	u32 *bdl;
+	__le32 *bdl;
 
 	ctx->cl_dev.bufsize = SKL_MAX_BUFFER_SIZE;
 
@@ -359,7 +359,7 @@ int skl_cldma_prepare(struct sst_dsp *ctx)
 		ctx->dsp_ops.free_dma_buf(ctx->dev, &ctx->cl_dev.dmab_data);
 		return ret;
 	}
-	bdl = (u32 *)ctx->cl_dev.dmab_bdl.area;
+	bdl = (__le32 *)ctx->cl_dev.dmab_bdl.area;
 
 	/* Allocate BDLs */
 	ctx->cl_dev.ops.cl_setup_bdle(ctx, &ctx->cl_dev.dmab_data,
diff --git a/sound/soc/intel/skylake/skl-sst-cldma.h b/sound/soc/intel/skylake/skl-sst-cldma.h
index 5b730a1a0ae4..ec736921a083 100644
--- a/sound/soc/intel/skylake/skl-sst-cldma.h
+++ b/sound/soc/intel/skylake/skl-sst-cldma.h
@@ -203,7 +203,7 @@ struct sst_dsp;
 struct skl_cl_dev_ops {
 	void (*cl_setup_bdle)(struct sst_dsp *ctx,
 			struct snd_dma_buffer *dmab_data,
-			u32 **bdlp, int size, int with_ioc);
+			__le32 **bdlp, int size, int with_ioc);
 	void (*cl_setup_controller)(struct sst_dsp *ctx,
 			struct snd_dma_buffer *dmab_bdl,
 			unsigned int max_size, u32 page_count);
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index fcdc716754b6..2620d77729c5 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -108,6 +108,9 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w,
 	case snd_soc_dapm_aif_out:
 	case snd_soc_dapm_dai_out:
 	case snd_soc_dapm_switch:
+	case snd_soc_dapm_output:
+	case snd_soc_dapm_mux:
+
 		return false;
 	default:
 		return true;
@@ -934,7 +937,7 @@ static int skl_tplg_find_moduleid_from_uuid(struct skl *skl,
 	struct soc_bytes_ext *sb = (void *) k->private_value;
 	struct skl_algo_data *bc = (struct skl_algo_data *)sb->dobj.private;
 	struct skl_kpb_params *uuid_params, *params;
-	struct hdac_bus *bus = ebus_to_hbus(skl_to_ebus(skl));
+	struct hdac_bus *bus = skl_to_bus(skl);
 	int i, size, module_id;
 
 	if (bc->set_params == SKL_PARAM_BIND && bc->max) {
@@ -3024,14 +3027,13 @@ void skl_cleanup_resources(struct skl *skl)
  * information to the driver about module and pipeline parameters which DSP
  * FW expects like ids, resource values, formats etc
  */
-static int skl_tplg_widget_load(struct snd_soc_component *cmpnt,
+static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, int index,
 				struct snd_soc_dapm_widget *w,
 				struct snd_soc_tplg_dapm_widget *tplg_w)
 {
 	int ret;
-	struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt);
-	struct skl *skl = ebus_to_skl(ebus);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt);
+	struct skl *skl = bus_to_skl(bus);
 	struct skl_module_cfg *mconfig;
 
 	if (!tplg_w->priv.size)
@@ -3131,14 +3133,14 @@ static int skl_init_enum_data(struct device *dev, struct soc_enum *se,
 }
 
 static int skl_tplg_control_load(struct snd_soc_component *cmpnt,
+				int index,
 				struct snd_kcontrol_new *kctl,
 				struct snd_soc_tplg_ctl_hdr *hdr)
 {
 	struct soc_bytes_ext *sb;
 	struct snd_soc_tplg_bytes_control *tplg_bc;
 	struct snd_soc_tplg_enum_control *tplg_ec;
-	struct hdac_ext_bus *ebus  = snd_soc_component_get_drvdata(cmpnt);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus  = snd_soc_component_get_drvdata(cmpnt);
 	struct soc_enum *se;
 
 	switch (hdr->ops.info) {
@@ -3619,12 +3621,11 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest,
 	return 0;
 }
 
-static int skl_manifest_load(struct snd_soc_component *cmpnt,
+static int skl_manifest_load(struct snd_soc_component *cmpnt, int index,
 				struct snd_soc_tplg_manifest *manifest)
 {
-	struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
-	struct skl *skl = ebus_to_skl(ebus);
+	struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt);
+	struct skl *skl = bus_to_skl(bus);
 
 	/* proceed only if we have private data defined */
 	if (manifest->priv.size == 0)
@@ -3713,12 +3714,11 @@ static void skl_tplg_set_pipe_type(struct skl *skl, struct skl_pipe *pipe)
 /*
  * SKL topology init routine
  */
-int skl_tplg_init(struct snd_soc_component *component, struct hdac_ext_bus *ebus)
+int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus)
 {
 	int ret;
 	const struct firmware *fw;
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
-	struct skl *skl = ebus_to_skl(ebus);
+	struct skl *skl = bus_to_skl(bus);
 	struct skl_pipeline *ppl;
 
 	ret = request_firmware(&fw, skl->tplg_name, bus->dev);
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index 6d7e0569695f..82282cac9751 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -458,9 +458,9 @@ enum skl_channel {
 
 static inline struct skl *get_skl_ctx(struct device *dev)
 {
-	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+	struct hdac_bus *bus = dev_get_drvdata(dev);
 
-	return ebus_to_skl(ebus);
+	return bus_to_skl(bus);
 }
 
 int skl_tplg_be_update_params(struct snd_soc_dai *dai,
@@ -470,7 +470,7 @@ int skl_dsp_set_dma_control(struct skl_sst *ctx, u32 *caps,
 void skl_tplg_set_be_dmic_config(struct snd_soc_dai *dai,
 	struct skl_pipe_params *params, int stream);
 int skl_tplg_init(struct snd_soc_component *component,
-				struct hdac_ext_bus *ebus);
+				struct hdac_bus *ebus);
 struct skl_module_cfg *skl_tplg_fe_get_cpr_module(
 		struct snd_soc_dai *dai, int stream);
 int skl_tplg_update_pipe_params(struct device *dev,
@@ -512,8 +512,9 @@ int skl_pcm_host_dma_prepare(struct device *dev,
 int skl_pcm_link_dma_prepare(struct device *dev,
 			struct skl_pipe_params *params);
 
-int skl_dai_load(struct snd_soc_component *cmp,
-		 struct snd_soc_dai_driver *pcm_dai);
+int skl_dai_load(struct snd_soc_component *cmp, int index,
+		struct snd_soc_dai_driver *dai_drv,
+		struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai);
 void skl_tplg_add_moduleid_in_bind_params(struct skl *skl,
 				struct snd_soc_dapm_widget *w);
 #endif
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index f0d9793f872a..dce649485649 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -29,6 +29,7 @@
 #include <linux/delay.h>
 #include <sound/pcm.h>
 #include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
 #include <sound/hda_register.h>
 #include <sound/hdaudio.h>
 #include <sound/hda_i915.h>
@@ -36,8 +37,6 @@
 #include "skl-sst-dsp.h"
 #include "skl-sst-ipc.h"
 
-static struct skl_machine_pdata skl_dmic_data;
-
 /*
  * initialize the PCI registers
  */
@@ -54,7 +53,7 @@ static void skl_update_pci_byte(struct pci_dev *pci, unsigned int reg,
 
 static void skl_init_pci(struct skl *skl)
 {
-	struct hdac_ext_bus *ebus = &skl->ebus;
+	struct hdac_bus *bus = skl_to_bus(skl);
 
 	/*
 	 * Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
@@ -63,7 +62,7 @@ static void skl_init_pci(struct skl *skl)
 	 * codecs.
 	 * The PCI register TCSEL is defined in the Intel manuals.
 	 */
-	dev_dbg(ebus_to_hbus(ebus)->dev, "Clearing TCSEL\n");
+	dev_dbg(bus->dev, "Clearing TCSEL\n");
 	skl_update_pci_byte(skl->pci, AZX_PCIREG_TCSEL, 0x07, 0);
 }
 
@@ -103,8 +102,7 @@ static void skl_enable_miscbdcge(struct device *dev, bool enable)
 static void skl_clock_power_gating(struct device *dev, bool enable)
 {
 	struct pci_dev *pci = to_pci_dev(dev);
-	struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = pci_get_drvdata(pci);
 	u32 val;
 
 	/* Update PDCGE bit of CGCTL register */
@@ -127,7 +125,6 @@ static void skl_clock_power_gating(struct device *dev, bool enable)
  */
 static int skl_init_chip(struct hdac_bus *bus, bool full_reset)
 {
-	struct hdac_ext_bus *ebus = hbus_to_ebus(bus);
 	struct hdac_ext_link *hlink;
 	int ret;
 
@@ -135,7 +132,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset)
 	ret = snd_hdac_bus_init_chip(bus, full_reset);
 
 	/* Reset stream-to-link mapping */
-	list_for_each_entry(hlink, &ebus->hlink_list, list)
+	list_for_each_entry(hlink, &bus->hlink_list, list)
 		bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV);
 
 	skl_enable_miscbdcge(bus->dev, true);
@@ -146,8 +143,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset)
 void skl_update_d0i3c(struct device *dev, bool enable)
 {
 	struct pci_dev *pci = to_pci_dev(dev);
-	struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = pci_get_drvdata(pci);
 	u8 reg;
 	int timeout = 50;
 
@@ -197,8 +193,7 @@ static void skl_stream_update(struct hdac_bus *bus, struct hdac_stream *hstr)
 
 static irqreturn_t skl_interrupt(int irq, void *dev_id)
 {
-	struct hdac_ext_bus *ebus = dev_id;
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = dev_id;
 	u32 status;
 
 	if (!pm_runtime_active(bus->dev))
@@ -227,8 +222,7 @@ static irqreturn_t skl_interrupt(int irq, void *dev_id)
 
 static irqreturn_t skl_threaded_handler(int irq, void *dev_id)
 {
-	struct hdac_ext_bus *ebus = dev_id;
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = dev_id;
 	u32 status;
 
 	status = snd_hdac_chip_readl(bus, INTSTS);
@@ -238,16 +232,15 @@ static irqreturn_t skl_threaded_handler(int irq, void *dev_id)
 	return IRQ_HANDLED;
 }
 
-static int skl_acquire_irq(struct hdac_ext_bus *ebus, int do_disconnect)
+static int skl_acquire_irq(struct hdac_bus *bus, int do_disconnect)
 {
-	struct skl *skl = ebus_to_skl(ebus);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct skl *skl = bus_to_skl(bus);
 	int ret;
 
 	ret = request_threaded_irq(skl->pci->irq, skl_interrupt,
 			skl_threaded_handler,
 			IRQF_SHARED,
-			KBUILD_MODNAME, ebus);
+			KBUILD_MODNAME, bus);
 	if (ret) {
 		dev_err(bus->dev,
 			"unable to grab IRQ %d, disabling device\n",
@@ -264,21 +257,20 @@ static int skl_acquire_irq(struct hdac_ext_bus *ebus, int do_disconnect)
 static int skl_suspend_late(struct device *dev)
 {
 	struct pci_dev *pci = to_pci_dev(dev);
-	struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
-	struct skl *skl = ebus_to_skl(ebus);
+	struct hdac_bus *bus = pci_get_drvdata(pci);
+	struct skl *skl = bus_to_skl(bus);
 
 	return skl_suspend_late_dsp(skl);
 }
 
 #ifdef CONFIG_PM
-static int _skl_suspend(struct hdac_ext_bus *ebus)
+static int _skl_suspend(struct hdac_bus *bus)
 {
-	struct skl *skl = ebus_to_skl(ebus);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct skl *skl = bus_to_skl(bus);
 	struct pci_dev *pci = to_pci_dev(bus->dev);
 	int ret;
 
-	snd_hdac_ext_bus_link_power_down_all(ebus);
+	snd_hdac_ext_bus_link_power_down_all(bus);
 
 	ret = skl_suspend_dsp(skl);
 	if (ret < 0)
@@ -295,10 +287,9 @@ static int _skl_suspend(struct hdac_ext_bus *ebus)
 	return 0;
 }
 
-static int _skl_resume(struct hdac_ext_bus *ebus)
+static int _skl_resume(struct hdac_bus *bus)
 {
-	struct skl *skl = ebus_to_skl(ebus);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct skl *skl = bus_to_skl(bus);
 
 	skl_init_pci(skl);
 	skl_init_chip(bus, true);
@@ -314,9 +305,8 @@ static int _skl_resume(struct hdac_ext_bus *ebus)
 static int skl_suspend(struct device *dev)
 {
 	struct pci_dev *pci = to_pci_dev(dev);
-	struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
-	struct skl *skl  = ebus_to_skl(ebus);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = pci_get_drvdata(pci);
+	struct skl *skl  = bus_to_skl(bus);
 	int ret = 0;
 
 	/*
@@ -325,15 +315,15 @@ static int skl_suspend(struct device *dev)
 	 */
 	if (skl->supend_active) {
 		/* turn off the links and stop the CORB/RIRB DMA if it is On */
-		snd_hdac_ext_bus_link_power_down_all(ebus);
+		snd_hdac_ext_bus_link_power_down_all(bus);
 
-		if (ebus->cmd_dma_state)
-			snd_hdac_bus_stop_cmd_io(&ebus->bus);
+		if (bus->cmd_dma_state)
+			snd_hdac_bus_stop_cmd_io(bus);
 
 		enable_irq_wake(bus->irq);
 		pci_save_state(pci);
 	} else {
-		ret = _skl_suspend(ebus);
+		ret = _skl_suspend(bus);
 		if (ret < 0)
 			return ret;
 		skl->skl_sst->fw_loaded = false;
@@ -352,9 +342,8 @@ static int skl_suspend(struct device *dev)
 static int skl_resume(struct device *dev)
 {
 	struct pci_dev *pci = to_pci_dev(dev);
-	struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
-	struct skl *skl  = ebus_to_skl(ebus);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = pci_get_drvdata(pci);
+	struct skl *skl  = bus_to_skl(bus);
 	struct hdac_ext_link *hlink = NULL;
 	int ret;
 
@@ -374,32 +363,32 @@ static int skl_resume(struct device *dev)
 	 */
 	if (skl->supend_active) {
 		pci_restore_state(pci);
-		snd_hdac_ext_bus_link_power_up_all(ebus);
+		snd_hdac_ext_bus_link_power_up_all(bus);
 		disable_irq_wake(bus->irq);
 		/*
 		 * turn On the links which are On before active suspend
 		 * and start the CORB/RIRB DMA if On before
 		 * active suspend.
 		 */
-		list_for_each_entry(hlink, &ebus->hlink_list, list) {
+		list_for_each_entry(hlink, &bus->hlink_list, list) {
 			if (hlink->ref_count)
 				snd_hdac_ext_bus_link_power_up(hlink);
 		}
 
-		if (ebus->cmd_dma_state)
-			snd_hdac_bus_init_cmd_io(&ebus->bus);
 		ret = 0;
+		if (bus->cmd_dma_state)
+			snd_hdac_bus_init_cmd_io(bus);
 	} else {
-		ret = _skl_resume(ebus);
+		ret = _skl_resume(bus);
 
 		/* turn off the links which are off before suspend */
-		list_for_each_entry(hlink, &ebus->hlink_list, list) {
+		list_for_each_entry(hlink, &bus->hlink_list, list) {
 			if (!hlink->ref_count)
 				snd_hdac_ext_bus_link_power_down(hlink);
 		}
 
-		if (!ebus->cmd_dma_state)
-			snd_hdac_bus_stop_cmd_io(&ebus->bus);
+		if (!bus->cmd_dma_state)
+			snd_hdac_bus_stop_cmd_io(bus);
 	}
 
 	return ret;
@@ -410,23 +399,21 @@ static int skl_resume(struct device *dev)
 static int skl_runtime_suspend(struct device *dev)
 {
 	struct pci_dev *pci = to_pci_dev(dev);
-	struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = pci_get_drvdata(pci);
 
 	dev_dbg(bus->dev, "in %s\n", __func__);
 
-	return _skl_suspend(ebus);
+	return _skl_suspend(bus);
 }
 
 static int skl_runtime_resume(struct device *dev)
 {
 	struct pci_dev *pci = to_pci_dev(dev);
-	struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = pci_get_drvdata(pci);
 
 	dev_dbg(bus->dev, "in %s\n", __func__);
 
-	return _skl_resume(ebus);
+	return _skl_resume(bus);
 }
 #endif /* CONFIG_PM */
 
@@ -439,20 +426,19 @@ static const struct dev_pm_ops skl_pm = {
 /*
  * destructor
  */
-static int skl_free(struct hdac_ext_bus *ebus)
+static int skl_free(struct hdac_bus *bus)
 {
-	struct skl *skl  = ebus_to_skl(ebus);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct skl *skl  = bus_to_skl(bus);
 
 	skl->init_done = 0; /* to be sure */
 
-	snd_hdac_ext_stop_streams(ebus);
+	snd_hdac_ext_stop_streams(bus);
 
 	if (bus->irq >= 0)
-		free_irq(bus->irq, (void *)ebus);
+		free_irq(bus->irq, (void *)bus);
 	snd_hdac_bus_free_stream_pages(bus);
-	snd_hdac_stream_free_all(ebus);
-	snd_hdac_link_free_all(ebus);
+	snd_hdac_stream_free_all(bus);
+	snd_hdac_link_free_all(bus);
 
 	if (bus->remap_addr)
 		iounmap(bus->remap_addr);
@@ -460,11 +446,11 @@ static int skl_free(struct hdac_ext_bus *ebus)
 	pci_release_regions(skl->pci);
 	pci_disable_device(skl->pci);
 
-	snd_hdac_ext_bus_exit(ebus);
+	snd_hdac_ext_bus_exit(bus);
 
 	cancel_work_sync(&skl->probe_work);
 	if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI))
-		snd_hdac_i915_exit(&ebus->bus);
+		snd_hdac_i915_exit(bus);
 
 	return 0;
 }
@@ -488,8 +474,8 @@ static struct skl_ssp_clk skl_ssp_clks[] = {
 
 static int skl_find_machine(struct skl *skl, void *driver_data)
 {
+	struct hdac_bus *bus = skl_to_bus(skl);
 	struct snd_soc_acpi_mach *mach = driver_data;
-	struct hdac_bus *bus = ebus_to_hbus(&skl->ebus);
 	struct skl_machine_pdata *pdata;
 
 	mach = snd_soc_acpi_find_machine(mach);
@@ -500,17 +486,19 @@ static int skl_find_machine(struct skl *skl, void *driver_data)
 
 	skl->mach = mach;
 	skl->fw_name = mach->fw_filename;
-	pdata = skl->mach->pdata;
+	pdata = mach->pdata;
 
-	if (mach->pdata)
+	if (pdata) {
 		skl->use_tplg_pcm = pdata->use_tplg_pcm;
+		pdata->dmic_num = skl_get_dmic_geo(skl);
+	}
 
 	return 0;
 }
 
 static int skl_machine_device_register(struct skl *skl)
 {
-	struct hdac_bus *bus = ebus_to_hbus(&skl->ebus);
+	struct hdac_bus *bus = skl_to_bus(skl);
 	struct snd_soc_acpi_mach *mach = skl->mach;
 	struct platform_device *pdev;
 	int ret;
@@ -544,7 +532,7 @@ static void skl_machine_device_unregister(struct skl *skl)
 
 static int skl_dmic_device_register(struct skl *skl)
 {
-	struct hdac_bus *bus = ebus_to_hbus(&skl->ebus);
+	struct hdac_bus *bus = skl_to_bus(skl);
 	struct platform_device *pdev;
 	int ret;
 
@@ -643,12 +631,13 @@ static void skl_clock_device_unregister(struct skl *skl)
 /*
  * Probe the given codec address
  */
-static int probe_codec(struct hdac_ext_bus *ebus, int addr)
+static int probe_codec(struct hdac_bus *bus, int addr)
 {
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
 	unsigned int cmd = (addr << 28) | (AC_NODE_ROOT << 20) |
 		(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
 	unsigned int res = -1;
+	struct skl *skl = bus_to_skl(bus);
+	struct hdac_device *hdev;
 
 	mutex_lock(&bus->cmd_mutex);
 	snd_hdac_bus_send_cmd(bus, cmd);
@@ -658,13 +647,16 @@ static int probe_codec(struct hdac_ext_bus *ebus, int addr)
 		return -EIO;
 	dev_dbg(bus->dev, "codec #%d probed OK\n", addr);
 
-	return snd_hdac_ext_bus_device_init(ebus, addr);
+	hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL);
+	if (!hdev)
+		return -ENOMEM;
+
+	return snd_hdac_ext_bus_device_init(bus, addr, hdev);
 }
 
 /* Codec initialization */
-static void skl_codec_create(struct hdac_ext_bus *ebus)
+static void skl_codec_create(struct hdac_bus *bus)
 {
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
 	int c, max_slots;
 
 	max_slots = HDA_MAX_CODECS;
@@ -672,7 +664,7 @@ static void skl_codec_create(struct hdac_ext_bus *ebus)
 	/* First try to probe all given codec slots */
 	for (c = 0; c < max_slots; c++) {
 		if ((bus->codec_mask & (1 << c))) {
-			if (probe_codec(ebus, c) < 0) {
+			if (probe_codec(bus, c) < 0) {
 				/*
 				 * Some BIOSen give you wrong codec addresses
 				 * that don't exist
@@ -722,8 +714,7 @@ static int skl_i915_init(struct hdac_bus *bus)
 static void skl_probe_work(struct work_struct *work)
 {
 	struct skl *skl = container_of(work, struct skl, probe_work);
-	struct hdac_ext_bus *ebus = &skl->ebus;
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = skl_to_bus(skl);
 	struct hdac_ext_link *hlink = NULL;
 	int err;
 
@@ -744,7 +735,7 @@ static void skl_probe_work(struct work_struct *work)
 		dev_info(bus->dev, "no hda codecs found!\n");
 
 	/* create codec instances */
-	skl_codec_create(ebus);
+	skl_codec_create(bus);
 
 	/* register platform dai and controls */
 	err = skl_platform_register(bus->dev);
@@ -773,8 +764,8 @@ static void skl_probe_work(struct work_struct *work)
 	/*
 	 * we are done probing so decrement link counts
 	 */
-	list_for_each_entry(hlink, &ebus->hlink_list, list)
-		snd_hdac_ext_bus_link_put(ebus, hlink);
+	list_for_each_entry(hlink, &bus->hlink_list, list)
+		snd_hdac_ext_bus_link_put(bus, hlink);
 
 	/* configure PM */
 	pm_runtime_put_noidle(bus->dev);
@@ -796,7 +787,7 @@ static int skl_create(struct pci_dev *pci,
 		      struct skl **rskl)
 {
 	struct skl *skl;
-	struct hdac_ext_bus *ebus;
+	struct hdac_bus *bus;
 
 	int err;
 
@@ -811,23 +802,22 @@ static int skl_create(struct pci_dev *pci,
 		pci_disable_device(pci);
 		return -ENOMEM;
 	}
-	ebus = &skl->ebus;
-	snd_hdac_ext_bus_init(ebus, &pci->dev, &bus_core_ops, io_ops);
-	ebus->bus.use_posbuf = 1;
+
+	bus = skl_to_bus(skl);
+	snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, NULL);
+	bus->use_posbuf = 1;
 	skl->pci = pci;
 	INIT_WORK(&skl->probe_work, skl_probe_work);
-
-	ebus->bus.bdl_pos_adj = 0;
+	bus->bdl_pos_adj = 0;
 
 	*rskl = skl;
 
 	return 0;
 }
 
-static int skl_first_init(struct hdac_ext_bus *ebus)
+static int skl_first_init(struct hdac_bus *bus)
 {
-	struct skl *skl = ebus_to_skl(ebus);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct skl *skl = bus_to_skl(bus);
 	struct pci_dev *pci = skl->pci;
 	int err;
 	unsigned short gcap;
@@ -848,7 +838,7 @@ static int skl_first_init(struct hdac_ext_bus *ebus)
 
 	snd_hdac_bus_parse_capabilities(bus);
 
-	if (skl_acquire_irq(ebus, 0) < 0)
+	if (skl_acquire_irq(bus, 0) < 0)
 		return -EBUSY;
 
 	pci_set_master(pci);
@@ -872,14 +862,14 @@ static int skl_first_init(struct hdac_ext_bus *ebus)
 	if (!pb_streams && !cp_streams)
 		return -EIO;
 
-	ebus->num_streams = cp_streams + pb_streams;
+	bus->num_streams = cp_streams + pb_streams;
 
 	/* initialize streams */
 	snd_hdac_ext_stream_init_all
-		(ebus, 0, cp_streams, SNDRV_PCM_STREAM_CAPTURE);
+		(bus, 0, cp_streams, SNDRV_PCM_STREAM_CAPTURE);
 	start_idx = cp_streams;
 	snd_hdac_ext_stream_init_all
-		(ebus, start_idx, pb_streams, SNDRV_PCM_STREAM_PLAYBACK);
+		(bus, start_idx, pb_streams, SNDRV_PCM_STREAM_PLAYBACK);
 
 	err = snd_hdac_bus_alloc_stream_pages(bus);
 	if (err < 0)
@@ -895,7 +885,6 @@ static int skl_probe(struct pci_dev *pci,
 		     const struct pci_device_id *pci_id)
 {
 	struct skl *skl;
-	struct hdac_ext_bus *ebus = NULL;
 	struct hdac_bus *bus = NULL;
 	int err;
 
@@ -904,10 +893,9 @@ static int skl_probe(struct pci_dev *pci,
 	if (err < 0)
 		return err;
 
-	ebus = &skl->ebus;
-	bus = ebus_to_hbus(ebus);
+	bus = skl_to_bus(skl);
 
-	err = skl_first_init(ebus);
+	err = skl_first_init(bus);
 	if (err < 0)
 		goto out_free;
 
@@ -928,9 +916,7 @@ static int skl_probe(struct pci_dev *pci,
 
 	skl_nhlt_update_topology_bin(skl);
 
-	pci_set_drvdata(skl->pci, ebus);
-
-	skl_dmic_data.dmic_num = skl_get_dmic_geo(skl);
+	pci_set_drvdata(skl->pci, bus);
 
 	/* check if dsp is there */
 	if (bus->ppcap) {
@@ -952,7 +938,7 @@ static int skl_probe(struct pci_dev *pci,
 		skl->skl_sst->clock_power_gating = skl_clock_power_gating;
 	}
 	if (bus->mlcap)
-		snd_hdac_ext_bus_get_ml_capabilities(ebus);
+		snd_hdac_ext_bus_get_ml_capabilities(bus);
 
 	snd_hdac_bus_stop_chip(bus);
 
@@ -972,31 +958,30 @@ out_clk_free:
 out_nhlt_free:
 	skl_nhlt_free(skl->nhlt);
 out_free:
-	skl_free(ebus);
+	skl_free(bus);
 
 	return err;
 }
 
 static void skl_shutdown(struct pci_dev *pci)
 {
-	struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
-	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_bus *bus = pci_get_drvdata(pci);
 	struct hdac_stream *s;
 	struct hdac_ext_stream *stream;
 	struct skl *skl;
 
-	if (ebus == NULL)
+	if (!bus)
 		return;
 
-	skl = ebus_to_skl(ebus);
+	skl = bus_to_skl(bus);
 
 	if (!skl->init_done)
 		return;
 
-	snd_hdac_ext_stop_streams(ebus);
+	snd_hdac_ext_stop_streams(bus);
 	list_for_each_entry(s, &bus->stream_list, list) {
 		stream = stream_to_hdac_ext_stream(s);
-		snd_hdac_ext_stream_decouple(ebus, stream, false);
+		snd_hdac_ext_stream_decouple(bus, stream, false);
 	}
 
 	snd_hdac_bus_stop_chip(bus);
@@ -1004,15 +989,15 @@ static void skl_shutdown(struct pci_dev *pci)
 
 static void skl_remove(struct pci_dev *pci)
 {
-	struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
-	struct skl *skl = ebus_to_skl(ebus);
+	struct hdac_bus *bus = pci_get_drvdata(pci);
+	struct skl *skl = bus_to_skl(bus);
 
 	release_firmware(skl->tplg);
 
 	pm_runtime_get_noresume(&pci->dev);
 
 	/* codec removal, invoke bus_device_remove */
-	snd_hdac_ext_bus_device_remove(ebus);
+	snd_hdac_ext_bus_device_remove(bus);
 
 	skl->debugfs = NULL;
 	skl_platform_unregister(&pci->dev);
@@ -1022,176 +1007,27 @@ static void skl_remove(struct pci_dev *pci)
 	skl_clock_device_unregister(skl);
 	skl_nhlt_remove_sysfs(skl);
 	skl_nhlt_free(skl->nhlt);
-	skl_free(ebus);
+	skl_free(bus);
 	dev_set_drvdata(&pci->dev, NULL);
 }
 
-static struct snd_soc_acpi_codecs skl_codecs = {
-	.num_codecs = 1,
-	.codecs = {"10508825"}
-};
-
-static struct snd_soc_acpi_codecs kbl_codecs = {
-	.num_codecs = 1,
-	.codecs = {"10508825"}
-};
-
-static struct snd_soc_acpi_codecs bxt_codecs = {
-	.num_codecs = 1,
-	.codecs = {"MX98357A"}
-};
-
-static struct snd_soc_acpi_codecs kbl_poppy_codecs = {
-	.num_codecs = 1,
-	.codecs = {"10EC5663"}
-};
-
-static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = {
-	.num_codecs = 2,
-	.codecs = {"10EC5663", "10EC5514"}
-};
-
-static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = {
-	.num_codecs = 1,
-	.codecs = {"MX98357A"}
-};
-
-static struct skl_machine_pdata cnl_pdata = {
-	.use_tplg_pcm = true,
-};
-
-static struct snd_soc_acpi_mach sst_skl_devdata[] = {
-	{
-		.id = "INT343A",
-		.drv_name = "skl_alc286s_i2s",
-		.fw_filename = "intel/dsp_fw_release.bin",
-	},
-	{
-		.id = "INT343B",
-		.drv_name = "skl_n88l25_s4567",
-		.fw_filename = "intel/dsp_fw_release.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &skl_codecs,
-		.pdata = &skl_dmic_data
-	},
-	{
-		.id = "MX98357A",
-		.drv_name = "skl_n88l25_m98357a",
-		.fw_filename = "intel/dsp_fw_release.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &skl_codecs,
-		.pdata = &skl_dmic_data
-	},
-	{}
-};
-
-static struct snd_soc_acpi_mach sst_bxtp_devdata[] = {
-	{
-		.id = "INT343A",
-		.drv_name = "bxt_alc298s_i2s",
-		.fw_filename = "intel/dsp_fw_bxtn.bin",
-	},
-	{
-		.id = "DLGS7219",
-		.drv_name = "bxt_da7219_max98357a_i2s",
-		.fw_filename = "intel/dsp_fw_bxtn.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &bxt_codecs,
-	},
-	{}
-};
-
-static struct snd_soc_acpi_mach sst_kbl_devdata[] = {
-	{
-		.id = "INT343A",
-		.drv_name = "kbl_alc286s_i2s",
-		.fw_filename = "intel/dsp_fw_kbl.bin",
-	},
-	{
-		.id = "INT343B",
-		.drv_name = "kbl_n88l25_s4567",
-		.fw_filename = "intel/dsp_fw_kbl.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &kbl_codecs,
-		.pdata = &skl_dmic_data
-	},
-	{
-		.id = "MX98357A",
-		.drv_name = "kbl_n88l25_m98357a",
-		.fw_filename = "intel/dsp_fw_kbl.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &kbl_codecs,
-		.pdata = &skl_dmic_data
-	},
-	{
-		.id = "MX98927",
-		.drv_name = "kbl_r5514_5663_max",
-		.fw_filename = "intel/dsp_fw_kbl.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &kbl_5663_5514_codecs,
-		.pdata = &skl_dmic_data
-	},
-	{
-		.id = "MX98927",
-		.drv_name = "kbl_rt5663_m98927",
-		.fw_filename = "intel/dsp_fw_kbl.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &kbl_poppy_codecs,
-		.pdata = &skl_dmic_data
-	},
-	{
-		.id = "10EC5663",
-		.drv_name = "kbl_rt5663",
-		.fw_filename = "intel/dsp_fw_kbl.bin",
-	},
-	{
-		.id = "DLGS7219",
-		.drv_name = "kbl_da7219_max98357a",
-		.fw_filename = "intel/dsp_fw_kbl.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &kbl_7219_98357_codecs,
-		.pdata = &skl_dmic_data
-	},
-
-	{}
-};
-
-static struct snd_soc_acpi_mach sst_glk_devdata[] = {
-	{
-		.id = "INT343A",
-		.drv_name = "glk_alc298s_i2s",
-		.fw_filename = "intel/dsp_fw_glk.bin",
-	},
-	{}
-};
-
-static const struct snd_soc_acpi_mach sst_cnl_devdata[] = {
-	{
-		.id = "INT34C2",
-		.drv_name = "cnl_rt274",
-		.fw_filename = "intel/dsp_fw_cnl.bin",
-		.pdata = &cnl_pdata,
-	},
-	{}
-};
-
 /* PCI IDs */
 static const struct pci_device_id skl_ids[] = {
 	/* Sunrise Point-LP */
 	{ PCI_DEVICE(0x8086, 0x9d70),
-		.driver_data = (unsigned long)&sst_skl_devdata},
+		.driver_data = (unsigned long)&snd_soc_acpi_intel_skl_machines},
 	/* BXT-P */
 	{ PCI_DEVICE(0x8086, 0x5a98),
-		.driver_data = (unsigned long)&sst_bxtp_devdata},
+		.driver_data = (unsigned long)&snd_soc_acpi_intel_bxt_machines},
 	/* KBL */
 	{ PCI_DEVICE(0x8086, 0x9D71),
-		.driver_data = (unsigned long)&sst_kbl_devdata},
+		.driver_data = (unsigned long)&snd_soc_acpi_intel_kbl_machines},
 	/* GLK */
 	{ PCI_DEVICE(0x8086, 0x3198),
-		.driver_data = (unsigned long)&sst_glk_devdata},
+		.driver_data = (unsigned long)&snd_soc_acpi_intel_glk_machines},
 	/* CNL */
 	{ PCI_DEVICE(0x8086, 0x9dc8),
-		.driver_data = (unsigned long)&sst_cnl_devdata},
+		.driver_data = (unsigned long)&snd_soc_acpi_intel_cnl_machines},
 	{ 0, }
 };
 MODULE_DEVICE_TABLE(pci, skl_ids);
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index 0d5375cbcf6e..78aa8bdcb619 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -71,7 +71,7 @@ struct skl_fw_config {
 };
 
 struct skl {
-	struct hdac_ext_bus ebus;
+	struct hdac_bus hbus;
 	struct pci_dev *pci;
 
 	unsigned int init_done:1; /* delayed init status */
@@ -105,9 +105,8 @@ struct skl {
 	struct snd_soc_acpi_mach *mach;
 };
 
-#define skl_to_ebus(s)	(&(s)->ebus)
-#define ebus_to_skl(sbus) \
-	container_of(sbus, struct skl, sbus)
+#define skl_to_bus(s)  (&(s)->hbus)
+#define bus_to_skl(bus) container_of(bus, struct skl, hbus)
 
 /* to pass dai dma data */
 struct skl_dma_params {
diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c
index 51ec4ff6ed95..697aa50aff9a 100644
--- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c
+++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c
@@ -15,20 +15,12 @@
 
 int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe)
 {
-	struct snd_soc_dai_driver *sub_dai_drivers;
+	struct mtk_base_afe_dai *dai;
 	size_t num_dai_drivers = 0, dai_idx = 0;
-	int i;
-
-	if (!afe->sub_dais) {
-		dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__);
-		return -EINVAL;
-	}
 
 	/* calcualte total dai driver size */
-	for (i = 0; i < afe->num_sub_dais; i++) {
-		if (afe->sub_dais[i].dai_drivers &&
-		    afe->sub_dais[i].num_dai_drivers != 0)
-			num_dai_drivers += afe->sub_dais[i].num_dai_drivers;
+	list_for_each_entry(dai, &afe->sub_dais, list) {
+		num_dai_drivers += dai->num_dai_drivers;
 	}
 
 	dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers);
@@ -42,19 +34,14 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe)
 	if (!afe->dai_drivers)
 		return -ENOMEM;
 
-	for (i = 0; i < afe->num_sub_dais; i++) {
-		if (afe->sub_dais[i].dai_drivers &&
-		    afe->sub_dais[i].num_dai_drivers != 0) {
-			sub_dai_drivers = afe->sub_dais[i].dai_drivers;
-			/* dai driver */
-			memcpy(&afe->dai_drivers[dai_idx],
-			       sub_dai_drivers,
-			       afe->sub_dais[i].num_dai_drivers *
-			       sizeof(struct snd_soc_dai_driver));
-			dai_idx += afe->sub_dais[i].num_dai_drivers;
-		}
+	list_for_each_entry(dai, &afe->sub_dais, list) {
+		/* dai driver */
+		memcpy(&afe->dai_drivers[dai_idx],
+		       dai->dai_drivers,
+		       dai->num_dai_drivers *
+		       sizeof(struct snd_soc_dai_driver));
+		dai_idx += dai->num_dai_drivers;
 	}
-
 	return 0;
 }
 EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai);
@@ -62,28 +49,25 @@ EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai);
 int mtk_afe_add_sub_dai_control(struct snd_soc_component *component)
 {
 	struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
-	int i;
+	struct mtk_base_afe_dai *dai;
 
-	if (!afe->sub_dais) {
-		dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__);
-		return -EINVAL;
-	}
-
-	for (i = 0; i < afe->num_sub_dais; i++) {
-		if (afe->sub_dais[i].controls)
+	list_for_each_entry(dai, &afe->sub_dais, list) {
+		if (dai->controls)
 			snd_soc_add_component_controls(component,
-				afe->sub_dais[i].controls,
-				afe->sub_dais[i].num_controls);
+						       dai->controls,
+						       dai->num_controls);
 
-		if (afe->sub_dais[i].dapm_widgets)
+		if (dai->dapm_widgets)
 			snd_soc_dapm_new_controls(&component->dapm,
-				afe->sub_dais[i].dapm_widgets,
-				afe->sub_dais[i].num_dapm_widgets);
-
-		if (afe->sub_dais[i].dapm_routes)
+						  dai->dapm_widgets,
+						  dai->num_dapm_widgets);
+	}
+	/* add routes after all widgets are added */
+	list_for_each_entry(dai, &afe->sub_dais, list) {
+		if (dai->dapm_routes)
 			snd_soc_dapm_add_routes(&component->dapm,
-				afe->sub_dais[i].dapm_routes,
-				afe->sub_dais[i].num_dapm_routes);
+						dai->dapm_routes,
+						dai->num_dapm_routes);
 	}
 
 	snd_soc_dapm_new_widgets(component->dapm.card);
diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h
index bcf562f029b6..bd8d5e0c6843 100644
--- a/sound/soc/mediatek/common/mtk-base-afe.h
+++ b/sound/soc/mediatek/common/mtk-base-afe.h
@@ -46,6 +46,7 @@ struct mtk_base_irq_data {
 };
 
 struct device;
+struct list_head;
 struct mtk_base_afe_memif;
 struct mtk_base_afe_irq;
 struct mtk_base_afe_dai;
@@ -72,8 +73,7 @@ struct mtk_base_afe {
 	struct mtk_base_afe_irq *irqs;
 	int irqs_size;
 
-	struct mtk_base_afe_dai *sub_dais;
-	int num_sub_dais;
+	struct list_head sub_dais;
 	struct snd_soc_dai_driver *dai_drivers;
 	unsigned int num_dai_drivers;
 
@@ -110,6 +110,8 @@ struct mtk_base_afe_dai {
 	unsigned int num_dapm_widgets;
 	const struct snd_soc_dapm_route *dapm_routes;
 	unsigned int num_dapm_routes;
+
+	struct list_head list;
 };
 
 #endif
diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-common.h b/sound/soc/mediatek/mt6797/mt6797-afe-common.h
index 22eb7b455cf1..4eac9977b2b0 100644
--- a/sound/soc/mediatek/mt6797/mt6797-afe-common.h
+++ b/sound/soc/mediatek/mt6797/mt6797-afe-common.h
@@ -10,6 +10,7 @@
 #define _MT_6797_AFE_COMMON_H_
 
 #include <sound/soc.h>
+#include <linux/list.h>
 #include <linux/regmap.h>
 #include "../common/mtk-base-afe.h"
 
diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
index 6c5dd9fc9976..192f4d7b37b6 100644
--- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
+++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
@@ -733,6 +733,34 @@ static const struct snd_soc_component_driver mt6797_afe_component = {
 	.probe = mt6797_afe_component_probe,
 };
 
+static int mt6797_dai_memif_register(struct mtk_base_afe *afe)
+{
+	struct mtk_base_afe_dai *dai;
+
+	dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL);
+	if (!dai)
+		return -ENOMEM;
+
+	list_add(&dai->list, &afe->sub_dais);
+
+	dai->dai_drivers = mt6797_memif_dai_driver;
+	dai->num_dai_drivers = ARRAY_SIZE(mt6797_memif_dai_driver);
+
+	dai->dapm_widgets = mt6797_memif_widgets;
+	dai->num_dapm_widgets = ARRAY_SIZE(mt6797_memif_widgets);
+	dai->dapm_routes = mt6797_memif_routes;
+	dai->num_dapm_routes = ARRAY_SIZE(mt6797_memif_routes);
+	return 0;
+}
+
+typedef int (*dai_register_cb)(struct mtk_base_afe *);
+static const dai_register_cb dai_register_cbs[] = {
+	mt6797_dai_adda_register,
+	mt6797_dai_pcm_register,
+	mt6797_dai_hostless_register,
+	mt6797_dai_memif_register,
+};
+
 static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev)
 {
 	struct mtk_base_afe *afe;
@@ -811,29 +839,24 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev)
 	}
 
 	/* init sub_dais */
-	afe->num_sub_dais = MT6797_DAI_NUM;
-	afe->sub_dais = devm_kcalloc(dev, afe->num_sub_dais,
-				     sizeof(*afe->sub_dais),
-				     GFP_KERNEL);
-	if (!afe->sub_dais)
-		return -ENOMEM;
-
-	mt6797_dai_adda_register(afe);
-	mt6797_dai_pcm_register(afe);
-	mt6797_dai_hostless_register(afe);
-
-	afe->sub_dais[MT6797_MEMIF_DL1].dai_drivers = mt6797_memif_dai_driver;
-	afe->sub_dais[MT6797_MEMIF_DL1].num_dai_drivers =
-		ARRAY_SIZE(mt6797_memif_dai_driver);
-	afe->sub_dais[MT6797_MEMIF_DL1].dapm_widgets = mt6797_memif_widgets;
-	afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_widgets =
-		ARRAY_SIZE(mt6797_memif_widgets);
-	afe->sub_dais[MT6797_MEMIF_DL1].dapm_routes = mt6797_memif_routes;
-	afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_routes =
-		ARRAY_SIZE(mt6797_memif_routes);
+	INIT_LIST_HEAD(&afe->sub_dais);
+
+	for (i = 0; i < ARRAY_SIZE(dai_register_cbs); i++) {
+		ret = dai_register_cbs[i](afe);
+		if (ret) {
+			dev_warn(afe->dev, "dai register i %d fail, ret %d\n",
+				 i, ret);
+			return ret;
+		}
+	}
 
 	/* init dai_driver and component_driver */
-	mtk_afe_combine_sub_dai(afe);
+	ret = mtk_afe_combine_sub_dai(afe);
+	if (ret) {
+		dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n",
+			 ret);
+		return ret;
+	}
 
 	afe->mtk_afe_hardware = &mt6797_afe_hardware;
 	afe->memif_fs = mt6797_memif_fs;
diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c
index ad083265ce94..0ac6409c6d61 100644
--- a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c
+++ b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c
@@ -383,14 +383,20 @@ static struct snd_soc_dai_driver mtk_dai_adda_driver[] = {
 
 int mt6797_dai_adda_register(struct mtk_base_afe *afe)
 {
-	int id = MT6797_DAI_ADDA;
+	struct mtk_base_afe_dai *dai;
 
-	afe->sub_dais[id].dai_drivers = mtk_dai_adda_driver;
-	afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver);
+	dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL);
+	if (!dai)
+		return -ENOMEM;
 
-	afe->sub_dais[id].dapm_widgets = mtk_dai_adda_widgets;
-	afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets);
-	afe->sub_dais[id].dapm_routes = mtk_dai_adda_routes;
-	afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes);
+	list_add(&dai->list, &afe->sub_dais);
+
+	dai->dai_drivers = mtk_dai_adda_driver;
+	dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver);
+
+	dai->dapm_widgets = mtk_dai_adda_widgets;
+	dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets);
+	dai->dapm_routes = mtk_dai_adda_routes;
+	dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes);
 	return 0;
 }
diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c
index 4cf985b15a11..ed23e6a53b08 100644
--- a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c
+++ b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c
@@ -100,13 +100,19 @@ static struct snd_soc_dai_driver mtk_dai_hostless_driver[] = {
 
 int mt6797_dai_hostless_register(struct mtk_base_afe *afe)
 {
-	int id = MT6797_DAI_HOSTLESS_LPBK;
+	struct mtk_base_afe_dai *dai;
 
-	afe->sub_dais[id].dai_drivers = mtk_dai_hostless_driver;
-	afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver);
+	dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL);
+	if (!dai)
+		return -ENOMEM;
 
-	afe->sub_dais[id].dapm_routes = mtk_dai_hostless_routes;
-	afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes);
+	list_add(&dai->list, &afe->sub_dais);
+
+	dai->dai_drivers = mtk_dai_hostless_driver;
+	dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver);
+
+	dai->dapm_routes = mtk_dai_hostless_routes;
+	dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes);
 
 	return 0;
 }
diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c
index 16d5b5067204..3136f0bc7827 100644
--- a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c
+++ b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c
@@ -298,15 +298,20 @@ static struct snd_soc_dai_driver mtk_dai_pcm_driver[] = {
 
 int mt6797_dai_pcm_register(struct mtk_base_afe *afe)
 {
-	int id = MT6797_DAI_PCM_1;
+	struct mtk_base_afe_dai *dai;
 
-	afe->sub_dais[id].dai_drivers = mtk_dai_pcm_driver;
-	afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver);
+	dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL);
+	if (!dai)
+		return -ENOMEM;
 
-	afe->sub_dais[id].dapm_widgets = mtk_dai_pcm_widgets;
-	afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets);
-	afe->sub_dais[id].dapm_routes = mtk_dai_pcm_routes;
-	afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes);
+	list_add(&dai->list, &afe->sub_dais);
 
+	dai->dai_drivers = mtk_dai_pcm_driver;
+	dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver);
+
+	dai->dapm_widgets = mtk_dai_pcm_widgets;
+	dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets);
+	dai->dapm_routes = mtk_dai_pcm_routes;
+	dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes);
 	return 0;
 }
diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig
new file mode 100644
index 000000000000..8af8bc358a90
--- /dev/null
+++ b/sound/soc/meson/Kconfig
@@ -0,0 +1,65 @@
+menu "ASoC support for Amlogic platforms"
+	depends on ARCH_MESON || COMPILE_TEST
+
+config SND_MESON_AXG_FIFO
+	tristate
+	select REGMAP_MMIO
+
+config SND_MESON_AXG_FRDDR
+	tristate "Amlogic AXG Playback FIFO support"
+	select SND_MESON_AXG_FIFO
+	help
+	  Select Y or M to add support for the frontend playback interfaces
+	  embedded in the Amlogic AXG SoC family
+
+config SND_MESON_AXG_TODDR
+	tristate "Amlogic AXG Capture FIFO support"
+	select SND_MESON_AXG_FIFO
+	help
+	  Select Y or M to add support for the frontend capture interfaces
+	  embedded in the Amlogic AXG SoC family
+
+config SND_MESON_AXG_TDM_FORMATTER
+	tristate
+	select REGMAP_MMIO
+
+config SND_MESON_AXG_TDM_INTERFACE
+	tristate
+	select SND_MESON_AXG_TDM_FORMATTER
+
+config SND_MESON_AXG_TDMIN
+	tristate "Amlogic AXG TDM Input Support"
+	select SND_MESON_AXG_TDM_FORMATTER
+	select SND_MESON_AXG_TDM_INTERFACE
+	help
+	  Select Y or M to add support for TDM input formatter embedded
+	  in the Amlogic AXG SoC family
+
+config SND_MESON_AXG_TDMOUT
+	tristate "Amlogic AXG TDM Output Support"
+	select SND_MESON_AXG_TDM_FORMATTER
+	select SND_MESON_AXG_TDM_INTERFACE
+	help
+	  Select Y or M to add support for TDM output formatter embedded
+	  in the Amlogic AXG SoC family
+
+config SND_MESON_AXG_SOUND_CARD
+	tristate "Amlogic AXG Sound Card Support"
+	select SND_MESON_AXG_TDM_INTERFACE
+	imply SND_MESON_AXG_FRDDR
+	imply SND_MESON_AXG_TODDR
+	imply SND_MESON_AXG_TDMIN
+	imply SND_MESON_AXG_TDMOUT
+	imply SND_MESON_AXG_SPDIFOUT
+	help
+	  Select Y or M to add support for the AXG SoC sound card
+
+config SND_MESON_AXG_SPDIFOUT
+	tristate "Amlogic AXG SPDIF Output Support"
+	select SND_PCM_IEC958
+	imply SND_SOC_SPDIF
+	help
+	  Select Y or M to add support for SPDIF output serializer embedded
+	  in the Amlogic AXG SoC family
+
+endmenu
diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile
new file mode 100644
index 000000000000..c5e003b093db
--- /dev/null
+++ b/sound/soc/meson/Makefile
@@ -0,0 +1,21 @@
+# SPDX-License-Identifier: (GPL-2.0 OR MIT)
+
+snd-soc-meson-axg-fifo-objs := axg-fifo.o
+snd-soc-meson-axg-frddr-objs := axg-frddr.o
+snd-soc-meson-axg-toddr-objs := axg-toddr.o
+snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o
+snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o
+snd-soc-meson-axg-tdmin-objs := axg-tdmin.o
+snd-soc-meson-axg-tdmout-objs := axg-tdmout.o
+snd-soc-meson-axg-sound-card-objs := axg-card.o
+snd-soc-meson-axg-spdifout-objs := axg-spdifout.o
+
+obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o
+obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o
+obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o
+obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o
+obj-$(CONFIG_SND_MESON_AXG_TDM_INTERFACE) += snd-soc-meson-axg-tdm-interface.o
+obj-$(CONFIG_SND_MESON_AXG_TDMIN) += snd-soc-meson-axg-tdmin.o
+obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o
+obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o
+obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
new file mode 100644
index 000000000000..2914ba0d965b
--- /dev/null
+++ b/sound/soc/meson/axg-card.c
@@ -0,0 +1,671 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-tdm.h"
+
+struct axg_card {
+	struct snd_soc_card card;
+	void **link_data;
+};
+
+struct axg_dai_link_tdm_mask {
+	u32 tx;
+	u32 rx;
+};
+
+struct axg_dai_link_tdm_data {
+	unsigned int mclk_fs;
+	unsigned int slots;
+	unsigned int slot_width;
+	u32 *tx_mask;
+	u32 *rx_mask;
+	struct axg_dai_link_tdm_mask *codec_masks;
+};
+
+#define PREFIX "amlogic,"
+
+static int axg_card_reallocate_links(struct axg_card *priv,
+				     unsigned int num_links)
+{
+	struct snd_soc_dai_link *links;
+	void **ldata;
+
+	links = krealloc(priv->card.dai_link,
+			 num_links * sizeof(*priv->card.dai_link),
+			 GFP_KERNEL | __GFP_ZERO);
+	ldata = krealloc(priv->link_data,
+			 num_links * sizeof(*priv->link_data),
+			 GFP_KERNEL | __GFP_ZERO);
+
+	if (!links || !ldata) {
+		dev_err(priv->card.dev, "failed to allocate links\n");
+		return -ENOMEM;
+	}
+
+	priv->card.dai_link = links;
+	priv->link_data = ldata;
+	priv->card.num_links = num_links;
+	return 0;
+}
+
+static int axg_card_parse_dai(struct snd_soc_card *card,
+			      struct device_node *node,
+			      struct device_node **dai_of_node,
+			      const char **dai_name)
+{
+	struct of_phandle_args args;
+	int ret;
+
+	if (!dai_name || !dai_of_node || !node)
+		return -EINVAL;
+
+	ret = of_parse_phandle_with_args(node, "sound-dai",
+					 "#sound-dai-cells", 0, &args);
+	if (ret) {
+		if (ret != -EPROBE_DEFER)
+			dev_err(card->dev, "can't parse dai %d\n", ret);
+		return ret;
+	}
+	*dai_of_node = args.np;
+
+	return snd_soc_get_dai_name(&args, dai_name);
+}
+
+static int axg_card_set_link_name(struct snd_soc_card *card,
+				  struct snd_soc_dai_link *link,
+				  const char *prefix)
+{
+	char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s",
+				    prefix, link->cpu_of_node->full_name);
+	if (!name)
+		return -ENOMEM;
+
+	link->name = name;
+	link->stream_name = name;
+
+	return 0;
+}
+
+static void axg_card_clean_references(struct axg_card *priv)
+{
+	struct snd_soc_card *card = &priv->card;
+	struct snd_soc_dai_link *link;
+	int i, j;
+
+	if (card->dai_link) {
+		for (i = 0; i < card->num_links; i++) {
+			link = &card->dai_link[i];
+			of_node_put(link->cpu_of_node);
+			for (j = 0; j < link->num_codecs; j++)
+				of_node_put(link->codecs[j].of_node);
+		}
+	}
+
+	if (card->aux_dev) {
+		for (i = 0; i < card->num_aux_devs; i++)
+			of_node_put(card->aux_dev[i].codec_of_node);
+	}
+
+	kfree(card->dai_link);
+	kfree(priv->link_data);
+}
+
+static int axg_card_add_aux_devices(struct snd_soc_card *card)
+{
+	struct device_node *node = card->dev->of_node;
+	struct snd_soc_aux_dev *aux;
+	int num, i;
+
+	num = of_count_phandle_with_args(node, "audio-aux-devs", NULL);
+	if (num == -ENOENT) {
+		/*
+		 * It is ok to have no auxiliary devices but for this card it
+		 * is a strange situtation. Let's warn the about it.
+		 */
+		dev_warn(card->dev, "card has no auxiliary devices\n");
+		return 0;
+	} else if (num < 0) {
+		dev_err(card->dev, "error getting auxiliary devices: %d\n",
+			num);
+		return num;
+	}
+
+	aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL);
+	if (!aux)
+		return -ENOMEM;
+	card->aux_dev = aux;
+	card->num_aux_devs = num;
+
+	for (i = 0; i < card->num_aux_devs; i++, aux++) {
+		aux->codec_of_node =
+			of_parse_phandle(node, "audio-aux-devs", i);
+		if (!aux->codec_of_node)
+			return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream,
+				     struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct axg_dai_link_tdm_data *be =
+		(struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
+	struct snd_soc_dai *codec_dai;
+	unsigned int mclk;
+	int ret, i;
+
+	if (be->mclk_fs) {
+		mclk = params_rate(params) * be->mclk_fs;
+
+		for (i = 0; i < rtd->num_codecs; i++) {
+			codec_dai = rtd->codec_dais[i];
+			ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
+						     SND_SOC_CLOCK_IN);
+			if (ret && ret != -ENOTSUPP)
+				return ret;
+		}
+
+		ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk,
+					     SND_SOC_CLOCK_OUT);
+		if (ret && ret != -ENOTSUPP)
+			return ret;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_ops axg_card_tdm_be_ops = {
+	.hw_params = axg_card_tdm_be_hw_params,
+};
+
+static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct axg_dai_link_tdm_data *be =
+		(struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
+	struct snd_soc_dai *codec_dai;
+	int ret, i;
+
+	for (i = 0; i < rtd->num_codecs; i++) {
+		codec_dai = rtd->codec_dais[i];
+		ret = snd_soc_dai_set_tdm_slot(codec_dai,
+					       be->codec_masks[i].tx,
+					       be->codec_masks[i].rx,
+					       be->slots, be->slot_width);
+		if (ret && ret != -ENOTSUPP) {
+			dev_err(codec_dai->dev,
+				"setting tdm link slots failed\n");
+			return ret;
+		}
+	}
+
+	ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, be->tx_mask, be->rx_mask,
+				    be->slots, be->slot_width);
+	if (ret) {
+		dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct axg_dai_link_tdm_data *be =
+		(struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
+	int ret;
+
+	/* The loopback rx_mask is the pad tx_mask */
+	ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, NULL, be->tx_mask,
+				    be->slots, be->slot_width);
+	if (ret) {
+		dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
+				     int *index)
+{
+	struct axg_card *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai_link *pad = &card->dai_link[*index];
+	struct snd_soc_dai_link *lb;
+	int ret;
+
+	/* extend links */
+	ret = axg_card_reallocate_links(priv, card->num_links + 1);
+	if (ret)
+		return ret;
+
+	lb = &card->dai_link[*index + 1];
+
+	lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name);
+	if (!lb->name)
+		return -ENOMEM;
+
+	lb->stream_name = lb->name;
+	lb->cpu_of_node = pad->cpu_of_node;
+	lb->cpu_dai_name = "TDM Loopback";
+	lb->codec_name = "snd-soc-dummy";
+	lb->codec_dai_name = "snd-soc-dummy-dai";
+	lb->dpcm_capture = 1;
+	lb->no_pcm = 1;
+	lb->ops = &axg_card_tdm_be_ops;
+	lb->init = axg_card_tdm_dai_lb_init;
+
+	/* Provide the same link data to the loopback */
+	priv->link_data[*index + 1] = priv->link_data[*index];
+
+	/*
+	 * axg_card_clean_references() will iterate over this link,
+	 * make sure the node count is balanced
+	 */
+	of_node_get(lb->cpu_of_node);
+
+	/* Let add_links continue where it should */
+	*index += 1;
+
+	return 0;
+}
+
+static unsigned int axg_card_parse_daifmt(struct device_node *node,
+					  struct device_node *cpu_node)
+{
+	struct device_node *bitclkmaster = NULL;
+	struct device_node *framemaster = NULL;
+	unsigned int daifmt;
+
+	daifmt = snd_soc_of_parse_daifmt(node, PREFIX,
+					 &bitclkmaster, &framemaster);
+	daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+
+	/* If no master is provided, default to cpu master */
+	if (!bitclkmaster || bitclkmaster == cpu_node) {
+		daifmt |= (!framemaster || framemaster == cpu_node) ?
+			SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM;
+	} else {
+		daifmt |= (!framemaster || framemaster == cpu_node) ?
+			SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM;
+	}
+
+	of_node_put(bitclkmaster);
+	of_node_put(framemaster);
+
+	return daifmt;
+}
+
+static int axg_card_parse_cpu_tdm_slots(struct snd_soc_card *card,
+					struct snd_soc_dai_link *link,
+					struct device_node *node,
+					struct axg_dai_link_tdm_data *be)
+{
+	char propname[32];
+	u32 tx, rx;
+	int i;
+
+	be->tx_mask = devm_kcalloc(card->dev, AXG_TDM_NUM_LANES,
+				   sizeof(*be->tx_mask), GFP_KERNEL);
+	be->rx_mask = devm_kcalloc(card->dev, AXG_TDM_NUM_LANES,
+				   sizeof(*be->rx_mask), GFP_KERNEL);
+	if (!be->tx_mask || !be->rx_mask)
+		return -ENOMEM;
+
+	for (i = 0, tx = 0; i < AXG_TDM_NUM_LANES; i++) {
+		snprintf(propname, 32, "dai-tdm-slot-tx-mask-%d", i);
+		snd_soc_of_get_slot_mask(node, propname, &be->tx_mask[i]);
+		tx = max(tx, be->tx_mask[i]);
+	}
+
+	/* Disable playback is the interface has no tx slots */
+	if (!tx)
+		link->dpcm_playback = 0;
+
+	for (i = 0, rx = 0; i < AXG_TDM_NUM_LANES; i++) {
+		snprintf(propname, 32, "dai-tdm-slot-rx-mask-%d", i);
+		snd_soc_of_get_slot_mask(node, propname, &be->rx_mask[i]);
+		rx = max(rx, be->rx_mask[i]);
+	}
+
+	/* Disable capture is the interface has no rx slots */
+	if (!rx)
+		link->dpcm_capture = 0;
+
+	/* ... but the interface should at least have one of them */
+	if (!tx && !rx) {
+		dev_err(card->dev, "tdm link has no cpu slots\n");
+		return -EINVAL;
+	}
+
+	of_property_read_u32(node, "dai-tdm-slot-num", &be->slots);
+	if (!be->slots) {
+		/*
+		 * If the slot number is not provided, set it such as it
+		 * accommodates the largest mask
+		 */
+		be->slots = fls(max(tx, rx));
+	} else if (be->slots < fls(max(tx, rx)) || be->slots > 32) {
+		/*
+		 * Error if the slots can't accommodate the largest mask or
+		 * if it is just too big
+		 */
+		dev_err(card->dev, "bad slot number\n");
+		return -EINVAL;
+	}
+
+	of_property_read_u32(node, "dai-tdm-slot-width", &be->slot_width);
+
+	return 0;
+}
+
+static int axg_card_parse_codecs_masks(struct snd_soc_card *card,
+				       struct snd_soc_dai_link *link,
+				       struct device_node *node,
+				       struct axg_dai_link_tdm_data *be)
+{
+	struct axg_dai_link_tdm_mask *codec_mask;
+	struct device_node *np;
+
+	codec_mask = devm_kcalloc(card->dev, link->num_codecs,
+				  sizeof(*codec_mask), GFP_KERNEL);
+	if (!codec_mask)
+		return -ENOMEM;
+
+	be->codec_masks = codec_mask;
+
+	for_each_child_of_node(node, np) {
+		snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask",
+					 &codec_mask->rx);
+		snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask",
+					 &codec_mask->tx);
+
+		codec_mask++;
+	}
+
+	return 0;
+}
+
+static int axg_card_parse_tdm(struct snd_soc_card *card,
+			      struct device_node *node,
+			      int *index)
+{
+	struct axg_card *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai_link *link = &card->dai_link[*index];
+	struct axg_dai_link_tdm_data *be;
+	int ret;
+
+	/* Allocate tdm link parameters */
+	be = devm_kzalloc(card->dev, sizeof(*be), GFP_KERNEL);
+	if (!be)
+		return -ENOMEM;
+	priv->link_data[*index] = be;
+
+	/* Setup tdm link */
+	link->ops = &axg_card_tdm_be_ops;
+	link->init = axg_card_tdm_dai_init;
+	link->dai_fmt = axg_card_parse_daifmt(node, link->cpu_of_node);
+
+	of_property_read_u32(node, "mclk-fs", &be->mclk_fs);
+
+	ret = axg_card_parse_cpu_tdm_slots(card, link, node, be);
+	if (ret) {
+		dev_err(card->dev, "error parsing tdm link slots\n");
+		return ret;
+	}
+
+	ret = axg_card_parse_codecs_masks(card, link, node, be);
+	if (ret)
+		return ret;
+
+	/* Add loopback if the pad dai has playback */
+	if (link->dpcm_playback) {
+		ret = axg_card_add_tdm_loopback(card, index);
+		if (ret)
+			return ret;
+	}
+
+	return 0;
+}
+
+static int axg_card_set_be_link(struct snd_soc_card *card,
+				struct snd_soc_dai_link *link,
+				struct device_node *node)
+{
+	struct snd_soc_dai_link_component *codec;
+	struct device_node *np;
+	int ret, num_codecs;
+
+	link->no_pcm = 1;
+	link->dpcm_playback = 1;
+	link->dpcm_capture = 1;
+
+	num_codecs = of_get_child_count(node);
+	if (!num_codecs) {
+		dev_err(card->dev, "be link %s has no codec\n",
+			node->full_name);
+		return -EINVAL;
+	}
+
+	codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL);
+	if (!codec)
+		return -ENOMEM;
+
+	link->codecs = codec;
+	link->num_codecs = num_codecs;
+
+	for_each_child_of_node(node, np) {
+		ret = axg_card_parse_dai(card, np, &codec->of_node,
+					 &codec->dai_name);
+		if (ret) {
+			of_node_put(np);
+			return ret;
+		}
+
+		codec++;
+	}
+
+	ret = axg_card_set_link_name(card, link, "be");
+	if (ret)
+		dev_err(card->dev, "error setting %s link name\n", np->name);
+
+	return ret;
+}
+
+static int axg_card_set_fe_link(struct snd_soc_card *card,
+				struct snd_soc_dai_link *link,
+				bool is_playback)
+{
+	link->dynamic = 1;
+	link->dpcm_merged_format = 1;
+	link->dpcm_merged_chan = 1;
+	link->dpcm_merged_rate = 1;
+	link->codec_dai_name = "snd-soc-dummy-dai";
+	link->codec_name = "snd-soc-dummy";
+
+	if (is_playback)
+		link->dpcm_playback = 1;
+	else
+		link->dpcm_capture = 1;
+
+	return axg_card_set_link_name(card, link, "fe");
+}
+
+static int axg_card_cpu_is_capture_fe(struct device_node *np)
+{
+	return of_device_is_compatible(np, PREFIX "axg-toddr");
+}
+
+static int axg_card_cpu_is_playback_fe(struct device_node *np)
+{
+	return of_device_is_compatible(np, PREFIX "axg-frddr");
+}
+
+static int axg_card_cpu_is_tdm_iface(struct device_node *np)
+{
+	return of_device_is_compatible(np, PREFIX "axg-tdm-iface");
+}
+
+static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np,
+			     int *index)
+{
+	struct snd_soc_dai_link *dai_link = &card->dai_link[*index];
+	int ret;
+
+	ret = axg_card_parse_dai(card, np, &dai_link->cpu_of_node,
+				 &dai_link->cpu_dai_name);
+	if (ret)
+		return ret;
+
+	if (axg_card_cpu_is_playback_fe(dai_link->cpu_of_node))
+		ret = axg_card_set_fe_link(card, dai_link, true);
+	else if (axg_card_cpu_is_capture_fe(dai_link->cpu_of_node))
+		ret = axg_card_set_fe_link(card, dai_link, false);
+	else
+		ret = axg_card_set_be_link(card, dai_link, np);
+
+	if (ret)
+		return ret;
+
+	if (axg_card_cpu_is_tdm_iface(dai_link->cpu_of_node))
+		ret = axg_card_parse_tdm(card, np, index);
+
+	return ret;
+}
+
+static int axg_card_add_links(struct snd_soc_card *card)
+{
+	struct axg_card *priv = snd_soc_card_get_drvdata(card);
+	struct device_node *node = card->dev->of_node;
+	struct device_node *np;
+	int num, i, ret;
+
+	num = of_get_child_count(node);
+	if (!num) {
+		dev_err(card->dev, "card has no links\n");
+		return -EINVAL;
+	}
+
+	ret = axg_card_reallocate_links(priv, num);
+	if (ret)
+		return ret;
+
+	i = 0;
+	for_each_child_of_node(node, np) {
+		ret = axg_card_add_link(card, np, &i);
+		if (ret) {
+			of_node_put(np);
+			return ret;
+		}
+
+		i++;
+	}
+
+	return 0;
+}
+
+static int axg_card_parse_of_optional(struct snd_soc_card *card,
+				      const char *propname,
+				      int (*func)(struct snd_soc_card *c,
+						  const char *p))
+{
+	/* If property is not provided, don't fail ... */
+	if (!of_property_read_bool(card->dev->of_node, propname))
+		return 0;
+
+	/* ... but do fail if it is provided and the parsing fails */
+	return func(card, propname);
+}
+
+static const struct of_device_id axg_card_of_match[] = {
+	{ .compatible = "amlogic,axg-sound-card", },
+	{}
+};
+MODULE_DEVICE_TABLE(of, axg_card_of_match);
+
+static int axg_card_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct axg_card *priv;
+	int ret;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, priv);
+	snd_soc_card_set_drvdata(&priv->card, priv);
+
+	priv->card.owner = THIS_MODULE;
+	priv->card.dev = dev;
+
+	ret = snd_soc_of_parse_card_name(&priv->card, "model");
+	if (ret < 0)
+		return ret;
+
+	ret = axg_card_parse_of_optional(&priv->card, "audio-routing",
+					 snd_soc_of_parse_audio_routing);
+	if (ret) {
+		dev_err(dev, "error while parsing routing\n");
+		return ret;
+	}
+
+	ret = axg_card_parse_of_optional(&priv->card, "audio-widgets",
+					 snd_soc_of_parse_audio_simple_widgets);
+	if (ret) {
+		dev_err(dev, "error while parsing widgets\n");
+		return ret;
+	}
+
+	ret = axg_card_add_links(&priv->card);
+	if (ret)
+		goto out_err;
+
+	ret = axg_card_add_aux_devices(&priv->card);
+	if (ret)
+		goto out_err;
+
+	ret = devm_snd_soc_register_card(dev, &priv->card);
+	if (ret)
+		goto out_err;
+
+	return 0;
+
+out_err:
+	axg_card_clean_references(priv);
+	return ret;
+}
+
+static int axg_card_remove(struct platform_device *pdev)
+{
+	struct axg_card *priv = platform_get_drvdata(pdev);
+
+	axg_card_clean_references(priv);
+
+	return 0;
+}
+
+static struct platform_driver axg_card_pdrv = {
+	.probe = axg_card_probe,
+	.remove = axg_card_remove,
+	.driver = {
+		.name = "axg-sound-card",
+		.of_match_table = axg_card_of_match,
+	},
+};
+module_platform_driver(axg_card_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG ALSA machine driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
new file mode 100644
index 000000000000..30262550e37b
--- /dev/null
+++ b/sound/soc/meson/axg-fifo.c
@@ -0,0 +1,341 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/of_irq.h>
+#include <linux/of_platform.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/reset.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-fifo.h"
+
+/*
+ * This file implements the platform operations common to the playback and
+ * capture frontend DAI. The logic behind this two types of fifo is very
+ * similar but some difference exist.
+ * These differences the respective DAI drivers
+ */
+
+static struct snd_pcm_hardware axg_fifo_hw = {
+	.info = (SNDRV_PCM_INFO_INTERLEAVED |
+		 SNDRV_PCM_INFO_MMAP |
+		 SNDRV_PCM_INFO_MMAP_VALID |
+		 SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		 SNDRV_PCM_INFO_PAUSE),
+
+	.formats = AXG_FIFO_FORMATS,
+	.rate_min = 5512,
+	.rate_max = 192000,
+	.channels_min = 1,
+	.channels_max = AXG_FIFO_CH_MAX,
+	.period_bytes_min = AXG_FIFO_MIN_DEPTH,
+	.period_bytes_max = UINT_MAX,
+	.periods_min = 2,
+	.periods_max = UINT_MAX,
+
+	/* No real justification for this */
+	.buffer_bytes_max = 1 * 1024 * 1024,
+};
+
+static struct snd_soc_dai *axg_fifo_dai(struct snd_pcm_substream *ss)
+{
+	struct snd_soc_pcm_runtime *rtd = ss->private_data;
+
+	return rtd->cpu_dai;
+}
+
+static struct axg_fifo *axg_fifo_data(struct snd_pcm_substream *ss)
+{
+	struct snd_soc_dai *dai = axg_fifo_dai(ss);
+
+	return snd_soc_dai_get_drvdata(dai);
+}
+
+static struct device *axg_fifo_dev(struct snd_pcm_substream *ss)
+{
+	struct snd_soc_dai *dai = axg_fifo_dai(ss);
+
+	return dai->dev;
+}
+
+static void __dma_enable(struct axg_fifo *fifo,  bool enable)
+{
+	regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_DMA_EN,
+			   enable ? CTRL0_DMA_EN : 0);
+}
+
+static int axg_fifo_pcm_trigger(struct snd_pcm_substream *ss, int cmd)
+{
+	struct axg_fifo *fifo = axg_fifo_data(ss);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		__dma_enable(fifo, true);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_STOP:
+		__dma_enable(fifo, false);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static snd_pcm_uframes_t axg_fifo_pcm_pointer(struct snd_pcm_substream *ss)
+{
+	struct axg_fifo *fifo = axg_fifo_data(ss);
+	struct snd_pcm_runtime *runtime = ss->runtime;
+	unsigned int addr;
+
+	regmap_read(fifo->map, FIFO_STATUS2, &addr);
+
+	return bytes_to_frames(runtime, addr - (unsigned int)runtime->dma_addr);
+}
+
+static int axg_fifo_pcm_hw_params(struct snd_pcm_substream *ss,
+				  struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = ss->runtime;
+	struct axg_fifo *fifo = axg_fifo_data(ss);
+	dma_addr_t end_ptr;
+	unsigned int burst_num;
+	int ret;
+
+	ret = snd_pcm_lib_malloc_pages(ss, params_buffer_bytes(params));
+	if (ret < 0)
+		return ret;
+
+	/* Setup dma memory pointers */
+	end_ptr = runtime->dma_addr + runtime->dma_bytes - AXG_FIFO_BURST;
+	regmap_write(fifo->map, FIFO_START_ADDR, runtime->dma_addr);
+	regmap_write(fifo->map, FIFO_FINISH_ADDR, end_ptr);
+
+	/* Setup interrupt periodicity */
+	burst_num = params_period_bytes(params) / AXG_FIFO_BURST;
+	regmap_write(fifo->map, FIFO_INT_ADDR, burst_num);
+
+	/* Enable block count irq */
+	regmap_update_bits(fifo->map, FIFO_CTRL0,
+			   CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT),
+			   CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT));
+
+	return 0;
+}
+
+static int axg_fifo_pcm_hw_free(struct snd_pcm_substream *ss)
+{
+	struct axg_fifo *fifo = axg_fifo_data(ss);
+
+	/* Disable the block count irq */
+	regmap_update_bits(fifo->map, FIFO_CTRL0,
+			   CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), 0);
+
+	return snd_pcm_lib_free_pages(ss);
+}
+
+static void axg_fifo_ack_irq(struct axg_fifo *fifo, u8 mask)
+{
+	regmap_update_bits(fifo->map, FIFO_CTRL1,
+			   CTRL1_INT_CLR(FIFO_INT_MASK),
+			   CTRL1_INT_CLR(mask));
+
+	/* Clear must also be cleared */
+	regmap_update_bits(fifo->map, FIFO_CTRL1,
+			   CTRL1_INT_CLR(FIFO_INT_MASK),
+			   0);
+}
+
+static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id)
+{
+	struct snd_pcm_substream *ss = dev_id;
+	struct axg_fifo *fifo = axg_fifo_data(ss);
+	unsigned int status;
+
+	regmap_read(fifo->map, FIFO_STATUS1, &status);
+
+	status = STATUS1_INT_STS(status) & FIFO_INT_MASK;
+	if (status & FIFO_INT_COUNT_REPEAT)
+		snd_pcm_period_elapsed(ss);
+	else
+		dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n",
+			status);
+
+	/* Ack irqs */
+	axg_fifo_ack_irq(fifo, status);
+
+	return IRQ_RETVAL(status);
+}
+
+static int axg_fifo_pcm_open(struct snd_pcm_substream *ss)
+{
+	struct axg_fifo *fifo = axg_fifo_data(ss);
+	struct device *dev = axg_fifo_dev(ss);
+	int ret;
+
+	snd_soc_set_runtime_hwparams(ss, &axg_fifo_hw);
+
+	/*
+	 * Make sure the buffer and period size are multiple of the FIFO
+	 * minimum depth size
+	 */
+	ret = snd_pcm_hw_constraint_step(ss->runtime, 0,
+					 SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+					 AXG_FIFO_MIN_DEPTH);
+	if (ret)
+		return ret;
+
+	ret = snd_pcm_hw_constraint_step(ss->runtime, 0,
+					 SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+					 AXG_FIFO_MIN_DEPTH);
+	if (ret)
+		return ret;
+
+	ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0,
+			  dev_name(dev), ss);
+
+	/* Enable pclk to access registers and clock the fifo ip */
+	ret = clk_prepare_enable(fifo->pclk);
+	if (ret)
+		return ret;
+
+	/* Setup status2 so it reports the memory pointer */
+	regmap_update_bits(fifo->map, FIFO_CTRL1,
+			   CTRL1_STATUS2_SEL_MASK,
+			   CTRL1_STATUS2_SEL(STATUS2_SEL_DDR_READ));
+
+	/* Make sure the dma is initially disabled */
+	__dma_enable(fifo, false);
+
+	/* Disable irqs until params are ready */
+	regmap_update_bits(fifo->map, FIFO_CTRL0,
+			   CTRL0_INT_EN(FIFO_INT_MASK), 0);
+
+	/* Clear any pending interrupt */
+	axg_fifo_ack_irq(fifo, FIFO_INT_MASK);
+
+	/* Take memory arbitror out of reset */
+	ret = reset_control_deassert(fifo->arb);
+	if (ret)
+		clk_disable_unprepare(fifo->pclk);
+
+	return ret;
+}
+
+static int axg_fifo_pcm_close(struct snd_pcm_substream *ss)
+{
+	struct axg_fifo *fifo = axg_fifo_data(ss);
+	int ret;
+
+	/* Put the memory arbitror back in reset */
+	ret = reset_control_assert(fifo->arb);
+
+	/* Disable fifo ip and register access */
+	clk_disable_unprepare(fifo->pclk);
+
+	/* remove IRQ */
+	free_irq(fifo->irq, ss);
+
+	return ret;
+}
+
+const struct snd_pcm_ops axg_fifo_pcm_ops = {
+	.open =		axg_fifo_pcm_open,
+	.close =        axg_fifo_pcm_close,
+	.ioctl =	snd_pcm_lib_ioctl,
+	.hw_params =	axg_fifo_pcm_hw_params,
+	.hw_free =      axg_fifo_pcm_hw_free,
+	.pointer =	axg_fifo_pcm_pointer,
+	.trigger =	axg_fifo_pcm_trigger,
+};
+EXPORT_SYMBOL_GPL(axg_fifo_pcm_ops);
+
+int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type)
+{
+	struct snd_card *card = rtd->card->snd_card;
+	size_t size = axg_fifo_hw.buffer_bytes_max;
+
+	return snd_pcm_lib_preallocate_pages(rtd->pcm->streams[type].substream,
+					     SNDRV_DMA_TYPE_DEV, card->dev,
+					     size, size);
+}
+EXPORT_SYMBOL_GPL(axg_fifo_pcm_new);
+
+static const struct regmap_config axg_fifo_regmap_cfg = {
+	.reg_bits	= 32,
+	.val_bits	= 32,
+	.reg_stride	= 4,
+	.max_register	= FIFO_STATUS2,
+};
+
+int axg_fifo_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	const struct axg_fifo_match_data *data;
+	struct axg_fifo *fifo;
+	struct resource *res;
+	void __iomem *regs;
+
+	data = of_device_get_match_data(dev);
+	if (!data) {
+		dev_err(dev, "failed to match device\n");
+		return -ENODEV;
+	}
+
+	fifo = devm_kzalloc(dev, sizeof(*fifo), GFP_KERNEL);
+	if (!fifo)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, fifo);
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	regs = devm_ioremap_resource(dev, res);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	fifo->map = devm_regmap_init_mmio(dev, regs, &axg_fifo_regmap_cfg);
+	if (IS_ERR(fifo->map)) {
+		dev_err(dev, "failed to init regmap: %ld\n",
+			PTR_ERR(fifo->map));
+		return PTR_ERR(fifo->map);
+	}
+
+	fifo->pclk = devm_clk_get(dev, NULL);
+	if (IS_ERR(fifo->pclk)) {
+		if (PTR_ERR(fifo->pclk) != -EPROBE_DEFER)
+			dev_err(dev, "failed to get pclk: %ld\n",
+				PTR_ERR(fifo->pclk));
+		return PTR_ERR(fifo->pclk);
+	}
+
+	fifo->arb = devm_reset_control_get_exclusive(dev, NULL);
+	if (IS_ERR(fifo->arb)) {
+		if (PTR_ERR(fifo->arb) != -EPROBE_DEFER)
+			dev_err(dev, "failed to get arb reset: %ld\n",
+				PTR_ERR(fifo->arb));
+		return PTR_ERR(fifo->arb);
+	}
+
+	fifo->irq = of_irq_get(dev->of_node, 0);
+	if (fifo->irq <= 0) {
+		dev_err(dev, "failed to get irq: %d\n", fifo->irq);
+		return fifo->irq;
+	}
+
+	return devm_snd_soc_register_component(dev, data->component_drv,
+					       data->dai_drv, 1);
+}
+EXPORT_SYMBOL_GPL(axg_fifo_probe);
+
+MODULE_DESCRIPTION("Amlogic AXG fifo driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h
new file mode 100644
index 000000000000..cb6c4013ca33
--- /dev/null
+++ b/sound/soc/meson/axg-fifo.h
@@ -0,0 +1,80 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */
+/*
+ * Copyright (c) 2018 BayLibre, SAS.
+ * Author: Jerome Brunet <jbrunet@baylibre.com>
+ */
+
+#ifndef _MESON_AXG_FIFO_H
+#define _MESON_AXG_FIFO_H
+
+struct clk;
+struct platform_device;
+struct regmap;
+struct reset_control;
+
+struct snd_soc_component_driver;
+struct snd_soc_dai;
+struct snd_soc_dai_driver;
+struct snd_pcm_ops;
+struct snd_soc_pcm_runtime;
+
+#define AXG_FIFO_CH_MAX			128
+#define AXG_FIFO_RATES			(SNDRV_PCM_RATE_5512 |		\
+					 SNDRV_PCM_RATE_8000_192000)
+#define AXG_FIFO_FORMATS		(SNDRV_PCM_FMTBIT_S8 |		\
+					 SNDRV_PCM_FMTBIT_S16_LE |	\
+					 SNDRV_PCM_FMTBIT_S20_LE |	\
+					 SNDRV_PCM_FMTBIT_S24_LE |	\
+					 SNDRV_PCM_FMTBIT_S32_LE)
+
+#define AXG_FIFO_BURST			8
+#define AXG_FIFO_MIN_CNT		64
+#define AXG_FIFO_MIN_DEPTH		(AXG_FIFO_BURST * AXG_FIFO_MIN_CNT)
+
+#define FIFO_INT_ADDR_FINISH		BIT(0)
+#define FIFO_INT_ADDR_INT		BIT(1)
+#define FIFO_INT_COUNT_REPEAT		BIT(2)
+#define FIFO_INT_COUNT_ONCE		BIT(3)
+#define FIFO_INT_FIFO_ZERO		BIT(4)
+#define FIFO_INT_FIFO_DEPTH		BIT(5)
+#define FIFO_INT_MASK			GENMASK(7, 0)
+
+#define FIFO_CTRL0			0x00
+#define  CTRL0_DMA_EN			BIT(31)
+#define  CTRL0_INT_EN(x)		((x) << 16)
+#define  CTRL0_SEL_MASK			GENMASK(2, 0)
+#define  CTRL0_SEL_SHIFT		0
+#define FIFO_CTRL1			0x04
+#define  CTRL1_INT_CLR(x)		((x) << 0)
+#define  CTRL1_STATUS2_SEL_MASK		GENMASK(11, 8)
+#define  CTRL1_STATUS2_SEL(x)		((x) << 8)
+#define   STATUS2_SEL_DDR_READ		0
+#define  CTRL1_THRESHOLD_MASK		GENMASK(23, 16)
+#define  CTRL1_THRESHOLD(x)		((x) << 16)
+#define  CTRL1_FRDDR_DEPTH_MASK		GENMASK(31, 24)
+#define  CTRL1_FRDDR_DEPTH(x)		((x) << 24)
+#define FIFO_START_ADDR			0x08
+#define FIFO_FINISH_ADDR		0x0c
+#define FIFO_INT_ADDR			0x10
+#define FIFO_STATUS1			0x14
+#define  STATUS1_INT_STS(x)		((x) << 0)
+#define FIFO_STATUS2			0x18
+
+struct axg_fifo {
+	struct regmap *map;
+	struct clk *pclk;
+	struct reset_control *arb;
+	int irq;
+};
+
+struct axg_fifo_match_data {
+	const struct snd_soc_component_driver *component_drv;
+	struct snd_soc_dai_driver *dai_drv;
+};
+
+extern const struct snd_pcm_ops axg_fifo_pcm_ops;
+
+int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type);
+int axg_fifo_probe(struct platform_device *pdev);
+
+#endif /* _MESON_AXG_FIFO_H */
diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c
new file mode 100644
index 000000000000..a6f6f6a2eca8
--- /dev/null
+++ b/sound/soc/meson/axg-frddr.c
@@ -0,0 +1,141 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+/* This driver implements the frontend playback DAI of AXG based SoCs */
+
+#include <linux/clk.h>
+#include <linux/regmap.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-fifo.h"
+
+#define CTRL0_FRDDR_PP_MODE	BIT(30)
+
+static int axg_frddr_dai_startup(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+	unsigned int fifo_depth, fifo_threshold;
+	int ret;
+
+	/* Enable pclk to access registers and clock the fifo ip */
+	ret = clk_prepare_enable(fifo->pclk);
+	if (ret)
+		return ret;
+
+	/* Apply single buffer mode to the interface */
+	regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_FRDDR_PP_MODE, 0);
+
+	/*
+	 * TODO: We could adapt the fifo depth and the fifo threshold
+	 * depending on the expected memory throughput and lantencies
+	 * For now, we'll just use the same values as the vendor kernel
+	 * Depth and threshold are zero based.
+	 */
+	fifo_depth = AXG_FIFO_MIN_CNT - 1;
+	fifo_threshold = (AXG_FIFO_MIN_CNT / 2) - 1;
+	regmap_update_bits(fifo->map, FIFO_CTRL1,
+			   CTRL1_FRDDR_DEPTH_MASK | CTRL1_THRESHOLD_MASK,
+			   CTRL1_FRDDR_DEPTH(fifo_depth) |
+			   CTRL1_THRESHOLD(fifo_threshold));
+
+	return 0;
+}
+
+static void axg_frddr_dai_shutdown(struct snd_pcm_substream *substream,
+				   struct snd_soc_dai *dai)
+{
+	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+
+	clk_disable_unprepare(fifo->pclk);
+}
+
+static int axg_frddr_pcm_new(struct snd_soc_pcm_runtime *rtd,
+			     struct snd_soc_dai *dai)
+{
+	return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_PLAYBACK);
+}
+
+static const struct snd_soc_dai_ops axg_frddr_ops = {
+	.startup	= axg_frddr_dai_startup,
+	.shutdown	= axg_frddr_dai_shutdown,
+};
+
+static struct snd_soc_dai_driver axg_frddr_dai_drv = {
+	.name = "FRDDR",
+	.playback = {
+		.stream_name	= "Playback",
+		.channels_min	= 1,
+		.channels_max	= AXG_FIFO_CH_MAX,
+		.rates		= AXG_FIFO_RATES,
+		.formats	= AXG_FIFO_FORMATS,
+	},
+	.ops		= &axg_frddr_ops,
+	.pcm_new	= axg_frddr_pcm_new,
+};
+
+static const char * const axg_frddr_sel_texts[] = {
+	"OUT 0", "OUT 1", "OUT 2", "OUT 3"
+};
+
+static SOC_ENUM_SINGLE_DECL(axg_frddr_sel_enum, FIFO_CTRL0, CTRL0_SEL_SHIFT,
+			    axg_frddr_sel_texts);
+
+static const struct snd_kcontrol_new axg_frddr_out_demux =
+	SOC_DAPM_ENUM("Output Sink", axg_frddr_sel_enum);
+
+static const struct snd_soc_dapm_widget axg_frddr_dapm_widgets[] = {
+	SND_SOC_DAPM_DEMUX("SINK SEL", SND_SOC_NOPM, 0, 0,
+			   &axg_frddr_out_demux),
+	SND_SOC_DAPM_AIF_OUT("OUT 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 3", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route axg_frddr_dapm_routes[] = {
+	{ "SINK SEL", NULL, "Playback" },
+	{ "OUT 0", "OUT 0",  "SINK SEL" },
+	{ "OUT 1", "OUT 1",  "SINK SEL" },
+	{ "OUT 2", "OUT 2",  "SINK SEL" },
+	{ "OUT 3", "OUT 3",  "SINK SEL" },
+};
+
+static const struct snd_soc_component_driver axg_frddr_component_drv = {
+	.dapm_widgets		= axg_frddr_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(axg_frddr_dapm_widgets),
+	.dapm_routes		= axg_frddr_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(axg_frddr_dapm_routes),
+	.ops			= &axg_fifo_pcm_ops
+};
+
+static const struct axg_fifo_match_data axg_frddr_match_data = {
+	.component_drv	= &axg_frddr_component_drv,
+	.dai_drv	= &axg_frddr_dai_drv
+};
+
+static const struct of_device_id axg_frddr_of_match[] = {
+	{
+		.compatible = "amlogic,axg-frddr",
+		.data = &axg_frddr_match_data,
+	}, {}
+};
+MODULE_DEVICE_TABLE(of, axg_frddr_of_match);
+
+static struct platform_driver axg_frddr_pdrv = {
+	.probe = axg_fifo_probe,
+	.driver = {
+		.name = "axg-frddr",
+		.of_match_table = axg_frddr_of_match,
+	},
+};
+module_platform_driver(axg_frddr_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG playback fifo driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c
new file mode 100644
index 000000000000..9dea528053ad
--- /dev/null
+++ b/sound/soc/meson/axg-spdifout.c
@@ -0,0 +1,456 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/pcm_params.h>
+#include <sound/pcm_iec958.h>
+
+/*
+ * NOTE:
+ * The meaning of bits SPDIFOUT_CTRL0_XXX_SEL is actually the opposite
+ * of what the documentation says. Manual control on V, U and C bits is
+ * applied when the related sel bits are cleared
+ */
+
+#define SPDIFOUT_STAT			0x00
+#define SPDIFOUT_GAIN0			0x04
+#define SPDIFOUT_GAIN1			0x08
+#define SPDIFOUT_CTRL0			0x0c
+#define  SPDIFOUT_CTRL0_EN		BIT(31)
+#define  SPDIFOUT_CTRL0_RST_OUT		BIT(29)
+#define  SPDIFOUT_CTRL0_RST_IN		BIT(28)
+#define  SPDIFOUT_CTRL0_USEL		BIT(26)
+#define  SPDIFOUT_CTRL0_USET		BIT(25)
+#define  SPDIFOUT_CTRL0_CHSTS_SEL	BIT(24)
+#define  SPDIFOUT_CTRL0_DATA_SEL	BIT(20)
+#define  SPDIFOUT_CTRL0_MSB_FIRST	BIT(19)
+#define  SPDIFOUT_CTRL0_VSEL		BIT(18)
+#define  SPDIFOUT_CTRL0_VSET		BIT(17)
+#define  SPDIFOUT_CTRL0_MASK_MASK	GENMASK(11, 4)
+#define  SPDIFOUT_CTRL0_MASK(x)		((x) << 4)
+#define SPDIFOUT_CTRL1			0x10
+#define  SPDIFOUT_CTRL1_MSB_POS_MASK	GENMASK(12, 8)
+#define  SPDIFOUT_CTRL1_MSB_POS(x)	((x) << 8)
+#define  SPDIFOUT_CTRL1_TYPE_MASK	GENMASK(6, 4)
+#define  SPDIFOUT_CTRL1_TYPE(x)		((x) << 4)
+#define SPDIFOUT_PREAMB			0x14
+#define SPDIFOUT_SWAP			0x18
+#define SPDIFOUT_CHSTS0			0x1c
+#define SPDIFOUT_CHSTS1			0x20
+#define SPDIFOUT_CHSTS2			0x24
+#define SPDIFOUT_CHSTS3			0x28
+#define SPDIFOUT_CHSTS4			0x2c
+#define SPDIFOUT_CHSTS5			0x30
+#define SPDIFOUT_CHSTS6			0x34
+#define SPDIFOUT_CHSTS7			0x38
+#define SPDIFOUT_CHSTS8			0x3c
+#define SPDIFOUT_CHSTS9			0x40
+#define SPDIFOUT_CHSTSA			0x44
+#define SPDIFOUT_CHSTSB			0x48
+#define SPDIFOUT_MUTE_VAL		0x4c
+
+struct axg_spdifout {
+	struct regmap *map;
+	struct clk *mclk;
+	struct clk *pclk;
+};
+
+static void axg_spdifout_enable(struct regmap *map)
+{
+	/* Apply both reset */
+	regmap_update_bits(map, SPDIFOUT_CTRL0,
+			   SPDIFOUT_CTRL0_RST_OUT | SPDIFOUT_CTRL0_RST_IN,
+			   0);
+
+	/* Clear out reset before in reset */
+	regmap_update_bits(map, SPDIFOUT_CTRL0,
+			   SPDIFOUT_CTRL0_RST_OUT, SPDIFOUT_CTRL0_RST_OUT);
+	regmap_update_bits(map, SPDIFOUT_CTRL0,
+			   SPDIFOUT_CTRL0_RST_IN,  SPDIFOUT_CTRL0_RST_IN);
+
+	/* Enable spdifout */
+	regmap_update_bits(map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_EN,
+			   SPDIFOUT_CTRL0_EN);
+}
+
+static void axg_spdifout_disable(struct regmap *map)
+{
+	regmap_update_bits(map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_EN, 0);
+}
+
+static int axg_spdifout_trigger(struct snd_pcm_substream *substream, int cmd,
+				struct snd_soc_dai *dai)
+{
+	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		axg_spdifout_enable(priv->map);
+		return 0;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		axg_spdifout_disable(priv->map);
+		return 0;
+
+	default:
+		return -EINVAL;
+	}
+}
+
+static int axg_spdifout_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+
+	/* Use spdif valid bit to perform digital mute */
+	regmap_update_bits(priv->map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_VSET,
+			   mute ? SPDIFOUT_CTRL0_VSET : 0);
+
+	return 0;
+}
+
+static int axg_spdifout_sample_fmt(struct snd_pcm_hw_params *params,
+				   struct snd_soc_dai *dai)
+{
+	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+	unsigned int val;
+
+	/* Set the samples spdifout will pull from the FIFO */
+	switch (params_channels(params)) {
+	case 1:
+		val = SPDIFOUT_CTRL0_MASK(0x1);
+		break;
+	case 2:
+		val = SPDIFOUT_CTRL0_MASK(0x3);
+		break;
+	default:
+		dev_err(dai->dev, "too many channels for spdif dai: %u\n",
+			params_channels(params));
+		return -EINVAL;
+	}
+
+	regmap_update_bits(priv->map, SPDIFOUT_CTRL0,
+			   SPDIFOUT_CTRL0_MASK_MASK, val);
+
+	/* FIFO data are arranged in chunks of 64bits */
+	switch (params_physical_width(params)) {
+	case 8:
+		/* 8 samples of 8 bits */
+		val = SPDIFOUT_CTRL1_TYPE(0);
+		break;
+	case 16:
+		/* 4 samples of 16 bits - right justified */
+		val = SPDIFOUT_CTRL1_TYPE(2);
+		break;
+	case 32:
+		/* 2 samples of 32 bits - right justified */
+		val = SPDIFOUT_CTRL1_TYPE(4);
+		break;
+	default:
+		dev_err(dai->dev, "Unsupported physical width: %u\n",
+			params_physical_width(params));
+		return -EINVAL;
+	}
+
+	/* Position of the MSB in FIFO samples */
+	val |= SPDIFOUT_CTRL1_MSB_POS(params_width(params) - 1);
+
+	regmap_update_bits(priv->map, SPDIFOUT_CTRL1,
+			   SPDIFOUT_CTRL1_MSB_POS_MASK |
+			   SPDIFOUT_CTRL1_TYPE_MASK, val);
+
+	regmap_update_bits(priv->map, SPDIFOUT_CTRL0,
+			   SPDIFOUT_CTRL0_MSB_FIRST | SPDIFOUT_CTRL0_DATA_SEL,
+			   0);
+
+	return 0;
+}
+
+static int axg_spdifout_set_chsts(struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
+{
+	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+	unsigned int offset;
+	int ret;
+	u8 cs[4];
+	u32 val;
+
+	ret = snd_pcm_create_iec958_consumer_hw_params(params, cs, 4);
+	if (ret < 0) {
+		dev_err(dai->dev, "Creating IEC958 channel status failed %d\n",
+			ret);
+		return ret;
+	}
+	val = cs[0] | cs[1] << 8 | cs[2] << 16 | cs[3] << 24;
+
+	/* Setup channel status A bits [31 - 0]*/
+	regmap_write(priv->map, SPDIFOUT_CHSTS0, val);
+
+	/* Clear channel status A bits [191 - 32] */
+	for (offset = SPDIFOUT_CHSTS1; offset <= SPDIFOUT_CHSTS5;
+	     offset += regmap_get_reg_stride(priv->map))
+		regmap_write(priv->map, offset, 0);
+
+	/* Setup channel status B bits [31 - 0]*/
+	regmap_write(priv->map, SPDIFOUT_CHSTS6, val);
+
+	/* Clear channel status B bits [191 - 32] */
+	for (offset = SPDIFOUT_CHSTS7; offset <= SPDIFOUT_CHSTSB;
+	     offset += regmap_get_reg_stride(priv->map))
+		regmap_write(priv->map, offset, 0);
+
+	return 0;
+}
+
+static int axg_spdifout_hw_params(struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
+{
+	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+	unsigned int rate = params_rate(params);
+	int ret;
+
+	/* 2 * 32bits per subframe * 2 channels = 128 */
+	ret = clk_set_rate(priv->mclk, rate * 128);
+	if (ret) {
+		dev_err(dai->dev, "failed to set spdif clock\n");
+		return ret;
+	}
+
+	ret = axg_spdifout_sample_fmt(params, dai);
+	if (ret) {
+		dev_err(dai->dev, "failed to setup sample format\n");
+		return ret;
+	}
+
+	ret = axg_spdifout_set_chsts(params, dai);
+	if (ret) {
+		dev_err(dai->dev, "failed to setup channel status words\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int axg_spdifout_startup(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	/* Clock the spdif output block */
+	ret = clk_prepare_enable(priv->pclk);
+	if (ret) {
+		dev_err(dai->dev, "failed to enable pclk\n");
+		return ret;
+	}
+
+	/* Make sure the block is initially stopped */
+	axg_spdifout_disable(priv->map);
+
+	/* Insert data from bit 27 lsb first */
+	regmap_update_bits(priv->map, SPDIFOUT_CTRL0,
+			   SPDIFOUT_CTRL0_MSB_FIRST | SPDIFOUT_CTRL0_DATA_SEL,
+			   0);
+
+	/* Manual control of V, C and U, U = 0 */
+	regmap_update_bits(priv->map, SPDIFOUT_CTRL0,
+			   SPDIFOUT_CTRL0_CHSTS_SEL | SPDIFOUT_CTRL0_VSEL |
+			   SPDIFOUT_CTRL0_USEL | SPDIFOUT_CTRL0_USET,
+			   0);
+
+	/* Static SWAP configuration ATM */
+	regmap_write(priv->map, SPDIFOUT_SWAP, 0x10);
+
+	return 0;
+}
+
+static void axg_spdifout_shutdown(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+
+	clk_disable_unprepare(priv->pclk);
+}
+
+static const struct snd_soc_dai_ops axg_spdifout_ops = {
+	.trigger	= axg_spdifout_trigger,
+	.digital_mute	= axg_spdifout_digital_mute,
+	.hw_params	= axg_spdifout_hw_params,
+	.startup	= axg_spdifout_startup,
+	.shutdown	= axg_spdifout_shutdown,
+};
+
+static struct snd_soc_dai_driver axg_spdifout_dai_drv[] = {
+	{
+		.name = "SPDIF Output",
+		.playback = {
+			.stream_name	= "Playback",
+			.channels_min	= 1,
+			.channels_max	= 2,
+			.rates		= (SNDRV_PCM_RATE_32000  |
+					   SNDRV_PCM_RATE_44100  |
+					   SNDRV_PCM_RATE_48000  |
+					   SNDRV_PCM_RATE_88200  |
+					   SNDRV_PCM_RATE_96000  |
+					   SNDRV_PCM_RATE_176400 |
+					   SNDRV_PCM_RATE_192000),
+			.formats	= (SNDRV_PCM_FMTBIT_S8     |
+					   SNDRV_PCM_FMTBIT_S16_LE |
+					   SNDRV_PCM_FMTBIT_S20_LE |
+					   SNDRV_PCM_FMTBIT_S24_LE),
+		},
+		.ops = &axg_spdifout_ops,
+	},
+};
+
+static const char * const spdifout_sel_texts[] = {
+	"IN 0", "IN 1", "IN 2",
+};
+
+static SOC_ENUM_SINGLE_DECL(axg_spdifout_sel_enum, SPDIFOUT_CTRL1, 24,
+			    spdifout_sel_texts);
+
+static const struct snd_kcontrol_new axg_spdifout_in_mux =
+	SOC_DAPM_ENUM("Input Source", axg_spdifout_sel_enum);
+
+static const struct snd_soc_dapm_widget axg_spdifout_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_spdifout_in_mux),
+};
+
+static const struct snd_soc_dapm_route axg_spdifout_dapm_routes[] = {
+	{ "SRC SEL", "IN 0", "IN 0" },
+	{ "SRC SEL", "IN 1", "IN 1" },
+	{ "SRC SEL", "IN 2", "IN 2" },
+	{ "Playback", NULL, "SRC SEL" },
+};
+
+static const struct snd_kcontrol_new axg_spdifout_controls[] = {
+	SOC_DOUBLE("Playback Volume", SPDIFOUT_GAIN0,  0,  8, 255, 0),
+	SOC_DOUBLE("Playback Switch", SPDIFOUT_CTRL0, 22, 21, 1, 1),
+	SOC_SINGLE("Playback Gain Enable Switch",
+		   SPDIFOUT_CTRL1, 26, 1, 0),
+	SOC_SINGLE("Playback Channels Mix Switch",
+		   SPDIFOUT_CTRL0, 23, 1, 0),
+};
+
+static int axg_spdifout_set_bias_level(struct snd_soc_component *component,
+				       enum snd_soc_bias_level level)
+{
+	struct axg_spdifout *priv = snd_soc_component_get_drvdata(component);
+	enum snd_soc_bias_level now =
+		snd_soc_component_get_bias_level(component);
+	int ret = 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		if (now == SND_SOC_BIAS_STANDBY)
+			ret = clk_prepare_enable(priv->mclk);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (now == SND_SOC_BIAS_PREPARE)
+			clk_disable_unprepare(priv->mclk);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+	case SND_SOC_BIAS_ON:
+		break;
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_component_driver axg_spdifout_component_drv = {
+	.controls		= axg_spdifout_controls,
+	.num_controls		= ARRAY_SIZE(axg_spdifout_controls),
+	.dapm_widgets		= axg_spdifout_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(axg_spdifout_dapm_widgets),
+	.dapm_routes		= axg_spdifout_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(axg_spdifout_dapm_routes),
+	.set_bias_level		= axg_spdifout_set_bias_level,
+};
+
+static const struct regmap_config axg_spdifout_regmap_cfg = {
+	.reg_bits	= 32,
+	.val_bits	= 32,
+	.reg_stride	= 4,
+	.max_register	= SPDIFOUT_MUTE_VAL,
+};
+
+static const struct of_device_id axg_spdifout_of_match[] = {
+	{ .compatible = "amlogic,axg-spdifout", },
+	{}
+};
+MODULE_DEVICE_TABLE(of, axg_spdifout_of_match);
+
+static int axg_spdifout_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct axg_spdifout *priv;
+	struct resource *res;
+	void __iomem *regs;
+	int ret;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, priv);
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	regs = devm_ioremap_resource(dev, res);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	priv->map = devm_regmap_init_mmio(dev, regs, &axg_spdifout_regmap_cfg);
+	if (IS_ERR(priv->map)) {
+		dev_err(dev, "failed to init regmap: %ld\n",
+			PTR_ERR(priv->map));
+		return PTR_ERR(priv->map);
+	}
+
+	priv->pclk = devm_clk_get(dev, "pclk");
+	if (IS_ERR(priv->pclk)) {
+		ret = PTR_ERR(priv->pclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get pclk: %d\n", ret);
+		return ret;
+	}
+
+	priv->mclk = devm_clk_get(dev, "mclk");
+	if (IS_ERR(priv->mclk)) {
+		ret = PTR_ERR(priv->mclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get mclk: %d\n", ret);
+		return ret;
+	}
+
+	return devm_snd_soc_register_component(dev, &axg_spdifout_component_drv,
+			axg_spdifout_dai_drv, ARRAY_SIZE(axg_spdifout_dai_drv));
+}
+
+static struct platform_driver axg_spdifout_pdrv = {
+	.probe = axg_spdifout_probe,
+	.driver = {
+		.name = "axg-spdifout",
+		.of_match_table = axg_spdifout_of_match,
+	},
+};
+module_platform_driver(axg_spdifout_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG SPDIF Output driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c
new file mode 100644
index 000000000000..43e390f9358a
--- /dev/null
+++ b/sound/soc/meson/axg-tdm-formatter.c
@@ -0,0 +1,381 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+
+#include "axg-tdm-formatter.h"
+
+struct axg_tdm_formatter {
+	struct list_head list;
+	struct axg_tdm_stream *stream;
+	const struct axg_tdm_formatter_driver *drv;
+	struct clk *pclk;
+	struct clk *sclk;
+	struct clk *lrclk;
+	struct clk *sclk_sel;
+	struct clk *lrclk_sel;
+	bool enabled;
+	struct regmap *map;
+};
+
+int axg_tdm_formatter_set_channel_masks(struct regmap *map,
+					struct axg_tdm_stream *ts,
+					unsigned int offset)
+{
+	unsigned int val, ch = ts->channels;
+	unsigned long mask;
+	int i, j;
+
+	/*
+	 * Distribute the channels of the stream over the available slots
+	 * of each TDM lane
+	 */
+	for (i = 0; i < AXG_TDM_NUM_LANES; i++) {
+		val = 0;
+		mask = ts->mask[i];
+
+		for (j = find_first_bit(&mask, 32);
+		     (j < 32) && ch;
+		     j = find_next_bit(&mask, 32, j + 1)) {
+			val |= 1 << j;
+			ch -= 1;
+		}
+
+		regmap_write(map, offset, val);
+		offset += regmap_get_reg_stride(map);
+	}
+
+	/*
+	 * If we still have channel left at the end of the process, it means
+	 * the stream has more channels than we can accommodate and we should
+	 * have caught this earlier.
+	 */
+	if (WARN_ON(ch != 0)) {
+		pr_err("channel mask error\n");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks);
+
+static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
+{
+	struct axg_tdm_stream *ts = formatter->stream;
+	bool invert = formatter->drv->invert_sclk;
+	int ret;
+
+	/* Do nothing if the formatter is already enabled */
+	if (formatter->enabled)
+		return 0;
+
+	/*
+	 * If sclk is inverted, invert it back and provide the inversion
+	 * required by the formatter
+	 */
+	invert ^= axg_tdm_sclk_invert(ts->iface->fmt);
+	ret = clk_set_phase(formatter->sclk, invert ? 180 : 0);
+	if (ret)
+		return ret;
+
+	/* Setup the stream parameter in the formatter */
+	ret = formatter->drv->ops->prepare(formatter->map, formatter->stream);
+	if (ret)
+		return ret;
+
+	/* Enable the signal clocks feeding the formatter */
+	ret = clk_prepare_enable(formatter->sclk);
+	if (ret)
+		return ret;
+
+	ret = clk_prepare_enable(formatter->lrclk);
+	if (ret) {
+		clk_disable_unprepare(formatter->sclk);
+		return ret;
+	}
+
+	/* Finally, actually enable the formatter */
+	formatter->drv->ops->enable(formatter->map);
+	formatter->enabled = true;
+
+	return 0;
+}
+
+static void axg_tdm_formatter_disable(struct axg_tdm_formatter *formatter)
+{
+	/* Do nothing if the formatter is already disabled */
+	if (!formatter->enabled)
+		return;
+
+	formatter->drv->ops->disable(formatter->map);
+	clk_disable_unprepare(formatter->lrclk);
+	clk_disable_unprepare(formatter->sclk);
+	formatter->enabled = false;
+}
+
+static int axg_tdm_formatter_attach(struct axg_tdm_formatter *formatter)
+{
+	struct axg_tdm_stream *ts = formatter->stream;
+	int ret = 0;
+
+	mutex_lock(&ts->lock);
+
+	/* Catch up if the stream is already running when we attach */
+	if (ts->ready) {
+		ret = axg_tdm_formatter_enable(formatter);
+		if (ret) {
+			pr_err("failed to enable formatter\n");
+			goto out;
+		}
+	}
+
+	list_add_tail(&formatter->list, &ts->formatter_list);
+out:
+	mutex_unlock(&ts->lock);
+	return ret;
+}
+
+static void axg_tdm_formatter_dettach(struct axg_tdm_formatter *formatter)
+{
+	struct axg_tdm_stream *ts = formatter->stream;
+
+	mutex_lock(&ts->lock);
+	list_del(&formatter->list);
+	mutex_unlock(&ts->lock);
+
+	axg_tdm_formatter_disable(formatter);
+}
+
+static int axg_tdm_formatter_power_up(struct axg_tdm_formatter *formatter,
+				      struct snd_soc_dapm_widget *w)
+{
+	struct axg_tdm_stream *ts = formatter->drv->ops->get_stream(w);
+	int ret;
+
+	/*
+	 * If we don't get a stream at this stage, it would mean that the
+	 * widget is powering up but is not attached to any backend DAI.
+	 * It should not happen, ever !
+	 */
+	if (WARN_ON(!ts))
+		return -ENODEV;
+
+	/* Clock our device */
+	ret = clk_prepare_enable(formatter->pclk);
+	if (ret)
+		return ret;
+
+	/* Reparent the bit clock to the TDM interface */
+	ret = clk_set_parent(formatter->sclk_sel, ts->iface->sclk);
+	if (ret)
+		goto disable_pclk;
+
+	/* Reparent the sample clock to the TDM interface */
+	ret = clk_set_parent(formatter->lrclk_sel, ts->iface->lrclk);
+	if (ret)
+		goto disable_pclk;
+
+	formatter->stream = ts;
+	ret = axg_tdm_formatter_attach(formatter);
+	if (ret)
+		goto disable_pclk;
+
+	return 0;
+
+disable_pclk:
+	clk_disable_unprepare(formatter->pclk);
+	return ret;
+}
+
+static void axg_tdm_formatter_power_down(struct axg_tdm_formatter *formatter)
+{
+	axg_tdm_formatter_dettach(formatter);
+	clk_disable_unprepare(formatter->pclk);
+	formatter->stream = NULL;
+}
+
+int axg_tdm_formatter_event(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *control,
+			    int event)
+{
+	struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm);
+	struct axg_tdm_formatter *formatter = snd_soc_component_get_drvdata(c);
+	int ret = 0;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		ret = axg_tdm_formatter_power_up(formatter, w);
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		axg_tdm_formatter_power_down(formatter);
+		break;
+
+	default:
+		dev_err(c->dev, "Unexpected event %d\n", event);
+		return -EINVAL;
+	}
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_formatter_event);
+
+int axg_tdm_formatter_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	const struct axg_tdm_formatter_driver *drv;
+	struct axg_tdm_formatter *formatter;
+	struct resource *res;
+	void __iomem *regs;
+	int ret;
+
+	drv = of_device_get_match_data(dev);
+	if (!drv) {
+		dev_err(dev, "failed to match device\n");
+		return -ENODEV;
+	}
+
+	formatter = devm_kzalloc(dev, sizeof(*formatter), GFP_KERNEL);
+	if (!formatter)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, formatter);
+	formatter->drv = drv;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	regs = devm_ioremap_resource(dev, res);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	formatter->map = devm_regmap_init_mmio(dev, regs, drv->regmap_cfg);
+	if (IS_ERR(formatter->map)) {
+		dev_err(dev, "failed to init regmap: %ld\n",
+			PTR_ERR(formatter->map));
+		return PTR_ERR(formatter->map);
+	}
+
+	/* Peripharal clock */
+	formatter->pclk = devm_clk_get(dev, "pclk");
+	if (IS_ERR(formatter->pclk)) {
+		ret = PTR_ERR(formatter->pclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get pclk: %d\n", ret);
+		return ret;
+	}
+
+	/* Formatter bit clock */
+	formatter->sclk = devm_clk_get(dev, "sclk");
+	if (IS_ERR(formatter->sclk)) {
+		ret = PTR_ERR(formatter->sclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get sclk: %d\n", ret);
+		return ret;
+	}
+
+	/* Formatter sample clock */
+	formatter->lrclk = devm_clk_get(dev, "lrclk");
+	if (IS_ERR(formatter->lrclk)) {
+		ret = PTR_ERR(formatter->lrclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get lrclk: %d\n", ret);
+		return ret;
+	}
+
+	/* Formatter bit clock input multiplexer */
+	formatter->sclk_sel = devm_clk_get(dev, "sclk_sel");
+	if (IS_ERR(formatter->sclk_sel)) {
+		ret = PTR_ERR(formatter->sclk_sel);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get sclk_sel: %d\n", ret);
+		return ret;
+	}
+
+	/* Formatter sample clock input multiplexer */
+	formatter->lrclk_sel = devm_clk_get(dev, "lrclk_sel");
+	if (IS_ERR(formatter->lrclk_sel)) {
+		ret = PTR_ERR(formatter->lrclk_sel);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get lrclk_sel: %d\n", ret);
+		return ret;
+	}
+
+	return devm_snd_soc_register_component(dev, drv->component_drv,
+					       NULL, 0);
+}
+EXPORT_SYMBOL_GPL(axg_tdm_formatter_probe);
+
+int axg_tdm_stream_start(struct axg_tdm_stream *ts)
+{
+	struct axg_tdm_formatter *formatter;
+	int ret = 0;
+
+	mutex_lock(&ts->lock);
+	ts->ready = true;
+
+	/* Start all the formatters attached to the stream */
+	list_for_each_entry(formatter, &ts->formatter_list, list) {
+		ret = axg_tdm_formatter_enable(formatter);
+		if (ret) {
+			pr_err("failed to start tdm stream\n");
+			goto out;
+		}
+	}
+
+out:
+	mutex_unlock(&ts->lock);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_stream_start);
+
+void axg_tdm_stream_stop(struct axg_tdm_stream *ts)
+{
+	struct axg_tdm_formatter *formatter;
+
+	mutex_lock(&ts->lock);
+	ts->ready = false;
+
+	/* Stop all the formatters attached to the stream */
+	list_for_each_entry(formatter, &ts->formatter_list, list) {
+		axg_tdm_formatter_disable(formatter);
+	}
+
+	mutex_unlock(&ts->lock);
+}
+EXPORT_SYMBOL_GPL(axg_tdm_stream_stop);
+
+struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface)
+{
+	struct axg_tdm_stream *ts;
+
+	ts = kzalloc(sizeof(*ts), GFP_KERNEL);
+	if (ts) {
+		INIT_LIST_HEAD(&ts->formatter_list);
+		mutex_init(&ts->lock);
+		ts->iface = iface;
+	}
+
+	return ts;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_stream_alloc);
+
+void axg_tdm_stream_free(struct axg_tdm_stream *ts)
+{
+	/*
+	 * If the list is not empty, it would mean that one of the formatter
+	 * widget is still powered and attached to the interface while we
+	 * we are removing the TDM DAI. It should not be possible
+	 */
+	WARN_ON(!list_empty(&ts->formatter_list));
+	mutex_destroy(&ts->lock);
+	kfree(ts);
+}
+EXPORT_SYMBOL_GPL(axg_tdm_stream_free);
+
+MODULE_DESCRIPTION("Amlogic AXG TDM formatter driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdm-formatter.h b/sound/soc/meson/axg-tdm-formatter.h
new file mode 100644
index 000000000000..cf947caf3cb1
--- /dev/null
+++ b/sound/soc/meson/axg-tdm-formatter.h
@@ -0,0 +1,39 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR MIT)
+ *
+ * Copyright (c) 2018 Baylibre SAS.
+ * Author: Jerome Brunet <jbrunet@baylibre.com>
+ */
+
+#ifndef _MESON_AXG_TDM_FORMATTER_H
+#define _MESON_AXG_TDM_FORMATTER_H
+
+#include "axg-tdm.h"
+
+struct platform_device;
+struct regmap;
+struct snd_soc_dapm_widget;
+struct snd_kcontrol;
+
+struct axg_tdm_formatter_ops {
+	struct axg_tdm_stream *(*get_stream)(struct snd_soc_dapm_widget *w);
+	void (*enable)(struct regmap *map);
+	void (*disable)(struct regmap *map);
+	int (*prepare)(struct regmap *map, struct axg_tdm_stream *ts);
+};
+
+struct axg_tdm_formatter_driver {
+	const struct snd_soc_component_driver *component_drv;
+	const struct regmap_config *regmap_cfg;
+	const struct axg_tdm_formatter_ops *ops;
+	bool invert_sclk;
+};
+
+int axg_tdm_formatter_set_channel_masks(struct regmap *map,
+					struct axg_tdm_stream *ts,
+					unsigned int offset);
+int axg_tdm_formatter_event(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *control,
+			    int event);
+int axg_tdm_formatter_probe(struct platform_device *pdev);
+
+#endif /* _MESON_AXG_TDM_FORMATTER_H */
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
new file mode 100644
index 000000000000..7b8baf46d968
--- /dev/null
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -0,0 +1,542 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-tdm.h"
+
+enum {
+	TDM_IFACE_PAD,
+	TDM_IFACE_LOOPBACK,
+};
+
+static unsigned int axg_tdm_slots_total(u32 *mask)
+{
+	unsigned int slots = 0;
+	int i;
+
+	if (!mask)
+		return 0;
+
+	/* Count the total number of slots provided by all 4 lanes */
+	for (i = 0; i < AXG_TDM_NUM_LANES; i++)
+		slots += hweight32(mask[i]);
+
+	return slots;
+}
+
+int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
+			  u32 *rx_mask, unsigned int slots,
+			  unsigned int slot_width)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+	struct axg_tdm_stream *tx = (struct axg_tdm_stream *)
+		dai->playback_dma_data;
+	struct axg_tdm_stream *rx = (struct axg_tdm_stream *)
+		dai->capture_dma_data;
+	unsigned int tx_slots, rx_slots;
+
+	tx_slots = axg_tdm_slots_total(tx_mask);
+	rx_slots = axg_tdm_slots_total(rx_mask);
+
+	/* We should at least have a slot for a valid interface */
+	if (!tx_slots && !rx_slots) {
+		dev_err(dai->dev, "interface has no slot\n");
+		return -EINVAL;
+	}
+
+	/*
+	 * Amend the dai driver channel number and let dpcm channel merge do
+	 * its job
+	 */
+	if (tx) {
+		tx->mask = tx_mask;
+		dai->driver->playback.channels_max = tx_slots;
+	}
+
+	if (rx) {
+		rx->mask = rx_mask;
+		dai->driver->capture.channels_max = rx_slots;
+	}
+
+	iface->slots = slots;
+
+	switch (slot_width) {
+	case 0:
+		/* defaults width to 32 if not provided */
+		iface->slot_width = 32;
+		break;
+	case 8:
+	case 16:
+	case 24:
+	case 32:
+		iface->slot_width = slot_width;
+		break;
+	default:
+		dev_err(dai->dev, "unsupported slot width: %d\n", slot_width);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_set_tdm_slots);
+
+static int axg_tdm_iface_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+				    unsigned int freq, int dir)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+	int ret = -ENOTSUPP;
+
+	if (dir == SND_SOC_CLOCK_OUT && clk_id == 0) {
+		if (!iface->mclk) {
+			dev_warn(dai->dev, "master clock not provided\n");
+		} else {
+			ret = clk_set_rate(iface->mclk, freq);
+			if (!ret)
+				iface->mclk_rate = freq;
+		}
+	}
+
+	return ret;
+}
+
+static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+
+	/* These modes are not supported */
+	if (fmt & (SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_CBM_CFS)) {
+		dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n");
+		return -EINVAL;
+	}
+
+	/* If the TDM interface is the clock master, it requires mclk */
+	if (!iface->mclk && (fmt & SND_SOC_DAIFMT_CBS_CFS)) {
+		dev_err(dai->dev, "cpu clock master: mclk missing\n");
+		return -ENODEV;
+	}
+
+	iface->fmt = fmt;
+	return 0;
+}
+
+static int axg_tdm_iface_startup(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+	struct axg_tdm_stream *ts =
+		snd_soc_dai_get_dma_data(dai, substream);
+	int ret;
+
+	if (!axg_tdm_slots_total(ts->mask)) {
+		dev_err(dai->dev, "interface has not slots\n");
+		return -EINVAL;
+	}
+
+	/* Apply component wide rate symmetry */
+	if (dai->component->active) {
+		ret = snd_pcm_hw_constraint_single(substream->runtime,
+						   SNDRV_PCM_HW_PARAM_RATE,
+						   iface->rate);
+		if (ret < 0) {
+			dev_err(dai->dev,
+				"can't set iface rate constraint\n");
+			return ret;
+		}
+	}
+
+	return 0;
+}
+
+static int axg_tdm_iface_set_stream(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params,
+				    struct snd_soc_dai *dai)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+	struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream);
+	unsigned int channels = params_channels(params);
+	unsigned int width = params_width(params);
+
+	/* Save rate and sample_bits for component symmetry */
+	iface->rate = params_rate(params);
+
+	/* Make sure this interface can cope with the stream */
+	if (axg_tdm_slots_total(ts->mask) < channels) {
+		dev_err(dai->dev, "not enough slots for channels\n");
+		return -EINVAL;
+	}
+
+	if (iface->slot_width < width) {
+		dev_err(dai->dev, "incompatible slots width for stream\n");
+		return -EINVAL;
+	}
+
+	/* Save the parameter for tdmout/tdmin widgets */
+	ts->physical_width = params_physical_width(params);
+	ts->width = params_width(params);
+	ts->channels = params_channels(params);
+
+	return 0;
+}
+
+static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai,
+				   struct snd_pcm_hw_params *params)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+	unsigned int ratio_num;
+	int ret;
+
+	ret = clk_set_rate(iface->lrclk, params_rate(params));
+	if (ret) {
+		dev_err(dai->dev, "setting sample clock failed: %d\n", ret);
+		return ret;
+	}
+
+	switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_RIGHT_J:
+		/* 50% duty cycle ratio */
+		ratio_num = 1;
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A:
+	case SND_SOC_DAIFMT_DSP_B:
+		/*
+		 * A zero duty cycle ratio will result in setting the mininum
+		 * ratio possible which, for this clock, is 1 cycle of the
+		 * parent bclk clock high and the rest low, This is exactly
+		 * what we want here.
+		 */
+		ratio_num = 0;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2);
+	if (ret) {
+		dev_err(dai->dev,
+			"setting sample clock duty cycle failed: %d\n", ret);
+		return ret;
+	}
+
+	/* Set sample clock inversion */
+	ret = clk_set_phase(iface->lrclk,
+			    axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0);
+	if (ret) {
+		dev_err(dai->dev,
+			"setting sample clock phase failed: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int axg_tdm_iface_set_sclk(struct snd_soc_dai *dai,
+				  struct snd_pcm_hw_params *params)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+	unsigned long srate;
+	int ret;
+
+	srate = iface->slots * iface->slot_width * params_rate(params);
+
+	if (!iface->mclk_rate) {
+		/* If no specific mclk is requested, default to bit clock * 4 */
+		clk_set_rate(iface->mclk, 4 * srate);
+	} else {
+		/* Check if we can actually get the bit clock from mclk */
+		if (iface->mclk_rate % srate) {
+			dev_err(dai->dev,
+				"can't derive sclk %lu from mclk %lu\n",
+				srate, iface->mclk_rate);
+			return -EINVAL;
+		}
+	}
+
+	ret = clk_set_rate(iface->sclk, srate);
+	if (ret) {
+		dev_err(dai->dev, "setting bit clock failed: %d\n", ret);
+		return ret;
+	}
+
+	/* Set the bit clock inversion */
+	ret = clk_set_phase(iface->sclk,
+			    axg_tdm_sclk_invert(iface->fmt) ? 0 : 180);
+	if (ret) {
+		dev_err(dai->dev, "setting bit clock phase failed: %d\n", ret);
+		return ret;
+	}
+
+	return ret;
+}
+
+static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream,
+				   struct snd_pcm_hw_params *params,
+				   struct snd_soc_dai *dai)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_RIGHT_J:
+		if (iface->slots > 2) {
+			dev_err(dai->dev, "bad slot number for format: %d\n",
+				iface->slots);
+			return -EINVAL;
+		}
+		break;
+
+	case SND_SOC_DAI_FORMAT_DSP_A:
+	case SND_SOC_DAI_FORMAT_DSP_B:
+		break;
+
+	default:
+		dev_err(dai->dev, "unsupported dai format\n");
+		return -EINVAL;
+	}
+
+	ret = axg_tdm_iface_set_stream(substream, params, dai);
+	if (ret)
+		return ret;
+
+	if (iface->fmt & SND_SOC_DAIFMT_CBS_CFS) {
+		ret = axg_tdm_iface_set_sclk(dai, params);
+		if (ret)
+			return ret;
+
+		ret = axg_tdm_iface_set_lrclk(dai, params);
+		if (ret)
+			return ret;
+	}
+
+	return 0;
+}
+
+static int axg_tdm_iface_hw_free(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream);
+
+	/* Stop all attached formatters */
+	axg_tdm_stream_stop(ts);
+
+	return 0;
+}
+
+static int axg_tdm_iface_prepare(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream);
+
+	/* Force all attached formatters to update */
+	return axg_tdm_stream_reset(ts);
+}
+
+static int axg_tdm_iface_remove_dai(struct snd_soc_dai *dai)
+{
+	if (dai->capture_dma_data)
+		axg_tdm_stream_free(dai->capture_dma_data);
+
+	if (dai->playback_dma_data)
+		axg_tdm_stream_free(dai->playback_dma_data);
+
+	return 0;
+}
+
+static int axg_tdm_iface_probe_dai(struct snd_soc_dai *dai)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+
+	if (dai->capture_widget) {
+		dai->capture_dma_data = axg_tdm_stream_alloc(iface);
+		if (!dai->capture_dma_data)
+			return -ENOMEM;
+	}
+
+	if (dai->playback_widget) {
+		dai->playback_dma_data = axg_tdm_stream_alloc(iface);
+		if (!dai->playback_dma_data) {
+			axg_tdm_iface_remove_dai(dai);
+			return -ENOMEM;
+		}
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops axg_tdm_iface_ops = {
+	.set_sysclk	= axg_tdm_iface_set_sysclk,
+	.set_fmt	= axg_tdm_iface_set_fmt,
+	.startup	= axg_tdm_iface_startup,
+	.hw_params	= axg_tdm_iface_hw_params,
+	.prepare	= axg_tdm_iface_prepare,
+	.hw_free	= axg_tdm_iface_hw_free,
+};
+
+/* TDM Backend DAIs */
+static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = {
+	[TDM_IFACE_PAD] = {
+		.name = "TDM Pad",
+		.playback = {
+			.stream_name	= "Playback",
+			.channels_min	= 1,
+			.channels_max	= AXG_TDM_CHANNEL_MAX,
+			.rates		= AXG_TDM_RATES,
+			.formats	= AXG_TDM_FORMATS,
+		},
+		.capture = {
+			.stream_name	= "Capture",
+			.channels_min	= 1,
+			.channels_max	= AXG_TDM_CHANNEL_MAX,
+			.rates		= AXG_TDM_RATES,
+			.formats	= AXG_TDM_FORMATS,
+		},
+		.id = TDM_IFACE_PAD,
+		.ops = &axg_tdm_iface_ops,
+		.probe = axg_tdm_iface_probe_dai,
+		.remove = axg_tdm_iface_remove_dai,
+	},
+	[TDM_IFACE_LOOPBACK] = {
+		.name = "TDM Loopback",
+		.capture = {
+			.stream_name	= "Loopback",
+			.channels_min	= 1,
+			.channels_max	= AXG_TDM_CHANNEL_MAX,
+			.rates		= AXG_TDM_RATES,
+			.formats	= AXG_TDM_FORMATS,
+		},
+		.id = TDM_IFACE_LOOPBACK,
+		.ops = &axg_tdm_iface_ops,
+		.probe = axg_tdm_iface_probe_dai,
+		.remove = axg_tdm_iface_remove_dai,
+	},
+};
+
+static int axg_tdm_iface_set_bias_level(struct snd_soc_component *component,
+					enum snd_soc_bias_level level)
+{
+	struct axg_tdm_iface *iface = snd_soc_component_get_drvdata(component);
+	enum snd_soc_bias_level now =
+		snd_soc_component_get_bias_level(component);
+	int ret = 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		if (now == SND_SOC_BIAS_STANDBY)
+			ret = clk_prepare_enable(iface->mclk);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (now == SND_SOC_BIAS_PREPARE)
+			clk_disable_unprepare(iface->mclk);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+	case SND_SOC_BIAS_ON:
+		break;
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_component_driver axg_tdm_iface_component_drv = {
+	.set_bias_level	= axg_tdm_iface_set_bias_level,
+};
+
+static const struct of_device_id axg_tdm_iface_of_match[] = {
+	{ .compatible = "amlogic,axg-tdm-iface", },
+	{}
+};
+MODULE_DEVICE_TABLE(of, axg_tdm_iface_of_match);
+
+static int axg_tdm_iface_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct snd_soc_dai_driver *dai_drv;
+	struct axg_tdm_iface *iface;
+	int ret, i;
+
+	iface = devm_kzalloc(dev, sizeof(*iface), GFP_KERNEL);
+	if (!iface)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, iface);
+
+	/*
+	 * Duplicate dai driver: depending on the slot masks configuration
+	 * We'll change the number of channel provided by DAI stream, so dpcm
+	 * channel merge can be done properly
+	 */
+	dai_drv = devm_kcalloc(dev, ARRAY_SIZE(axg_tdm_iface_dai_drv),
+			       sizeof(*dai_drv), GFP_KERNEL);
+	if (!dai_drv)
+		return -ENOMEM;
+
+	for (i = 0; i < ARRAY_SIZE(axg_tdm_iface_dai_drv); i++)
+		memcpy(&dai_drv[i], &axg_tdm_iface_dai_drv[i],
+		       sizeof(*dai_drv));
+
+	/* Bit clock provided on the pad */
+	iface->sclk = devm_clk_get(dev, "sclk");
+	if (IS_ERR(iface->sclk)) {
+		ret = PTR_ERR(iface->sclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get sclk: %d\n", ret);
+		return ret;
+	}
+
+	/* Sample clock provided on the pad */
+	iface->lrclk = devm_clk_get(dev, "lrclk");
+	if (IS_ERR(iface->lrclk)) {
+		ret = PTR_ERR(iface->lrclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get lrclk: %d\n", ret);
+		return ret;
+	}
+
+	/*
+	 * mclk maybe be missing when the cpu dai is in slave mode and
+	 * the codec does not require it to provide a master clock.
+	 * At this point, ignore the error if mclk is missing. We'll
+	 * throw an error if the cpu dai is master and mclk is missing
+	 */
+	iface->mclk = devm_clk_get(dev, "mclk");
+	if (IS_ERR(iface->mclk)) {
+		ret = PTR_ERR(iface->mclk);
+		if (ret == -ENOENT) {
+			iface->mclk = NULL;
+		} else {
+			if (ret != -EPROBE_DEFER)
+				dev_err(dev, "failed to get mclk: %d\n", ret);
+			return ret;
+		}
+	}
+
+	return devm_snd_soc_register_component(dev,
+					&axg_tdm_iface_component_drv, dai_drv,
+					ARRAY_SIZE(axg_tdm_iface_dai_drv));
+}
+
+static struct platform_driver axg_tdm_iface_pdrv = {
+	.probe = axg_tdm_iface_probe,
+	.driver = {
+		.name = "axg-tdm-iface",
+		.of_match_table = axg_tdm_iface_of_match,
+	},
+};
+module_platform_driver(axg_tdm_iface_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG TDM interface driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h
new file mode 100644
index 000000000000..e578b6f40a07
--- /dev/null
+++ b/sound/soc/meson/axg-tdm.h
@@ -0,0 +1,78 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR MIT)
+ *
+ * Copyright (c) 2018 Baylibre SAS.
+ * Author: Jerome Brunet <jbrunet@baylibre.com>
+ */
+
+#ifndef _MESON_AXG_TDM_H
+#define _MESON_AXG_TDM_H
+
+#include <linux/clk.h>
+#include <linux/regmap.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#define AXG_TDM_NUM_LANES	4
+#define AXG_TDM_CHANNEL_MAX	128
+#define AXG_TDM_RATES		(SNDRV_PCM_RATE_5512 |		\
+				 SNDRV_PCM_RATE_8000_192000)
+#define AXG_TDM_FORMATS		(SNDRV_PCM_FMTBIT_S8 |		\
+				 SNDRV_PCM_FMTBIT_S16_LE |	\
+				 SNDRV_PCM_FMTBIT_S20_LE |	\
+				 SNDRV_PCM_FMTBIT_S24_LE |	\
+				 SNDRV_PCM_FMTBIT_S32_LE)
+
+struct axg_tdm_iface {
+	struct clk *sclk;
+	struct clk *lrclk;
+	struct clk *mclk;
+	unsigned long mclk_rate;
+
+	/* format is common to all the DAIs of the iface */
+	unsigned int fmt;
+	unsigned int slots;
+	unsigned int slot_width;
+
+	/* For component wide symmetry */
+	int rate;
+};
+
+static inline bool axg_tdm_lrclk_invert(unsigned int fmt)
+{
+	return (fmt & SND_SOC_DAIFMT_I2S) ^
+		!!(fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_NB_IF));
+}
+
+static inline bool axg_tdm_sclk_invert(unsigned int fmt)
+{
+	return fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_IB_NF);
+}
+
+struct axg_tdm_stream {
+	struct axg_tdm_iface *iface;
+	struct list_head formatter_list;
+	struct mutex lock;
+	unsigned int channels;
+	unsigned int width;
+	unsigned int physical_width;
+	u32 *mask;
+	bool ready;
+};
+
+struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface);
+void axg_tdm_stream_free(struct axg_tdm_stream *ts);
+int axg_tdm_stream_start(struct axg_tdm_stream *ts);
+void axg_tdm_stream_stop(struct axg_tdm_stream *ts);
+
+static inline int axg_tdm_stream_reset(struct axg_tdm_stream *ts)
+{
+	axg_tdm_stream_stop(ts);
+	return axg_tdm_stream_start(ts);
+}
+
+int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
+			  u32 *rx_mask, unsigned int slots,
+			  unsigned int slot_width);
+
+#endif /* _MESON_AXG_TDM_H */
diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c
new file mode 100644
index 000000000000..bbac44c81688
--- /dev/null
+++ b/sound/soc/meson/axg-tdmin.c
@@ -0,0 +1,229 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-tdm-formatter.h"
+
+#define TDMIN_CTRL			0x00
+#define  TDMIN_CTRL_ENABLE		BIT(31)
+#define  TDMIN_CTRL_I2S_MODE		BIT(30)
+#define  TDMIN_CTRL_RST_OUT		BIT(29)
+#define  TDMIN_CTRL_RST_IN		BIT(28)
+#define  TDMIN_CTRL_WS_INV		BIT(25)
+#define  TDMIN_CTRL_SEL_SHIFT		20
+#define  TDMIN_CTRL_IN_BIT_SKEW_MASK	GENMASK(18, 16)
+#define  TDMIN_CTRL_IN_BIT_SKEW(x)	((x) << 16)
+#define  TDMIN_CTRL_LSB_FIRST		BIT(5)
+#define  TDMIN_CTRL_BITNUM_MASK	GENMASK(4, 0)
+#define  TDMIN_CTRL_BITNUM(x)		((x) << 0)
+#define TDMIN_SWAP			0x04
+#define TDMIN_MASK0			0x08
+#define TDMIN_MASK1			0x0c
+#define TDMIN_MASK2			0x10
+#define TDMIN_MASK3			0x14
+#define TDMIN_STAT			0x18
+#define TDMIN_MUTE_VAL			0x1c
+#define TDMIN_MUTE0			0x20
+#define TDMIN_MUTE1			0x24
+#define TDMIN_MUTE2			0x28
+#define TDMIN_MUTE3			0x2c
+
+static const struct regmap_config axg_tdmin_regmap_cfg = {
+	.reg_bits	= 32,
+	.val_bits	= 32,
+	.reg_stride	= 4,
+	.max_register	= TDMIN_MUTE3,
+};
+
+static const char * const axg_tdmin_sel_texts[] = {
+	"IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 5",
+};
+
+/* Change to special mux control to reset dapm */
+static SOC_ENUM_SINGLE_DECL(axg_tdmin_sel_enum, TDMIN_CTRL,
+			    TDMIN_CTRL_SEL_SHIFT, axg_tdmin_sel_texts);
+
+static const struct snd_kcontrol_new axg_tdmin_in_mux =
+	SOC_DAPM_ENUM("Input Source", axg_tdmin_sel_enum);
+
+static struct snd_soc_dai *
+axg_tdmin_get_be(struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dapm_path *p = NULL;
+	struct snd_soc_dai *be;
+
+	snd_soc_dapm_widget_for_each_source_path(w, p) {
+		if (!p->connect)
+			continue;
+
+		if (p->source->id == snd_soc_dapm_dai_out)
+			return (struct snd_soc_dai *)p->source->priv;
+
+		be = axg_tdmin_get_be(p->source);
+		if (be)
+			return be;
+	}
+
+	return NULL;
+}
+
+static struct axg_tdm_stream *
+axg_tdmin_get_tdm_stream(struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dai *be = axg_tdmin_get_be(w);
+
+	if (!be)
+		return NULL;
+
+	return be->capture_dma_data;
+}
+
+static void axg_tdmin_enable(struct regmap *map)
+{
+	/* Apply both reset */
+	regmap_update_bits(map, TDMIN_CTRL,
+			   TDMIN_CTRL_RST_OUT | TDMIN_CTRL_RST_IN, 0);
+
+	/* Clear out reset before in reset */
+	regmap_update_bits(map, TDMIN_CTRL,
+			   TDMIN_CTRL_RST_OUT, TDMIN_CTRL_RST_OUT);
+	regmap_update_bits(map, TDMIN_CTRL,
+			   TDMIN_CTRL_RST_IN,  TDMIN_CTRL_RST_IN);
+
+	/* Actually enable tdmin */
+	regmap_update_bits(map, TDMIN_CTRL,
+			   TDMIN_CTRL_ENABLE, TDMIN_CTRL_ENABLE);
+}
+
+static void axg_tdmin_disable(struct regmap *map)
+{
+	regmap_update_bits(map, TDMIN_CTRL, TDMIN_CTRL_ENABLE, 0);
+}
+
+static int axg_tdmin_prepare(struct regmap *map, struct axg_tdm_stream *ts)
+{
+	unsigned int val = 0;
+
+	/* Set stream skew */
+	switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_DSP_A:
+		val |= TDMIN_CTRL_IN_BIT_SKEW(3);
+		break;
+
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_RIGHT_J:
+	case SND_SOC_DAIFMT_DSP_B:
+		val = TDMIN_CTRL_IN_BIT_SKEW(2);
+		break;
+
+	default:
+		pr_err("Unsupported format: %u\n",
+		       ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+		return -EINVAL;
+	}
+
+	/* Set stream format mode */
+	switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_RIGHT_J:
+		val |= TDMIN_CTRL_I2S_MODE;
+		break;
+	}
+
+	/* If the sample clock is inverted, invert it back for the formatter */
+	if (axg_tdm_lrclk_invert(ts->iface->fmt))
+		val |= TDMIN_CTRL_WS_INV;
+
+	/* Set the slot width */
+	val |= TDMIN_CTRL_BITNUM(ts->iface->slot_width - 1);
+
+	/*
+	 * The following also reset LSB_FIRST which result in the formatter
+	 * placing the first bit received at bit 31
+	 */
+	regmap_update_bits(map, TDMIN_CTRL,
+			   (TDMIN_CTRL_IN_BIT_SKEW_MASK | TDMIN_CTRL_WS_INV |
+			    TDMIN_CTRL_I2S_MODE | TDMIN_CTRL_LSB_FIRST |
+			    TDMIN_CTRL_BITNUM_MASK), val);
+
+	/* Set static swap mask configuration */
+	regmap_write(map, TDMIN_SWAP, 0x76543210);
+
+	return axg_tdm_formatter_set_channel_masks(map, ts, TDMIN_MASK0);
+}
+
+static const struct snd_soc_dapm_widget axg_tdmin_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 5", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmin_in_mux),
+	SND_SOC_DAPM_PGA_E("DEC", SND_SOC_NOPM, 0, 0, NULL, 0,
+			   axg_tdm_formatter_event,
+			   (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)),
+	SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route axg_tdmin_dapm_routes[] = {
+	{ "SRC SEL", "IN 0", "IN 0" },
+	{ "SRC SEL", "IN 1", "IN 1" },
+	{ "SRC SEL", "IN 2", "IN 2" },
+	{ "SRC SEL", "IN 3", "IN 3" },
+	{ "SRC SEL", "IN 4", "IN 4" },
+	{ "SRC SEL", "IN 5", "IN 5" },
+	{ "DEC", NULL, "SRC SEL" },
+	{ "OUT", NULL, "DEC" },
+};
+
+static const struct snd_soc_component_driver axg_tdmin_component_drv = {
+	.dapm_widgets		= axg_tdmin_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(axg_tdmin_dapm_widgets),
+	.dapm_routes		= axg_tdmin_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(axg_tdmin_dapm_routes),
+};
+
+static const struct axg_tdm_formatter_ops axg_tdmin_ops = {
+	.get_stream	= axg_tdmin_get_tdm_stream,
+	.prepare	= axg_tdmin_prepare,
+	.enable		= axg_tdmin_enable,
+	.disable	= axg_tdmin_disable,
+};
+
+static const struct axg_tdm_formatter_driver axg_tdmin_drv = {
+	.component_drv	= &axg_tdmin_component_drv,
+	.regmap_cfg	= &axg_tdmin_regmap_cfg,
+	.ops		= &axg_tdmin_ops,
+	.invert_sclk	= false,
+};
+
+static const struct of_device_id axg_tdmin_of_match[] = {
+	{
+		.compatible = "amlogic,axg-tdmin",
+		.data = &axg_tdmin_drv,
+	}, {}
+};
+MODULE_DEVICE_TABLE(of, axg_tdmin_of_match);
+
+static struct platform_driver axg_tdmin_pdrv = {
+	.probe = axg_tdm_formatter_probe,
+	.driver = {
+		.name = "axg-tdmin",
+		.of_match_table = axg_tdmin_of_match,
+	},
+};
+module_platform_driver(axg_tdmin_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG TDM input formatter driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c
new file mode 100644
index 000000000000..f73368ee1088
--- /dev/null
+++ b/sound/soc/meson/axg-tdmout.c
@@ -0,0 +1,259 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-tdm-formatter.h"
+
+#define TDMOUT_CTRL0			0x00
+#define  TDMOUT_CTRL0_BITNUM_MASK	GENMASK(4, 0)
+#define  TDMOUT_CTRL0_BITNUM(x)		((x) << 0)
+#define  TDMOUT_CTRL0_SLOTNUM_MASK	GENMASK(9, 5)
+#define  TDMOUT_CTRL0_SLOTNUM(x)	((x) << 5)
+#define  TDMOUT_CTRL0_INIT_BITNUM_MASK	GENMASK(19, 15)
+#define  TDMOUT_CTRL0_INIT_BITNUM(x)	((x) << 15)
+#define  TDMOUT_CTRL0_ENABLE		BIT(31)
+#define  TDMOUT_CTRL0_RST_OUT		BIT(29)
+#define  TDMOUT_CTRL0_RST_IN		BIT(28)
+#define TDMOUT_CTRL1			0x04
+#define  TDMOUT_CTRL1_TYPE_MASK		GENMASK(6, 4)
+#define  TDMOUT_CTRL1_TYPE(x)		((x) << 4)
+#define  TDMOUT_CTRL1_MSB_POS_MASK	GENMASK(12, 8)
+#define  TDMOUT_CTRL1_MSB_POS(x)	((x) << 8)
+#define  TDMOUT_CTRL1_SEL_SHIFT		24
+#define  TDMOUT_CTRL1_GAIN_EN		26
+#define  TDMOUT_CTRL1_WS_INV		BIT(28)
+#define TDMOUT_SWAP			0x08
+#define TDMOUT_MASK0			0x0c
+#define TDMOUT_MASK1			0x10
+#define TDMOUT_MASK2			0x14
+#define TDMOUT_MASK3			0x18
+#define TDMOUT_STAT			0x1c
+#define TDMOUT_GAIN0			0x20
+#define TDMOUT_GAIN1			0x24
+#define TDMOUT_MUTE_VAL			0x28
+#define TDMOUT_MUTE0			0x2c
+#define TDMOUT_MUTE1			0x30
+#define TDMOUT_MUTE2			0x34
+#define TDMOUT_MUTE3			0x38
+#define TDMOUT_MASK_VAL			0x3c
+
+static const struct regmap_config axg_tdmout_regmap_cfg = {
+	.reg_bits	= 32,
+	.val_bits	= 32,
+	.reg_stride	= 4,
+	.max_register	= TDMOUT_MASK_VAL,
+};
+
+static const struct snd_kcontrol_new axg_tdmout_controls[] = {
+	SOC_DOUBLE("Lane 0 Volume", TDMOUT_GAIN0,  0,  8, 255, 0),
+	SOC_DOUBLE("Lane 1 Volume", TDMOUT_GAIN0, 16, 24, 255, 0),
+	SOC_DOUBLE("Lane 2 Volume", TDMOUT_GAIN1,  0,  8, 255, 0),
+	SOC_DOUBLE("Lane 3 Volume", TDMOUT_GAIN1, 16, 24, 255, 0),
+	SOC_SINGLE("Gain Enable Switch", TDMOUT_CTRL1,
+		   TDMOUT_CTRL1_GAIN_EN, 1, 0),
+};
+
+static const char * const tdmout_sel_texts[] = {
+	"IN 0", "IN 1", "IN 2",
+};
+
+static SOC_ENUM_SINGLE_DECL(axg_tdmout_sel_enum, TDMOUT_CTRL1,
+			    TDMOUT_CTRL1_SEL_SHIFT, tdmout_sel_texts);
+
+static const struct snd_kcontrol_new axg_tdmout_in_mux =
+	SOC_DAPM_ENUM("Input Source", axg_tdmout_sel_enum);
+
+static struct snd_soc_dai *
+axg_tdmout_get_be(struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dapm_path *p = NULL;
+	struct snd_soc_dai *be;
+
+	snd_soc_dapm_widget_for_each_sink_path(w, p) {
+		if (!p->connect)
+			continue;
+
+		if (p->sink->id == snd_soc_dapm_dai_in)
+			return (struct snd_soc_dai *)p->sink->priv;
+
+		be = axg_tdmout_get_be(p->sink);
+		if (be)
+			return be;
+	}
+
+	return NULL;
+}
+
+static struct axg_tdm_stream *
+axg_tdmout_get_tdm_stream(struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dai *be = axg_tdmout_get_be(w);
+
+	if (!be)
+		return NULL;
+
+	return be->playback_dma_data;
+}
+
+static void axg_tdmout_enable(struct regmap *map)
+{
+	/* Apply both reset */
+	regmap_update_bits(map, TDMOUT_CTRL0,
+			   TDMOUT_CTRL0_RST_OUT | TDMOUT_CTRL0_RST_IN, 0);
+
+	/* Clear out reset before in reset */
+	regmap_update_bits(map, TDMOUT_CTRL0,
+			   TDMOUT_CTRL0_RST_OUT, TDMOUT_CTRL0_RST_OUT);
+	regmap_update_bits(map, TDMOUT_CTRL0,
+			   TDMOUT_CTRL0_RST_IN,  TDMOUT_CTRL0_RST_IN);
+
+	/* Actually enable tdmout */
+	regmap_update_bits(map, TDMOUT_CTRL0,
+			   TDMOUT_CTRL0_ENABLE, TDMOUT_CTRL0_ENABLE);
+}
+
+static void axg_tdmout_disable(struct regmap *map)
+{
+	regmap_update_bits(map, TDMOUT_CTRL0, TDMOUT_CTRL0_ENABLE, 0);
+}
+
+static int axg_tdmout_prepare(struct regmap *map, struct axg_tdm_stream *ts)
+{
+	unsigned int val = 0;
+
+	/* Set the stream skew */
+	switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_DSP_A:
+		val |= TDMOUT_CTRL0_INIT_BITNUM(1);
+		break;
+
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_RIGHT_J:
+	case SND_SOC_DAIFMT_DSP_B:
+		val |= TDMOUT_CTRL0_INIT_BITNUM(2);
+		break;
+
+	default:
+		pr_err("Unsupported format: %u\n",
+		       ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+		return -EINVAL;
+	}
+
+	/* Set the slot width */
+	val |= TDMOUT_CTRL0_BITNUM(ts->iface->slot_width - 1);
+
+	/* Set the slot number */
+	val |= TDMOUT_CTRL0_SLOTNUM(ts->iface->slots - 1);
+
+	regmap_update_bits(map, TDMOUT_CTRL0,
+			   TDMOUT_CTRL0_INIT_BITNUM_MASK |
+			   TDMOUT_CTRL0_BITNUM_MASK |
+			   TDMOUT_CTRL0_SLOTNUM_MASK, val);
+
+	/* Set the sample width */
+	val = TDMOUT_CTRL1_MSB_POS(ts->width - 1);
+
+	/* FIFO data are arranged in chunks of 64bits */
+	switch (ts->physical_width) {
+	case 8:
+		/* 8 samples of 8 bits */
+		val |= TDMOUT_CTRL1_TYPE(0);
+		break;
+	case 16:
+		/* 4 samples of 16 bits - right justified */
+		val |= TDMOUT_CTRL1_TYPE(2);
+		break;
+	case 32:
+		/* 2 samples of 32 bits - right justified */
+		val |= TDMOUT_CTRL1_TYPE(4);
+		break;
+	default:
+		pr_err("Unsupported physical width: %u\n",
+		       ts->physical_width);
+		return -EINVAL;
+	}
+
+	/* If the sample clock is inverted, invert it back for the formatter */
+	if (axg_tdm_lrclk_invert(ts->iface->fmt))
+		val |= TDMOUT_CTRL1_WS_INV;
+
+	regmap_update_bits(map, TDMOUT_CTRL1,
+			   (TDMOUT_CTRL1_TYPE_MASK | TDMOUT_CTRL1_MSB_POS_MASK |
+			    TDMOUT_CTRL1_WS_INV), val);
+
+	/* Set static swap mask configuration */
+	regmap_write(map, TDMOUT_SWAP, 0x76543210);
+
+	return axg_tdm_formatter_set_channel_masks(map, ts, TDMOUT_MASK0);
+}
+
+static const struct snd_soc_dapm_widget axg_tdmout_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmout_in_mux),
+	SND_SOC_DAPM_PGA_E("ENC", SND_SOC_NOPM, 0, 0, NULL, 0,
+			   axg_tdm_formatter_event,
+			   (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)),
+	SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route axg_tdmout_dapm_routes[] = {
+	{ "SRC SEL", "IN 0", "IN 0" },
+	{ "SRC SEL", "IN 1", "IN 1" },
+	{ "SRC SEL", "IN 2", "IN 2" },
+	{ "ENC", NULL, "SRC SEL" },
+	{ "OUT", NULL, "ENC" },
+};
+
+static const struct snd_soc_component_driver axg_tdmout_component_drv = {
+	.controls		= axg_tdmout_controls,
+	.num_controls		= ARRAY_SIZE(axg_tdmout_controls),
+	.dapm_widgets		= axg_tdmout_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(axg_tdmout_dapm_widgets),
+	.dapm_routes		= axg_tdmout_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(axg_tdmout_dapm_routes),
+};
+
+static const struct axg_tdm_formatter_ops axg_tdmout_ops = {
+	.get_stream	= axg_tdmout_get_tdm_stream,
+	.prepare	= axg_tdmout_prepare,
+	.enable		= axg_tdmout_enable,
+	.disable	= axg_tdmout_disable,
+};
+
+static const struct axg_tdm_formatter_driver axg_tdmout_drv = {
+	.component_drv	= &axg_tdmout_component_drv,
+	.regmap_cfg	= &axg_tdmout_regmap_cfg,
+	.ops		= &axg_tdmout_ops,
+	.invert_sclk	= true,
+};
+
+static const struct of_device_id axg_tdmout_of_match[] = {
+	{
+		.compatible = "amlogic,axg-tdmout",
+		.data = &axg_tdmout_drv,
+	}, {}
+};
+MODULE_DEVICE_TABLE(of, axg_tdmout_of_match);
+
+static struct platform_driver axg_tdmout_pdrv = {
+	.probe = axg_tdm_formatter_probe,
+	.driver = {
+		.name = "axg-tdmout",
+		.of_match_table = axg_tdmout_of_match,
+	},
+};
+module_platform_driver(axg_tdmout_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG TDM output formatter driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c
new file mode 100644
index 000000000000..c2c9bb312586
--- /dev/null
+++ b/sound/soc/meson/axg-toddr.c
@@ -0,0 +1,199 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+/* This driver implements the frontend capture DAI of AXG based SoCs */
+
+#include <linux/clk.h>
+#include <linux/regmap.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-fifo.h"
+
+#define CTRL0_TODDR_SEL_RESAMPLE	BIT(30)
+#define CTRL0_TODDR_EXT_SIGNED		BIT(29)
+#define CTRL0_TODDR_PP_MODE		BIT(28)
+#define CTRL0_TODDR_TYPE_MASK		GENMASK(15, 13)
+#define CTRL0_TODDR_TYPE(x)		((x) << 13)
+#define CTRL0_TODDR_MSB_POS_MASK	GENMASK(12, 8)
+#define CTRL0_TODDR_MSB_POS(x)		((x) << 8)
+#define CTRL0_TODDR_LSB_POS_MASK	GENMASK(7, 3)
+#define CTRL0_TODDR_LSB_POS(x)		((x) << 3)
+
+static int axg_toddr_pcm_new(struct snd_soc_pcm_runtime *rtd,
+			     struct snd_soc_dai *dai)
+{
+	return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_CAPTURE);
+}
+
+static int axg_toddr_dai_hw_params(struct snd_pcm_substream *substream,
+				   struct snd_pcm_hw_params *params,
+				   struct snd_soc_dai *dai)
+{
+	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+	unsigned int type, width, msb = 31;
+
+	/*
+	 * NOTE:
+	 * Almost all backend will place the MSB at bit 31, except SPDIF Input
+	 * which will put it at index 28. When adding support for the SPDIF
+	 * Input, we'll need to find which type of backend we are connected to.
+	 */
+
+	switch (params_physical_width(params)) {
+	case 8:
+		type = 0; /* 8 samples of 8 bits */
+		break;
+	case 16:
+		type = 2; /* 4 samples of 16 bits - right justified */
+		break;
+	case 32:
+		type = 4; /* 2 samples of 32 bits - right justified */
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	width = params_width(params);
+
+	regmap_update_bits(fifo->map, FIFO_CTRL0,
+			   CTRL0_TODDR_TYPE_MASK |
+			   CTRL0_TODDR_MSB_POS_MASK |
+			   CTRL0_TODDR_LSB_POS_MASK,
+			   CTRL0_TODDR_TYPE(type) |
+			   CTRL0_TODDR_MSB_POS(msb) |
+			   CTRL0_TODDR_LSB_POS(msb - (width - 1)));
+
+	return 0;
+}
+
+static int axg_toddr_dai_startup(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+	unsigned int fifo_threshold;
+	int ret;
+
+	/* Enable pclk to access registers and clock the fifo ip */
+	ret = clk_prepare_enable(fifo->pclk);
+	if (ret)
+		return ret;
+
+	/* Select orginal data - resampling not supported ATM */
+	regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_SEL_RESAMPLE, 0);
+
+	/* Only signed format are supported ATM */
+	regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_EXT_SIGNED,
+			   CTRL0_TODDR_EXT_SIGNED);
+
+	/* Apply single buffer mode to the interface */
+	regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_PP_MODE, 0);
+
+	/* TODDR does not have a configurable fifo depth */
+	fifo_threshold = AXG_FIFO_MIN_CNT - 1;
+	regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_THRESHOLD_MASK,
+			   CTRL1_THRESHOLD(fifo_threshold));
+
+	return 0;
+}
+
+static void axg_toddr_dai_shutdown(struct snd_pcm_substream *substream,
+				   struct snd_soc_dai *dai)
+{
+	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+
+	clk_disable_unprepare(fifo->pclk);
+}
+
+static const struct snd_soc_dai_ops axg_toddr_ops = {
+	.hw_params	= axg_toddr_dai_hw_params,
+	.startup	= axg_toddr_dai_startup,
+	.shutdown	= axg_toddr_dai_shutdown,
+};
+
+static struct snd_soc_dai_driver axg_toddr_dai_drv = {
+	.name = "TODDR",
+	.capture = {
+		.stream_name	= "Capture",
+		.channels_min	= 1,
+		.channels_max	= AXG_FIFO_CH_MAX,
+		.rates		= AXG_FIFO_RATES,
+		.formats	= AXG_FIFO_FORMATS,
+	},
+	.ops		= &axg_toddr_ops,
+	.pcm_new	= axg_toddr_pcm_new,
+};
+
+static const char * const axg_toddr_sel_texts[] = {
+	"IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 6"
+};
+
+static const unsigned int axg_toddr_sel_values[] = {
+	0, 1, 2, 3, 4, 6
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(axg_toddr_sel_enum, FIFO_CTRL0,
+				  CTRL0_SEL_SHIFT, CTRL0_SEL_MASK,
+				  axg_toddr_sel_texts, axg_toddr_sel_values);
+
+static const struct snd_kcontrol_new axg_toddr_in_mux =
+	SOC_DAPM_ENUM("Input Source", axg_toddr_sel_enum);
+
+static const struct snd_soc_dapm_widget axg_toddr_dapm_widgets[] = {
+	SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_toddr_in_mux),
+	SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 6", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route axg_toddr_dapm_routes[] = {
+	{ "Capture", NULL, "SRC SEL" },
+	{ "SRC SEL", "IN 0", "IN 0" },
+	{ "SRC SEL", "IN 1", "IN 1" },
+	{ "SRC SEL", "IN 2", "IN 2" },
+	{ "SRC SEL", "IN 3", "IN 3" },
+	{ "SRC SEL", "IN 4", "IN 4" },
+	{ "SRC SEL", "IN 6", "IN 6" },
+};
+
+static const struct snd_soc_component_driver axg_toddr_component_drv = {
+	.dapm_widgets		= axg_toddr_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(axg_toddr_dapm_widgets),
+	.dapm_routes		= axg_toddr_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(axg_toddr_dapm_routes),
+	.ops			= &axg_fifo_pcm_ops
+};
+
+static const struct axg_fifo_match_data axg_toddr_match_data = {
+	.component_drv	= &axg_toddr_component_drv,
+	.dai_drv	= &axg_toddr_dai_drv
+};
+
+static const struct of_device_id axg_toddr_of_match[] = {
+	{
+		.compatible = "amlogic,axg-toddr",
+		.data = &axg_toddr_match_data,
+	}, {}
+};
+MODULE_DEVICE_TABLE(of, axg_toddr_of_match);
+
+static struct platform_driver axg_toddr_pdrv = {
+	.probe = axg_fifo_probe,
+	.driver = {
+		.name = "axg-toddr",
+		.of_match_table = axg_toddr_of_match,
+	},
+};
+module_platform_driver(axg_toddr_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG capture fifo driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 15ccbf479c96..d5ae9eb8c756 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -40,7 +40,7 @@ struct abe_twl6040 {
 	int	mclk_freq;	/* MCLK frequency speed for twl6040 */
 };
 
-struct platform_device *dmic_codec_dev;
+static struct platform_device *dmic_codec_dev;
 
 static int omap_abe_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 51dd7c65096b..fe966272bd0c 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -213,8 +213,10 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
 	switch (channels) {
 	case 6:
 		dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE;
+		/* fall through */
 	case 4:
 		dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE;
+		/* fall through */
 	case 2:
 		dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE;
 		break;
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 0e97360f9890..4c1be36c2207 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -310,15 +310,19 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
 			/* up to 3 channels for capture */
 			return -EINVAL;
 		link_mask |= 1 << 4;
+		/* fall through */
 	case 4:
 		if (stream == SNDRV_PCM_STREAM_CAPTURE)
 			/* up to 3 channels for capture */
 			return -EINVAL;
 		link_mask |= 1 << 3;
+		/* fall through */
 	case 3:
 		link_mask |= 1 << 2;
+		/* fall through */
 	case 2:
 		link_mask |= 1 << 1;
+		/* fall through */
 	case 1:
 		link_mask |= 1 << 0;
 		break;
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 960744e46edc..776e148b0aa2 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -24,15 +24,19 @@ config SND_PXA2XX_AC97
 config SND_PXA2XX_SOC_AC97
 	tristate
 	select AC97_BUS
+	select SND_PXA2XX_LIB
 	select SND_PXA2XX_LIB_AC97
 	select SND_SOC_AC97_BUS
 
 config SND_PXA2XX_SOC_I2S
+	select SND_PXA2XX_LIB
 	tristate
 
 config SND_PXA_SOC_SSP
-	tristate
+	tristate "Soc Audio via PXA2xx/PXA3xx SSP ports"
+	depends on PLAT_PXA
 	select PXA_SSP
+	select SND_PXA2XX_LIB
 
 config SND_MMP_SOC_SSPA
 	tristate
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 2fc012b06c43..935a248e5bf6 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -90,95 +90,9 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	unsigned int acps, acds, width;
-	unsigned int div4 = PXA_SSP_CLK_SCDB_4;
+	unsigned int width;
 	int ret = 0;
 
-	width = snd_pcm_format_physical_width(params_format(params));
-
-	/*
-	 * rate = SSPSCLK / (2 * width(16 or 32))
-	 * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
-	 */
-	switch (params_rate(params)) {
-	case 8000:
-		/* off by a factor of 2: bug in the PXA27x audio clock? */
-		acps = 32842000;
-		switch (width) {
-		case 16:
-			/* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_16;
-			break;
-		default: /* 32 */
-			/* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_8;
-		}
-		break;
-	case 11025:
-		acps = 5622000;
-		switch (width) {
-		case 16:
-			/* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_4;
-			break;
-		default: /* 32 */
-			/* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-		}
-		break;
-	case 22050:
-		acps = 5622000;
-		switch (width) {
-		case 16:
-			/* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			break;
-		default: /* 32 */
-			/* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-		}
-		break;
-	case 44100:
-		acps = 5622000;
-		switch (width) {
-		case 16:
-			/* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			break;
-		default: /* 32 */
-			/* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-		}
-		break;
-	case 48000:
-		acps = 12235000;
-		switch (width) {
-		case 16:
-			/* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			break;
-		default: /* 32 */
-			/* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-		}
-		break;
-	case 96000:
-	default:
-		acps = 12235000;
-		switch (width) {
-		case 16:
-			/* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-			break;
-		default: /* 32 */
-			/* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			div4 = PXA_SSP_CLK_SCDB_1;
-			break;
-		}
-		break;
-	}
-
 	/* set codec DAI configuration */
 	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
 			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
@@ -191,6 +105,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
+	width = snd_pcm_format_physical_width(params_format(params));
 	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width);
 	if (ret < 0)
 		return ret;
@@ -201,23 +116,6 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
-	/* set the SSP audio system clock ACDS divider */
-	ret = snd_soc_dai_set_clkdiv(cpu_dai,
-			PXA_SSP_AUDIO_DIV_ACDS, acds);
-	if (ret < 0)
-		return ret;
-
-	/* set the SSP audio system clock SCDB divider4 */
-	ret = snd_soc_dai_set_clkdiv(cpu_dai,
-			PXA_SSP_AUDIO_DIV_SCDB, div4);
-	if (ret < 0)
-		return ret;
-
-	/* set SSP audio pll clock */
-	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps);
-	if (ret < 0)
-		return ret;
-
 	return 0;
 }
 
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 6fc986080130..69033e1a84e6 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -34,7 +34,6 @@
 #include <sound/pxa2xx-lib.h>
 #include <sound/dmaengine_pcm.h>
 
-#include "../../arm/pxa2xx-pcm.h"
 #include "pxa-ssp.h"
 
 /*
@@ -42,6 +41,8 @@
  */
 struct ssp_priv {
 	struct ssp_device *ssp;
+	struct clk *extclk;
+	unsigned long ssp_clk;
 	unsigned int sysclk;
 	unsigned int dai_fmt;
 	unsigned int configured_dai_fmt;
@@ -105,9 +106,8 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 	dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL);
 	if (!dma)
 		return -ENOMEM;
-
-	dma->filter_data = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
-				&ssp->drcmr_tx : &ssp->drcmr_rx;
+	dma->chan_name = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+		"tx" : "rx";
 
 	snd_soc_dai_set_dma_data(cpu_dai, substream, dma);
 
@@ -194,21 +194,6 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div)
 	pxa_ssp_write_reg(ssp, SSCR0, sscr0);
 }
 
-/**
- * pxa_ssp_get_clkdiv - get SSP clock divider
- */
-static u32 pxa_ssp_get_scr(struct ssp_device *ssp)
-{
-	u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
-	u32 div;
-
-	if (ssp->type == PXA25x_SSP)
-		div = ((sscr0 >> 8) & 0xff) * 2 + 2;
-	else
-		div = ((sscr0 >> 8) & 0xfff) + 1;
-	return div;
-}
-
 /*
  * Set the SSP ports SYSCLK.
  */
@@ -221,6 +206,21 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) &
 		~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
 
+	if (priv->extclk) {
+		int ret;
+
+		/*
+		 * For DT based boards, if an extclk is given, use it
+		 * here and configure PXA_SSP_CLK_EXT.
+		 */
+
+		ret = clk_set_rate(priv->extclk, freq);
+		if (ret < 0)
+			return ret;
+
+		clk_id = PXA_SSP_CLK_EXT;
+	}
+
 	dev_dbg(&ssp->pdev->dev,
 		"pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n",
 		cpu_dai->id, clk_id, freq);
@@ -265,66 +265,17 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 }
 
 /*
- * Set the SSP clock dividers.
- */
-static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
-	int div_id, int div)
-{
-	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
-	struct ssp_device *ssp = priv->ssp;
-	int val;
-
-	switch (div_id) {
-	case PXA_SSP_AUDIO_DIV_ACDS:
-		val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div);
-		pxa_ssp_write_reg(ssp, SSACD, val);
-		break;
-	case PXA_SSP_AUDIO_DIV_SCDB:
-		val = pxa_ssp_read_reg(ssp, SSACD);
-		val &= ~SSACD_SCDB;
-		if (ssp->type == PXA3xx_SSP)
-			val &= ~SSACD_SCDX8;
-		switch (div) {
-		case PXA_SSP_CLK_SCDB_1:
-			val |= SSACD_SCDB;
-			break;
-		case PXA_SSP_CLK_SCDB_4:
-			break;
-		case PXA_SSP_CLK_SCDB_8:
-			if (ssp->type == PXA3xx_SSP)
-				val |= SSACD_SCDX8;
-			else
-				return -EINVAL;
-			break;
-		default:
-			return -EINVAL;
-		}
-		pxa_ssp_write_reg(ssp, SSACD, val);
-		break;
-	case PXA_SSP_DIV_SCR:
-		pxa_ssp_set_scr(ssp, div);
-		break;
-	default:
-		return -ENODEV;
-	}
-
-	return 0;
-}
-
-/*
  * Configure the PLL frequency pxa27x and (afaik - pxa320 only)
  */
-static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
-	int source, unsigned int freq_in, unsigned int freq_out)
+static int pxa_ssp_set_pll(struct ssp_priv *priv, unsigned int freq)
 {
-	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 	u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70;
 
 	if (ssp->type == PXA3xx_SSP)
 		pxa_ssp_write_reg(ssp, SSACDD, 0);
 
-	switch (freq_out) {
+	switch (freq) {
 	case 5622000:
 		break;
 	case 11345000:
@@ -355,7 +306,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 			u64 tmp = 19968;
 
 			tmp *= 1000000;
-			do_div(tmp, freq_out);
+			do_div(tmp, freq);
 			val = tmp;
 
 			val = (val << 16) | 64;
@@ -365,7 +316,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 
 			dev_dbg(&ssp->pdev->dev,
 				"Using SSACDD %x to supply %uHz\n",
-				val, freq_out);
+				val, freq);
 			break;
 		}
 
@@ -535,6 +486,7 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv)
 
 	case SND_SOC_DAIFMT_DSP_A:
 		sspsp |= SSPSP_FSRT;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_B:
 		sscr0 |= SSCR0_MOD | SSCR0_PSP;
 		sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
@@ -570,6 +522,24 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv)
 	return 0;
 }
 
+struct pxa_ssp_clock_mode {
+	int rate;
+	int pll;
+	u8 acds;
+	u8 scdb;
+};
+
+static const struct pxa_ssp_clock_mode pxa_ssp_clock_modes[] = {
+	{ .rate =  8000, .pll = 32842000, .acds = SSACD_ACDS_32, .scdb = SSACD_SCDB_4X },
+	{ .rate = 11025, .pll =  5622000, .acds = SSACD_ACDS_4,  .scdb = SSACD_SCDB_4X },
+	{ .rate = 16000, .pll = 32842000, .acds = SSACD_ACDS_16, .scdb = SSACD_SCDB_4X },
+	{ .rate = 22050, .pll =  5622000, .acds = SSACD_ACDS_2,  .scdb = SSACD_SCDB_4X },
+	{ .rate = 44100, .pll = 11345000, .acds = SSACD_ACDS_2,  .scdb = SSACD_SCDB_4X },
+	{ .rate = 48000, .pll = 12235000, .acds = SSACD_ACDS_2,  .scdb = SSACD_SCDB_4X },
+	{ .rate = 96000, .pll = 12235000, .acds = SSACD_ACDS_4,  .scdb = SSACD_SCDB_1X },
+	{}
+};
+
 /*
  * Set the SSP audio DMA parameters and sample size.
  * Can be called multiple times by oss emulation.
@@ -581,11 +551,12 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 	int chn = params_channels(params);
-	u32 sscr0;
-	u32 sspsp;
+	u32 sscr0, sspsp;
 	int width = snd_pcm_format_physical_width(params_format(params));
 	int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf;
 	struct snd_dmaengine_dai_dma_data *dma_data;
+	int rate = params_rate(params);
+	int bclk = rate * chn * (width / 8);
 	int ret;
 
 	dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream);
@@ -625,11 +596,57 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	}
 	pxa_ssp_write_reg(ssp, SSCR0, sscr0);
 
+	if (sscr0 & SSCR0_ACS) {
+		ret = pxa_ssp_set_pll(priv, bclk);
+
+		/*
+		 * If we were able to generate the bclk directly,
+		 * all is fine. Otherwise, look up the closest rate
+		 * from the table and also set the dividers.
+		 */
+
+		if (ret < 0) {
+			const struct pxa_ssp_clock_mode *m;
+			int ssacd, acds;
+
+			for (m = pxa_ssp_clock_modes; m->rate; m++) {
+				if (m->rate == rate)
+					break;
+			}
+
+			if (!m->rate)
+				return -EINVAL;
+
+			acds = m->acds;
+
+			/* The values in the table are for 16 bits */
+			if (width == 32)
+				acds--;
+
+			ret = pxa_ssp_set_pll(priv, bclk);
+			if (ret < 0)
+				return ret;
+
+			ssacd = pxa_ssp_read_reg(ssp, SSACD);
+			ssacd &= ~(SSACD_ACDS(7) | SSACD_SCDB_1X);
+			ssacd |= SSACD_ACDS(m->acds);
+			ssacd |= m->scdb;
+			pxa_ssp_write_reg(ssp, SSACD, ssacd);
+		}
+	} else if (sscr0 & SSCR0_ECS) {
+		/*
+		 * For setups with external clocking, the PLL and its diviers
+		 * are not active. Instead, the SCR bits in SSCR0 can be used
+		 * to divide the clock.
+		 */
+		pxa_ssp_set_scr(ssp, bclk / rate);
+	}
+
 	switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 	       sspsp = pxa_ssp_read_reg(ssp, SSPSP);
 
-		if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) {
+		if (((priv->sysclk / bclk) == 64) && (width == 16)) {
 			/* This is a special case where the bitclk is 64fs
 			 * and we're not dealing with 2*32 bits of audio
 			 * samples.
@@ -773,6 +790,15 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai)
 			ret = -ENODEV;
 			goto err_priv;
 		}
+
+		priv->extclk = devm_clk_get(dev, "extclk");
+		if (IS_ERR(priv->extclk)) {
+			ret = PTR_ERR(priv->extclk);
+			if (ret == -EPROBE_DEFER)
+				return ret;
+
+			priv->extclk = NULL;
+		}
 	} else {
 		priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio");
 		if (priv->ssp == NULL) {
@@ -814,8 +840,6 @@ static const struct snd_soc_dai_ops pxa_ssp_dai_ops = {
 	.trigger	= pxa_ssp_trigger,
 	.hw_params	= pxa_ssp_hw_params,
 	.set_sysclk	= pxa_ssp_set_dai_sysclk,
-	.set_clkdiv	= pxa_ssp_set_dai_clkdiv,
-	.set_pll	= pxa_ssp_set_dai_pll,
 	.set_fmt	= pxa_ssp_set_dai_fmt,
 	.set_tdm_slot	= pxa_ssp_set_dai_tdm_slot,
 	.set_tristate	= pxa_ssp_set_dai_tristate,
@@ -843,6 +867,9 @@ static struct snd_soc_dai_driver pxa_ssp_dai = {
 
 static const struct snd_soc_component_driver pxa_ssp_component = {
 	.name		= "pxa-ssp",
+	.ops		= &pxa2xx_pcm_ops,
+	.pcm_new	= pxa2xx_soc_pcm_new,
+	.pcm_free	= pxa2xx_pcm_free_dma_buffers,
 };
 
 #ifdef CONFIG_OF
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 803818aabee9..9f779657bc86 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -68,61 +68,39 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
 	.reset	= pxa2xx_ac97_cold_reset,
 };
 
-static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = {
-	.prio = PXAD_PRIO_LOWEST,
-	.drcmr = 11,
-};
-
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
 	.addr		= __PREG(PCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.chan_name	= "pcm_pcm_stereo_in",
 	.maxburst	= 32,
-	.filter_data	= &pxa2xx_ac97_pcm_stereo_in_req,
-};
-
-static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = {
-	.prio = PXAD_PRIO_LOWEST,
-	.drcmr = 12,
 };
 
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
 	.addr		= __PREG(PCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.chan_name	= "pcm_pcm_stereo_out",
 	.maxburst	= 32,
-	.filter_data	= &pxa2xx_ac97_pcm_stereo_out_req,
 };
 
-static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = {
-	.prio = PXAD_PRIO_LOWEST,
-	.drcmr = 10,
-};
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = {
 	.addr		= __PREG(MODR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
+	.chan_name	= "pcm_aux_mono_out",
 	.maxburst	= 16,
-	.filter_data	= &pxa2xx_ac97_pcm_aux_mono_out_req,
 };
 
-static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = {
-	.prio = PXAD_PRIO_LOWEST,
-	.drcmr = 9,
-};
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = {
 	.addr		= __PREG(MODR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
+	.chan_name	= "pcm_aux_mono_in",
 	.maxburst	= 16,
-	.filter_data	= &pxa2xx_ac97_pcm_aux_mono_in_req,
 };
 
-static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = {
-	.prio = PXAD_PRIO_LOWEST,
-	.drcmr = 8,
-};
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = {
 	.addr		= __PREG(MCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
+	.chan_name	= "pcm_aux_mic_mono",
 	.maxburst	= 16,
-	.filter_data	= &pxa2xx_ac97_pcm_aux_mic_mono_req,
 };
 
 static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream,
@@ -236,7 +214,21 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = {
 
 static const struct snd_soc_component_driver pxa_ac97_component = {
 	.name		= "pxa-ac97",
+	.ops		= &pxa2xx_pcm_ops,
+	.pcm_new	= pxa2xx_soc_pcm_new,
+	.pcm_free	= pxa2xx_pcm_free_dma_buffers,
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id pxa2xx_ac97_dt_ids[] = {
+	{ .compatible = "marvell,pxa250-ac97", },
+	{ .compatible = "marvell,pxa270-ac97", },
+	{ .compatible = "marvell,pxa300-ac97", },
+	{ }
 };
+MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids);
+
+#endif
 
 static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 {
@@ -296,6 +288,7 @@ static struct platform_driver pxa2xx_ac97_driver = {
 #ifdef CONFIG_PM_SLEEP
 		.pm	= &pxa2xx_ac97_pm_ops,
 #endif
+		.of_match_table = of_match_ptr(pxa2xx_ac97_dt_ids),
 	},
 };
 
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 3fb60baf6eab..42820121e5b9 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -82,20 +82,18 @@ static struct pxa_i2s_port pxa_i2s;
 static struct clk *clk_i2s;
 static int clk_ena = 0;
 
-static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3;
 static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = {
 	.addr		= __PREG(SADR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.chan_name	= "tx",
 	.maxburst	= 32,
-	.filter_data	= &pxa2xx_i2s_pcm_stereo_out_req,
 };
 
-static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2;
 static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = {
 	.addr		= __PREG(SADR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.chan_name	= "rx",
 	.maxburst	= 32,
-	.filter_data	= &pxa2xx_i2s_pcm_stereo_in_req,
 };
 
 static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
@@ -366,6 +364,9 @@ static struct snd_soc_dai_driver pxa_i2s_dai = {
 
 static const struct snd_soc_component_driver pxa_i2s_component = {
 	.name		= "pxa-i2s",
+	.ops		= &pxa2xx_pcm_ops,
+	.pcm_new	= pxa2xx_soc_pcm_new,
+	.pcm_free	= pxa2xx_pcm_free_dma_buffers,
 };
 
 static int pxa2xx_i2s_drv_probe(struct platform_device *pdev)
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 8b6a70e94c01..72eaaef1b426 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -20,70 +20,6 @@
 #include <sound/pxa2xx-lib.h>
 #include <sound/dmaengine_pcm.h>
 
-#include "../../arm/pxa2xx-pcm.h"
-
-static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_dmaengine_dai_dma_data *dma;
-
-	dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
-	/* return if this is a bufferless transfer e.g.
-	 * codec <--> BT codec or GSM modem -- lg FIXME */
-	if (!dma)
-		return 0;
-
-	return __pxa2xx_pcm_hw_params(substream, params);
-}
-
-static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
-{
-	__pxa2xx_pcm_hw_free(substream);
-
-	return 0;
-}
-
-static const struct snd_pcm_ops pxa2xx_pcm_ops = {
-	.open		= __pxa2xx_pcm_open,
-	.close		= __pxa2xx_pcm_close,
-	.ioctl		= snd_pcm_lib_ioctl,
-	.hw_params	= pxa2xx_pcm_hw_params,
-	.hw_free	= pxa2xx_pcm_hw_free,
-	.prepare	= __pxa2xx_pcm_prepare,
-	.trigger	= pxa2xx_pcm_trigger,
-	.pointer	= pxa2xx_pcm_pointer,
-	.mmap		= pxa2xx_pcm_mmap,
-};
-
-static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_card *card = rtd->card->snd_card;
-	struct snd_pcm *pcm = rtd->pcm;
-	int ret;
-
-	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
-	if (ret)
-		return ret;
-
-	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
-		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_PLAYBACK);
-		if (ret)
-			goto out;
-	}
-
-	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
-		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_CAPTURE);
-		if (ret)
-			goto out;
-	}
- out:
-	return ret;
-}
-
 static const struct snd_soc_component_driver pxa2xx_soc_platform = {
 	.ops		= &pxa2xx_pcm_ops,
 	.pcm_new	= pxa2xx_soc_pcm_new,
@@ -96,18 +32,9 @@ static int pxa2xx_soc_platform_probe(struct platform_device *pdev)
 					       NULL, 0);
 }
 
-#ifdef CONFIG_OF
-static const struct of_device_id snd_soc_pxa_audio_match[] = {
-	{ .compatible   = "mrvl,pxa-pcm-audio" },
-	{ }
-};
-MODULE_DEVICE_TABLE(of, snd_soc_pxa_audio_match);
-#endif
-
 static struct platform_driver pxa_pcm_driver = {
 	.driver = {
 		.name = "pxa-pcm-audio",
-		.of_match_table = of_match_ptr(snd_soc_pxa_audio_match),
 	},
 
 	.probe = pxa2xx_soc_platform_probe,
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index ba468e560dd2..230eee450f45 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -83,11 +83,9 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	unsigned int pll_out = 0;
 	unsigned int wm9713_div = 0;
 	int ret = 0;
 	int rate = params_rate(params);
-	int width = snd_pcm_format_physical_width(params_format(params));
 
 	/* Only support ratios that we can generate neatly from the AC97
 	 * based master clock - in particular, this excludes 44.1kHz.
@@ -109,17 +107,10 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	/* Add 1 to the width for the leading clock cycle */
-	pll_out = rate * (width + 1) * 8;
-
 	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out);
-	if (ret < 0)
-		return ret;
-
 	if (clk_pout)
 		ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV,
 					     WM9713_PCMDIV(wm9713_div));
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 87838fa27997..2a4c912d1e48 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -41,6 +41,9 @@ config SND_SOC_APQ8016_SBC
           APQ8016 SOC-based systems.
           Say Y if you want to use audio devices on MI2S.
 
+config SND_SOC_QCOM_COMMON
+	tristate
+
 config SND_SOC_QDSP6_COMMON
 	tristate
 
@@ -86,7 +89,18 @@ config SND_SOC_MSM8996
 	tristate "SoC Machine driver for MSM8996 and APQ8096 boards"
 	depends on QCOM_APR
 	select SND_SOC_QDSP6
+	select SND_SOC_QCOM_COMMON
 	help
           Support for Qualcomm Technologies LPASS audio block in
           APQ8096 SoC-based systems.
           Say Y if you want to use audio device on this SoCs
+
+config SND_SOC_SDM845
+	tristate "SoC Machine driver for SDM845 boards"
+	depends on QCOM_APR
+	select SND_SOC_QDSP6
+	select SND_SOC_QCOM_COMMON
+	help
+	  To add support for audio on Qualcomm Technologies Inc.
+	  SDM845 SoC-based systems.
+	  Say Y if you want to use audio device on this SoCs.
diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile
index 206945bb9ba1..41b2c7a23a4d 100644
--- a/sound/soc/qcom/Makefile
+++ b/sound/soc/qcom/Makefile
@@ -14,10 +14,14 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o
 snd-soc-storm-objs := storm.o
 snd-soc-apq8016-sbc-objs := apq8016_sbc.o
 snd-soc-apq8096-objs := apq8096.o
+snd-soc-sdm845-objs := sdm845.o
+snd-soc-qcom-common-objs := common.o
 
 obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o
 obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o
 obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o
+obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o
+obj-$(CONFIG_SND_SOC_QCOM_COMMON) += snd-soc-qcom-common.o
 
 #DSP lib
 obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 561cd429e6f2..1543e85629f8 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -1,14 +1,13 @@
 // SPDX-License-Identifier: GPL-2.0
 // Copyright (c) 2018, Linaro Limited
 
-#include <linux/soc/qcom/apr.h>
 #include <linux/module.h>
-#include <linux/component.h>
 #include <linux/platform_device.h>
 #include <linux/of_device.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 #include <sound/pcm.h>
+#include "common.h"
 
 static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 				      struct snd_pcm_hw_params *params)
@@ -24,211 +23,57 @@ static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 	return 0;
 }
 
-static int apq8096_sbc_parse_of(struct snd_soc_card *card)
+static void apq8096_add_be_ops(struct snd_soc_card *card)
 {
-	struct device_node *np;
-	struct device_node *codec = NULL;
-	struct device_node *platform = NULL;
-	struct device_node *cpu = NULL;
-	struct device *dev = card->dev;
-	struct snd_soc_dai_link *link;
-	int ret, num_links;
-
-	ret = snd_soc_of_parse_card_name(card, "qcom,model");
-	if (ret) {
-		dev_err(dev, "Error parsing card name: %d\n", ret);
-		return ret;
-	}
-
-	/* DAPM routes */
-	if (of_property_read_bool(dev->of_node, "qcom,audio-routing")) {
-		ret = snd_soc_of_parse_audio_routing(card,
-					"qcom,audio-routing");
-		if (ret)
-			return ret;
-	}
-
-	/* Populate links */
-	num_links = of_get_child_count(dev->of_node);
+	struct snd_soc_dai_link *link = card->dai_link;
+	int i, num_links = card->num_links;
 
-	/* Allocate the DAI link array */
-	card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL);
-	if (!card->dai_link)
-		return -ENOMEM;
-
-	card->num_links	= num_links;
-	link = card->dai_link;
-
-	for_each_child_of_node(dev->of_node, np) {
-		cpu = of_get_child_by_name(np, "cpu");
-		if (!cpu) {
-			dev_err(dev, "Can't find cpu DT node\n");
-			ret = -EINVAL;
-			goto err;
-		}
-
-		link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0);
-		if (!link->cpu_of_node) {
-			dev_err(card->dev, "error getting cpu phandle\n");
-			ret = -EINVAL;
-			goto err;
-		}
-
-		ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
-		if (ret) {
-			dev_err(card->dev, "error getting cpu dai name\n");
-			goto err;
-		}
-
-		platform = of_get_child_by_name(np, "platform");
-		codec = of_get_child_by_name(np, "codec");
-		if (codec && platform) {
-			link->platform_of_node = of_parse_phandle(platform,
-								  "sound-dai",
-								   0);
-			if (!link->platform_of_node) {
-				dev_err(card->dev, "platform dai not found\n");
-				ret = -EINVAL;
-				goto err;
-			}
-
-			ret = snd_soc_of_get_dai_link_codecs(dev, codec, link);
-			if (ret < 0) {
-				dev_err(card->dev, "codec dai not found\n");
-				goto err;
-			}
-			link->no_pcm = 1;
-			link->ignore_pmdown_time = 1;
+	for (i = 0; i < num_links; i++) {
+		if (link->no_pcm == 1)
 			link->be_hw_params_fixup = apq8096_be_hw_params_fixup;
-		} else {
-			link->platform_of_node = link->cpu_of_node;
-			link->codec_dai_name = "snd-soc-dummy-dai";
-			link->codec_name = "snd-soc-dummy";
-			link->dynamic = 1;
-		}
-
-		link->ignore_suspend = 1;
-		ret = of_property_read_string(np, "link-name", &link->name);
-		if (ret) {
-			dev_err(card->dev, "error getting codec dai_link name\n");
-			goto err;
-		}
-
-		link->dpcm_playback = 1;
-		link->dpcm_capture = 1;
-		link->stream_name = link->name;
 		link++;
 	}
-
-	return 0;
-err:
-	of_node_put(cpu);
-	of_node_put(codec);
-	of_node_put(platform);
-	kfree(card->dai_link);
-	return ret;
 }
 
-static int apq8096_bind(struct device *dev)
+static int apq8096_platform_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card;
+	struct device *dev = &pdev->dev;
 	int ret;
 
 	card = kzalloc(sizeof(*card), GFP_KERNEL);
 	if (!card)
 		return -ENOMEM;
 
-	component_bind_all(dev, card);
 	card->dev = dev;
-	ret = apq8096_sbc_parse_of(card);
+	dev_set_drvdata(dev, card);
+	ret = qcom_snd_parse_of(card);
 	if (ret) {
 		dev_err(dev, "Error parsing OF data\n");
 		goto err;
 	}
 
+	apq8096_add_be_ops(card);
 	ret = snd_soc_register_card(card);
 	if (ret)
-		goto err;
+		goto err_card_register;
 
 	return 0;
 
+err_card_register:
+	kfree(card->dai_link);
 err:
-	component_unbind_all(dev, card);
 	kfree(card);
 	return ret;
 }
 
-static void apq8096_unbind(struct device *dev)
+static int apq8096_platform_remove(struct platform_device *pdev)
 {
-	struct snd_soc_card *card = dev_get_drvdata(dev);
+	struct snd_soc_card *card = dev_get_drvdata(&pdev->dev);
 
 	snd_soc_unregister_card(card);
-	component_unbind_all(dev, card);
 	kfree(card->dai_link);
 	kfree(card);
-}
-
-static const struct component_master_ops apq8096_ops = {
-	.bind = apq8096_bind,
-	.unbind = apq8096_unbind,
-};
-
-static int apq8016_compare_of(struct device *dev, void *data)
-{
-	return dev->of_node == data;
-}
-
-static void apq8016_release_of(struct device *dev, void *data)
-{
-	of_node_put(data);
-}
-
-static int add_audio_components(struct device *dev,
-				struct component_match **matchptr)
-{
-	struct device_node *np, *platform, *cpu, *node, *dai_node;
-
-	node = dev->of_node;
-
-	for_each_child_of_node(node, np) {
-		cpu = of_get_child_by_name(np, "cpu");
-		if (cpu) {
-			dai_node = of_parse_phandle(cpu, "sound-dai", 0);
-			of_node_get(dai_node);
-			component_match_add_release(dev, matchptr,
-						    apq8016_release_of,
-						    apq8016_compare_of,
-						    dai_node);
-		}
-
-		platform = of_get_child_by_name(np, "platform");
-		if (platform) {
-			dai_node = of_parse_phandle(platform, "sound-dai", 0);
-			component_match_add_release(dev, matchptr,
-						    apq8016_release_of,
-						    apq8016_compare_of,
-						    dai_node);
-		}
-	}
-
-	return 0;
-}
-
-static int apq8096_platform_probe(struct platform_device *pdev)
-{
-	struct component_match *match = NULL;
-	int ret;
-
-	ret = add_audio_components(&pdev->dev, &match);
-	if (ret)
-		return ret;
-
-	return component_master_add_with_match(&pdev->dev, &apq8096_ops, match);
-}
-
-static int apq8096_platform_remove(struct platform_device *pdev)
-{
-	component_master_del(&pdev->dev, &apq8096_ops);
 
 	return 0;
 }
@@ -245,7 +90,6 @@ static struct platform_driver msm_snd_apq8096_driver = {
 	.remove = apq8096_platform_remove,
 	.driver = {
 		.name = "msm-snd-apq8096",
-		.owner = THIS_MODULE,
 		.of_match_table = msm_snd_apq8096_dt_match,
 	},
 };
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
new file mode 100644
index 000000000000..eb1b9da05dd4
--- /dev/null
+++ b/sound/soc/qcom/common.c
@@ -0,0 +1,112 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2018, Linaro Limited.
+// Copyright (c) 2018, The Linux Foundation. All rights reserved.
+
+#include <linux/module.h>
+#include "common.h"
+
+int qcom_snd_parse_of(struct snd_soc_card *card)
+{
+	struct device_node *np;
+	struct device_node *codec = NULL;
+	struct device_node *platform = NULL;
+	struct device_node *cpu = NULL;
+	struct device *dev = card->dev;
+	struct snd_soc_dai_link *link;
+	int ret, num_links;
+
+	ret = snd_soc_of_parse_card_name(card, "model");
+	if (ret) {
+		dev_err(dev, "Error parsing card name: %d\n", ret);
+		return ret;
+	}
+
+	/* DAPM routes */
+	if (of_property_read_bool(dev->of_node, "audio-routing")) {
+		ret = snd_soc_of_parse_audio_routing(card,
+				"audio-routing");
+		if (ret)
+			return ret;
+	}
+
+	/* Populate links */
+	num_links = of_get_child_count(dev->of_node);
+
+	/* Allocate the DAI link array */
+	card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL);
+	if (!card->dai_link)
+		return -ENOMEM;
+
+	card->num_links = num_links;
+	link = card->dai_link;
+	for_each_child_of_node(dev->of_node, np) {
+		cpu = of_get_child_by_name(np, "cpu");
+		if (!cpu) {
+			dev_err(dev, "Can't find cpu DT node\n");
+			ret = -EINVAL;
+			goto err;
+		}
+
+		link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0);
+		if (!link->cpu_of_node) {
+			dev_err(card->dev, "error getting cpu phandle\n");
+			ret = -EINVAL;
+			goto err;
+		}
+
+		ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
+		if (ret) {
+			dev_err(card->dev, "error getting cpu dai name\n");
+			goto err;
+		}
+
+		platform = of_get_child_by_name(np, "platform");
+		codec = of_get_child_by_name(np, "codec");
+		if (codec && platform) {
+			link->platform_of_node = of_parse_phandle(platform,
+					"sound-dai",
+					0);
+			if (!link->platform_of_node) {
+				dev_err(card->dev, "platform dai not found\n");
+				ret = -EINVAL;
+				goto err;
+			}
+
+			ret = snd_soc_of_get_dai_link_codecs(dev, codec, link);
+			if (ret < 0) {
+				dev_err(card->dev, "codec dai not found\n");
+				goto err;
+			}
+			link->no_pcm = 1;
+			link->ignore_pmdown_time = 1;
+		} else {
+			link->platform_of_node = link->cpu_of_node;
+			link->codec_dai_name = "snd-soc-dummy-dai";
+			link->codec_name = "snd-soc-dummy";
+			link->dynamic = 1;
+		}
+
+		link->ignore_suspend = 1;
+		ret = of_property_read_string(np, "link-name", &link->name);
+		if (ret) {
+			dev_err(card->dev, "error getting codec dai_link name\n");
+			goto err;
+		}
+
+		link->dpcm_playback = 1;
+		link->dpcm_capture = 1;
+		link->stream_name = link->name;
+		link++;
+	}
+
+	return 0;
+err:
+	of_node_put(cpu);
+	of_node_put(codec);
+	of_node_put(platform);
+	kfree(card->dai_link);
+	return ret;
+}
+EXPORT_SYMBOL(qcom_snd_parse_of);
+
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/common.h b/sound/soc/qcom/common.h
new file mode 100644
index 000000000000..f05c05b12bd7
--- /dev/null
+++ b/sound/soc/qcom/common.h
@@ -0,0 +1,11 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+// Copyright (c) 2018, The Linux Foundation. All rights reserved.
+
+#ifndef __QCOM_SND_COMMON_H__
+#define __QCOM_SND_COMMON_H__
+
+#include <sound/soc.h>
+
+int qcom_snd_parse_of(struct snd_soc_card *card);
+
+#endif
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index 31fe78aa207f..d07271ea4c45 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -458,7 +458,7 @@ static irqreturn_t lpass_dma_interrupt_handler(
 			return IRQ_NONE;
 		}
 		dev_warn(soc_runtime->dev, "xrun warning\n");
-		snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+		snd_pcm_stop_xrun(substream);
 		ret = IRQ_HANDLED;
 	}
 
diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c
index 9983c665a941..932c3ebfd252 100644
--- a/sound/soc/qcom/qdsp6/q6adm.c
+++ b/sound/soc/qcom/qdsp6/q6adm.c
@@ -64,7 +64,6 @@ struct q6adm {
 	struct aprv2_ibasic_rsp_result_t result;
 	struct mutex lock;
 	wait_queue_head_t matrix_map_wait;
-	struct platform_device *pdev_routing;
 };
 
 struct q6adm_cmd_device_open_v5 {
@@ -588,7 +587,6 @@ EXPORT_SYMBOL_GPL(q6adm_close);
 static int q6adm_probe(struct apr_device *adev)
 {
 	struct device *dev = &adev->dev;
-	struct device_node *dais_np;
 	struct q6adm *adm;
 
 	adm = devm_kzalloc(&adev->dev, sizeof(*adm), GFP_KERNEL);
@@ -605,22 +603,12 @@ static int q6adm_probe(struct apr_device *adev)
 	INIT_LIST_HEAD(&adm->copps_list);
 	spin_lock_init(&adm->copps_list_lock);
 
-	dais_np = of_get_child_by_name(dev->of_node, "routing");
-	if (dais_np) {
-		adm->pdev_routing = of_platform_device_create(dais_np,
-							   "q6routing", dev);
-		of_node_put(dais_np);
-	}
-
-	return 0;
+	return of_platform_populate(dev->of_node, NULL, NULL, dev);
 }
 
 static int q6adm_remove(struct apr_device *adev)
 {
-	struct q6adm *adm = dev_get_drvdata(&adev->dev);
-
-	if (adm->pdev_routing)
-		of_platform_device_destroy(&adm->pdev_routing->dev, NULL);
+	of_platform_depopulate(&adev->dev);
 
 	return 0;
 }
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 5002dd05bf27..60ff4a2d3577 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -4,7 +4,6 @@
 
 #include <linux/err.h>
 #include <linux/init.h>
-#include <linux/component.h>
 #include <linux/module.h>
 #include <linux/device.h>
 #include <linux/platform_device.h>
@@ -81,7 +80,6 @@ static int q6slim_hw_params(struct snd_pcm_substream *substream,
 	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
 	struct q6afe_slim_cfg *slim = &dai_data->port_config[dai->id].slim;
 
-	slim->num_channels = params_channels(params);
 	slim->sample_rate = params_rate(params);
 
 	switch (params_format(params)) {
@@ -315,6 +313,9 @@ static void q6afe_dai_shutdown(struct snd_pcm_substream *substream,
 	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
 	int rc;
 
+	if (!dai_data->is_port_started[dai->id])
+		return;
+
 	rc = q6afe_port_stop(dai_data->port[dai->id]);
 	if (rc < 0)
 		dev_err(dai->dev, "fail to close AFE port (%d)\n", rc);
@@ -382,23 +383,31 @@ static int q6slim_set_channel_map(struct snd_soc_dai *dai,
 	struct q6afe_port_config *pcfg = &dai_data->port_config[dai->id];
 	int i;
 
-	if (!rx_slot) {
-		pr_err("%s: rx slot not found\n", __func__);
-		return -EINVAL;
-	}
+	if (dai->id & 0x1) {
+		/* TX */
+		if (!tx_slot) {
+			pr_err("%s: tx slot not found\n", __func__);
+			return -EINVAL;
+		}
 
-	for (i = 0; i < rx_num; i++) {
-		pcfg->slim.ch_mapping[i] =   rx_slot[i];
-		pr_debug("%s: find number of channels[%d] ch[%d]\n",
-		       __func__, i, rx_slot[i]);
-	}
+		for (i = 0; i < tx_num; i++)
+			pcfg->slim.ch_mapping[i] = tx_slot[i];
+
+		pcfg->slim.num_channels = tx_num;
+
+
+	} else {
+		if (!rx_slot) {
+			pr_err("%s: rx slot not found\n", __func__);
+			return -EINVAL;
+		}
 
-	pcfg->slim.num_channels = rx_num;
+		for (i = 0; i < rx_num; i++)
+			pcfg->slim.ch_mapping[i] =   rx_slot[i];
 
-	pr_debug("%s: SLIMBUS_%d_RX cnt[%d] ch[%d %d]\n", __func__,
-		(dai->id - SLIMBUS_0_RX) / 2, rx_num,
-		pcfg->slim.ch_mapping[0],
-		pcfg->slim.ch_mapping[1]);
+		pcfg->slim.num_channels = rx_num;
+
+	}
 
 	return 0;
 }
@@ -443,6 +452,14 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
 	{"Slimbus5 Playback", NULL, "SLIMBUS_5_RX"},
 	{"Slimbus6 Playback", NULL, "SLIMBUS_6_RX"},
 
+	{"SLIMBUS_0_TX", NULL, "Slimbus Capture"},
+	{"SLIMBUS_1_TX", NULL, "Slimbus1 Capture"},
+	{"SLIMBUS_2_TX", NULL, "Slimbus2 Capture"},
+	{"SLIMBUS_3_TX", NULL, "Slimbus3 Capture"},
+	{"SLIMBUS_4_TX", NULL, "Slimbus4 Capture"},
+	{"SLIMBUS_5_TX", NULL, "Slimbus5 Capture"},
+	{"SLIMBUS_6_TX", NULL, "Slimbus6 Capture"},
+
 	{"Primary MI2S Playback", NULL, "PRI_MI2S_RX"},
 	{"Secondary MI2S Playback", NULL, "SEC_MI2S_RX"},
 	{"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"},
@@ -637,6 +654,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 			.rate_max = 192000,
 		},
 	}, {
+		.name = "SLIMBUS_0_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_0_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
+	}, {
 		.playback = {
 			.stream_name = "Slimbus1 Playback",
 			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
@@ -655,6 +690,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
 	}, {
+		.name = "SLIMBUS_1_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_1_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus1 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
+	}, {
 		.playback = {
 			.stream_name = "Slimbus2 Playback",
 			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
@@ -672,6 +725,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_2_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_2_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_2_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus2 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus3 Playback",
@@ -690,6 +762,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_3_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_3_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_3_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus3 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus4 Playback",
@@ -708,6 +799,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_4_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_4_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_4_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus4 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus5 Playback",
@@ -726,6 +836,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_5_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_5_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_5_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus5 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus6 Playback",
@@ -744,6 +873,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_6_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_6_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_6_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus6 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Primary MI2S Playback",
@@ -972,6 +1120,13 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
 	SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback", 0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_RX", "Slimbus5 Playback", 0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_RX", "Slimbus6 Playback", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_0_TX", "Slimbus Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_1_TX", "Slimbus1 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_2_TX", "Slimbus2 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_3_TX", "Slimbus3 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_4_TX", "Slimbus4 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_5_TX", "Slimbus5 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_6_TX", "Slimbus6 Capture", 0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_RX", "Quaternary MI2S Playback",
 						0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUAT_MI2S_TX", "Quaternary MI2S Capture",
@@ -1180,7 +1335,7 @@ static void of_q6afe_parse_dai_data(struct device *dev,
 		int id, i, num_lines;
 
 		ret = of_property_read_u32(node, "reg", &id);
-		if (ret || id > AFE_PORT_MAX) {
+		if (ret || id < 0 || id >= AFE_PORT_MAX) {
 			dev_err(dev, "valid dai id not found:%d\n", ret);
 			continue;
 		}
@@ -1249,11 +1404,12 @@ static void of_q6afe_parse_dai_data(struct device *dev,
 	}
 }
 
-static int q6afe_dai_bind(struct device *dev, struct device *master, void *data)
+static int q6afe_dai_dev_probe(struct platform_device *pdev)
 {
 	struct q6afe_dai_data *dai_data;
+	struct device *dev = &pdev->dev;
 
-	dai_data = kzalloc(sizeof(*dai_data), GFP_KERNEL);
+	dai_data = devm_kzalloc(dev, sizeof(*dai_data), GFP_KERNEL);
 	if (!dai_data)
 		return -ENOMEM;
 
@@ -1261,41 +1417,22 @@ static int q6afe_dai_bind(struct device *dev, struct device *master, void *data)
 
 	of_q6afe_parse_dai_data(dev, dai_data);
 
-	return snd_soc_register_component(dev, &q6afe_dai_component,
+	return devm_snd_soc_register_component(dev, &q6afe_dai_component,
 					  q6afe_dais, ARRAY_SIZE(q6afe_dais));
 }
 
-static void q6afe_dai_unbind(struct device *dev, struct device *master,
-			     void *data)
-{
-	struct q6afe_dai_data *dai_data = dev_get_drvdata(dev);
-
-	snd_soc_unregister_component(dev);
-	kfree(dai_data);
-}
-
-static const struct component_ops q6afe_dai_comp_ops = {
-	.bind   = q6afe_dai_bind,
-	.unbind = q6afe_dai_unbind,
+static const struct of_device_id q6afe_dai_device_id[] = {
+	{ .compatible = "qcom,q6afe-dais" },
+	{},
 };
-
-static int q6afe_dai_dev_probe(struct platform_device *pdev)
-{
-	return component_add(&pdev->dev, &q6afe_dai_comp_ops);
-}
-
-static int q6afe_dai_dev_remove(struct platform_device *pdev)
-{
-	component_del(&pdev->dev, &q6afe_dai_comp_ops);
-	return 0;
-}
+MODULE_DEVICE_TABLE(of, q6afe_dai_device_id);
 
 static struct platform_driver q6afe_dai_platform_driver = {
 	.driver = {
 		.name = "q6afe-dai",
+		.of_match_table = of_match_ptr(q6afe_dai_device_id),
 	},
 	.probe = q6afe_dai_dev_probe,
-	.remove = q6afe_dai_dev_remove,
 };
 module_platform_driver(q6afe_dai_platform_driver);
 
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 01f43218984b..000775b4bba8 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -316,7 +316,6 @@ struct q6afe {
 	struct mutex lock;
 	struct list_head port_list;
 	spinlock_t port_list_lock;
-	struct platform_device *pdev_dais;
 };
 
 struct afe_port_cmd_device_start {
@@ -515,6 +514,20 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = {
 				SLIMBUS_5_RX, 1, 1},
 	[SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX,
 				SLIMBUS_6_RX, 1, 1},
+	[SLIMBUS_0_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX,
+				SLIMBUS_0_TX, 0, 1},
+	[SLIMBUS_1_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX,
+				SLIMBUS_1_TX, 0, 1},
+	[SLIMBUS_2_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX,
+				SLIMBUS_2_TX, 0, 1},
+	[SLIMBUS_3_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX,
+				SLIMBUS_3_TX, 0, 1},
+	[SLIMBUS_4_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX,
+				SLIMBUS_4_TX, 0, 1},
+	[SLIMBUS_5_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX,
+				SLIMBUS_5_TX, 0, 1},
+	[SLIMBUS_6_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX,
+				SLIMBUS_6_TX, 0, 1},
 	[PRIMARY_MI2S_RX] = { AFE_PORT_ID_PRIMARY_MI2S_RX,
 				PRIMARY_MI2S_RX, 1, 1},
 	[PRIMARY_MI2S_TX] = { AFE_PORT_ID_PRIMARY_MI2S_TX,
@@ -777,7 +790,7 @@ static int q6afe_callback(struct apr_device *adev, struct apr_resp_pkt *data)
  */
 int q6afe_get_port_id(int index)
 {
-	if (index < 0 || index > AFE_PORT_MAX)
+	if (index < 0 || index >= AFE_PORT_MAX)
 		return -EINVAL;
 
 	return port_maps[index].port_id;
@@ -1014,7 +1027,7 @@ int q6afe_port_stop(struct q6afe_port *port)
 
 	port_id = port->id;
 	index = port->token;
-	if (index < 0 || index > AFE_PORT_MAX) {
+	if (index < 0 || index >= AFE_PORT_MAX) {
 		dev_err(afe->dev, "AFE port index[%d] invalid!\n", index);
 		return -EINVAL;
 	}
@@ -1355,7 +1368,7 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
 	unsigned long flags;
 	int cfg_type;
 
-	if (id < 0 || id > AFE_PORT_MAX) {
+	if (id < 0 || id >= AFE_PORT_MAX) {
 		dev_err(dev, "AFE port token[%d] invalid!\n", id);
 		return ERR_PTR(-EINVAL);
 	}
@@ -1373,6 +1386,13 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
 	case AFE_PORT_ID_MULTICHAN_HDMI_RX:
 		cfg_type = AFE_PARAM_ID_HDMI_CONFIG;
 		break;
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX:
 	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX:
 	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX:
 	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX:
@@ -1438,7 +1458,6 @@ static int q6afe_probe(struct apr_device *adev)
 {
 	struct q6afe *afe;
 	struct device *dev = &adev->dev;
-	struct device_node *dais_np;
 
 	afe = devm_kzalloc(dev, sizeof(*afe), GFP_KERNEL);
 	if (!afe)
@@ -1453,22 +1472,12 @@ static int q6afe_probe(struct apr_device *adev)
 
 	dev_set_drvdata(dev, afe);
 
-	dais_np = of_get_child_by_name(dev->of_node, "dais");
-	if (dais_np) {
-		afe->pdev_dais = of_platform_device_create(dais_np,
-							   "q6afe-dai", dev);
-		of_node_put(dais_np);
-	}
-
-	return 0;
+	return of_platform_populate(dev->of_node, NULL, NULL, dev);
 }
 
 static int q6afe_remove(struct apr_device *adev)
 {
-	struct q6afe *afe = dev_get_drvdata(&adev->dev);
-
-	if (afe->pdev_dais)
-		of_platform_device_destroy(&afe->pdev_dais->dev, NULL);
+	of_platform_depopulate(&adev->dev);
 
 	return 0;
 }
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 349c6a883c63..9db9a2944ef2 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -7,7 +7,6 @@
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
-#include <linux/component.h>
 #include <sound/soc.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
@@ -390,7 +389,9 @@ static int q6asm_dai_close(struct snd_pcm_substream *substream)
 	struct q6asm_dai_rtd *prtd = runtime->private_data;
 
 	if (prtd->audio_client) {
-		q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+		if (prtd->state)
+			q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+
 		q6asm_unmap_memory_regions(substream->stream,
 					   prtd->audio_client);
 		q6asm_audio_client_free(prtd->audio_client);
@@ -561,14 +562,15 @@ static struct snd_soc_dai_driver q6asm_fe_dais[] = {
 	Q6ASM_FEDAI_DRIVER(8),
 };
 
-static int q6asm_dai_bind(struct device *dev, struct device *master, void *data)
+static int q6asm_dai_probe(struct platform_device *pdev)
 {
+	struct device *dev = &pdev->dev;
 	struct device_node *node = dev->of_node;
 	struct of_phandle_args args;
 	struct q6asm_dai_data *pdata;
 	int rc;
 
-	pdata = kzalloc(sizeof(struct q6asm_dai_data), GFP_KERNEL);
+	pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL);
 	if (!pdata)
 		return -ENOMEM;
 
@@ -580,43 +582,23 @@ static int q6asm_dai_bind(struct device *dev, struct device *master, void *data)
 
 	dev_set_drvdata(dev, pdata);
 
-	return snd_soc_register_component(dev, &q6asm_fe_dai_component,
+	return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component,
 					q6asm_fe_dais,
 					ARRAY_SIZE(q6asm_fe_dais));
 }
-static void q6asm_dai_unbind(struct device *dev, struct device *master,
-			     void *data)
-{
-	struct q6asm_dai_data *pdata = dev_get_drvdata(dev);
-
-	snd_soc_unregister_component(dev);
-
-	kfree(pdata);
-
-}
 
-static const struct component_ops q6asm_dai_comp_ops = {
-	.bind   = q6asm_dai_bind,
-	.unbind = q6asm_dai_unbind,
+static const struct of_device_id q6asm_dai_device_id[] = {
+	{ .compatible = "qcom,q6asm-dais" },
+	{},
 };
-
-static int q6asm_dai_probe(struct platform_device *pdev)
-{
-	return component_add(&pdev->dev, &q6asm_dai_comp_ops);
-}
-
-static int q6asm_dai_dev_remove(struct platform_device *pdev)
-{
-	component_del(&pdev->dev, &q6asm_dai_comp_ops);
-	return 0;
-}
+MODULE_DEVICE_TABLE(of, q6asm_dai_device_id);
 
 static struct platform_driver q6asm_dai_platform_driver = {
 	.driver = {
 		.name = "q6asm-dai",
+		.of_match_table = of_match_ptr(q6asm_dai_device_id),
 	},
 	.probe = q6asm_dai_probe,
-	.remove = q6asm_dai_dev_remove,
 };
 module_platform_driver(q6asm_dai_platform_driver);
 
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 530852385cad..2b2c7233bb5f 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -174,10 +174,8 @@ struct q6asm {
 	struct device *dev;
 	struct q6core_svc_api_info ainfo;
 	wait_queue_head_t mem_wait;
-	struct platform_device *pcmdev;
 	spinlock_t slock;
 	struct audio_client *session[MAX_SESSIONS + 1];
-	struct platform_device *pdev_dais;
 };
 
 struct audio_client {
@@ -1344,7 +1342,6 @@ EXPORT_SYMBOL_GPL(q6asm_cmd_nowait);
 static int q6asm_probe(struct apr_device *adev)
 {
 	struct device *dev = &adev->dev;
-	struct device_node *dais_np;
 	struct q6asm *q6asm;
 
 	q6asm = devm_kzalloc(dev, sizeof(*q6asm), GFP_KERNEL);
@@ -1359,22 +1356,12 @@ static int q6asm_probe(struct apr_device *adev)
 	spin_lock_init(&q6asm->slock);
 	dev_set_drvdata(dev, q6asm);
 
-	dais_np = of_get_child_by_name(dev->of_node, "dais");
-	if (dais_np) {
-		q6asm->pdev_dais = of_platform_device_create(dais_np,
-							   "q6asm-dai", dev);
-		of_node_put(dais_np);
-	}
-
-	return 0;
+	return of_platform_populate(dev->of_node, NULL, NULL, dev);
 }
 
 static int q6asm_remove(struct apr_device *adev)
 {
-	struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
-
-	if (q6asm->pdev_dais)
-		of_platform_device_destroy(&q6asm->pdev_dais->dev, NULL);
+	of_platform_depopulate(&adev->dev);
 
 	return 0;
 }
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index 593f66b8622f..dc94c5c53788 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -8,7 +8,6 @@
 #include <linux/platform_device.h>
 #include <linux/of_platform.h>
 #include <linux/bitops.h>
-#include <linux/component.h>
 #include <linux/mutex.h>
 #include <linux/of_device.h>
 #include <linux/slab.h>
@@ -68,6 +67,13 @@
 	{ mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" },	\
 	{ mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" },	\
 	{ mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" },		\
+	{ mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" },		\
+	{ mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" },		\
+	{ mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" },		\
+	{ mix_name, "SLIMBUS_3_TX", "SLIMBUS_3_TX" },		\
+	{ mix_name, "SLIMBUS_4_TX", "SLIMBUS_4_TX" },		\
+	{ mix_name, "SLIMBUS_5_TX", "SLIMBUS_5_TX" },		\
+	{ mix_name, "SLIMBUS_6_TX", "SLIMBUS_6_TX" },		\
 	{ mix_name, "PRIMARY_TDM_TX_0", "PRIMARY_TDM_TX_0"},	\
 	{ mix_name, "PRIMARY_TDM_TX_1", "PRIMARY_TDM_TX_1"},	\
 	{ mix_name, "PRIMARY_TDM_TX_2", "PRIMARY_TDM_TX_2"},	\
@@ -122,6 +128,27 @@
 	SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,		\
 		id, 1, 0, msm_routing_get_audio_mixer,			\
 		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_1_TX", SLIMBUS_1_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_2_TX", SLIMBUS_2_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_3_TX", SLIMBUS_3_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_4_TX", SLIMBUS_4_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_5_TX", SLIMBUS_5_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_6_TX", SLIMBUS_6_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
 	SOC_SINGLE_EXT("PRIMARY_TDM_TX_0", PRIMARY_TDM_TX_0,		\
 		id, 1, 0, msm_routing_get_audio_mixer,			\
 		msm_routing_put_audio_mixer),				\
@@ -310,7 +337,7 @@ int q6routing_stream_open(int fedai_id, int perf_mode,
 			      session->channels, topology, perf_mode,
 			      session->bits_per_sample, 0, 0);
 
-	if (!copp) {
+	if (IS_ERR_OR_NULL(copp)) {
 		mutex_unlock(&routing_data->lock);
 		return -EINVAL;
 	}
@@ -899,7 +926,7 @@ static int routing_hw_params(struct snd_pcm_substream *substream,
 	else
 		path_type = ADM_PATH_LIVE_REC;
 
-	if (be_id > AFE_MAX_PORTS)
+	if (be_id >= AFE_MAX_PORTS)
 		return -EINVAL;
 
 	session = &data->port_data[be_id];
@@ -949,9 +976,10 @@ static const struct snd_soc_component_driver msm_soc_routing_component = {
 	.num_dapm_routes = ARRAY_SIZE(intercon),
 };
 
-static int q6routing_dai_bind(struct device *dev, struct device *master,
-			      void *data)
+static int q6pcm_routing_probe(struct platform_device *pdev)
 {
+	struct device *dev = &pdev->dev;
+
 	routing_data = kzalloc(sizeof(*routing_data), GFP_KERNEL);
 	if (!routing_data)
 		return -ENOMEM;
@@ -961,41 +989,28 @@ static int q6routing_dai_bind(struct device *dev, struct device *master,
 	mutex_init(&routing_data->lock);
 	dev_set_drvdata(dev, routing_data);
 
-	return snd_soc_register_component(dev, &msm_soc_routing_component,
+	return devm_snd_soc_register_component(dev, &msm_soc_routing_component,
 					  NULL, 0);
 }
 
-static void q6routing_dai_unbind(struct device *dev, struct device *master,
-				 void *d)
+static int q6pcm_routing_remove(struct platform_device *pdev)
 {
-	struct msm_routing_data *data = dev_get_drvdata(dev);
-
-	snd_soc_unregister_component(dev);
-
-	kfree(data);
-
+	kfree(routing_data);
 	routing_data = NULL;
-}
-
-static const struct component_ops q6routing_dai_comp_ops = {
-	.bind   = q6routing_dai_bind,
-	.unbind = q6routing_dai_unbind,
-};
 
-static int q6pcm_routing_probe(struct platform_device *pdev)
-{
-	return component_add(&pdev->dev, &q6routing_dai_comp_ops);
-}
-
-static int q6pcm_routing_remove(struct platform_device *pdev)
-{
-	component_del(&pdev->dev, &q6routing_dai_comp_ops);
 	return 0;
 }
 
+static const struct of_device_id q6pcm_routing_device_id[] = {
+	{ .compatible = "qcom,q6adm-routing" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, q6pcm_routing_device_id);
+
 static struct platform_driver q6pcm_routing_platform_driver = {
 	.driver = {
 		.name = "q6routing",
+		.of_match_table = of_match_ptr(q6pcm_routing_device_id),
 	},
 	.probe = q6pcm_routing_probe,
 	.remove = q6pcm_routing_remove,
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
new file mode 100644
index 000000000000..2a781d87ee65
--- /dev/null
+++ b/sound/soc/qcom/sdm845.c
@@ -0,0 +1,285 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * Copyright (c) 2018, The Linux Foundation. All rights reserved.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/of_device.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include "common.h"
+#include "qdsp6/q6afe.h"
+
+#define DEFAULT_SAMPLE_RATE_48K		48000
+#define DEFAULT_MCLK_RATE		24576000
+#define DEFAULT_BCLK_RATE		12288000
+
+struct sdm845_snd_data {
+	struct snd_soc_card *card;
+	uint32_t pri_mi2s_clk_count;
+	uint32_t quat_tdm_clk_count;
+};
+
+static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
+
+static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret = 0;
+	int channels, slot_width;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		slot_width = 32;
+		break;
+	default:
+		dev_err(rtd->dev, "%s: invalid param format 0x%x\n",
+				__func__, params_format(params));
+		return -EINVAL;
+	}
+
+	channels = params_channels(params);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0, 0x3,
+				8, slot_width);
+		if (ret < 0) {
+			dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n",
+					__func__, ret);
+			goto end;
+		}
+
+		ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL,
+				channels, tdm_slot_offset);
+		if (ret < 0) {
+			dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n",
+					__func__, ret);
+			goto end;
+		}
+	} else {
+		ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xf, 0,
+				8, slot_width);
+		if (ret < 0) {
+			dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n",
+					__func__, ret);
+			goto end;
+		}
+
+		ret = snd_soc_dai_set_channel_map(cpu_dai, channels,
+				tdm_slot_offset, 0, NULL);
+		if (ret < 0) {
+			dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n",
+					__func__, ret);
+			goto end;
+		}
+	}
+end:
+	return ret;
+}
+
+static int sdm845_snd_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret = 0;
+
+	switch (cpu_dai->id) {
+	case QUATERNARY_TDM_RX_0:
+	case QUATERNARY_TDM_TX_0:
+		ret = sdm845_tdm_snd_hw_params(substream, params);
+		break;
+	default:
+		pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
+		break;
+	}
+	return ret;
+}
+
+static int sdm845_snd_startup(struct snd_pcm_substream *substream)
+{
+	unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+	switch (cpu_dai->id) {
+	case PRIMARY_MI2S_RX:
+	case PRIMARY_MI2S_TX:
+		if (++(data->pri_mi2s_clk_count) == 1) {
+			snd_soc_dai_set_sysclk(cpu_dai,
+				Q6AFE_LPASS_CLK_ID_MCLK_1,
+				DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+			snd_soc_dai_set_sysclk(cpu_dai,
+				Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
+				DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+		}
+		snd_soc_dai_set_fmt(cpu_dai, fmt);
+		break;
+
+	case QUATERNARY_TDM_RX_0:
+	case QUATERNARY_TDM_TX_0:
+		if (++(data->quat_tdm_clk_count) == 1) {
+			snd_soc_dai_set_sysclk(cpu_dai,
+				Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT,
+				DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+		}
+		break;
+
+	default:
+		pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
+		break;
+	}
+	return 0;
+}
+
+static void  sdm845_snd_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+	switch (cpu_dai->id) {
+	case PRIMARY_MI2S_RX:
+	case PRIMARY_MI2S_TX:
+		if (--(data->pri_mi2s_clk_count) == 0) {
+			snd_soc_dai_set_sysclk(cpu_dai,
+				Q6AFE_LPASS_CLK_ID_MCLK_1,
+				0, SNDRV_PCM_STREAM_PLAYBACK);
+			snd_soc_dai_set_sysclk(cpu_dai,
+				Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
+				0, SNDRV_PCM_STREAM_PLAYBACK);
+		};
+		break;
+
+	case QUATERNARY_TDM_RX_0:
+	case QUATERNARY_TDM_TX_0:
+		if (--(data->quat_tdm_clk_count) == 0) {
+			snd_soc_dai_set_sysclk(cpu_dai,
+				Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT,
+				0, SNDRV_PCM_STREAM_PLAYBACK);
+		}
+		break;
+
+	default:
+		pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
+		break;
+	}
+}
+
+static struct snd_soc_ops sdm845_be_ops = {
+	.hw_params = sdm845_snd_hw_params,
+	.startup = sdm845_snd_startup,
+	.shutdown = sdm845_snd_shutdown,
+};
+
+static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+					SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+					SNDRV_PCM_HW_PARAM_CHANNELS);
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	rate->min = rate->max = DEFAULT_SAMPLE_RATE_48K;
+	channels->min = channels->max = 2;
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
+
+	return 0;
+}
+
+static void sdm845_add_be_ops(struct snd_soc_card *card)
+{
+	struct snd_soc_dai_link *link = card->dai_link;
+	int i, num_links = card->num_links;
+
+	for (i = 0; i < num_links; i++) {
+		if (link->no_pcm == 1) {
+			link->ops = &sdm845_be_ops;
+			link->be_hw_params_fixup = sdm845_be_hw_params_fixup;
+		}
+		link++;
+	}
+}
+
+static int sdm845_snd_platform_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card;
+	struct sdm845_snd_data *data;
+	struct device *dev = &pdev->dev;
+	int ret;
+
+	card = kzalloc(sizeof(*card), GFP_KERNEL);
+	if (!card)
+		return -ENOMEM;
+
+	/* Allocate the private data */
+	data = kzalloc(sizeof(*data), GFP_KERNEL);
+	if (!data) {
+		ret = -ENOMEM;
+		goto data_alloc_fail;
+	}
+
+	card->dev = dev;
+	dev_set_drvdata(dev, card);
+	ret = qcom_snd_parse_of(card);
+	if (ret) {
+		dev_err(dev, "Error parsing OF data\n");
+		goto parse_dt_fail;
+	}
+
+	data->card = card;
+	snd_soc_card_set_drvdata(card, data);
+
+	sdm845_add_be_ops(card);
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(dev, "Sound card registration failed\n");
+		goto register_card_fail;
+	}
+	return ret;
+
+register_card_fail:
+	kfree(card->dai_link);
+parse_dt_fail:
+	kfree(data);
+data_alloc_fail:
+	kfree(card);
+	return ret;
+}
+
+static int sdm845_snd_platform_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = dev_get_drvdata(&pdev->dev);
+	struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
+
+	snd_soc_unregister_card(card);
+	kfree(card->dai_link);
+	kfree(data);
+	kfree(card);
+	return 0;
+}
+
+static const struct of_device_id sdm845_snd_device_id[]  = {
+	{ .compatible = "qcom,sdm845-sndcard" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, sdm845_snd_device_id);
+
+static struct platform_driver sdm845_snd_driver = {
+	.probe = sdm845_snd_platform_probe,
+	.remove = sdm845_snd_platform_remove,
+	.driver = {
+		.name = "msm-snd-sdm845",
+		.of_match_table = sdm845_snd_device_id,
+	},
+};
+module_platform_driver(sdm845_snd_driver);
+
+MODULE_DESCRIPTION("sdm845 ASoC Machine Driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile
index 05b078e7b87f..65e814d46006 100644
--- a/sound/soc/rockchip/Makefile
+++ b/sound/soc/rockchip/Makefile
@@ -1,10 +1,11 @@
 # SPDX-License-Identifier: GPL-2.0
 # ROCKCHIP Platform Support
 snd-soc-rockchip-i2s-objs := rockchip_i2s.o
+snd-soc-rockchip-pcm-objs := rockchip_pcm.o
 snd-soc-rockchip-pdm-objs := rockchip_pdm.o
 snd-soc-rockchip-spdif-objs := rockchip_spdif.o
 
-obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o
+obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o snd-soc-rockchip-pcm.o
 obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o
 obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o
 
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 950823d69e9c..60d43d53a8f5 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -22,6 +22,7 @@
 #include <sound/dmaengine_pcm.h>
 
 #include "rockchip_i2s.h"
+#include "rockchip_pcm.h"
 
 #define DRV_NAME "rockchip-i2s"
 
@@ -674,7 +675,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
 		goto err_suspend;
 	}
 
-	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+	ret = rockchip_pcm_platform_register(&pdev->dev);
 	if (ret) {
 		dev_err(&pdev->dev, "Could not register PCM\n");
 		return ret;
diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c
new file mode 100644
index 000000000000..f77538319221
--- /dev/null
+++ b/sound/soc/rockchip/rockchip_pcm.c
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2018 Rockchip Electronics Co. Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/device.h>
+#include <linux/init.h>
+#include <linux/module.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "rockchip_pcm.h"
+
+static const struct snd_pcm_hardware snd_rockchip_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_PAUSE |
+				  SNDRV_PCM_INFO_RESUME,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 8192,
+	.periods_min		= 1,
+	.periods_max		= 52,
+	.buffer_bytes_max	= 64 * 1024,
+	.fifo_size		= 32,
+};
+
+static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = {
+	.pcm_hardware = &snd_rockchip_hardware,
+	.prealloc_buffer_size = 32 * 1024,
+};
+
+int rockchip_pcm_platform_register(struct device *dev)
+{
+	return devm_snd_dmaengine_pcm_register(dev, &rk_dmaengine_pcm_config,
+		SND_DMAENGINE_PCM_FLAG_COMPAT);
+}
+EXPORT_SYMBOL_GPL(rockchip_pcm_platform_register);
+
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/rockchip/rockchip_pcm.h b/sound/soc/rockchip/rockchip_pcm.h
new file mode 100644
index 000000000000..d6c36115c60a
--- /dev/null
+++ b/sound/soc/rockchip/rockchip_pcm.h
@@ -0,0 +1,14 @@
+/*
+ * Copyright (c) 2018 Rockchip Electronics Co. Ltd.
+ *
+ * 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 _ROCKCHIP_PCM_H
+#define _ROCKCHIP_PCM_H
+
+int rockchip_pcm_platform_register(struct device *dev);
+
+#endif
diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c
index 4db4fd56db35..881c32498808 100644
--- a/sound/soc/rockchip/rockchip_rt5645.c
+++ b/sound/soc/rockchip/rockchip_rt5645.c
@@ -181,7 +181,8 @@ static int snd_rk_mc_probe(struct platform_device *pdev)
 	if (!rk_dailink.cpu_of_node) {
 		dev_err(&pdev->dev,
 			"Property 'rockchip,i2s-controller' missing or invalid\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto put_codec_of_node;
 	}
 
 	rk_dailink.platform_of_node = rk_dailink.cpu_of_node;
@@ -190,17 +191,36 @@ static int snd_rk_mc_probe(struct platform_device *pdev)
 	if (ret) {
 		dev_err(&pdev->dev,
 			"Soc parse card name failed %d\n", ret);
-		return ret;
+		goto put_cpu_of_node;
 	}
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret) {
 		dev_err(&pdev->dev,
 			"Soc register card failed %d\n", ret);
-		return ret;
+		goto put_cpu_of_node;
 	}
 
 	return ret;
+
+put_cpu_of_node:
+	of_node_put(rk_dailink.cpu_of_node);
+	rk_dailink.cpu_of_node = NULL;
+put_codec_of_node:
+	of_node_put(rk_dailink.codec_of_node);
+	rk_dailink.codec_of_node = NULL;
+
+	return ret;
+}
+
+static int snd_rk_mc_remove(struct platform_device *pdev)
+{
+	of_node_put(rk_dailink.cpu_of_node);
+	rk_dailink.cpu_of_node = NULL;
+	of_node_put(rk_dailink.codec_of_node);
+	rk_dailink.codec_of_node = NULL;
+
+	return 0;
 }
 
 static const struct of_device_id rockchip_rt5645_of_match[] = {
@@ -212,6 +232,7 @@ MODULE_DEVICE_TABLE(of, rockchip_rt5645_of_match);
 
 static struct platform_driver snd_rk_mc_driver = {
 	.probe = snd_rk_mc_probe,
+	.remove = snd_rk_mc_remove,
 	.driver = {
 		.name = DRV_NAME,
 		.pm = &snd_soc_pm_ops,
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index f914ed45db7d..d6c62aa13041 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -710,6 +710,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream,
 	switch (params_channels(params)) {
 	case 6:
 		val |= MOD_DC2_EN;
+		/* fall through */
 	case 4:
 		val |= MOD_DC1_EN;
 		break;
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 0ae0800bf3a8..dc20f0f7080a 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -1,3 +1,4 @@
+# SPDX-License-Identifier: GPL-2.0
 menu "SoC Audio support for Renesas SoCs"
 	depends on SUPERH || ARCH_RENESAS || COMPILE_TEST
 
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 2dc3b762fdd9..922fb6aa3ed1 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -1,16 +1,14 @@
-/*
- * SH7760 ("camelot") DMABRG audio DMA unit support
- *
- * Copyright (C) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
- *  licensed under the terms outlined in the file COPYING at the root
- *  of the linux kernel sources.
- *
- * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which
- * trigger an interrupt when one half of the programmed transfer size
- * has been xmitted.
- *
- * FIXME: little-endian only for now
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// SH7760 ("camelot") DMABRG audio DMA unit support
+//
+// Copyright (C) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+//
+// The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which
+// trigger an interrupt when one half of the programmed transfer size
+// has been xmitted.
+//
+// FIXME: little-endian only for now
 
 #include <linux/module.h>
 #include <linux/gfp.h>
@@ -341,6 +339,6 @@ static struct platform_driver sh7760_pcm_driver = {
 
 module_platform_driver(sh7760_pcm_driver);
 
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 3bae06dd121f..aa7e902f0c02 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1,16 +1,12 @@
-/*
- * Fifo-attached Serial Interface (FSI) support for SH7724
- *
- * Copyright (C) 2009 Renesas Solutions Corp.
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * Based on ssi.c
- * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Fifo-attached Serial Interface (FSI) support for SH7724
+//
+// Copyright (C) 2009 Renesas Solutions Corp.
+// Kuninori Morimoto <morimoto.kuninori@renesas.com>
+//
+// Based on ssi.c
+// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
 
 #include <linux/delay.h>
 #include <linux/dma-mapping.h>
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index 624aaf569fef..c2b496398e6b 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -1,13 +1,11 @@
-/*
- * Hitachi Audio Controller (AC97) support for SH7760/SH7780
- *
- * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
- *  licensed under the terms outlined in the file COPYING at the root
- *  of the linux kernel sources.
- *
- * dont forget to set IPSEL/OMSEL register bits (in your board code) to
- * enable HAC output pins!
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Hitachi Audio Controller (AC97) support for SH7760/SH7780
+//
+// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+//
+// dont forget to set IPSEL/OMSEL register bits (in your board code) to
+// enable HAC output pins!
 
 /* BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only
  * the FIRST can be used since ASoC does not pass any information to the
@@ -343,6 +341,6 @@ static struct platform_driver hac_pcm_driver = {
 
 module_platform_driver(hac_pcm_driver);
 
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index ecb057ff9fbb..8739c9f60672 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -1,12 +1,8 @@
-/*
- * ALSA SoC driver for Migo-R
- *
- * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ALSA SoC driver for Migo-R
+//
+// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
 
 #include <linux/clkdev.h>
 #include <linux/device.h>
diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile
index 9c3d5aed99d1..5d1ff8ef26f9 100644
--- a/sound/soc/sh/rcar/Makefile
+++ b/sound/soc/sh/rcar/Makefile
@@ -1,2 +1,3 @@
+# SPDX-License-Identifier: GPL-2.0
 snd-soc-rcar-objs	:= core.o gen.o dma.o adg.o ssi.o ssiu.o src.o ctu.o mix.o dvc.o cmd.o
 obj-$(CONFIG_SND_SOC_RCAR)	+= snd-soc-rcar.o
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 4672688cac32..3a3064dda57f 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -1,12 +1,9 @@
-/*
- * Helper routines for R-Car sound ADG.
- *
- *  Copyright (C) 2013  Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This file is subject to the terms and conditions of the GNU General Public
- * License.  See the file "COPYING" in the main directory of this archive
- * for more details.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Helper routines for R-Car sound ADG.
+//
+//  Copyright (C) 2013  Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #include <linux/clk-provider.h>
 #include "rsnd.h"
 
diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c
index 5900fb535a2b..cc191cd5fb82 100644
--- a/sound/soc/sh/rcar/cmd.c
+++ b/sound/soc/sh/rcar/cmd.c
@@ -1,13 +1,10 @@
-/*
- * Renesas R-Car CMD support
- *
- * Copyright (C) 2015 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car CMD support
+//
+// Copyright (C) 2015 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #include "rsnd.h"
 
 struct rsnd_cmd {
@@ -89,7 +86,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod,
 			cmd_case[rsnd_mod_id(src)] << 16;
 	}
 
-	dev_dbg(dev, "ctu/mix path = 0x%08x", data);
+	dev_dbg(dev, "ctu/mix path = 0x%08x\n", data);
 
 	rsnd_mod_write(mod, CMD_ROUTE_SLCT, data);
 	rsnd_mod_write(mod, CMD_BUSIF_MODE, rsnd_get_busif_shift(io, mod) | 1);
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index f237002180c0..f8425d8b44d2 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1,16 +1,12 @@
-/*
- * Renesas R-Car SRU/SCU/SSIU/SSI support
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * Based on fsi.c
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car SRU/SCU/SSIU/SSI support
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// Based on fsi.c
+// Kuninori Morimoto <morimoto.kuninori@renesas.com>
 
 /*
  * Renesas R-Car sound device structure
@@ -552,6 +548,15 @@ struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id)
 	return priv->rdai + id;
 }
 
+static struct snd_soc_dai_driver
+*rsnd_daidrv_get(struct rsnd_priv *priv, int id)
+{
+	if ((id < 0) || (id >= rsnd_rdai_nr(priv)))
+		return NULL;
+
+	return priv->daidrv + id;
+}
+
 #define rsnd_dai_to_priv(dai) snd_soc_dai_get_drvdata(dai)
 static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai)
 {
@@ -1037,7 +1042,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv,
 	int io_i;
 
 	rdai		= rsnd_rdai_get(priv, dai_i);
-	drv		= priv->daidrv + dai_i;
+	drv		= rsnd_daidrv_get(priv, dai_i);
 	io_playback	= &rdai->playback;
 	io_capture	= &rdai->capture;
 
@@ -1085,6 +1090,12 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv,
 		of_node_put(capture);
 	}
 
+	if (rsnd_ssi_is_pin_sharing(io_capture) ||
+	    rsnd_ssi_is_pin_sharing(io_playback)) {
+		/* should have symmetric_rates if pin sharing */
+		drv->symmetric_rates = 1;
+	}
+
 	dev_dbg(dev, "%s (%s/%s)\n", rdai->name,
 		rsnd_io_to_mod_ssi(io_playback) ? "play"    : " -- ",
 		rsnd_io_to_mod_ssi(io_capture) ? "capture" : "  --   ");
@@ -1606,7 +1617,7 @@ static struct platform_driver rsnd_driver = {
 };
 module_platform_driver(rsnd_driver);
 
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_DESCRIPTION("Renesas R-Car audio driver");
 MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
 MODULE_ALIAS("platform:rcar-pcm-audio");
diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c
index 83be7d3ae0a8..6a55aa753003 100644
--- a/sound/soc/sh/rcar/ctu.c
+++ b/sound/soc/sh/rcar/ctu.c
@@ -1,12 +1,9 @@
-/*
- * ctu.c
- *
- * Copyright (c) 2015 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ctu.c
+//
+// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #include "rsnd.h"
 
 #define CTU_NAME_SIZE	16
diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index ef82b94d038b..fe63ef8600d0 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -1,13 +1,10 @@
-/*
- * Renesas R-Car Audio DMAC support
- *
- * Copyright (C) 2015 Renesas Electronics Corp.
- * Copyright (c) 2015 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car Audio DMAC support
+//
+// Copyright (C) 2015 Renesas Electronics Corp.
+// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #include <linux/delay.h>
 #include <linux/of_dma.h>
 #include "rsnd.h"
diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c
index ca1780e0b830..2b16e0ce6bc5 100644
--- a/sound/soc/sh/rcar/dvc.c
+++ b/sound/soc/sh/rcar/dvc.c
@@ -1,13 +1,9 @@
-/*
- * Renesas R-Car DVC support
- *
- * Copyright (C) 2014 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car DVC support
+//
+// Copyright (C) 2014 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
 
 /*
  * Playback Volume
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 25642e92dae0..0230301fe078 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -1,13 +1,9 @@
-/*
- * Renesas R-Car Gen1 SRU/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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car Gen1 SRU/SSI support
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
 
 /*
  * #define DEBUG
diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c
index 1881b2de9126..8e3b57eaa708 100644
--- a/sound/soc/sh/rcar/mix.c
+++ b/sound/soc/sh/rcar/mix.c
@@ -1,12 +1,8 @@
-/*
- * mix.c
- *
- * Copyright (c) 2015 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// mix.c
+//
+// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
 
 /*
  *		    CTUn	MIXn
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 6d7280d2d9be..96d93330b1e1 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -1,13 +1,10 @@
-/*
- * Renesas R-Car
- *
- * 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #ifndef RSND_H
 #define RSND_H
 
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 6c72d1a81cf5..beccfbac7581 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -1,13 +1,9 @@
-/*
- * Renesas R-Car SRC 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car SRC support
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
 
 /*
  * you can enable below define if you don't need
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 6e1166ec24a0..8304e4ec9242 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -1,16 +1,12 @@
-/*
- * Renesas R-Car SSIU/SSI support
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * Based on fsi.c
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car SSIU/SSI support
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// Based on fsi.c
+// Kuninori Morimoto <morimoto.kuninori@renesas.com>
 
 /*
  * you can enable below define if you don't need
@@ -37,6 +33,7 @@
 #define	CHNL_4		(1 << 22)	/* Channels */
 #define	CHNL_6		(2 << 22)	/* Channels */
 #define	CHNL_8		(3 << 22)	/* Channels */
+#define DWL_MASK	(7 << 19)	/* Data Word Length mask */
 #define	DWL_8		(0 << 19)	/* Data Word Length */
 #define	DWL_16		(1 << 19)	/* Data Word Length */
 #define	DWL_18		(2 << 19)	/* Data Word Length */
@@ -353,21 +350,18 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
 	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
 	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
-	u32 cr_own;
-	u32 cr_mode;
-	u32 wsr;
+	u32 cr_own	= ssi->cr_own;
+	u32 cr_mode	= ssi->cr_mode;
+	u32 wsr		= ssi->wsr;
 	int is_tdm;
 
-	if (rsnd_ssi_is_parent(mod, io))
-		return;
-
 	is_tdm = rsnd_runtime_is_ssi_tdm(io);
 
 	/*
 	 * always use 32bit system word.
 	 * see also rsnd_ssi_master_clk_enable()
 	 */
-	cr_own = FORCE | SWL_32;
+	cr_own |= FORCE | SWL_32;
 
 	if (rdai->bit_clk_inv)
 		cr_own |= SCKP;
@@ -377,9 +371,18 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 		cr_own |= SDTA;
 	if (rdai->sys_delay)
 		cr_own |= DEL;
+
+	/*
+	 * We shouldn't exchange SWSP after running.
+	 * This means, parent needs to care it.
+	 */
+	if (rsnd_ssi_is_parent(mod, io))
+		goto init_end;
+
 	if (rsnd_io_is_play(io))
 		cr_own |= TRMD;
 
+	cr_own &= ~DWL_MASK;
 	switch (snd_pcm_format_width(runtime->format)) {
 	case 16:
 		cr_own |= DWL_16;
@@ -406,7 +409,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 		wsr	|= WS_MODE;
 		cr_own	|= CHNL_8;
 	}
-
+init_end:
 	ssi->cr_own	= cr_own;
 	ssi->cr_mode	= cr_mode;
 	ssi->wsr	= wsr;
@@ -470,15 +473,18 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod,
 		return -EIO;
 	}
 
-	if (!rsnd_ssi_is_parent(mod, io))
-		ssi->cr_own	= 0;
-
 	rsnd_ssi_master_clk_stop(mod, io);
 
 	rsnd_mod_power_off(mod);
 
 	ssi->usrcnt--;
 
+	if (!ssi->usrcnt) {
+		ssi->cr_own	= 0;
+		ssi->cr_mode	= 0;
+		ssi->wsr	= 0;
+	}
+
 	return 0;
 }
 
@@ -1055,9 +1061,10 @@ struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id)
 
 int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
 {
-	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+	if (!mod)
+		return 0;
 
-	return !!(rsnd_flags_has(ssi, RSND_SSI_CLK_PIN_SHARE));
+	return !!(rsnd_flags_has(rsnd_mod_to_ssi(mod), RSND_SSI_CLK_PIN_SHARE));
 }
 
 static u32 *rsnd_ssi_get_status(struct rsnd_dai_stream *io,
diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index 47bdba9fc582..016fbf5ac242 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -1,12 +1,9 @@
-/*
- * Renesas R-Car SSIU support
- *
- * Copyright (c) 2015 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.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car SSIU support
+//
+// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #include "rsnd.h"
 
 #define SSIU_NAME "ssiu"
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
index 4a3568a9bf59..4bb4c13cf860 100644
--- a/sound/soc/sh/sh7760-ac97.c
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -1,10 +1,8 @@
-/*
- * Generic AC97 sound support for SH7760
- *
- * (c) 2007 Manuel Lauss
- *
- * Licensed under the GPLv2.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Generic AC97 sound support for SH7760
+//
+// (c) 2007 Manuel Lauss
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
@@ -68,6 +66,6 @@ static void __exit sh7760_ac97_exit(void)
 module_init(sh7760_ac97_init);
 module_exit(sh7760_ac97_exit);
 
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_DESCRIPTION("Generic SH7760 AC97 sound machine");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h
index 6088d627c0e4..63a508fdfe78 100644
--- a/sound/soc/sh/siu.h
+++ b/sound/soc/sh/siu.h
@@ -1,23 +1,9 @@
-/*
- * siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral.
- *
- * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- * Copyright (C) 2006 Carlos Munoz <carlos@kenati.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.
- *
- * 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., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral.
+//
+// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
+// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
 
 #ifndef SIU_H
 #define SIU_H
diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c
index ee2211635e92..f2a386fcd92e 100644
--- a/sound/soc/sh/siu_dai.c
+++ b/sound/soc/sh/siu_dai.c
@@ -1,23 +1,9 @@
-/*
- * siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral.
- *
- * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- * Copyright (C) 2006 Carlos Munoz <carlos@kenati.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.
- *
- * 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., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral.
+//
+// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
+// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
 
 #include <linux/delay.h>
 #include <linux/firmware.h>
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index 172909570ed5..e263757e4a69 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -1,23 +1,10 @@
-/*
- * siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral.
- *
- * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- * Copyright (C) 2006 Carlos Munoz <carlos@kenati.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.
- *
- * 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., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral.
+//
+// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
+// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
+
 #include <linux/delay.h>
 #include <linux/dma-mapping.h>
 #include <linux/dmaengine.h>
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 89ed1b107ac5..8125fa3840b6 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -1,14 +1,11 @@
-/*
- * Serial Sound Interface (I2S) support for SH7760/SH7780
- *
- * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
- *
- *  licensed under the terms outlined in the file COPYING at the root
- *  of the linux kernel sources.
- *
- * dont forget to set IPSEL/OMSEL register bits (in your board code) to
- * enable SSI output pins!
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Serial Sound Interface (I2S) support for SH7760/SH7780
+//
+// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+//
+// dont forget to set IPSEL/OMSEL register bits (in your board code) to
+// enable SSI output pins!
 
 /*
  * LIMITATIONS:
@@ -400,6 +397,6 @@ static struct platform_driver sh4_ssi_driver = {
 
 module_platform_driver(sh4_ssi_driver);
 
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c
index 77e7dcf969d0..d70fcd4a1adf 100644
--- a/sound/soc/sirf/sirf-usp.c
+++ b/sound/soc/sirf/sirf-usp.c
@@ -370,10 +370,9 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, usp);
 
 	mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_ioremap(&pdev->dev, mem_res->start,
-		resource_size(mem_res));
-	if (base == NULL)
-		return -ENOMEM;
+	base = devm_ioremap_resource(&pdev->dev, mem_res);
+	if (IS_ERR(base))
+		return PTR_ERR(base);
 	usp->regmap = devm_regmap_init_mmio(&pdev->dev, base,
 					    &sirf_usp_regmap_config);
 	if (IS_ERR(usp->regmap))
diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c
index 3f424f214bca..c086786e4471 100644
--- a/sound/soc/soc-ac97.c
+++ b/sound/soc/soc-ac97.c
@@ -1,20 +1,15 @@
-/*
- * soc-ac97.c  --  ALSA SoC Audio Layer AC97 support
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- * Copyright (C) 2010 Slimlogic Ltd.
- * Copyright (C) 2010 Texas Instruments Inc.
- *
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
- *         with code, comments and ideas from :-
- *         Richard Purdie <richard@openedhand.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.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-ac97.c  --  ALSA SoC Audio Layer AC97 support
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Copyright 2005 Openedhand Ltd.
+// Copyright (C) 2010 Slimlogic Ltd.
+// Copyright (C) 2010 Texas Instruments Inc.
+//
+// Author: Liam Girdwood <lrg@slimlogic.co.uk>
+//         with code, comments and ideas from :-
+//         Richard Purdie <richard@openedhand.com>
 
 #include <linux/ctype.h>
 #include <linux/delay.h>
diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c
index 3d7e1ff79139..b8e72b52db30 100644
--- a/sound/soc/soc-acpi.c
+++ b/sound/soc/soc-acpi.c
@@ -1,18 +1,8 @@
-/*
- * soc-apci.c - support for ACPI enumeration.
- *
- * Copyright (c) 2013-15, Intel Corporation.
- *
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms and conditions of the GNU General Public License,
- * version 2, as published by the Free Software Foundation.
- *
- * This program is distributed in the hope it will be useful, but WITHOUT
- * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
- * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
- * more details.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// soc-apci.c - support for ACPI enumeration.
+//
+// Copyright (c) 2013-15, Intel Corporation.
 
 #include <sound/soc-acpi.h>
 
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index e095115fa9f9..409d082e80d1 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -1,18 +1,12 @@
-/*
- * soc-compress.c  --  ALSA SoC Compress
- *
- * Copyright (C) 2012 Intel Corp.
- *
- * Authors: Namarta Kohli <namartax.kohli@intel.com>
- *          Ramesh Babu K V <ramesh.babu@linux.intel.com>
- *          Vinod Koul <vinod.koul@linux.intel.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.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-compress.c  --  ALSA SoC Compress
+//
+// Copyright (C) 2012 Intel Corp.
+//
+// Authors: Namarta Kohli <namartax.kohli@intel.com>
+//          Ramesh Babu K V <ramesh.babu@linux.intel.com>
+//          Vinod Koul <vinod.koul@linux.intel.com>
 
 #include <linux/kernel.h>
 #include <linux/init.h>
@@ -146,6 +140,30 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
 		stream = SNDRV_PCM_STREAM_CAPTURE;
 
 	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+	fe->dpcm[stream].runtime = fe_substream->runtime;
+
+	ret = dpcm_path_get(fe, stream, &list);
+	if (ret < 0)
+		goto be_err;
+	else if (ret == 0)
+		dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n",
+			fe->dai_link->name, stream ? "capture" : "playback");
+	/* calculate valid and active FE <-> BE dpcms */
+	dpcm_process_paths(fe, stream, &list, 1);
+	fe->dpcm[stream].runtime = fe_substream->runtime;
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+	ret = dpcm_be_dai_startup(fe, stream);
+	if (ret < 0) {
+		/* clean up all links */
+		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+			dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
+
+		dpcm_be_disconnect(fe, stream);
+		fe->dpcm[stream].runtime = NULL;
+		goto out;
+	}
 
 	if (cpu_dai->driver->cops && cpu_dai->driver->cops->startup) {
 		ret = cpu_dai->driver->cops->startup(cstream, cpu_dai);
@@ -159,7 +177,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
 
 	ret = soc_compr_components_open(cstream, &component);
 	if (ret < 0)
-		goto machine_err;
+		goto open_err;
 
 	if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->startup) {
 		ret = fe->dai_link->compr_ops->startup(cstream);
@@ -170,31 +188,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
 		}
 	}
 
-	fe->dpcm[stream].runtime = fe_substream->runtime;
-
-	ret = dpcm_path_get(fe, stream, &list);
-	if (ret < 0)
-		goto fe_err;
-	else if (ret == 0)
-		dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n",
-			fe->dai_link->name, stream ? "capture" : "playback");
-
-	/* calculate valid and active FE <-> BE dpcms */
-	dpcm_process_paths(fe, stream, &list, 1);
-
-	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
-
-	ret = dpcm_be_dai_startup(fe, stream);
-	if (ret < 0) {
-		/* clean up all links */
-		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
-			dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
-
-		dpcm_be_disconnect(fe, stream);
-		fe->dpcm[stream].runtime = NULL;
-		goto path_err;
-	}
-
 	dpcm_clear_pending_state(fe, stream);
 	dpcm_path_put(&list);
 
@@ -207,17 +200,14 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
 
 	return 0;
 
-path_err:
-	dpcm_path_put(&list);
-fe_err:
-	if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown)
-		fe->dai_link->compr_ops->shutdown(cstream);
 machine_err:
 	soc_compr_components_free(cstream, component);
-
+open_err:
 	if (cpu_dai->driver->cops && cpu_dai->driver->cops->shutdown)
 		cpu_dai->driver->cops->shutdown(cstream, cpu_dai);
 out:
+	dpcm_path_put(&list);
+be_err:
 	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
 	mutex_unlock(&fe->card->mutex);
 	return ret;
@@ -557,6 +547,24 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream,
 
 	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
 
+	/*
+	 * Create an empty hw_params for the BE as the machine driver must
+	 * fix this up to match DSP decoder and ASRC configuration.
+	 * I.e. machine driver fixup for compressed BE is mandatory.
+	 */
+	memset(&fe->dpcm[fe_substream->stream].hw_params, 0,
+		sizeof(struct snd_pcm_hw_params));
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+	ret = dpcm_be_dai_hw_params(fe, stream);
+	if (ret < 0)
+		goto out;
+
+	ret = dpcm_be_dai_prepare(fe, stream);
+	if (ret < 0)
+		goto out;
+
 	if (cpu_dai->driver->cops && cpu_dai->driver->cops->set_params) {
 		ret = cpu_dai->driver->cops->set_params(cstream, params, cpu_dai);
 		if (ret < 0)
@@ -583,24 +591,6 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream,
 			goto out;
 	}
 
-	/*
-	 * Create an empty hw_params for the BE as the machine driver must
-	 * fix this up to match DSP decoder and ASRC configuration.
-	 * I.e. machine driver fixup for compressed BE is mandatory.
-	 */
-	memset(&fe->dpcm[fe_substream->stream].hw_params, 0,
-		sizeof(struct snd_pcm_hw_params));
-
-	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
-
-	ret = dpcm_be_dai_hw_params(fe, stream);
-	if (ret < 0)
-		goto out;
-
-	ret = dpcm_be_dai_prepare(fe, stream);
-	if (ret < 0)
-		goto out;
-
 	dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START);
 	fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE;
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4663de3cf495..9cfe10d8040c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1,26 +1,21 @@
-/*
- * soc-core.c  --  ALSA SoC Audio Layer
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- * Copyright (C) 2010 Slimlogic Ltd.
- * Copyright (C) 2010 Texas Instruments Inc.
- *
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
- *         with code, comments and ideas from :-
- *         Richard Purdie <richard@openedhand.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.
- *
- *  TODO:
- *   o Add hw rules to enforce rates, etc.
- *   o More testing with other codecs/machines.
- *   o Add more codecs and platforms to ensure good API coverage.
- *   o Support TDM on PCM and I2S
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-core.c  --  ALSA SoC Audio Layer
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Copyright 2005 Openedhand Ltd.
+// Copyright (C) 2010 Slimlogic Ltd.
+// Copyright (C) 2010 Texas Instruments Inc.
+//
+// Author: Liam Girdwood <lrg@slimlogic.co.uk>
+//         with code, comments and ideas from :-
+//         Richard Purdie <richard@openedhand.com>
+//
+//  TODO:
+//   o Add hw rules to enforce rates, etc.
+//   o More testing with other codecs/machines.
+//   o Add more codecs and platforms to ensure good API coverage.
+//   o Support TDM on PCM and I2S
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
@@ -533,6 +528,7 @@ int snd_soc_suspend(struct device *dev)
 						"ASoC: idle_bias_off CODEC on over suspend\n");
 					break;
 				}
+				/* fall through */
 
 			case SND_SOC_BIAS_OFF:
 				if (component->driver->suspend)
@@ -852,6 +848,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 	const char *platform_name;
 	int i;
 
+	if (dai_link->ignore)
+		return 0;
+
 	dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name);
 
 	if (soc_is_dai_link_bound(card, dai_link)) {
@@ -1195,15 +1194,27 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card,
 }
 EXPORT_SYMBOL_GPL(snd_soc_remove_dai_link);
 
+static void soc_set_of_name_prefix(struct snd_soc_component *component)
+{
+	struct device_node *component_of_node = component->dev->of_node;
+	const char *str;
+	int ret;
+
+	if (!component_of_node && component->dev->parent)
+		component_of_node = component->dev->parent->of_node;
+
+	ret = of_property_read_string(component_of_node, "sound-name-prefix",
+				      &str);
+	if (!ret)
+		component->name_prefix = str;
+}
+
 static void soc_set_name_prefix(struct snd_soc_card *card,
 				struct snd_soc_component *component)
 {
 	int i;
 
-	if (card->codec_conf == NULL)
-		return;
-
-	for (i = 0; i < card->num_configs; i++) {
+	for (i = 0; i < card->num_configs && card->codec_conf; i++) {
 		struct snd_soc_codec_conf *map = &card->codec_conf[i];
 		struct device_node *component_of_node = component->dev->of_node;
 
@@ -1215,8 +1226,14 @@ static void soc_set_name_prefix(struct snd_soc_card *card,
 		if (map->dev_name && strcmp(component->name, map->dev_name))
 			continue;
 		component->name_prefix = map->name_prefix;
-		break;
+		return;
 	}
+
+	/*
+	 * If there is no configuration table or no match in the table,
+	 * check if a prefix is provided in the node
+	 */
+	soc_set_of_name_prefix(component);
 }
 
 static int soc_probe_component(struct snd_soc_card *card,
@@ -1461,7 +1478,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 {
 	struct snd_soc_dai_link *dai_link = rtd->dai_link;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	int i, ret;
+	struct snd_soc_rtdcom_list *rtdcom;
+	struct snd_soc_component *component;
+	int i, ret, num;
 
 	dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n",
 			card->name, rtd->num, order);
@@ -1507,9 +1526,28 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 		soc_dpcm_debugfs_add(rtd);
 #endif
 
+	num = rtd->num;
+
+	/*
+	 * most drivers will register their PCMs using DAI link ordering but
+	 * topology based drivers can use the DAI link id field to set PCM
+	 * device number and then use rtd + a base offset of the BEs.
+	 */
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		if (!component->driver->use_dai_pcm_id)
+			continue;
+
+		if (rtd->dai_link->no_pcm)
+			num += component->driver->be_pcm_base;
+		else
+			num = rtd->dai_link->id;
+	}
+
 	if (cpu_dai->driver->compress_new) {
 		/*create compress_device"*/
-		ret = cpu_dai->driver->compress_new(rtd, rtd->num);
+		ret = cpu_dai->driver->compress_new(rtd, num);
 		if (ret < 0) {
 			dev_err(card->dev, "ASoC: can't create compress %s\n",
 					 dai_link->stream_name);
@@ -1519,7 +1557,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 
 		if (!dai_link->params) {
 			/* create the pcm */
-			ret = soc_new_pcm(rtd, rtd->num);
+			ret = soc_new_pcm(rtd, num);
 			if (ret < 0) {
 				dev_err(card->dev, "ASoC: can't create pcm %s :%d\n",
 				       dai_link->stream_name, ret);
@@ -1846,6 +1884,74 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
 EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name);
 #endif /* CONFIG_DMI */
 
+static void soc_check_tplg_fes(struct snd_soc_card *card)
+{
+	struct snd_soc_component *component;
+	const struct snd_soc_component_driver *comp_drv;
+	struct snd_soc_dai_link *dai_link;
+	int i;
+
+	list_for_each_entry(component, &component_list, list) {
+
+		/* does this component override FEs ? */
+		if (!component->driver->ignore_machine)
+			continue;
+
+		/* for this machine ? */
+		if (strcmp(component->driver->ignore_machine,
+			   card->dev->driver->name))
+			continue;
+
+		/* machine matches, so override the rtd data */
+		for (i = 0; i < card->num_links; i++) {
+
+			dai_link = &card->dai_link[i];
+
+			/* ignore this FE */
+			if (dai_link->dynamic) {
+				dai_link->ignore = true;
+				continue;
+			}
+
+			dev_info(card->dev, "info: override FE DAI link %s\n",
+				 card->dai_link[i].name);
+
+			/* override platform component */
+			dai_link->platform_name = component->name;
+
+			/* convert non BE into BE */
+			dai_link->no_pcm = 1;
+
+			/* override any BE fixups */
+			dai_link->be_hw_params_fixup =
+				component->driver->be_hw_params_fixup;
+
+			/* most BE links don't set stream name, so set it to
+			 * dai link name if it's NULL to help bind widgets.
+			 */
+			if (!dai_link->stream_name)
+				dai_link->stream_name = dai_link->name;
+		}
+
+		/* Inform userspace we are using alternate topology */
+		if (component->driver->topology_name_prefix) {
+
+			/* topology shortname created ? */
+			if (!card->topology_shortname_created) {
+				comp_drv = component->driver;
+
+				snprintf(card->topology_shortname, 32, "%s-%s",
+					 comp_drv->topology_name_prefix,
+					 card->name);
+				card->topology_shortname_created = true;
+			}
+
+			/* use topology shortname */
+			card->name = card->topology_shortname;
+		}
+	}
+}
+
 static int snd_soc_instantiate_card(struct snd_soc_card *card)
 {
 	struct snd_soc_pcm_runtime *rtd;
@@ -1855,6 +1961,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	mutex_lock(&client_mutex);
 	mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT);
 
+	/* check whether any platform is ignore machine FE and using topology */
+	soc_check_tplg_fes(card);
+
 	/* bind DAIs */
 	for (i = 0; i < card->num_links; i++) {
 		ret = soc_bind_dai_link(card, &card->dai_link[i]);
@@ -2523,6 +2632,28 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
 EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map);
 
 /**
+ * snd_soc_dai_get_channel_map - Get DAI audio channel map
+ * @dai: DAI
+ * @tx_num: how many TX channels
+ * @tx_slot: pointer to an array which imply the TX slot number channel
+ *           0~num-1 uses
+ * @rx_num: how many RX channels
+ * @rx_slot: pointer to an array which imply the RX slot number channel
+ *           0~num-1 uses
+ */
+int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
+	unsigned int *tx_num, unsigned int *tx_slot,
+	unsigned int *rx_num, unsigned int *rx_slot)
+{
+	if (dai->driver->ops->get_channel_map)
+		return dai->driver->ops->get_channel_map(dai, tx_num, tx_slot,
+			rx_num, rx_slot);
+	else
+		return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_get_channel_map);
+
+/**
  * snd_soc_dai_set_tristate - configure DAI system or master clock.
  * @dai: DAI
  * @tristate: tristate enable
@@ -3258,9 +3389,9 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
 }
 EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets);
 
-static int snd_soc_of_get_slot_mask(struct device_node *np,
-				    const char *prop_name,
-				    unsigned int *mask)
+int snd_soc_of_get_slot_mask(struct device_node *np,
+			     const char *prop_name,
+			     unsigned int *mask)
 {
 	u32 val;
 	const __be32 *of_slot_mask = of_get_property(np, prop_name, &val);
@@ -3275,6 +3406,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np,
 
 	return val;
 }
+EXPORT_SYMBOL_GPL(snd_soc_of_get_slot_mask);
 
 int snd_soc_of_parse_tdm_slot(struct device_node *np,
 			      unsigned int *tx_mask,
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 229c12349803..7e96793050c9 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1,27 +1,21 @@
-/*
- * soc-dapm.c  --  ALSA SoC Dynamic Audio Power Management
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
- *
- *  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.
- *
- *  Features:
- *    o Changes power status of internal codec blocks depending on the
- *      dynamic configuration of codec internal audio paths and active
- *      DACs/ADCs.
- *    o Platform power domain - can support external components i.e. amps and
- *      mic/headphone insertion events.
- *    o Automatic Mic Bias support
- *    o Jack insertion power event initiation - e.g. hp insertion will enable
- *      sinks, dacs, etc
- *    o Delayed power down of audio subsystem to reduce pops between a quick
- *      device reopen.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-dapm.c  --  ALSA SoC Dynamic Audio Power Management
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Author: Liam Girdwood <lrg@slimlogic.co.uk>
+//
+//  Features:
+//    o Changes power status of internal codec blocks depending on the
+//      dynamic configuration of codec internal audio paths and active
+//      DACs/ADCs.
+//    o Platform power domain - can support external components i.e. amps and
+//      mic/headphone insertion events.
+//    o Automatic Mic Bias support
+//    o Jack insertion power event initiation - e.g. hp insertion will enable
+//      sinks, dacs, etc
+//    o Delayed power down of audio subsystem to reduce pops between a quick
+//      device reopen.
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
@@ -3662,7 +3656,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 	struct snd_pcm_substream substream;
 	struct snd_pcm_hw_params *params = NULL;
 	struct snd_pcm_runtime *runtime = NULL;
-	u64 fmt;
+	unsigned int fmt;
 	int ret;
 
 	if (WARN_ON(!config) ||
@@ -4073,6 +4067,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
 			continue;
 		}
 
+		/* let users know there is no DAI to link */
+		if (!dai_w->priv) {
+			dev_dbg(card->dev, "dai widget %s has no DAI\n",
+				dai_w->name);
+			continue;
+		}
+
 		dai = dai_w->priv;
 
 		/* ...find all widgets with the same stream and link them */
diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index 7ac745df1412..a9ea172a66a7 100644
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -1,13 +1,8 @@
-/*
- * soc-devres.c  --  ALSA SoC Audio Layer devres functions
- *
- * Copyright (C) 2013 Linaro Ltd
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-devres.c  --  ALSA SoC Audio Layer devres functions
+//
+// Copyright (C) 2013 Linaro Ltd
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 56a541b9ff9e..52fd7af952a5 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -1,17 +1,8 @@
-/*
- *  Copyright (C) 2013, Analog Devices Inc.
- *	Author: Lars-Peter Clausen <lars@metafoo.de>
- *
- *  This program is free software; you can redistribute it and/or modify it
- *  under  the terms of the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the License, or (at your
- *  option) any later version.
- *
- *  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.,
- *  675 Mass Ave, Cambridge, MA 02139, USA.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+//  Copyright (C) 2013, Analog Devices Inc.
+//	Author: Lars-Peter Clausen <lars@metafoo.de>
+
 #include <linux/module.h>
 #include <linux/init.h>
 #include <linux/dmaengine.h>
@@ -197,7 +188,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea
 			case 32:
 			case 64:
 				if (addr_widths & (1 << (bits / 8)))
-					hw.formats |= (1LL << i);
+					hw.formats |= pcm_format_to_bits(i);
 				break;
 			default:
 				/* Unsupported types */
@@ -343,7 +334,7 @@ static snd_pcm_uframes_t dmaengine_pcm_pointer(
 
 static int dmaengine_copy_user(struct snd_pcm_substream *substream,
 			       int channel, unsigned long hwoff,
-			       void *buf, unsigned long bytes)
+			       void __user *buf, unsigned long bytes)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_component *component =
@@ -359,18 +350,17 @@ static int dmaengine_copy_user(struct snd_pcm_substream *substream,
 	int ret;
 
 	if (is_playback)
-		if (copy_from_user(dma_ptr, (void __user *)buf, bytes))
+		if (copy_from_user(dma_ptr, buf, bytes))
 			return -EFAULT;
 
 	if (process) {
-		ret = process(substream, channel, hwoff,
-			      (void __user *)buf, bytes);
+		ret = process(substream, channel, hwoff, (__force void *)buf, bytes);
 		if (ret < 0)
 			return ret;
 	}
 
 	if (!is_playback)
-		if (copy_to_user((void __user *)buf, dma_ptr, bytes))
+		if (copy_to_user(buf, dma_ptr, bytes))
 			return -EFAULT;
 
 	return 0;
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index 026cd5347e53..1ff9175e9d5e 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -1,15 +1,10 @@
-/*
- * soc-io.c  --  ASoC register I/O helpers
- *
- * Copyright 2009-2011 Wolfson Microelectronics PLC.
- *
- * Author: Mark Brown <broonie@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 as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-io.c  --  ASoC register I/O helpers
+//
+// Copyright 2009-2011 Wolfson Microelectronics PLC.
+//
+// Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
 
 #include <linux/i2c.h>
 #include <linux/spi/spi.h>
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index b2b16044ae80..c7b990abdbaa 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -1,15 +1,10 @@
-/*
- * soc-jack.c  --  ALSA SoC jack handling
- *
- * Copyright 2008 Wolfson Microelectronics PLC.
- *
- * Author: Mark Brown <broonie@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 as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-jack.c  --  ALSA SoC jack handling
+//
+// Copyright 2008 Wolfson Microelectronics PLC.
+//
+// Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
 
 #include <sound/jack.h>
 #include <sound/soc.h>
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 7144a51ddfa9..592efb370c44 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -1,20 +1,15 @@
-/*
- * soc-ops.c  --  Generic ASoC operations
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- * Copyright (C) 2010 Slimlogic Ltd.
- * Copyright (C) 2010 Texas Instruments Inc.
- *
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
- *         with code, comments and ideas from :-
- *         Richard Purdie <richard@openedhand.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.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-ops.c  --  Generic ASoC operations
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Copyright 2005 Openedhand Ltd.
+// Copyright (C) 2010 Slimlogic Ltd.
+// Copyright (C) 2010 Texas Instruments Inc.
+//
+// Author: Liam Girdwood <lrg@slimlogic.co.uk>
+//         with code, comments and ideas from :-
+//         Richard Purdie <richard@openedhand.com>
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 5e7ae47a9658..e8b98bfd4cf1 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1,20 +1,14 @@
-/*
- * soc-pcm.c  --  ALSA SoC PCM
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- * Copyright (C) 2010 Slimlogic Ltd.
- * Copyright (C) 2010 Texas Instruments Inc.
- *
- * Authors: Liam Girdwood <lrg@ti.com>
- *          Mark Brown <broonie@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 as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-pcm.c  --  ALSA SoC PCM
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Copyright 2005 Openedhand Ltd.
+// Copyright (C) 2010 Slimlogic Ltd.
+// Copyright (C) 2010 Texas Instruments Inc.
+//
+// Authors: Liam Girdwood <lrg@ti.com>
+//          Mark Brown <broonie@opensource.wolfsonmicro.com>
 
 #include <linux/kernel.h>
 #include <linux/init.h>
@@ -448,6 +442,29 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 	hw->rate_max = min_not_zero(hw->rate_max, rate_max);
 }
 
+static int soc_pcm_components_close(struct snd_pcm_substream *substream,
+				    struct snd_soc_component *last)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_rtdcom_list *rtdcom;
+	struct snd_soc_component *component;
+
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		if (component == last)
+			break;
+
+		if (!component->driver->ops ||
+		    !component->driver->ops->close)
+			continue;
+
+		component->driver->ops->close(substream);
+	}
+
+	return 0;
+}
+
 /*
  * Called by ALSA when a PCM substream is opened, the runtime->hw record is
  * then initialized and any private data can be allocated. This also calls
@@ -462,7 +479,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai *codec_dai;
 	const char *codec_dai_name = "multicodec";
-	int i, ret = 0, __ret;
+	int i, ret = 0;
 
 	pinctrl_pm_select_default_state(cpu_dai->dev);
 	for (i = 0; i < rtd->num_codecs; i++)
@@ -486,7 +503,6 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		}
 	}
 
-	ret = 0;
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
 
@@ -494,16 +510,15 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		    !component->driver->ops->open)
 			continue;
 
-		__ret = component->driver->ops->open(substream);
-		if (__ret < 0) {
+		ret = component->driver->ops->open(substream);
+		if (ret < 0) {
 			dev_err(component->dev,
 				"ASoC: can't open component %s: %d\n",
-				component->name, __ret);
-			ret = __ret;
+				component->name, ret);
+			goto component_err;
 		}
 	}
-	if (ret < 0)
-		goto component_err;
+	component = NULL;
 
 	for (i = 0; i < rtd->num_codecs; i++) {
 		codec_dai = rtd->codec_dais[i];
@@ -612,15 +627,7 @@ codec_dai_err:
 	}
 
 component_err:
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->close)
-			continue;
-
-		component->driver->ops->close(substream);
-	}
+	soc_pcm_components_close(substream, component);
 
 	if (cpu_dai->driver->ops->shutdown)
 		cpu_dai->driver->ops->shutdown(substream, cpu_dai);
@@ -714,15 +721,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	if (rtd->dai_link->ops->shutdown)
 		rtd->dai_link->ops->shutdown(substream);
 
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->close)
-			continue;
-
-		component->driver->ops->close(substream);
-	}
+	soc_pcm_components_close(substream, NULL);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (snd_soc_runtime_ignore_pmdown_time(rtd)) {
@@ -860,8 +859,20 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream,
 		      struct snd_pcm_hw_params *params,
 		      struct snd_soc_dai *dai)
 {
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	int ret;
 
+	/* perform any topology hw_params fixups before DAI  */
+	if (rtd->dai_link->be_hw_params_fixup) {
+		ret = rtd->dai_link->be_hw_params_fixup(rtd, params);
+		if (ret < 0) {
+			dev_err(rtd->dev,
+				"ASoC: hw_params topology fixup failed %d\n",
+				ret);
+			return ret;
+		}
+	}
+
 	if (dai->driver->ops->hw_params) {
 		ret = dai->driver->ops->hw_params(substream, params, dai);
 		if (ret < 0) {
@@ -874,6 +885,29 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream,
+				      struct snd_soc_component *last)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_rtdcom_list *rtdcom;
+	struct snd_soc_component *component;
+
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		if (component == last)
+			break;
+
+		if (!component->driver->ops ||
+		    !component->driver->ops->hw_free)
+			continue;
+
+		component->driver->ops->hw_free(substream);
+	}
+
+	return 0;
+}
+
 /*
  * Called by ALSA when the hardware params are set by application. This
  * function can also be called multiple times and can allocate buffers
@@ -886,7 +920,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_component *component;
 	struct snd_soc_rtdcom_list *rtdcom;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	int i, ret = 0, __ret;
+	int i, ret = 0;
 
 	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
 	if (rtd->dai_link->ops->hw_params) {
@@ -944,7 +978,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		goto interface_err;
 
-	ret = 0;
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
 
@@ -952,16 +985,15 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		    !component->driver->ops->hw_params)
 			continue;
 
-		__ret = component->driver->ops->hw_params(substream, params);
-		if (__ret < 0) {
+		ret = component->driver->ops->hw_params(substream, params);
+		if (ret < 0) {
 			dev_err(component->dev,
 				"ASoC: %s hw params failed: %d\n",
-				component->name, __ret);
-			ret = __ret;
+				component->name, ret);
+			goto component_err;
 		}
 	}
-	if (ret < 0)
-		goto component_err;
+	component = NULL;
 
 	/* store the parameters for each DAIs */
 	cpu_dai->rate = params_rate(params);
@@ -977,15 +1009,7 @@ out:
 	return ret;
 
 component_err:
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->hw_free)
-			continue;
-
-		component->driver->ops->hw_free(substream);
-	}
+	soc_pcm_components_hw_free(substream, component);
 
 	if (cpu_dai->driver->ops->hw_free)
 		cpu_dai->driver->ops->hw_free(substream, cpu_dai);
@@ -1014,8 +1038,6 @@ codec_err:
 static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_component *component;
-	struct snd_soc_rtdcom_list *rtdcom;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai *codec_dai;
 	bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
@@ -1052,15 +1074,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 		rtd->dai_link->ops->hw_free(substream);
 
 	/* free any component resources */
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->hw_free)
-			continue;
-
-		component->driver->ops->hw_free(substream);
-	}
+	soc_pcm_components_hw_free(substream, NULL);
 
 	/* now free hw params for the DAIs  */
 	for (i = 0; i < rtd->num_codecs; i++) {
@@ -1165,6 +1179,9 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
 	snd_pcm_sframes_t codec_delay = 0;
 	int i;
 
+	/* clearing the previous total delay */
+	runtime->delay = 0;
+
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
 
@@ -1176,6 +1193,8 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
 		offset = component->driver->ops->pointer(substream);
 		break;
 	}
+	/* base delay if assigned in pointer callback */
+	delay = runtime->delay;
 
 	if (cpu_dai->driver->ops->delay)
 		delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
@@ -1658,29 +1677,28 @@ unwind:
 }
 
 static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
-				 struct snd_soc_pcm_stream *stream,
-				 u64 formats)
+				 struct snd_soc_pcm_stream *stream)
 {
 	runtime->hw.rate_min = stream->rate_min;
 	runtime->hw.rate_max = stream->rate_max;
 	runtime->hw.channels_min = stream->channels_min;
 	runtime->hw.channels_max = stream->channels_max;
 	if (runtime->hw.formats)
-		runtime->hw.formats &= formats & stream->formats;
+		runtime->hw.formats &= stream->formats;
 	else
-		runtime->hw.formats = formats & stream->formats;
+		runtime->hw.formats = stream->formats;
 	runtime->hw.rates = stream->rates;
 }
 
-static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream)
+static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream,
+				      u64 *formats)
 {
 	struct snd_soc_pcm_runtime *fe = substream->private_data;
 	struct snd_soc_dpcm *dpcm;
-	u64 formats = ULLONG_MAX;
 	int stream = substream->stream;
 
 	if (!fe->dai_link->dpcm_merged_format)
-		return formats;
+		return;
 
 	/*
 	 * It returns merged BE codec format
@@ -1694,17 +1712,132 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream)
 		int i;
 
 		for (i = 0; i < be->num_codecs; i++) {
+			/*
+			 * Skip CODECs which don't support the current stream
+			 * type. See soc_pcm_init_runtime_hw() for more details
+			 */
+			if (!snd_soc_dai_stream_valid(be->codec_dais[i],
+						      stream))
+				continue;
+
 			codec_dai_drv = be->codec_dais[i]->driver;
 			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
 				codec_stream = &codec_dai_drv->playback;
 			else
 				codec_stream = &codec_dai_drv->capture;
 
-			formats &= codec_stream->formats;
+			*formats &= codec_stream->formats;
 		}
 	}
+}
+
+static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream,
+				    unsigned int *channels_min,
+				    unsigned int *channels_max)
+{
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
+	struct snd_soc_dpcm *dpcm;
+	int stream = substream->stream;
 
-	return formats;
+	if (!fe->dai_link->dpcm_merged_chan)
+		return;
+
+	/*
+	 * It returns merged BE codec channel;
+	 * if FE want to use it (= dpcm_merged_chan)
+	 */
+
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		struct snd_soc_dai_driver *cpu_dai_drv =  be->cpu_dai->driver;
+		struct snd_soc_dai_driver *codec_dai_drv;
+		struct snd_soc_pcm_stream *codec_stream;
+		struct snd_soc_pcm_stream *cpu_stream;
+
+		if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+			cpu_stream = &cpu_dai_drv->playback;
+		else
+			cpu_stream = &cpu_dai_drv->capture;
+
+		*channels_min = max(*channels_min, cpu_stream->channels_min);
+		*channels_max = min(*channels_max, cpu_stream->channels_max);
+
+		/*
+		 * chan min/max cannot be enforced if there are multiple CODEC
+		 * DAIs connected to a single CPU DAI, use CPU DAI's directly
+		 */
+		if (be->num_codecs == 1) {
+			codec_dai_drv = be->codec_dais[0]->driver;
+
+			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+				codec_stream = &codec_dai_drv->playback;
+			else
+				codec_stream = &codec_dai_drv->capture;
+
+			*channels_min = max(*channels_min,
+					    codec_stream->channels_min);
+			*channels_max = min(*channels_max,
+					    codec_stream->channels_max);
+		}
+	}
+}
+
+static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream,
+				    unsigned int *rates,
+				    unsigned int *rate_min,
+				    unsigned int *rate_max)
+{
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
+	struct snd_soc_dpcm *dpcm;
+	int stream = substream->stream;
+
+	if (!fe->dai_link->dpcm_merged_rate)
+		return;
+
+	/*
+	 * It returns merged BE codec channel;
+	 * if FE want to use it (= dpcm_merged_chan)
+	 */
+
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		struct snd_soc_dai_driver *cpu_dai_drv =  be->cpu_dai->driver;
+		struct snd_soc_dai_driver *codec_dai_drv;
+		struct snd_soc_pcm_stream *codec_stream;
+		struct snd_soc_pcm_stream *cpu_stream;
+		int i;
+
+		if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+			cpu_stream = &cpu_dai_drv->playback;
+		else
+			cpu_stream = &cpu_dai_drv->capture;
+
+		*rate_min = max(*rate_min, cpu_stream->rate_min);
+		*rate_max = min_not_zero(*rate_max, cpu_stream->rate_max);
+		*rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates);
+
+		for (i = 0; i < be->num_codecs; i++) {
+			/*
+			 * Skip CODECs which don't support the current stream
+			 * type. See soc_pcm_init_runtime_hw() for more details
+			 */
+			if (!snd_soc_dai_stream_valid(be->codec_dais[i],
+						      stream))
+				continue;
+
+			codec_dai_drv = be->codec_dais[i]->driver;
+			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+				codec_stream = &codec_dai_drv->playback;
+			else
+				codec_stream = &codec_dai_drv->capture;
+
+			*rate_min = max(*rate_min, codec_stream->rate_min);
+			*rate_max = min_not_zero(*rate_max,
+						 codec_stream->rate_max);
+			*rates = snd_pcm_rate_mask_intersect(*rates,
+						codec_stream->rates);
+		}
+	}
 }
 
 static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream)
@@ -1713,12 +1846,17 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
-	u64 format = dpcm_runtime_base_format(substream);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback, format);
+		dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback);
 	else
-		dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture, format);
+		dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture);
+
+	dpcm_runtime_merge_format(substream, &runtime->hw.formats);
+	dpcm_runtime_merge_chan(substream, &runtime->hw.channels_min,
+				&runtime->hw.channels_max);
+	dpcm_runtime_merge_rate(substream, &runtime->hw.rates,
+				&runtime->hw.rate_min, &runtime->hw.rate_max);
 }
 
 static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd);
@@ -2543,106 +2681,113 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream)
 	return ret;
 }
 
-/* Called by DAPM mixer/mux changes to update audio routing between PCMs and
- * any DAI links.
- */
-int soc_dpcm_runtime_update(struct snd_soc_card *card)
+static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
 {
-	struct snd_soc_pcm_runtime *fe;
-	int old, new, paths;
+	struct snd_soc_dapm_widget_list *list;
+	int count, paths;
 
-	mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
-	list_for_each_entry(fe, &card->rtd_list, list) {
-		struct snd_soc_dapm_widget_list *list;
+	if (!fe->dai_link->dynamic)
+		return 0;
 
-		/* make sure link is FE */
-		if (!fe->dai_link->dynamic)
-			continue;
+	/* only check active links */
+	if (!fe->cpu_dai->active)
+		return 0;
 
-		/* only check active links */
-		if (!fe->cpu_dai->active)
-			continue;
+	/* DAPM sync will call this to update DSP paths */
+	dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n",
+		new ? "new" : "old", fe->dai_link->name);
 
-		/* DAPM sync will call this to update DSP paths */
-		dev_dbg(fe->dev, "ASoC: DPCM runtime update for FE %s\n",
-			fe->dai_link->name);
+	/* skip if FE doesn't have playback capability */
+	if (!fe->cpu_dai->driver->playback.channels_min ||
+	    !fe->codec_dai->driver->playback.channels_min)
+		goto capture;
 
-		/* skip if FE doesn't have playback capability */
-		if (!fe->cpu_dai->driver->playback.channels_min
-		    || !fe->codec_dai->driver->playback.channels_min)
-			goto capture;
-
-		/* skip if FE isn't currently playing */
-		if (!fe->cpu_dai->playback_active
-		    || !fe->codec_dai->playback_active)
-			goto capture;
-
-		paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
-		if (paths < 0) {
-			dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
-					fe->dai_link->name,  "playback");
-			mutex_unlock(&card->mutex);
-			return paths;
-		}
+	/* skip if FE isn't currently playing */
+	if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active)
+		goto capture;
 
-		/* update any new playback paths */
-		new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1);
-		if (new) {
-			dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
-			dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
-			dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
-		}
+	paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
+	if (paths < 0) {
+		dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
+			 fe->dai_link->name,  "playback");
+		return paths;
+	}
 
-		/* update any old playback paths */
-		old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0);
-		if (old) {
+	/* update any playback paths */
+	count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new);
+	if (count) {
+		if (new)
+			dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
+		else
 			dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
-			dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
-			dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
-		}
 
-		dpcm_path_put(&list);
+		dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
+		dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
+	}
+
+	dpcm_path_put(&list);
+
 capture:
-		/* skip if FE doesn't have capture capability */
-		if (!fe->cpu_dai->driver->capture.channels_min
-		    || !fe->codec_dai->driver->capture.channels_min)
-			continue;
+	/* skip if FE doesn't have capture capability */
+	if (!fe->cpu_dai->driver->capture.channels_min ||
+	    !fe->codec_dai->driver->capture.channels_min)
+		return 0;
 
-		/* skip if FE isn't currently capturing */
-		if (!fe->cpu_dai->capture_active
-		    || !fe->codec_dai->capture_active)
-			continue;
+	/* skip if FE isn't currently capturing */
+	if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active)
+		return 0;
 
-		paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
-		if (paths < 0) {
-			dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
-					fe->dai_link->name,  "capture");
-			mutex_unlock(&card->mutex);
-			return paths;
-		}
+	paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
+	if (paths < 0) {
+		dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
+			 fe->dai_link->name,  "capture");
+		return paths;
+	}
 
-		/* update any new capture paths */
-		new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1);
-		if (new) {
+	/* update any old capture paths */
+	count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new);
+	if (count) {
+		if (new)
 			dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE);
-			dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
-			dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
-		}
-
-		/* update any old capture paths */
-		old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0);
-		if (old) {
+		else
 			dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE);
-			dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
-			dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
-		}
 
-		dpcm_path_put(&list);
+		dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
+		dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
 	}
 
-	mutex_unlock(&card->mutex);
+	dpcm_path_put(&list);
+
 	return 0;
 }
+
+/* Called by DAPM mixer/mux changes to update audio routing between PCMs and
+ * any DAI links.
+ */
+int soc_dpcm_runtime_update(struct snd_soc_card *card)
+{
+	struct snd_soc_pcm_runtime *fe;
+	int ret = 0;
+
+	mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+	/* shutdown all old paths first */
+	list_for_each_entry(fe, &card->rtd_list, list) {
+		ret = soc_dpcm_fe_runtime_update(fe, 0);
+		if (ret)
+			goto out;
+	}
+
+	/* bring new paths up */
+	list_for_each_entry(fe, &card->rtd_list, list) {
+		ret = soc_dpcm_fe_runtime_update(fe, 1);
+		if (ret)
+			goto out;
+	}
+
+out:
+	mutex_unlock(&card->mutex);
+	return ret;
+}
 int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
 {
 	struct snd_soc_dpcm *dpcm;
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 53f121a50c97..66e77e020745 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1,29 +1,24 @@
-/*
- * soc-topology.c  --  ALSA SoC Topology
- *
- * Copyright (C) 2012 Texas Instruments Inc.
- * Copyright (C) 2015 Intel Corporation.
- *
- * Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
- *		K, Mythri P <mythri.p.k@intel.com>
- *		Prusty, Subhransu S <subhransu.s.prusty@intel.com>
- *		B, Jayachandran <jayachandran.b@intel.com>
- *		Abdullah, Omair M <omair.m.abdullah@intel.com>
- *		Jin, Yao <yao.jin@intel.com>
- *		Lin, Mengdong <mengdong.lin@intel.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.
- *
- *  Add support to read audio firmware topology alongside firmware text. The
- *  topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links,
- *  equalizers, firmware, coefficients etc.
- *
- *  This file only manages the core ALSA and ASoC components, all other bespoke
- *  firmware topology data is passed to component drivers for bespoke handling.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-topology.c  --  ALSA SoC Topology
+//
+// Copyright (C) 2012 Texas Instruments Inc.
+// Copyright (C) 2015 Intel Corporation.
+//
+// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
+//		K, Mythri P <mythri.p.k@intel.com>
+//		Prusty, Subhransu S <subhransu.s.prusty@intel.com>
+//		B, Jayachandran <jayachandran.b@intel.com>
+//		Abdullah, Omair M <omair.m.abdullah@intel.com>
+//		Jin, Yao <yao.jin@intel.com>
+//		Lin, Mengdong <mengdong.lin@intel.com>
+//
+//  Add support to read audio firmware topology alongside firmware text. The
+//  topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links,
+//  equalizers, firmware, coefficients etc.
+//
+//  This file only manages the core ALSA and ASoC components, all other bespoke
+//  firmware topology data is passed to component drivers for bespoke handling.
 
 #include <linux/kernel.h>
 #include <linux/export.h>
@@ -259,7 +254,7 @@ static int soc_tplg_vendor_load_(struct soc_tplg *tplg,
 	int ret = 0;
 
 	if (tplg->comp && tplg->ops && tplg->ops->vendor_load)
-		ret = tplg->ops->vendor_load(tplg->comp, hdr);
+		ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr);
 	else {
 		dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n",
 			hdr->vendor_type);
@@ -291,7 +286,8 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg,
 	struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w)
 {
 	if (tplg->comp && tplg->ops && tplg->ops->widget_load)
-		return tplg->ops->widget_load(tplg->comp, w, tplg_w);
+		return tplg->ops->widget_load(tplg->comp, tplg->index, w,
+			tplg_w);
 
 	return 0;
 }
@@ -302,27 +298,30 @@ static int soc_tplg_widget_ready(struct soc_tplg *tplg,
 	struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w)
 {
 	if (tplg->comp && tplg->ops && tplg->ops->widget_ready)
-		return tplg->ops->widget_ready(tplg->comp, w, tplg_w);
+		return tplg->ops->widget_ready(tplg->comp, tplg->index, w,
+			tplg_w);
 
 	return 0;
 }
 
 /* pass DAI configurations to component driver for extra initialization */
 static int soc_tplg_dai_load(struct soc_tplg *tplg,
-	struct snd_soc_dai_driver *dai_drv)
+	struct snd_soc_dai_driver *dai_drv,
+	struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai)
 {
 	if (tplg->comp && tplg->ops && tplg->ops->dai_load)
-		return tplg->ops->dai_load(tplg->comp, dai_drv);
+		return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv,
+			pcm, dai);
 
 	return 0;
 }
 
 /* pass link configurations to component driver for extra initialization */
 static int soc_tplg_dai_link_load(struct soc_tplg *tplg,
-	struct snd_soc_dai_link *link)
+	struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg)
 {
 	if (tplg->comp && tplg->ops && tplg->ops->link_load)
-		return tplg->ops->link_load(tplg->comp, link);
+		return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg);
 
 	return 0;
 }
@@ -643,7 +642,8 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg,
 	struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr)
 {
 	if (tplg->comp && tplg->ops && tplg->ops->control_load)
-		return tplg->ops->control_load(tplg->comp, k, hdr);
+		return tplg->ops->control_load(tplg->comp, tplg->index, k,
+			hdr);
 
 	return 0;
 }
@@ -1100,6 +1100,17 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg,
 	return 0;
 }
 
+/* optionally pass new dynamic kcontrol to component driver. */
+static int soc_tplg_add_route(struct soc_tplg *tplg,
+	struct snd_soc_dapm_route *route)
+{
+	if (tplg->comp && tplg->ops && tplg->ops->dapm_route_load)
+		return tplg->ops->dapm_route_load(tplg->comp, tplg->index,
+			route);
+
+	return 0;
+}
+
 static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
 	struct snd_soc_tplg_hdr *hdr)
 {
@@ -1148,6 +1159,8 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
 		else
 			route.control = elem->control;
 
+		soc_tplg_add_route(tplg, &route);
+
 		/* add route, but keep going if some fail */
 		snd_soc_dapm_add_routes(dapm, &route, 1);
 	}
@@ -1702,7 +1715,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
 		dai_drv->compress_new = snd_soc_new_compress;
 
 	/* pass control to component driver for optional further init */
-	ret = soc_tplg_dai_load(tplg, dai_drv);
+	ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL);
 	if (ret < 0) {
 		dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n");
 		kfree(dai_drv);
@@ -1772,7 +1785,7 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg,
 		set_link_flags(link, pcm->flag_mask, pcm->flags);
 
 	/* pass control to component driver for optional further init */
-	ret = soc_tplg_dai_link_load(tplg, link);
+	ret = soc_tplg_dai_link_load(tplg, link, NULL);
 	if (ret < 0) {
 		dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n");
 		kfree(link);
@@ -2080,7 +2093,7 @@ static int soc_tplg_link_config(struct soc_tplg *tplg,
 		set_link_flags(link, cfg->flag_mask, cfg->flags);
 
 	/* pass control to component driver for optional further init */
-	ret = soc_tplg_dai_link_load(tplg, link);
+	ret = soc_tplg_dai_link_load(tplg, link, cfg);
 	if (ret < 0) {
 		dev_err(tplg->dev, "ASoC: physical link loading failed\n");
 		return ret;
@@ -2202,7 +2215,7 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg,
 		set_dai_flags(dai_drv, d->flag_mask, d->flags);
 
 	/* pass control to component driver for optional further init */
-	ret = soc_tplg_dai_load(tplg, dai_drv);
+	ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai);
 	if (ret < 0) {
 		dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n");
 		return ret;
@@ -2311,7 +2324,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg,
 
 	/* pass control to component driver for optional further init */
 	if (tplg->comp && tplg->ops && tplg->ops->manifest)
-		return tplg->ops->manifest(tplg->comp, _manifest);
+		return tplg->ops->manifest(tplg->comp, tplg->index, _manifest);
 
 	if (!abi_match)	/* free the duplicated one */
 		kfree(_manifest);
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 2d9e98bd1530..e0c93496c0cd 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -1,17 +1,11 @@
-/*
- * soc-util.c  --  ALSA SoC Audio Layer utility functions
- *
- * Copyright 2009 Wolfson Microelectronics PLC.
- *
- * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
- *         Liam Girdwood <lrg@slimlogic.co.uk>
- *         
- *
- *  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.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-util.c  --  ALSA SoC Audio Layer utility functions
+//
+// Copyright 2009 Wolfson Microelectronics PLC.
+//
+// Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+//         Liam Girdwood <lrg@slimlogic.co.uk>
 
 #include <linux/platform_device.h>
 #include <linux/export.h>
@@ -381,6 +375,6 @@ int __init snd_soc_util_init(void)
 
 void __exit snd_soc_util_exit(void)
 {
-	platform_device_unregister(soc_dummy_dev);
 	platform_driver_unregister(&soc_dummy_driver);
+	platform_device_unregister(soc_dummy_dev);
 }
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index d8b6936e544e..313dab2857ef 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -91,7 +91,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
 			SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player);
 
 			/* Stop the player */
-			snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN);
+			snd_pcm_stop_xrun(player->substream);
 		}
 
 		ret = IRQ_HANDLED;
@@ -105,7 +105,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
 		SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player);
 
 		/* Stop the player */
-		snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN);
+		snd_pcm_stop_xrun(player->substream);
 
 		ret = IRQ_HANDLED;
 	}
@@ -138,7 +138,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
 		dev_err(player->dev, "Underflow recovery failed\n");
 
 		/* Stop the player */
-		snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN);
+		snd_pcm_stop_xrun(player->substream);
 
 		ret = IRQ_HANDLED;
 	}
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index ee0055e60852..7b63d35ef428 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -65,7 +65,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id)
 	if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) {
 		dev_err(reader->dev, "FIFO error detected\n");
 
-		snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN);
+		snd_pcm_stop_xrun(reader->substream);
 
 		ret = IRQ_HANDLED;
 	}
diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig
index 48f9ddd94016..9b2681397dba 100644
--- a/sound/soc/stm/Kconfig
+++ b/sound/soc/stm/Kconfig
@@ -6,6 +6,7 @@ config SND_SOC_STM32_SAI
 	depends on SND_SOC
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 	select REGMAP_MMIO
+	select SND_PCM_IEC958
 	help
 	  Say Y if you want to enable SAI for STM32
 
diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c
index db73fef3e500..706ff005234f 100644
--- a/sound/soc/stm/stm32_adfsdm.c
+++ b/sound/soc/stm/stm32_adfsdm.c
@@ -149,7 +149,7 @@ static int stm32_afsdm_pcm_cb(const void *data, size_t size, void *private)
 	unsigned int old_pos = priv->pos;
 	unsigned int cur_size = size;
 
-	dev_dbg(rtd->dev, "%s: buff_add :%p, pos = %d, size = %zu\n",
+	dev_dbg(rtd->dev, "%s: buff_add :%pK, pos = %d, size = %zu\n",
 		__func__, &pcm_buff[priv->pos], priv->pos, size);
 
 	if ((priv->pos + size) > buff_size) {
@@ -269,16 +269,10 @@ static int stm32_adfsdm_pcm_new(struct snd_soc_pcm_runtime *rtd)
 static void stm32_adfsdm_pcm_free(struct snd_pcm *pcm)
 {
 	struct snd_pcm_substream *substream;
-	struct snd_soc_pcm_runtime *rtd;
-	struct stm32_adfsdm_priv *priv;
 
 	substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
-	if (substream) {
-		rtd = substream->private_data;
-		priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
-
+	if (substream)
 		snd_pcm_lib_preallocate_free_for_all(pcm);
-	}
 }
 
 static struct snd_soc_component_driver stm32_adfsdm_soc_platform = {
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index cfeb219e1d78..06fba9650ac4 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -96,7 +96,8 @@
  * @slot_mask: rx or tx active slots mask. set at init or at runtime
  * @data_size: PCM data width. corresponds to PCM substream width.
  * @spdif_frm_cnt: S/PDIF playback frame counter
- * @spdif_status_bits: S/PDIF status bits
+ * @snd_aes_iec958: iec958 data
+ * @ctrl_lock: control lock
  */
 struct stm32_sai_sub_data {
 	struct platform_device *pdev;
@@ -125,7 +126,8 @@ struct stm32_sai_sub_data {
 	int slot_mask;
 	int data_size;
 	unsigned int spdif_frm_cnt;
-	unsigned char spdif_status_bits[SAI_IEC60958_STATUS_BYTES];
+	struct snd_aes_iec958 iec958;
+	struct mutex ctrl_lock; /* protect resources accessed by controls */
 };
 
 enum stm32_sai_fifo_th {
@@ -184,10 +186,6 @@ static bool stm32_sai_sub_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static const unsigned char default_status_bits[SAI_IEC60958_STATUS_BYTES] = {
-	0, 0, 0, IEC958_AES3_CON_FS_48000,
-};
-
 static const struct regmap_config stm32_sai_sub_regmap_config_f4 = {
 	.reg_bits = 32,
 	.reg_stride = 4,
@@ -210,6 +208,49 @@ static const struct regmap_config stm32_sai_sub_regmap_config_h7 = {
 	.fast_io = true,
 };
 
+static int snd_pcm_iec958_info(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+	uinfo->count = 1;
+
+	return 0;
+}
+
+static int snd_pcm_iec958_get(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *uctl)
+{
+	struct stm32_sai_sub_data *sai = snd_kcontrol_chip(kcontrol);
+
+	mutex_lock(&sai->ctrl_lock);
+	memcpy(uctl->value.iec958.status, sai->iec958.status, 4);
+	mutex_unlock(&sai->ctrl_lock);
+
+	return 0;
+}
+
+static int snd_pcm_iec958_put(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *uctl)
+{
+	struct stm32_sai_sub_data *sai = snd_kcontrol_chip(kcontrol);
+
+	mutex_lock(&sai->ctrl_lock);
+	memcpy(sai->iec958.status, uctl->value.iec958.status, 4);
+	mutex_unlock(&sai->ctrl_lock);
+
+	return 0;
+}
+
+static const struct snd_kcontrol_new iec958_ctls = {
+	.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+			SNDRV_CTL_ELEM_ACCESS_VOLATILE),
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+	.name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
+	.info = snd_pcm_iec958_info,
+	.get = snd_pcm_iec958_get,
+	.put = snd_pcm_iec958_put,
+};
+
 static irqreturn_t stm32_sai_isr(int irq, void *devid)
 {
 	struct stm32_sai_sub_data *sai = (struct stm32_sai_sub_data *)devid;
@@ -259,11 +300,8 @@ static irqreturn_t stm32_sai_isr(int irq, void *devid)
 		status = SNDRV_PCM_STATE_XRUN;
 	}
 
-	if (status != SNDRV_PCM_STATE_RUNNING) {
-		snd_pcm_stream_lock(sai->substream);
-		snd_pcm_stop(sai->substream, SNDRV_PCM_STATE_XRUN);
-		snd_pcm_stream_unlock(sai->substream);
-	}
+	if (status != SNDRV_PCM_STATE_RUNNING)
+		snd_pcm_stop_xrun(sai->substream);
 
 	return IRQ_HANDLED;
 }
@@ -619,6 +657,59 @@ static void stm32_sai_set_frame(struct snd_soc_dai *cpu_dai)
 	}
 }
 
+static void stm32_sai_init_iec958_status(struct stm32_sai_sub_data *sai)
+{
+	unsigned char *cs = sai->iec958.status;
+
+	cs[0] = IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_NONE;
+	cs[1] = IEC958_AES1_CON_GENERAL;
+	cs[2] = IEC958_AES2_CON_SOURCE_UNSPEC | IEC958_AES2_CON_CHANNEL_UNSPEC;
+	cs[3] = IEC958_AES3_CON_CLOCK_1000PPM | IEC958_AES3_CON_FS_NOTID;
+}
+
+static void stm32_sai_set_iec958_status(struct stm32_sai_sub_data *sai,
+					struct snd_pcm_runtime *runtime)
+{
+	if (!runtime)
+		return;
+
+	/* Force the sample rate according to runtime rate */
+	mutex_lock(&sai->ctrl_lock);
+	switch (runtime->rate) {
+	case 22050:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_22050;
+		break;
+	case 44100:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_44100;
+		break;
+	case 88200:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_88200;
+		break;
+	case 176400:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_176400;
+		break;
+	case 24000:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_24000;
+		break;
+	case 48000:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_48000;
+		break;
+	case 96000:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_96000;
+		break;
+	case 192000:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_192000;
+		break;
+	case 32000:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_32000;
+		break;
+	default:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_NOTID;
+		break;
+	}
+	mutex_unlock(&sai->ctrl_lock);
+}
+
 static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
 				     struct snd_pcm_hw_params *params)
 {
@@ -709,7 +800,11 @@ static int stm32_sai_hw_params(struct snd_pcm_substream *substream,
 
 	sai->data_size = params_width(params);
 
-	if (!STM_SAI_PROTOCOL_IS_SPDIF(sai)) {
+	if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) {
+		/* Rate not already set in runtime structure */
+		substream->runtime->rate = params_rate(params);
+		stm32_sai_set_iec958_status(sai, substream->runtime);
+	} else {
 		ret = stm32_sai_set_slots(cpu_dai);
 		if (ret < 0)
 			return ret;
@@ -789,6 +884,20 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream,
 	sai->substream = NULL;
 }
 
+static int stm32_sai_pcm_new(struct snd_soc_pcm_runtime *rtd,
+			     struct snd_soc_dai *cpu_dai)
+{
+	struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev);
+
+	if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) {
+		dev_dbg(&sai->pdev->dev, "%s: register iec controls", __func__);
+		return snd_ctl_add(rtd->pcm->card,
+				   snd_ctl_new1(&iec958_ctls, sai));
+	}
+
+	return 0;
+}
+
 static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 {
 	struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev);
@@ -809,6 +918,10 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 	else
 		snd_soc_dai_init_dma_data(cpu_dai, NULL, &sai->dma_params);
 
+	/* Next settings are not relevant for spdif mode */
+	if (STM_SAI_PROTOCOL_IS_SPDIF(sai))
+		return 0;
+
 	cr1_mask = SAI_XCR1_RX_TX;
 	if (STM_SAI_IS_CAPTURE(sai))
 		cr1 |= SAI_XCR1_RX_TX;
@@ -820,10 +933,6 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 				     sai->synco, sai->synci);
 	}
 
-	if (STM_SAI_PROTOCOL_IS_SPDIF(sai))
-		memcpy(sai->spdif_status_bits, default_status_bits,
-		       sizeof(default_status_bits));
-
 	cr1_mask |= SAI_XCR1_SYNCEN_MASK;
 	cr1 |= SAI_XCR1_SYNCEN_SET(sai->sync);
 
@@ -861,7 +970,7 @@ static int stm32_sai_pcm_process_spdif(struct snd_pcm_substream *substream,
 		/* Set channel status bit */
 		byte = frm_cnt >> 3;
 		mask = 1 << (frm_cnt - (byte << 3));
-		if (sai->spdif_status_bits[byte] & mask)
+		if (sai->iec958.status[byte] & mask)
 			*ptr |= 0x04000000;
 		ptr++;
 
@@ -888,6 +997,7 @@ static const struct snd_pcm_hardware stm32_sai_pcm_hw = {
 static struct snd_soc_dai_driver stm32_sai_playback_dai[] = {
 {
 		.probe = stm32_sai_dai_probe,
+		.pcm_new = stm32_sai_pcm_new,
 		.id = 1, /* avoid call to fmt_single_name() */
 		.playback = {
 			.channels_min = 1,
@@ -998,6 +1108,7 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev,
 			dev_err(&pdev->dev, "S/PDIF IEC60958 not supported\n");
 			return -EINVAL;
 		}
+		stm32_sai_init_iec958_status(sai);
 		sai->spdif = true;
 		sai->master = true;
 	}
@@ -1114,6 +1225,7 @@ static int stm32_sai_sub_probe(struct platform_device *pdev)
 	sai->id = (uintptr_t)of_id->data;
 
 	sai->pdev = pdev;
+	mutex_init(&sai->ctrl_lock);
 	platform_set_drvdata(pdev, sai);
 
 	sai->pdata = dev_get_drvdata(pdev->dev.parent);
diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c
index affad46bf188..682ef33afb5f 100644
--- a/sound/soc/tegra/tegra20_ac97.c
+++ b/sound/soc/tegra/tegra20_ac97.c
@@ -377,7 +377,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev)
 	ret = clk_prepare_enable(ac97->clk_ac97);
 	if (ret) {
 		dev_err(&pdev->dev, "clk_enable failed: %d\n", ret);
-		goto err;
+		goto err_clk_put;
 	}
 
 	ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops);
diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h
index 774fc6ad2026..2e561e946de2 100644
--- a/sound/soc/tegra/tegra30_i2s.h
+++ b/sound/soc/tegra/tegra30_i2s.h
@@ -173,7 +173,7 @@
 /* Number of slots in frame, minus 1 */
 #define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT		16
 #define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US	7
-#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK		(TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_SHIFT)
+#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK		(TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT)
 
 /* TDM mode slot enable bitmask */
 #define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT	8
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 5197d6b18cb6..98d87801d57a 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -190,14 +190,14 @@ static int tegra_alc5632_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev,
 			"Property 'nvidia,i2s-controller' missing or invalid\n");
 		ret = -EINVAL;
-		goto err;
+		goto err_put_codec_of_node;
 	}
 
 	tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node;
 
 	ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev);
 	if (ret)
-		goto err;
+		goto err_put_cpu_of_node;
 
 	ret = snd_soc_register_card(card);
 	if (ret) {
@@ -210,6 +210,13 @@ static int tegra_alc5632_probe(struct platform_device *pdev)
 
 err_fini_utils:
 	tegra_asoc_utils_fini(&alc5632->util_data);
+err_put_cpu_of_node:
+	of_node_put(tegra_alc5632_dai.cpu_of_node);
+	tegra_alc5632_dai.cpu_of_node = NULL;
+	tegra_alc5632_dai.platform_of_node = NULL;
+err_put_codec_of_node:
+	of_node_put(tegra_alc5632_dai.codec_of_node);
+	tegra_alc5632_dai.codec_of_node = NULL;
 err:
 	return ret;
 }
@@ -223,6 +230,12 @@ static int tegra_alc5632_remove(struct platform_device *pdev)
 
 	tegra_asoc_utils_fini(&machine->util_data);
 
+	of_node_put(tegra_alc5632_dai.cpu_of_node);
+	tegra_alc5632_dai.cpu_of_node = NULL;
+	tegra_alc5632_dai.platform_of_node = NULL;
+	of_node_put(tegra_alc5632_dai.codec_of_node);
+	tegra_alc5632_dai.codec_of_node = NULL;
+
 	return 0;
 }
 
diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c
index 0e4805c7b4ca..7081f15302cc 100644
--- a/sound/soc/tegra/tegra_rt5677.c
+++ b/sound/soc/tegra/tegra_rt5677.c
@@ -264,13 +264,13 @@ static int tegra_rt5677_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev,
 			"Property 'nvidia,i2s-controller' missing or invalid\n");
 		ret = -EINVAL;
-		goto err;
+		goto err_put_codec_of_node;
 	}
 	tegra_rt5677_dai.platform_of_node = tegra_rt5677_dai.cpu_of_node;
 
 	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
 	if (ret)
-		goto err;
+		goto err_put_cpu_of_node;
 
 	ret = snd_soc_register_card(card);
 	if (ret) {
@@ -283,6 +283,13 @@ static int tegra_rt5677_probe(struct platform_device *pdev)
 
 err_fini_utils:
 	tegra_asoc_utils_fini(&machine->util_data);
+err_put_cpu_of_node:
+	of_node_put(tegra_rt5677_dai.cpu_of_node);
+	tegra_rt5677_dai.cpu_of_node = NULL;
+	tegra_rt5677_dai.platform_of_node = NULL;
+err_put_codec_of_node:
+	of_node_put(tegra_rt5677_dai.codec_of_node);
+	tegra_rt5677_dai.codec_of_node = NULL;
 err:
 	return ret;
 }
@@ -296,6 +303,12 @@ static int tegra_rt5677_remove(struct platform_device *pdev)
 
 	tegra_asoc_utils_fini(&machine->util_data);
 
+	tegra_rt5677_dai.platform_of_node = NULL;
+	of_node_put(tegra_rt5677_dai.codec_of_node);
+	tegra_rt5677_dai.codec_of_node = NULL;
+	of_node_put(tegra_rt5677_dai.cpu_of_node);
+	tegra_rt5677_dai.cpu_of_node = NULL;
+
 	return 0;
 }
 
diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c
index 638cb3fc5f7b..9bcba06ba52e 100644
--- a/sound/soc/uniphier/aio-core.c
+++ b/sound/soc/uniphier/aio-core.c
@@ -265,6 +265,57 @@ void aio_port_reset(struct uniphier_aio_sub *sub)
 }
 
 /**
+ * aio_port_set_ch - set channels of LPCM
+ * @sub: the AIO substream pointer, PCM substream only
+ * @ch : count of channels
+ *
+ * Set suitable slot selecting to input/output port block of AIO.
+ *
+ * This function may return error if non-PCM substream.
+ *
+ * Return: Zero if successful, otherwise a negative value on error.
+ */
+static int aio_port_set_ch(struct uniphier_aio_sub *sub)
+{
+	struct regmap *r = sub->aio->chip->regmap;
+	u32 slotsel_2ch[] = {
+		0, 0, 0, 0, 0,
+	};
+	u32 slotsel_multi[] = {
+		OPORTMXTYSLOTCTR_SLOTSEL_SLOT0,
+		OPORTMXTYSLOTCTR_SLOTSEL_SLOT1,
+		OPORTMXTYSLOTCTR_SLOTSEL_SLOT2,
+		OPORTMXTYSLOTCTR_SLOTSEL_SLOT3,
+		OPORTMXTYSLOTCTR_SLOTSEL_SLOT4,
+	};
+	u32 mode, *slotsel;
+	int i;
+
+	switch (params_channels(&sub->params)) {
+	case 8:
+	case 6:
+		mode = OPORTMXTYSLOTCTR_MODE;
+		slotsel = slotsel_multi;
+		break;
+	case 2:
+		mode = 0;
+		slotsel = slotsel_2ch;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	for (i = 0; i < AUD_MAX_SLOTSEL; i++) {
+		regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i),
+				   OPORTMXTYSLOTCTR_MODE, mode);
+		regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i),
+				   OPORTMXTYSLOTCTR_SLOTSEL_MASK, slotsel[i]);
+	}
+
+	return 0;
+}
+
+/**
  * aio_port_set_rate - set sampling rate of LPCM
  * @sub: the AIO substream pointer, PCM substream only
  * @rate: Sampling rate in Hz.
@@ -276,7 +327,7 @@ void aio_port_reset(struct uniphier_aio_sub *sub)
  *
  * Return: Zero if successful, otherwise a negative value on error.
  */
-int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate)
+static int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate)
 {
 	struct regmap *r = sub->aio->chip->regmap;
 	struct device *dev = &sub->aio->chip->pdev->dev;
@@ -395,7 +446,7 @@ int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate)
  *
  * Return: Zero if successful, otherwise a negative value on error.
  */
-int aio_port_set_fmt(struct uniphier_aio_sub *sub)
+static int aio_port_set_fmt(struct uniphier_aio_sub *sub)
 {
 	struct regmap *r = sub->aio->chip->regmap;
 	struct device *dev = &sub->aio->chip->pdev->dev;
@@ -460,7 +511,7 @@ int aio_port_set_fmt(struct uniphier_aio_sub *sub)
  *
  * Return: Zero if successful, otherwise a negative value on error.
  */
-int aio_port_set_clk(struct uniphier_aio_sub *sub)
+static int aio_port_set_clk(struct uniphier_aio_sub *sub)
 {
 	struct uniphier_aio_chip *chip = sub->aio->chip;
 	struct device *dev = &sub->aio->chip->pdev->dev;
@@ -575,6 +626,10 @@ int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through,
 			rate = params_rate(params);
 		}
 
+		ret = aio_port_set_ch(sub);
+		if (ret)
+			return ret;
+
 		ret = aio_port_set_rate(sub, rate);
 		if (ret)
 			return ret;
@@ -731,15 +786,28 @@ void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol)
 int aio_if_set_param(struct uniphier_aio_sub *sub, int pass_through)
 {
 	struct regmap *r = sub->aio->chip->regmap;
-	u32 v;
+	u32 memfmt, v;
 
 	if (sub->swm->dir == PORT_DIR_OUTPUT) {
-		if (pass_through)
+		if (pass_through) {
 			v = PBOUTMXCTR0_ENDIAN_0123 |
 				PBOUTMXCTR0_MEMFMT_STREAM;
-		else
-			v = PBOUTMXCTR0_ENDIAN_3210 |
-				PBOUTMXCTR0_MEMFMT_2CH;
+		} else {
+			switch (params_channels(&sub->params)) {
+			case 2:
+				memfmt = PBOUTMXCTR0_MEMFMT_2CH;
+				break;
+			case 6:
+				memfmt = PBOUTMXCTR0_MEMFMT_6CH;
+				break;
+			case 8:
+				memfmt = PBOUTMXCTR0_MEMFMT_8CH;
+				break;
+			default:
+				return -EINVAL;
+			}
+			v = PBOUTMXCTR0_ENDIAN_3210 | memfmt;
+		}
 
 		regmap_write(r, PBOUTMXCTR0(sub->swm->oif.map), v);
 		regmap_write(r, PBOUTMXCTR1(sub->swm->oif.map), 0);
diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c
index 2d9b7dde2ffa..ee90e6c3937c 100644
--- a/sound/soc/uniphier/aio-cpu.c
+++ b/sound/soc/uniphier/aio-cpu.c
@@ -219,15 +219,10 @@ static int uniphier_aio_set_pll(struct snd_soc_dai *dai, int pll_id,
 				unsigned int freq_out)
 {
 	struct uniphier_aio *aio = uniphier_priv(dai);
-	struct device *dev = &aio->chip->pdev->dev;
 	int ret;
 
 	if (!is_valid_pll(aio->chip, pll_id))
 		return -EINVAL;
-	if (!aio->chip->plls[pll_id].enable) {
-		dev_err(dev, "PLL(%d) is not implemented\n", pll_id);
-		return -ENOTSUPP;
-	}
 
 	ret = aio_chip_set_pll(aio->chip, pll_id, freq_out);
 	if (ret < 0)
diff --git a/sound/soc/uniphier/aio-ld11.c b/sound/soc/uniphier/aio-ld11.c
index ab04d3331be9..de962df245ba 100644
--- a/sound/soc/uniphier/aio-ld11.c
+++ b/sound/soc/uniphier/aio-ld11.c
@@ -286,7 +286,7 @@ static struct snd_soc_dai_driver uniphier_aio_dai_ld11[] = {
 			.formats     = SNDRV_PCM_FMTBIT_S32_LE,
 			.rates       = SNDRV_PCM_RATE_48000,
 			.channels_min = 2,
-			.channels_max = 2,
+			.channels_max = 8,
 		},
 		.ops = &uniphier_aio_i2s_ops,
 	},
diff --git a/sound/soc/uniphier/aio-reg.h b/sound/soc/uniphier/aio-reg.h
index 45fdc6ae358a..734395dbcffb 100644
--- a/sound/soc/uniphier/aio-reg.h
+++ b/sound/soc/uniphier/aio-reg.h
@@ -374,6 +374,7 @@
 #define OPORTMXTYVOLGAINSTATUS(n, m)     (0x42108 + 0x400 * (n) + 0x20 * (m))
 #define   OPORTMXTYVOLGAINSTATUS_CUR_MASK  GENMASK(15, 0)
 #define OPORTMXTYSLOTCTR(n, m)           (0x42114 + 0x400 * (n) + 0x20 * (m))
+#define   OPORTMXTYSLOTCTR_MODE            BIT(15)
 #define   OPORTMXTYSLOTCTR_SLOTSEL_MASK    GENMASK(11, 8)
 #define   OPORTMXTYSLOTCTR_SLOTSEL_SLOT0   (0x8 << 8)
 #define   OPORTMXTYSLOTCTR_SLOTSEL_SLOT1   (0x9 << 8)
diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h
index aa89c2f6fa24..ca6ccbae0ee8 100644
--- a/sound/soc/uniphier/aio.h
+++ b/sound/soc/uniphier/aio.h
@@ -141,6 +141,9 @@ enum IEC61937_PC {
 #define AUD_MIN_FRAGMENT_SIZE    (4 * 1024)
 #define AUD_MAX_FRAGMENT_SIZE    (16 * 1024)
 
+/* max 5 slots, 10 channels, 2 channel in 1 slot */
+#define AUD_MAX_SLOTSEL    5
+
 /*
  * This is a selector for virtual register map of AIO.
  *
@@ -322,9 +325,6 @@ int aio_chip_set_pll(struct uniphier_aio_chip *chip, int pll_id,
 void aio_chip_init(struct uniphier_aio_chip *chip);
 int aio_init(struct uniphier_aio_sub *sub);
 void aio_port_reset(struct uniphier_aio_sub *sub);
-int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate);
-int aio_port_set_fmt(struct uniphier_aio_sub *sub);
-int aio_port_set_clk(struct uniphier_aio_sub *sub);
 int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through,
 		       const struct snd_pcm_hw_params *params);
 void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable);
diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c
index dc955272f58b..389272eeba9a 100644
--- a/sound/soc/zte/zx-tdm.c
+++ b/sound/soc/zte/zx-tdm.c
@@ -144,8 +144,8 @@ static void zx_tdm_rx_dma_en(struct zx_tdm_info *tdm, bool on)
 #define ZX_TDM_RATES	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000)
 
 #define ZX_TDM_FMTBIT \
-	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_MU_LAW | \
-	SNDRV_PCM_FORMAT_A_LAW)
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_MU_LAW | \
+	SNDRV_PCM_FMTBIT_A_LAW)
 
 static int zx_tdm_dai_probe(struct snd_soc_dai *dai)
 {
diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c
index b840ff2dcfbb..64f3141a3e1b 100644
--- a/sound/synth/emux/emux.c
+++ b/sound/synth/emux/emux.c
@@ -163,20 +163,3 @@ int snd_emux_free(struct snd_emux *emu)
 }
 
 EXPORT_SYMBOL(snd_emux_free);
-
-
-/*
- *  INIT part
- */
-
-static int __init alsa_emux_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_emux_exit(void)
-{
-}
-
-module_init(alsa_emux_init)
-module_exit(alsa_emux_exit)
diff --git a/sound/synth/util_mem.c b/sound/synth/util_mem.c
index 8e34bc4e07ec..4bd1e98200d2 100644
--- a/sound/synth/util_mem.c
+++ b/sound/synth/util_mem.c
@@ -193,19 +193,3 @@ EXPORT_SYMBOL(snd_util_mem_avail);
 EXPORT_SYMBOL(__snd_util_mem_alloc);
 EXPORT_SYMBOL(__snd_util_mem_free);
 EXPORT_SYMBOL(__snd_util_memblk_new);
-
-/*
- *  INIT part
- */
-
-static int __init alsa_util_mem_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_util_mem_exit(void)
-{
-}
-
-module_init(alsa_util_mem_init)
-module_exit(alsa_util_mem_exit)
diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c
index 2dd2518a71d3..f8ef3e2a8ca0 100644
--- a/sound/usb/6fire/pcm.c
+++ b/sound/usb/6fire/pcm.c
@@ -565,7 +565,6 @@ static const struct snd_pcm_ops pcm_ops = {
 	.trigger = usb6fire_pcm_trigger,
 	.pointer = usb6fire_pcm_pointer,
 	.page = snd_pcm_lib_get_vmalloc_page,
-	.mmap = snd_pcm_lib_mmap_vmalloc,
 };
 
 static void usb6fire_pcm_init_urb(struct pcm_urb *urb,
diff --git a/sound/usb/Makefile b/sound/usb/Makefile
index 05440e2df8d9..d330f74c90e6 100644
--- a/sound/usb/Makefile
+++ b/sound/usb/Makefile
@@ -13,6 +13,7 @@ snd-usb-audio-objs := 	card.o \
 			mixer_scarlett.o \
 			mixer_us16x08.o \
 			pcm.o \
+			power.o \
 			proc.o \
 			quirks.o \
 			stream.o
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index f35d29f49ffe..c6108a3d7f8f 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -348,7 +348,6 @@ static const struct snd_pcm_ops snd_usb_caiaq_ops = {
 	.trigger =	snd_usb_caiaq_pcm_trigger,
 	.pointer =	snd_usb_caiaq_pcm_pointer,
 	.page =		snd_pcm_lib_get_vmalloc_page,
-	.mmap =		snd_pcm_lib_mmap_vmalloc,
 };
 
 static void check_for_elapsed_periods(struct snd_usb_caiaqdev *cdev,
@@ -636,6 +635,7 @@ static void read_completed(struct urb *urb)
 	struct device *dev;
 	struct urb *out = NULL;
 	int i, frame, len, send_it = 0, outframe = 0;
+	unsigned long flags;
 	size_t offset = 0;
 
 	if (urb->status || !info)
@@ -672,10 +672,10 @@ static void read_completed(struct urb *urb)
 		offset += len;
 
 		if (len > 0) {
-			spin_lock(&cdev->spinlock);
+			spin_lock_irqsave(&cdev->spinlock, flags);
 			fill_out_urb(cdev, out, &out->iso_frame_desc[outframe]);
 			read_in_urb(cdev, urb, &urb->iso_frame_desc[frame]);
-			spin_unlock(&cdev->spinlock);
+			spin_unlock_irqrestore(&cdev->spinlock, flags);
 			check_for_elapsed_periods(cdev, cdev->sub_playback);
 			check_for_elapsed_periods(cdev, cdev->sub_capture);
 			send_it = 1;
diff --git a/sound/usb/card.c b/sound/usb/card.c
index a1ed798a1c6b..2bfe4e80a6b9 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -809,6 +809,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
 	if (!chip->num_suspended_intf++) {
 		list_for_each_entry(as, &chip->pcm_list, list) {
 			snd_pcm_suspend_all(as->pcm);
+			snd_usb_pcm_suspend(as);
 			as->substream[0].need_setup_ep =
 				as->substream[1].need_setup_ep = true;
 		}
@@ -824,6 +825,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
 static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
 {
 	struct snd_usb_audio *chip = usb_get_intfdata(intf);
+	struct snd_usb_stream *as;
 	struct usb_mixer_interface *mixer;
 	struct list_head *p;
 	int err = 0;
@@ -834,6 +836,13 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
 		return 0;
 
 	atomic_inc(&chip->active); /* avoid autopm */
+
+	list_for_each_entry(as, &chip->pcm_list, list) {
+		err = snd_usb_pcm_resume(as);
+		if (err < 0)
+			goto err_out;
+	}
+
 	/*
 	 * ALSA leaves material resumption to user space
 	 * we just notify and restart the mixers
diff --git a/sound/usb/card.h b/sound/usb/card.h
index 9b41b7dda84f..ac785d15ced4 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -37,6 +37,7 @@ struct audioformat {
 
 struct snd_usb_substream;
 struct snd_usb_endpoint;
+struct snd_usb_power_domain;
 
 struct snd_urb_ctx {
 	struct urb *urb;
@@ -115,6 +116,7 @@ struct snd_usb_substream {
 	int interface;	/* current interface */
 	int endpoint;	/* assigned endpoint */
 	struct audioformat *cur_audiofmt;	/* current audioformat pointer (for hw_params callback) */
+	struct snd_usb_power_domain *str_pd;	/* UAC3 Power Domain for streaming path */
 	snd_pcm_format_t pcm_format;	/* current audio format (for hw_params callback) */
 	unsigned int channels;		/* current number of channels (for hw_params callback) */
 	unsigned int channels_max;	/* max channels in the all audiofmts */
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index c79749613fa6..db5e39d67a90 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -513,14 +513,28 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface,
 	bool writeable;
 	u32 bmControls;
 
+	/* First, try to find a valid clock. This may trigger
+	 * automatic clock selection if the current clock is not
+	 * valid.
+	 */
 	clock = snd_usb_clock_find_source(chip, fmt->protocol,
 					  fmt->clock, true);
-	if (clock < 0)
-		return clock;
+	if (clock < 0) {
+		/* We did not find a valid clock, but that might be
+		 * because the current sample rate does not match an
+		 * external clock source. Try again without validation
+		 * and we will do another validation after setting the
+		 * rate.
+		 */
+		clock = snd_usb_clock_find_source(chip, fmt->protocol,
+						  fmt->clock, false);
+		if (clock < 0)
+			return clock;
+	}
 
 	prev_rate = get_sample_rate_v2v3(chip, iface, fmt->altsetting, clock);
 	if (prev_rate == rate)
-		return 0;
+		goto validation;
 
 	if (fmt->protocol == UAC_VERSION_3) {
 		struct uac3_clock_source_descriptor *cs_desc;
@@ -577,6 +591,10 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface,
 		snd_usb_set_interface_quirk(dev);
 	}
 
+validation:
+	/* validate clock after rate change */
+	if (!uac_clock_source_is_valid(chip, fmt->protocol, clock))
+		return -ENXIO;
 	return 0;
 }
 
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index c90607ebe155..d86be8bfe412 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -325,7 +325,6 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep)
 		unsigned long flags;
 		struct snd_usb_packet_info *uninitialized_var(packet);
 		struct snd_urb_ctx *ctx = NULL;
-		struct urb *urb;
 		int err, i;
 
 		spin_lock_irqsave(&ep->lock, flags);
@@ -345,7 +344,6 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep)
 			return;
 
 		list_del_init(&ctx->ready_list);
-		urb = ctx->urb;
 
 		/* copy over the length information */
 		for (i = 0; i < packet->packets; i++)
diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c
index 396c317115b1..e1fbb9cc9ea7 100644
--- a/sound/usb/hiface/pcm.c
+++ b/sound/usb/hiface/pcm.c
@@ -523,7 +523,6 @@ static const struct snd_pcm_ops pcm_ops = {
 	.trigger = hiface_pcm_trigger,
 	.pointer = hiface_pcm_pointer,
 	.page = snd_pcm_lib_get_vmalloc_page,
-	.mmap = snd_pcm_lib_mmap_vmalloc,
 };
 
 static int hiface_pcm_init_urb(struct pcm_urb *urb,
diff --git a/sound/usb/line6/toneport.c b/sound/usb/line6/toneport.c
index 750467fb95db..f47ba94e6f4a 100644
--- a/sound/usb/line6/toneport.c
+++ b/sound/usb/line6/toneport.c
@@ -367,12 +367,13 @@ static bool toneport_has_source_select(struct usb_line6_toneport *toneport)
 */
 static void toneport_setup(struct usb_line6_toneport *toneport)
 {
-	int ticks;
+	u32 ticks;
 	struct usb_line6 *line6 = &toneport->line6;
 	struct usb_device *usbdev = line6->usbdev;
 
 	/* sync time on device with host: */
-	ticks = (int)get_seconds();
+	/* note: 32-bit timestamps overflow in year 2106 */
+	ticks = (u32)ktime_get_real_seconds();
 	line6_write_data(line6, 0x80c6, &ticks, 4);
 
 	/* enable device: */
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 2c1aaa3292bf..dcfc546d81b9 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -281,15 +281,16 @@ static void snd_usbmidi_out_urb_complete(struct urb *urb)
 	struct out_urb_context *context = urb->context;
 	struct snd_usb_midi_out_endpoint *ep = context->ep;
 	unsigned int urb_index;
+	unsigned long flags;
 
-	spin_lock(&ep->buffer_lock);
+	spin_lock_irqsave(&ep->buffer_lock, flags);
 	urb_index = context - ep->urbs;
 	ep->active_urbs &= ~(1 << urb_index);
 	if (unlikely(ep->drain_urbs)) {
 		ep->drain_urbs &= ~(1 << urb_index);
 		wake_up(&ep->drain_wait);
 	}
-	spin_unlock(&ep->buffer_lock);
+	spin_unlock_irqrestore(&ep->buffer_lock, flags);
 	if (urb->status < 0) {
 		int err = snd_usbmidi_urb_error(urb);
 		if (err < 0) {
diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c
index 386fbfd5c617..a0b6d039017f 100644
--- a/sound/usb/misc/ua101.c
+++ b/sound/usb/misc/ua101.c
@@ -900,7 +900,6 @@ static const struct snd_pcm_ops capture_pcm_ops = {
 	.trigger = capture_pcm_trigger,
 	.pointer = capture_pcm_pointer,
 	.page = snd_pcm_lib_get_vmalloc_page,
-	.mmap = snd_pcm_lib_mmap_vmalloc,
 };
 
 static const struct snd_pcm_ops playback_pcm_ops = {
@@ -913,7 +912,6 @@ static const struct snd_pcm_ops playback_pcm_ops = {
 	.trigger = playback_pcm_trigger,
 	.pointer = playback_pcm_pointer,
 	.page = snd_pcm_lib_get_vmalloc_page,
-	.mmap = snd_pcm_lib_mmap_vmalloc,
 };
 
 static const struct uac_format_type_i_discrete_descriptor *
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index ca963e94ec03..c63c84b54969 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -675,16 +675,16 @@ static int get_term_name(struct snd_usb_audio *chip, struct usb_audio_term *iter
 		if (term_only)
 			return 0;
 		switch (iterm->type >> 16) {
-		case UAC_SELECTOR_UNIT:
+		case UAC3_SELECTOR_UNIT:
 			strcpy(name, "Selector");
 			return 8;
-		case UAC1_PROCESSING_UNIT:
+		case UAC3_PROCESSING_UNIT:
 			strcpy(name, "Process Unit");
 			return 12;
-		case UAC1_EXTENSION_UNIT:
+		case UAC3_EXTENSION_UNIT:
 			strcpy(name, "Ext Unit");
 			return 8;
-		case UAC_MIXER_UNIT:
+		case UAC3_MIXER_UNIT:
 			strcpy(name, "Mixer");
 			return 5;
 		default:
@@ -832,7 +832,7 @@ static int check_input_term(struct mixer_build *state, int id,
 			case UAC_MIXER_UNIT: {
 				struct uac_mixer_unit_descriptor *d = p1;
 
-				term->type = d->bDescriptorSubtype << 16; /* virtual type */
+				term->type = UAC3_MIXER_UNIT << 16; /* virtual type */
 				term->channels = uac_mixer_unit_bNrChannels(d);
 				term->chconfig = uac_mixer_unit_wChannelConfig(d, protocol);
 				term->name = uac_mixer_unit_iMixer(d);
@@ -845,15 +845,25 @@ static int check_input_term(struct mixer_build *state, int id,
 				err = check_input_term(state, d->baSourceID[0], term);
 				if (err < 0)
 					return err;
-				term->type = d->bDescriptorSubtype << 16; /* virtual type */
+				term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */
 				term->id = id;
 				term->name = uac_selector_unit_iSelector(d);
 				return 0;
 			}
 			case UAC1_PROCESSING_UNIT:
+			/* UAC2_EFFECT_UNIT */
+				if (protocol == UAC_VERSION_1)
+					term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */
+				else /* UAC_VERSION_2 */
+					term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */
+				/* fall through */
 			case UAC1_EXTENSION_UNIT:
 			/* UAC2_PROCESSING_UNIT_V2 */
-			/* UAC2_EFFECT_UNIT */
+				if (protocol == UAC_VERSION_1 && !term->type)
+					term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */
+				else if (protocol == UAC_VERSION_2 && !term->type)
+					term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */
+				/* fall through */
 			case UAC2_EXTENSION_UNIT_V2: {
 				struct uac_processing_unit_descriptor *d = p1;
 
@@ -869,7 +879,9 @@ static int check_input_term(struct mixer_build *state, int id,
 					id = d->baSourceID[0];
 					break; /* continue to parse */
 				}
-				term->type = d->bDescriptorSubtype << 16; /* virtual type */
+				if (!term->type)
+					term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */
+
 				term->channels = uac_processing_unit_bNrChannels(d);
 				term->chconfig = uac_processing_unit_wChannelConfig(d, protocol);
 				term->name = uac_processing_unit_iProcessing(d, protocol);
@@ -878,7 +890,7 @@ static int check_input_term(struct mixer_build *state, int id,
 			case UAC2_CLOCK_SOURCE: {
 				struct uac_clock_source_descriptor *d = p1;
 
-				term->type = d->bDescriptorSubtype << 16; /* virtual type */
+				term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */
 				term->id = id;
 				term->name = d->iClockSource;
 				return 0;
@@ -923,7 +935,7 @@ static int check_input_term(struct mixer_build *state, int id,
 			case UAC3_CLOCK_SOURCE: {
 				struct uac3_clock_source_descriptor *d = p1;
 
-				term->type = d->bDescriptorSubtype << 16; /* virtual type */
+				term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */
 				term->id = id;
 				term->name = le16_to_cpu(d->wClockSourceStr);
 				return 0;
@@ -936,7 +948,37 @@ static int check_input_term(struct mixer_build *state, int id,
 					return err;
 
 				term->channels = err;
-				term->type = d->bDescriptorSubtype << 16; /* virtual type */
+				term->type = UAC3_MIXER_UNIT << 16; /* virtual type */
+
+				return 0;
+			}
+			case UAC3_SELECTOR_UNIT:
+			case UAC3_CLOCK_SELECTOR: {
+				struct uac_selector_unit_descriptor *d = p1;
+				/* call recursively to retrieve the channel info */
+				err = check_input_term(state, d->baSourceID[0], term);
+				if (err < 0)
+					return err;
+				term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */
+				term->id = id;
+				term->name = 0; /* TODO: UAC3 Class-specific strings */
+
+				return 0;
+			}
+			case UAC3_PROCESSING_UNIT: {
+				struct uac_processing_unit_descriptor *d = p1;
+
+				if (!d->bNrInPins)
+					return -EINVAL;
+
+				/* call recursively to retrieve the channel info */
+				err = check_input_term(state, d->baSourceID[0], term);
+				if (err < 0)
+					return err;
+
+				term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */
+				term->id = id;
+				term->name = 0; /* TODO: UAC3 Class-specific strings */
 
 				return 0;
 			}
@@ -2167,6 +2209,11 @@ struct procunit_info {
 	struct procunit_value_info *values;
 };
 
+static struct procunit_value_info undefined_proc_info[] = {
+	{ 0x00, "Control Undefined", 0 },
+	{ 0 }
+};
+
 static struct procunit_value_info updown_proc_info[] = {
 	{ UAC_UD_ENABLE, "Switch", USB_MIXER_BOOLEAN },
 	{ UAC_UD_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 },
@@ -2215,6 +2262,23 @@ static struct procunit_info procunits[] = {
 	{ UAC_PROCESS_DYN_RANGE_COMP, "DCR", dcr_proc_info },
 	{ 0 },
 };
+
+static struct procunit_value_info uac3_updown_proc_info[] = {
+	{ UAC3_UD_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 },
+	{ 0 }
+};
+static struct procunit_value_info uac3_stereo_ext_proc_info[] = {
+	{ UAC3_EXT_WIDTH_CONTROL, "Width Control", USB_MIXER_U8 },
+	{ 0 }
+};
+
+static struct procunit_info uac3_procunits[] = {
+	{ UAC3_PROCESS_UP_DOWNMIX, "Up Down", uac3_updown_proc_info },
+	{ UAC3_PROCESS_STEREO_EXTENDER, "3D Stereo Extender", uac3_stereo_ext_proc_info },
+	{ UAC3_PROCESS_MULTI_FUNCTION, "Multi-Function", undefined_proc_info },
+	{ 0 },
+};
+
 /*
  * predefined data for extension units
  */
@@ -2287,8 +2351,16 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 	for (valinfo = info->values; valinfo->control; valinfo++) {
 		__u8 *controls = uac_processing_unit_bmControls(desc, state->mixer->protocol);
 
-		if (!(controls[valinfo->control / 8] & (1 << ((valinfo->control % 8) - 1))))
-			continue;
+		if (state->mixer->protocol == UAC_VERSION_1) {
+			if (!(controls[valinfo->control / 8] &
+					(1 << ((valinfo->control % 8) - 1))))
+				continue;
+		} else { /* UAC_VERSION_2/3 */
+			if (!uac_v2v3_control_is_readable(controls[valinfo->control / 8],
+							  valinfo->control))
+				continue;
+		}
+
 		map = find_map(state->map, unitid, valinfo->control);
 		if (check_ignored_ctl(map))
 			continue;
@@ -2300,26 +2372,55 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 		cval->val_type = valinfo->val_type;
 		cval->channels = 1;
 
+		if (state->mixer->protocol > UAC_VERSION_1 &&
+		    !uac_v2v3_control_is_writeable(controls[valinfo->control / 8],
+						   valinfo->control))
+			cval->master_readonly = 1;
+
 		/* get min/max values */
-		if (type == UAC_PROCESS_UP_DOWNMIX && cval->control == UAC_UD_MODE_SELECT) {
-			__u8 *control_spec = uac_processing_unit_specific(desc, state->mixer->protocol);
-			/* FIXME: hard-coded */
-			cval->min = 1;
-			cval->max = control_spec[0];
-			cval->res = 1;
-			cval->initialized = 1;
-		} else {
-			if (type == USB_XU_CLOCK_RATE) {
-				/*
-				 * E-Mu USB 0404/0202/TrackerPre/0204
-				 * samplerate control quirk
-				 */
-				cval->min = 0;
-				cval->max = 5;
+		switch (type) {
+		case UAC_PROCESS_UP_DOWNMIX: {
+			bool mode_sel = false;
+
+			switch (state->mixer->protocol) {
+			case UAC_VERSION_1:
+			case UAC_VERSION_2:
+			default:
+				if (cval->control == UAC_UD_MODE_SELECT)
+					mode_sel = true;
+				break;
+			case UAC_VERSION_3:
+				if (cval->control == UAC3_UD_MODE_SELECT)
+					mode_sel = true;
+				break;
+			}
+
+			if (mode_sel) {
+				__u8 *control_spec = uac_processing_unit_specific(desc,
+								state->mixer->protocol);
+				cval->min = 1;
+				cval->max = control_spec[0];
 				cval->res = 1;
 				cval->initialized = 1;
-			} else
-				get_min_max(cval, valinfo->min_value);
+				break;
+			}
+
+			get_min_max(cval, valinfo->min_value);
+			break;
+		}
+		case USB_XU_CLOCK_RATE:
+			/*
+			 * E-Mu USB 0404/0202/TrackerPre/0204
+			 * samplerate control quirk
+			 */
+			cval->min = 0;
+			cval->max = 5;
+			cval->res = 1;
+			cval->initialized = 1;
+			break;
+		default:
+			get_min_max(cval, valinfo->min_value);
+			break;
 		}
 
 		kctl = snd_ctl_new1(&mixer_procunit_ctl, cval);
@@ -2362,8 +2463,16 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 static int parse_audio_processing_unit(struct mixer_build *state, int unitid,
 				       void *raw_desc)
 {
-	return build_audio_procunit(state, unitid, raw_desc,
-				    procunits, "Processing Unit");
+	switch (state->mixer->protocol) {
+	case UAC_VERSION_1:
+	case UAC_VERSION_2:
+	default:
+		return build_audio_procunit(state, unitid, raw_desc,
+				procunits, "Processing Unit");
+	case UAC_VERSION_3:
+		return build_audio_procunit(state, unitid, raw_desc,
+				uac3_procunits, "Processing Unit");
+	}
 }
 
 static int parse_audio_extension_unit(struct mixer_build *state, int unitid,
@@ -2509,11 +2618,20 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 	cval->res = 1;
 	cval->initialized = 1;
 
-	if (state->mixer->protocol == UAC_VERSION_1)
+	switch (state->mixer->protocol) {
+	case UAC_VERSION_1:
+	default:
 		cval->control = 0;
-	else /* UAC_VERSION_2 */
-		cval->control = (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) ?
-			UAC2_CX_CLOCK_SELECTOR : UAC2_SU_SELECTOR;
+		break;
+	case UAC_VERSION_2:
+	case UAC_VERSION_3:
+		if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR ||
+		    desc->bDescriptorSubtype == UAC3_CLOCK_SELECTOR)
+			cval->control = UAC2_CX_CLOCK_SELECTOR;
+		else /* UAC2/3_SELECTOR_UNIT */
+			cval->control = UAC2_SU_SELECTOR;
+		break;
+	}
 
 	namelist = kmalloc_array(desc->bNrInPins, sizeof(char *), GFP_KERNEL);
 	if (!namelist) {
@@ -2555,12 +2673,22 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 	len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name));
 	if (!len) {
 		/* no mapping ? */
+		switch (state->mixer->protocol) {
+		case UAC_VERSION_1:
+		case UAC_VERSION_2:
+		default:
 		/* if iSelector is given, use it */
-		nameid = uac_selector_unit_iSelector(desc);
-		if (nameid)
-			len = snd_usb_copy_string_desc(state->chip, nameid,
-						       kctl->id.name,
-						       sizeof(kctl->id.name));
+			nameid = uac_selector_unit_iSelector(desc);
+			if (nameid)
+				len = snd_usb_copy_string_desc(state->chip,
+							nameid, kctl->id.name,
+							sizeof(kctl->id.name));
+			break;
+		case UAC_VERSION_3:
+			/* TODO: Class-Specific strings not yet supported */
+			break;
+		}
+
 		/* ... or pick up the terminal name at next */
 		if (!len)
 			len = get_term_name(state->chip, &state->oterm,
@@ -2570,7 +2698,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 			strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name));
 
 		/* and add the proper suffix */
-		if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR)
+		if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR ||
+		    desc->bDescriptorSubtype == UAC3_CLOCK_SELECTOR)
 			append_ctl_name(kctl, " Clock Source");
 		else if ((state->oterm.type & 0xff00) == 0x0100)
 			append_ctl_name(kctl, " Capture Source");
@@ -2641,6 +2770,7 @@ static int parse_audio_unit(struct mixer_build *state, int unitid)
 			return parse_audio_mixer_unit(state, unitid, p1);
 		case UAC3_CLOCK_SOURCE:
 			return parse_clock_source_unit(state, unitid, p1);
+		case UAC3_SELECTOR_UNIT:
 		case UAC3_CLOCK_SELECTOR:
 			return parse_audio_selector_unit(state, unitid, p1);
 		case UAC3_FEATURE_UNIT:
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index e02653465e29..3d12af8bf191 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -109,4 +109,6 @@ int snd_usb_get_cur_mix_value(struct usb_mixer_elem_info *cval,
 
 extern void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl);
 
+extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl;
+
 #endif /* __USBMIXER_H */
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index e82a72fea9a1..cbfb48bdea51 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -47,8 +47,6 @@
 #include "mixer_us16x08.h"
 #include "helper.h"
 
-extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl;
-
 struct std_mono_table {
 	unsigned int unitid, control, cmask;
 	int val_type;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 160f52c4871b..382847154227 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -711,6 +711,54 @@ static int configure_endpoint(struct snd_usb_substream *subs)
 	return ret;
 }
 
+static int snd_usb_pcm_change_state(struct snd_usb_substream *subs, int state)
+{
+	int ret;
+
+	if (!subs->str_pd)
+		return 0;
+
+	ret = snd_usb_power_domain_set(subs->stream->chip, subs->str_pd, state);
+	if (ret < 0) {
+		dev_err(&subs->dev->dev,
+			"Cannot change Power Domain ID: %d to state: %d. Err: %d\n",
+			subs->str_pd->pd_id, state, ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+int snd_usb_pcm_suspend(struct snd_usb_stream *as)
+{
+	int ret;
+
+	ret = snd_usb_pcm_change_state(&as->substream[0], UAC3_PD_STATE_D2);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_usb_pcm_change_state(&as->substream[1], UAC3_PD_STATE_D2);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+int snd_usb_pcm_resume(struct snd_usb_stream *as)
+{
+	int ret;
+
+	ret = snd_usb_pcm_change_state(&as->substream[0], UAC3_PD_STATE_D1);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_usb_pcm_change_state(&as->substream[1], UAC3_PD_STATE_D1);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
 /*
  * hw_params callback
  *
@@ -755,16 +803,22 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
 	ret = snd_usb_lock_shutdown(subs->stream->chip);
 	if (ret < 0)
 		return ret;
+
+	ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D0);
+	if (ret < 0)
+		goto unlock;
+
 	ret = set_format(subs, fmt);
-	snd_usb_unlock_shutdown(subs->stream->chip);
 	if (ret < 0)
-		return ret;
+		goto unlock;
 
 	subs->interface = fmt->iface;
 	subs->altset_idx = fmt->altset_idx;
 	subs->need_setup_ep = true;
 
-	return 0;
+ unlock:
+	snd_usb_unlock_shutdown(subs->stream->chip);
+	return ret;
 }
 
 /*
@@ -821,6 +875,10 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
 	snd_usb_endpoint_sync_pending_stop(subs->sync_endpoint);
 	snd_usb_endpoint_sync_pending_stop(subs->data_endpoint);
 
+	ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D0);
+	if (ret < 0)
+		goto unlock;
+
 	ret = set_format(subs, subs->cur_audiofmt);
 	if (ret < 0)
 		goto unlock;
@@ -1265,6 +1323,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream)
 	int direction = substream->stream;
 	struct snd_usb_stream *as = snd_pcm_substream_chip(substream);
 	struct snd_usb_substream *subs = &as->substream[direction];
+	int ret;
 
 	stop_endpoints(subs, true);
 
@@ -1273,7 +1332,10 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream)
 	    !snd_usb_lock_shutdown(subs->stream->chip)) {
 		usb_set_interface(subs->dev, subs->interface, 0);
 		subs->interface = -1;
+		ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D1);
 		snd_usb_unlock_shutdown(subs->stream->chip);
+		if (ret < 0)
+			return ret;
 	}
 
 	subs->pcm_substream = NULL;
@@ -1632,6 +1694,7 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		subs->trigger_tstamp_pending_update = true;
+		/* fall through */
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		subs->data_endpoint->prepare_data_urb = prepare_playback_urb;
 		subs->data_endpoint->retire_data_urb = retire_playback_urb;
@@ -1694,7 +1757,6 @@ static const struct snd_pcm_ops snd_usb_playback_ops = {
 	.trigger =	snd_usb_substream_playback_trigger,
 	.pointer =	snd_usb_pcm_pointer,
 	.page =		snd_pcm_lib_get_vmalloc_page,
-	.mmap =		snd_pcm_lib_mmap_vmalloc,
 };
 
 static const struct snd_pcm_ops snd_usb_capture_ops = {
@@ -1707,7 +1769,6 @@ static const struct snd_pcm_ops snd_usb_capture_ops = {
 	.trigger =	snd_usb_substream_capture_trigger,
 	.pointer =	snd_usb_pcm_pointer,
 	.page =		snd_pcm_lib_get_vmalloc_page,
-	.mmap =		snd_pcm_lib_mmap_vmalloc,
 };
 
 static const struct snd_pcm_ops snd_usb_playback_dev_ops = {
diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h
index f77ec58bf1a1..9833627c1eca 100644
--- a/sound/usb/pcm.h
+++ b/sound/usb/pcm.h
@@ -6,6 +6,8 @@ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs,
 				    unsigned int rate);
 
 void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream);
+int snd_usb_pcm_suspend(struct snd_usb_stream *as);
+int snd_usb_pcm_resume(struct snd_usb_stream *as);
 
 int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
 		       struct usb_host_interface *alts,
diff --git a/sound/usb/power.c b/sound/usb/power.c
new file mode 100644
index 000000000000..bd303a1ba1b7
--- /dev/null
+++ b/sound/usb/power.c
@@ -0,0 +1,104 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ *   UAC3 Power Domain state management functions
+ */
+
+#include <linux/slab.h>
+#include <linux/usb.h>
+#include <linux/usb/audio.h>
+#include <linux/usb/audio-v2.h>
+#include <linux/usb/audio-v3.h>
+
+#include "usbaudio.h"
+#include "helper.h"
+#include "power.h"
+
+struct snd_usb_power_domain *
+snd_usb_find_power_domain(struct usb_host_interface *ctrl_iface,
+			  unsigned char id)
+{
+	struct snd_usb_power_domain *pd;
+	void *p;
+
+	pd = kzalloc(sizeof(*pd), GFP_KERNEL);
+	if (!pd)
+		return NULL;
+
+	p = NULL;
+	while ((p = snd_usb_find_csint_desc(ctrl_iface->extra,
+					    ctrl_iface->extralen,
+					    p, UAC3_POWER_DOMAIN)) != NULL) {
+		struct uac3_power_domain_descriptor *pd_desc = p;
+		int i;
+
+		for (i = 0; i < pd_desc->bNrEntities; i++) {
+			if (pd_desc->baEntityID[i] == id) {
+				pd->pd_id = pd_desc->bPowerDomainID;
+				pd->pd_d1d0_rec =
+					le16_to_cpu(pd_desc->waRecoveryTime1);
+				pd->pd_d2d0_rec =
+					le16_to_cpu(pd_desc->waRecoveryTime2);
+				return pd;
+			}
+		}
+	}
+
+	kfree(pd);
+	return NULL;
+}
+
+int snd_usb_power_domain_set(struct snd_usb_audio *chip,
+			     struct snd_usb_power_domain *pd,
+			     unsigned char state)
+{
+	struct usb_device *dev = chip->dev;
+	unsigned char current_state;
+	int err, idx;
+
+	idx = snd_usb_ctrl_intf(chip) | (pd->pd_id << 8);
+
+	err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0),
+			      UAC2_CS_CUR,
+			      USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
+			      UAC3_AC_POWER_DOMAIN_CONTROL << 8, idx,
+			      &current_state, sizeof(current_state));
+	if (err < 0) {
+		dev_err(&dev->dev, "Can't get UAC3 power state for id %d\n",
+			pd->pd_id);
+		return err;
+	}
+
+	if (current_state == state) {
+		dev_dbg(&dev->dev, "UAC3 power domain id %d already in state %d\n",
+			pd->pd_id, state);
+		return 0;
+	}
+
+	err = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), UAC2_CS_CUR,
+			      USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
+			      UAC3_AC_POWER_DOMAIN_CONTROL << 8, idx,
+			      &state, sizeof(state));
+	if (err < 0) {
+		dev_err(&dev->dev, "Can't set UAC3 power state to %d for id %d\n",
+			state, pd->pd_id);
+		return err;
+	}
+
+	if (state == UAC3_PD_STATE_D0) {
+		switch (current_state) {
+		case UAC3_PD_STATE_D2:
+			udelay(pd->pd_d2d0_rec * 50);
+			break;
+		case UAC3_PD_STATE_D1:
+			udelay(pd->pd_d1d0_rec * 50);
+			break;
+		default:
+			return -EINVAL;
+		}
+	}
+
+	dev_dbg(&dev->dev, "UAC3 power domain id %d change to state %d\n",
+		pd->pd_id, state);
+
+	return 0;
+}
diff --git a/sound/usb/power.h b/sound/usb/power.h
index b2e25f60c5a2..6004231a7c75 100644
--- a/sound/usb/power.h
+++ b/sound/usb/power.h
@@ -2,6 +2,25 @@
 #ifndef __USBAUDIO_POWER_H
 #define __USBAUDIO_POWER_H
 
+struct snd_usb_power_domain {
+	int pd_id;              /* UAC3 Power Domain ID */
+	int pd_d1d0_rec;        /* D1 to D0 recovery time */
+	int pd_d2d0_rec;        /* D2 to D0 recovery time */
+};
+
+enum {
+	UAC3_PD_STATE_D0,
+	UAC3_PD_STATE_D1,
+	UAC3_PD_STATE_D2,
+};
+
+int snd_usb_power_domain_set(struct snd_usb_audio *chip,
+			     struct snd_usb_power_domain *pd,
+			     unsigned char state);
+struct snd_usb_power_domain *
+snd_usb_find_power_domain(struct usb_host_interface *ctrl_iface,
+			  unsigned char id);
+
 #ifdef CONFIG_PM
 int snd_usb_autoresume(struct snd_usb_audio *chip);
 void snd_usb_autosuspend(struct snd_usb_audio *chip);
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 8aac48f9c322..08aa78007020 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2875,7 +2875,8 @@ YAMAHA_DEVICE(0x7010, "UB99"),
  */
 
 #define AU0828_DEVICE(vid, pid, vname, pname) { \
-	USB_DEVICE_VENDOR_SPEC(vid, pid), \
+	.idVendor = vid, \
+	.idProduct = pid, \
 	.match_flags = USB_DEVICE_ID_MATCH_DEVICE | \
 		       USB_DEVICE_ID_MATCH_INT_CLASS | \
 		       USB_DEVICE_ID_MATCH_INT_SUBCLASS, \
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 02b6cc02767f..8a945ece9869 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1213,7 +1213,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs,
 		if (err < 0)
 			return err;
 
-		mdelay(20); /* Delay needed after setting the interface */
+		msleep(20); /* Delay needed after setting the interface */
 
 		/* Vendor mode switch cmd is required. */
 		if (fmt->formats & SNDRV_PCM_FMTBIT_DSD_U32_BE) {
@@ -1234,7 +1234,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs,
 				return err;
 
 		}
-		mdelay(20);
+		msleep(20);
 	}
 	return 0;
 }
@@ -1281,7 +1281,7 @@ void snd_usb_set_interface_quirk(struct usb_device *dev)
 	switch (USB_ID_VENDOR(chip->usb_id)) {
 	case 0x23ba: /* Playback Design */
 	case 0x0644: /* TEAC Corp. */
-		mdelay(50);
+		msleep(50);
 		break;
 	}
 }
@@ -1301,7 +1301,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
 	 */
 	if (USB_ID_VENDOR(chip->usb_id) == 0x23ba &&
 	    (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
-		mdelay(20);
+		msleep(20);
 
 	/*
 	 * "TEAC Corp." products need a 20ms delay after each
@@ -1309,14 +1309,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
 	 */
 	if (USB_ID_VENDOR(chip->usb_id) == 0x0644 &&
 	    (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
-		mdelay(20);
+		msleep(20);
 
 	/* ITF-USB DSD based DACs functionality need a delay
 	 * after each class compliant request
 	 */
 	if (is_itf_usb_dsd_dac(chip->usb_id)
 	    && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
-		mdelay(20);
+		msleep(20);
 
 	/* Zoom R16/24, Logitech H650e, Jabra 550a needs a tiny delay here,
 	 * otherwise requests like get/set frequency return as failed despite
@@ -1326,7 +1326,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
 	     chip->usb_id == USB_ID(0x046d, 0x0a46) ||
 	     chip->usb_id == USB_ID(0x0b0e, 0x0349)) &&
 	    (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
-		mdelay(1);
+		usleep_range(1000, 2000);
 }
 
 /*
@@ -1373,6 +1373,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 			return SNDRV_PCM_FMTBIT_DSD_U32_BE;
 		break;
 
+	case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */
 	case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */
 	case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */
 	case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */
@@ -1443,6 +1444,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 	 */
 	switch (USB_ID_VENDOR(chip->usb_id)) {
 	case 0x20b1:  /* XMOS based devices */
+	case 0x152a:  /* Thesycon devices */
 	case 0x25ce:  /* Mytek devices */
 		if (fp->dsd_raw)
 			return SNDRV_PCM_FMTBIT_DSD_U32_BE;
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 729afd808cc4..67cf849aa16b 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -37,6 +37,7 @@
 #include "format.h"
 #include "clock.h"
 #include "stream.h"
+#include "power.h"
 
 /*
  * free a substream
@@ -53,6 +54,7 @@ static void free_substream(struct snd_usb_substream *subs)
 		kfree(fp);
 	}
 	kfree(subs->rate_list.list);
+	kfree(subs->str_pd);
 }
 
 
@@ -82,7 +84,8 @@ static void snd_usb_audio_pcm_free(struct snd_pcm *pcm)
 
 static void snd_usb_init_substream(struct snd_usb_stream *as,
 				   int stream,
-				   struct audioformat *fp)
+				   struct audioformat *fp,
+				   struct snd_usb_power_domain *pd)
 {
 	struct snd_usb_substream *subs = &as->substream[stream];
 
@@ -107,6 +110,13 @@ static void snd_usb_init_substream(struct snd_usb_stream *as,
 	if (fp->channels > subs->channels_max)
 		subs->channels_max = fp->channels;
 
+	if (pd) {
+		subs->str_pd = pd;
+		/* Initialize Power Domain to idle status D1 */
+		snd_usb_power_domain_set(subs->stream->chip, pd,
+					 UAC3_PD_STATE_D1);
+	}
+
 	snd_usb_preallocate_buffer(subs);
 }
 
@@ -468,9 +478,11 @@ snd_pcm_chmap_elem *convert_chmap_v3(struct uac3_cluster_header_descriptor
  * fmt_list and will be freed on the chip instance release. do not free
  * fp or do remove it from the substream fmt_list to avoid double-free.
  */
-int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
-			     int stream,
-			     struct audioformat *fp)
+static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip,
+				      int stream,
+				      struct audioformat *fp,
+				      struct snd_usb_power_domain *pd)
+
 {
 	struct snd_usb_stream *as;
 	struct snd_usb_substream *subs;
@@ -498,7 +510,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
 		err = snd_pcm_new_stream(as->pcm, stream, 1);
 		if (err < 0)
 			return err;
-		snd_usb_init_substream(as, stream, fp);
+		snd_usb_init_substream(as, stream, fp, pd);
 		return add_chmap(as->pcm, stream, subs);
 	}
 
@@ -526,7 +538,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
 	else
 		strcpy(pcm->name, "USB Audio");
 
-	snd_usb_init_substream(as, stream, fp);
+	snd_usb_init_substream(as, stream, fp, pd);
 
 	/*
 	 * Keep using head insertion for M-Audio Audiophile USB (tm) which has a
@@ -544,6 +556,21 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
 	return add_chmap(pcm, stream, &as->substream[stream]);
 }
 
+int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
+			     int stream,
+			     struct audioformat *fp)
+{
+	return __snd_usb_add_audio_stream(chip, stream, fp, NULL);
+}
+
+static int snd_usb_add_audio_stream_v3(struct snd_usb_audio *chip,
+				       int stream,
+				       struct audioformat *fp,
+				       struct snd_usb_power_domain *pd)
+{
+	return __snd_usb_add_audio_stream(chip, stream, fp, pd);
+}
+
 static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip,
 					 struct usb_host_interface *alts,
 					 int protocol, int iface_no)
@@ -819,6 +846,7 @@ found_clock:
 static struct audioformat *
 snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip,
 			     struct usb_host_interface *alts,
+			     struct snd_usb_power_domain **pd_out,
 			     int iface_no, int altset_idx,
 			     int altno, int stream)
 {
@@ -829,6 +857,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip,
 	struct uac3_as_header_descriptor *as = NULL;
 	struct uac3_hc_descriptor_header hc_header;
 	struct snd_pcm_chmap_elem *chmap;
+	struct snd_usb_power_domain *pd;
 	unsigned char badd_profile;
 	u64 badd_formats = 0;
 	unsigned int num_channels;
@@ -1008,12 +1037,28 @@ found_clock:
 		fp->rate_max = UAC3_BADD_SAMPLING_RATE;
 		fp->rates = SNDRV_PCM_RATE_CONTINUOUS;
 
+		pd = kzalloc(sizeof(*pd), GFP_KERNEL);
+		if (!pd) {
+			kfree(fp->rate_table);
+			kfree(fp);
+			return NULL;
+		}
+		pd->pd_id = (stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+					UAC3_BADD_PD_ID10 : UAC3_BADD_PD_ID11;
+		pd->pd_d1d0_rec = UAC3_BADD_PD_RECOVER_D1D0;
+		pd->pd_d2d0_rec = UAC3_BADD_PD_RECOVER_D2D0;
+
 	} else {
 		fp->attributes = parse_uac_endpoint_attributes(chip, alts,
 							       UAC_VERSION_3,
 							       iface_no);
+
+		pd = snd_usb_find_power_domain(chip->ctrl_intf,
+					       as->bTerminalLink);
+
 		/* ok, let's parse further... */
 		if (snd_usb_parse_audio_format_v3(chip, fp, as, stream) < 0) {
+			kfree(pd);
 			kfree(fp->chmap);
 			kfree(fp->rate_table);
 			kfree(fp);
@@ -1021,6 +1066,9 @@ found_clock:
 		}
 	}
 
+	if (pd)
+		*pd_out = pd;
+
 	return fp;
 }
 
@@ -1032,6 +1080,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
 	struct usb_interface_descriptor *altsd;
 	int i, altno, err, stream;
 	struct audioformat *fp = NULL;
+	struct snd_usb_power_domain *pd = NULL;
 	int num, protocol;
 
 	dev = chip->dev;
@@ -1114,7 +1163,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
 			break;
 		}
 		case UAC_VERSION_3:
-			fp = snd_usb_get_audioformat_uac3(chip, alts,
+			fp = snd_usb_get_audioformat_uac3(chip, alts, &pd,
 						iface_no, i, altno, stream);
 			break;
 		}
@@ -1125,9 +1174,14 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
 			return PTR_ERR(fp);
 
 		dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint);
-		err = snd_usb_add_audio_stream(chip, stream, fp);
+		if (protocol == UAC_VERSION_3)
+			err = snd_usb_add_audio_stream_v3(chip, stream, fp, pd);
+		else
+			err = snd_usb_add_audio_stream(chip, stream, fp);
+
 		if (err < 0) {
 			list_del(&fp->list); /* unlink for avoiding double-free */
+			kfree(pd);
 			kfree(fp->rate_table);
 			kfree(fp->chmap);
 			kfree(fp);
diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c
index 4ed9d0c41843..fa7dca5a68c8 100644
--- a/sound/x86/intel_hdmi_audio.c
+++ b/sound/x86/intel_hdmi_audio.c
@@ -290,7 +290,6 @@ static void had_reset_audio(struct snd_intelhad *intelhaddata)
 static int had_prog_status_reg(struct snd_pcm_substream *substream,
 			struct snd_intelhad *intelhaddata)
 {
-	union aud_cfg cfg_val = {.regval = 0};
 	union aud_ch_status_0 ch_stat0 = {.regval = 0};
 	union aud_ch_status_1 ch_stat1 = {.regval = 0};
 
@@ -298,7 +297,6 @@ static int had_prog_status_reg(struct snd_pcm_substream *substream,
 					  IEC958_AES0_NONAUDIO) >> 1;
 	ch_stat0.regx.clk_acc = (intelhaddata->aes_bits &
 					  IEC958_AES3_CON_CLOCK) >> 4;
-	cfg_val.regx.val_bit = ch_stat0.regx.lpcm_id;
 
 	switch (substream->runtime->rate) {
 	case AUD_SAMPLE_RATE_32:
@@ -1854,7 +1852,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev)
 		/* setup private data which can be retrieved when required */
 		pcm->private_data = ctx;
 		pcm->info_flags = 0;
-		strncpy(pcm->name, card->shortname, strlen(card->shortname));
+		strlcpy(pcm->name, card->shortname, strlen(card->shortname));
 		/* setup the ops for playabck */
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &had_pcm_ops);
 
diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c
index 5a2bd70a2fa1..129180e17db1 100644
--- a/sound/xen/xen_snd_front_alsa.c
+++ b/sound/xen/xen_snd_front_alsa.c
@@ -188,7 +188,7 @@ static u64 to_sndif_formats_mask(u64 alsa_formats)
 
 	mask = 0;
 	for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++)
-		if (1 << ALSA_SNDIF_FORMATS[i].alsa & alsa_formats)
+		if (pcm_format_to_bits(ALSA_SNDIF_FORMATS[i].alsa) & alsa_formats)
 			mask |= 1 << ALSA_SNDIF_FORMATS[i].sndif;
 
 	return mask;
@@ -202,7 +202,7 @@ static u64 to_alsa_formats_mask(u64 sndif_formats)
 	mask = 0;
 	for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++)
 		if (1 << ALSA_SNDIF_FORMATS[i].sndif & sndif_formats)
-			mask |= 1 << ALSA_SNDIF_FORMATS[i].alsa;
+			mask |= pcm_format_to_bits(ALSA_SNDIF_FORMATS[i].alsa);
 
 	return mask;
 }