Merge tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There have been little changes in ALSA core stuff, but ASoC core still
  kept rolling for the continued restructuring. The rest are lots of
  small driver-specific changes and some minor API updates. Here are
  highlights:

  General:
  - Appropriate fall-through annotations everywhere
  - Some code cleanup in memalloc code, handling non-cacahed pages more
    commonly in the helper
  - Deployment of SNDRV_PCM_INFO_SYNC_APPLPTR flag consistently

  Drivers:
  - More HD-audio CA0132 codec improvement for supporting other Creative
    boards
  - Plumbing legacy HD-audio codecs as ASoC BE on Intel SST; this will
    give move support of existing HD-audio devices with DSP
  - A few device-specific HD-audio quirks as usual
  - New quirk for RME CC devices and correction for B&W PX for USB-audio
  - FireWire: code refactoring including devres usages

  ASoC Core:
  - Continued componentization works; it's almost done!
  - A bunch of new for_each_foo macros
  - Cleanups and fixes in DAPM code

  ASoC Drivers:
  - MCLK support for several different devices, including CS42L51, STM32
    SAI, and MAX98373
  - Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
    MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
    TI PCM3060"

* tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (299 commits)
  ASoC: stm32: sai: fix master clock naming
  ASoC: stm32: add clock dependency for sai
  ALSA: hda/ca0132 - Actually fix microphone issue
  ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks
  ASoC: wm2000: Remove wm2000_read helper function
  ASoC: cs42l51: fix mclk support
  ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate
  ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer
  ASoC: cs42l51: add mclk support
  ASoC: stm32: sai: set sai as mclk clock provider
  ASoC: dt-bindings: add mclk support to cs42l51
  ASoC: dt-bindings: add mclk provider support to stm32 sai
  ASoC: soc-core: fix trivial checkpatch issues
  ASoC: dapm: Add support for hw_free on CODEC to CODEC links
  ASoC: Intel: kbl_da7219_max98927: minor white space clean up
  ALSA: i2c/cs8427: Fix int to char conversion
  ALSA: doc: Brush up the old writing-an-alsa-driver
  ASoC: rsnd: tidyup SSICR::SWSP for TDM
  ASoC: rsnd: enable TDM settings for SSI parent
  ASoC: pcm3168a: add hw constraint for capture channel
  ...

244 files changed, 10779 insertions, 2627 deletions

diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.txt b/Documentation/devicetree/bindings/sound/adi,adau1977.txt
new file mode 100644
index 000000000000..e79aeef73f28
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,adau1977.txt
@@ -0,0 +1,54 @@
+Analog Devices ADAU1977/ADAU1978/ADAU1979
+
+Datasheets:
+http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf
+http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf
+http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf
+
+This driver supports both the I2C and SPI bus.
+
+Required properties:
+ - compatible: Should contain one of the following:
+               "adi,adau1977"
+               "adi,adau1978"
+               "adi,adau1979"
+
+ - AVDD-supply: analog power supply for the device, please consult
+                Documentation/devicetree/bindings/regulator/regulator.txt
+
+Optional properties:
+ - reset-gpio:  the reset pin for the chip, for more details consult
+                Documentation/devicetree/bindings/gpio/gpio.txt
+
+ - DVDD-supply: supply voltage for the digital core, please consult
+                Documentation/devicetree/bindings/regulator/regulator.txt
+
+For required properties on SPI, please consult
+Documentation/devicetree/bindings/spi/spi-bus.txt
+
+Required properties on I2C:
+
+ - reg:         The i2c address. Value depends on the state of ADDR0
+                and ADDR1, as wired in hardware.
+
+Examples:
+
+        adau1977_spi: adau1977@0 {
+                compatible = "adi,adau1977";
+                spi-max-frequency = <600000>;
+
+                AVDD-supply = <&regulator>;
+                DVDD-supply = <&regulator_digital>;
+
+                reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>;
+        };
+
+        adau1977_i2c: adau1977@11 {
+                compatible = "adi,adau1977";
+                reg = <0x11>;
+
+                AVDD-supply = <&regulator>;
+                DVDD-supply = <&regulator_digital>;
+
+                reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>;
+        };
diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt
new file mode 100644
index 000000000000..5672d0bc5b16
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt
@@ -0,0 +1,24 @@
+* Amlogic Audio PDM input
+
+Required properties:
+- compatible: 'amlogic,axg-pdm'
+- 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.
+  * "dclk"   : pdm digital clock
+  * "sysclk" : dsp system clock
+- #sound-dai-cells: must be 0.
+
+Example of PDM on the A113 SoC:
+
+pdm: audio-controller@ff632000 {
+        compatible = "amlogic,axg-pdm";
+        reg = <0x0 0xff632000 0x0 0x34>;
+        #sound-dai-cells = <0>;
+        clocks = <&clkc_audio AUD_CLKID_PDM>,
+                 <&clkc_audio AUD_CLKID_PDM_DCLK>,
+                 <&clkc_audio AUD_CLKID_PDM_SYSCLK>;
+        clock-names = "pclk", "dclk", "sysclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/cs42l51.txt b/Documentation/devicetree/bindings/sound/cs42l51.txt
new file mode 100644
index 000000000000..4b5de33ce377
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs42l51.txt
@@ -0,0 +1,17 @@
+CS42L51 audio CODEC
+
+Optional properties:
+
+  - clocks : a list of phandles + clock-specifiers, one for each entry in
+    clock-names
+
+  - clock-names : must contain "MCLK"
+
+Example:
+
+cs42l51: cs42l51@4a {
+        compatible = "cirrus,cs42l51";
+        reg = <0x4a>;
+        clocks = <&mclk_prov>;
+        clock-names = "MCLK";
+};
diff --git a/Documentation/devicetree/bindings/sound/maxim,max98088.txt b/Documentation/devicetree/bindings/sound/maxim,max98088.txt
new file mode 100644
index 000000000000..da764d913319
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/maxim,max98088.txt
@@ -0,0 +1,23 @@
+MAX98088 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible: "maxim,max98088" or "maxim,max98089".
+- reg: The I2C address of the device.
+
+Optional properties:
+
+- clocks: the clock provider of MCLK, see ../clock/clock-bindings.txt section
+  "consumer" for more information.
+- clock-names: must be set to "mclk"
+
+Example:
+
+max98089: codec@10 {
+        compatible = "maxim,max98089";
+        reg = <0x10>;
+        clocks = <&clks IMX6QDL_CLK_CKO2>;
+        clock-names = "mclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt b/Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt
new file mode 100644
index 000000000000..912f8fae11c5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt
@@ -0,0 +1,23 @@
+Mikroe-PROTO audio board
+
+Required properties:
+  - compatible: "mikroe,mikroe-proto"
+  - dai-format: Must be "i2s".
+  - i2s-controller: The phandle of the I2S controller.
+  - audio-codec: The phandle of the WM8731 audio codec.
+Optional properties:
+  - model: The user-visible name of this sound complex.
+  - bitclock-master: Indicates dai-link bit clock master; for details see simple-card.txt (1).
+  - frame-master: Indicates dai-link frame master; for details see simple-card.txt (1).
+
+(1) : There must be the same master for both bit and frame clocks.
+
+Example:
+        sound {
+                compatible = "mikroe,mikroe-proto";
+                model = "wm8731 @ sama5d2_xplained";
+                i2s-controller = <&i2s0>;
+                audio-codec = <&wm8731>;
+                dai-format = "i2s";
+        };
+};
diff --git a/Documentation/devicetree/bindings/sound/nau8822.txt b/Documentation/devicetree/bindings/sound/nau8822.txt
new file mode 100644
index 000000000000..a471d162d4e5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nau8822.txt
@@ -0,0 +1,16 @@
+NAU8822 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "nuvoton,nau8822"
+
+  - reg : the I2C address of the device.
+
+Example:
+
+codec: nau8822@1a {
+        compatible = "nuvoton,nau8822";
+        reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/pcm3060.txt b/Documentation/devicetree/bindings/sound/pcm3060.txt
new file mode 100644
index 000000000000..90fcb8523099
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/pcm3060.txt
@@ -0,0 +1,17 @@
+PCM3060 audio CODEC
+
+This driver supports both I2C and SPI.
+
+Required properties:
+
+- compatible: "ti,pcm3060"
+
+- reg : the I2C address of the device for I2C, the chip select
+        number for SPI.
+
+Examples:
+
+        pcm3060: pcm3060@46 {
+                 compatible = "ti,pcm3060";
+                 reg = <0x46>;
+        };
diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
index a8179409c194..d74888b9f1bb 100644
--- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
+++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
@@ -49,7 +49,7 @@ configuration of each dai. Must contain the following properties.
         Usage: required for mi2s interface
         Value type: <prop-encoded-array>
         Definition: Must be list of serial data lines used by this dai.
-        should be one or more of the 1-4 sd lines.
+        should be one or more of the 0-3 sd lines.
 
  - qcom,tdm-sync-mode:
         Usage: required for tdm interface
@@ -137,42 +137,42 @@ q6afe@4 {
 
                 prim-mi2s-rx@16 {
                         reg = <16>;
-                        qcom,sd-lines = <1 3>;
+                        qcom,sd-lines = <0 2>;
                 };
 
                 prim-mi2s-tx@17 {
                         reg = <17>;
-                        qcom,sd-lines = <2>;
+                        qcom,sd-lines = <1>;
                 };
 
                 sec-mi2s-rx@18 {
                         reg = <18>;
-                        qcom,sd-lines = <1 4>;
+                        qcom,sd-lines = <0 3>;
                 };
 
                 sec-mi2s-tx@19 {
                         reg = <19>;
-                        qcom,sd-lines = <2>;
+                        qcom,sd-lines = <1>;
                 };
 
                 tert-mi2s-rx@20 {
                         reg = <20>;
-                        qcom,sd-lines = <2 4>;
+                        qcom,sd-lines = <1 3>;
                 };
 
                 tert-mi2s-tx@21 {
                         reg = <21>;
-                        qcom,sd-lines = <1>;
+                        qcom,sd-lines = <0>;
                 };
 
                 quat-mi2s-rx@22 {
                         reg = <22>;
-                        qcom,sd-lines = <1>;
+                        qcom,sd-lines = <0>;
                 };
 
                 quat-mi2s-tx@23 {
                         reg = <23>;
-                        qcom,sd-lines = <2>;
+                        qcom,sd-lines = <1>;
                 };
         };
 };
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index 9e764270c36b..d92b705e7917 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -340,10 +340,12 @@ Required properties:
 - compatible                    : "renesas,rcar_sound-<soctype>", fallbacks
                                   "renesas,rcar_sound-gen1" if generation1, and
                                   "renesas,rcar_sound-gen2" if generation2 (or RZ/G1)
-                                  "renesas,rcar_sound-gen3" if generation3
+                                  "renesas,rcar_sound-gen3" if generation3 (or RZ/G2)
                                   Examples with soctypes are:
                                     - "renesas,rcar_sound-r8a7743" (RZ/G1M)
+                                    - "renesas,rcar_sound-r8a7744" (RZ/G1N)
                                     - "renesas,rcar_sound-r8a7745" (RZ/G1E)
+                                    - "renesas,rcar_sound-r8a774a1" (RZ/G2M)
                                     - "renesas,rcar_sound-r8a7778" (R-Car M1A)
                                     - "renesas,rcar_sound-r8a7779" (R-Car H1)
                                     - "renesas,rcar_sound-r8a7790" (R-Car H2)
@@ -353,6 +355,7 @@ Required properties:
                                     - "renesas,rcar_sound-r8a7795" (R-Car H3)
                                     - "renesas,rcar_sound-r8a7796" (R-Car M3-W)
                                     - "renesas,rcar_sound-r8a77965" (R-Car M3-N)
+                                    - "renesas,rcar_sound-r8a77990" (R-Car E3)
 - reg                           : Should contain the register physical address.
                                   required register is
                                    SRU/ADG/SSI      if generation1
diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt
index 255de3ae5b2f..52265fb757c5 100644
--- a/Documentation/devicetree/bindings/sound/st,sta32x.txt
+++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt
@@ -19,6 +19,10 @@ Required properties:
 
 Optional properties:
 
+  - clocks, clock-names: Clock specifier for XTI input clock.
+        If specified, the clock will be enabled when the codec is probed,
+        and disabled when it is removed. The 'clock-names' must be set to 'xti'.
+
   -  st,output-conf: number, Selects the output configuration:
         0: 2-channel (full-bridge) power, 2-channel data-out
         1: 2 (half-bridge). 1 (full-bridge) on-board power
@@ -39,6 +43,9 @@ Optional properties:
   -  st,thermal-warning-recover:
         If present, thermal warning recovery is enabled.
 
+  - st,fault-detect-recovery:
+        If present, fault detect recovery is enabled.
+
   -  st,thermal-warning-adjustment:
         If present, thermal warning adjustment is enabled.
 
@@ -76,6 +83,8 @@ Example:
 codec: sta32x@38 {
         compatible = "st,sta32x";
         reg = <0x1c>;
+        clocks = <&clock>;
+        clock-names = "xti";
         reset-gpios = <&gpio1 19 0>;
         power-down-gpios = <&gpio1 16 0>;
         st,output-conf = /bits/ 8  <0x3>;       // set output to 2-channel
diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt
index 3a3fc506e43a..3f4467ff0aa2 100644
--- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt
+++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt
@@ -31,7 +31,11 @@ SAI subnodes required properties:
   - reg: Base address and size of SAI sub-block register set.
   - clocks: Must contain one phandle and clock specifier pair
         for sai_ck which feeds the internal clock generator.
+        If the SAI shares a master clock, with another SAI set as MCLK
+        clock provider, SAI provider phandle must be specified here.
   - clock-names: Must contain "sai_ck".
+        Must also contain "MCLK", if SAI shares a master clock,
+        with a SAI set as MCLK clock provider.
   - dmas: see Documentation/devicetree/bindings/dma/stm32-dma.txt
   - dma-names: identifier string for each DMA request line
         "tx": if sai sub-block is configured as playback DAI
@@ -51,6 +55,9 @@ SAI subnodes Optional properties:
         configured according to protocol defined in related DAI link node,
         such as i2s, left justified, right justified, dsp and pdm protocols.
         Note: ac97 protocol is not supported by SAI driver
+   - #clock-cells: should be 0. This property must be present if the SAI device
+        is a master clock provider, according to clocks bindings, described in
+        Documentation/devicetree/bindings/clock/clock-bindings.txt.
 
 The device node should contain one 'port' child node with one child 'endpoint'
 node, according to the bindings defined in Documentation/devicetree/bindings/
diff --git a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
index b9d50d6cdef3..61e71c1729e0 100644
--- a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
+++ b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
@@ -10,6 +10,7 @@ Required properties:
    - "allwinner,sun6i-a31-i2s"
    - "allwinner,sun8i-a83t-i2s"
    - "allwinner,sun8i-h3-i2s"
+   - "allwinner,sun50i-a64-codec-i2s"
 - reg: physical base address of the controller and length of memory mapped
   region.
 - interrupts: should contain the I2S interrupt.
@@ -26,6 +27,7 @@ Required properties for the following compatibles:
         - "allwinner,sun6i-a31-i2s"
         - "allwinner,sun8i-a83t-i2s"
         - "allwinner,sun8i-h3-i2s"
+        - "allwinner,sun50i-a64-codec-i2s"
 - resets: phandle to the reset line for this codec
 
 Example:
diff --git a/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt b/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt
new file mode 100644
index 000000000000..4f8ad0e04d20
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt
@@ -0,0 +1,12 @@
+* Allwinner A64 Codec Analog Controls
+
+Required properties:
+- compatible: must be one of the following compatibles:
+                - "allwinner,sun50i-a64-codec-analog"
+- reg: must contain the registers location and length
+
+Example:
+        codec_analog: codec-analog@1f015c0 {
+                compatible = "allwinner,sun50i-a64-codec-analog";
+                reg = <0x01f015c0 0x4>;
+        };
diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt
index 3ed8359144d3..21ab45bc7e8f 100644
--- a/Documentation/devicetree/bindings/sound/ts3a227e.txt
+++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt
@@ -14,7 +14,7 @@ Required properties:
 
 Optional properies:
  - ti,micbias:   Intended MICBIAS voltage (datasheet section 9.6.7).
-      Select 0/1/2/3/4/5/6/7 to specify MACBIAS voltage
+      Select 0/1/2/3/4/5/6/7 to specify MICBIAS voltage
       2.1V/2.2V/2.3V/2.4V/2.5V/2.6V/2.7V/2.8V
       Default value is "1" (2.2V).
 
diff --git a/Documentation/devicetree/bindings/sound/wm8782.txt b/Documentation/devicetree/bindings/sound/wm8782.txt
new file mode 100644
index 000000000000..256cdec6ec4d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8782.txt
@@ -0,0 +1,17 @@
+WM8782 stereo ADC
+
+This device does not have any control interface or reset pins.
+
+Required properties:
+
+ - compatible  : "wlf,wm8782"
+ - Vdda-supply : phandle to a regulator for the analog power supply (2.7V - 5.5V)
+ - Vdd-supply  : phandle to a regulator for the digital power supply (2.7V - 3.6V)
+
+Example:
+
+wm8782: stereo-adc {
+        compatible = "wlf,wm8782";
+        Vdda-supply = <&vdda_supply>;
+        Vdd-supply = <&vdd_supply>;
+};
diff --git a/Documentation/devicetree/bindings/trivial-devices.txt b/Documentation/devicetree/bindings/trivial-devices.txt
index 763a2808a95c..69c934aec13b 100644
--- a/Documentation/devicetree/bindings/trivial-devices.txt
+++ b/Documentation/devicetree/bindings/trivial-devices.txt
@@ -35,7 +35,6 @@ at,24c08		i2c serial eeprom  (24cxx)
 atmel,at97sc3204t       i2c trusted platform module (TPM)
 capella,cm32181         CM32181: Ambient Light Sensor
 capella,cm3232          CM3232: Ambient Light Sensor
-cirrus,cs42l51          Cirrus Logic CS42L51 audio codec
 dallas,ds1374           I2C, 32-Bit Binary Counter Watchdog RTC with Trickle Charger and Reset Input/Output
 dallas,ds1631           High-Precision Digital Thermometer
 dallas,ds1672           Dallas DS1672 Real-time Clock
diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt
index 2c3fc512e746..f26bf667e530 100644
--- a/Documentation/devicetree/bindings/vendor-prefixes.txt
+++ b/Documentation/devicetree/bindings/vendor-prefixes.txt
@@ -235,6 +235,7 @@ micrel	Micrel Inc.
 microchip       Microchip Technology Inc.
 microcrystal    Micro Crystal AG
 micron  Micron Technology Inc.
+mikroe          MikroElektronika d.o.o.
 minix   MINIX Technology Ltd.
 miramems        MiraMEMS Sensing Technology Co., Ltd.
 mitsubishi      Mitsubishi Electric Corporation
diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst
index e06238131f77..368a07a165f5 100644
--- a/Documentation/sound/hd-audio/models.rst
+++ b/Documentation/sound/hd-audio/models.rst
@@ -309,6 +309,8 @@ asus-nx50
     ASUS Nx50 fixups
 asus-nx51
     ASUS Nx51 fixups
+asus-g751
+    ASUS G751 fixups
 alc891-headset
     Headset mode support on ALC891
 alc891-headset-multi
diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
index a0b268466cb1..b37234afdfa1 100644
--- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
+++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
@@ -3,8 +3,6 @@ Writing an ALSA Driver
 ======================
 
 :Author: Takashi Iwai <tiwai@suse.de>
-:Date:   Oct 15, 2007
-:Edition: 0.3.7
 
 Preface
 =======
@@ -21,11 +19,6 @@ explain the general topic of linux kernel coding and doesn't cover
 low-level driver implementation details. It only describes the standard
 way to write a PCI sound driver on ALSA.
 
-If you are already familiar with the older ALSA ver.0.5.x API, you can
-check the drivers such as ``sound/pci/es1938.c`` or
-``sound/pci/maestro3.c`` which have also almost the same code-base in
-the ALSA 0.5.x tree, so you can compare the differences.
-
 This document is still a draft version. Any feedback and corrections,
 please!!
 
@@ -35,24 +28,7 @@ File Tree Structure
 General
 -------
 
-The ALSA drivers are provided in two ways.
-
-One is the trees provided as a tarball or via cvs from the ALSA's ftp
-site, and another is the 2.6 (or later) Linux kernel tree. To
-synchronize both, the ALSA driver tree is split into two different
-trees: alsa-kernel and alsa-driver. The former contains purely the
-source code for the Linux 2.6 (or later) tree. This tree is designed
-only for compilation on 2.6 or later environment. The latter,
-alsa-driver, contains many subtle files for compiling ALSA drivers
-outside of the Linux kernel tree, wrapper functions for older 2.2 and
-2.4 kernels, to adapt the latest kernel API, and additional drivers
-which are still in development or in tests. The drivers in alsa-driver
-tree will be moved to alsa-kernel (and eventually to the 2.6 kernel
-tree) when they are finished and confirmed to work fine.
-
-The file tree structure of ALSA driver is depicted below. Both
-alsa-kernel and alsa-driver have almost the same file structure, except
-for “core” directory. It's named as “acore” in alsa-driver tree.
+The file tree structure of ALSA driver is depicted below.
 
 ::
 
@@ -61,14 +37,11 @@ for “core” directory. It's named as “acore” in alsa-driver tree.
                             /oss
                             /seq
                                     /oss
-                                    /instr
-                    /ioctl32
                     /include
                     /drivers
                             /mpu401
                             /opl3
                     /i2c
-                            /l3
                     /synth
                             /emux
                     /pci
@@ -80,6 +53,7 @@ for “core” directory. It's named as “acore” in alsa-driver tree.
                     /sparc
                     /usb
                     /pcmcia /(cards)
+                    /soc
                     /oss
 
 
@@ -99,13 +73,6 @@ directory. The rawmidi OSS emulation is included in the ALSA rawmidi
 code since it's quite small. The sequencer code is stored in
 ``core/seq/oss`` directory (see `below <#core-seq-oss>`__).
 
-core/ioctl32
-~~~~~~~~~~~~
-
-This directory contains the 32bit-ioctl wrappers for 64bit architectures
-such like x86-64, ppc64 and sparc64. For 32bit and alpha architectures,
-these are not compiled.
-
 core/seq
 ~~~~~~~~
 
@@ -119,11 +86,6 @@ core/seq/oss
 
 This contains the OSS sequencer emulation codes.
 
-core/seq/instr
-~~~~~~~~~~~~~~
-
-This directory contains the modules for the sequencer instrument layer.
-
 include directory
 -----------------
 
@@ -161,11 +123,6 @@ Although there is a standard i2c layer on Linux, ALSA has its own i2c
 code for some cards, because the soundcard needs only a simple operation
 and the standard i2c API is too complicated for such a purpose.
 
-i2c/l3
-~~~~~~
-
-This is a sub-directory for ARM L3 i2c.
-
 synth directory
 ---------------
 
@@ -209,11 +166,19 @@ The PCMCIA, especially PCCard drivers will go here. CardBus drivers will
 be in the pci directory, because their API is identical to that of
 standard PCI cards.
 
+soc directory
+-------------
+
+This directory contains the codes for ASoC (ALSA System on Chip)
+layer including ASoC core, codec and machine drivers.
+
 oss directory
 -------------
 
-The OSS/Lite source files are stored here in Linux 2.6 (or later) tree.
-In the ALSA driver tarball, this directory is empty, of course :)
+Here contains OSS/Lite codes.
+All codes have been deprecated except for dmasound on m68k as of
+writing this.
+
 
 Basic Flow for PCI Drivers
 ==========================
@@ -352,10 +317,8 @@ to details explained in the following section.
 
               /* (3) */
               err = snd_mychip_create(card, pci, &chip);
-              if (err < 0) {
-                      snd_card_free(card);
-                      return err;
-              }
+              if (err < 0)
+                      goto error;
 
               /* (4) */
               strcpy(card->driver, "My Chip");
@@ -368,22 +331,23 @@ to details explained in the following section.
 
               /* (6) */
               err = snd_card_register(card);
-              if (err < 0) {
-                      snd_card_free(card);
-                      return err;
-              }
+              if (err < 0)
+                      goto error;
 
               /* (7) */
               pci_set_drvdata(pci, card);
               dev++;
               return 0;
+
+      error:
+              snd_card_free(card);
+              return err;
       }
 
       /* destructor -- see the "Destructor" sub-section */
       static void snd_mychip_remove(struct pci_dev *pci)
       {
               snd_card_free(pci_get_drvdata(pci));
-              pci_set_drvdata(pci, NULL);
       }
 
 
@@ -445,14 +409,26 @@ In this part, the PCI resources are allocated.
   struct mychip *chip;
   ....
   err = snd_mychip_create(card, pci, &chip);
-  if (err < 0) {
-          snd_card_free(card);
-          return err;
-  }
+  if (err < 0)
+          goto error;
 
 The details will be explained in the section `PCI Resource
 Management`_.
 
+When something goes wrong, the probe function needs to deal with the
+error.  In this example, we have a single error handling path placed
+at the end of the function.
+
+::
+
+  error:
+          snd_card_free(card);
+          return err;
+
+Since each component can be properly freed, the single
+:c:func:`snd_card_free()` call should suffice in most cases.
+
+
 4) Set the driver ID and name strings.
 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
 
@@ -486,10 +462,8 @@ too.
 ::
 
   err = snd_card_register(card);
-  if (err < 0) {
-          snd_card_free(card);
-          return err;
-  }
+  if (err < 0)
+          goto error;
 
 Will be explained in the section `Management of Cards and
 Components`_, too.
@@ -513,14 +487,13 @@ The destructor, remove callback, simply releases the card instance. Then
 the ALSA middle layer will release all the attached components
 automatically.
 
-It would be typically like the following:
+It would be typically just :c:func:`calling snd_card_free()`:
 
 ::
 
   static void snd_mychip_remove(struct pci_dev *pci)
   {
           snd_card_free(pci_get_drvdata(pci));
-          pci_set_drvdata(pci, NULL);
   }
 
 
@@ -546,7 +519,7 @@ in the source file. If the code is split into several files, the files
 without module options don't need them.
 
 In addition to these headers, you'll need ``<linux/interrupt.h>`` for
-interrupt handling, and ``<asm/io.h>`` for I/O access. If you use the
+interrupt handling, and ``<linux/io.h>`` for I/O access. If you use the
 :c:func:`mdelay()` or :c:func:`udelay()` functions, you'll need
 to include ``<linux/delay.h>`` too.
 
@@ -720,6 +693,13 @@ function, which will call the real destructor.
 
 where :c:func:`snd_mychip_free()` is the real destructor.
 
+The demerit of this method is the obviously more amount of codes.
+The merit is, however, you can trigger the own callback at registering
+and disconnecting the card via setting in snd_device_ops.
+About the registering and disconnecting the card, see the subsections
+below.
+
+
 Registration and Release
 ------------------------
 
@@ -905,10 +885,8 @@ Resource Allocation
 -------------------
 
 The allocation of I/O ports and irqs is done via standard kernel
-functions. Unlike ALSA ver.0.5.x., there are no helpers for that. And
-these resources must be released in the destructor function (see below).
-Also, on ALSA 0.9.x, you don't need to allocate (pseudo-)DMA for PCI
-like in ALSA 0.5.x.
+functions.  These resources must be released in the destructor
+function (see below).
 
 Now assume that the PCI device has an I/O port with 8 bytes and an
 interrupt. Then :c:type:`struct mychip <mychip>` will have the
@@ -1064,7 +1042,8 @@ and the allocation would be like below:
 
 ::
 
-  if ((err = pci_request_regions(pci, "My Chip")) < 0) {
+  err = pci_request_regions(pci, "My Chip");
+  if (err < 0) {
           kfree(chip);
           return err;
   }
@@ -1086,6 +1065,21 @@ and the corresponding destructor would be:
           ....
   }
 
+Of course, a modern way with :c:func:`pci_iomap()` will make things a
+bit easier, too.
+
+::
+
+  err = pci_request_regions(pci, "My Chip");
+  if (err < 0) {
+          kfree(chip);
+          return err;
+  }
+  chip->iobase_virt = pci_iomap(pci, 0, 0);
+
+which is paired with :c:func:`pci_iounmap()` at destructor.
+
+
 PCI Entries
 -----------
 
@@ -1154,13 +1148,6 @@ And at last, the module entries:
 Note that these module entries are tagged with ``__init`` and ``__exit``
 prefixes.
 
-Oh, one thing was forgotten. If you have no exported symbols, you need
-to declare it in 2.2 or 2.4 kernels (it's not necessary in 2.6 kernels).
-
-::
-
-  EXPORT_NO_SYMBOLS;
-
 That's all!
 
 PCM Interface
@@ -2113,6 +2100,16 @@ non-contiguous buffers. The mmap calls this callback to get the page
 address. Some examples will be explained in the later section `Buffer
 and Memory Management`_, too.
 
+mmap calllback
+~~~~~~~~~~~~~~
+
+This is another optional callback for controlling mmap behavior.
+Once when defined, PCM core calls this callback when a page is
+memory-mapped instead of dealing via the standard helper.
+If you need special handling (due to some architecture or
+device-specific issues), implement everything here as you like.
+
+
 PCM Interrupt Handler
 ---------------------
 
@@ -2370,6 +2367,27 @@ to define the inverse rule:
                       hw_rule_format_by_channels, NULL,
                       SNDRV_PCM_HW_PARAM_CHANNELS, -1);
 
+One typical usage of the hw constraints is to align the buffer size
+with the period size.  As default, ALSA PCM core doesn't enforce the
+buffer size to be aligned with the period size.  For example, it'd be
+possible to have a combination like 256 period bytes with 999 buffer
+bytes.
+
+Many device chips, however, require the buffer to be a multiple of
+periods.  In such a case, call
+:c:func:`snd_pcm_hw_constraint_integer()` for
+``SNDRV_PCM_HW_PARAM_PERIODS``.
+
+::
+
+  snd_pcm_hw_constraint_integer(substream->runtime,
+                                SNDRV_PCM_HW_PARAM_PERIODS);
+
+This assures that the number of periods is integer, hence the buffer
+size is aligned with the period size.
+
+The hw constraint is a very much powerful mechanism to define the
+preferred PCM configuration, and there are relevant helpers.
 I won't give more details here, rather I would like to say, “Luke, use
 the source.”
 
@@ -3712,7 +3730,14 @@ example, for an intermediate buffer. Since the allocated pages are not
 contiguous, you need to set the ``page`` callback to obtain the physical
 address at every offset.
 
-The implementation of ``page`` callback would be like this:
+The easiest way to achieve it would be to use
+:c:func:`snd_pcm_lib_alloc_vmalloc_buffer()` for allocating the buffer
+via :c:func:`vmalloc()`, and set :c:func:`snd_pcm_sgbuf_ops_page()` to
+the ``page`` callback.  At release, you need to call
+:c:func:`snd_pcm_lib_free_vmalloc_buffer()`.
+
+If you want to implementation the ``page`` manually, it would be like
+this:
 
 ::
 
@@ -3848,7 +3873,9 @@ Power Management
 
 If the chip is supposed to work with suspend/resume functions, you need
 to add power-management code to the driver. The additional code for
-power-management should be ifdef-ed with ``CONFIG_PM``.
+power-management should be ifdef-ed with ``CONFIG_PM``, or annotated
+with __maybe_unused attribute; otherwise the compiler will complain
+you.
 
 If the driver *fully* supports suspend/resume that is, the device can be
 properly resumed to its state when suspend was called, you can set the
@@ -3879,18 +3906,16 @@ the case of PCI drivers, the callbacks look like below:
 
 ::
 
-  #ifdef CONFIG_PM
-  static int snd_my_suspend(struct pci_dev *pci, pm_message_t state)
+  static int __maybe_unused snd_my_suspend(struct device *dev)
   {
           .... /* do things for suspend */
           return 0;
   }
-  static int snd_my_resume(struct pci_dev *pci)
+  static int __maybe_unused snd_my_resume(struct device *dev)
   {
           .... /* do things for suspend */
           return 0;
   }
-  #endif
 
 The scheme of the real suspend job is as follows.
 
@@ -3909,18 +3934,14 @@ The scheme of the real suspend job is as follows.
 
 6. Stop the hardware if necessary.
 
-7. Disable the PCI device by calling
-   :c:func:`pci_disable_device()`. Then, call
-   :c:func:`pci_save_state()` at last.
-
 A typical code would be like:
 
 ::
 
-  static int mychip_suspend(struct pci_dev *pci, pm_message_t state)
+  static int __maybe_unused mychip_suspend(struct device *dev)
   {
           /* (1) */
-          struct snd_card *card = pci_get_drvdata(pci);
+          struct snd_card *card = dev_get_drvdata(dev);
           struct mychip *chip = card->private_data;
           /* (2) */
           snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -3932,9 +3953,6 @@ A typical code would be like:
           snd_mychip_save_registers(chip);
           /* (6) */
           snd_mychip_stop_hardware(chip);
-          /* (7) */
-          pci_disable_device(pci);
-          pci_save_state(pci);
           return 0;
   }
 
@@ -3943,44 +3961,35 @@ The scheme of the real resume job is as follows.
 
 1. Retrieve the card and the chip data.
 
-2. Set up PCI. First, call :c:func:`pci_restore_state()`. Then
-   enable the pci device again by calling
-   :c:func:`pci_enable_device()`. Call
-   :c:func:`pci_set_master()` if necessary, too.
+2. Re-initialize the chip.
 
-3. Re-initialize the chip.
+3. Restore the saved registers if necessary.
 
-4. Restore the saved registers if necessary.
+4. Resume the mixer, e.g. calling :c:func:`snd_ac97_resume()`.
 
-5. Resume the mixer, e.g. calling :c:func:`snd_ac97_resume()`.
+5. Restart the hardware (if any).
 
-6. Restart the hardware (if any).
-
-7. Call :c:func:`snd_power_change_state()` with
+6. Call :c:func:`snd_power_change_state()` with
    ``SNDRV_CTL_POWER_D0`` to notify the processes.
 
 A typical code would be like:
 
 ::
 
-  static int mychip_resume(struct pci_dev *pci)
+  static int __maybe_unused mychip_resume(struct pci_dev *pci)
   {
           /* (1) */
-          struct snd_card *card = pci_get_drvdata(pci);
+          struct snd_card *card = dev_get_drvdata(dev);
           struct mychip *chip = card->private_data;
           /* (2) */
-          pci_restore_state(pci);
-          pci_enable_device(pci);
-          pci_set_master(pci);
-          /* (3) */
           snd_mychip_reinit_chip(chip);
-          /* (4) */
+          /* (3) */
           snd_mychip_restore_registers(chip);
-          /* (5) */
+          /* (4) */
           snd_ac97_resume(chip->ac97);
-          /* (6) */
+          /* (5) */
           snd_mychip_restart_chip(chip);
-          /* (7) */
+          /* (6) */
           snd_power_change_state(card, SNDRV_CTL_POWER_D0);
           return 0;
   }
@@ -4046,15 +4055,14 @@ And next, set suspend/resume callbacks to the pci_driver.
 
 ::
 
+  static SIMPLE_DEV_PM_OPS(snd_my_pm_ops, mychip_suspend, mychip_resume);
+
   static struct pci_driver driver = {
           .name = KBUILD_MODNAME,
           .id_table = snd_my_ids,
           .probe = snd_my_probe,
           .remove = snd_my_remove,
-  #ifdef CONFIG_PM
-          .suspend = snd_my_suspend,
-          .resume = snd_my_resume,
-  #endif
+          .driver.pm = &snd_my_pm_ops,
   };
 
 Module Parameters
@@ -4078,7 +4086,7 @@ variables, instead. ``enable`` option is not always necessary in this
 case, but it would be better to have a dummy option for compatibility.
 
 The module parameters must be declared with the standard
-``module_param()()``, ``module_param_array()()`` and
+``module_param()``, ``module_param_array()`` and
 :c:func:`MODULE_PARM_DESC()` macros.
 
 The typical coding would be like below:
@@ -4094,15 +4102,14 @@ The typical coding would be like below:
   module_param_array(enable, bool, NULL, 0444);
   MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
 
-Also, don't forget to define the module description, classes, license
-and devices. Especially, the recent modprobe requires to define the
+Also, don't forget to define the module description and the license.
+Especially, the recent modprobe requires to define the
 module license as GPL, etc., otherwise the system is shown as “tainted”.
 
 ::
 
-  MODULE_DESCRIPTION("My Chip");
+  MODULE_DESCRIPTION("Sound driver for My Chip");
   MODULE_LICENSE("GPL");
-  MODULE_SUPPORTED_DEVICE("{{Vendor,My Chip Name}}");
 
 
 How To Put Your Driver Into ALSA Tree
@@ -4117,21 +4124,17 @@ a question now: how to put my own driver into the ALSA driver tree? Here
 
 Suppose that you create a new PCI driver for the card “xyz”. The card
 module name would be snd-xyz. The new driver is usually put into the
-alsa-driver tree, ``alsa-driver/pci`` directory in the case of PCI
-cards. Then the driver is evaluated, audited and tested by developers
-and users. After a certain time, the driver will go to the alsa-kernel
-tree (to the corresponding directory, such as ``alsa-kernel/pci``) and
-eventually will be integrated into the Linux 2.6 tree (the directory
-would be ``linux/sound/pci``).
+alsa-driver tree, ``sound/pci`` directory in the case of PCI
+cards.
 
 In the following sections, the driver code is supposed to be put into
-alsa-driver tree. The two cases are covered: a driver consisting of a
+Linux kernel tree. The two cases are covered: a driver consisting of a
 single source file and one consisting of several source files.
 
 Driver with A Single Source File
 --------------------------------
 
-1. Modify alsa-driver/pci/Makefile
+1. Modify sound/pci/Makefile
 
    Suppose you have a file xyz.c. Add the following two lines
 
@@ -4160,52 +4163,43 @@ Driver with A Single Source File
 
    For the details of Kconfig script, refer to the kbuild documentation.
 
-3. Run cvscompile script to re-generate the configure script and build
-   the whole stuff again.
-
 Drivers with Several Source Files
 ---------------------------------
 
 Suppose that the driver snd-xyz have several source files. They are
-located in the new subdirectory, pci/xyz.
+located in the new subdirectory, sound/pci/xyz.
 
-1. Add a new directory (``xyz``) in ``alsa-driver/pci/Makefile`` as
-   below
+1. Add a new directory (``sound/pci/xyz``) in ``sound/pci/Makefile``
+   as below
 
 ::
 
-  obj-$(CONFIG_SND) += xyz/
+  obj-$(CONFIG_SND) += sound/pci/xyz/
 
 
-2. Under the directory ``xyz``, create a Makefile
+2. Under the directory ``sound/pci/xyz``, create a Makefile
 
 ::
 
-         ifndef SND_TOPDIR
-         SND_TOPDIR=../..
-         endif
-
-         include $(SND_TOPDIR)/toplevel.config
-         include $(SND_TOPDIR)/Makefile.conf
-
          snd-xyz-objs := xyz.o abc.o def.o
-
          obj-$(CONFIG_SND_XYZ) += snd-xyz.o
 
-         include $(SND_TOPDIR)/Rules.make
-
 3. Create the Kconfig entry
 
    This procedure is as same as in the last section.
 
-4. Run cvscompile script to re-generate the configure script and build
-   the whole stuff again.
 
 Useful Functions
 ================
 
 :c:func:`snd_printk()` and friends
----------------------------------------
+----------------------------------
+
+.. note:: This subsection describes a few helper functions for
+   decorating a bit more on the standard :c:func:`printk()` & co.
+   However, in general, the use of such helpers is no longer recommended.
+   If possible, try to stick with the standard functions like
+   :c:func:`dev_err()` or :c:func:`pr_err()`.
 
 ALSA provides a verbose version of the :c:func:`printk()` function.
 If a kernel config ``CONFIG_SND_VERBOSE_PRINTK`` is set, this function
@@ -4221,13 +4215,10 @@ just like :c:func:`snd_printk()`. If the ALSA is compiled without
 the debugging flag, it's ignored.
 
 :c:func:`snd_printdd()` is compiled in only when
-``CONFIG_SND_DEBUG_VERBOSE`` is set. Please note that
-``CONFIG_SND_DEBUG_VERBOSE`` is not set as default even if you configure
-the alsa-driver with ``--with-debug=full`` option. You need to give
-explicitly ``--with-debug=detect`` option instead.
+``CONFIG_SND_DEBUG_VERBOSE`` is set.
 
 :c:func:`snd_BUG()`
-------------------------
+-------------------
 
 It shows the ``BUG?`` message and stack trace as well as
 :c:func:`snd_BUG_ON()` at the point. It's useful to show that a
@@ -4236,7 +4227,7 @@ fatal error happens there.
 When no debug flag is set, this macro is ignored.
 
 :c:func:`snd_BUG_ON()`
-----------------------------
+----------------------
 
 :c:func:`snd_BUG_ON()` macro is similar with
 :c:func:`WARN_ON()` macro. For example, snd_BUG_ON(!pointer); or
diff --git a/MAINTAINERS b/MAINTAINERS
index 46a7656bf96b..2b928aa857ce 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -14759,6 +14759,13 @@ L:	netdev@vger.kernel.org
 S:      Maintained
 F:      drivers/net/ethernet/ti/netcp*
 
+TI PCM3060 ASoC CODEC DRIVER
+M:      Kirill Marinushkin <kmarinushkin@birdec.tech>
+L:      alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:      Maintained
+F:      Documentation/devicetree/bindings/sound/pcm3060.txt
+F:      sound/soc/codecs/pcm3060*
+
 TI TAS571X FAMILY ASoC CODEC DRIVER
 M:      Kevin Cernekee <cernekee@chromium.org>
 L:      alsa-devel@alsa-project.org (moderated for non-subscribers)
diff --git a/sound/pci/hda/hda_codec.h b/include/sound/hda_codec.h
index 0d98bb9068b1..0d98bb9068b1 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/include/sound/hda_codec.h
diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h
index 67561b997915..af3fa577fa06 100644
--- a/include/sound/memalloc.h
+++ b/include/sound/memalloc.h
@@ -47,10 +47,13 @@ struct snd_dma_device {
 #define SNDRV_DMA_TYPE_UNKNOWN          0       /* not defined */
 #define SNDRV_DMA_TYPE_CONTINUOUS       1       /* continuous no-DMA memory */
 #define SNDRV_DMA_TYPE_DEV              2       /* generic device continuous */
+#define SNDRV_DMA_TYPE_DEV_UC           5       /* continuous non-cahced */
 #ifdef CONFIG_SND_DMA_SGBUF
 #define SNDRV_DMA_TYPE_DEV_SG           3       /* generic device SG-buffer */
+#define SNDRV_DMA_TYPE_DEV_UC_SG        6       /* SG non-cached */
 #else
 #define SNDRV_DMA_TYPE_DEV_SG   SNDRV_DMA_TYPE_DEV /* no SG-buf support */
+#define SNDRV_DMA_TYPE_DEV_UC_SG        SNDRV_DMA_TYPE_DEV_UC
 #endif
 #ifdef CONFIG_GENERIC_ALLOCATOR
 #define SNDRV_DMA_TYPE_DEV_IRAM         4       /* generic device iram-buffer */
diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h
index 6665cb29e1a2..3b5a061132b6 100644
--- a/include/sound/rawmidi.h
+++ b/include/sound/rawmidi.h
@@ -171,6 +171,7 @@ int __snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream,
                               unsigned char *buffer, int count);
 int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream,
                                int count);
+int snd_rawmidi_proceed(struct snd_rawmidi_substream *substream);
 
 /* main midi functions */
 
diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h
index 8bc5e2d8b13c..fb0318f9b10f 100644
--- a/include/sound/simple_card_utils.h
+++ b/include/sound/simple_card_utils.h
@@ -51,29 +51,35 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card,
 
 #define asoc_simple_card_parse_clk_cpu(dev, node, dai_link, simple_dai)         \
         asoc_simple_card_parse_clk(dev, node, dai_link->cpu_of_node, simple_dai, \
-                                   dai_link->cpu_dai_name)
+                                   dai_link->cpu_dai_name, NULL)
 #define asoc_simple_card_parse_clk_codec(dev, node, dai_link, simple_dai)       \
         asoc_simple_card_parse_clk(dev, node, dai_link->codec_of_node, simple_dai,\
-                                   dai_link->codec_dai_name)
+                                   dai_link->codec_dai_name, dai_link->codecs)
 int asoc_simple_card_parse_clk(struct device *dev,
                                struct device_node *node,
                                struct device_node *dai_of_node,
                                struct asoc_simple_dai *simple_dai,
-                               const char *name);
+                               const char *dai_name,
+                               struct snd_soc_dai_link_component *dlc);
 int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai);
 void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai);
 
 #define asoc_simple_card_parse_cpu(node, dai_link,                              \
                                    list_name, cells_name, is_single_link)       \
-        asoc_simple_card_parse_dai(node, &dai_link->cpu_of_node,                \
+        asoc_simple_card_parse_dai(node, NULL,                                  \
+                &dai_link->cpu_of_node,                                         \
                 &dai_link->cpu_dai_name, list_name, cells_name, is_single_link)
 #define asoc_simple_card_parse_codec(node, dai_link, list_name, cells_name)     \
-        asoc_simple_card_parse_dai(node, &dai_link->codec_of_node,              \
-                &dai_link->codec_dai_name, list_name, cells_name, NULL)
+        asoc_simple_card_parse_dai(node, dai_link->codecs,                      \
+                                   &dai_link->codec_of_node,                    \
+                                   &dai_link->codec_dai_name,                   \
+                                   list_name, cells_name, NULL)
 #define asoc_simple_card_parse_platform(node, dai_link, list_name, cells_name)  \
-        asoc_simple_card_parse_dai(node, &dai_link->platform_of_node,           \
+        asoc_simple_card_parse_dai(node, dai_link->platform,                                    \
+                &dai_link->platform_of_node,                                    \
                 NULL, list_name, cells_name, NULL)
 int asoc_simple_card_parse_dai(struct device_node *node,
+                                  struct snd_soc_dai_link_component *dlc,
                                   struct device_node **endpoint_np,
                                   const char **dai_name,
                                   const char *list_name,
@@ -81,12 +87,15 @@ int asoc_simple_card_parse_dai(struct device_node *node,
                                   int *is_single_links);
 
 #define asoc_simple_card_parse_graph_cpu(ep, dai_link)                  \
-        asoc_simple_card_parse_graph_dai(ep, &dai_link->cpu_of_node,    \
+        asoc_simple_card_parse_graph_dai(ep, NULL,                      \
+                                         &dai_link->cpu_of_node,        \
                                          &dai_link->cpu_dai_name)
 #define asoc_simple_card_parse_graph_codec(ep, dai_link)                \
-        asoc_simple_card_parse_graph_dai(ep, &dai_link->codec_of_node,  \
+        asoc_simple_card_parse_graph_dai(ep, dai_link->codecs,          \
+                                         &dai_link->codec_of_node,      \
                                          &dai_link->codec_dai_name)
 int asoc_simple_card_parse_graph_dai(struct device_node *ep,
+                                     struct snd_soc_dai_link_component *dlc,
                                      struct device_node **endpoint_np,
                                      const char **dai_name);
 
diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h
index bb1d24b703fb..f48f59e5b7b0 100644
--- a/include/sound/soc-acpi-intel-match.h
+++ b/include/sound/soc-acpi-intel-match.h
@@ -25,4 +25,10 @@ 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[];
 
+/*
+ * generic table used for HDA codec-based platforms, possibly with
+ * additional ACPI-enumerated codecs
+ */
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[];
+
 #endif
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index fdaaafdc7a00..bd8163f151cb 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -406,12 +406,6 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
                                  struct snd_soc_dai *dai);
 int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card);
 void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card);
-int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
-                         struct snd_soc_pcm_runtime *rtd,
-                         const struct snd_soc_pcm_stream *params,
-                         unsigned int num_params,
-                         struct snd_soc_dapm_widget *source,
-                         struct snd_soc_dapm_widget *sink);
 
 /* dapm path setup */
 int snd_soc_dapm_new_widgets(struct snd_soc_card *card);
@@ -590,9 +584,6 @@ struct snd_soc_dapm_widget {
         void *priv;                             /* widget specific data */
         struct regulator *regulator;            /* attached regulator */
         struct pinctrl *pinctrl;                /* attached pinctrl */
-        const struct snd_soc_pcm_stream *params; /* params for dai links */
-        unsigned int num_params; /* number of params for dai links */
-        unsigned int params_select; /* currently selected param for dai link */
 
         /* dapm control */
         int reg;                                /* negative reg = no direct dapm */
diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h
index 9bb92f187af8..4be3a2b7c106 100644
--- a/include/sound/soc-dpcm.h
+++ b/include/sound/soc-dpcm.h
@@ -103,6 +103,16 @@ struct snd_soc_dpcm_runtime {
         int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */
 };
 
+#define for_each_dpcm_fe(be, stream, dpcm)                              \
+        list_for_each_entry(dpcm, &(be)->dpcm[stream].fe_clients, list_fe)
+
+#define for_each_dpcm_be(fe, stream, dpcm)                              \
+        list_for_each_entry(dpcm, &(fe)->dpcm[stream].be_clients, list_be)
+#define for_each_dpcm_be_safe(fe, stream, dpcm, _dpcm)                  \
+        list_for_each_entry_safe(dpcm, _dpcm, &(fe)->dpcm[stream].be_clients, list_be)
+#define for_each_dpcm_be_rollback(fe, stream, dpcm)                     \
+        list_for_each_entry_continue_reverse(dpcm, &(fe)->dpcm[stream].be_clients, list_be)
+
 /* can this BE stop and free */
 int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
                 struct snd_soc_pcm_runtime *be, int stream);
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 41cec42fb456..f1dab1f4b194 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -372,6 +372,11 @@
 #define SND_SOC_COMP_ORDER_LATE         1
 #define SND_SOC_COMP_ORDER_LAST         2
 
+#define for_each_comp_order(order)              \
+        for (order  = SND_SOC_COMP_ORDER_FIRST; \
+             order <= SND_SOC_COMP_ORDER_LAST;  \
+             order++)
+
 /*
  * Bias levels
  *
@@ -859,6 +864,11 @@ struct snd_soc_component {
 #endif
 };
 
+#define for_each_component_dais(component, dai)\
+        list_for_each_entry(dai, &(component)->dai_list, list)
+#define for_each_component_dais_safe(component, dai, _dai)\
+        list_for_each_entry_safe(dai, _dai, &(component)->dai_list, list)
+
 struct snd_soc_rtdcom_list {
         struct snd_soc_component *component;
         struct list_head list; /* rtd::component_list */
@@ -915,6 +925,8 @@ struct snd_soc_dai_link {
          */
         const char *platform_name;
         struct device_node *platform_of_node;
+        struct snd_soc_dai_link_component *platform;
+
         int id; /* optional ID for machine driver link identification */
 
         const struct snd_soc_pcm_stream *params;
@@ -976,6 +988,10 @@ struct snd_soc_dai_link {
         struct list_head list; /* DAI link list of the soc card */
         struct snd_soc_dobj dobj; /* For topology */
 };
+#define for_each_link_codecs(link, i, codec)                            \
+        for ((i) = 0;                                                   \
+             ((i) < link->num_codecs) && ((codec) = &link->codecs[i]);  \
+             (i)++)
 
 struct snd_soc_codec_conf {
         /*
@@ -1054,7 +1070,6 @@ struct snd_soc_card {
         struct snd_soc_dai_link *dai_link;  /* predefined links only */
         int num_links;  /* predefined links only */
         struct list_head dai_link_list; /* all links */
-        int num_dai_links;
 
         struct list_head rtd_list;
         int num_rtd;
@@ -1092,6 +1107,7 @@ struct snd_soc_card {
 
         /* lists of probed devices belonging to this card */
         struct list_head component_dev_list;
+        struct list_head list;
 
         struct list_head widgets;
         struct list_head paths;
@@ -1114,6 +1130,23 @@ struct snd_soc_card {
 
         void *drvdata;
 };
+#define for_each_card_prelinks(card, i, link)                           \
+        for ((i) = 0;                                                   \
+             ((i) < (card)->num_links) && ((link) = &(card)->dai_link[i]); \
+             (i)++)
+
+#define for_each_card_links(card, link)                         \
+        list_for_each_entry(dai_link, &(card)->dai_link_list, list)
+#define for_each_card_links_safe(card, link, _link)                     \
+        list_for_each_entry_safe(link, _link, &(card)->dai_link_list, list)
+
+#define for_each_card_rtds(card, rtd)                   \
+        list_for_each_entry(rtd, &(card)->rtd_list, list)
+#define for_each_card_rtds_safe(card, rtd, _rtd)        \
+        list_for_each_entry_safe(rtd, _rtd, &(card)->rtd_list, list)
+
+#define for_each_card_components(card, component)                       \
+        list_for_each_entry(component, &(card)->component_dev_list, card_list)
 
 /* SoC machine DAI configuration, glues a codec and cpu DAI together */
 struct snd_soc_pcm_runtime {
@@ -1124,6 +1157,8 @@ struct snd_soc_pcm_runtime {
         enum snd_soc_pcm_subclass pcm_subclass;
         struct snd_pcm_ops ops;
 
+        unsigned int params_select; /* currently selected param for dai link */
+
         /* Dynamic PCM BE runtime data */
         struct snd_soc_dpcm_runtime dpcm[2];
         int fe_compr;
@@ -1152,6 +1187,13 @@ struct snd_soc_pcm_runtime {
         unsigned int dev_registered:1;
         unsigned int pop_wait:1;
 };
+#define for_each_rtd_codec_dai(rtd, i, dai)\
+        for ((i) = 0;                                                  \
+             ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \
+             (i)++)
+#define for_each_rtd_codec_dai_rollback(rtd, i, dai)            \
+        for (; ((i--) >= 0) && ((dai) = rtd->codec_dais[i]);)
+
 
 /* mixer control */
 struct soc_mixer_control {
@@ -1359,6 +1401,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
         INIT_LIST_HEAD(&card->dapm_list);
         INIT_LIST_HEAD(&card->aux_comp_list);
         INIT_LIST_HEAD(&card->component_dev_list);
+        INIT_LIST_HEAD(&card->list);
 }
 
 static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc)
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index ed0a120d4f08..404d4b9ffe76 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -752,7 +752,7 @@ struct snd_timer_info {
 #define SNDRV_TIMER_PSFLG_EARLY_EVENT   (1<<2)  /* write early event to the poll queue */
 
 struct snd_timer_params {
-        unsigned int flags;             /* flags - SNDRV_MIXER_PSFLG_* */
+        unsigned int flags;             /* flags - SNDRV_TIMER_PSFLG_* */
         unsigned int ticks;             /* requested resolution in ticks */
         unsigned int queue_size;        /* total size of queue (32-1024) */
         unsigned int reserved0;         /* reserved, was: failure locations */
diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c
index 000b58522106..bd7c5029fc59 100644
--- a/sound/aoa/soundbus/i2sbus/core.c
+++ b/sound/aoa/soundbus/i2sbus/core.c
@@ -157,18 +157,19 @@ static int i2sbus_add_dev(struct macio_dev *macio,
         struct device_node *child = NULL, *sound = NULL;
         struct resource *r;
         int i, layout = 0, rlen, ok = force;
-        static const char *rnames[] = { "i2sbus: %s (control)",
-                                        "i2sbus: %s (tx)",
-                                        "i2sbus: %s (rx)" };
+        char node_name[6];
+        static const char *rnames[] = { "i2sbus: %pOFn (control)",
+                                        "i2sbus: %pOFn (tx)",
+                                        "i2sbus: %pOFn (rx)" };
         static irq_handler_t ints[] = {
                 i2sbus_bus_intr,
                 i2sbus_tx_intr,
                 i2sbus_rx_intr
         };
 
-        if (strlen(np->name) != 5)
+        if (snprintf(node_name, sizeof(node_name), "%pOFn", np) != 5)
                 return 0;
-        if (strncmp(np->name, "i2s-", 4))
+        if (strncmp(node_name, "i2s-", 4))
                 return 0;
 
         dev = kzalloc(sizeof(struct i2sbus_dev), GFP_KERNEL);
@@ -228,13 +229,13 @@ static int i2sbus_add_dev(struct macio_dev *macio,
         dev->sound.pcmid = -1;
         dev->macio = macio;
         dev->control = control;
-        dev->bus_number = np->name[4] - 'a';
+        dev->bus_number = node_name[4] - 'a';
         INIT_LIST_HEAD(&dev->sound.codec_list);
 
         for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) {
                 dev->interrupts[i] = -1;
                 snprintf(dev->rnames[i], sizeof(dev->rnames[i]),
-                         rnames[i], np->name);
+                         rnames[i], np);
         }
         for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) {
                 int irq = irq_of_parse_and_map(np, i);
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index 5fbd47a9177e..28867732a318 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -31,7 +31,6 @@ endif	# SND_ARM
 
 config SND_PXA2XX_LIB
         tristate
-        select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
         select SND_DMAENGINE_PCM
 
 config SND_PXA2XX_LIB_AC97
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 753d5fc4b284..59a4adc286ed 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -25,6 +25,9 @@
 #include <linux/mm.h>
 #include <linux/dma-mapping.h>
 #include <linux/genalloc.h>
+#ifdef CONFIG_X86
+#include <asm/set_memory.h>
+#endif
 #include <sound/memalloc.h>
 
 /*
@@ -82,31 +85,32 @@ EXPORT_SYMBOL(snd_free_pages);
 
 #ifdef CONFIG_HAS_DMA
 /* allocate the coherent DMA pages */
-static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma)
+static void snd_malloc_dev_pages(struct snd_dma_buffer *dmab, size_t size)
 {
-        int pg;
         gfp_t gfp_flags;
 
-        if (WARN_ON(!dma))
-                return NULL;
-        pg = get_order(size);
         gfp_flags = GFP_KERNEL
                 | __GFP_COMP    /* compound page lets parts be mapped */
                 | __GFP_NORETRY /* don't trigger OOM-killer */
                 | __GFP_NOWARN; /* no stack trace print - this call is non-critical */
-        return dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags);
+        dmab->area = dma_alloc_coherent(dmab->dev.dev, size, &dmab->addr,
+                                        gfp_flags);
+#ifdef CONFIG_X86
+        if (dmab->area && dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC)
+                set_memory_wc((unsigned long)dmab->area,
+                              PAGE_ALIGN(size) >> PAGE_SHIFT);
+#endif
 }
 
 /* free the coherent DMA pages */
-static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr,
-                               dma_addr_t dma)
+static void snd_free_dev_pages(struct snd_dma_buffer *dmab)
 {
-        int pg;
-
-        if (ptr == NULL)
-                return;
-        pg = get_order(size);
-        dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma);
+#ifdef CONFIG_X86
+        if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC)
+                set_memory_wb((unsigned long)dmab->area,
+                              PAGE_ALIGN(dmab->bytes) >> PAGE_SHIFT);
+#endif
+        dma_free_coherent(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr);
 }
 
 #ifdef CONFIG_GENERIC_ALLOCATOR
@@ -199,12 +203,15 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
                  */
                 dmab->dev.type = SNDRV_DMA_TYPE_DEV;
 #endif /* CONFIG_GENERIC_ALLOCATOR */
+                /* fall through */
         case SNDRV_DMA_TYPE_DEV:
-                dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr);
+        case SNDRV_DMA_TYPE_DEV_UC:
+                snd_malloc_dev_pages(dmab, size);
                 break;
 #endif
 #ifdef CONFIG_SND_DMA_SGBUF
         case SNDRV_DMA_TYPE_DEV_SG:
+        case SNDRV_DMA_TYPE_DEV_UC_SG:
                 snd_malloc_sgbuf_pages(device, size, dmab, NULL);
                 break;
 #endif
@@ -275,11 +282,13 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab)
                 break;
 #endif /* CONFIG_GENERIC_ALLOCATOR */
         case SNDRV_DMA_TYPE_DEV:
-                snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr);
+        case SNDRV_DMA_TYPE_DEV_UC:
+                snd_free_dev_pages(dmab);
                 break;
 #endif
 #ifdef CONFIG_SND_DMA_SGBUF
         case SNDRV_DMA_TYPE_DEV_SG:
+        case SNDRV_DMA_TYPE_DEV_UC_SG:
                 snd_free_sgbuf_pages(dmab);
                 break;
 #endif
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index 0391cb1a4f19..141c5f3a9575 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -111,7 +111,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames)
                 while (plugin->next) {
                         if (plugin->dst_frames)
                                 frames = plugin->dst_frames(plugin, frames);
-                        if (snd_BUG_ON(frames <= 0))
+                        if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0))
                                 return -ENXIO;
                         plugin = plugin->next;
                         err = snd_pcm_plugin_alloc(plugin, frames);
@@ -123,7 +123,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames)
                 while (plugin->prev) {
                         if (plugin->src_frames)
                                 frames = plugin->src_frames(plugin, frames);
-                        if (snd_BUG_ON(frames <= 0))
+                        if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0))
                                 return -ENXIO;
                         plugin = plugin->prev;
                         err = snd_pcm_plugin_alloc(plugin, frames);
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 4e6110d778bd..40013b26f671 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -2172,18 +2172,25 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream,
         if (err < 0)
                 goto _end_unlock;
 
+        runtime->twake = runtime->control->avail_min ? : 1;
+        if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
+                snd_pcm_update_hw_ptr(substream);
+
         if (!is_playback &&
-            runtime->status->state == SNDRV_PCM_STATE_PREPARED &&
-            size >= runtime->start_threshold) {
-                err = snd_pcm_start(substream);
-                if (err < 0)
+            runtime->status->state == SNDRV_PCM_STATE_PREPARED) {
+                if (size >= runtime->start_threshold) {
+                        err = snd_pcm_start(substream);
+                        if (err < 0)
+                                goto _end_unlock;
+                } else {
+                        /* nothing to do */
+                        err = 0;
                         goto _end_unlock;
+                }
         }
 
-        runtime->twake = runtime->control->avail_min ? : 1;
-        if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
-                snd_pcm_update_hw_ptr(substream);
         avail = snd_pcm_avail(substream);
+
         while (size > 0) {
                 snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
                 snd_pcm_uframes_t cont;
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 08d5662039e3..ee601d7f0926 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -1236,6 +1236,28 @@ int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream,
 }
 EXPORT_SYMBOL(snd_rawmidi_transmit);
 
+/**
+ * snd_rawmidi_proceed - Discard the all pending bytes and proceed
+ * @substream: rawmidi substream
+ *
+ * Return: the number of discarded bytes
+ */
+int snd_rawmidi_proceed(struct snd_rawmidi_substream *substream)
+{
+        struct snd_rawmidi_runtime *runtime = substream->runtime;
+        unsigned long flags;
+        int count = 0;
+
+        spin_lock_irqsave(&runtime->lock, flags);
+        if (runtime->avail < runtime->buffer_size) {
+                count = runtime->buffer_size - runtime->avail;
+                __snd_rawmidi_transmit_ack(substream, count);
+        }
+        spin_unlock_irqrestore(&runtime->lock, flags);
+        return count;
+}
+EXPORT_SYMBOL(snd_rawmidi_proceed);
+
 static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream,
                                       const unsigned char __user *userbuf,
                                       const unsigned char *kernelbuf,
diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c
index ba127c22539a..0778d28421da 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;
-                        /* fall through and continue to next */
+                        /* fall through */
                 case TMR_WAIT_ABS:
                         if (parm == 0) {
                                 rec->realtime = 1;
diff --git a/sound/core/seq/seq_system.c b/sound/core/seq/seq_system.c
index 8ce1d0b40dce..0dc5d5a45ecc 100644
--- a/sound/core/seq/seq_system.c
+++ b/sound/core/seq/seq_system.c
@@ -123,6 +123,7 @@ int __init snd_seq_system_client_init(void)
 {
         struct snd_seq_port_callback pcallbacks;
         struct snd_seq_port_info *port;
+        int err;
 
         port = kzalloc(sizeof(*port), GFP_KERNEL);
         if (!port)
@@ -134,6 +135,10 @@ int __init snd_seq_system_client_init(void)
 
         /* register client */
         sysclient = snd_seq_create_kernel_client(NULL, 0, "System");
+        if (sysclient < 0) {
+                kfree(port);
+                return sysclient;
+        }
 
         /* register timer */
         strcpy(port->name, "Timer");
@@ -144,7 +149,10 @@ int __init snd_seq_system_client_init(void)
         port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT;
         port->addr.client = sysclient;
         port->addr.port = SNDRV_SEQ_PORT_SYSTEM_TIMER;
-        snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port);
+        err = snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT,
+                                        port);
+        if (err < 0)
+                goto error_port;
 
         /* register announcement port */
         strcpy(port->name, "Announce");
@@ -154,16 +162,24 @@ int __init snd_seq_system_client_init(void)
         port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT;
         port->addr.client = sysclient;
         port->addr.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE;
-        snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port);
+        err = snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT,
+                                        port);
+        if (err < 0)
+                goto error_port;
         announce_port = port->addr.port;
 
         kfree(port);
         return 0;
+
+ error_port:
+        snd_seq_system_client_done();
+        kfree(port);
+        return err;
 }
 
 
 /* unregister our internal client */
-void __exit snd_seq_system_client_done(void)
+void snd_seq_system_client_done(void)
 {
         int oldsysclient = sysclient;
 
diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c
index cb988efd1ed0..e5a40795914a 100644
--- a/sound/core/seq/seq_virmidi.c
+++ b/sound/core/seq/seq_virmidi.c
@@ -149,9 +149,7 @@ static void snd_vmidi_output_work(struct work_struct *work)
         /* discard the outputs in dispatch mode unless subscribed */
         if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH &&
             !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) {
-                char buf[32];
-                while (snd_rawmidi_transmit(substream, buf, sizeof(buf)) > 0)
-                        ; /* ignored */
+                snd_rawmidi_proceed(substream);
                 return;
         }
 
diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c
index 84fffabdd129..c1cfaa01a5cb 100644
--- a/sound/core/sgbuf.c
+++ b/sound/core/sgbuf.c
@@ -23,6 +23,7 @@
 #include <linux/mm.h>
 #include <linux/vmalloc.h>
 #include <linux/export.h>
+#include <asm/pgtable.h>
 #include <sound/memalloc.h>
 
 
@@ -43,6 +44,8 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab)
         dmab->area = NULL;
 
         tmpb.dev.type = SNDRV_DMA_TYPE_DEV;
+        if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC_SG)
+                tmpb.dev.type = SNDRV_DMA_TYPE_DEV_UC;
         tmpb.dev.dev = sgbuf->dev;
         for (i = 0; i < sgbuf->pages; i++) {
                 if (!(sgbuf->table[i].addr & ~PAGE_MASK))
@@ -72,12 +75,20 @@ void *snd_malloc_sgbuf_pages(struct device *device,
         struct snd_dma_buffer tmpb;
         struct snd_sg_page *table;
         struct page **pgtable;
+        int type = SNDRV_DMA_TYPE_DEV;
+        pgprot_t prot = PAGE_KERNEL;
 
         dmab->area = NULL;
         dmab->addr = 0;
         dmab->private_data = sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL);
         if (! sgbuf)
                 return NULL;
+        if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC_SG) {
+                type = SNDRV_DMA_TYPE_DEV_UC;
+#ifdef pgprot_noncached
+                prot = pgprot_noncached(PAGE_KERNEL);
+#endif
+        }
         sgbuf->dev = device;
         pages = snd_sgbuf_aligned_pages(size);
         sgbuf->tblsize = sgbuf_align_table(pages);
@@ -98,7 +109,7 @@ void *snd_malloc_sgbuf_pages(struct device *device,
                 if (chunk > maxpages)
                         chunk = maxpages;
                 chunk <<= PAGE_SHIFT;
-                if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, device,
+                if (snd_dma_alloc_pages_fallback(type, device,
                                                  chunk, &tmpb) < 0) {
                         if (!sgbuf->pages)
                                 goto _failed;
@@ -125,7 +136,7 @@ void *snd_malloc_sgbuf_pages(struct device *device,
         }
 
         sgbuf->size = size;
-        dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, PAGE_KERNEL);
+        dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, prot);
         if (! dmab->area)
                 goto _failed;
         if (res_size)
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index 529d9f405fa9..8a146b039276 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -147,7 +147,9 @@ config SND_FIREWIRE_MOTU
         help
          Say Y here to enable support for FireWire devices which MOTU produced:
           * 828mk2
+          * Traveler
           * 828mk3
+          * Audio Express
 
          To compile this driver as a module, choose M here: the module
          will be called snd-firewire-motu.
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index cb9acfe60f6a..fcd965f1d69e 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -140,6 +140,59 @@ const unsigned int amdtp_rate_table[CIP_SFC_COUNT] = {
 };
 EXPORT_SYMBOL(amdtp_rate_table);
 
+static int apply_constraint_to_size(struct snd_pcm_hw_params *params,
+                                    struct snd_pcm_hw_rule *rule)
+{
+        struct snd_interval *s = hw_param_interval(params, rule->var);
+        const struct snd_interval *r =
+                hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE);
+        struct snd_interval t = {
+                .min = s->min, .max = s->max, .integer = 1,
+        };
+        int i;
+
+        for (i = 0; i < CIP_SFC_COUNT; ++i) {
+                unsigned int rate = amdtp_rate_table[i];
+                unsigned int step = amdtp_syt_intervals[i];
+
+                if (!snd_interval_test(r, rate))
+                        continue;
+
+                t.min = roundup(t.min, step);
+                t.max = rounddown(t.max, step);
+        }
+
+        if (snd_interval_checkempty(&t))
+                return -EINVAL;
+
+        return snd_interval_refine(s, &t);
+}
+
+static int apply_constraint_to_rate(struct snd_pcm_hw_params *params,
+                                    struct snd_pcm_hw_rule *rule)
+{
+        struct snd_interval *r =
+                        hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+        const struct snd_interval *s = hw_param_interval_c(params, rule->deps[0]);
+        struct snd_interval t = {
+                .min = UINT_MAX, .max = 0, .integer = 1,
+        };
+        int i;
+
+        for (i = 0; i < CIP_SFC_COUNT; ++i) {
+                unsigned int step = amdtp_syt_intervals[i];
+                unsigned int rate = amdtp_rate_table[i];
+
+                if (s->min % step || s->max % step)
+                        continue;
+
+                t.min = min(t.min, rate);
+                t.max = max(t.max, rate);
+        }
+
+        return snd_interval_refine(r, &t);
+}
+
 /**
  * amdtp_stream_add_pcm_hw_constraints - add hw constraints for PCM substream
  * @s:          the AMDTP stream, which must be initialized.
@@ -194,16 +247,27 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
          * number equals to SYT_INTERVAL. So the number is 8, 16 or 32,
          * depending on its sampling rate. For accurate period interrupt, it's
          * preferrable to align period/buffer sizes to current SYT_INTERVAL.
-         *
-         * TODO: These constraints can be improved with proper rules.
-         * Currently apply LCM of SYT_INTERVALs.
          */
-        err = snd_pcm_hw_constraint_step(runtime, 0,
-                                         SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32);
+        err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+                                  apply_constraint_to_size, NULL,
+                                  SNDRV_PCM_HW_PARAM_RATE, -1);
+        if (err < 0)
+                goto end;
+        err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+                                  apply_constraint_to_rate, NULL,
+                                  SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1);
+        if (err < 0)
+                goto end;
+        err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+                                  apply_constraint_to_size, NULL,
+                                  SNDRV_PCM_HW_PARAM_RATE, -1);
+        if (err < 0)
+                goto end;
+        err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+                                  apply_constraint_to_rate, NULL,
+                                  SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1);
         if (err < 0)
                 goto end;
-        err = snd_pcm_hw_constraint_step(runtime, 0,
-                                         SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32);
 end:
         return err;
 }
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 93676354f87f..672d13488454 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -126,23 +126,6 @@ end:
         return err;
 }
 
-static void bebob_free(struct snd_bebob *bebob)
-{
-        snd_bebob_stream_destroy_duplex(bebob);
-        fw_unit_put(bebob->unit);
-
-        kfree(bebob->maudio_special_quirk);
-
-        mutex_destroy(&bebob->mutex);
-        kfree(bebob);
-}
-
-/*
- * This module releases the FireWire unit data after all ALSA character devices
- * are released by applications. This is for releasing stream data or finishing
- * transactions safely. Thus at returning from .remove(), this module still keep
- * references for the unit.
- */
 static void
 bebob_card_free(struct snd_card *card)
 {
@@ -152,7 +135,7 @@ bebob_card_free(struct snd_card *card)
         clear_bit(bebob->card_index, devices_used);
         mutex_unlock(&devices_mutex);
 
-        bebob_free(card->private_data);
+        snd_bebob_stream_destroy_duplex(bebob);
 }
 
 static const struct snd_bebob_spec *
@@ -192,7 +175,6 @@ do_registration(struct work_struct *work)
                 return;
 
         mutex_lock(&devices_mutex);
-
         for (card_index = 0; card_index < SNDRV_CARDS; card_index++) {
                 if (!test_bit(card_index, devices_used) && enable[card_index])
                         break;
@@ -208,6 +190,11 @@ do_registration(struct work_struct *work)
                 mutex_unlock(&devices_mutex);
                 return;
         }
+        set_bit(card_index, devices_used);
+        mutex_unlock(&devices_mutex);
+
+        bebob->card->private_free = bebob_card_free;
+        bebob->card->private_data = bebob;
 
         err = name_device(bebob);
         if (err < 0)
@@ -248,23 +235,10 @@ do_registration(struct work_struct *work)
         if (err < 0)
                 goto error;
 
-        set_bit(card_index, devices_used);
-        mutex_unlock(&devices_mutex);
-
-        /*
-         * After registered, bebob instance can be released corresponding to
-         * releasing the sound card instance.
-         */
-        bebob->card->private_free = bebob_card_free;
-        bebob->card->private_data = bebob;
         bebob->registered = true;
 
         return;
 error:
-        mutex_unlock(&devices_mutex);
-        snd_bebob_stream_destroy_duplex(bebob);
-        kfree(bebob->maudio_special_quirk);
-        bebob->maudio_special_quirk = NULL;
         snd_card_free(bebob->card);
         dev_info(&bebob->unit->device,
                  "Sound card registration failed: %d\n", err);
@@ -295,15 +269,15 @@ bebob_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry)
         }
 
         /* Allocate this independent of sound card instance. */
-        bebob = kzalloc(sizeof(struct snd_bebob), GFP_KERNEL);
-        if (bebob == NULL)
+        bebob = devm_kzalloc(&unit->device, sizeof(struct snd_bebob),
+                             GFP_KERNEL);
+        if (!bebob)
                 return -ENOMEM;
-
         bebob->unit = fw_unit_get(unit);
-        bebob->entry = entry;
-        bebob->spec = spec;
         dev_set_drvdata(&unit->device, bebob);
 
+        bebob->entry = entry;
+        bebob->spec = spec;
         mutex_init(&bebob->mutex);
         spin_lock_init(&bebob->lock);
         init_waitqueue_head(&bebob->hwdep_wait);
@@ -379,12 +353,12 @@ static void bebob_remove(struct fw_unit *unit)
         cancel_delayed_work_sync(&bebob->dwork);
 
         if (bebob->registered) {
-                /* No need to wait for releasing card object in this context. */
-                snd_card_free_when_closed(bebob->card);
-        } else {
-                /* Don't forget this case. */
-                bebob_free(bebob);
+                // Block till all of ALSA character devices are released.
+                snd_card_free(bebob->card);
         }
+
+        mutex_destroy(&bebob->mutex);
+        fw_unit_put(bebob->unit);
 }
 
 static const struct snd_bebob_rate_spec normal_rate_spec = {
diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c
index c266997ad299..51152ca4af57 100644
--- a/sound/firewire/bebob/bebob_maudio.c
+++ b/sound/firewire/bebob/bebob_maudio.c
@@ -261,8 +261,9 @@ snd_bebob_maudio_special_discover(struct snd_bebob *bebob, bool is1814)
         struct special_params *params;
         int err;
 
-        params = kzalloc(sizeof(struct special_params), GFP_KERNEL);
-        if (params == NULL)
+        params = devm_kzalloc(&bebob->card->card_dev,
+                              sizeof(struct special_params), GFP_KERNEL);
+        if (!params)
                 return -ENOMEM;
 
         mutex_lock(&bebob->mutex);
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 774eb2205668..0f6dbcffe711 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -122,25 +122,12 @@ static void dice_card_strings(struct snd_dice *dice)
         strcpy(card->mixername, "DICE");
 }
 
-static void dice_free(struct snd_dice *dice)
+static void dice_card_free(struct snd_card *card)
 {
+        struct snd_dice *dice = card->private_data;
+
         snd_dice_stream_destroy_duplex(dice);
         snd_dice_transaction_destroy(dice);
-        fw_unit_put(dice->unit);
-
-        mutex_destroy(&dice->mutex);
-        kfree(dice);
-}
-
-/*
- * This module releases the FireWire unit data after all ALSA character devices
- * are released by applications. This is for releasing stream data or finishing
- * transactions safely. Thus at returning from .remove(), this module still keep
- * references for the unit.
- */
-static void dice_card_free(struct snd_card *card)
-{
-        dice_free(card->private_data);
 }
 
 static void do_registration(struct work_struct *work)
@@ -155,6 +142,8 @@ static void do_registration(struct work_struct *work)
                            &dice->card);
         if (err < 0)
                 return;
+        dice->card->private_free = dice_card_free;
+        dice->card->private_data = dice;
 
         err = snd_dice_transaction_init(dice);
         if (err < 0)
@@ -192,19 +181,10 @@ static void do_registration(struct work_struct *work)
         if (err < 0)
                 goto error;
 
-        /*
-         * After registered, dice instance can be released corresponding to
-         * releasing the sound card instance.
-         */
-        dice->card->private_free = dice_card_free;
-        dice->card->private_data = dice;
         dice->registered = true;
 
         return;
 error:
-        snd_dice_stream_destroy_duplex(dice);
-        snd_dice_transaction_destroy(dice);
-        snd_dice_stream_destroy_duplex(dice);
         snd_card_free(dice->card);
         dev_info(&dice->unit->device,
                  "Sound card registration failed: %d\n", err);
@@ -223,10 +203,9 @@ static int dice_probe(struct fw_unit *unit,
         }
 
         /* Allocate this independent of sound card instance. */
-        dice = kzalloc(sizeof(struct snd_dice), GFP_KERNEL);
-        if (dice == NULL)
+        dice = devm_kzalloc(&unit->device, sizeof(struct snd_dice), GFP_KERNEL);
+        if (!dice)
                 return -ENOMEM;
-
         dice->unit = fw_unit_get(unit);
         dev_set_drvdata(&unit->device, dice);
 
@@ -263,10 +242,10 @@ static void dice_remove(struct fw_unit *unit)
         if (dice->registered) {
                 /* No need to wait for releasing card object in this context. */
                 snd_card_free_when_closed(dice->card);
-        } else {
-                /* Don't forget this case. */
-                dice_free(dice);
         }
+
+        mutex_destroy(&dice->mutex);
+        fw_unit_put(dice->unit);
 }
 
 static void dice_bus_reset(struct fw_unit *unit)
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
index ef689997d6a5..6c6ea149ef6b 100644
--- a/sound/firewire/digi00x/digi00x.c
+++ b/sound/firewire/digi00x/digi00x.c
@@ -41,20 +41,12 @@ static int name_card(struct snd_dg00x *dg00x)
         return 0;
 }
 
-static void dg00x_free(struct snd_dg00x *dg00x)
+static void dg00x_card_free(struct snd_card *card)
 {
+        struct snd_dg00x *dg00x = card->private_data;
+
         snd_dg00x_stream_destroy_duplex(dg00x);
         snd_dg00x_transaction_unregister(dg00x);
-
-        fw_unit_put(dg00x->unit);
-
-        mutex_destroy(&dg00x->mutex);
-        kfree(dg00x);
-}
-
-static void dg00x_card_free(struct snd_card *card)
-{
-        dg00x_free(card->private_data);
 }
 
 static void do_registration(struct work_struct *work)
@@ -70,6 +62,8 @@ static void do_registration(struct work_struct *work)
                            &dg00x->card);
         if (err < 0)
                 return;
+        dg00x->card->private_free = dg00x_card_free;
+        dg00x->card->private_data = dg00x;
 
         err = name_card(dg00x);
         if (err < 0)
@@ -101,14 +95,10 @@ static void do_registration(struct work_struct *work)
         if (err < 0)
                 goto error;
 
-        dg00x->card->private_free = dg00x_card_free;
-        dg00x->card->private_data = dg00x;
         dg00x->registered = true;
 
         return;
 error:
-        snd_dg00x_transaction_unregister(dg00x);
-        snd_dg00x_stream_destroy_duplex(dg00x);
         snd_card_free(dg00x->card);
         dev_info(&dg00x->unit->device,
                  "Sound card registration failed: %d\n", err);
@@ -120,8 +110,9 @@ static int snd_dg00x_probe(struct fw_unit *unit,
         struct snd_dg00x *dg00x;
 
         /* Allocate this independent of sound card instance. */
-        dg00x = kzalloc(sizeof(struct snd_dg00x), GFP_KERNEL);
-        if (dg00x == NULL)
+        dg00x = devm_kzalloc(&unit->device, sizeof(struct snd_dg00x),
+                             GFP_KERNEL);
+        if (!dg00x)
                 return -ENOMEM;
 
         dg00x->unit = fw_unit_get(unit);
@@ -173,12 +164,12 @@ static void snd_dg00x_remove(struct fw_unit *unit)
         cancel_delayed_work_sync(&dg00x->dwork);
 
         if (dg00x->registered) {
-                /* No need to wait for releasing card object in this context. */
-                snd_card_free_when_closed(dg00x->card);
-        } else {
-                /* Don't forget this case. */
-                dg00x_free(dg00x);
+                // Block till all of ALSA character devices are released.
+                snd_card_free(dg00x->card);
         }
+
+        mutex_destroy(&dg00x->mutex);
+        fw_unit_put(dg00x->unit);
 }
 
 static const struct ieee1394_device_id snd_dg00x_id_table[] = {
diff --git a/sound/firewire/fireface/ff.c b/sound/firewire/fireface/ff.c
index 4974bc7980e9..3f61cfeace69 100644
--- a/sound/firewire/fireface/ff.c
+++ b/sound/firewire/fireface/ff.c
@@ -27,20 +27,12 @@ static void name_card(struct snd_ff *ff)
                  dev_name(&ff->unit->device), 100 << fw_dev->max_speed);
 }
 
-static void ff_free(struct snd_ff *ff)
+static void ff_card_free(struct snd_card *card)
 {
+        struct snd_ff *ff = card->private_data;
+
         snd_ff_stream_destroy_duplex(ff);
         snd_ff_transaction_unregister(ff);
-
-        fw_unit_put(ff->unit);
-
-        mutex_destroy(&ff->mutex);
-        kfree(ff);
-}
-
-static void ff_card_free(struct snd_card *card)
-{
-        ff_free(card->private_data);
 }
 
 static void do_registration(struct work_struct *work)
@@ -55,6 +47,8 @@ static void do_registration(struct work_struct *work)
                            &ff->card);
         if (err < 0)
                 return;
+        ff->card->private_free = ff_card_free;
+        ff->card->private_data = ff;
 
         err = snd_ff_transaction_register(ff);
         if (err < 0)
@@ -84,14 +78,10 @@ static void do_registration(struct work_struct *work)
         if (err < 0)
                 goto error;
 
-        ff->card->private_free = ff_card_free;
-        ff->card->private_data = ff;
         ff->registered = true;
 
         return;
 error:
-        snd_ff_transaction_unregister(ff);
-        snd_ff_stream_destroy_duplex(ff);
         snd_card_free(ff->card);
         dev_info(&ff->unit->device,
                  "Sound card registration failed: %d\n", err);
@@ -102,11 +92,9 @@ static int snd_ff_probe(struct fw_unit *unit,
 {
         struct snd_ff *ff;
 
-        ff = kzalloc(sizeof(struct snd_ff), GFP_KERNEL);
-        if (ff == NULL)
+        ff = devm_kzalloc(&unit->device, sizeof(struct snd_ff), GFP_KERNEL);
+        if (!ff)
                 return -ENOMEM;
-
-        /* initialize myself */
         ff->unit = fw_unit_get(unit);
         dev_set_drvdata(&unit->device, ff);
 
@@ -149,12 +137,12 @@ static void snd_ff_remove(struct fw_unit *unit)
         cancel_work_sync(&ff->dwork.work);
 
         if (ff->registered) {
-                /* No need to wait for releasing card object in this context. */
-                snd_card_free_when_closed(ff->card);
-        } else {
-                /* Don't forget this case. */
-                ff_free(ff);
+                // Block till all of ALSA character devices are released.
+                snd_card_free(ff->card);
         }
+
+        mutex_destroy(&ff->mutex);
+        fw_unit_put(ff->unit);
 }
 
 static const struct snd_ff_spec spec_ff400 = {
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index f2d073365cf6..faf0e001c4c5 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -184,36 +184,17 @@ end:
         return err;
 }
 
-static void efw_free(struct snd_efw *efw)
-{
-        snd_efw_stream_destroy_duplex(efw);
-        snd_efw_transaction_remove_instance(efw);
-        fw_unit_put(efw->unit);
-
-        kfree(efw->resp_buf);
-
-        mutex_destroy(&efw->mutex);
-        kfree(efw);
-}
-
-/*
- * This module releases the FireWire unit data after all ALSA character devices
- * are released by applications. This is for releasing stream data or finishing
- * transactions safely. Thus at returning from .remove(), this module still keep
- * references for the unit.
- */
 static void
 efw_card_free(struct snd_card *card)
 {
         struct snd_efw *efw = card->private_data;
 
-        if (efw->card_index >= 0) {
-                mutex_lock(&devices_mutex);
-                clear_bit(efw->card_index, devices_used);
-                mutex_unlock(&devices_mutex);
-        }
+        mutex_lock(&devices_mutex);
+        clear_bit(efw->card_index, devices_used);
+        mutex_unlock(&devices_mutex);
 
-        efw_free(card->private_data);
+        snd_efw_stream_destroy_duplex(efw);
+        snd_efw_transaction_remove_instance(efw);
 }
 
 static void
@@ -226,9 +207,8 @@ do_registration(struct work_struct *work)
         if (efw->registered)
                 return;
 
-        mutex_lock(&devices_mutex);
-
         /* check registered cards */
+        mutex_lock(&devices_mutex);
         for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) {
                 if (!test_bit(card_index, devices_used) && enable[card_index])
                         break;
@@ -244,12 +224,18 @@ do_registration(struct work_struct *work)
                 mutex_unlock(&devices_mutex);
                 return;
         }
+        set_bit(card_index, devices_used);
+        mutex_unlock(&devices_mutex);
+
+        efw->card->private_free = efw_card_free;
+        efw->card->private_data = efw;
 
         /* prepare response buffer */
         snd_efw_resp_buf_size = clamp(snd_efw_resp_buf_size,
                                       SND_EFW_RESPONSE_MAXIMUM_BYTES, 4096U);
-        efw->resp_buf = kzalloc(snd_efw_resp_buf_size, GFP_KERNEL);
-        if (efw->resp_buf == NULL) {
+        efw->resp_buf = devm_kzalloc(&efw->card->card_dev,
+                                     snd_efw_resp_buf_size, GFP_KERNEL);
+        if (!efw->resp_buf) {
                 err = -ENOMEM;
                 goto error;
         }
@@ -284,25 +270,11 @@ do_registration(struct work_struct *work)
         if (err < 0)
                 goto error;
 
-        set_bit(card_index, devices_used);
-        mutex_unlock(&devices_mutex);
-
-        /*
-         * After registered, efw instance can be released corresponding to
-         * releasing the sound card instance.
-         */
-        efw->card->private_free = efw_card_free;
-        efw->card->private_data = efw;
         efw->registered = true;
 
         return;
 error:
-        mutex_unlock(&devices_mutex);
-        snd_efw_transaction_remove_instance(efw);
-        snd_efw_stream_destroy_duplex(efw);
         snd_card_free(efw->card);
-        kfree(efw->resp_buf);
-        efw->resp_buf = NULL;
         dev_info(&efw->unit->device,
                  "Sound card registration failed: %d\n", err);
 }
@@ -312,10 +284,9 @@ efw_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry)
 {
         struct snd_efw *efw;
 
-        efw = kzalloc(sizeof(struct snd_efw), GFP_KERNEL);
+        efw = devm_kzalloc(&unit->device, sizeof(struct snd_efw), GFP_KERNEL);
         if (efw == NULL)
                 return -ENOMEM;
-
         efw->unit = fw_unit_get(unit);
         dev_set_drvdata(&unit->device, efw);
 
@@ -363,12 +334,12 @@ static void efw_remove(struct fw_unit *unit)
         cancel_delayed_work_sync(&efw->dwork);
 
         if (efw->registered) {
-                /* No need to wait for releasing card object in this context. */
-                snd_card_free_when_closed(efw->card);
-        } else {
-                /* Don't forget this case. */
-                efw_free(efw);
+                // Block till all of ALSA character devices are released.
+                snd_card_free(efw->card);
         }
+
+        mutex_destroy(&efw->mutex);
+        fw_unit_put(efw->unit);
 }
 
 static const struct ieee1394_device_id efw_id_table[] = {
diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index 30957477e005..9ebe510ea26b 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -602,8 +602,6 @@ static void isight_card_free(struct snd_card *card)
         struct isight *isight = card->private_data;
 
         fw_iso_resources_destroy(&isight->resources);
-        fw_unit_put(isight->unit);
-        mutex_destroy(&isight->mutex);
 }
 
 static u64 get_unit_base(struct fw_unit *unit)
@@ -640,7 +638,7 @@ static int isight_probe(struct fw_unit *unit,
         if (!isight->audio_base) {
                 dev_err(&unit->device, "audio unit base not found\n");
                 err = -ENXIO;
-                goto err_unit;
+                goto error;
         }
         fw_iso_resources_init(&isight->resources, unit);
 
@@ -669,12 +667,12 @@ static int isight_probe(struct fw_unit *unit,
         dev_set_drvdata(&unit->device, isight);
 
         return 0;
-
-err_unit:
-        fw_unit_put(isight->unit);
-        mutex_destroy(&isight->mutex);
 error:
         snd_card_free(card);
+
+        mutex_destroy(&isight->mutex);
+        fw_unit_put(isight->unit);
+
         return err;
 }
 
@@ -703,7 +701,11 @@ static void isight_remove(struct fw_unit *unit)
         isight_stop_streaming(isight);
         mutex_unlock(&isight->mutex);
 
-        snd_card_free_when_closed(isight->card);
+        // Block till all of ALSA character devices are released.
+        snd_card_free(isight->card);
+
+        mutex_destroy(&isight->mutex);
+        fw_unit_put(isight->unit);
 }
 
 static const struct ieee1394_device_id isight_id_table[] = {
diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c
index 300d31b6f191..220e61926ea4 100644
--- a/sound/firewire/motu/motu.c
+++ b/sound/firewire/motu/motu.c
@@ -52,26 +52,12 @@ static void name_card(struct snd_motu *motu)
                  dev_name(&motu->unit->device), 100 << fw_dev->max_speed);
 }
 
-static void motu_free(struct snd_motu *motu)
+static void motu_card_free(struct snd_card *card)
 {
-        snd_motu_transaction_unregister(motu);
+        struct snd_motu *motu = card->private_data;
 
+        snd_motu_transaction_unregister(motu);
         snd_motu_stream_destroy_duplex(motu);
-        fw_unit_put(motu->unit);
-
-        mutex_destroy(&motu->mutex);
-        kfree(motu);
-}
-
-/*
- * This module releases the FireWire unit data after all ALSA character devices
- * are released by applications. This is for releasing stream data or finishing
- * transactions safely. Thus at returning from .remove(), this module still keep
- * references for the unit.
- */
-static void motu_card_free(struct snd_card *card)
-{
-        motu_free(card->private_data);
 }
 
 static void do_registration(struct work_struct *work)
@@ -86,6 +72,8 @@ static void do_registration(struct work_struct *work)
                            &motu->card);
         if (err < 0)
                 return;
+        motu->card->private_free = motu_card_free;
+        motu->card->private_data = motu;
 
         name_card(motu);
 
@@ -120,18 +108,10 @@ static void do_registration(struct work_struct *work)
         if (err < 0)
                 goto error;
 
-        /*
-         * After registered, motu instance can be released corresponding to
-         * releasing the sound card instance.
-         */
-        motu->card->private_free = motu_card_free;
-        motu->card->private_data = motu;
         motu->registered = true;
 
         return;
 error:
-        snd_motu_transaction_unregister(motu);
-        snd_motu_stream_destroy_duplex(motu);
         snd_card_free(motu->card);
         dev_info(&motu->unit->device,
                  "Sound card registration failed: %d\n", err);
@@ -143,14 +123,13 @@ static int motu_probe(struct fw_unit *unit,
         struct snd_motu *motu;
 
         /* Allocate this independently of sound card instance. */
-        motu = kzalloc(sizeof(struct snd_motu), GFP_KERNEL);
-        if (motu == NULL)
+        motu = devm_kzalloc(&unit->device, sizeof(struct snd_motu), GFP_KERNEL);
+        if (!motu)
                 return -ENOMEM;
-
-        motu->spec = (const struct snd_motu_spec *)entry->driver_data;
         motu->unit = fw_unit_get(unit);
         dev_set_drvdata(&unit->device, motu);
 
+        motu->spec = (const struct snd_motu_spec *)entry->driver_data;
         mutex_init(&motu->mutex);
         spin_lock_init(&motu->lock);
         init_waitqueue_head(&motu->hwdep_wait);
@@ -174,12 +153,12 @@ static void motu_remove(struct fw_unit *unit)
         cancel_delayed_work_sync(&motu->dwork);
 
         if (motu->registered) {
-                /* No need to wait for releasing card object in this context. */
-                snd_card_free_when_closed(motu->card);
-        } else {
-                /* Don't forget this case. */
-                motu_free(motu);
+                // Block till all of ALSA character devices are released.
+                snd_card_free(motu->card);
         }
+
+        mutex_destroy(&motu->mutex);
+        fw_unit_put(motu->unit);
 }
 
 static void motu_bus_update(struct fw_unit *unit)
diff --git a/sound/firewire/oxfw/oxfw-scs1x.c b/sound/firewire/oxfw/oxfw-scs1x.c
index f33497cdc706..9d9545880a28 100644
--- a/sound/firewire/oxfw/oxfw-scs1x.c
+++ b/sound/firewire/oxfw/oxfw-scs1x.c
@@ -372,8 +372,9 @@ int snd_oxfw_scs1x_add(struct snd_oxfw *oxfw)
         struct fw_scs1x *scs;
         int err;
 
-        scs = kzalloc(sizeof(struct fw_scs1x), GFP_KERNEL);
-        if (scs == NULL)
+        scs = devm_kzalloc(&oxfw->card->card_dev, sizeof(struct fw_scs1x),
+                           GFP_KERNEL);
+        if (!scs)
                 return -ENOMEM;
         scs->fw_dev = fw_parent_device(oxfw->unit);
         oxfw->spec = scs;
diff --git a/sound/firewire/oxfw/oxfw-spkr.c b/sound/firewire/oxfw/oxfw-spkr.c
index cb905af0660d..66d4b1f73f0f 100644
--- a/sound/firewire/oxfw/oxfw-spkr.c
+++ b/sound/firewire/oxfw/oxfw-spkr.c
@@ -270,8 +270,9 @@ int snd_oxfw_add_spkr(struct snd_oxfw *oxfw, bool is_lacie)
         unsigned int i, first_ch;
         int err;
 
-        spkr = kzalloc(sizeof(struct fw_spkr), GFP_KERNEL);
-        if (spkr == NULL)
+        spkr = devm_kzalloc(&oxfw->card->card_dev, sizeof(struct fw_spkr),
+                            GFP_KERNEL);
+        if (!spkr)
                 return -ENOMEM;
         oxfw->spec = spkr;
 
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index d9361f352133..f230a9e44c3c 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -517,8 +517,9 @@ assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir,
         if (err < 0)
                 goto end;
 
-        formats[eid] = kmemdup(buf, *len, GFP_KERNEL);
-        if (formats[eid] == NULL) {
+        formats[eid] = devm_kmemdup(&oxfw->card->card_dev, buf, *len,
+                                    GFP_KERNEL);
+        if (!formats[eid]) {
                 err = -ENOMEM;
                 goto end;
         }
@@ -535,7 +536,8 @@ assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir,
                         continue;
 
                 eid++;
-                formats[eid] = kmemdup(buf, *len, GFP_KERNEL);
+                formats[eid] = devm_kmemdup(&oxfw->card->card_dev, buf, *len,
+                                            GFP_KERNEL);
                 if (formats[eid] == NULL) {
                         err = -ENOMEM;
                         goto end;
@@ -597,8 +599,9 @@ static int fill_stream_formats(struct snd_oxfw *oxfw,
                 if (err < 0)
                         break;
 
-                formats[eid] = kmemdup(buf, len, GFP_KERNEL);
-                if (formats[eid] == NULL) {
+                formats[eid] = devm_kmemdup(&oxfw->card->card_dev, buf, len,
+                                            GFP_KERNEL);
+                if (!formats[eid]) {
                         err = -ENOMEM;
                         break;
                 }
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 2ea8be6c8584..afb78d90384b 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -113,35 +113,13 @@ end:
         return err;
 }
 
-static void oxfw_free(struct snd_oxfw *oxfw)
+static void oxfw_card_free(struct snd_card *card)
 {
-        unsigned int i;
+        struct snd_oxfw *oxfw = card->private_data;
 
         snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
         if (oxfw->has_output)
                 snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream);
-
-        fw_unit_put(oxfw->unit);
-
-        for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
-                kfree(oxfw->tx_stream_formats[i]);
-                kfree(oxfw->rx_stream_formats[i]);
-        }
-
-        kfree(oxfw->spec);
-        mutex_destroy(&oxfw->mutex);
-        kfree(oxfw);
-}
-
-/*
- * This module releases the FireWire unit data after all ALSA character devices
- * are released by applications. This is for releasing stream data or finishing
- * transactions safely. Thus at returning from .remove(), this module still keep
- * references for the unit.
- */
-static void oxfw_card_free(struct snd_card *card)
-{
-        oxfw_free(card->private_data);
 }
 
 static int detect_quirks(struct snd_oxfw *oxfw)
@@ -208,7 +186,6 @@ static int detect_quirks(struct snd_oxfw *oxfw)
 static void do_registration(struct work_struct *work)
 {
         struct snd_oxfw *oxfw = container_of(work, struct snd_oxfw, dwork.work);
-        int i;
         int err;
 
         if (oxfw->registered)
@@ -218,6 +195,8 @@ static void do_registration(struct work_struct *work)
                            &oxfw->card);
         if (err < 0)
                 return;
+        oxfw->card->private_free = oxfw_card_free;
+        oxfw->card->private_data = oxfw;
 
         err = name_card(oxfw);
         if (err < 0)
@@ -258,28 +237,11 @@ static void do_registration(struct work_struct *work)
         if (err < 0)
                 goto error;
 
-        /*
-         * After registered, oxfw instance can be released corresponding to
-         * releasing the sound card instance.
-         */
-        oxfw->card->private_free = oxfw_card_free;
-        oxfw->card->private_data = oxfw;
         oxfw->registered = true;
 
         return;
 error:
-        snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
-        if (oxfw->has_output)
-                snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream);
-        for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; ++i) {
-                kfree(oxfw->tx_stream_formats[i]);
-                oxfw->tx_stream_formats[i] = NULL;
-                kfree(oxfw->rx_stream_formats[i]);
-                oxfw->rx_stream_formats[i] = NULL;
-        }
         snd_card_free(oxfw->card);
-        kfree(oxfw->spec);
-        oxfw->spec = NULL;
         dev_info(&oxfw->unit->device,
                  "Sound card registration failed: %d\n", err);
 }
@@ -293,14 +255,13 @@ static int oxfw_probe(struct fw_unit *unit,
                 return -ENODEV;
 
         /* Allocate this independent of sound card instance. */
-        oxfw = kzalloc(sizeof(struct snd_oxfw), GFP_KERNEL);
-        if (oxfw == NULL)
+        oxfw = devm_kzalloc(&unit->device, sizeof(struct snd_oxfw), GFP_KERNEL);
+        if (!oxfw)
                 return -ENOMEM;
-
-        oxfw->entry = entry;
         oxfw->unit = fw_unit_get(unit);
         dev_set_drvdata(&unit->device, oxfw);
 
+        oxfw->entry = entry;
         mutex_init(&oxfw->mutex);
         spin_lock_init(&oxfw->lock);
         init_waitqueue_head(&oxfw->hwdep_wait);
@@ -347,12 +308,12 @@ static void oxfw_remove(struct fw_unit *unit)
         cancel_delayed_work_sync(&oxfw->dwork);
 
         if (oxfw->registered) {
-                /* No need to wait for releasing card object in this context. */
-                snd_card_free_when_closed(oxfw->card);
-        } else {
-                /* Don't forget this case. */
-                oxfw_free(oxfw);
+                // Block till all of ALSA character devices are released.
+                snd_card_free(oxfw->card);
         }
+
+        mutex_destroy(&oxfw->mutex);
+        fw_unit_put(oxfw->unit);
 }
 
 static const struct compat_info griffin_firewave = {
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index d3fdc463a884..ef57fa4db323 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -85,20 +85,12 @@ static int identify_model(struct snd_tscm *tscm)
         return 0;
 }
 
-static void tscm_free(struct snd_tscm *tscm)
+static void tscm_card_free(struct snd_card *card)
 {
+        struct snd_tscm *tscm = card->private_data;
+
         snd_tscm_transaction_unregister(tscm);
         snd_tscm_stream_destroy_duplex(tscm);
-
-        fw_unit_put(tscm->unit);
-
-        mutex_destroy(&tscm->mutex);
-        kfree(tscm);
-}
-
-static void tscm_card_free(struct snd_card *card)
-{
-        tscm_free(card->private_data);
 }
 
 static void do_registration(struct work_struct *work)
@@ -110,6 +102,8 @@ static void do_registration(struct work_struct *work)
                            &tscm->card);
         if (err < 0)
                 return;
+        tscm->card->private_free = tscm_card_free;
+        tscm->card->private_data = tscm;
 
         err = identify_model(tscm);
         if (err < 0)
@@ -141,18 +135,10 @@ static void do_registration(struct work_struct *work)
         if (err < 0)
                 goto error;
 
-        /*
-         * After registered, tscm instance can be released corresponding to
-         * releasing the sound card instance.
-         */
-        tscm->card->private_free = tscm_card_free;
-        tscm->card->private_data = tscm;
         tscm->registered = true;
 
         return;
 error:
-        snd_tscm_transaction_unregister(tscm);
-        snd_tscm_stream_destroy_duplex(tscm);
         snd_card_free(tscm->card);
         dev_info(&tscm->unit->device,
                  "Sound card registration failed: %d\n", err);
@@ -164,11 +150,9 @@ static int snd_tscm_probe(struct fw_unit *unit,
         struct snd_tscm *tscm;
 
         /* Allocate this independent of sound card instance. */
-        tscm = kzalloc(sizeof(struct snd_tscm), GFP_KERNEL);
-        if (tscm == NULL)
+        tscm = devm_kzalloc(&unit->device, sizeof(struct snd_tscm), GFP_KERNEL);
+        if (!tscm)
                 return -ENOMEM;
-
-        /* initialize myself */
         tscm->unit = fw_unit_get(unit);
         dev_set_drvdata(&unit->device, tscm);
 
@@ -216,12 +200,12 @@ static void snd_tscm_remove(struct fw_unit *unit)
         cancel_delayed_work_sync(&tscm->dwork);
 
         if (tscm->registered) {
-                /* No need to wait for releasing card object in this context. */
-                snd_card_free_when_closed(tscm->card);
-        } else {
-                /* Don't forget this case. */
-                tscm_free(tscm);
+                // Block till all of ALSA character devices are released.
+                snd_card_free(tscm->card);
         }
+
+        mutex_destroy(&tscm->mutex);
+        fw_unit_put(tscm->unit);
 }
 
 static const struct ieee1394_device_id snd_tscm_id_table[] = {
diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c
index 5bc4a1d587d4..60cb00fd0c69 100644
--- a/sound/hda/ext/hdac_ext_controller.c
+++ b/sound/hda/ext/hdac_ext_controller.c
@@ -48,9 +48,11 @@ void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *bus, bool enable)
         }
 
         if (enable)
-                snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_GPROCEN);
+                snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL,
+                                 AZX_PPCTL_GPROCEN, AZX_PPCTL_GPROCEN);
         else
-                snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_GPROCEN, 0);
+                snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL,
+                                 AZX_PPCTL_GPROCEN, 0);
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_enable);
 
@@ -68,9 +70,11 @@ void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *bus, bool enable)
         }
 
         if (enable)
-                snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_PIE);
+                snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL,
+                                 AZX_PPCTL_PIE, AZX_PPCTL_PIE);
         else
-                snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_PIE, 0);
+                snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL,
+                                 AZX_PPCTL_PIE, 0);
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_int_enable);
 
@@ -194,7 +198,8 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable)
  */
 int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link)
 {
-        snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA);
+        snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL,
+                         AZX_MLCTL_SPA, AZX_MLCTL_SPA);
 
         return check_hdac_link_power_active(link, true);
 }
@@ -222,8 +227,8 @@ int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus)
         int ret;
 
         list_for_each_entry(hlink, &bus->hlink_list, list) {
-                snd_hdac_updatel(hlink->ml_addr,
-                                AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA);
+                snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL,
+                                 AZX_MLCTL_SPA, AZX_MLCTL_SPA);
                 ret = check_hdac_link_power_active(hlink, true);
                 if (ret < 0)
                         return ret;
@@ -243,7 +248,8 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus)
         int ret;
 
         list_for_each_entry(hlink, &bus->hlink_list, list) {
-                snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0);
+                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)
                         return ret;
diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c
index 2647309bc675..8afa2f888466 100644
--- a/sound/i2c/cs8427.c
+++ b/sound/i2c/cs8427.c
@@ -118,7 +118,7 @@ static int snd_cs8427_send_corudata(struct snd_i2c_device *device,
         struct cs8427 *chip = device->private_data;
         char *hw_data = udata ?
                 chip->playback.hw_udata : chip->playback.hw_status;
-        char data[32];
+        unsigned char data[32];
         int err, idx;
 
         if (!memcmp(hw_data, ndata, count))
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index ac0ab6eb40f0..47e0b2820ace 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -389,7 +389,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip,
         case OPTi9XX_HW_82C931:
                 /* disable 3D sound (set GPIO1 as output, low) */
                 snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c);
-        case OPTi9XX_HW_82C933: /* FALL THROUGH */
+                /* fall through */
+        case OPTi9XX_HW_82C933:
                 /*
                  * The BTC 1817DW has QS1000 wavetable which is connected
                  * to the serial digital input of the OPTI931.
@@ -400,7 +401,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip,
                  * or digital input signal.
                  */
                 snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01);
-        case OPTi9XX_HW_82C930: /* FALL THROUGH */
+                /* fall through */
+        case OPTi9XX_HW_82C930:
                 snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03);
                 snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff);
                 snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 |
diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c
index 481797744b3c..8288fae90085 100644
--- a/sound/isa/sb/sb8_main.c
+++ b/sound/isa/sb/sb8_main.c
@@ -130,13 +130,13 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream)
                         chip->playback_format = SB_DSP_HI_OUTPUT_AUTO;
                         break;
                 }
-                /* fallthru */
+                /* fall through */
         case SB_HW_201:
                 if (rate > 23000) {
                         chip->playback_format = SB_DSP_HI_OUTPUT_AUTO;
                         break;
                 }
-                /* fallthru */
+                /* fall through */
         case SB_HW_20:
                 chip->playback_format = SB_DSP_LO_OUTPUT_AUTO;
                 break;
@@ -287,7 +287,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream)
                         chip->capture_format = SB_DSP_HI_INPUT_AUTO;
                         break;
                 }
-                /* fallthru */
+                /* fall through */
         case SB_HW_20:
                 chip->capture_format = SB_DSP_LO_INPUT_AUTO;
                 break;
@@ -387,7 +387,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip)
         case SB_MODE_PLAYBACK_16:       /* ok.. playback is active */
                 if (chip->hardware != SB_HW_JAZZ16)
                         break;
-                /* fallthru */
+                /* fall through */
         case SB_MODE_PLAYBACK_8:
                 substream = chip->playback_substream;
                 if (chip->playback_format == SB_DSP_OUTPUT)
@@ -397,7 +397,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip)
         case SB_MODE_CAPTURE_16:
                 if (chip->hardware != SB_HW_JAZZ16)
                         break;
-                /* fallthru */
+                /* fall through */
         case SB_MODE_CAPTURE_8:
                 substream = chip->capture_substream;
                 if (chip->capture_format == SB_DSP_INPUT)
diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c
index c8904e732aaa..a4ed54aeaf1d 100644
--- a/sound/mips/hal2.c
+++ b/sound/mips/hal2.c
@@ -500,7 +500,8 @@ static const struct snd_pcm_hardware hal2_pcm_hw = {
         .info = (SNDRV_PCM_INFO_MMAP |
                  SNDRV_PCM_INFO_MMAP_VALID |
                  SNDRV_PCM_INFO_INTERLEAVED |
-                 SNDRV_PCM_INFO_BLOCK_TRANSFER),
+                 SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                 SNDRV_PCM_INFO_SYNC_APPLPTR),
         .formats =          SNDRV_PCM_FMTBIT_S16_BE,
         .rates =            SNDRV_PCM_RATE_8000_48000,
         .rate_min =         8000,
@@ -563,6 +564,8 @@ static int hal2_playback_prepare(struct snd_pcm_substream *substream)
         dac->sample_rate = hal2_compute_rate(dac, runtime->rate);
         memset(&dac->pcm_indirect, 0, sizeof(dac->pcm_indirect));
         dac->pcm_indirect.hw_buffer_size = H2_BUF_SIZE;
+        dac->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
+        dac->pcm_indirect.hw_io = dac->buffer_dma;
         dac->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
         dac->substream = substream;
         hal2_setup_dac(hal2);
@@ -575,9 +578,6 @@ static int hal2_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 
         switch (cmd) {
         case SNDRV_PCM_TRIGGER_START:
-                hal2->dac.pcm_indirect.hw_io = hal2->dac.buffer_dma;
-                hal2->dac.pcm_indirect.hw_data = 0;
-                substream->ops->ack(substream);
                 hal2_start_dac(hal2);
                 break;
         case SNDRV_PCM_TRIGGER_STOP:
@@ -615,7 +615,6 @@ static int hal2_playback_ack(struct snd_pcm_substream *substream)
         struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
         struct hal2_codec *dac = &hal2->dac;
 
-        dac->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
         return snd_pcm_indirect_playback_transfer(substream,
                                                   &dac->pcm_indirect,
                                                   hal2_playback_transfer);
@@ -655,6 +654,7 @@ static int hal2_capture_prepare(struct snd_pcm_substream *substream)
         memset(&adc->pcm_indirect, 0, sizeof(adc->pcm_indirect));
         adc->pcm_indirect.hw_buffer_size = H2_BUF_SIZE;
         adc->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
+        adc->pcm_indirect.hw_io = adc->buffer_dma;
         adc->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
         adc->substream = substream;
         hal2_setup_adc(hal2);
@@ -667,9 +667,6 @@ static int hal2_capture_trigger(struct snd_pcm_substream *substream, int cmd)
 
         switch (cmd) {
         case SNDRV_PCM_TRIGGER_START:
-                hal2->adc.pcm_indirect.hw_io = hal2->adc.buffer_dma;
-                hal2->adc.pcm_indirect.hw_data = 0;
-                printk(KERN_DEBUG "buffer_dma %x\n", hal2->adc.buffer_dma);
                 hal2_start_adc(hal2);
                 break;
         case SNDRV_PCM_TRIGGER_STOP:
diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c
index 5ef4fe964366..7c91330af719 100644
--- a/sound/pci/asihpi/hpios.c
+++ b/sound/pci/asihpi/hpios.c
@@ -49,7 +49,7 @@ u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
         /*?? any benefit in using managed dmam_alloc_coherent? */
         p_mem_area->vaddr =
                 dma_alloc_coherent(&pdev->dev, size, &p_mem_area->dma_handle,
-                GFP_DMA32 | GFP_KERNEL);
+                GFP_KERNEL);
 
         if (p_mem_area->vaddr) {
                 HPI_DEBUG_LOG(DEBUG, "allocated %d bytes, dma 0x%x vma %p\n",
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index a1e4944dcfe8..1a41f8c80243 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -903,15 +903,15 @@ static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream)
         case 8:
                 data |= ATI_REG_OUT_DMA_SLOT_BIT(10) |
                         ATI_REG_OUT_DMA_SLOT_BIT(11);
-                /* fallthru */
+                /* fall through */
         case 6:
                 data |= ATI_REG_OUT_DMA_SLOT_BIT(7) |
                         ATI_REG_OUT_DMA_SLOT_BIT(8);
-                /* fallthru */
+                /* fall through */
         case 4:
                 data |= ATI_REG_OUT_DMA_SLOT_BIT(6) |
                         ATI_REG_OUT_DMA_SLOT_BIT(9);
-                /* fallthru */
+                /* fall through */
         default:
                 data |= ATI_REG_OUT_DMA_SLOT_BIT(3) |
                         ATI_REG_OUT_DMA_SLOT_BIT(4);
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 2e5b460a847c..96ece1a71cf1 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1115,6 +1115,7 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma,
                 hwwrite(vortex->mmio,
                         VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0xc,
                         snd_pcm_sgbuf_get_addr(dma->substream, psize * 3));
+                /* fall through */
                 /* 3 pages */
         case 3:
                 dma->cfg0 |= 0x12000000;
@@ -1122,12 +1123,14 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma,
                 hwwrite(vortex->mmio,
                         VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0x8,
                         snd_pcm_sgbuf_get_addr(dma->substream, psize * 2));
+                /* fall through */
                 /* 2 pages */
         case 2:
                 dma->cfg0 |= 0x88000000 | 0x44000000 | 0x10000000 | (psize - 1);
                 hwwrite(vortex->mmio,
                         VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0x4,
                         snd_pcm_sgbuf_get_addr(dma->substream, psize));
+                /* fall through */
                 /* 1 page */
         case 1:
                 dma->cfg0 |= 0x80000000 | 0x40000000 | ((psize - 1) << 0xc);
@@ -1390,17 +1393,20 @@ vortex_wtdma_setbuffers(vortex_t * vortex, int wtdma,
                 dma->cfg1 |= 0x88000000 | 0x44000000 | 0x30000000 | (psize-1);
                 hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0xc,
                         snd_pcm_sgbuf_get_addr(dma->substream, psize * 3));
+                /* fall through */
                 /* 3 pages */
         case 3:
                 dma->cfg0 |= 0x12000000;
                 dma->cfg1 |= 0x80000000 | 0x40000000 | ((psize-1) << 0xc);
                 hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4)  + 0x8,
                         snd_pcm_sgbuf_get_addr(dma->substream, psize * 2));
+                /* fall through */
                 /* 2 pages */
         case 2:
                 dma->cfg0 |= 0x88000000 | 0x44000000 | 0x10000000 | (psize-1);
                 hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0x4,
                         snd_pcm_sgbuf_get_addr(dma->substream, psize));
+                /* fall through */
                 /* 1 page */
         case 1:
                 dma->cfg0 |= 0x80000000 | 0x40000000 | ((psize-1) << 0xc);
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 146e1a3498c7..750eec437a79 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -1443,7 +1443,8 @@ static const struct snd_pcm_hardware snd_cs46xx_playback =
         .info =                 (SNDRV_PCM_INFO_MMAP |
                                  SNDRV_PCM_INFO_INTERLEAVED | 
                                  SNDRV_PCM_INFO_BLOCK_TRANSFER /*|*/
-                                 /*SNDRV_PCM_INFO_RESUME*/),
+                                 /*SNDRV_PCM_INFO_RESUME*/ |
+                                 SNDRV_PCM_INFO_SYNC_APPLPTR),
         .formats =              (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |
                                  SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
                                  SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE),
@@ -1465,7 +1466,8 @@ static const struct snd_pcm_hardware snd_cs46xx_capture =
         .info =                 (SNDRV_PCM_INFO_MMAP |
                                  SNDRV_PCM_INFO_INTERLEAVED |
                                  SNDRV_PCM_INFO_BLOCK_TRANSFER /*|*/
-                                 /*SNDRV_PCM_INFO_RESUME*/),
+                                 /*SNDRV_PCM_INFO_RESUME*/ |
+                                 SNDRV_PCM_INFO_SYNC_APPLPTR),
         .formats =              SNDRV_PCM_FMTBIT_S16_LE,
         .rates =                SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
         .rate_min =             5500,
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index 9f2b6097f486..30b3472d0b75 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -1753,7 +1753,8 @@ static const struct snd_pcm_hardware snd_emu10k1_fx8010_playback =
 {
         .info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
                                  SNDRV_PCM_INFO_RESUME |
-                                 /* SNDRV_PCM_INFO_MMAP_VALID | */ SNDRV_PCM_INFO_PAUSE),
+                                 /* SNDRV_PCM_INFO_MMAP_VALID | */ SNDRV_PCM_INFO_PAUSE |
+                                 SNDRV_PCM_INFO_SYNC_APPLPTR),
         .formats =              SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
         .rates =                SNDRV_PCM_RATE_48000,
         .rate_min =             48000,
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index b9a6b66aeb0e..df0d636145f8 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -13,7 +13,7 @@
 #include <linux/export.h>
 #include <linux/sort.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index d1a6a9c1329a..f1457c6b3969 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -9,7 +9,7 @@
 #ifndef __SOUND_HDA_BEEP_H
 #define __SOUND_HDA_BEEP_H
 
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 
 #define HDA_BEEP_MODE_OFF       0
 #define HDA_BEEP_MODE_ON        1
diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index d361bb77ca00..9174f1b3a987 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -11,7 +11,7 @@
 #include <linux/pm.h>
 #include <linux/pm_runtime.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 
 /*
@@ -81,6 +81,12 @@ static int hda_codec_driver_probe(struct device *dev)
         hda_codec_patch_t patch;
         int err;
 
+        if (codec->bus->core.ext_ops) {
+                if (WARN_ON(!codec->bus->core.ext_ops->hdev_attach))
+                        return -EINVAL;
+                return codec->bus->core.ext_ops->hdev_attach(&codec->core);
+        }
+
         if (WARN_ON(!codec->preset))
                 return -EINVAL;
 
@@ -134,6 +140,12 @@ static int hda_codec_driver_remove(struct device *dev)
 {
         struct hda_codec *codec = dev_to_hda_codec(dev);
 
+        if (codec->bus->core.ext_ops) {
+                if (WARN_ON(!codec->bus->core.ext_ops->hdev_detach))
+                        return -EINVAL;
+                return codec->bus->core.ext_ops->hdev_detach(&codec->core);
+        }
+
         if (codec->patch_ops.free)
                 codec->patch_ops.free(codec);
         snd_hda_codec_cleanup_for_unbind(codec);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 26d348b47867..0957813939e5 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -27,7 +27,7 @@
 #include <linux/pm.h>
 #include <linux/pm_runtime.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include <sound/asoundef.h>
 #include <sound/tlv.h>
 #include <sound/initval.h>
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index a12e594d4e3b..fe2506672a72 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -130,8 +130,9 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
         azx_dev->core.bufsize = 0;
         azx_dev->core.period_bytes = 0;
         azx_dev->core.format_val = 0;
-        ret = chip->ops->substream_alloc_pages(chip, substream,
-                                          params_buffer_bytes(hw_params));
+        ret = snd_pcm_lib_malloc_pages(substream,
+                                       params_buffer_bytes(hw_params));
+
 unlock:
         dsp_unlock(azx_dev);
         return ret;
@@ -141,7 +142,6 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
         struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
         struct azx_dev *azx_dev = get_azx_dev(substream);
-        struct azx *chip = apcm->chip;
         struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream);
         int err;
 
@@ -152,7 +152,7 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
 
         snd_hda_codec_cleanup(apcm->codec, hinfo, substream);
 
-        err = chip->ops->substream_free_pages(chip, substream);
+        err = snd_pcm_lib_free_pages(substream);
         azx_stream(azx_dev)->prepared = 0;
         dsp_unlock(azx_dev);
         return err;
@@ -732,6 +732,7 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec,
         int pcm_dev = cpcm->device;
         unsigned int size;
         int s, err;
+        int type = SNDRV_DMA_TYPE_DEV_SG;
 
         list_for_each_entry(apcm, &chip->pcm_list, list) {
                 if (apcm->pcm->device == pcm_dev) {
@@ -770,7 +771,9 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec,
         size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024;
         if (size > MAX_PREALLOC_SIZE)
                 size = MAX_PREALLOC_SIZE;
-        snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+        if (chip->uc_buffer)
+                type = SNDRV_DMA_TYPE_DEV_UC_SG;
+        snd_pcm_lib_preallocate_pages_for_all(pcm, type,
                                               chip->card->dev,
                                               size, MAX_PREALLOC_SIZE);
         return 0;
@@ -1220,27 +1223,6 @@ void snd_hda_bus_reset(struct hda_bus *bus)
         bus->in_reset = 0;
 }
 
-static int get_jackpoll_interval(struct azx *chip)
-{
-        int i;
-        unsigned int j;
-
-        if (!chip->jackpoll_ms)
-                return 0;
-
-        i = chip->jackpoll_ms[chip->dev_index];
-        if (i == 0)
-                return 0;
-        if (i < 50 || i > 60000)
-                j = 0;
-        else
-                j = msecs_to_jiffies(i);
-        if (j == 0)
-                dev_warn(chip->card->dev,
-                         "jackpoll_ms value out of range: %d\n", i);
-        return j;
-}
-
 /* HD-audio bus initialization */
 int azx_bus_init(struct azx *chip, const char *model,
                  const struct hdac_io_ops *io_ops)
@@ -1323,7 +1305,7 @@ int azx_probe_codecs(struct azx *chip, unsigned int max_slots)
                         err = snd_hda_codec_new(&chip->bus, chip->card, c, &codec);
                         if (err < 0)
                                 continue;
-                        codec->jackpoll_interval = get_jackpoll_interval(chip);
+                        codec->jackpoll_interval = chip->jackpoll_interval;
                         codec->beep_mode = chip->beep_mode;
                         codecs++;
                 }
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index a68e75b00ea3..c95097bb5a0c 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -20,7 +20,7 @@
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/initval.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include <sound/hda_register.h>
 
 #define AZX_MAX_CODECS          HDA_MAX_CODECS
@@ -76,7 +76,6 @@ struct azx_dev {
          *  when link position is not greater than FIFO size
          */
         unsigned int insufficient:1;
-        unsigned int wc_marked:1;
 };
 
 #define azx_stream(dev)         (&(dev)->core)
@@ -88,11 +87,6 @@ struct azx;
 struct hda_controller_ops {
         /* Disable msi if supported, PCI only */
         int (*disable_msi_reset_irq)(struct azx *);
-        int (*substream_alloc_pages)(struct azx *chip,
-                                     struct snd_pcm_substream *substream,
-                                     size_t size);
-        int (*substream_free_pages)(struct azx *chip,
-                                    struct snd_pcm_substream *substream);
         void (*pcm_mmap_prepare)(struct snd_pcm_substream *substream,
                                  struct vm_area_struct *area);
         /* Check if current position is acceptable */
@@ -127,7 +121,7 @@ struct azx {
         int capture_streams;
         int capture_index_offset;
         int num_streams;
-        const int *jackpoll_ms; /* per-card jack poll interval */
+        int jackpoll_interval; /* jack poll interval in jiffies */
 
         /* Register interaction. */
         const struct hda_controller_ops *ops;
@@ -160,6 +154,7 @@ struct azx {
         unsigned int msi:1;
         unsigned int probing:1; /* codec probing phase */
         unsigned int snoop:1;
+        unsigned int uc_buffer:1; /* non-cached pages for stream buffers */
         unsigned int align_buffer_size:1;
         unsigned int region_requested:1;
         unsigned int disabled:1; /* disabled by vga_switcheroo */
@@ -175,11 +170,10 @@ struct azx {
 #define azx_bus(chip)   (&(chip)->bus.core)
 #define bus_to_azx(_bus)        container_of(_bus, struct azx, bus.core)
 
-#ifdef CONFIG_X86
-#define azx_snoop(chip)         ((chip)->snoop)
-#else
-#define azx_snoop(chip)         true
-#endif
+static inline bool azx_snoop(struct azx *chip)
+{
+        return !IS_ENABLED(CONFIG_X86) || chip->snoop;
+}
 
 /*
  * macros for easy use
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index ba7fe9b6655c..806b12ed44a2 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -27,7 +27,7 @@
 #include <sound/core.h>
 #include <asm/unaligned.h>
 #include <sound/hda_chmap.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 
 enum eld_versions {
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 579984ecdec3..276150f29cda 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -32,7 +32,7 @@
 #include <sound/core.h>
 #include <sound/jack.h>
 #include <sound/tlv.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index cc009a4a3d1d..268bba6ec985 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -23,7 +23,7 @@
 #include <linux/compat.h>
 #include <linux/nospec.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include <sound/hda_hwdep.h>
 #include <sound/minors.h>
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index aa4c672dbaf7..d8eb2b5f51ae 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -63,7 +63,7 @@
 #include <linux/vgaarb.h>
 #include <linux/vga_switcheroo.h>
 #include <linux/firmware.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_controller.h"
 #include "hda_intel.h"
 
@@ -399,61 +399,6 @@ static char *driver_short_names[] = {
         [AZX_DRIVER_GENERIC] = "HD-Audio Generic",
 };
 
-#ifdef CONFIG_X86
-static void __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, bool on)
-{
-        int pages;
-
-        if (azx_snoop(chip))
-                return;
-        if (!dmab || !dmab->area || !dmab->bytes)
-                return;
-
-#ifdef CONFIG_SND_DMA_SGBUF
-        if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_SG) {
-                struct snd_sg_buf *sgbuf = dmab->private_data;
-                if (chip->driver_type == AZX_DRIVER_CMEDIA)
-                        return; /* deal with only CORB/RIRB buffers */
-                if (on)
-                        set_pages_array_wc(sgbuf->page_table, sgbuf->pages);
-                else
-                        set_pages_array_wb(sgbuf->page_table, sgbuf->pages);
-                return;
-        }
-#endif
-
-        pages = (dmab->bytes + PAGE_SIZE - 1) >> PAGE_SHIFT;
-        if (on)
-                set_memory_wc((unsigned long)dmab->area, pages);
-        else
-                set_memory_wb((unsigned long)dmab->area, pages);
-}
-
-static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf,
-                                 bool on)
-{
-        __mark_pages_wc(chip, buf, on);
-}
-static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev,
-                                   struct snd_pcm_substream *substream, bool on)
-{
-        if (azx_dev->wc_marked != on) {
-                __mark_pages_wc(chip, snd_pcm_get_dma_buf(substream), on);
-                azx_dev->wc_marked = on;
-        }
-}
-#else
-/* NOP for other archs */
-static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf,
-                                 bool on)
-{
-}
-static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev,
-                                   struct snd_pcm_substream *substream, bool on)
-{
-}
-#endif
-
 static int azx_acquire_irq(struct azx *chip, int do_disconnect);
 static void set_default_power_save(struct azx *chip);
 
@@ -1678,6 +1623,7 @@ static void azx_check_snoop_available(struct azx *chip)
                 dev_info(chip->card->dev, "Force to %s mode by module option\n",
                          snoop ? "snoop" : "non-snoop");
                 chip->snoop = snoop;
+                chip->uc_buffer = !snoop;
                 return;
         }
 
@@ -1698,8 +1644,12 @@ static void azx_check_snoop_available(struct azx *chip)
                 snoop = false;
 
         chip->snoop = snoop;
-        if (!snoop)
+        if (!snoop) {
                 dev_info(chip->card->dev, "Force to non-snoop mode\n");
+                /* C-Media requires non-cached pages only for CORB/RIRB */
+                if (chip->driver_type != AZX_DRIVER_CMEDIA)
+                        chip->uc_buffer = true;
+        }
 }
 
 static void azx_probe_work(struct work_struct *work)
@@ -1767,7 +1717,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci,
         chip->driver_type = driver_caps & 0xff;
         check_msi(chip);
         chip->dev_index = dev;
-        chip->jackpoll_ms = jackpoll_ms;
+        if (jackpoll_ms[dev] >= 50 && jackpoll_ms[dev] <= 60000)
+                chip->jackpoll_interval = msecs_to_jiffies(jackpoll_ms[dev]);
         INIT_LIST_HEAD(&chip->pcm_list);
         INIT_WORK(&hda->irq_pending_work, azx_irq_pending_work);
         INIT_LIST_HEAD(&hda->list);
@@ -2090,55 +2041,24 @@ static int dma_alloc_pages(struct hdac_bus *bus,
                            struct snd_dma_buffer *buf)
 {
         struct azx *chip = bus_to_azx(bus);
-        int err;
 
-        err = snd_dma_alloc_pages(type,
-                                  bus->dev,
-                                  size, buf);
-        if (err < 0)
-                return err;
-        mark_pages_wc(chip, buf, true);
-        return 0;
+        if (!azx_snoop(chip) && type == SNDRV_DMA_TYPE_DEV)
+                type = SNDRV_DMA_TYPE_DEV_UC;
+        return snd_dma_alloc_pages(type, bus->dev, size, buf);
 }
 
 static void dma_free_pages(struct hdac_bus *bus, struct snd_dma_buffer *buf)
 {
-        struct azx *chip = bus_to_azx(bus);
-
-        mark_pages_wc(chip, buf, false);
         snd_dma_free_pages(buf);
 }
 
-static int substream_alloc_pages(struct azx *chip,
-                                 struct snd_pcm_substream *substream,
-                                 size_t size)
-{
-        struct azx_dev *azx_dev = get_azx_dev(substream);
-        int ret;
-
-        mark_runtime_wc(chip, azx_dev, substream, false);
-        ret = snd_pcm_lib_malloc_pages(substream, size);
-        if (ret < 0)
-                return ret;
-        mark_runtime_wc(chip, azx_dev, substream, true);
-        return 0;
-}
-
-static int substream_free_pages(struct azx *chip,
-                                struct snd_pcm_substream *substream)
-{
-        struct azx_dev *azx_dev = get_azx_dev(substream);
-        mark_runtime_wc(chip, azx_dev, substream, false);
-        return snd_pcm_lib_free_pages(substream);
-}
-
 static void pcm_mmap_prepare(struct snd_pcm_substream *substream,
                              struct vm_area_struct *area)
 {
 #ifdef CONFIG_X86
         struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
         struct azx *chip = apcm->chip;
-        if (!azx_snoop(chip) && chip->driver_type != AZX_DRIVER_CMEDIA)
+        if (chip->uc_buffer)
                 area->vm_page_prot = pgprot_writecombine(area->vm_page_prot);
 #endif
 }
@@ -2156,8 +2076,6 @@ static const struct hdac_io_ops pci_hda_io_ops = {
 
 static const struct hda_controller_ops pci_hda_ops = {
         .disable_msi_reset_irq = disable_msi_reset_irq,
-        .substream_alloc_pages = substream_alloc_pages,
-        .substream_free_pages = substream_free_pages,
         .pcm_mmap_prepare = pcm_mmap_prepare,
         .position_check = azx_position_check,
         .link_power = azx_intel_link_power,
@@ -2257,8 +2175,12 @@ static struct snd_pci_quirk power_save_blacklist[] = {
         /* https://bugzilla.redhat.com/show_bug.cgi?id=1581607 */
         SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", 0),
         /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
+        SND_PCI_QUIRK(0x1028, 0x0497, "Dell Precision T3600", 0),
+        /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
         /* Note the P55A-UD3 and Z87-D3HP share the subsys id for the HDA dev */
         SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P55A-UD3 / Z87-D3HP", 0),
+        /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
+        SND_PCI_QUIRK(0x8086, 0x2040, "Intel DZ77BH-55K", 0),
         /* https://bugzilla.kernel.org/show_bug.cgi?id=199607 */
         SND_PCI_QUIRK(0x8086, 0x2057, "Intel NUC5i7RYB", 0),
         /* https://bugzilla.redhat.com/show_bug.cgi?id=1520902 */
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index a33234e04d4f..c499727920e6 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -15,7 +15,7 @@
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/jack.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index c6b778b2580c..a65740419650 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -25,7 +25,7 @@
 #include <linux/slab.h>
 #include <sound/core.h>
 #include <linux/module.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 
 static int dump_coef = -1;
diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c
index 6ec79c58d48d..c154b19a0c45 100644
--- a/sound/pci/hda/hda_sysfs.c
+++ b/sound/pci/hda/hda_sysfs.c
@@ -14,7 +14,7 @@
 #include <linux/string.h>
 #include <linux/export.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include <sound/hda_hwdep.h>
 #include <sound/minors.h>
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 0621920f7617..dd7d4242d6d2 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -35,7 +35,7 @@
 #include <sound/core.h>
 #include <sound/initval.h>
 
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_controller.h"
 
 /* Defines for Nvidia Tegra HDA support */
@@ -99,19 +99,6 @@ static void dma_free_pages(struct hdac_bus *bus, struct snd_dma_buffer *buf)
         snd_dma_free_pages(buf);
 }
 
-static int substream_alloc_pages(struct azx *chip,
-                                 struct snd_pcm_substream *substream,
-                                 size_t size)
-{
-        return snd_pcm_lib_malloc_pages(substream, size);
-}
-
-static int substream_free_pages(struct azx *chip,
-                                struct snd_pcm_substream *substream)
-{
-        return snd_pcm_lib_free_pages(substream);
-}
-
 /*
  * Register access ops. Tegra HDA register access is DWORD only.
  */
@@ -180,10 +167,7 @@ static const struct hdac_io_ops hda_tegra_io_ops = {
         .dma_free_pages = dma_free_pages,
 };
 
-static const struct hda_controller_ops hda_tegra_ops = {
-        .substream_alloc_pages = substream_alloc_pages,
-        .substream_free_pages = substream_free_pages,
-};
+static const struct hda_controller_ops hda_tegra_ops; /* nothing special */
 
 static void hda_tegra_init(struct hda_tegra *hda)
 {
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index fd476fb40e1b..ebfd0be885b3 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -24,7 +24,7 @@
 #include <linux/module.h>
 
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_beep.h"
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index c2d9ee9cfdc0..21d0f0610913 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -22,7 +22,7 @@
 #include <linux/slab.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 0166a3d7cd55..0a24037184c3 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -31,8 +31,9 @@
 #include <linux/types.h>
 #include <linux/io.h>
 #include <linux/pci.h>
+#include <asm/io.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
@@ -81,12 +82,12 @@
 #define SCP_GET    1
 
 #define EFX_FILE   "ctefx.bin"
-#define SBZ_EFX_FILE   "ctefx-sbz.bin"
+#define DESKTOP_EFX_FILE   "ctefx-desktop.bin"
 #define R3DI_EFX_FILE  "ctefx-r3di.bin"
 
 #ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP
 MODULE_FIRMWARE(EFX_FILE);
-MODULE_FIRMWARE(SBZ_EFX_FILE);
+MODULE_FIRMWARE(DESKTOP_EFX_FILE);
 MODULE_FIRMWARE(R3DI_EFX_FILE);
 #endif
 
@@ -152,7 +153,10 @@ enum {
         XBASS_XOVER,
         EQ_PRESET_ENUM,
         SMART_VOLUME_ENUM,
-        MIC_BOOST_ENUM
+        MIC_BOOST_ENUM,
+        AE5_HEADPHONE_GAIN_ENUM,
+        AE5_SOUND_FILTER_ENUM,
+        ZXR_HEADPHONE_GAIN
 #define EFFECTS_COUNT  (EFFECT_END_NID - EFFECT_START_NID)
 };
 
@@ -666,6 +670,65 @@ static const struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = {
         }
 };
 
+/* Values for ca0113_mmio_command_set for selecting output. */
+#define AE5_CA0113_OUT_SET_COMMANDS 6
+struct ae5_ca0113_output_set {
+        unsigned int group[AE5_CA0113_OUT_SET_COMMANDS];
+        unsigned int target[AE5_CA0113_OUT_SET_COMMANDS];
+        unsigned int vals[AE5_CA0113_OUT_SET_COMMANDS];
+};
+
+static const struct ae5_ca0113_output_set ae5_ca0113_output_presets[] = {
+        { .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+          .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+          .vals =   { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f }
+        },
+        { .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+          .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+          .vals =   { 0x3f, 0x3f, 0x00, 0x00, 0x00, 0x00 }
+        },
+        { .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+          .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+          .vals =   { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f }
+        }
+};
+
+/* ae5 ca0113 command sequences to set headphone gain levels. */
+#define AE5_HEADPHONE_GAIN_PRESET_MAX_COMMANDS 4
+struct ae5_headphone_gain_set {
+        char *name;
+        unsigned int vals[AE5_HEADPHONE_GAIN_PRESET_MAX_COMMANDS];
+};
+
+static const struct ae5_headphone_gain_set ae5_headphone_gain_presets[] = {
+        { .name = "Low (16-31",
+          .vals = { 0xff, 0x2c, 0xf5, 0x32 }
+        },
+        { .name = "Medium (32-149",
+          .vals = { 0x38, 0xa8, 0x3e, 0x4c }
+        },
+        { .name = "High (150-600",
+          .vals = { 0xff, 0xff, 0xff, 0x7f }
+        }
+};
+
+struct ae5_filter_set {
+        char *name;
+        unsigned int val;
+};
+
+static const struct ae5_filter_set ae5_filter_presets[] = {
+        { .name = "Slow Roll Off",
+          .val = 0xa0
+        },
+        { .name = "Minimum Phase",
+          .val = 0xc0
+        },
+        { .name = "Fast Roll Off",
+          .val = 0x80
+        }
+};
+
 enum hda_cmd_vendor_io {
         /* for DspIO node */
         VENDOR_DSPIO_SCP_WRITE_DATA_LOW      = 0x000,
@@ -685,6 +748,9 @@ enum hda_cmd_vendor_io {
         VENDOR_CHIPIO_DATA_LOW               = 0x300,
         VENDOR_CHIPIO_DATA_HIGH              = 0x400,
 
+        VENDOR_CHIPIO_8051_WRITE_DIRECT      = 0x500,
+        VENDOR_CHIPIO_8051_READ_DIRECT       = 0xD00,
+
         VENDOR_CHIPIO_GET_PARAMETER          = 0xF00,
         VENDOR_CHIPIO_STATUS                 = 0xF01,
         VENDOR_CHIPIO_HIC_POST_READ          = 0x702,
@@ -692,6 +758,9 @@ enum hda_cmd_vendor_io {
 
         VENDOR_CHIPIO_8051_DATA_WRITE        = 0x707,
         VENDOR_CHIPIO_8051_DATA_READ         = 0xF07,
+        VENDOR_CHIPIO_8051_PMEM_READ         = 0xF08,
+        VENDOR_CHIPIO_8051_IRAM_WRITE        = 0x709,
+        VENDOR_CHIPIO_8051_IRAM_READ         = 0xF09,
 
         VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE   = 0x70A,
         VENDOR_CHIPIO_CT_EXTENSIONS_GET      = 0xF0A,
@@ -798,6 +867,12 @@ enum control_param_id {
          * impedance is selected*/
         CONTROL_PARAM_PORTD_160OHM_GAIN        = 10,
 
+        /*
+         * This control param name was found in the 8051 memory, and makes
+         * sense given the fact the AE-5 uses it and has the ASI flag set.
+         */
+        CONTROL_PARAM_ASI                      = 23,
+
         /* Stream Control */
 
         /* Select stream with the given ID */
@@ -955,7 +1030,11 @@ struct ca0132_spec {
         long eq_preset_val;
         unsigned int tlv[4];
         struct hda_vmaster_mute_hook vmaster_mute;
-
+        /* AE-5 Control values */
+        unsigned char ae5_headphone_gain_val;
+        unsigned char ae5_filter_val;
+        /* ZxR Control Values */
+        unsigned char zxr_gain_set;
 
         struct hda_codec *codec;
         struct delayed_work unsol_hp_work;
@@ -995,8 +1074,11 @@ enum {
         QUIRK_ALIENWARE,
         QUIRK_ALIENWARE_M17XR4,
         QUIRK_SBZ,
+        QUIRK_ZXR,
+        QUIRK_ZXR_DBPRO,
         QUIRK_R3DI,
         QUIRK_R3D,
+        QUIRK_AE5,
 };
 
 static const struct hda_pintbl alienware_pincfgs[] = {
@@ -1028,6 +1110,21 @@ static const struct hda_pintbl sbz_pincfgs[] = {
         {}
 };
 
+/* Sound Blaster ZxR pin configs taken from Windows Driver */
+static const struct hda_pintbl zxr_pincfgs[] = {
+        { 0x0b, 0x01047110 }, /* Port G -- Lineout FRONT L/R */
+        { 0x0c, 0x414510f0 }, /* SPDIF Out 1 - Disabled*/
+        { 0x0d, 0x014510f0 }, /* Digital Out */
+        { 0x0e, 0x41c520f0 }, /* SPDIF In - Disabled*/
+        { 0x0f, 0x0122711f }, /* Port A -- BackPanel HP */
+        { 0x10, 0x01017111 }, /* Port D -- Center/LFE */
+        { 0x11, 0x01017114 }, /* Port B -- LineMicIn2 / Rear L/R */
+        { 0x12, 0x01a271f0 }, /* Port C -- LineIn1 */
+        { 0x13, 0x908700f0 }, /* What U Hear In*/
+        { 0x18, 0x50d000f0 }, /* N/A */
+        {}
+};
+
 /* Recon3D pin configs taken from Windows Driver */
 static const struct hda_pintbl r3d_pincfgs[] = {
         { 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */
@@ -1043,6 +1140,21 @@ static const struct hda_pintbl r3d_pincfgs[] = {
         {}
 };
 
+/* Sound Blaster AE-5 pin configs taken from Windows Driver */
+static const struct hda_pintbl ae5_pincfgs[] = {
+        { 0x0b, 0x01017010 }, /* Port G -- Lineout FRONT L/R */
+        { 0x0c, 0x014510f0 }, /* SPDIF Out 1 */
+        { 0x0d, 0x014510f0 }, /* Digital Out */
+        { 0x0e, 0x01c510f0 }, /* SPDIF In */
+        { 0x0f, 0x01017114 }, /* Port A -- Rear L/R. */
+        { 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */
+        { 0x11, 0x01a170ff }, /* Port B -- LineMicIn2 / Rear Headphone */
+        { 0x12, 0x01a170f0 }, /* 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 */
@@ -1069,6 +1181,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
         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),
+        SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5),
         {}
 };
 
@@ -1454,6 +1567,20 @@ static void chipio_set_conn_rate(struct hda_codec *codec,
 }
 
 /*
+ * Writes to the 8051's internal address space directly instead of indirectly,
+ * giving access to the special function registers located at addresses
+ * 0x80-0xFF.
+ */
+static void chipio_8051_write_direct(struct hda_codec *codec,
+                unsigned int addr, unsigned int data)
+{
+        unsigned int verb;
+
+        verb = VENDOR_CHIPIO_8051_WRITE_DIRECT | data;
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, verb, addr);
+}
+
+/*
  * Enable clocks.
  */
 static void chipio_enable_clocks(struct hda_codec *codec)
@@ -3088,7 +3215,9 @@ static bool dspload_wait_loaded(struct hda_codec *codec)
 }
 
 /*
- * Setup GPIO for the other variants of Core3D.
+ * ca0113 related functions. The ca0113 acts as the HDA bus for the pci-e
+ * based cards, and has a second mmio region, region2, that's used for special
+ * commands.
  */
 
 /*
@@ -3096,8 +3225,11 @@ static bool dspload_wait_loaded(struct hda_codec *codec)
  * 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.
+ * AE-5 note: The AE-5 seems to use pins 2 and 3 to somehow set the color value
+ * of the on-card LED. It seems to use pin 2 for data, then toggles 3 to on and
+ * then off to send that bit.
  */
-static void ca0132_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin,
+static void ca0113_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin,
                 bool enable)
 {
         struct ca0132_spec *spec = codec->spec;
@@ -3110,6 +3242,89 @@ static void ca0132_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin,
 }
 
 /*
+ * Special pci region2 commands that are only used by the AE-5. They follow
+ * a set format, and require reads at certain points to seemingly 'clear'
+ * the response data. My first tests didn't do these reads, and would cause
+ * the card to get locked up until the memory was read. These commands
+ * seem to work with three distinct values that I've taken to calling group,
+ * target-id, and value.
+ */
+static void ca0113_mmio_command_set(struct hda_codec *codec, unsigned int group,
+                unsigned int target, unsigned int value)
+{
+        struct ca0132_spec *spec = codec->spec;
+        unsigned int write_val;
+
+        writel(0x0000007e, spec->mem_base + 0x210);
+        readl(spec->mem_base + 0x210);
+        writel(0x0000005a, spec->mem_base + 0x210);
+        readl(spec->mem_base + 0x210);
+        readl(spec->mem_base + 0x210);
+
+        writel(0x00800005, spec->mem_base + 0x20c);
+        writel(group, spec->mem_base + 0x804);
+
+        writel(0x00800005, spec->mem_base + 0x20c);
+        write_val = (target & 0xff);
+        write_val |= (value << 8);
+
+
+        writel(write_val, spec->mem_base + 0x204);
+        /*
+         * Need delay here or else it goes too fast and works inconsistently.
+         */
+        msleep(20);
+
+        readl(spec->mem_base + 0x860);
+        readl(spec->mem_base + 0x854);
+        readl(spec->mem_base + 0x840);
+
+        writel(0x00800004, spec->mem_base + 0x20c);
+        writel(0x00000000, spec->mem_base + 0x210);
+        readl(spec->mem_base + 0x210);
+        readl(spec->mem_base + 0x210);
+}
+
+/*
+ * This second type of command is used for setting the sound filter type.
+ */
+static void ca0113_mmio_command_set_type2(struct hda_codec *codec,
+                unsigned int group, unsigned int target, unsigned int value)
+{
+        struct ca0132_spec *spec = codec->spec;
+        unsigned int write_val;
+
+        writel(0x0000007e, spec->mem_base + 0x210);
+        readl(spec->mem_base + 0x210);
+        writel(0x0000005a, spec->mem_base + 0x210);
+        readl(spec->mem_base + 0x210);
+        readl(spec->mem_base + 0x210);
+
+        writel(0x00800003, spec->mem_base + 0x20c);
+        writel(group, spec->mem_base + 0x804);
+
+        writel(0x00800005, spec->mem_base + 0x20c);
+        write_val = (target & 0xff);
+        write_val |= (value << 8);
+
+
+        writel(write_val, spec->mem_base + 0x204);
+        msleep(20);
+        readl(spec->mem_base + 0x860);
+        readl(spec->mem_base + 0x854);
+        readl(spec->mem_base + 0x840);
+
+        writel(0x00800004, spec->mem_base + 0x20c);
+        writel(0x00000000, spec->mem_base + 0x210);
+        readl(spec->mem_base + 0x210);
+        readl(spec->mem_base + 0x210);
+}
+
+/*
+ * Setup GPIO for the other variants of Core3D.
+ */
+
+/*
  * Sets up the GPIO pins so that they are discoverable. If this isn't done,
  * the card shows as having no GPIO pins.
  */
@@ -3119,6 +3334,7 @@ static void ca0132_gpio_init(struct hda_codec *codec)
 
         switch (spec->quirk) {
         case QUIRK_SBZ:
+        case QUIRK_AE5:
                 snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
                 snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
                 snd_hda_codec_write(codec, 0x01, 0, 0x790, 0x23);
@@ -3928,6 +4144,138 @@ exit:
         return err < 0 ? err : 0;
 }
 
+static int ae5_headphone_gain_set(struct hda_codec *codec, long val);
+static int zxr_headphone_gain_set(struct hda_codec *codec, long val);
+static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val);
+
+static void ae5_mmio_select_out(struct hda_codec *codec)
+{
+        struct ca0132_spec *spec = codec->spec;
+        unsigned int i;
+
+        for (i = 0; i < AE5_CA0113_OUT_SET_COMMANDS; i++)
+                ca0113_mmio_command_set(codec,
+                        ae5_ca0113_output_presets[spec->cur_out_type].group[i],
+                        ae5_ca0113_output_presets[spec->cur_out_type].target[i],
+                        ae5_ca0113_output_presets[spec->cur_out_type].vals[i]);
+}
+
+/*
+ * These are the commands needed to setup output on each of the different card
+ * types.
+ */
+static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec)
+{
+        struct ca0132_spec *spec = codec->spec;
+        unsigned int tmp;
+
+        switch (spec->cur_out_type) {
+        case SPEAKER_OUT:
+                switch (spec->quirk) {
+                case QUIRK_SBZ:
+                        ca0113_mmio_gpio_set(codec, 7, false);
+                        ca0113_mmio_gpio_set(codec, 4, true);
+                        ca0113_mmio_gpio_set(codec, 1, true);
+                        chipio_set_control_param(codec, 0x0d, 0x18);
+                        break;
+                case QUIRK_ZXR:
+                        ca0113_mmio_gpio_set(codec, 2, true);
+                        ca0113_mmio_gpio_set(codec, 3, true);
+                        ca0113_mmio_gpio_set(codec, 5, false);
+                        zxr_headphone_gain_set(codec, 0);
+                        chipio_set_control_param(codec, 0x0d, 0x24);
+                        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);
+                        ca0113_mmio_gpio_set(codec, 1, true);
+                        break;
+                case QUIRK_AE5:
+                        ae5_mmio_select_out(codec);
+                        ae5_headphone_gain_set(codec, 2);
+                        tmp = FLOAT_ZERO;
+                        dspio_set_uint_param(codec, 0x96, 0x29, tmp);
+                        dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
+                        chipio_set_control_param(codec, 0x0d, 0xa4);
+                        chipio_write(codec, 0x18b03c, 0x00000012);
+                        break;
+                }
+                break;
+        case HEADPHONE_OUT:
+                switch (spec->quirk) {
+                case QUIRK_SBZ:
+                        ca0113_mmio_gpio_set(codec, 7, true);
+                        ca0113_mmio_gpio_set(codec, 4, true);
+                        ca0113_mmio_gpio_set(codec, 1, false);
+                        chipio_set_control_param(codec, 0x0d, 0x12);
+                        break;
+                case QUIRK_ZXR:
+                        ca0113_mmio_gpio_set(codec, 2, false);
+                        ca0113_mmio_gpio_set(codec, 3, false);
+                        ca0113_mmio_gpio_set(codec, 5, true);
+                        zxr_headphone_gain_set(codec, spec->zxr_gain_set);
+                        chipio_set_control_param(codec, 0x0d, 0x21);
+                        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);
+                        ca0113_mmio_gpio_set(codec, 0x1, false);
+                        break;
+                case QUIRK_AE5:
+                        ae5_mmio_select_out(codec);
+                        ae5_headphone_gain_set(codec,
+                                        spec->ae5_headphone_gain_val);
+                        tmp = FLOAT_ONE;
+                        dspio_set_uint_param(codec, 0x96, 0x29, tmp);
+                        dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
+                        chipio_set_control_param(codec, 0x0d, 0xa1);
+                        chipio_write(codec, 0x18b03c, 0x00000012);
+                        break;
+                }
+                break;
+        case SURROUND_OUT:
+                switch (spec->quirk) {
+                case QUIRK_SBZ:
+                        ca0113_mmio_gpio_set(codec, 7, false);
+                        ca0113_mmio_gpio_set(codec, 4, true);
+                        ca0113_mmio_gpio_set(codec, 1, true);
+                        chipio_set_control_param(codec, 0x0d, 0x18);
+                        break;
+                case QUIRK_ZXR:
+                        ca0113_mmio_gpio_set(codec, 2, true);
+                        ca0113_mmio_gpio_set(codec, 3, true);
+                        ca0113_mmio_gpio_set(codec, 5, false);
+                        zxr_headphone_gain_set(codec, 0);
+                        chipio_set_control_param(codec, 0x0d, 0x24);
+                        break;
+                case QUIRK_R3DI:
+                        chipio_set_control_param(codec, 0x0d, 0x24);
+                        r3di_gpio_out_set(codec, R3DI_LINE_OUT);
+                        break;
+                case QUIRK_R3D:
+                        ca0113_mmio_gpio_set(codec, 1, true);
+                        chipio_set_control_param(codec, 0x0d, 0x24);
+                        break;
+                case QUIRK_AE5:
+                        ae5_mmio_select_out(codec);
+                        ae5_headphone_gain_set(codec, 2);
+                        tmp = FLOAT_ZERO;
+                        dspio_set_uint_param(codec, 0x96, 0x29, tmp);
+                        dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
+                        chipio_set_control_param(codec, 0x0d, 0xa4);
+                        chipio_write(codec, 0x18b03c, 0x00000012);
+                        break;
+                }
+                break;
+        }
+}
+
 /*
  * This function behaves similarly to the ca0132_select_out funciton above,
  * except with a few differences. It adds the ability to select the current
@@ -3978,26 +4326,11 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
         if (err < 0)
                 goto exit;
 
+        ca0132_alt_select_out_quirk_handler(codec);
+
         switch (spec->cur_out_type) {
         case SPEAKER_OUT:
                 codec_dbg(codec, "%s speaker\n", __func__);
-                /*speaker out config*/
-                switch (spec->quirk) {
-                case QUIRK_SBZ:
-                        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 */
                 pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0,
@@ -4021,23 +4354,6 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
                 break;
         case HEADPHONE_OUT:
                 codec_dbg(codec, "%s hp\n", __func__);
-                /* Headphone out config*/
-                switch (spec->quirk) {
-                case QUIRK_SBZ:
-                        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,
                         AC_VERB_SET_EAPD_BTLENABLE, 0x00);
@@ -4067,23 +4383,7 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
                 break;
         case SURROUND_OUT:
                 codec_dbg(codec, "%s surround\n", __func__);
-                /* Surround out config*/
-                switch (spec->quirk) {
-                case QUIRK_SBZ:
-                        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,
                                 AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
@@ -4108,14 +4408,21 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
                 snd_hda_set_pin_ctl(codec, spec->out_pins[3],
                                     pin_ctl | PIN_OUT);
 
-                if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
-                        dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
-                else
-                        dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
+                dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
                 break;
         }
+        /*
+         * Surround always sets it's scp command to req 0x04 to FLOAT_EIGHT.
+         * With this set though, X_BASS cannot be enabled. So, if we have OutFX
+         * enabled, we need to make sure X_BASS is off, otherwise everything
+         * sounds all muffled. Running ca0132_effects_set with X_BASS as the
+         * effect should sort this out.
+         */
+        if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
+                ca0132_effects_set(codec, X_BASS,
+                        spec->effects_switch[X_BASS - EFFECT_START_NID]);
 
-        /* run through the output dsp commands for line-out */
+        /* run through the output dsp commands for the selected output. */
         for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; i++) {
                 err = dspio_set_uint_param(codec,
                 alt_out_presets[spec->cur_out_type].mids[i],
@@ -4152,7 +4459,6 @@ static void ca0132_unsol_hp_delayed(struct work_struct *work)
 
 static void ca0132_set_dmic(struct hda_codec *codec, int enable);
 static int ca0132_mic_boost_set(struct hda_codec *codec, long val);
-static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val);
 static void resume_mic1(struct hda_codec *codec, unsigned int oldval);
 static int stop_mic1(struct hda_codec *codec);
 static int ca0132_cvoice_switch_set(struct hda_codec *codec);
@@ -4341,13 +4647,20 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
                 switch (spec->quirk) {
                 case QUIRK_SBZ:
                 case QUIRK_R3D:
-                        ca0132_mmio_gpio_set(codec, 0, false);
+                        ca0113_mmio_gpio_set(codec, 0, false);
+                        tmp = FLOAT_THREE;
+                        break;
+                case QUIRK_ZXR:
                         tmp = FLOAT_THREE;
                         break;
                 case QUIRK_R3DI:
                         r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
                         tmp = FLOAT_ONE;
                         break;
+                case QUIRK_AE5:
+                        ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00);
+                        tmp = FLOAT_THREE;
+                        break;
                 default:
                         tmp = FLOAT_ONE;
                         break;
@@ -4362,10 +4675,19 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 
                 chipio_set_stream_control(codec, 0x03, 1);
                 chipio_set_stream_control(codec, 0x04, 1);
-
-                if (spec->quirk == QUIRK_SBZ) {
+                switch (spec->quirk) {
+                case QUIRK_SBZ:
                         chipio_write(codec, 0x18B098, 0x0000000C);
                         chipio_write(codec, 0x18B09C, 0x0000000C);
+                        break;
+                case QUIRK_ZXR:
+                        chipio_write(codec, 0x18B098, 0x0000000C);
+                        chipio_write(codec, 0x18B09C, 0x000000CC);
+                        break;
+                case QUIRK_AE5:
+                        chipio_write(codec, 0x18B098, 0x0000000C);
+                        chipio_write(codec, 0x18B09C, 0x0000004C);
+                        break;
                 }
                 ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
                 break;
@@ -4374,11 +4696,14 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
                 switch (spec->quirk) {
                 case QUIRK_SBZ:
                 case QUIRK_R3D:
-                        ca0132_mmio_gpio_set(codec, 0, false);
+                        ca0113_mmio_gpio_set(codec, 0, false);
                         break;
                 case QUIRK_R3DI:
                         r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
                         break;
+                case QUIRK_AE5:
+                        ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00);
+                        break;
                 }
 
                 chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
@@ -4389,11 +4714,13 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
                 tmp = FLOAT_ZERO;
                 dspio_set_uint_param(codec, 0x80, 0x00, tmp);
 
-                if (spec->quirk == QUIRK_SBZ) {
+                switch (spec->quirk) {
+                case QUIRK_SBZ:
+                case QUIRK_AE5:
                         chipio_write(codec, 0x18B098, 0x00000000);
                         chipio_write(codec, 0x18B09C, 0x00000000);
+                        break;
                 }
-
                 chipio_set_stream_control(codec, 0x03, 1);
                 chipio_set_stream_control(codec, 0x04, 1);
                 break;
@@ -4401,14 +4728,18 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
                 switch (spec->quirk) {
                 case QUIRK_SBZ:
                 case QUIRK_R3D:
-                        ca0132_mmio_gpio_set(codec, 0, true);
-                        ca0132_mmio_gpio_set(codec, 5, false);
+                        ca0113_mmio_gpio_set(codec, 0, true);
+                        ca0113_mmio_gpio_set(codec, 5, false);
                         tmp = FLOAT_THREE;
                         break;
                 case QUIRK_R3DI:
                         r3di_gpio_mic_set(codec, R3DI_FRONT_MIC);
                         tmp = FLOAT_ONE;
                         break;
+                case QUIRK_AE5:
+                        ca0113_mmio_command_set(codec, 0x48, 0x28, 0x3f);
+                        tmp = FLOAT_THREE;
+                        break;
                 default:
                         tmp = FLOAT_ONE;
                         break;
@@ -4424,9 +4755,15 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
                 chipio_set_stream_control(codec, 0x03, 1);
                 chipio_set_stream_control(codec, 0x04, 1);
 
-                if (spec->quirk == QUIRK_SBZ) {
+                switch (spec->quirk) {
+                case QUIRK_SBZ:
                         chipio_write(codec, 0x18B098, 0x0000000C);
                         chipio_write(codec, 0x18B09C, 0x000000CC);
+                        break;
+                case QUIRK_AE5:
+                        chipio_write(codec, 0x18B098, 0x0000000C);
+                        chipio_write(codec, 0x18B09C, 0x0000004C);
+                        break;
                 }
                 ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
                 break;
@@ -4435,7 +4772,6 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 
         snd_hda_power_down_pm(codec);
         return 0;
-
 }
 
 /*
@@ -4507,6 +4843,8 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
                 /* if PE if off, turn off out effects. */
                 if (!spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
                         val = 0;
+                if (spec->cur_out_type == SURROUND_OUT && nid == X_BASS)
+                        val = 0;
         }
 
         /* for in effect, qualify with CrystalVoice */
@@ -4520,7 +4858,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
                         val = 0;
 
                 /* If Voice Focus on SBZ, set to two channel. */
-                if ((nid == VOICE_FOCUS) && (spec->quirk == QUIRK_SBZ)
+                if ((nid == VOICE_FOCUS) && (spec->use_pci_mmio)
                                 && (spec->cur_mic_type != REAR_LINE_IN)) {
                         if (spec->effects_switch[CRYSTAL_VOICE -
                                                  EFFECT_START_NID]) {
@@ -4539,7 +4877,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
                  * For SBZ noise reduction, there's an extra command
                  * to module ID 0x47. No clue why.
                  */
-                if ((nid == NOISE_REDUCTION) && (spec->quirk == QUIRK_SBZ)
+                if ((nid == NOISE_REDUCTION) && (spec->use_pci_mmio)
                                 && (spec->cur_mic_type != REAR_LINE_IN)) {
                         if (spec->effects_switch[CRYSTAL_VOICE -
                                                  EFFECT_START_NID]) {
@@ -4678,6 +5016,27 @@ static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val)
         return ret;
 }
 
+static int ae5_headphone_gain_set(struct hda_codec *codec, long val)
+{
+        unsigned int i;
+
+        for (i = 0; i < 4; i++)
+                ca0113_mmio_command_set(codec, 0x48, 0x11 + i,
+                                ae5_headphone_gain_presets[val].vals[i]);
+        return 0;
+}
+
+/*
+ * gpio pin 1 is a relay that switches on/off, apparently setting the headphone
+ * amplifier to handle a 600 ohm load.
+ */
+static int zxr_headphone_gain_set(struct hda_codec *codec, long val)
+{
+        ca0113_mmio_gpio_set(codec, 1, val);
+
+        return 0;
+}
+
 static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
                                 struct snd_ctl_elem_value *ucontrol)
 {
@@ -4942,6 +5301,112 @@ static int ca0132_alt_mic_boost_put(struct snd_kcontrol *kcontrol,
         return 1;
 }
 
+/*
+ * Sound BlasterX AE-5 Headphone Gain Controls.
+ */
+#define AE5_HEADPHONE_GAIN_MAX 3
+static int ae5_headphone_gain_info(struct snd_kcontrol *kcontrol,
+                                 struct snd_ctl_elem_info *uinfo)
+{
+        char *sfx = " Ohms)";
+        char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+        uinfo->count = 1;
+        uinfo->value.enumerated.items = AE5_HEADPHONE_GAIN_MAX;
+        if (uinfo->value.enumerated.item >= AE5_HEADPHONE_GAIN_MAX)
+                uinfo->value.enumerated.item = AE5_HEADPHONE_GAIN_MAX - 1;
+        sprintf(namestr, "%s %s",
+                ae5_headphone_gain_presets[uinfo->value.enumerated.item].name,
+                sfx);
+        strcpy(uinfo->value.enumerated.name, namestr);
+        return 0;
+}
+
+static int ae5_headphone_gain_get(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_value *ucontrol)
+{
+        struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+        struct ca0132_spec *spec = codec->spec;
+
+        ucontrol->value.enumerated.item[0] = spec->ae5_headphone_gain_val;
+        return 0;
+}
+
+static int ae5_headphone_gain_put(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_value *ucontrol)
+{
+        struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+        struct ca0132_spec *spec = codec->spec;
+        int sel = ucontrol->value.enumerated.item[0];
+        unsigned int items = AE5_HEADPHONE_GAIN_MAX;
+
+        if (sel >= items)
+                return 0;
+
+        codec_dbg(codec, "ae5_headphone_gain: boost=%d\n",
+                    sel);
+
+        spec->ae5_headphone_gain_val = sel;
+
+        if (spec->out_enum_val == HEADPHONE_OUT)
+                ae5_headphone_gain_set(codec, spec->ae5_headphone_gain_val);
+
+        return 1;
+}
+
+/*
+ * Sound BlasterX AE-5 sound filter enumerated control.
+ */
+#define AE5_SOUND_FILTER_MAX 3
+
+static int ae5_sound_filter_info(struct snd_kcontrol *kcontrol,
+                                 struct snd_ctl_elem_info *uinfo)
+{
+        char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+        uinfo->count = 1;
+        uinfo->value.enumerated.items = AE5_SOUND_FILTER_MAX;
+        if (uinfo->value.enumerated.item >= AE5_SOUND_FILTER_MAX)
+                uinfo->value.enumerated.item = AE5_SOUND_FILTER_MAX - 1;
+        sprintf(namestr, "%s",
+                        ae5_filter_presets[uinfo->value.enumerated.item].name);
+        strcpy(uinfo->value.enumerated.name, namestr);
+        return 0;
+}
+
+static int ae5_sound_filter_get(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_value *ucontrol)
+{
+        struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+        struct ca0132_spec *spec = codec->spec;
+
+        ucontrol->value.enumerated.item[0] = spec->ae5_filter_val;
+        return 0;
+}
+
+static int ae5_sound_filter_put(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_value *ucontrol)
+{
+        struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+        struct ca0132_spec *spec = codec->spec;
+        int sel = ucontrol->value.enumerated.item[0];
+        unsigned int items = AE5_SOUND_FILTER_MAX;
+
+        if (sel >= items)
+                return 0;
+
+        codec_dbg(codec, "ae5_sound_filter: %s\n",
+                        ae5_filter_presets[sel].name);
+
+        spec->ae5_filter_val = sel;
+
+        ca0113_mmio_command_set_type2(codec, 0x48, 0x07,
+                        ae5_filter_presets[sel].val);
+
+        return 1;
+}
 
 /*
  * Input Select Control for alternative ca0132 codecs. This exists because
@@ -5330,6 +5795,16 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
                 goto exit;
         }
 
+        if (nid == ZXR_HEADPHONE_GAIN) {
+                spec->zxr_gain_set = *valp;
+                if (spec->cur_out_type == HEADPHONE_OUT)
+                        changed = zxr_headphone_gain_set(codec, *valp);
+                else
+                        changed = 0;
+
+                goto exit;
+        }
+
 exit:
         snd_hda_power_down(codec);
         return changed;
@@ -5705,6 +6180,50 @@ static int ca0132_alt_add_mic_boost_enum(struct hda_codec *codec)
 }
 
 /*
+ * Add headphone gain enumerated control for the AE-5. This switches between
+ * three modes, low, medium, and high. When non-headphone outputs are selected,
+ * it is automatically set to high. This is the same behavior as Windows.
+ */
+static int ae5_add_headphone_gain_enum(struct hda_codec *codec)
+{
+        struct snd_kcontrol_new knew =
+                HDA_CODEC_MUTE_MONO("AE-5: Headphone Gain",
+                                    AE5_HEADPHONE_GAIN_ENUM, 1, 0, HDA_OUTPUT);
+        knew.info = ae5_headphone_gain_info;
+        knew.get = ae5_headphone_gain_get;
+        knew.put = ae5_headphone_gain_put;
+        return snd_hda_ctl_add(codec, AE5_HEADPHONE_GAIN_ENUM,
+                                snd_ctl_new1(&knew, codec));
+}
+
+/*
+ * Add sound filter enumerated control for the AE-5. This adds three different
+ * settings: Slow Roll Off, Minimum Phase, and Fast Roll Off. From what I've
+ * read into it, it changes the DAC's interpolation filter.
+ */
+static int ae5_add_sound_filter_enum(struct hda_codec *codec)
+{
+        struct snd_kcontrol_new knew =
+                HDA_CODEC_MUTE_MONO("AE-5: Sound Filter",
+                                    AE5_SOUND_FILTER_ENUM, 1, 0, HDA_OUTPUT);
+        knew.info = ae5_sound_filter_info;
+        knew.get = ae5_sound_filter_get;
+        knew.put = ae5_sound_filter_put;
+        return snd_hda_ctl_add(codec, AE5_SOUND_FILTER_ENUM,
+                                snd_ctl_new1(&knew, codec));
+}
+
+static int zxr_add_headphone_gain_switch(struct hda_codec *codec)
+{
+        struct snd_kcontrol_new knew =
+                CA0132_CODEC_MUTE_MONO("ZxR: 600 Ohm Gain",
+                                    ZXR_HEADPHONE_GAIN, 1, HDA_OUTPUT);
+
+        return snd_hda_ctl_add(codec, ZXR_HEADPHONE_GAIN,
+                                snd_ctl_new1(&knew, codec));
+}
+
+/*
  * Need to create slave controls for the alternate codecs that have surround
  * capabilities.
  */
@@ -5847,7 +6366,8 @@ static int ca0132_build_controls(struct hda_codec *codec)
                                             NULL, ca0132_alt_slave_pfxs,
                                             "Playback Switch",
                                             true, &spec->vmaster_mute.sw_kctl);
-
+                if (err < 0)
+                        return err;
         }
 
         /* Add in and out effects controls.
@@ -5855,8 +6375,8 @@ 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 and R3D break if Echo Cancellation is used. */
-                if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3D) {
+                /* Desktop cards break if Echo Cancellation is used. */
+                if (spec->use_pci_mmio) {
                         if (i == (ECHO_CANCELLATION - IN_EFFECT_START_NID +
                                                 OUT_EFFECTS_COUNT))
                                 continue;
@@ -5874,8 +6394,14 @@ static int ca0132_build_controls(struct hda_codec *codec)
          * prefix, and change PlayEnhancement and CrystalVoice to match.
          */
         if (spec->use_alt_controls) {
-                ca0132_alt_add_svm_enum(codec);
-                add_ca0132_alt_eq_presets(codec);
+                err = ca0132_alt_add_svm_enum(codec);
+                if (err < 0)
+                        return err;
+
+                err = add_ca0132_alt_eq_presets(codec);
+                if (err < 0)
+                        return err;
+
                 err = add_fx_switch(codec, PLAY_ENHANCEMENT,
                                         "Enable OutFX", 0);
                 if (err < 0)
@@ -5912,7 +6438,9 @@ static int ca0132_build_controls(struct hda_codec *codec)
                 if (err < 0)
                         return err;
         }
-        add_voicefx(codec);
+        err = add_voicefx(codec);
+        if (err < 0)
+                return err;
 
         /*
          * If the codec uses alt_functions, you need the enumerated controls
@@ -5920,9 +6448,36 @@ static int ca0132_build_controls(struct hda_codec *codec)
          * setting control.
          */
         if (spec->use_alt_functions) {
-                ca0132_alt_add_output_enum(codec);
-                ca0132_alt_add_input_enum(codec);
-                ca0132_alt_add_mic_boost_enum(codec);
+                err = ca0132_alt_add_output_enum(codec);
+                if (err < 0)
+                        return err;
+                err = ca0132_alt_add_mic_boost_enum(codec);
+                if (err < 0)
+                        return err;
+                /*
+                 * ZxR only has microphone input, there is no front panel
+                 * header on the card, and aux-in is handled by the DBPro board.
+                 */
+                if (spec->quirk != QUIRK_ZXR) {
+                        err = ca0132_alt_add_input_enum(codec);
+                        if (err < 0)
+                                return err;
+                }
+        }
+
+        if (spec->quirk == QUIRK_AE5) {
+                err = ae5_add_headphone_gain_enum(codec);
+                if (err < 0)
+                        return err;
+                err = ae5_add_sound_filter_enum(codec);
+                if (err < 0)
+                        return err;
+        }
+
+        if (spec->quirk == QUIRK_ZXR) {
+                err = zxr_add_headphone_gain_switch(codec);
+                if (err < 0)
+                        return err;
         }
 #ifdef ENABLE_TUNING_CONTROLS
         add_tuning_ctls(codec);
@@ -5955,6 +6510,27 @@ static int ca0132_build_controls(struct hda_codec *codec)
         return 0;
 }
 
+static int dbpro_build_controls(struct hda_codec *codec)
+{
+        struct ca0132_spec *spec = codec->spec;
+        int err = 0;
+
+        if (spec->dig_out) {
+                err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out,
+                                spec->dig_out);
+                if (err < 0)
+                        return err;
+        }
+
+        if (spec->dig_in) {
+                err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
+                if (err < 0)
+                        return err;
+        }
+
+        return 0;
+}
+
 /*
  * PCM
  */
@@ -6058,6 +6634,40 @@ static int ca0132_build_pcms(struct hda_codec *codec)
         return 0;
 }
 
+static int dbpro_build_pcms(struct hda_codec *codec)
+{
+        struct ca0132_spec *spec = codec->spec;
+        struct hda_pcm *info;
+
+        info = snd_hda_codec_pcm_new(codec, "CA0132 Alt Analog");
+        if (!info)
+                return -ENOMEM;
+        info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture;
+        info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1;
+        info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0];
+
+
+        if (!spec->dig_out && !spec->dig_in)
+                return 0;
+
+        info = snd_hda_codec_pcm_new(codec, "CA0132 Digital");
+        if (!info)
+                return -ENOMEM;
+        info->pcm_type = HDA_PCM_TYPE_SPDIF;
+        if (spec->dig_out) {
+                info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+                        ca0132_pcm_digital_playback;
+                info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dig_out;
+        }
+        if (spec->dig_in) {
+                info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+                        ca0132_pcm_digital_capture;
+                info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in;
+        }
+
+        return 0;
+}
+
 static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
 {
         if (pin) {
@@ -6238,69 +6848,48 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec)
 }
 
 /*
- * Recon3D r3d_setup_defaults sub functions.
+ * Creates a dummy stream to bind the output to. This seems to have to be done
+ * after changing the main outputs source and destination streams.
  */
-
-static void r3d_dsp_scp_startup(struct hda_codec *codec)
+static void ca0132_alt_create_dummy_stream(struct hda_codec *codec)
 {
-        unsigned int tmp;
-
-        tmp = 0x00000000;
-        dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
-
-        tmp = 0x00000001;
-        dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
-
-        tmp = 0x00000004;
-        dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
-        tmp = 0x00000005;
-        dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
-        tmp = 0x00000000;
-        dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
-}
+        struct ca0132_spec *spec = codec->spec;
+        unsigned int stream_format;
 
-static void r3d_dsp_initial_mic_setup(struct hda_codec *codec)
-{
-        unsigned int tmp;
+        stream_format = snd_hdac_calc_stream_format(48000, 2,
+                        SNDRV_PCM_FORMAT_S32_LE, 32, 0);
 
-        /* Mic 1 Setup */
-        chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
-        chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
-        /* This ConnPointID is unique to Recon3Di. Haven't seen it elsewhere */
-        chipio_set_conn_rate(codec, 0x0F, SR_96_000);
-        tmp = FLOAT_ONE;
-        dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+        snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
+                                        0, stream_format);
 
-        /* Mic 2 Setup, even though it isn't connected on SBZ */
-        chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000);
-        chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000);
-        chipio_set_conn_rate(codec, 0x0F, SR_96_000);
-        tmp = FLOAT_ZERO;
-        dspio_set_uint_param(codec, 0x80, 0x01, tmp);
+        snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
 }
 
 /*
- * Initialize Sound Blaster Z analog microphones.
+ * Initialize mic for non-chromebook ca0132 implementations.
  */
-static void sbz_init_analog_mics(struct hda_codec *codec)
+static void ca0132_alt_init_analog_mics(struct hda_codec *codec)
 {
+        struct ca0132_spec *spec = codec->spec;
         unsigned int tmp;
 
         /* Mic 1 Setup */
         chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
         chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
-        tmp = FLOAT_THREE;
+        if (spec->quirk == QUIRK_R3DI) {
+                chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+                tmp = FLOAT_ONE;
+        } else
+                tmp = FLOAT_THREE;
         dspio_set_uint_param(codec, 0x80, 0x00, tmp);
 
-        /* Mic 2 Setup, even though it isn't connected on SBZ */
+        /* Mic 2 setup (not present on desktop cards) */
         chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000);
         chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000);
+        if (spec->quirk == QUIRK_R3DI)
+                chipio_set_conn_rate(codec, 0x0F, SR_96_000);
         tmp = FLOAT_ZERO;
         dspio_set_uint_param(codec, 0x80, 0x01, tmp);
-
 }
 
 /*
@@ -6333,7 +6922,6 @@ static void sbz_connect_streams(struct hda_codec *codec)
         codec_dbg(codec, "Connect Streams exited, mutex released.\n");
 
         mutex_unlock(&spec->chipio_mutex);
-
 }
 
 /*
@@ -6360,19 +6948,29 @@ static void sbz_chipio_startup_data(struct hda_codec *codec)
         chipio_set_stream_channels(codec, 0x0C, 6);
         chipio_set_stream_control(codec, 0x0C, 1);
         /* No clue what these control */
-        chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0);
-        chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1);
-        chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2);
-        chipio_write_no_mutex(codec, 0x19003c, 0x0001e5c3);
-        chipio_write_no_mutex(codec, 0x190040, 0x0001e2c4);
-        chipio_write_no_mutex(codec, 0x190044, 0x0001e3c5);
-        chipio_write_no_mutex(codec, 0x190048, 0x0001e8c6);
-        chipio_write_no_mutex(codec, 0x19004c, 0x0001e9c7);
-        chipio_write_no_mutex(codec, 0x190050, 0x0001ecc8);
-        chipio_write_no_mutex(codec, 0x190054, 0x0001edc9);
-        chipio_write_no_mutex(codec, 0x190058, 0x0001eaca);
-        chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb);
-
+        if (spec->quirk == QUIRK_SBZ) {
+                chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0);
+                chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1);
+                chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2);
+                chipio_write_no_mutex(codec, 0x19003c, 0x0001e5c3);
+                chipio_write_no_mutex(codec, 0x190040, 0x0001e2c4);
+                chipio_write_no_mutex(codec, 0x190044, 0x0001e3c5);
+                chipio_write_no_mutex(codec, 0x190048, 0x0001e8c6);
+                chipio_write_no_mutex(codec, 0x19004c, 0x0001e9c7);
+                chipio_write_no_mutex(codec, 0x190050, 0x0001ecc8);
+                chipio_write_no_mutex(codec, 0x190054, 0x0001edc9);
+                chipio_write_no_mutex(codec, 0x190058, 0x0001eaca);
+                chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb);
+        } else if (spec->quirk == QUIRK_ZXR) {
+                chipio_write_no_mutex(codec, 0x190038, 0x000140c2);
+                chipio_write_no_mutex(codec, 0x19003c, 0x000141c3);
+                chipio_write_no_mutex(codec, 0x190040, 0x000150c4);
+                chipio_write_no_mutex(codec, 0x190044, 0x000151c5);
+                chipio_write_no_mutex(codec, 0x190050, 0x000142c8);
+                chipio_write_no_mutex(codec, 0x190054, 0x000143c9);
+                chipio_write_no_mutex(codec, 0x190058, 0x000152ca);
+                chipio_write_no_mutex(codec, 0x19005c, 0x000153cb);
+        }
         chipio_write_no_mutex(codec, 0x19042c, 0x00000001);
 
         codec_dbg(codec, "Startup Data exited, mutex released.\n");
@@ -6380,35 +6978,56 @@ static void sbz_chipio_startup_data(struct hda_codec *codec)
 }
 
 /*
- * Sound Blaster Z uses these after DSP is loaded. Weird SCP commands
- * without a 0x20 source like normal.
+ * Custom DSP SCP commands where the src value is 0x00 instead of 0x20. This is
+ * done after the DSP is loaded.
  */
-static void sbz_dsp_scp_startup(struct hda_codec *codec)
+static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec)
 {
-        unsigned int tmp;
-
-        tmp = 0x00000003;
-        dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
-        tmp = 0x00000000;
-        dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
-
-        tmp = 0x00000001;
-        dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
-
-        tmp = 0x00000004;
-        dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
-        tmp = 0x00000005;
-        dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
-        tmp = 0x00000000;
-        dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+        struct ca0132_spec *spec = codec->spec;
+        unsigned int tmp, i;
 
+        /*
+         * Gotta run these twice, or else mic works inconsistently. Not clear
+         * why this is, but multiple tests have confirmed it.
+         */
+        for (i = 0; i < 2; i++) {
+                switch (spec->quirk) {
+                case QUIRK_SBZ:
+                case QUIRK_AE5:
+                        tmp = 0x00000003;
+                        dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+                        tmp = 0x00000000;
+                        dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
+                        tmp = 0x00000001;
+                        dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
+                        tmp = 0x00000004;
+                        dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+                        tmp = 0x00000005;
+                        dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+                        tmp = 0x00000000;
+                        dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+                        break;
+                case QUIRK_R3D:
+                case QUIRK_R3DI:
+                        tmp = 0x00000000;
+                        dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
+                        tmp = 0x00000001;
+                        dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
+                        tmp = 0x00000004;
+                        dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+                        tmp = 0x00000005;
+                        dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+                        tmp = 0x00000000;
+                        dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+                        break;
+                }
+                msleep(100);
+        }
 }
 
-static void sbz_dsp_initial_mic_setup(struct hda_codec *codec)
+static void ca0132_alt_dsp_initial_mic_setup(struct hda_codec *codec)
 {
+        struct ca0132_spec *spec = codec->spec;
         unsigned int tmp;
 
         chipio_set_stream_control(codec, 0x03, 0);
@@ -6423,8 +7042,161 @@ static void sbz_dsp_initial_mic_setup(struct hda_codec *codec)
         chipio_set_stream_control(codec, 0x03, 1);
         chipio_set_stream_control(codec, 0x04, 1);
 
-        chipio_write(codec, 0x18b098, 0x0000000c);
-        chipio_write(codec, 0x18b09C, 0x0000000c);
+        switch (spec->quirk) {
+        case QUIRK_SBZ:
+                chipio_write(codec, 0x18b098, 0x0000000c);
+                chipio_write(codec, 0x18b09C, 0x0000000c);
+                break;
+        case QUIRK_AE5:
+                chipio_write(codec, 0x18b098, 0x0000000c);
+                chipio_write(codec, 0x18b09c, 0x0000004c);
+                break;
+        }
+}
+
+static void ae5_post_dsp_register_set(struct hda_codec *codec)
+{
+        struct ca0132_spec *spec = codec->spec;
+
+        chipio_8051_write_direct(codec, 0x93, 0x10);
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44);
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2);
+
+        writeb(0xff, spec->mem_base + 0x304);
+        writeb(0xff, spec->mem_base + 0x304);
+        writeb(0xff, spec->mem_base + 0x304);
+        writeb(0xff, spec->mem_base + 0x304);
+        writeb(0x00, spec->mem_base + 0x100);
+        writeb(0xff, spec->mem_base + 0x304);
+        writeb(0x00, spec->mem_base + 0x100);
+        writeb(0xff, spec->mem_base + 0x304);
+        writeb(0x00, spec->mem_base + 0x100);
+        writeb(0xff, spec->mem_base + 0x304);
+        writeb(0x00, spec->mem_base + 0x100);
+        writeb(0xff, spec->mem_base + 0x304);
+
+        ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x3f);
+        ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f);
+        ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+}
+
+static void ae5_post_dsp_param_setup(struct hda_codec *codec)
+{
+        /*
+         * Param3 in the 8051's memory is represented by the ascii string 'mch'
+         * which seems to be 'multichannel'. This is also mentioned in the
+         * AE-5's registry values in Windows.
+         */
+        chipio_set_control_param(codec, 3, 0);
+        /*
+         * I believe ASI is 'audio serial interface' and that it's used to
+         * change colors on the external LED strip connected to the AE-5.
+         */
+        chipio_set_control_flag(codec, CONTROL_FLAG_ASI_96KHZ, 1);
+
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x724, 0x83);
+        chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0);
+
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x92);
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0xfa);
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_8051_DATA_WRITE, 0x22);
+}
+
+static void ae5_post_dsp_pll_setup(struct hda_codec *codec)
+{
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x41);
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc8);
+
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x45);
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_PLL_PMU_WRITE, 0xcc);
+
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x40);
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_PLL_PMU_WRITE, 0xcb);
+
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43);
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7);
+
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x51);
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_PLL_PMU_WRITE, 0x8d);
+}
+
+static void ae5_post_dsp_stream_setup(struct hda_codec *codec)
+{
+        struct ca0132_spec *spec = codec->spec;
+
+        mutex_lock(&spec->chipio_mutex);
+
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x725, 0x81);
+
+        chipio_set_conn_rate_no_mutex(codec, 0x70, SR_96_000);
+
+        chipio_set_stream_channels(codec, 0x0C, 6);
+        chipio_set_stream_control(codec, 0x0C, 1);
+
+        chipio_set_stream_source_dest(codec, 0x5, 0x43, 0x0);
+
+        chipio_set_stream_source_dest(codec, 0x18, 0x9, 0xd0);
+        chipio_set_conn_rate_no_mutex(codec, 0xd0, SR_96_000);
+        chipio_set_stream_channels(codec, 0x18, 6);
+        chipio_set_stream_control(codec, 0x18, 1);
+
+        chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 4);
+
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43);
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7);
+
+        ca0113_mmio_command_set(codec, 0x48, 0x01, 0x80);
+
+        mutex_unlock(&spec->chipio_mutex);
+}
+
+static void ae5_post_dsp_startup_data(struct hda_codec *codec)
+{
+        struct ca0132_spec *spec = codec->spec;
+
+        mutex_lock(&spec->chipio_mutex);
+
+        chipio_write_no_mutex(codec, 0x189000, 0x0001f101);
+        chipio_write_no_mutex(codec, 0x189004, 0x0001f101);
+        chipio_write_no_mutex(codec, 0x189024, 0x00014004);
+        chipio_write_no_mutex(codec, 0x189028, 0x0002000f);
+
+        ca0113_mmio_command_set(codec, 0x48, 0x0a, 0x05);
+        chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 7);
+        ca0113_mmio_command_set(codec, 0x48, 0x0b, 0x12);
+        ca0113_mmio_command_set(codec, 0x48, 0x04, 0x00);
+        ca0113_mmio_command_set(codec, 0x48, 0x06, 0x48);
+        ca0113_mmio_command_set(codec, 0x48, 0x0a, 0x05);
+        ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+        ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00);
+        ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00);
+        ca0113_mmio_gpio_set(codec, 0, true);
+        ca0113_mmio_gpio_set(codec, 1, true);
+        ca0113_mmio_command_set(codec, 0x48, 0x07, 0x80);
+
+        chipio_write_no_mutex(codec, 0x18b03c, 0x00000012);
+
+        ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00);
+        ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00);
+
+        mutex_unlock(&spec->chipio_mutex);
 }
 
 /*
@@ -6485,9 +7257,8 @@ static void r3d_setup_defaults(struct hda_codec *codec)
         if (spec->dsp_state != DSP_DOWNLOADED)
                 return;
 
-        r3d_dsp_scp_startup(codec);
-
-        r3d_dsp_initial_mic_setup(codec);
+        ca0132_alt_dsp_scp_startup(codec);
+        ca0132_alt_init_analog_mics(codec);
 
         /*remove DSP headroom*/
         tmp = FLOAT_ZERO;
@@ -6523,19 +7294,16 @@ static void r3d_setup_defaults(struct hda_codec *codec)
 static void sbz_setup_defaults(struct hda_codec *codec)
 {
         struct ca0132_spec *spec = codec->spec;
-        unsigned int tmp, stream_format;
+        unsigned int tmp;
         int num_fx;
         int idx, i;
 
         if (spec->dsp_state != DSP_DOWNLOADED)
                 return;
 
-        sbz_dsp_scp_startup(codec);
-
-        sbz_init_analog_mics(codec);
-
+        ca0132_alt_dsp_scp_startup(codec);
+        ca0132_alt_init_analog_mics(codec);
         sbz_connect_streams(codec);
-
         sbz_chipio_startup_data(codec);
 
         chipio_set_stream_control(codec, 0x03, 1);
@@ -6561,8 +7329,7 @@ static void sbz_setup_defaults(struct hda_codec *codec)
         /* Set speaker source? */
         dspio_set_uint_param(codec, 0x32, 0x00, tmp);
 
-        sbz_dsp_initial_mic_setup(codec);
-
+        ca0132_alt_dsp_initial_mic_setup(codec);
 
         /* out, in effects + voicefx */
         num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1;
@@ -6575,23 +7342,74 @@ static void sbz_setup_defaults(struct hda_codec *codec)
                 }
         }
 
-        /*
-         * Have to make a stream to bind the sound output to, otherwise
-         * you'll get dead audio. Before I did this, it would bind to an
-         * audio input, and would never work
-         */
-        stream_format = snd_hdac_calc_stream_format(48000, 2,
-                        SNDRV_PCM_FORMAT_S32_LE, 32, 0);
+        ca0132_alt_create_dummy_stream(codec);
+}
 
-        snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
-                                        0, stream_format);
+/*
+ * Setup default parameters for the Sound BlasterX AE-5 DSP.
+ */
+static void ae5_setup_defaults(struct hda_codec *codec)
+{
+        struct ca0132_spec *spec = codec->spec;
+        unsigned int tmp;
+        int num_fx;
+        int idx, i;
 
-        snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
+        if (spec->dsp_state != DSP_DOWNLOADED)
+                return;
 
-        snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
-                                        0, stream_format);
+        ca0132_alt_dsp_scp_startup(codec);
+        ca0132_alt_init_analog_mics(codec);
+        chipio_set_stream_control(codec, 0x03, 1);
+        chipio_set_stream_control(codec, 0x04, 1);
 
-        snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
+        /* New, unknown SCP req's */
+        tmp = FLOAT_ZERO;
+        dspio_set_uint_param(codec, 0x96, 0x29, tmp);
+        dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
+        dspio_set_uint_param(codec, 0x80, 0x0d, tmp);
+        dspio_set_uint_param(codec, 0x80, 0x0e, tmp);
+
+        ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f);
+        ca0113_mmio_gpio_set(codec, 0, false);
+        ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
+
+        /* Internal loopback off */
+        tmp = FLOAT_ONE;
+        dspio_set_uint_param(codec, 0x37, 0x08, tmp);
+        dspio_set_uint_param(codec, 0x37, 0x10, tmp);
+
+        /*remove DSP headroom*/
+        tmp = FLOAT_ZERO;
+        dspio_set_uint_param(codec, 0x96, 0x3C, tmp);
+
+        /* set WUH source */
+        tmp = FLOAT_TWO;
+        dspio_set_uint_param(codec, 0x31, 0x00, tmp);
+        chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000);
+
+        /* Set speaker source? */
+        dspio_set_uint_param(codec, 0x32, 0x00, tmp);
+
+        ca0132_alt_dsp_initial_mic_setup(codec);
+        ae5_post_dsp_register_set(codec);
+        ae5_post_dsp_param_setup(codec);
+        ae5_post_dsp_pll_setup(codec);
+        ae5_post_dsp_stream_setup(codec);
+        ae5_post_dsp_startup_data(codec);
+
+        /* out, in effects + voicefx */
+        num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1;
+        for (idx = 0; idx < num_fx; idx++) {
+                for (i = 0; i <= ca0132_effects[idx].params; i++) {
+                        dspio_set_uint_param(codec,
+                                        ca0132_effects[idx].mid,
+                                        ca0132_effects[idx].reqs[i],
+                                        ca0132_effects[idx].def_vals[i]);
+                }
+        }
+
+        ca0132_alt_create_dummy_stream(codec);
 }
 
 /*
@@ -6673,12 +7491,14 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec)
          */
         switch (spec->quirk) {
         case QUIRK_SBZ:
-                if (request_firmware(&fw_entry, SBZ_EFX_FILE,
+        case QUIRK_R3D:
+        case QUIRK_AE5:
+                if (request_firmware(&fw_entry, DESKTOP_EFX_FILE,
                                         codec->card->dev) != 0) {
-                        codec_dbg(codec, "SBZ alt firmware not detected. ");
+                        codec_dbg(codec, "Desktop firmware not found.");
                         spec->alt_firmware_present = false;
                 } else {
-                        codec_dbg(codec, "Sound Blaster Z firmware selected.");
+                        codec_dbg(codec, "Desktop firmware selected.");
                         spec->alt_firmware_present = true;
                 }
                 break;
@@ -6921,6 +7741,14 @@ static void ca0132_init_chip(struct hda_codec *codec)
         spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID] = 1;
         spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] = 0;
 
+        /*
+         * The ZxR doesn't have a front panel header, and it's line-in is on
+         * the daughter board. So, there is no input enum control, and we need
+         * to make sure that spec->in_enum_val is set properly.
+         */
+        if (spec->quirk == QUIRK_ZXR)
+                spec->in_enum_val = REAR_MIC;
+
 #ifdef ENABLE_TUNING_CONTROLS
         ca0132_init_tuning_defaults(codec);
 #endif
@@ -6948,11 +7776,11 @@ static void sbz_region2_exit(struct hda_codec *codec)
         for (i = 0; i < 8; i++)
                 writeb(0xb3, spec->mem_base + 0x304);
 
-        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);
+        ca0113_mmio_gpio_set(codec, 0, false);
+        ca0113_mmio_gpio_set(codec, 1, false);
+        ca0113_mmio_gpio_set(codec, 4, true);
+        ca0113_mmio_gpio_set(codec, 5, false);
+        ca0113_mmio_gpio_set(codec, 7, false);
 }
 
 static void sbz_set_pin_ctl_default(struct hda_codec *codec)
@@ -6995,6 +7823,16 @@ static void sbz_gpio_shutdown_commands(struct hda_codec *codec, int dir,
                                 AC_VERB_SET_GPIO_DATA, data);
 }
 
+static void zxr_dbpro_power_state_shutdown(struct hda_codec *codec)
+{
+        hda_nid_t pins[7] = {0x05, 0x0c, 0x09, 0x0e, 0x08, 0x11, 0x01};
+        unsigned int i;
+
+        for (i = 0; i < 7; i++)
+                snd_hda_codec_write(codec, pins[i], 0,
+                                AC_VERB_SET_POWER_STATE, 0x03);
+}
+
 static void sbz_exit_chip(struct hda_codec *codec)
 {
         chipio_set_stream_control(codec, 0x03, 0);
@@ -7037,6 +7875,61 @@ static void r3d_exit_chip(struct hda_codec *codec)
         snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5b);
 }
 
+static void ae5_exit_chip(struct hda_codec *codec)
+{
+        chipio_set_stream_control(codec, 0x03, 0);
+        chipio_set_stream_control(codec, 0x04, 0);
+
+        ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
+        ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+        ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+        ca0113_mmio_command_set(codec, 0x30, 0x30, 0x00);
+        ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00);
+        ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x00);
+        ca0113_mmio_gpio_set(codec, 0, false);
+        ca0113_mmio_gpio_set(codec, 1, false);
+
+        snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
+        snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
+
+        chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0);
+
+        chipio_set_stream_control(codec, 0x18, 0);
+        chipio_set_stream_control(codec, 0x0c, 0);
+
+        snd_hda_codec_write(codec, 0x01, 0, 0x724, 0x83);
+}
+
+static void zxr_exit_chip(struct hda_codec *codec)
+{
+        chipio_set_stream_control(codec, 0x03, 0);
+        chipio_set_stream_control(codec, 0x04, 0);
+        chipio_set_stream_control(codec, 0x14, 0);
+        chipio_set_stream_control(codec, 0x0C, 0);
+
+        chipio_set_conn_rate(codec, 0x41, SR_192_000);
+        chipio_set_conn_rate(codec, 0x91, SR_192_000);
+
+        chipio_write(codec, 0x18a020, 0x00000083);
+
+        snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
+        snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
+
+        ca0132_clear_unsolicited(codec);
+        sbz_set_pin_ctl_default(codec);
+        snd_hda_codec_write(codec, 0x0B, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x00);
+
+        ca0113_mmio_gpio_set(codec, 5, false);
+        ca0113_mmio_gpio_set(codec, 2, false);
+        ca0113_mmio_gpio_set(codec, 3, false);
+        ca0113_mmio_gpio_set(codec, 0, false);
+        ca0113_mmio_gpio_set(codec, 4, true);
+        ca0113_mmio_gpio_set(codec, 0, true);
+        ca0113_mmio_gpio_set(codec, 5, true);
+        ca0113_mmio_gpio_set(codec, 2, false);
+        ca0113_mmio_gpio_set(codec, 3, false);
+}
+
 static void ca0132_exit_chip(struct hda_codec *codec)
 {
         /* put any chip cleanup stuffs here. */
@@ -7140,11 +8033,6 @@ static void sbz_pre_dsp_setup(struct hda_codec *codec)
         writel(0x00820680, spec->mem_base + 0x01C);
         writel(0x00820680, spec->mem_base + 0x01C);
 
-        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, 0x11, 0,
@@ -7153,12 +8041,6 @@ static void sbz_pre_dsp_setup(struct hda_codec *codec)
 
 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,
@@ -7205,23 +8087,116 @@ static void ca0132_mmio_init(struct hda_codec *codec)
 {
         struct ca0132_spec *spec = codec->spec;
 
-        writel(0x00000000, spec->mem_base + 0x400);
-        writel(0x00000000, spec->mem_base + 0x408);
-        writel(0x00000000, spec->mem_base + 0x40C);
-        writel(0x00880680, spec->mem_base + 0x01C);
-        writel(0x00000083, spec->mem_base + 0xC0C);
+        if (spec->quirk == QUIRK_AE5)
+                writel(0x00000001, spec->mem_base + 0x400);
+        else
+                writel(0x00000000, spec->mem_base + 0x400);
+
+        if (spec->quirk == QUIRK_AE5)
+                writel(0x00000001, spec->mem_base + 0x408);
+        else
+                writel(0x00000000, spec->mem_base + 0x408);
+
+        if (spec->quirk == QUIRK_AE5)
+                writel(0x00000001, spec->mem_base + 0x40c);
+        else
+                writel(0x00000000, spec->mem_base + 0x40C);
+
+        if (spec->quirk == QUIRK_ZXR)
+                writel(0x00880640, spec->mem_base + 0x01C);
+        else
+                writel(0x00880680, spec->mem_base + 0x01C);
+
+        if (spec->quirk == QUIRK_AE5)
+                writel(0x00000080, spec->mem_base + 0xC0C);
+        else
+                writel(0x00000083, spec->mem_base + 0xC0C);
+
         writel(0x00000030, spec->mem_base + 0xC00);
         writel(0x00000000, spec->mem_base + 0xC04);
+
+        if (spec->quirk == QUIRK_AE5)
+                writel(0x00000000, spec->mem_base + 0xC0C);
+        else
+                writel(0x00000003, spec->mem_base + 0xC0C);
+
         writel(0x00000003, spec->mem_base + 0xC0C);
         writel(0x00000003, spec->mem_base + 0xC0C);
         writel(0x00000003, spec->mem_base + 0xC0C);
-        writel(0x00000003, spec->mem_base + 0xC0C);
-        writel(0x000000C1, spec->mem_base + 0xC08);
+
+        if (spec->quirk == QUIRK_AE5)
+                writel(0x00000001, spec->mem_base + 0xC08);
+        else
+                writel(0x000000C1, spec->mem_base + 0xC08);
+
         writel(0x000000F1, spec->mem_base + 0xC08);
         writel(0x00000001, spec->mem_base + 0xC08);
         writel(0x000000C7, spec->mem_base + 0xC08);
         writel(0x000000C1, spec->mem_base + 0xC08);
         writel(0x00000080, spec->mem_base + 0xC04);
+
+        if (spec->quirk == QUIRK_AE5) {
+                writel(0x00000000, spec->mem_base + 0x42c);
+                writel(0x00000000, spec->mem_base + 0x46c);
+                writel(0x00000000, spec->mem_base + 0x4ac);
+                writel(0x00000000, spec->mem_base + 0x4ec);
+                writel(0x00000000, spec->mem_base + 0x43c);
+                writel(0x00000000, spec->mem_base + 0x47c);
+                writel(0x00000000, spec->mem_base + 0x4bc);
+                writel(0x00000000, spec->mem_base + 0x4fc);
+                writel(0x00000600, spec->mem_base + 0x100);
+                writel(0x00000014, spec->mem_base + 0x410);
+                writel(0x0000060f, spec->mem_base + 0x100);
+                writel(0x0000070f, spec->mem_base + 0x100);
+                writel(0x00000aff, spec->mem_base + 0x830);
+                writel(0x00000000, spec->mem_base + 0x86c);
+                writel(0x0000006b, spec->mem_base + 0x800);
+                writel(0x00000001, spec->mem_base + 0x86c);
+                writel(0x0000006b, spec->mem_base + 0x800);
+                writel(0x00000057, spec->mem_base + 0x804);
+                writel(0x00800000, spec->mem_base + 0x20c);
+        }
+}
+
+/*
+ * This function writes to some SFR's, does some region2 writes, and then
+ * eventually resets the codec with the 0x7ff verb. Not quite sure why it does
+ * what it does.
+ */
+static void ae5_register_set(struct hda_codec *codec)
+{
+        struct ca0132_spec *spec = codec->spec;
+
+        chipio_8051_write_direct(codec, 0x93, 0x10);
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44);
+        snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                            VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2);
+
+        writeb(0x0f, spec->mem_base + 0x304);
+        writeb(0x0f, spec->mem_base + 0x304);
+        writeb(0x0f, spec->mem_base + 0x304);
+        writeb(0x0f, spec->mem_base + 0x304);
+        writeb(0x0e, spec->mem_base + 0x100);
+        writeb(0x1f, spec->mem_base + 0x304);
+        writeb(0x0c, spec->mem_base + 0x100);
+        writeb(0x3f, spec->mem_base + 0x304);
+        writeb(0x08, spec->mem_base + 0x100);
+        writeb(0x7f, spec->mem_base + 0x304);
+        writeb(0x00, spec->mem_base + 0x100);
+        writeb(0xff, spec->mem_base + 0x304);
+
+        ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f);
+
+        chipio_8051_write_direct(codec, 0x90, 0x00);
+        chipio_8051_write_direct(codec, 0x90, 0x10);
+
+        ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+
+        chipio_write(codec, 0x18b0a4, 0x000000c2);
+
+        snd_hda_codec_write(codec, 0x01, 0, 0x7ff, 0x00);
+        snd_hda_codec_write(codec, 0x01, 0, 0x7ff, 0x00);
 }
 
 /*
@@ -7257,6 +8232,21 @@ static void ca0132_alt_init(struct hda_codec *codec)
                 snd_hda_sequence_write(codec, spec->chip_init_verbs);
                 snd_hda_sequence_write(codec, spec->desktop_init_verbs);
                 break;
+        case QUIRK_AE5:
+                ca0132_gpio_init(codec);
+                snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                                VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x49);
+                snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+                                VENDOR_CHIPIO_PLL_PMU_WRITE, 0x88);
+                chipio_write(codec, 0x18b030, 0x00000020);
+                snd_hda_sequence_write(codec, spec->chip_init_verbs);
+                snd_hda_sequence_write(codec, spec->desktop_init_verbs);
+                ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
+                break;
+        case QUIRK_ZXR:
+                snd_hda_sequence_write(codec, spec->chip_init_verbs);
+                snd_hda_sequence_write(codec, spec->desktop_init_verbs);
+                break;
         }
 }
 
@@ -7298,6 +8288,9 @@ static int ca0132_init(struct hda_codec *codec)
 
         snd_hda_power_up_pm(codec);
 
+        if (spec->quirk == QUIRK_AE5)
+                ae5_register_set(codec);
+
         ca0132_init_unsol(codec);
         ca0132_init_params(codec);
         ca0132_init_flags(codec);
@@ -7317,8 +8310,12 @@ static int ca0132_init(struct hda_codec *codec)
                 r3d_setup_defaults(codec);
                 break;
         case QUIRK_SBZ:
+        case QUIRK_ZXR:
                 sbz_setup_defaults(codec);
                 break;
+        case QUIRK_AE5:
+                ae5_setup_defaults(codec);
+                break;
         default:
                 ca0132_setup_defaults(codec);
                 ca0132_init_analog_mic2(codec);
@@ -7372,6 +8369,21 @@ static int ca0132_init(struct hda_codec *codec)
         return 0;
 }
 
+static int dbpro_init(struct hda_codec *codec)
+{
+        struct ca0132_spec *spec = codec->spec;
+        struct auto_pin_cfg *cfg = &spec->autocfg;
+        unsigned int i;
+
+        init_output(codec, cfg->dig_out_pins[0], spec->dig_out);
+        init_input(codec, cfg->dig_in_pin, spec->dig_in);
+
+        for (i = 0; i < spec->num_inputs; i++)
+                init_input(codec, spec->input_pins[i], spec->adcs[i]);
+
+        return 0;
+}
+
 static void ca0132_free(struct hda_codec *codec)
 {
         struct ca0132_spec *spec = codec->spec;
@@ -7382,9 +8394,15 @@ static void ca0132_free(struct hda_codec *codec)
         case QUIRK_SBZ:
                 sbz_exit_chip(codec);
                 break;
+        case QUIRK_ZXR:
+                zxr_exit_chip(codec);
+                break;
         case QUIRK_R3D:
                 r3d_exit_chip(codec);
                 break;
+        case QUIRK_AE5:
+                ae5_exit_chip(codec);
+                break;
         case QUIRK_R3DI:
                 r3di_gpio_shutdown(codec);
                 break;
@@ -7400,6 +8418,16 @@ static void ca0132_free(struct hda_codec *codec)
         kfree(codec->spec);
 }
 
+static void dbpro_free(struct hda_codec *codec)
+{
+        struct ca0132_spec *spec = codec->spec;
+
+        zxr_dbpro_power_state_shutdown(codec);
+
+        kfree(spec->spec_init_verbs);
+        kfree(codec->spec);
+}
+
 static void ca0132_reboot_notify(struct hda_codec *codec)
 {
         codec->patch_ops.free(codec);
@@ -7414,6 +8442,13 @@ static const struct hda_codec_ops ca0132_patch_ops = {
         .reboot_notify = ca0132_reboot_notify,
 };
 
+static const struct hda_codec_ops dbpro_patch_ops = {
+        .build_controls = dbpro_build_controls,
+        .build_pcms = dbpro_build_pcms,
+        .init = dbpro_init,
+        .free = dbpro_free,
+};
+
 static void ca0132_config(struct hda_codec *codec)
 {
         struct ca0132_spec *spec = codec->spec;
@@ -7432,9 +8467,33 @@ static void ca0132_config(struct hda_codec *codec)
 
         switch (spec->quirk) {
         case QUIRK_ALIENWARE:
-                codec_dbg(codec, "ca0132_config: QUIRK_ALIENWARE applied.\n");
+                codec_dbg(codec, "%s: QUIRK_ALIENWARE applied.\n", __func__);
                 snd_hda_apply_pincfgs(codec, alienware_pincfgs);
+                break;
+        case QUIRK_SBZ:
+                codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__);
+                snd_hda_apply_pincfgs(codec, sbz_pincfgs);
+                break;
+        case QUIRK_ZXR:
+                codec_dbg(codec, "%s: QUIRK_ZXR applied.\n", __func__);
+                snd_hda_apply_pincfgs(codec, zxr_pincfgs);
+                break;
+        case QUIRK_R3D:
+                codec_dbg(codec, "%s: QUIRK_R3D applied.\n", __func__);
+                snd_hda_apply_pincfgs(codec, r3d_pincfgs);
+                break;
+        case QUIRK_R3DI:
+                codec_dbg(codec, "%s: QUIRK_R3DI applied.\n", __func__);
+                snd_hda_apply_pincfgs(codec, r3di_pincfgs);
+                break;
+        case QUIRK_AE5:
+                codec_dbg(codec, "%s: QUIRK_AE5 applied.\n", __func__);
+                snd_hda_apply_pincfgs(codec, r3di_pincfgs);
+                break;
+        }
 
+        switch (spec->quirk) {
+        case QUIRK_ALIENWARE:
                 spec->num_outputs = 2;
                 spec->out_pins[0] = 0x0b; /* speaker out */
                 spec->out_pins[1] = 0x0f;
@@ -7454,15 +8513,6 @@ static void ca0132_config(struct hda_codec *codec)
                 break;
         case QUIRK_SBZ:
         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 */
                 spec->out_pins[1] = 0x0F; /* Rear headphone out */
@@ -7487,10 +8537,62 @@ static void ca0132_config(struct hda_codec *codec)
                 spec->multiout.dig_out_nid = spec->dig_out;
                 spec->dig_in = 0x09;
                 break;
-        case QUIRK_R3DI:
-                codec_dbg(codec, "%s: QUIRK_R3DI applied.\n", __func__);
-                snd_hda_apply_pincfgs(codec, r3di_pincfgs);
+        case QUIRK_ZXR:
+                spec->num_outputs = 2;
+                spec->out_pins[0] = 0x0B; /* Line out */
+                spec->out_pins[1] = 0x0F; /* Rear headphone out */
+                spec->out_pins[2] = 0x10; /* Center/LFE */
+                spec->out_pins[3] = 0x11; /* Rear surround */
+                spec->shared_out_nid = 0x2;
+                spec->unsol_tag_hp = spec->out_pins[1];
+                spec->unsol_tag_front_hp = spec->out_pins[2];
 
+                spec->adcs[0] = 0x7; /* Rear Mic / Line-in */
+                spec->adcs[1] = 0x8; /* Not connected, no front mic */
+                spec->adcs[2] = 0xa; /* what u hear */
+
+                spec->num_inputs = 2;
+                spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */
+                spec->input_pins[1] = 0x13; /* What U Hear */
+                spec->shared_mic_nid = 0x7;
+                spec->unsol_tag_amic1 = spec->input_pins[0];
+                break;
+        case QUIRK_ZXR_DBPRO:
+                spec->adcs[0] = 0x8; /* ZxR DBPro Aux In */
+
+                spec->num_inputs = 1;
+                spec->input_pins[0] = 0x11; /* RCA Line-in */
+
+                spec->dig_out = 0x05;
+                spec->multiout.dig_out_nid = spec->dig_out;
+
+                spec->dig_in = 0x09;
+                break;
+        case QUIRK_AE5:
+                spec->num_outputs = 2;
+                spec->out_pins[0] = 0x0B; /* Line out */
+                spec->out_pins[1] = 0x11; /* Rear headphone out */
+                spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/
+                spec->out_pins[3] = 0x0F; /* Rear surround */
+                spec->shared_out_nid = 0x2;
+                spec->unsol_tag_hp = spec->out_pins[1];
+                spec->unsol_tag_front_hp = spec->out_pins[2];
+
+                spec->adcs[0] = 0x7; /* Rear Mic / Line-in */
+                spec->adcs[1] = 0x8; /* Front Mic, but only if no DSP */
+                spec->adcs[2] = 0xa; /* what u hear */
+
+                spec->num_inputs = 2;
+                spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */
+                spec->input_pins[1] = 0x13; /* What U Hear */
+                spec->shared_mic_nid = 0x7;
+                spec->unsol_tag_amic1 = spec->input_pins[0];
+
+                /* SPDIF I/O */
+                spec->dig_out = 0x05;
+                spec->multiout.dig_out_nid = spec->dig_out;
+                break;
+        case QUIRK_R3DI:
                 spec->num_outputs = 2;
                 spec->out_pins[0] = 0x0B; /* Line out */
                 spec->out_pins[1] = 0x0F; /* Rear headphone out */
@@ -7547,7 +8649,11 @@ 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->quirk == QUIRK_R3D)
+        /*
+         * Since desktop cards use pci_mmio, this can be used to determine
+         * whether or not to use these verbs instead of a separate bool.
+         */
+        if (spec->use_pci_mmio)
                 spec->desktop_init_verbs = ca0132_init_verbs1;
         spec->spec_init_verbs = kcalloc(NUM_SPEC_VERBS,
                                         sizeof(struct hda_verb),
@@ -7579,6 +8685,29 @@ static int ca0132_prepare_verbs(struct hda_codec *codec)
         return 0;
 }
 
+/*
+ * The Sound Blaster ZxR shares the same PCI subsystem ID as some regular
+ * Sound Blaster Z cards. However, they have different HDA codec subsystem
+ * ID's. So, we check for the ZxR's subsystem ID, as well as the DBPro
+ * daughter boards ID.
+ */
+static void sbz_detect_quirk(struct hda_codec *codec)
+{
+        struct ca0132_spec *spec = codec->spec;
+
+        switch (codec->core.subsystem_id) {
+        case 0x11020033:
+                spec->quirk = QUIRK_ZXR;
+                break;
+        case 0x1102003f:
+                spec->quirk = QUIRK_ZXR_DBPRO;
+                break;
+        default:
+                spec->quirk = QUIRK_SBZ;
+                break;
+        }
+}
+
 static int patch_ca0132(struct hda_codec *codec)
 {
         struct ca0132_spec *spec;
@@ -7593,10 +8722,6 @@ static int patch_ca0132(struct hda_codec *codec)
         codec->spec = spec;
         spec->codec = codec;
 
-        codec->patch_ops = ca0132_patch_ops;
-        codec->pcm_format_first = 1;
-        codec->no_sticky_stream = 1;
-
         /* Detect codec quirk */
         quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks);
         if (quirk)
@@ -7604,6 +8729,18 @@ static int patch_ca0132(struct hda_codec *codec)
         else
                 spec->quirk = QUIRK_NONE;
 
+        if (spec->quirk == QUIRK_SBZ)
+                sbz_detect_quirk(codec);
+
+        if (spec->quirk == QUIRK_ZXR_DBPRO)
+                codec->patch_ops = dbpro_patch_ops;
+        else
+                codec->patch_ops = ca0132_patch_ops;
+
+        codec->pcm_format_first = 1;
+        codec->no_sticky_stream = 1;
+
+
         spec->dsp_state = DSP_DOWNLOAD_INIT;
         spec->num_mixers = 1;
 
@@ -7613,6 +8750,12 @@ static int patch_ca0132(struct hda_codec *codec)
                 spec->mixers[0] = desktop_mixer;
                 snd_hda_codec_set_name(codec, "Sound Blaster Z");
                 break;
+        case QUIRK_ZXR:
+                spec->mixers[0] = desktop_mixer;
+                snd_hda_codec_set_name(codec, "Sound Blaster ZxR");
+                break;
+        case QUIRK_ZXR_DBPRO:
+                break;
         case QUIRK_R3D:
                 spec->mixers[0] = desktop_mixer;
                 snd_hda_codec_set_name(codec, "Recon3D");
@@ -7621,6 +8764,10 @@ static int patch_ca0132(struct hda_codec *codec)
                 spec->mixers[0] = r3di_mixer;
                 snd_hda_codec_set_name(codec, "Recon3Di");
                 break;
+        case QUIRK_AE5:
+                spec->mixers[0] = desktop_mixer;
+                snd_hda_codec_set_name(codec, "Sound BlasterX AE-5");
+                break;
         default:
                 spec->mixers[0] = ca0132_mixer;
                 break;
@@ -7630,6 +8777,8 @@ static int patch_ca0132(struct hda_codec *codec)
         switch (spec->quirk) {
         case QUIRK_SBZ:
         case QUIRK_R3D:
+        case QUIRK_AE5:
+        case QUIRK_ZXR:
                 spec->use_alt_controls = true;
                 spec->use_alt_functions = true;
                 spec->use_pci_mmio = true;
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index a7f91be45194..64fa5a82bb9f 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -23,7 +23,7 @@
 #include <linux/module.h>
 #include <sound/core.h>
 #include <sound/tlv.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 1b2195dd2b26..52642ba3e2c0 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -25,7 +25,7 @@
 #include <linux/slab.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index cfd4e4f97f8f..950e02e71766 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -27,7 +27,7 @@
 #include <sound/core.h>
 #include <sound/jack.h>
 
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_beep.h"
@@ -943,6 +943,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
         SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410),
         SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410),
         SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD),
+        SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC),
         SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC),
         SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
         SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC),
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index cb587dce67a9..67099cbb6be2 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -41,7 +41,7 @@
 #include <sound/hdaudio.h>
 #include <sound/hda_i915.h>
 #include <sound/hda_chmap.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_jack.h"
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3ac7ba9b342d..fa61674a5605 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -32,7 +32,7 @@
 #include <linux/input.h>
 #include <sound/core.h>
 #include <sound/jack.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
@@ -6843,6 +6843,12 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
                 {0x21, 0x0221101f}),
         SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC294_FIXUP_LENOVO_MIC_LOCATION,
                 {0x14, 0x90170110},
+                {0x19, 0x02a11030},
+                {0x1a, 0x02a11040},
+                {0x1b, 0x01011020},
+                {0x21, 0x0221101f}),
+        SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC294_FIXUP_LENOVO_MIC_LOCATION,
+                {0x14, 0x90170110},
                 {0x19, 0x02a11020},
                 {0x1a, 0x02a11030},
                 {0x21, 0x0221101f}),
@@ -7738,6 +7744,8 @@ enum {
         ALC662_FIXUP_ASUS_Nx50,
         ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE,
         ALC668_FIXUP_ASUS_Nx51,
+        ALC668_FIXUP_MIC_COEF,
+        ALC668_FIXUP_ASUS_G751,
         ALC891_FIXUP_HEADSET_MODE,
         ALC891_FIXUP_DELL_MIC_NO_PRESENCE,
         ALC662_FIXUP_ACER_VERITON,
@@ -8007,6 +8015,23 @@ static const struct hda_fixup alc662_fixups[] = {
                 .chained = true,
                 .chain_id = ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE,
         },
+        [ALC668_FIXUP_MIC_COEF] = {
+                .type = HDA_FIXUP_VERBS,
+                .v.verbs = (const struct hda_verb[]) {
+                        { 0x20, AC_VERB_SET_COEF_INDEX, 0xc3 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x4000 },
+                        {}
+                },
+        },
+        [ALC668_FIXUP_ASUS_G751] = {
+                .type = HDA_FIXUP_PINS,
+                .v.pins = (const struct hda_pintbl[]) {
+                        { 0x16, 0x0421101f }, /* HP */
+                        {}
+                },
+                .chained = true,
+                .chain_id = ALC668_FIXUP_MIC_COEF
+        },
         [ALC891_FIXUP_HEADSET_MODE] = {
                 .type = HDA_FIXUP_FUNC,
                 .v.func = alc_fixup_headset_mode,
@@ -8080,6 +8105,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50),
         SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A),
         SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50),
+        SND_PCI_QUIRK(0x1043, 0x12ff, "ASUS G751", ALC668_FIXUP_ASUS_G751),
         SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP),
         SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16),
         SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51),
@@ -8184,6 +8210,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = {
         {.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 = ALC668_FIXUP_ASUS_G751, .name = "asus-g751"},
         {.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"},
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c
index f63acb1b965c..c49d25bcd7f2 100644
--- a/sound/pci/hda/patch_si3054.c
+++ b/sound/pci/hda/patch_si3054.c
@@ -27,7 +27,7 @@
 #include <linux/slab.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 
 /* si3054 verbs */
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 046705b4691a..1b6ecfb01759 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -32,7 +32,7 @@
 #include <linux/module.h>
 #include <sound/core.h>
 #include <sound/jack.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_beep.h"
@@ -77,6 +77,7 @@ enum {
         STAC_DELL_M6_BOTH,
         STAC_DELL_EQ,
         STAC_ALIENWARE_M17X,
+        STAC_ELO_VUPOINT_15MX,
         STAC_92HD89XX_HP_FRONT_JACK,
         STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK,
         STAC_92HD73XX_ASUS_MOBO,
@@ -1879,6 +1880,18 @@ static void stac92hd73xx_fixup_no_jd(struct hda_codec *codec,
                 codec->no_jack_detect = 1;
 }
 
+
+static void stac92hd73xx_disable_automute(struct hda_codec *codec,
+                                     const struct hda_fixup *fix, int action)
+{
+        struct sigmatel_spec *spec = codec->spec;
+
+        if (action != HDA_FIXUP_ACT_PRE_PROBE)
+                return;
+
+        spec->gen.suppress_auto_mute = 1;
+}
+
 static const struct hda_fixup stac92hd73xx_fixups[] = {
         [STAC_92HD73XX_REF] = {
                 .type = HDA_FIXUP_FUNC,
@@ -1904,6 +1917,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = {
                 .type = HDA_FIXUP_FUNC,
                 .v.func = stac92hd73xx_fixup_alienware_m17x,
         },
+        [STAC_ELO_VUPOINT_15MX] = {
+                .type = HDA_FIXUP_FUNC,
+                .v.func = stac92hd73xx_disable_automute,
+        },
         [STAC_92HD73XX_INTEL] = {
                 .type = HDA_FIXUP_PINS,
                 .v.pins = intel_dg45id_pin_configs,
@@ -1942,6 +1959,7 @@ static const struct hda_model_fixup stac92hd73xx_models[] = {
         { .id = STAC_DELL_M6_BOTH, .name = "dell-m6" },
         { .id = STAC_DELL_EQ, .name = "dell-eq" },
         { .id = STAC_ALIENWARE_M17X, .name = "alienware" },
+        { .id = STAC_ELO_VUPOINT_15MX, .name = "elo-vupoint-15mx" },
         { .id = STAC_92HD73XX_ASUS_MOBO, .name = "asus-mobo" },
         {}
 };
@@ -1991,6 +2009,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = {
                       "Alienware M17x", STAC_ALIENWARE_M17X),
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490,
                       "Alienware M17x R3", STAC_DELL_EQ),
+        SND_PCI_QUIRK(0x1059, 0x1011,
+                      "ELO VuPoint 15MX", STAC_ELO_VUPOINT_15MX),
         SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1927,
                                 "HP Z1 G2", STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK),
         SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17,
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 6b9617aee0e6..9f6f13e25145 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -52,7 +52,7 @@
 #include <linux/module.h>
 #include <sound/core.h>
 #include <sound/asoundef.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 5ee468d1aefe..ffddcdfe0c66 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -38,11 +38,6 @@
 #include <sound/ac97_codec.h>
 #include <sound/info.h>
 #include <sound/initval.h>
-/* for 440MX workaround */
-#include <asm/pgtable.h>
-#ifdef CONFIG_X86
-#include <asm/set_memory.h>
-#endif
 
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455");
@@ -374,7 +369,6 @@ struct ichdev {
         unsigned int ali_slot;                  /* ALI DMA slot */
         struct ac97_pcm *pcm;
         int pcm_open_flag;
-        unsigned int page_attr_changed: 1;
         unsigned int suspended: 1;
 };
 
@@ -724,25 +718,6 @@ static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ich
         iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI);
 }
 
-#ifdef __i386__
-/*
- * Intel 82443MX running a 100MHz processor system bus has a hardware bug,
- * which aborts PCI busmaster for audio transfer.  A workaround is to set
- * the pages as non-cached.  For details, see the errata in
- *      http://download.intel.com/design/chipsets/specupdt/24505108.pdf
- */
-static void fill_nocache(void *buf, int size, int nocache)
-{
-        size = (size + PAGE_SIZE - 1) >> PAGE_SHIFT;
-        if (nocache)
-                set_pages_uc(virt_to_page(buf), size);
-        else
-                set_pages_wb(virt_to_page(buf), size);
-}
-#else
-#define fill_nocache(buf, size, nocache) do { ; } while (0)
-#endif
-
 /*
  *  Interrupt handler
  */
@@ -850,7 +825,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd
         switch (cmd) {
         case SNDRV_PCM_TRIGGER_RESUME:
                 ichdev->suspended = 0;
-                /* fallthru */
+                /* fall through */
         case SNDRV_PCM_TRIGGER_START:
         case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
                 val = ICH_IOCE | ICH_STARTBM;
@@ -858,7 +833,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd
                 break;
         case SNDRV_PCM_TRIGGER_SUSPEND:
                 ichdev->suspended = 1;
-                /* fallthru */
+                /* fall through */
         case SNDRV_PCM_TRIGGER_STOP:
                 val = 0;
                 break;
@@ -892,7 +867,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd
         switch (cmd) {
         case SNDRV_PCM_TRIGGER_RESUME:
                 ichdev->suspended = 0;
-                /* fallthru */
+                /* fall through */
         case SNDRV_PCM_TRIGGER_START:
         case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
                 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -909,7 +884,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd
                 break;
         case SNDRV_PCM_TRIGGER_SUSPEND:
                 ichdev->suspended = 1;
-                /* fallthru */
+                /* fall through */
         case SNDRV_PCM_TRIGGER_STOP:
         case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
                 /* pause */
@@ -938,23 +913,12 @@ static int snd_intel8x0_hw_params(struct snd_pcm_substream *substream,
 {
         struct intel8x0 *chip = snd_pcm_substream_chip(substream);
         struct ichdev *ichdev = get_ichdev(substream);
-        struct snd_pcm_runtime *runtime = substream->runtime;
         int dbl = params_rate(hw_params) > 48000;
         int err;
 
-        if (chip->fix_nocache && ichdev->page_attr_changed) {
-                fill_nocache(runtime->dma_area, runtime->dma_bytes, 0); /* clear */
-                ichdev->page_attr_changed = 0;
-        }
         err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
         if (err < 0)
                 return err;
-        if (chip->fix_nocache) {
-                if (runtime->dma_area && ! ichdev->page_attr_changed) {
-                        fill_nocache(runtime->dma_area, runtime->dma_bytes, 1);
-                        ichdev->page_attr_changed = 1;
-                }
-        }
         if (ichdev->pcm_open_flag) {
                 snd_ac97_pcm_close(ichdev->pcm);
                 ichdev->pcm_open_flag = 0;
@@ -974,17 +938,12 @@ static int snd_intel8x0_hw_params(struct snd_pcm_substream *substream,
 
 static int snd_intel8x0_hw_free(struct snd_pcm_substream *substream)
 {
-        struct intel8x0 *chip = snd_pcm_substream_chip(substream);
         struct ichdev *ichdev = get_ichdev(substream);
 
         if (ichdev->pcm_open_flag) {
                 snd_ac97_pcm_close(ichdev->pcm);
                 ichdev->pcm_open_flag = 0;
         }
-        if (chip->fix_nocache && ichdev->page_attr_changed) {
-                fill_nocache(substream->runtime->dma_area, substream->runtime->dma_bytes, 0);
-                ichdev->page_attr_changed = 0;
-        }
         return snd_pcm_lib_free_pages(substream);
 }
 
@@ -1510,6 +1469,9 @@ struct ich_pcm_table {
         int ac97_idx;
 };
 
+#define intel8x0_dma_type(chip) \
+        ((chip)->fix_nocache ? SNDRV_DMA_TYPE_DEV_UC : SNDRV_DMA_TYPE_DEV)
+
 static int snd_intel8x0_pcm1(struct intel8x0 *chip, int device,
                              struct ich_pcm_table *rec)
 {
@@ -1540,7 +1502,7 @@ static int snd_intel8x0_pcm1(struct intel8x0 *chip, int device,
                 strcpy(pcm->name, chip->card->shortname);
         chip->pcm[device] = pcm;
 
-        snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+        snd_pcm_lib_preallocate_pages_for_all(pcm, intel8x0_dma_type(chip),
                                               snd_dma_pci_data(chip->pci),
                                               rec->prealloc_size, rec->prealloc_max_size);
 
@@ -2629,11 +2591,8 @@ static int snd_intel8x0_free(struct intel8x0 *chip)
       __hw_end:
         if (chip->irq >= 0)
                 free_irq(chip->irq, chip);
-        if (chip->bdbars.area) {
-                if (chip->fix_nocache)
-                        fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 0);
+        if (chip->bdbars.area)
                 snd_dma_free_pages(&chip->bdbars);
-        }
         if (chip->addr)
                 pci_iounmap(chip->pci, chip->addr);
         if (chip->bmaddr)
@@ -2657,17 +2616,6 @@ static int intel8x0_suspend(struct device *dev)
         snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
         for (i = 0; i < chip->pcm_devs; i++)
                 snd_pcm_suspend_all(chip->pcm[i]);
-        /* clear nocache */
-        if (chip->fix_nocache) {
-                for (i = 0; i < chip->bdbars_count; i++) {
-                        struct ichdev *ichdev = &chip->ichd[i];
-                        if (ichdev->substream && ichdev->page_attr_changed) {
-                                struct snd_pcm_runtime *runtime = ichdev->substream->runtime;
-                                if (runtime->dma_area)
-                                        fill_nocache(runtime->dma_area, runtime->dma_bytes, 0);
-                        }
-                }
-        }
         for (i = 0; i < chip->ncodecs; i++)
                 snd_ac97_suspend(chip->ac97[i]);
         if (chip->device_type == DEVICE_INTEL_ICH4)
@@ -2708,25 +2656,9 @@ static int intel8x0_resume(struct device *dev)
                           ICH_PCM_SPDIF_1011);
         }
 
-        /* refill nocache */
-        if (chip->fix_nocache)
-                fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 1);
-
         for (i = 0; i < chip->ncodecs; i++)
                 snd_ac97_resume(chip->ac97[i]);
 
-        /* refill nocache */
-        if (chip->fix_nocache) {
-                for (i = 0; i < chip->bdbars_count; i++) {
-                        struct ichdev *ichdev = &chip->ichd[i];
-                        if (ichdev->substream && ichdev->page_attr_changed) {
-                                struct snd_pcm_runtime *runtime = ichdev->substream->runtime;
-                                if (runtime->dma_area)
-                                        fill_nocache(runtime->dma_area, runtime->dma_bytes, 1);
-                        }
-                }
-        }
-
         /* resume status */
         for (i = 0; i < chip->bdbars_count; i++) {
                 struct ichdev *ichdev = &chip->ichd[i];
@@ -3057,6 +2989,12 @@ static int snd_intel8x0_create(struct snd_card *card,
 
         chip->inside_vm = snd_intel8x0_inside_vm(pci);
 
+        /*
+         * Intel 82443MX running a 100MHz processor system bus has a hardware
+         * bug, which aborts PCI busmaster for audio transfer.  A workaround
+         * is to set the pages as non-cached.  For details, see the errata in
+         *     http://download.intel.com/design/chipsets/specupdt/24505108.pdf
+         */
         if (pci->vendor == PCI_VENDOR_ID_INTEL &&
             pci->device == PCI_DEVICE_ID_INTEL_440MX)
                 chip->fix_nocache = 1; /* enable workaround */
@@ -3128,7 +3066,7 @@ static int snd_intel8x0_create(struct snd_card *card,
 
         /* allocate buffer descriptor lists */
         /* the start of each lists must be aligned to 8 bytes */
-        if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+        if (snd_dma_alloc_pages(intel8x0_dma_type(chip), snd_dma_pci_data(pci),
                                 chip->bdbars_count * sizeof(u32) * ICH_MAX_FRAGS * 2,
                                 &chip->bdbars) < 0) {
                 snd_intel8x0_free(chip);
@@ -3137,9 +3075,6 @@ static int snd_intel8x0_create(struct snd_card *card,
         }
         /* tables must be aligned to 8 bytes here, but the kernel pages
            are much bigger, so we don't care (on i386) */
-        /* workaround for 440MX */
-        if (chip->fix_nocache)
-                fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 1);
         int_sta_masks = 0;
         for (i = 0; i < chip->bdbars_count; i++) {
                 ichdev = &chip->ichd[i];
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 943a726b1c1b..c84629190cba 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -1171,16 +1171,6 @@ static int snd_intel8x0m_create(struct snd_card *card,
         }
 
  port_inited:
-        if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED,
-                        KBUILD_MODNAME, chip)) {
-                dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq);
-                snd_intel8x0m_free(chip);
-                return -EBUSY;
-        }
-        chip->irq = pci->irq;
-        pci_set_master(pci);
-        synchronize_irq(chip->irq);
-
         /* initialize offsets */
         chip->bdbars_count = 2;
         tbl = intel_regs;
@@ -1224,11 +1214,21 @@ static int snd_intel8x0m_create(struct snd_card *card,
         chip->int_sta_reg = ICH_REG_GLOB_STA;
         chip->int_sta_mask = int_sta_masks;
 
+        pci_set_master(pci);
+
         if ((err = snd_intel8x0m_chip_init(chip, 1)) < 0) {
                 snd_intel8x0m_free(chip);
                 return err;
         }
 
+        if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED,
+                        KBUILD_MODNAME, chip)) {
+                dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq);
+                snd_intel8x0m_free(chip);
+                return -EBUSY;
+        }
+        chip->irq = pci->irq;
+
         if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
                 snd_intel8x0m_free(chip);
                 return err;
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index f0906ba416d4..3ac8c71d567c 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -319,7 +319,8 @@ static const struct snd_pcm_hardware snd_rme32_spdif_info = {
                          SNDRV_PCM_INFO_MMAP_VALID |
                          SNDRV_PCM_INFO_INTERLEAVED | 
                          SNDRV_PCM_INFO_PAUSE |
-                         SNDRV_PCM_INFO_SYNC_START),
+                         SNDRV_PCM_INFO_SYNC_START |
+                         SNDRV_PCM_INFO_SYNC_APPLPTR),
         .formats =      (SNDRV_PCM_FMTBIT_S16_LE | 
                          SNDRV_PCM_FMTBIT_S32_LE),
         .rates =        (SNDRV_PCM_RATE_32000 |
@@ -346,7 +347,8 @@ static const struct snd_pcm_hardware snd_rme32_adat_info =
                               SNDRV_PCM_INFO_MMAP_VALID |
                               SNDRV_PCM_INFO_INTERLEAVED |
                               SNDRV_PCM_INFO_PAUSE |
-                              SNDRV_PCM_INFO_SYNC_START),
+                              SNDRV_PCM_INFO_SYNC_START |
+                              SNDRV_PCM_INFO_SYNC_APPLPTR),
         .formats=            SNDRV_PCM_FMTBIT_S16_LE,
         .rates =             (SNDRV_PCM_RATE_44100 | 
                               SNDRV_PCM_RATE_48000),
@@ -370,7 +372,8 @@ static const struct snd_pcm_hardware snd_rme32_spdif_fd_info = {
                          SNDRV_PCM_INFO_MMAP_VALID |
                          SNDRV_PCM_INFO_INTERLEAVED | 
                          SNDRV_PCM_INFO_PAUSE |
-                         SNDRV_PCM_INFO_SYNC_START),
+                         SNDRV_PCM_INFO_SYNC_START |
+                         SNDRV_PCM_INFO_SYNC_APPLPTR),
         .formats =      (SNDRV_PCM_FMTBIT_S16_LE | 
                          SNDRV_PCM_FMTBIT_S32_LE),
         .rates =        (SNDRV_PCM_RATE_32000 |
@@ -397,7 +400,8 @@ static const struct snd_pcm_hardware snd_rme32_adat_fd_info =
                               SNDRV_PCM_INFO_MMAP_VALID |
                               SNDRV_PCM_INFO_INTERLEAVED |
                               SNDRV_PCM_INFO_PAUSE |
-                              SNDRV_PCM_INFO_SYNC_START),
+                              SNDRV_PCM_INFO_SYNC_START |
+                              SNDRV_PCM_INFO_SYNC_APPLPTR),
         .formats=            SNDRV_PCM_FMTBIT_S16_LE,
         .rates =             (SNDRV_PCM_RATE_44100 | 
                               SNDRV_PCM_RATE_48000),
@@ -1104,16 +1108,6 @@ snd_rme32_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
                 snd_pcm_trigger_done(s, substream);
         }
         
-        /* prefill playback buffer */
-        if (cmd == SNDRV_PCM_TRIGGER_START && rme32->fullduplex_mode) {
-                snd_pcm_group_for_each_entry(s, substream) {
-                        if (s == rme32->playback_substream) {
-                                s->ops->ack(s);
-                                break;
-                        }
-                }
-        }
-
         switch (cmd) {
         case SNDRV_PCM_TRIGGER_START:
                 if (rme32->running && ! RME32_ISWORKING(rme32))
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 11b5b5e0e058..679ad0415e3b 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6534,7 +6534,7 @@ static int snd_hdspm_create_alsa_devices(struct snd_card *card,
         dev_dbg(card->dev, "Update mixer controls...\n");
         hdspm_update_simple_mixer_controls(hdspm);
 
-        dev_dbg(card->dev, "Initializeing complete ???\n");
+        dev_dbg(card->dev, "Initializing complete?\n");
 
         err = snd_card_register(card);
         if (err < 0) {
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index 8e3275a96a82..3f813ea5210a 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -42,7 +42,7 @@
 #include "../codecs/da7219.h"
 #include "../codecs/da7219-aad.h"
 
-#define CZ_PLAT_CLK 25000000
+#define CZ_PLAT_CLK 48000000
 #define DUAL_CHANNEL            2
 
 static struct snd_soc_jack cz_jack;
@@ -75,7 +75,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd)
         da7219_dai_clk = clk_get(component->dev, "da7219-dai-clks");
 
         ret = snd_soc_card_jack_new(card, "Headset Jack",
-                                SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
+                                SND_JACK_HEADSET | SND_JACK_LINEOUT |
                                 SND_JACK_BTN_0 | SND_JACK_BTN_1 |
                                 SND_JACK_BTN_2 | SND_JACK_BTN_3,
                                 &cz_jack, NULL, 0);
@@ -133,7 +133,7 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = {
         .mask = 0,
 };
 
-static int cz_da7219_startup(struct snd_pcm_substream *substream)
+static int cz_da7219_play_startup(struct snd_pcm_substream *substream)
 {
         struct snd_pcm_runtime *runtime = substream->runtime;
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -150,7 +150,28 @@ 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_SP_INSTANCE;
+        machine->play_i2s_instance = I2S_SP_INSTANCE;
+        return da7219_clk_enable(substream);
+}
+
+static int cz_da7219_cap_startup(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        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);
+
+        /*
+         * On this platform for PCM device we support stereo
+         */
+
+        runtime->hw.channels_max = DUAL_CHANNEL;
+        snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+                                   &constraints_channels);
+        snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+                                   &constraints_rates);
+
+        machine->cap_i2s_instance = I2S_SP_INSTANCE;
         machine->capture_channel = CAP_CHANNEL1;
         return da7219_clk_enable(substream);
 }
@@ -162,11 +183,22 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream)
 
 static int cz_max_startup(struct snd_pcm_substream *substream)
 {
+        struct snd_pcm_runtime *runtime = substream->runtime;
         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;
+        /*
+         * On this platform for PCM device we support stereo
+         */
+
+        runtime->hw.channels_max = DUAL_CHANNEL;
+        snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+                                   &constraints_channels);
+        snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+                                   &constraints_rates);
+
+        machine->play_i2s_instance = I2S_BT_INSTANCE;
         return da7219_clk_enable(substream);
 }
 
@@ -177,21 +209,43 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream)
 
 static int cz_dmic0_startup(struct snd_pcm_substream *substream)
 {
+        struct snd_pcm_runtime *runtime = substream->runtime;
         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;
+        /*
+         * On this platform for PCM device we support stereo
+         */
+
+        runtime->hw.channels_max = DUAL_CHANNEL;
+        snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+                                   &constraints_channels);
+        snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+                                   &constraints_rates);
+
+        machine->cap_i2s_instance = I2S_BT_INSTANCE;
         return da7219_clk_enable(substream);
 }
 
 static int cz_dmic1_startup(struct snd_pcm_substream *substream)
 {
+        struct snd_pcm_runtime *runtime = substream->runtime;
         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;
+        /*
+         * On this platform for PCM device we support stereo
+         */
+
+        runtime->hw.channels_max = DUAL_CHANNEL;
+        snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+                                   &constraints_channels);
+        snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+                                   &constraints_rates);
+
+        machine->cap_i2s_instance = I2S_SP_INSTANCE;
         machine->capture_channel = CAP_CHANNEL0;
         return da7219_clk_enable(substream);
 }
@@ -201,8 +255,13 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream)
         da7219_clk_disable();
 }
 
+static const struct snd_soc_ops cz_da7219_play_ops = {
+        .startup = cz_da7219_play_startup,
+        .shutdown = cz_da7219_shutdown,
+};
+
 static const struct snd_soc_ops cz_da7219_cap_ops = {
-        .startup = cz_da7219_startup,
+        .startup = cz_da7219_cap_startup,
         .shutdown = cz_da7219_shutdown,
 };
 
@@ -233,7 +292,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
                                 | SND_SOC_DAIFMT_CBM_CFM,
                 .init = cz_da7219_init,
                 .dpcm_playback = 1,
-                .ops = &cz_da7219_cap_ops,
+                .ops = &cz_da7219_play_ops,
         },
         {
                 .name = "amd-da7219-cap",
diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c
index 77b265bd0505..cdebab2f8ce5 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -867,8 +867,12 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
                 return -EINVAL;
 
         if (pinfo) {
-                rtd->i2s_instance = pinfo->i2s_instance;
-                rtd->capture_channel = pinfo->capture_channel;
+                if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                        rtd->i2s_instance = pinfo->play_i2s_instance;
+                } else {
+                        rtd->i2s_instance = pinfo->cap_i2s_instance;
+                        rtd->capture_channel = pinfo->capture_channel;
+                }
         }
         if (adata->asic_type == CHIP_STONEY) {
                 val = acp_reg_read(adata->acp_mmio,
@@ -1036,16 +1040,22 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream)
 
         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)
+                if (bytescount >= rtd->bytescount)
                         bytescount -= rtd->bytescount;
-                delay = do_div(bytescount, period_bytes);
-                runtime->delay = bytes_to_frames(runtime, delay);
+                if (bytescount < period_bytes) {
+                        pos = 0;
+                } else {
+                        dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr);
+                        if (dscr == rtd->dma_dscr_idx_1)
+                                pos = period_bytes;
+                        else
+                                pos = 0;
+                }
+                if (bytescount > 0) {
+                        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);
diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h
index be3963e8f4fa..dbbb1a85638d 100644
--- a/sound/soc/amd/acp.h
+++ b/sound/soc/amd/acp.h
@@ -158,7 +158,8 @@ struct audio_drv_data {
  * and dma driver
  */
 struct acp_platform_info {
-        u16 i2s_instance;
+        u16 play_i2s_instance;
+        u16 cap_i2s_instance;
         u16 capture_channel;
 };
 
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index 64b784e96f84..64f86f0b87e5 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -97,4 +97,16 @@ config SND_ATMEL_SOC_I2S
         help
           Say Y or M if you want to add support for Atmel ASoc driver for boards
           using I2S.
+
+config SND_SOC_MIKROE_PROTO
+        tristate "Support for Mikroe-PROTO board"
+        depends on OF
+        depends on SND_SOC_I2C_AND_SPI
+        select SND_SOC_WM8731
+        help
+          Say Y or M if you want to add support for MikroElektronika PROTO Audio
+          Board. This board contains the WM8731 codec, which can be configured
+          using I2C over SDA (MPU Data Input) and SCL (MPU Clock Input) pins.
+          Both playback and capture are supported.
+
 endif
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index cd87cb4bcff5..9f41bfa0fea3 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -17,6 +17,7 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
 snd-atmel-soc-classd-objs := atmel-classd.o
 snd-atmel-soc-pdmic-objs := atmel-pdmic.o
 snd-atmel-soc-tse850-pcm5142-objs := tse850-pcm5142.o
+snd-soc-mikroe-proto-objs := mikroe-proto.o
 
 obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
 obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
@@ -24,3 +25,4 @@ obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o
 obj-$(CONFIG_SND_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o
 obj-$(CONFIG_SND_ATMEL_SOC_PDMIC) += snd-atmel-soc-pdmic.o
 obj-$(CONFIG_SND_ATMEL_SOC_TSE850_PCM5142) += snd-atmel-soc-tse850-pcm5142.o
+obj-$(CONFIG_SND_SOC_MIKROE_PROTO) += snd-soc-mikroe-proto.o
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index d3b69682d9c2..6291ec7f9dd6 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -1005,11 +1005,11 @@ static int asoc_ssc_init(struct device *dev)
         struct ssc_device *ssc = dev_get_drvdata(dev);
         int ret;
 
-        ret = snd_soc_register_component(dev, &atmel_ssc_component,
+        ret = devm_snd_soc_register_component(dev, &atmel_ssc_component,
                                          &atmel_ssc_dai, 1);
         if (ret) {
                 dev_err(dev, "Could not register DAI: %d\n", ret);
-                goto err;
+                return ret;
         }
 
         if (ssc->pdata->use_dma)
@@ -1019,15 +1019,10 @@ static int asoc_ssc_init(struct device *dev)
 
         if (ret) {
                 dev_err(dev, "Could not register PCM: %d\n", ret);
-                goto err_unregister_dai;
+                return ret;
         }
 
         return 0;
-
-err_unregister_dai:
-        snd_soc_unregister_component(dev);
-err:
-        return ret;
 }
 
 static void asoc_ssc_exit(struct device *dev)
@@ -1038,8 +1033,6 @@ static void asoc_ssc_exit(struct device *dev)
                 atmel_pcm_dma_platform_unregister(dev);
         else
                 atmel_pcm_pdc_platform_unregister(dev);
-
-        snd_soc_unregister_component(dev);
 }
 
 /**
diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c
new file mode 100644
index 000000000000..d47aaa5bf75a
--- /dev/null
+++ b/sound/soc/atmel/mikroe-proto.c
@@ -0,0 +1,165 @@
+/*
+ * ASoC driver for PROTO AudioCODEC (with a WM8731)
+ *
+ * Author:      Florian Meier, <koalo@koalo.de>
+ *            Copyright 2013
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "../codecs/wm8731.h"
+
+#define XTAL_RATE 12288000      /* This is fixed on this board */
+
+static int snd_proto_init(struct snd_soc_pcm_runtime *rtd)
+{
+        struct snd_soc_card *card = rtd->card;
+        struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+        /* Set proto sysclk */
+        int ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
+                                         XTAL_RATE, SND_SOC_CLOCK_IN);
+        if (ret < 0) {
+                dev_err(card->dev, "Failed to set WM8731 SYSCLK: %d\n",
+                        ret);
+                return ret;
+        }
+
+        return 0;
+}
+
+static const struct snd_soc_dapm_widget snd_proto_widget[] = {
+        SND_SOC_DAPM_MIC("Microphone Jack", NULL),
+        SND_SOC_DAPM_HP("Headphone Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route snd_proto_route[] = {
+        /* speaker connected to LHPOUT/RHPOUT */
+        {"Headphone Jack", NULL, "LHPOUT"},
+        {"Headphone Jack", NULL, "RHPOUT"},
+
+        /* mic is connected to Mic Jack, with WM8731 Mic Bias */
+        {"MICIN", NULL, "Mic Bias"},
+        {"Mic Bias", NULL, "Microphone Jack"},
+};
+
+/* audio machine driver */
+static struct snd_soc_card snd_proto = {
+        .name           = "snd_mikroe_proto",
+        .owner          = THIS_MODULE,
+        .dapm_widgets   = snd_proto_widget,
+        .num_dapm_widgets = ARRAY_SIZE(snd_proto_widget),
+        .dapm_routes    = snd_proto_route,
+        .num_dapm_routes = ARRAY_SIZE(snd_proto_route),
+};
+
+static int snd_proto_probe(struct platform_device *pdev)
+{
+        struct snd_soc_dai_link *dai;
+        struct device_node *np = pdev->dev.of_node;
+        struct device_node *codec_np, *cpu_np;
+        struct device_node *bitclkmaster = NULL;
+        struct device_node *framemaster = NULL;
+        unsigned int dai_fmt;
+        int ret = 0;
+
+        if (!np) {
+                dev_err(&pdev->dev, "No device node supplied\n");
+                return -EINVAL;
+        }
+
+        snd_proto.dev = &pdev->dev;
+        ret = snd_soc_of_parse_card_name(&snd_proto, "model");
+        if (ret)
+                return ret;
+
+        dai = devm_kzalloc(&pdev->dev, sizeof(*dai), GFP_KERNEL);
+        if (!dai)
+                return -ENOMEM;
+
+        snd_proto.dai_link = dai;
+        snd_proto.num_links = 1;
+
+        dai->name = "WM8731";
+        dai->stream_name = "WM8731 HiFi";
+        dai->codec_dai_name = "wm8731-hifi";
+        dai->init = &snd_proto_init;
+
+        codec_np = of_parse_phandle(np, "audio-codec", 0);
+        if (!codec_np) {
+                dev_err(&pdev->dev, "audio-codec node missing\n");
+                return -EINVAL;
+        }
+        dai->codec_of_node = codec_np;
+
+        cpu_np = of_parse_phandle(np, "i2s-controller", 0);
+        if (!cpu_np) {
+                dev_err(&pdev->dev, "i2s-controller missing\n");
+                return -EINVAL;
+        }
+        dai->cpu_of_node = cpu_np;
+        dai->platform_of_node = cpu_np;
+
+        dai_fmt = snd_soc_of_parse_daifmt(np, NULL,
+                                          &bitclkmaster, &framemaster);
+        if (bitclkmaster != framemaster) {
+                dev_err(&pdev->dev, "Must be the same bitclock and frame master\n");
+                return -EINVAL;
+        }
+        if (bitclkmaster) {
+                dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+                if (codec_np == bitclkmaster)
+                        dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+                else
+                        dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+        }
+        of_node_put(bitclkmaster);
+        of_node_put(framemaster);
+        dai->dai_fmt = dai_fmt;
+
+        of_node_put(codec_np);
+        of_node_put(cpu_np);
+
+        ret = snd_soc_register_card(&snd_proto);
+        if (ret && ret != -EPROBE_DEFER)
+                dev_err(&pdev->dev,
+                        "snd_soc_register_card() failed: %d\n", ret);
+
+        return ret;
+}
+
+static int snd_proto_remove(struct platform_device *pdev)
+{
+        return snd_soc_unregister_card(&snd_proto);
+}
+
+static const struct of_device_id snd_proto_of_match[] = {
+        { .compatible = "mikroe,mikroe-proto", },
+        {},
+};
+MODULE_DEVICE_TABLE(of, snd_proto_of_match);
+
+static struct platform_driver snd_proto_driver = {
+        .driver = {
+                .name   = "snd-mikroe-proto",
+                .of_match_table = snd_proto_of_match,
+        },
+        .probe    = snd_proto_probe,
+        .remove  = snd_proto_remove,
+};
+
+module_platform_driver(snd_proto_driver);
+
+MODULE_AUTHOR("Florian Meier");
+MODULE_DESCRIPTION("ASoC Driver for PROTO board (WM8731)");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c
index 3a1393283156..214adcad5419 100644
--- a/sound/soc/atmel/tse850-pcm5142.c
+++ b/sound/soc/atmel/tse850-pcm5142.c
@@ -1,44 +1,38 @@
-/*
- * TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec
- *
- * Copyright (C) 2016 Axentia Technologies AB
- *
- * Author: Peter Rosin <peda@axentia.se>
- *
- * 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.
- */
-
-/*
- *               loop1 relays
- *   IN1 +---o  +------------+  o---+ OUT1
- *            \                /
- *             +              +
- *             |   /          |
- *             +--o  +--.     |
- *             |  add   |     |
- *             |        V     |
- *             |      .---.   |
- *   DAC +----------->|Sum|---+
- *             |      '---'   |
- *             |              |
- *             +              +
- *
- *   IN2 +---o--+------------+--o---+ OUT2
- *               loop2 relays
- *
- * The 'loop1' gpio pin controlls two relays, which are either in loop
- * position, meaning that input and output are directly connected, or
- * they are in mixer position, meaning that the signal is passed through
- * the 'Sum' mixer. Similarly for 'loop2'.
- *
- * In the above, the 'loop1' relays are inactive, thus feeding IN1 to the
- * mixer (if 'add' is active) and feeding the mixer output to OUT1. The
- * 'loop2' relays are active, short-cutting the TSE-850 from channel 2.
- * IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name
- * of the (filtered) output from the PCM5142 codec.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec
+//
+// Copyright (C) 2016 Axentia Technologies AB
+//
+// Author: Peter Rosin <peda@axentia.se>
+//
+//               loop1 relays
+//   IN1 +---o  +------------+  o---+ OUT1
+//            \                /
+//             +              +
+//             |   /          |
+//             +--o  +--.     |
+//             |  add   |     |
+//             |        V     |
+//             |      .---.   |
+//   DAC +----------->|Sum|---+
+//             |      '---'   |
+//             |              |
+//             +              +
+//
+//   IN2 +---o--+------------+--o---+ OUT2
+//               loop2 relays
+//
+// The 'loop1' gpio pin controlls two relays, which are either in loop
+// position, meaning that input and output are directly connected, or
+// they are in mixer position, meaning that the signal is passed through
+// the 'Sum' mixer. Similarly for 'loop2'.
+//
+// In the above, the 'loop1' relays are inactive, thus feeding IN1 to the
+// mixer (if 'add' is active) and feeding the mixer output to OUT1. The
+// 'loop2' relays are active, short-cutting the TSE-850 from channel 2.
+// IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name
+// of the (filtered) output from the PCM5142 codec.
 
 #include <linux/clk.h>
 #include <linux/gpio.h>
@@ -452,4 +446,4 @@ module_platform_driver(tse850_driver);
 /* Module information */
 MODULE_AUTHOR("Peter Rosin <peda@axentia.se>");
 MODULE_DESCRIPTION("ALSA SoC driver for TSE-850 with PCM5142 codec");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c
index b733f1446353..b7c358b48d8d 100644
--- a/sound/soc/bcm/cygnus-ssp.c
+++ b/sound/soc/bcm/cygnus-ssp.c
@@ -1334,7 +1334,7 @@ static int cygnus_ssp_probe(struct platform_device *pdev)
         cygaud->active_ports = 0;
 
         dev_dbg(dev, "Registering %d DAIs\n", active_port_count);
-        err = snd_soc_register_component(dev, &cygnus_ssp_component,
+        err = devm_snd_soc_register_component(dev, &cygnus_ssp_component,
                                 cygnus_ssp_dai, active_port_count);
         if (err) {
                 dev_err(dev, "snd_soc_register_dai failed\n");
@@ -1345,32 +1345,27 @@ static int cygnus_ssp_probe(struct platform_device *pdev)
         if (cygaud->irq_num <= 0) {
                 dev_err(dev, "platform_get_irq failed\n");
                 err = cygaud->irq_num;
-                goto err_irq;
+                return err;
         }
 
         err = audio_clk_init(pdev, cygaud);
         if (err) {
                 dev_err(dev, "audio clock initialization failed\n");
-                goto err_irq;
+                return err;
         }
 
         err = cygnus_soc_platform_register(dev, cygaud);
         if (err) {
                 dev_err(dev, "platform reg error %d\n", err);
-                goto err_irq;
+                return err;
         }
 
         return 0;
-
-err_irq:
-        snd_soc_unregister_component(dev);
-        return err;
 }
 
 static int cygnus_ssp_remove(struct platform_device *pdev)
 {
         cygnus_soc_platform_unregister(&pdev->dev);
-        snd_soc_unregister_component(&pdev->dev);
 
         return 0;
 }
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index efb095dbcd71..9cc4f1848c9b 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -82,6 +82,7 @@ config SND_SOC_ALL_CODECS
         select SND_SOC_ES7241
         select SND_SOC_GTM601
         select SND_SOC_HDAC_HDMI
+        select SND_SOC_HDAC_HDA
         select SND_SOC_ICS43432
         select SND_SOC_INNO_RK3036
         select SND_SOC_ISABELLE if I2C
@@ -109,6 +110,7 @@ config SND_SOC_ALL_CODECS
         select SND_SOC_MT6351 if MTK_PMIC_WRAP
         select SND_SOC_NAU8540 if I2C
         select SND_SOC_NAU8810 if I2C
+        select SND_SOC_NAU8822 if I2C
         select SND_SOC_NAU8824 if I2C
         select SND_SOC_NAU8825 if I2C
         select SND_SOC_HDMI_CODEC
@@ -119,6 +121,8 @@ config SND_SOC_ALL_CODECS
         select SND_SOC_PCM186X_I2C if I2C
         select SND_SOC_PCM186X_SPI if SPI_MASTER
         select SND_SOC_PCM3008
+        select SND_SOC_PCM3060_I2C if I2C
+        select SND_SOC_PCM3060_SPI if SPI_MASTER
         select SND_SOC_PCM3168A_I2C if I2C
         select SND_SOC_PCM3168A_SPI if SPI_MASTER
         select SND_SOC_PCM5102A
@@ -575,7 +579,11 @@ config SND_SOC_DA9055
         tristate
 
 config SND_SOC_DMIC
-        tristate
+        tristate "Generic Digital Microphone CODEC"
+        depends on GPIOLIB
+        help
+          Enable support for the Generic Digital Microphone CODEC.
+          Select this if your sound card has DMICs.
 
 config SND_SOC_HDMI_CODEC
         tristate
@@ -615,6 +623,10 @@ config SND_SOC_HDAC_HDMI
         select SND_PCM_ELD
         select HDMI
 
+config SND_SOC_HDAC_HDA
+        tristate
+        select SND_HDA
+
 config SND_SOC_ICS43432
         tristate
 
@@ -629,7 +641,8 @@ config SND_SOC_LM49453
         tristate
 
 config SND_SOC_MAX98088
-       tristate
+        tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec"
+        depends on I2C
 
 config SND_SOC_MAX98090
        tristate
@@ -732,6 +745,21 @@ config SND_SOC_PCM186X_SPI
 config SND_SOC_PCM3008
        tristate
 
+config SND_SOC_PCM3060
+       tristate
+
+config SND_SOC_PCM3060_I2C
+        tristate "Texas Instruments PCM3060 CODEC - I2C"
+        depends on I2C
+        select SND_SOC_PCM3060
+        select REGMAP_I2C
+
+config SND_SOC_PCM3060_SPI
+        tristate "Texas Instruments PCM3060 CODEC - SPI"
+        depends on SPI_MASTER
+        select SND_SOC_PCM3060
+        select REGMAP_SPI
+
 config SND_SOC_PCM3168A
         tristate
 
@@ -1299,6 +1327,10 @@ config SND_SOC_NAU8810
         tristate "Nuvoton Technology Corporation NAU88C10 CODEC"
         depends on I2C
 
+config SND_SOC_NAU8822
+        tristate "Nuvoton Technology Corporation NAU88C22 CODEC"
+        depends on I2C
+
 config SND_SOC_NAU8824
         tristate "Nuvoton Technology Corporation NAU88L24 CODEC"
         depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 7ae7c85e8219..8ffab8c8dbfa 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -78,6 +78,7 @@ snd-soc-es8328-i2c-objs := es8328-i2c.o
 snd-soc-es8328-spi-objs := es8328-spi.o
 snd-soc-gtm601-objs := gtm601.o
 snd-soc-hdac-hdmi-objs := hdac_hdmi.o
+snd-soc-hdac-hda-objs := hdac_hda.o
 snd-soc-ics43432-objs := ics43432.o
 snd-soc-inno-rk3036-objs := inno_rk3036.o
 snd-soc-isabelle-objs := isabelle.o
@@ -106,6 +107,7 @@ snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o
 snd-soc-mt6351-objs := mt6351.o
 snd-soc-nau8540-objs := nau8540.o
 snd-soc-nau8810-objs := nau8810.o
+snd-soc-nau8822-objs := nau8822.o
 snd-soc-nau8824-objs := nau8824.o
 snd-soc-nau8825-objs := nau8825.o
 snd-soc-hdmi-codec-objs := hdmi-codec.o
@@ -119,6 +121,9 @@ snd-soc-pcm186x-objs := pcm186x.o
 snd-soc-pcm186x-i2c-objs := pcm186x-i2c.o
 snd-soc-pcm186x-spi-objs := pcm186x-spi.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-pcm3060-objs := pcm3060.o
+snd-soc-pcm3060-i2c-objs := pcm3060-i2c.o
+snd-soc-pcm3060-spi-objs := pcm3060-spi.o
 snd-soc-pcm3168a-objs := pcm3168a.o
 snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o
 snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o
@@ -338,6 +343,7 @@ obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
 obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o
 obj-$(CONFIG_SND_SOC_GTM601)    += snd-soc-gtm601.o
 obj-$(CONFIG_SND_SOC_HDAC_HDMI) += snd-soc-hdac-hdmi.o
+obj-$(CONFIG_SND_SOC_HDAC_HDA) += snd-soc-hdac-hda.o
 obj-$(CONFIG_SND_SOC_ICS43432)  += snd-soc-ics43432.o
 obj-$(CONFIG_SND_SOC_INNO_RK3036)       += snd-soc-inno-rk3036.o
 obj-$(CONFIG_SND_SOC_ISABELLE)  += snd-soc-isabelle.o
@@ -366,6 +372,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o
 obj-$(CONFIG_SND_SOC_MT6351)    += snd-soc-mt6351.o
 obj-$(CONFIG_SND_SOC_NAU8540)   += snd-soc-nau8540.o
 obj-$(CONFIG_SND_SOC_NAU8810)   += snd-soc-nau8810.o
+obj-$(CONFIG_SND_SOC_NAU8822)   += snd-soc-nau8822.o
 obj-$(CONFIG_SND_SOC_NAU8824)   += snd-soc-nau8824.o
 obj-$(CONFIG_SND_SOC_NAU8825)   += snd-soc-nau8825.o
 obj-$(CONFIG_SND_SOC_HDMI_CODEC)        += snd-soc-hdmi-codec.o
@@ -379,6 +386,9 @@ obj-$(CONFIG_SND_SOC_PCM186X)	+= snd-soc-pcm186x.o
 obj-$(CONFIG_SND_SOC_PCM186X_I2C)       += snd-soc-pcm186x-i2c.o
 obj-$(CONFIG_SND_SOC_PCM186X_SPI)       += snd-soc-pcm186x-spi.o
 obj-$(CONFIG_SND_SOC_PCM3008)   += snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_PCM3060)   += snd-soc-pcm3060.o
+obj-$(CONFIG_SND_SOC_PCM3060_I2C)       += snd-soc-pcm3060-i2c.o
+obj-$(CONFIG_SND_SOC_PCM3060_SPI)       += snd-soc-pcm3060-spi.o
 obj-$(CONFIG_SND_SOC_PCM3168A)  += snd-soc-pcm3168a.o
 obj-$(CONFIG_SND_SOC_PCM3168A_I2C)      += snd-soc-pcm3168a-i2c.o
 obj-$(CONFIG_SND_SOC_PCM3168A_SPI)      += snd-soc-pcm3168a-spi.o
diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c
index be136e981653..bef3e9e74c26 100644
--- a/sound/soc/codecs/adau1761.c
+++ b/sound/soc/codecs/adau1761.c
@@ -518,7 +518,8 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_component *component)
                         ARRAY_SIZE(adau1761_jack_detect_controls));
                 if (ret)
                         return ret;
-        case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */
+                /* fall through */
+        case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE:
                 ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes,
                         ARRAY_SIZE(adau1761_no_dmic_routes));
                 if (ret)
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 57169b8ff14e..3959e6ad113d 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -21,11 +21,18 @@
 #include <linux/i2c.h>
 #include <linux/spi/spi.h>
 #include <linux/regmap.h>
+#include <asm/unaligned.h>
 
 #include "sigmadsp.h"
 #include "adau17x1.h"
 #include "adau-utils.h"
 
+#define ADAU17X1_SAFELOAD_TARGET_ADDRESS 0x0006
+#define ADAU17X1_SAFELOAD_TRIGGER 0x0007
+#define ADAU17X1_SAFELOAD_DATA 0x0001
+#define ADAU17X1_SAFELOAD_DATA_SIZE 20
+#define ADAU17X1_WORD_SIZE 4
+
 static const char * const adau17x1_capture_mixer_boost_text[] = {
         "Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3",
 };
@@ -60,6 +67,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = {
         SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum),
 };
 
+static int adau17x1_setup_firmware(struct snd_soc_component *component,
+        unsigned int rate);
+
 static int adau17x1_pll_event(struct snd_soc_dapm_widget *w,
         struct snd_kcontrol *kcontrol, int event)
 {
@@ -313,7 +323,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = {
         { "Capture", NULL, "Right Decimator" },
 };
 
-bool adau17x1_has_dsp(struct adau *adau)
+static bool adau17x1_has_dsp(struct adau *adau)
 {
         switch (adau->type) {
         case ADAU1761:
@@ -324,7 +334,17 @@ bool adau17x1_has_dsp(struct adau *adau)
                 return false;
         }
 }
-EXPORT_SYMBOL_GPL(adau17x1_has_dsp);
+
+static bool adau17x1_has_safeload(struct adau *adau)
+{
+        switch (adau->type) {
+        case ADAU1761:
+        case ADAU1781:
+                return true;
+        default:
+                return false;
+        }
+}
 
 static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id,
         int source, unsigned int freq_in, unsigned int freq_out)
@@ -836,7 +856,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg)
 }
 EXPORT_SYMBOL_GPL(adau17x1_volatile_register);
 
-int adau17x1_setup_firmware(struct snd_soc_component *component,
+static int adau17x1_setup_firmware(struct snd_soc_component *component,
         unsigned int rate)
 {
         int ret;
@@ -880,7 +900,6 @@ err:
 
         return ret;
 }
-EXPORT_SYMBOL_GPL(adau17x1_setup_firmware);
 
 int adau17x1_add_widgets(struct snd_soc_component *component)
 {
@@ -957,6 +976,56 @@ int adau17x1_resume(struct snd_soc_component *component)
 }
 EXPORT_SYMBOL_GPL(adau17x1_resume);
 
+static int adau17x1_safeload(struct sigmadsp *sigmadsp, unsigned int addr,
+        const uint8_t bytes[], size_t len)
+{
+        uint8_t buf[ADAU17X1_WORD_SIZE];
+        uint8_t data[ADAU17X1_SAFELOAD_DATA_SIZE];
+        unsigned int addr_offset;
+        unsigned int nbr_words;
+        int ret;
+
+        /* write data to safeload addresses. Check if len is not a multiple of
+         * 4 bytes, if so we need to zero pad.
+         */
+        nbr_words = len / ADAU17X1_WORD_SIZE;
+        if ((len - nbr_words * ADAU17X1_WORD_SIZE) == 0) {
+                ret = regmap_raw_write(sigmadsp->control_data,
+                        ADAU17X1_SAFELOAD_DATA, bytes, len);
+        } else {
+                nbr_words++;
+                memset(data, 0, ADAU17X1_SAFELOAD_DATA_SIZE);
+                memcpy(data, bytes, len);
+                ret = regmap_raw_write(sigmadsp->control_data,
+                        ADAU17X1_SAFELOAD_DATA, data,
+                        nbr_words * ADAU17X1_WORD_SIZE);
+        }
+
+        if (ret < 0)
+                return ret;
+
+        /* Write target address, target address is offset by 1 */
+        addr_offset = addr - 1;
+        put_unaligned_be32(addr_offset, buf);
+        ret = regmap_raw_write(sigmadsp->control_data,
+                ADAU17X1_SAFELOAD_TARGET_ADDRESS, buf, ADAU17X1_WORD_SIZE);
+        if (ret < 0)
+                return ret;
+
+        /* write nbr of words to trigger address */
+        put_unaligned_be32(nbr_words, buf);
+        ret = regmap_raw_write(sigmadsp->control_data,
+                ADAU17X1_SAFELOAD_TRIGGER, buf, ADAU17X1_WORD_SIZE);
+        if (ret < 0)
+                return ret;
+
+        return 0;
+}
+
+static const struct sigmadsp_ops adau17x1_sigmadsp_ops = {
+        .safeload = adau17x1_safeload,
+};
+
 int adau17x1_probe(struct device *dev, struct regmap *regmap,
         enum adau17x1_type type, void (*switch_mode)(struct device *dev),
         const char *firmware_name)
@@ -1002,8 +1071,13 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap,
         dev_set_drvdata(dev, adau);
 
         if (firmware_name) {
-                adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, NULL,
-                        firmware_name);
+                if (adau17x1_has_safeload(adau)) {
+                        adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap,
+                                &adau17x1_sigmadsp_ops, firmware_name);
+                } else {
+                        adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap,
+                                NULL, firmware_name);
+                }
                 if (IS_ERR(adau->sigmadsp)) {
                         dev_warn(dev, "Could not find firmware file: %ld\n",
                                 PTR_ERR(adau->sigmadsp));
diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h
index e6fe87beec07..98a3b6f5bc96 100644
--- a/sound/soc/codecs/adau17x1.h
+++ b/sound/soc/codecs/adau17x1.h
@@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component);
 
 extern const struct snd_soc_dai_ops adau17x1_dai_ops;
 
-int adau17x1_setup_firmware(struct snd_soc_component *component,
-        unsigned int rate);
-bool adau17x1_has_dsp(struct adau *adau);
-
 #define ADAU17X1_CLOCK_CONTROL                  0x4000
 #define ADAU17X1_PLL_CONTROL                    0x4002
 #define ADAU17X1_REC_POWER_MGMT                 0x4009
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index 407554175282..ab27d2b94d02 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -154,11 +154,11 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = {
         SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1,
                                 6, 1, 0),
         SOC_ENUM("C Data Access", cam_mode_enum),
+        SOC_SINGLE("SPDIF Switch", CS4265_SPDIF_CTL2, 5, 1, 1),
         SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2,
                                 3, 1, 0),
         SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum),
-        SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2,
-                                0, 1, 0),
+        SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, 0, 1, 0),
         SOC_ENUM("Mono Channel Select", spdif_mono_select_enum),
         SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24),
 };
@@ -221,10 +221,11 @@ static const struct snd_soc_dapm_route cs4265_audio_map[] = {
         {"LINEOUTR", NULL, "DAC"},
         {"SPDIFOUT", NULL, "SPDIF"},
 
+        {"Pre-amp MIC", NULL, "MICL"},
+        {"Pre-amp MIC", NULL, "MICR"},
+        {"ADC Mux", "MIC", "Pre-amp MIC"},
         {"ADC Mux", "LINEIN", "LINEINL"},
         {"ADC Mux", "LINEIN", "LINEINR"},
-        {"ADC Mux", "MIC", "MICL"},
-        {"ADC Mux", "MIC", "MICR"},
         {"ADC", NULL, "ADC Mux"},
         {"DOUT", NULL, "ADC"},
         {"DAI1 Capture", NULL, "DOUT"},
@@ -496,7 +497,8 @@ static int cs4265_set_bias_level(struct snd_soc_component *component,
                         SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
 
 #define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \
-                        SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE)
+                        SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE | \
+                        SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE)
 
 static const struct snd_soc_dai_ops cs4265_ops = {
         .hw_params      = cs4265_pcm_hw_params,
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 5080d7a3c279..fd2bd74024c1 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -21,6 +21,7 @@
  *  - master mode *NOT* supported
  */
 
+#include <linux/clk.h>
 #include <linux/module.h>
 #include <linux/slab.h>
 #include <sound/core.h>
@@ -41,6 +42,7 @@ enum master_slave_mode {
 
 struct cs42l51_private {
         unsigned int mclk;
+        struct clk *mclk_handle;
         unsigned int audio_mode;        /* The mode (I2S or left-justified) */
         enum master_slave_mode func;
 };
@@ -237,6 +239,10 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = {
                 &cs42l51_adcr_mux_controls),
 };
 
+static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = {
+        SND_SOC_DAPM_CLOCK_SUPPLY("MCLK")
+};
+
 static const struct snd_soc_dapm_route cs42l51_routes[] = {
         {"HPL", NULL, "Left DAC"},
         {"HPR", NULL, "Right DAC"},
@@ -487,6 +493,14 @@ static struct snd_soc_dai_driver cs42l51_dai = {
 static int cs42l51_component_probe(struct snd_soc_component *component)
 {
         int ret, reg;
+        struct snd_soc_dapm_context *dapm;
+        struct cs42l51_private *cs42l51;
+
+        cs42l51 = snd_soc_component_get_drvdata(component);
+        dapm = snd_soc_component_get_dapm(component);
+
+        if (cs42l51->mclk_handle)
+                snd_soc_dapm_new_controls(dapm, cs42l51_dapm_mclk_widgets, 1);
 
         /*
          * DAC configuration
@@ -540,6 +554,13 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap)
 
         dev_set_drvdata(dev, cs42l51);
 
+        cs42l51->mclk_handle = devm_clk_get(dev, "MCLK");
+        if (IS_ERR(cs42l51->mclk_handle)) {
+                if (PTR_ERR(cs42l51->mclk_handle) != -ENOENT)
+                        return PTR_ERR(cs42l51->mclk_handle);
+                cs42l51->mclk_handle = NULL;
+        }
+
         /* Verify that we have a CS42L51 */
         ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val);
         if (ret < 0) {
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index 8c4926df9286..71322e0410ee 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -148,6 +148,7 @@ static const struct of_device_id dmic_dev_match[] = {
         {.compatible = "dmic-codec"},
         {}
 };
+MODULE_DEVICE_TABLE(of, dmic_dev_match);
 
 static struct platform_driver dmic_driver = {
         .driver = {
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 4b5827dc23aa..04a3aa770722 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -566,14 +566,14 @@ static int es8328_set_sysclk(struct snd_soc_dai *codec_dai,
                 break;
         case 22579200:
                 mclkdiv2 = 1;
-                /* fallthru */
+                /* fall through */
         case 11289600:
                 es8328->sysclk_constraints = &constraints_11289;
                 es8328->mclk_ratios = ratios_11289;
                 break;
         case 24576000:
                 mclkdiv2 = 1;
-                /* fallthru */
+                /* fall through */
         case 12288000:
                 es8328->sysclk_constraints = &constraints_12288;
                 es8328->mclk_ratios = ratios_12288;
diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c
new file mode 100644
index 000000000000..2aaa83028e55
--- /dev/null
+++ b/sound/soc/codecs/hdac_hda.c
@@ -0,0 +1,483 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2015-18 Intel Corporation.
+
+/*
+ * hdac_hda.c - ASoC extensions to reuse the legacy HDA codec drivers
+ * with ASoC platform drivers. These APIs are called by the legacy HDA
+ * codec drivers using hdac_ext_bus_ops ops.
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/pm_runtime.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/hdaudio_ext.h>
+#include <sound/hda_codec.h>
+#include <sound/hda_register.h>
+#include "hdac_hda.h"
+
+#define HDAC_ANALOG_DAI_ID              0
+#define HDAC_DIGITAL_DAI_ID             1
+#define HDAC_ALT_ANALOG_DAI_ID          2
+
+#define STUB_FORMATS    (SNDRV_PCM_FMTBIT_S8 | \
+                        SNDRV_PCM_FMTBIT_U8 | \
+                        SNDRV_PCM_FMTBIT_S16_LE | \
+                        SNDRV_PCM_FMTBIT_U16_LE | \
+                        SNDRV_PCM_FMTBIT_S24_LE | \
+                        SNDRV_PCM_FMTBIT_U24_LE | \
+                        SNDRV_PCM_FMTBIT_S32_LE | \
+                        SNDRV_PCM_FMTBIT_U32_LE | \
+                        SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
+
+static int hdac_hda_dai_open(struct snd_pcm_substream *substream,
+                             struct snd_soc_dai *dai);
+static void hdac_hda_dai_close(struct snd_pcm_substream *substream,
+                               struct snd_soc_dai *dai);
+static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream,
+                                struct snd_soc_dai *dai);
+static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream,
+                                struct snd_soc_dai *dai);
+static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai,
+                                     unsigned int tx_mask, unsigned int rx_mask,
+                                     int slots, int slot_width);
+static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt,
+                                                 struct snd_soc_dai *dai);
+
+static struct snd_soc_dai_ops hdac_hda_dai_ops = {
+        .startup = hdac_hda_dai_open,
+        .shutdown = hdac_hda_dai_close,
+        .prepare = hdac_hda_dai_prepare,
+        .hw_free = hdac_hda_dai_hw_free,
+        .set_tdm_slot = hdac_hda_dai_set_tdm_slot,
+};
+
+static struct snd_soc_dai_driver hdac_hda_dais[] = {
+{
+        .id = HDAC_ANALOG_DAI_ID,
+        .name = "Analog Codec DAI",
+        .ops = &hdac_hda_dai_ops,
+        .playback = {
+                .stream_name    = "Analog Codec Playback",
+                .channels_min   = 1,
+                .channels_max   = 16,
+                .rates          = SNDRV_PCM_RATE_8000_192000,
+                .formats        = STUB_FORMATS,
+                .sig_bits       = 24,
+        },
+        .capture = {
+                .stream_name    = "Analog Codec Capture",
+                .channels_min   = 1,
+                .channels_max   = 16,
+                .rates = SNDRV_PCM_RATE_8000_192000,
+                .formats = STUB_FORMATS,
+                .sig_bits = 24,
+        },
+},
+{
+        .id = HDAC_DIGITAL_DAI_ID,
+        .name = "Digital Codec DAI",
+        .ops = &hdac_hda_dai_ops,
+        .playback = {
+                .stream_name    = "Digital Codec Playback",
+                .channels_min   = 1,
+                .channels_max   = 16,
+                .rates          = SNDRV_PCM_RATE_8000_192000,
+                .formats        = STUB_FORMATS,
+                .sig_bits = 24,
+        },
+        .capture = {
+                .stream_name    = "Digital Codec Capture",
+                .channels_min   = 1,
+                .channels_max   = 16,
+                .rates = SNDRV_PCM_RATE_8000_192000,
+                .formats = STUB_FORMATS,
+                .sig_bits = 24,
+        },
+},
+{
+        .id = HDAC_ALT_ANALOG_DAI_ID,
+        .name = "Alt Analog Codec DAI",
+        .ops = &hdac_hda_dai_ops,
+        .playback = {
+                .stream_name    = "Alt Analog Codec Playback",
+                .channels_min   = 1,
+                .channels_max   = 16,
+                .rates          = SNDRV_PCM_RATE_8000_192000,
+                .formats        = STUB_FORMATS,
+                .sig_bits       = 24,
+        },
+        .capture = {
+                .stream_name    = "Alt Analog Codec Capture",
+                .channels_min   = 1,
+                .channels_max   = 16,
+                .rates = SNDRV_PCM_RATE_8000_192000,
+                .formats = STUB_FORMATS,
+                .sig_bits = 24,
+        },
+}
+
+};
+
+static int hdac_hda_dai_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;
+        struct hdac_hda_priv *hda_pvt;
+        struct hdac_hda_pcm *pcm;
+
+        hda_pvt = snd_soc_component_get_drvdata(component);
+        pcm = &hda_pvt->pcm[dai->id];
+        if (tx_mask)
+                pcm[dai->id].stream_tag[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask;
+        else
+                pcm[dai->id].stream_tag[SNDRV_PCM_STREAM_CAPTURE] = rx_mask;
+
+        return 0;
+}
+
+static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream,
+                                struct snd_soc_dai *dai)
+{
+        struct snd_soc_component *component = dai->component;
+        struct hdac_hda_priv *hda_pvt;
+        struct hda_pcm_stream *hda_stream;
+        struct hda_pcm *pcm;
+
+        hda_pvt = snd_soc_component_get_drvdata(component);
+        pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+        if (!pcm)
+                return -EINVAL;
+
+        hda_stream = &pcm->stream[substream->stream];
+        snd_hda_codec_cleanup(&hda_pvt->codec, hda_stream, substream);
+
+        return 0;
+}
+
+static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream,
+                                struct snd_soc_dai *dai)
+{
+        struct snd_soc_component *component = dai->component;
+        struct hdac_hda_priv *hda_pvt;
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct hdac_device *hdev;
+        struct hda_pcm_stream *hda_stream;
+        unsigned int format_val;
+        struct hda_pcm *pcm;
+        unsigned int stream;
+        int ret = 0;
+
+        hda_pvt = snd_soc_component_get_drvdata(component);
+        hdev = &hda_pvt->codec.core;
+        pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+        if (!pcm)
+                return -EINVAL;
+
+        hda_stream = &pcm->stream[substream->stream];
+
+        format_val = snd_hdac_calc_stream_format(runtime->rate,
+                                                 runtime->channels,
+                                                 runtime->format,
+                                                 hda_stream->maxbps,
+                                                 0);
+        if (!format_val) {
+                dev_err(&hdev->dev,
+                        "invalid format_val, rate=%d, ch=%d, format=%d\n",
+                        runtime->rate, runtime->channels, runtime->format);
+                return -EINVAL;
+        }
+
+        stream = hda_pvt->pcm[dai->id].stream_tag[substream->stream];
+
+        ret = snd_hda_codec_prepare(&hda_pvt->codec, hda_stream,
+                                    stream, format_val, substream);
+        if (ret < 0)
+                dev_err(&hdev->dev, "codec prepare failed %d\n", ret);
+
+        return ret;
+}
+
+static int hdac_hda_dai_open(struct snd_pcm_substream *substream,
+                             struct snd_soc_dai *dai)
+{
+        struct snd_soc_component *component = dai->component;
+        struct hdac_hda_priv *hda_pvt;
+        struct hda_pcm_stream *hda_stream;
+        struct hda_pcm *pcm;
+        int ret;
+
+        hda_pvt = snd_soc_component_get_drvdata(component);
+        pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+        if (!pcm)
+                return -EINVAL;
+
+        snd_hda_codec_pcm_get(pcm);
+
+        hda_stream = &pcm->stream[substream->stream];
+
+        ret = hda_stream->ops.open(hda_stream, &hda_pvt->codec, substream);
+        if (ret < 0)
+                snd_hda_codec_pcm_put(pcm);
+
+        return ret;
+}
+
+static void hdac_hda_dai_close(struct snd_pcm_substream *substream,
+                               struct snd_soc_dai *dai)
+{
+        struct snd_soc_component *component = dai->component;
+        struct hdac_hda_priv *hda_pvt;
+        struct hda_pcm_stream *hda_stream;
+        struct hda_pcm *pcm;
+
+        hda_pvt = snd_soc_component_get_drvdata(component);
+        pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+        if (!pcm)
+                return;
+
+        hda_stream = &pcm->stream[substream->stream];
+
+        hda_stream->ops.close(hda_stream, &hda_pvt->codec, substream);
+
+        snd_hda_codec_pcm_put(pcm);
+}
+
+static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt,
+                                                 struct snd_soc_dai *dai)
+{
+        struct hda_codec *hcodec = &hda_pvt->codec;
+        struct hda_pcm *cpcm;
+        const char *pcm_name;
+
+        switch (dai->id) {
+        case HDAC_ANALOG_DAI_ID:
+                pcm_name = "Analog";
+                break;
+        case HDAC_DIGITAL_DAI_ID:
+                pcm_name = "Digital";
+                break;
+        case HDAC_ALT_ANALOG_DAI_ID:
+                pcm_name = "Alt Analog";
+                break;
+        default:
+                dev_err(&hcodec->core.dev, "invalid dai id %d\n", dai->id);
+                return NULL;
+        }
+
+        list_for_each_entry(cpcm, &hcodec->pcm_list_head, list) {
+                if (strpbrk(cpcm->name, pcm_name))
+                        return cpcm;
+        }
+
+        dev_err(&hcodec->core.dev, "didn't find PCM for DAI %s\n", dai->name);
+        return NULL;
+}
+
+static int hdac_hda_codec_probe(struct snd_soc_component *component)
+{
+        struct hdac_hda_priv *hda_pvt =
+                        snd_soc_component_get_drvdata(component);
+        struct snd_soc_dapm_context *dapm =
+                        snd_soc_component_get_dapm(component);
+        struct hdac_device *hdev = &hda_pvt->codec.core;
+        struct hda_codec *hcodec = &hda_pvt->codec;
+        struct hdac_ext_link *hlink;
+        hda_codec_patch_t patch;
+        int ret;
+
+        hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+        if (!hlink) {
+                dev_err(&hdev->dev, "hdac link not found\n");
+                return -EIO;
+        }
+
+        snd_hdac_ext_bus_link_get(hdev->bus, hlink);
+
+        ret = snd_hda_codec_device_new(hcodec->bus, component->card->snd_card,
+                                       hdev->addr, hcodec);
+        if (ret < 0) {
+                dev_err(&hdev->dev, "failed to create hda codec %d\n", ret);
+                goto error_no_pm;
+        }
+
+        /*
+         * snd_hda_codec_device_new decrements the usage count so call get pm
+         * else the device will be powered off
+         */
+        pm_runtime_get_noresume(&hdev->dev);
+
+        hcodec->bus->card = dapm->card->snd_card;
+
+        ret = snd_hda_codec_set_name(hcodec, hcodec->preset->name);
+        if (ret < 0) {
+                dev_err(&hdev->dev, "name failed %s\n", hcodec->preset->name);
+                goto error;
+        }
+
+        ret = snd_hdac_regmap_init(&hcodec->core);
+        if (ret < 0) {
+                dev_err(&hdev->dev, "regmap init failed\n");
+                goto error;
+        }
+
+        patch = (hda_codec_patch_t)hcodec->preset->driver_data;
+        if (patch) {
+                ret = patch(hcodec);
+                if (ret < 0) {
+                        dev_err(&hdev->dev, "patch failed %d\n", ret);
+                        goto error;
+                }
+        } else {
+                dev_dbg(&hdev->dev, "no patch file found\n");
+        }
+
+        ret = snd_hda_codec_parse_pcms(hcodec);
+        if (ret < 0) {
+                dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret);
+                goto error;
+        }
+
+        ret = snd_hda_codec_build_controls(hcodec);
+        if (ret < 0) {
+                dev_err(&hdev->dev, "unable to create controls %d\n", ret);
+                goto error;
+        }
+
+        hcodec->core.lazy_cache = true;
+
+        /*
+         * 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 is also called during codec registeration
+         */
+        pm_runtime_put(&hdev->dev);
+        pm_runtime_suspend(&hdev->dev);
+
+        return 0;
+
+error:
+        pm_runtime_put(&hdev->dev);
+error_no_pm:
+        snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+        return ret;
+}
+
+static void hdac_hda_codec_remove(struct snd_soc_component *component)
+{
+        struct hdac_hda_priv *hda_pvt =
+                      snd_soc_component_get_drvdata(component);
+        struct hdac_device *hdev = &hda_pvt->codec.core;
+        struct hdac_ext_link *hlink = NULL;
+
+        hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+        if (!hlink) {
+                dev_err(&hdev->dev, "hdac link not found\n");
+                return;
+        }
+
+        snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+        pm_runtime_disable(&hdev->dev);
+}
+
+static const struct snd_soc_dapm_route hdac_hda_dapm_routes[] = {
+        {"AIF1TX", NULL, "Codec Input Pin1"},
+        {"AIF2TX", NULL, "Codec Input Pin2"},
+        {"AIF3TX", NULL, "Codec Input Pin3"},
+
+        {"Codec Output Pin1", NULL, "AIF1RX"},
+        {"Codec Output Pin2", NULL, "AIF2RX"},
+        {"Codec Output Pin3", NULL, "AIF3RX"},
+};
+
+static const struct snd_soc_dapm_widget hdac_hda_dapm_widgets[] = {
+        /* Audio Interface */
+        SND_SOC_DAPM_AIF_IN("AIF1RX", "Analog Codec Playback", 0,
+                            SND_SOC_NOPM, 0, 0),
+        SND_SOC_DAPM_AIF_IN("AIF2RX", "Digital Codec Playback", 0,
+                            SND_SOC_NOPM, 0, 0),
+        SND_SOC_DAPM_AIF_IN("AIF3RX", "Alt Analog Codec Playback", 0,
+                            SND_SOC_NOPM, 0, 0),
+        SND_SOC_DAPM_AIF_OUT("AIF1TX", "Analog Codec Capture", 0,
+                             SND_SOC_NOPM, 0, 0),
+        SND_SOC_DAPM_AIF_OUT("AIF2TX", "Digital Codec Capture", 0,
+                             SND_SOC_NOPM, 0, 0),
+        SND_SOC_DAPM_AIF_OUT("AIF3TX", "Alt Analog Codec Capture", 0,
+                             SND_SOC_NOPM, 0, 0),
+
+        /* Input Pins */
+        SND_SOC_DAPM_INPUT("Codec Input Pin1"),
+        SND_SOC_DAPM_INPUT("Codec Input Pin2"),
+        SND_SOC_DAPM_INPUT("Codec Input Pin3"),
+
+        /* Output Pins */
+        SND_SOC_DAPM_OUTPUT("Codec Output Pin1"),
+        SND_SOC_DAPM_OUTPUT("Codec Output Pin2"),
+        SND_SOC_DAPM_OUTPUT("Codec Output Pin3"),
+};
+
+static const struct snd_soc_component_driver hdac_hda_codec = {
+        .probe          = hdac_hda_codec_probe,
+        .remove         = hdac_hda_codec_remove,
+        .idle_bias_on   = false,
+        .dapm_widgets           = hdac_hda_dapm_widgets,
+        .num_dapm_widgets       = ARRAY_SIZE(hdac_hda_dapm_widgets),
+        .dapm_routes            = hdac_hda_dapm_routes,
+        .num_dapm_routes        = ARRAY_SIZE(hdac_hda_dapm_routes),
+};
+
+static int hdac_hda_dev_probe(struct hdac_device *hdev)
+{
+        struct hdac_ext_link *hlink;
+        struct hdac_hda_priv *hda_pvt;
+        int ret;
+
+        /* hold the ref while we probe */
+        hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+        if (!hlink) {
+                dev_err(&hdev->dev, "hdac link not found\n");
+                return -EIO;
+        }
+        snd_hdac_ext_bus_link_get(hdev->bus, hlink);
+
+        hda_pvt = hdac_to_hda_priv(hdev);
+        if (!hda_pvt)
+                return -ENOMEM;
+
+        /* ASoC specific initialization */
+        ret = devm_snd_soc_register_component(&hdev->dev,
+                                         &hdac_hda_codec, hdac_hda_dais,
+                                         ARRAY_SIZE(hdac_hda_dais));
+        if (ret < 0) {
+                dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret);
+                return ret;
+        }
+
+        dev_set_drvdata(&hdev->dev, hda_pvt);
+        snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+
+        return ret;
+}
+
+static int hdac_hda_dev_remove(struct hdac_device *hdev)
+{
+        return 0;
+}
+
+static struct hdac_ext_bus_ops hdac_ops = {
+        .hdev_attach = hdac_hda_dev_probe,
+        .hdev_detach = hdac_hda_dev_remove,
+};
+
+struct hdac_ext_bus_ops *snd_soc_hdac_hda_get_ops(void)
+{
+        return &hdac_ops;
+}
+EXPORT_SYMBOL_GPL(snd_soc_hdac_hda_get_ops);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("ASoC Extensions for legacy HDA Drivers");
+MODULE_AUTHOR("Rakesh Ughreja<rakesh.a.ughreja@intel.com>");
diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h
new file mode 100644
index 000000000000..e444ef593360
--- /dev/null
+++ b/sound/soc/codecs/hdac_hda.h
@@ -0,0 +1,24 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright(c) 2015-18 Intel Corporation.
+ */
+
+#ifndef __HDAC_HDA_H__
+#define __HDAC_HDA_H__
+
+struct hdac_hda_pcm {
+        int stream_tag[2];
+};
+
+struct hdac_hda_priv {
+        struct hda_codec codec;
+        struct hdac_hda_pcm pcm[2];
+};
+
+#define hdac_to_hda_priv(_hdac) \
+                        container_of(_hdac, struct hdac_hda_priv, codec.core)
+#define hdac_to_hda_codec(_hdac) container_of(_hdac, struct hda_codec, core)
+
+struct hdac_ext_bus_ops *snd_soc_hdac_hda_get_ops(void);
+
+#endif /* __HDAC_HDA_H__ */
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 7b8533abf637..4e9854889a95 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1410,6 +1410,12 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev,
                 if (ret)
                         return ret;
 
+                /* Filter out 44.1, 88.2 and 176.4Khz */
+                rates &= ~(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 |
+                           SNDRV_PCM_RATE_176400);
+                if (!rates)
+                        return -EINVAL;
+
                 sprintf(dai_name, "intel-hdmi-hifi%d", i+1);
                 hdmi_dais[i].name = devm_kstrdup(&hdev->dev,
                                         dai_name, GFP_KERNEL);
@@ -1598,7 +1604,7 @@ static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card,
 {
         struct snd_soc_pcm_runtime *rtd;
 
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
                 if (rtd->pcm && (rtd->pcm->device == device))
                         return rtd->pcm;
         }
@@ -1961,9 +1967,6 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdev, int pcm_idx)
 
         port = list_first_entry(&pcm->port_list, struct hdac_hdmi_port, head);
 
-        if (!port)
-                return 0;
-
         if (!port || !port->eld.eld_valid)
                 return 0;
 
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index fb515aaa54fc..ca172a4b6849 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -16,6 +16,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/regmap.h>
+#include <linux/clk.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -42,6 +43,7 @@ struct max98088_priv {
         struct regmap *regmap;
         enum max98088_type devtype;
         struct max98088_pdata *pdata;
+        struct clk *mclk;
         unsigned int sysclk;
         struct max98088_cdata dai[2];
         int eq_textcnt;
@@ -1103,6 +1105,11 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai,
        if (freq == max98088->sysclk)
                return 0;
 
+        if (!IS_ERR(max98088->mclk)) {
+                freq = clk_round_rate(max98088->mclk, freq);
+                clk_set_rate(max98088->mclk, freq);
+        }
+
        /* Setup clocks for slave mode, and using the PLL
         * PSCLK = 0x01 (when master clk is 10MHz to 20MHz)
         *         0x02 (when master clk is 20MHz to 30MHz)..
@@ -1310,6 +1317,20 @@ static int max98088_set_bias_level(struct snd_soc_component *component,
                 break;
 
         case SND_SOC_BIAS_PREPARE:
+                /*
+                 * SND_SOC_BIAS_PREPARE is called while preparing for a
+                 * transition to ON or away from ON. If current bias_level
+                 * is SND_SOC_BIAS_ON, then it is preparing for a transition
+                 * away from ON. Disable the clock in that case, otherwise
+                 * enable it.
+                 */
+                if (!IS_ERR(max98088->mclk)) {
+                        if (snd_soc_component_get_bias_level(component) ==
+                            SND_SOC_BIAS_ON)
+                                clk_disable_unprepare(max98088->mclk);
+                        else
+                                clk_prepare_enable(max98088->mclk);
+                }
                 break;
 
         case SND_SOC_BIAS_STANDBY:
@@ -1725,6 +1746,11 @@ static int max98088_i2c_probe(struct i2c_client *i2c,
        if (IS_ERR(max98088->regmap))
                return PTR_ERR(max98088->regmap);
 
+        max98088->mclk = devm_clk_get(&i2c->dev, "mclk");
+        if (IS_ERR(max98088->mclk))
+                if (PTR_ERR(max98088->mclk) == -EPROBE_DEFER)
+                        return PTR_ERR(max98088->mclk);
+
        max98088->devtype = id->driver_data;
 
        i2c_set_clientdata(i2c, max98088);
@@ -1742,9 +1768,19 @@ static const struct i2c_device_id max98088_i2c_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, max98088_i2c_id);
 
+#if defined(CONFIG_OF)
+static const struct of_device_id max98088_of_match[] = {
+        { .compatible = "maxim,max98088" },
+        { .compatible = "maxim,max98089" },
+        { }
+};
+MODULE_DEVICE_TABLE(of, max98088_of_match);
+#endif
+
 static struct i2c_driver max98088_i2c_driver = {
         .driver = {
                 .name = "max98088",
+                .of_match_table = of_match_ptr(max98088_of_match),
         },
         .probe  = max98088_i2c_probe,
         .id_table = max98088_i2c_id,
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index 1093f766d0d2..a09d01318f79 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -2,6 +2,7 @@
 // Copyright (c) 2017, Maxim Integrated
 
 #include <linux/acpi.h>
+#include <linux/delay.h>
 #include <linux/i2c.h>
 #include <linux/module.h>
 #include <linux/regmap.h>
@@ -454,7 +455,7 @@ SND_SOC_DAPM_SIGGEN("IMON"),
 SND_SOC_DAPM_SIGGEN("FBMON"),
 };
 
-static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, 0, -50, 0);
+static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, -6350, 50, 1);
 static const DECLARE_TLV_DB_RANGE(max98373_spk_tlv,
         0, 8, TLV_DB_SCALE_ITEM(0, 50, 0),
         9, 10, TLV_DB_SCALE_ITEM(500, 100, 0),
@@ -470,19 +471,19 @@ static const DECLARE_TLV_DB_RANGE(max98373_dht_spkgain_min_tlv,
         0, 9, TLV_DB_SCALE_ITEM(800, 100, 0),
 );
 static const DECLARE_TLV_DB_RANGE(max98373_dht_rotation_point_tlv,
-        0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0),
-        2, 7, TLV_DB_SCALE_ITEM(-200, -100, 0),
-        8, 9, TLV_DB_SCALE_ITEM(-1000, -200, 0),
-        10, 11, TLV_DB_SCALE_ITEM(-1500, -300, 0),
-        12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0),
-        14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0),
+        0, 1, TLV_DB_SCALE_ITEM(-3000, 500, 0),
+        2, 4, TLV_DB_SCALE_ITEM(-2200, 200, 0),
+        5, 6, TLV_DB_SCALE_ITEM(-1500, 300, 0),
+        7, 9, TLV_DB_SCALE_ITEM(-1000, 200, 0),
+        10, 13, TLV_DB_SCALE_ITEM(-500, 100, 0),
+        14, 15, TLV_DB_SCALE_ITEM(-100, 50, 0),
 );
 static const DECLARE_TLV_DB_RANGE(max98373_limiter_thresh_tlv,
-        0, 15, TLV_DB_SCALE_ITEM(0, -100, 0),
+        0, 15, TLV_DB_SCALE_ITEM(-1500, 100, 0),
 );
 
 static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv,
-        0, 60, TLV_DB_SCALE_ITEM(0, -25, 0),
+        0, 60, TLV_DB_SCALE_ITEM(-1500, 25, 0),
 );
 
 static bool max98373_readable_register(struct device *dev, unsigned int reg)
@@ -604,7 +605,7 @@ SOC_SINGLE("Dither Switch", MAX98373_R203F_AMP_DSP_CFG,
 SOC_SINGLE("DC Blocker Switch", MAX98373_R203F_AMP_DSP_CFG,
         MAX98373_AMP_DSP_CFG_DCBLK_SHIFT, 1, 0),
 SOC_SINGLE_TLV("Digital Volume", MAX98373_R203D_AMP_DIG_VOL_CTRL,
-        0, 0x7F, 0, max98373_digital_tlv),
+        0, 0x7F, 1, max98373_digital_tlv),
 SOC_SINGLE_TLV("Speaker Volume", MAX98373_R203E_AMP_PATH_GAIN,
         MAX98373_SPK_DIGI_GAIN_SHIFT, 10, 0, max98373_spk_tlv),
 SOC_SINGLE_TLV("FS Max Volume", MAX98373_R203E_AMP_PATH_GAIN,
@@ -616,7 +617,7 @@ SOC_SINGLE("DHT Switch", MAX98373_R20D4_DHT_EN,
 SOC_SINGLE_TLV("DHT Min Volume", MAX98373_R20D1_DHT_CFG,
         MAX98373_DHT_SPK_GAIN_MIN_SHIFT, 9, 0, max98373_dht_spkgain_min_tlv),
 SOC_SINGLE_TLV("DHT Rot Pnt Volume", MAX98373_R20D1_DHT_CFG,
-        MAX98373_DHT_ROT_PNT_SHIFT, 15, 0, max98373_dht_rotation_point_tlv),
+        MAX98373_DHT_ROT_PNT_SHIFT, 15, 1, max98373_dht_rotation_point_tlv),
 SOC_SINGLE_TLV("DHT Attack Step Volume", MAX98373_R20D2_DHT_ATTACK_CFG,
         MAX98373_DHT_ATTACK_STEP_SHIFT, 4, 0, max98373_dht_step_size_tlv),
 SOC_SINGLE_TLV("DHT Release Step Volume", MAX98373_R20D3_DHT_RELEASE_CFG,
@@ -653,29 +654,29 @@ SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0),
 SOC_SINGLE("BDE Attack Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 4, 0xF, 0),
 SOC_SINGLE("BDE Release Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 0, 0xF, 0),
 SOC_SINGLE_TLV("BDE LVL1 Clip Thresh Volume", MAX98373_R20A9_BDE_L1_CFG_2,
-        0, 0x3C, 0, max98373_bde_gain_tlv),
+        0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL2 Clip Thresh Volume", MAX98373_R20AC_BDE_L2_CFG_2,
-        0, 0x3C, 0, max98373_bde_gain_tlv),
+        0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL3 Clip Thresh Volume", MAX98373_R20AF_BDE_L3_CFG_2,
-        0, 0x3C, 0, max98373_bde_gain_tlv),
+        0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL4 Clip Thresh Volume", MAX98373_R20B2_BDE_L4_CFG_2,
-        0, 0x3C, 0, max98373_bde_gain_tlv),
+        0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL1 Clip Reduction Volume", MAX98373_R20AA_BDE_L1_CFG_3,
-        0, 0x3C, 0, max98373_bde_gain_tlv),
+        0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL2 Clip Reduction Volume", MAX98373_R20AD_BDE_L2_CFG_3,
-        0, 0x3C, 0, max98373_bde_gain_tlv),
+        0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL3 Clip Reduction Volume", MAX98373_R20B0_BDE_L3_CFG_3,
-        0, 0x3C, 0, max98373_bde_gain_tlv),
+        0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL4 Clip Reduction Volume", MAX98373_R20B3_BDE_L4_CFG_3,
-        0, 0x3C, 0, max98373_bde_gain_tlv),
+        0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL1 Limiter Thresh Volume", MAX98373_R20A8_BDE_L1_CFG_1,
-        0, 0xF, 0, max98373_limiter_thresh_tlv),
+        0, 0xF, 1, max98373_limiter_thresh_tlv),
 SOC_SINGLE_TLV("BDE LVL2 Limiter Thresh Volume", MAX98373_R20AB_BDE_L2_CFG_1,
-        0, 0xF, 0, max98373_limiter_thresh_tlv),
+        0, 0xF, 1, max98373_limiter_thresh_tlv),
 SOC_SINGLE_TLV("BDE LVL3 Limiter Thresh Volume", MAX98373_R20AE_BDE_L3_CFG_1,
-        0, 0xF, 0, max98373_limiter_thresh_tlv),
+        0, 0xF, 1, max98373_limiter_thresh_tlv),
 SOC_SINGLE_TLV("BDE LVL4 Limiter Thresh Volume", MAX98373_R20B1_BDE_L4_CFG_1,
-        0, 0xF, 0, max98373_limiter_thresh_tlv),
+        0, 0xF, 1, max98373_limiter_thresh_tlv),
 /* Limiter */
 SOC_SINGLE("Limiter Switch", MAX98373_R20E2_LIMITER_EN,
         MAX98373_LIMITER_EN_SHIFT, 1, 0),
diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c
new file mode 100644
index 000000000000..622ce947f134
--- /dev/null
+++ b/sound/soc/codecs/nau8822.c
@@ -0,0 +1,1136 @@
+/*
+ * nau8822.c  --  NAU8822 ALSA Soc Audio Codec driver
+ *
+ * Copyright 2017 Nuvoton Technology Corp.
+ *
+ * Author: David Lin <ctlin0@nuvoton.com>
+ * Co-author: John Hsu <kchsu0@nuvoton.com>
+ * Co-author: Seven Li <wtli@nuvoton.com>
+ *
+ * Based on WM8974.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.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <asm/div64.h>
+#include "nau8822.h"
+
+#define NAU_PLL_FREQ_MAX 100000000
+#define NAU_PLL_FREQ_MIN 90000000
+#define NAU_PLL_REF_MAX 33000000
+#define NAU_PLL_REF_MIN 8000000
+#define NAU_PLL_OPTOP_MIN 6
+
+static const int nau8822_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 };
+
+static const struct reg_default nau8822_reg_defaults[] = {
+        { NAU8822_REG_POWER_MANAGEMENT_1, 0x0000 },
+        { NAU8822_REG_POWER_MANAGEMENT_2, 0x0000 },
+        { NAU8822_REG_POWER_MANAGEMENT_3, 0x0000 },
+        { NAU8822_REG_AUDIO_INTERFACE, 0x0050 },
+        { NAU8822_REG_COMPANDING_CONTROL, 0x0000 },
+        { NAU8822_REG_CLOCKING, 0x0140 },
+        { NAU8822_REG_ADDITIONAL_CONTROL, 0x0000 },
+        { NAU8822_REG_GPIO_CONTROL, 0x0000 },
+        { NAU8822_REG_JACK_DETECT_CONTROL_1, 0x0000 },
+        { NAU8822_REG_DAC_CONTROL, 0x0000 },
+        { NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, 0x00ff },
+        { NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0x00ff },
+        { NAU8822_REG_JACK_DETECT_CONTROL_2, 0x0000 },
+        { NAU8822_REG_ADC_CONTROL, 0x0100 },
+        { NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, 0x00ff },
+        { NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0x00ff },
+        { NAU8822_REG_EQ1, 0x012c },
+        { NAU8822_REG_EQ2, 0x002c },
+        { NAU8822_REG_EQ3, 0x002c },
+        { NAU8822_REG_EQ4, 0x002c },
+        { NAU8822_REG_EQ5, 0x002c },
+        { NAU8822_REG_DAC_LIMITER_1, 0x0032 },
+        { NAU8822_REG_DAC_LIMITER_2, 0x0000 },
+        { NAU8822_REG_NOTCH_FILTER_1, 0x0000 },
+        { NAU8822_REG_NOTCH_FILTER_2, 0x0000 },
+        { NAU8822_REG_NOTCH_FILTER_3, 0x0000 },
+        { NAU8822_REG_NOTCH_FILTER_4, 0x0000 },
+        { NAU8822_REG_ALC_CONTROL_1, 0x0038 },
+        { NAU8822_REG_ALC_CONTROL_2, 0x000b },
+        { NAU8822_REG_ALC_CONTROL_3, 0x0032 },
+        { NAU8822_REG_NOISE_GATE, 0x0010 },
+        { NAU8822_REG_PLL_N, 0x0008 },
+        { NAU8822_REG_PLL_K1, 0x000c },
+        { NAU8822_REG_PLL_K2, 0x0093 },
+        { NAU8822_REG_PLL_K3, 0x00e9 },
+        { NAU8822_REG_3D_CONTROL, 0x0000 },
+        { NAU8822_REG_RIGHT_SPEAKER_CONTROL, 0x0000 },
+        { NAU8822_REG_INPUT_CONTROL, 0x0033 },
+        { NAU8822_REG_LEFT_INP_PGA_CONTROL, 0x0010 },
+        { NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0x0010 },
+        { NAU8822_REG_LEFT_ADC_BOOST_CONTROL, 0x0100 },
+        { NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0x0100 },
+        { NAU8822_REG_OUTPUT_CONTROL, 0x0002 },
+        { NAU8822_REG_LEFT_MIXER_CONTROL, 0x0001 },
+        { NAU8822_REG_RIGHT_MIXER_CONTROL, 0x0001 },
+        { NAU8822_REG_LHP_VOLUME, 0x0039 },
+        { NAU8822_REG_RHP_VOLUME, 0x0039 },
+        { NAU8822_REG_LSPKOUT_VOLUME, 0x0039 },
+        { NAU8822_REG_RSPKOUT_VOLUME, 0x0039 },
+        { NAU8822_REG_AUX2_MIXER, 0x0001 },
+        { NAU8822_REG_AUX1_MIXER, 0x0001 },
+        { NAU8822_REG_POWER_MANAGEMENT_4, 0x0000 },
+        { NAU8822_REG_LEFT_TIME_SLOT, 0x0000 },
+        { NAU8822_REG_MISC, 0x0020 },
+        { NAU8822_REG_RIGHT_TIME_SLOT, 0x0000 },
+        { NAU8822_REG_DEVICE_REVISION, 0x007f },
+        { NAU8822_REG_DEVICE_ID, 0x001a },
+        { NAU8822_REG_DAC_DITHER, 0x0114 },
+        { NAU8822_REG_ALC_ENHANCE_1, 0x0000 },
+        { NAU8822_REG_ALC_ENHANCE_2, 0x0000 },
+        { NAU8822_REG_192KHZ_SAMPLING, 0x0008 },
+        { NAU8822_REG_MISC_CONTROL, 0x0000 },
+        { NAU8822_REG_INPUT_TIEOFF, 0x0000 },
+        { NAU8822_REG_POWER_REDUCTION, 0x0000 },
+        { NAU8822_REG_AGC_PEAK2PEAK, 0x0000 },
+        { NAU8822_REG_AGC_PEAK_DETECT, 0x0000 },
+        { NAU8822_REG_AUTOMUTE_CONTROL, 0x0000 },
+        { NAU8822_REG_OUTPUT_TIEOFF, 0x0000 },
+};
+
+static bool nau8822_readable_reg(struct device *dev, unsigned int reg)
+{
+        switch (reg) {
+        case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1:
+        case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME:
+        case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME:
+        case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5:
+        case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2:
+        case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4:
+        case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3:
+        case NAU8822_REG_3D_CONTROL:
+        case NAU8822_REG_RIGHT_SPEAKER_CONTROL:
+        case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL:
+        case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER:
+        case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID:
+        case NAU8822_REG_DAC_DITHER:
+        case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL:
+        case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF:
+                return true;
+        default:
+                return false;
+        }
+}
+
+static bool nau8822_writeable_reg(struct device *dev, unsigned int reg)
+{
+        switch (reg) {
+        case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1:
+        case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME:
+        case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME:
+        case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5:
+        case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2:
+        case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4:
+        case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3:
+        case NAU8822_REG_3D_CONTROL:
+        case NAU8822_REG_RIGHT_SPEAKER_CONTROL:
+        case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL:
+        case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER:
+        case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID:
+        case NAU8822_REG_DAC_DITHER:
+        case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL:
+        case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF:
+                return true;
+        default:
+                return false;
+        }
+}
+
+static bool nau8822_volatile(struct device *dev, unsigned int reg)
+{
+        switch (reg) {
+        case NAU8822_REG_RESET:
+        case NAU8822_REG_DEVICE_REVISION:
+        case NAU8822_REG_DEVICE_ID:
+        case NAU8822_REG_AGC_PEAK2PEAK:
+        case NAU8822_REG_AGC_PEAK_DETECT:
+        case NAU8822_REG_AUTOMUTE_CONTROL:
+                return true;
+        default:
+                return false;
+        }
+}
+
+/* The EQ parameters get function is to get the 5 band equalizer control.
+ * The regmap raw read can't work here because regmap doesn't provide
+ * value format for value width of 9 bits. Therefore, the driver reads data
+ * from cache and makes value format according to the endianness of
+ * bytes type control element.
+ */
+static int nau8822_eq_get(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_component *component =
+                        snd_soc_kcontrol_component(kcontrol);
+        struct soc_bytes_ext *params = (void *)kcontrol->private_value;
+        int i, reg;
+        u16 reg_val, *val;
+
+        val = (u16 *)ucontrol->value.bytes.data;
+        reg = NAU8822_REG_EQ1;
+        for (i = 0; i < params->max / sizeof(u16); i++) {
+                reg_val = snd_soc_component_read32(component, reg + i);
+                /* conversion of 16-bit integers between native CPU format
+                 * and big endian format
+                 */
+                reg_val = cpu_to_be16(reg_val);
+                memcpy(val + i, &reg_val, sizeof(reg_val));
+        }
+
+        return 0;
+}
+
+/* The EQ parameters put function is to make configuration of 5 band equalizer
+ * control. These configuration includes central frequency, equalizer gain,
+ * cut-off frequency, bandwidth control, and equalizer path.
+ * The regmap raw write can't work here because regmap doesn't provide
+ * register and value format for register with address 7 bits and value 9 bits.
+ * Therefore, the driver makes value format according to the endianness of
+ * bytes type control element and writes data to codec.
+ */
+static int nau8822_eq_put(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_component *component =
+                        snd_soc_kcontrol_component(kcontrol);
+        struct soc_bytes_ext *params = (void *)kcontrol->private_value;
+        void *data;
+        u16 *val, value;
+        int i, reg, ret;
+
+        data = kmemdup(ucontrol->value.bytes.data,
+                params->max, GFP_KERNEL | GFP_DMA);
+        if (!data)
+                return -ENOMEM;
+
+        val = (u16 *)data;
+        reg = NAU8822_REG_EQ1;
+        for (i = 0; i < params->max / sizeof(u16); i++) {
+                /* conversion of 16-bit integers between native CPU format
+                 * and big endian format
+                 */
+                value = be16_to_cpu(*(val + i));
+                ret = snd_soc_component_write(component, reg + i, value);
+                if (ret) {
+                        dev_err(component->dev,
+                            "EQ configuration fail, register: %x ret: %d\n",
+                            reg + i, ret);
+                        kfree(data);
+                        return ret;
+                }
+        }
+        kfree(data);
+
+        return 0;
+}
+
+static const char * const nau8822_companding[] = {
+        "Off", "NC", "u-law", "A-law"};
+
+static const struct soc_enum nau8822_companding_adc_enum =
+        SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_ADCCM_SFT,
+                ARRAY_SIZE(nau8822_companding), nau8822_companding);
+
+static const struct soc_enum nau8822_companding_dac_enum =
+        SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_DACCM_SFT,
+                ARRAY_SIZE(nau8822_companding), nau8822_companding);
+
+static const char * const nau8822_eqmode[] = {"Capture", "Playback"};
+
+static const struct soc_enum nau8822_eqmode_enum =
+        SOC_ENUM_SINGLE(NAU8822_REG_EQ1, NAU8822_EQM_SFT,
+                ARRAY_SIZE(nau8822_eqmode), nau8822_eqmode);
+
+static const char * const nau8822_alc1[] = {"Off", "Right", "Left", "Both"};
+static const char * const nau8822_alc3[] = {"Normal", "Limiter"};
+
+static const struct soc_enum nau8822_alc_enable_enum =
+        SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_1, NAU8822_ALCEN_SFT,
+                ARRAY_SIZE(nau8822_alc1), nau8822_alc1);
+
+static const struct soc_enum nau8822_alc_mode_enum =
+        SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_3, NAU8822_ALCM_SFT,
+                ARRAY_SIZE(nau8822_alc3), nau8822_alc3);
+
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0);
+static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0);
+static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0);
+static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1);
+static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0);
+
+static const struct snd_kcontrol_new nau8822_snd_controls[] = {
+        SOC_ENUM("ADC Companding", nau8822_companding_adc_enum),
+        SOC_ENUM("DAC Companding", nau8822_companding_dac_enum),
+
+        SOC_ENUM("EQ Function", nau8822_eqmode_enum),
+        SND_SOC_BYTES_EXT("EQ Parameters", 10,
+                  nau8822_eq_get, nau8822_eq_put),
+
+        SOC_DOUBLE("DAC Inversion Switch",
+                NAU8822_REG_DAC_CONTROL, 0, 1, 1, 0),
+        SOC_DOUBLE_R_TLV("PCM Volume",
+                NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME,
+                NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv),
+
+        SOC_SINGLE("High Pass Filter Switch",
+                NAU8822_REG_ADC_CONTROL, 8, 1, 0),
+        SOC_SINGLE("High Pass Cut Off",
+                NAU8822_REG_ADC_CONTROL, 4, 7, 0),
+
+        SOC_DOUBLE("ADC Inversion Switch",
+                NAU8822_REG_ADC_CONTROL, 0, 1, 1, 0),
+        SOC_DOUBLE_R_TLV("ADC Volume",
+                NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME,
+                NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv),
+
+        SOC_SINGLE("DAC Limiter Switch",
+                NAU8822_REG_DAC_LIMITER_1, 8, 1, 0),
+        SOC_SINGLE("DAC Limiter Decay",
+                NAU8822_REG_DAC_LIMITER_1, 4, 15, 0),
+        SOC_SINGLE("DAC Limiter Attack",
+                NAU8822_REG_DAC_LIMITER_1, 0, 15, 0),
+        SOC_SINGLE("DAC Limiter Threshold",
+                NAU8822_REG_DAC_LIMITER_2, 4, 7, 0),
+        SOC_SINGLE_TLV("DAC Limiter Volume",
+                NAU8822_REG_DAC_LIMITER_2, 0, 12, 0, limiter_tlv),
+
+        SOC_ENUM("ALC Mode", nau8822_alc_mode_enum),
+        SOC_ENUM("ALC Enable Switch", nau8822_alc_enable_enum),
+        SOC_SINGLE("ALC Min Gain",
+                NAU8822_REG_ALC_CONTROL_1, 0, 7, 0),
+        SOC_SINGLE("ALC Max Gain",
+                NAU8822_REG_ALC_CONTROL_1, 3, 7, 0),
+        SOC_SINGLE("ALC Hold",
+                NAU8822_REG_ALC_CONTROL_2, 4, 10, 0),
+        SOC_SINGLE("ALC Target",
+                NAU8822_REG_ALC_CONTROL_2, 0, 15, 0),
+        SOC_SINGLE("ALC Decay",
+                NAU8822_REG_ALC_CONTROL_3, 4, 10, 0),
+        SOC_SINGLE("ALC Attack",
+                NAU8822_REG_ALC_CONTROL_3, 0, 10, 0),
+        SOC_SINGLE("ALC Noise Gate Switch",
+                NAU8822_REG_NOISE_GATE, 3, 1, 0),
+        SOC_SINGLE("ALC Noise Gate Threshold",
+                NAU8822_REG_NOISE_GATE, 0, 7, 0),
+
+        SOC_DOUBLE_R("PGA ZC Switch",
+                NAU8822_REG_LEFT_INP_PGA_CONTROL,
+                NAU8822_REG_RIGHT_INP_PGA_CONTROL,
+                7, 1, 0),
+        SOC_DOUBLE_R_TLV("PGA Volume",
+                NAU8822_REG_LEFT_INP_PGA_CONTROL,
+                NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0, 63, 0, inpga_tlv),
+
+        SOC_DOUBLE_R("Headphone ZC Switch",
+                NAU8822_REG_LHP_VOLUME,
+                NAU8822_REG_RHP_VOLUME, 7, 1, 0),
+        SOC_DOUBLE_R("Headphone Playback Switch",
+                NAU8822_REG_LHP_VOLUME,
+                NAU8822_REG_RHP_VOLUME, 6, 1, 1),
+        SOC_DOUBLE_R_TLV("Headphone Volume",
+                NAU8822_REG_LHP_VOLUME,
+                NAU8822_REG_RHP_VOLUME, 0, 63, 0, spk_tlv),
+
+        SOC_DOUBLE_R("Speaker ZC Switch",
+                NAU8822_REG_LSPKOUT_VOLUME,
+                NAU8822_REG_RSPKOUT_VOLUME, 7, 1, 0),
+        SOC_DOUBLE_R("Speaker Playback Switch",
+                NAU8822_REG_LSPKOUT_VOLUME,
+                NAU8822_REG_RSPKOUT_VOLUME, 6, 1, 1),
+        SOC_DOUBLE_R_TLV("Speaker Volume",
+                NAU8822_REG_LSPKOUT_VOLUME,
+                NAU8822_REG_RSPKOUT_VOLUME, 0, 63, 0, spk_tlv),
+
+        SOC_DOUBLE_R("AUXOUT Playback Switch",
+                NAU8822_REG_AUX2_MIXER,
+                NAU8822_REG_AUX1_MIXER, 6, 1, 1),
+
+        SOC_DOUBLE_R_TLV("PGA Boost Volume",
+                NAU8822_REG_LEFT_ADC_BOOST_CONTROL,
+                NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 8, 1, 0, pga_boost_tlv),
+        SOC_DOUBLE_R_TLV("L2/R2 Boost Volume",
+                NAU8822_REG_LEFT_ADC_BOOST_CONTROL,
+                NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 4, 7, 0, boost_tlv),
+        SOC_DOUBLE_R_TLV("Aux Boost Volume",
+                NAU8822_REG_LEFT_ADC_BOOST_CONTROL,
+                NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0, 7, 0, boost_tlv),
+
+        SOC_SINGLE("DAC 128x Oversampling Switch",
+                NAU8822_REG_DAC_CONTROL, 5, 1, 0),
+        SOC_SINGLE("ADC 128x Oversampling Switch",
+                NAU8822_REG_ADC_CONTROL, 5, 1, 0),
+};
+
+/* LMAIN and RMAIN Mixer */
+static const struct snd_kcontrol_new nau8822_left_out_mixer[] = {
+        SOC_DAPM_SINGLE("LINMIX Switch",
+                NAU8822_REG_LEFT_MIXER_CONTROL, 1, 1, 0),
+        SOC_DAPM_SINGLE("LAUX Switch",
+                NAU8822_REG_LEFT_MIXER_CONTROL, 5, 1, 0),
+        SOC_DAPM_SINGLE("LDAC Switch",
+                NAU8822_REG_LEFT_MIXER_CONTROL, 0, 1, 0),
+        SOC_DAPM_SINGLE("RDAC Switch",
+                NAU8822_REG_OUTPUT_CONTROL, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new nau8822_right_out_mixer[] = {
+        SOC_DAPM_SINGLE("RINMIX Switch",
+                NAU8822_REG_RIGHT_MIXER_CONTROL, 1, 1, 0),
+        SOC_DAPM_SINGLE("RAUX Switch",
+                NAU8822_REG_RIGHT_MIXER_CONTROL, 5, 1, 0),
+        SOC_DAPM_SINGLE("RDAC Switch",
+                NAU8822_REG_RIGHT_MIXER_CONTROL, 0, 1, 0),
+        SOC_DAPM_SINGLE("LDAC Switch",
+                NAU8822_REG_OUTPUT_CONTROL, 6, 1, 0),
+};
+
+/* AUX1 and AUX2 Mixer */
+static const struct snd_kcontrol_new nau8822_auxout1_mixer[] = {
+        SOC_DAPM_SINGLE("RDAC Switch", NAU8822_REG_AUX1_MIXER, 0, 1, 0),
+        SOC_DAPM_SINGLE("RMIX Switch", NAU8822_REG_AUX1_MIXER, 1, 1, 0),
+        SOC_DAPM_SINGLE("RINMIX Switch", NAU8822_REG_AUX1_MIXER, 2, 1, 0),
+        SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX1_MIXER, 3, 1, 0),
+        SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX1_MIXER, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new nau8822_auxout2_mixer[] = {
+        SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX2_MIXER, 0, 1, 0),
+        SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX2_MIXER, 1, 1, 0),
+        SOC_DAPM_SINGLE("LINMIX Switch", NAU8822_REG_AUX2_MIXER, 2, 1, 0),
+        SOC_DAPM_SINGLE("AUX1MIX Output Switch",
+                NAU8822_REG_AUX2_MIXER, 3, 1, 0),
+};
+
+/* Input PGA */
+static const struct snd_kcontrol_new nau8822_left_input_mixer[] = {
+        SOC_DAPM_SINGLE("L2 Switch", NAU8822_REG_INPUT_CONTROL, 2, 1, 0),
+        SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 1, 1, 0),
+        SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 0, 1, 0),
+};
+static const struct snd_kcontrol_new nau8822_right_input_mixer[] = {
+        SOC_DAPM_SINGLE("R2 Switch", NAU8822_REG_INPUT_CONTROL, 6, 1, 0),
+        SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 5, 1, 0),
+        SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 4, 1, 0),
+};
+
+/* Loopback Switch */
+static const struct snd_kcontrol_new nau8822_loopback =
+        SOC_DAPM_SINGLE("Switch", NAU8822_REG_COMPANDING_CONTROL,
+                NAU8822_ADDAP_SFT, 1, 0);
+
+static int check_mclk_select_pll(struct snd_soc_dapm_widget *source,
+                         struct snd_soc_dapm_widget *sink)
+{
+        struct snd_soc_component *component =
+                snd_soc_dapm_to_component(source->dapm);
+        unsigned int value;
+
+        value = snd_soc_component_read32(component, NAU8822_REG_CLOCKING);
+
+        return (value & NAU8822_CLKM_MASK);
+}
+
+static const struct snd_soc_dapm_widget nau8822_dapm_widgets[] = {
+        SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
+                NAU8822_REG_POWER_MANAGEMENT_3, 0, 0),
+        SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
+                NAU8822_REG_POWER_MANAGEMENT_3, 1, 0),
+        SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
+                NAU8822_REG_POWER_MANAGEMENT_2, 0, 0),
+        SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
+                NAU8822_REG_POWER_MANAGEMENT_2, 1, 0),
+
+        SOC_MIXER_ARRAY("Left Output Mixer",
+                NAU8822_REG_POWER_MANAGEMENT_3, 2, 0, nau8822_left_out_mixer),
+        SOC_MIXER_ARRAY("Right Output Mixer",
+                NAU8822_REG_POWER_MANAGEMENT_3, 3, 0, nau8822_right_out_mixer),
+        SOC_MIXER_ARRAY("AUX1 Output Mixer",
+                NAU8822_REG_POWER_MANAGEMENT_1, 7, 0, nau8822_auxout1_mixer),
+        SOC_MIXER_ARRAY("AUX2 Output Mixer",
+                NAU8822_REG_POWER_MANAGEMENT_1, 6, 0, nau8822_auxout2_mixer),
+
+        SOC_MIXER_ARRAY("Left Input Mixer",
+                NAU8822_REG_POWER_MANAGEMENT_2,
+                2, 0, nau8822_left_input_mixer),
+        SOC_MIXER_ARRAY("Right Input Mixer",
+                NAU8822_REG_POWER_MANAGEMENT_2,
+                3, 0, nau8822_right_input_mixer),
+
+        SND_SOC_DAPM_PGA("Left Boost Mixer",
+                NAU8822_REG_POWER_MANAGEMENT_2, 4, 0, NULL, 0),
+        SND_SOC_DAPM_PGA("Right Boost Mixer",
+                NAU8822_REG_POWER_MANAGEMENT_2, 5, 0, NULL, 0),
+
+        SND_SOC_DAPM_PGA("Left Capture PGA",
+                NAU8822_REG_LEFT_INP_PGA_CONTROL, 6, 1, NULL, 0),
+        SND_SOC_DAPM_PGA("Right Capture PGA",
+                NAU8822_REG_RIGHT_INP_PGA_CONTROL, 6, 1, NULL, 0),
+
+        SND_SOC_DAPM_PGA("Left Headphone Out",
+                NAU8822_REG_POWER_MANAGEMENT_2, 7, 0, NULL, 0),
+        SND_SOC_DAPM_PGA("Right Headphone Out",
+                NAU8822_REG_POWER_MANAGEMENT_2, 8, 0, NULL, 0),
+
+        SND_SOC_DAPM_PGA("Left Speaker Out",
+                 NAU8822_REG_POWER_MANAGEMENT_3, 6, 0, NULL, 0),
+        SND_SOC_DAPM_PGA("Right Speaker Out",
+                NAU8822_REG_POWER_MANAGEMENT_3, 5, 0, NULL, 0),
+
+        SND_SOC_DAPM_PGA("AUX1 Out",
+                NAU8822_REG_POWER_MANAGEMENT_3, 8, 0, NULL, 0),
+        SND_SOC_DAPM_PGA("AUX2 Out",
+                NAU8822_REG_POWER_MANAGEMENT_3, 7, 0, NULL, 0),
+
+        SND_SOC_DAPM_SUPPLY("Mic Bias",
+                NAU8822_REG_POWER_MANAGEMENT_1, 4, 0, NULL, 0),
+        SND_SOC_DAPM_SUPPLY("PLL",
+                NAU8822_REG_POWER_MANAGEMENT_1, 5, 0, NULL, 0),
+
+        SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0,
+                &nau8822_loopback),
+
+        SND_SOC_DAPM_INPUT("LMICN"),
+        SND_SOC_DAPM_INPUT("LMICP"),
+        SND_SOC_DAPM_INPUT("RMICN"),
+        SND_SOC_DAPM_INPUT("RMICP"),
+        SND_SOC_DAPM_INPUT("LAUX"),
+        SND_SOC_DAPM_INPUT("RAUX"),
+        SND_SOC_DAPM_INPUT("L2"),
+        SND_SOC_DAPM_INPUT("R2"),
+        SND_SOC_DAPM_OUTPUT("LHP"),
+        SND_SOC_DAPM_OUTPUT("RHP"),
+        SND_SOC_DAPM_OUTPUT("LSPK"),
+        SND_SOC_DAPM_OUTPUT("RSPK"),
+        SND_SOC_DAPM_OUTPUT("AUXOUT1"),
+        SND_SOC_DAPM_OUTPUT("AUXOUT2"),
+};
+
+static const struct snd_soc_dapm_route nau8822_dapm_routes[] = {
+        {"Right DAC", NULL, "PLL", check_mclk_select_pll},
+        {"Left DAC", NULL, "PLL", check_mclk_select_pll},
+
+        /* LMAIN and RMAIN Mixer */
+        {"Right Output Mixer", "LDAC Switch", "Left DAC"},
+        {"Right Output Mixer", "RDAC Switch", "Right DAC"},
+        {"Right Output Mixer", "RAUX Switch", "RAUX"},
+        {"Right Output Mixer", "RINMIX Switch", "Right Boost Mixer"},
+
+        {"Left Output Mixer", "LDAC Switch", "Left DAC"},
+        {"Left Output Mixer", "RDAC Switch", "Right DAC"},
+        {"Left Output Mixer", "LAUX Switch", "LAUX"},
+        {"Left Output Mixer", "LINMIX Switch", "Left Boost Mixer"},
+
+        /* AUX1 and AUX2 Mixer */
+        {"AUX1 Output Mixer", "RDAC Switch", "Right DAC"},
+        {"AUX1 Output Mixer", "RMIX Switch", "Right Output Mixer"},
+        {"AUX1 Output Mixer", "RINMIX Switch", "Right Boost Mixer"},
+        {"AUX1 Output Mixer", "LDAC Switch", "Left DAC"},
+        {"AUX1 Output Mixer", "LMIX Switch", "Left Output Mixer"},
+
+        {"AUX2 Output Mixer", "LDAC Switch", "Left DAC"},
+        {"AUX2 Output Mixer", "LMIX Switch", "Left Output Mixer"},
+        {"AUX2 Output Mixer", "LINMIX Switch", "Left Boost Mixer"},
+        {"AUX2 Output Mixer", "AUX1MIX Output Switch", "AUX1 Output Mixer"},
+
+        /* Outputs */
+        {"Right Headphone Out", NULL, "Right Output Mixer"},
+        {"RHP", NULL, "Right Headphone Out"},
+
+        {"Left Headphone Out", NULL, "Left Output Mixer"},
+        {"LHP", NULL, "Left Headphone Out"},
+
+        {"Right Speaker Out", NULL, "Right Output Mixer"},
+        {"RSPK", NULL, "Right Speaker Out"},
+
+        {"Left Speaker Out", NULL, "Left Output Mixer"},
+        {"LSPK", NULL, "Left Speaker Out"},
+
+        {"AUX1 Out", NULL, "AUX1 Output Mixer"},
+        {"AUX2 Out", NULL, "AUX2 Output Mixer"},
+        {"AUXOUT1", NULL, "AUX1 Out"},
+        {"AUXOUT2", NULL, "AUX2 Out"},
+
+        /* Boost Mixer */
+        {"Right ADC", NULL, "PLL", check_mclk_select_pll},
+        {"Left ADC", NULL, "PLL", check_mclk_select_pll},
+
+        {"Right ADC", NULL, "Right Boost Mixer"},
+
+        {"Right Boost Mixer", NULL, "RAUX"},
+        {"Right Boost Mixer", NULL, "Right Capture PGA"},
+        {"Right Boost Mixer", NULL, "R2"},
+
+        {"Left ADC", NULL, "Left Boost Mixer"},
+
+        {"Left Boost Mixer", NULL, "LAUX"},
+        {"Left Boost Mixer", NULL, "Left Capture PGA"},
+        {"Left Boost Mixer", NULL, "L2"},
+
+        /* Input PGA */
+        {"Right Capture PGA", NULL, "Right Input Mixer"},
+        {"Left Capture PGA", NULL, "Left Input Mixer"},
+
+        /* Enable Microphone Power */
+        {"Right Capture PGA", NULL, "Mic Bias"},
+        {"Left Capture PGA", NULL, "Mic Bias"},
+
+        {"Right Input Mixer", "R2 Switch", "R2"},
+        {"Right Input Mixer", "MicN Switch", "RMICN"},
+        {"Right Input Mixer", "MicP Switch", "RMICP"},
+
+        {"Left Input Mixer", "L2 Switch", "L2"},
+        {"Left Input Mixer", "MicN Switch", "LMICN"},
+        {"Left Input Mixer", "MicP Switch", "LMICP"},
+
+        /* Digital Loopback */
+        {"Digital Loopback", "Switch", "Left ADC"},
+        {"Digital Loopback", "Switch", "Right ADC"},
+        {"Left DAC", NULL, "Digital Loopback"},
+        {"Right DAC", NULL, "Digital Loopback"},
+};
+
+static int nau8822_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+                                 unsigned int freq, int dir)
+{
+        struct snd_soc_component *component = dai->component;
+        struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+
+        nau8822->div_id = clk_id;
+        nau8822->sysclk = freq;
+        dev_dbg(component->dev, "master sysclk %dHz, source %s\n", freq,
+                clk_id == NAU8822_CLK_PLL ? "PLL" : "MCLK");
+
+        return 0;
+}
+
+static int nau8822_calc_pll(unsigned int pll_in, unsigned int fs,
+                                struct nau8822_pll *pll_param)
+{
+        u64 f2, f2_max, pll_ratio;
+        int i, scal_sel;
+
+        if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN)
+                return -EINVAL;
+        f2_max = 0;
+        scal_sel = ARRAY_SIZE(nau8822_mclk_scaler);
+
+        for (i = 0; i < scal_sel; i++) {
+                f2 = 256 * fs * 4 * nau8822_mclk_scaler[i] / 10;
+                if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX &&
+                        f2_max < f2) {
+                        f2_max = f2;
+                        scal_sel = i;
+                }
+        }
+
+        if (ARRAY_SIZE(nau8822_mclk_scaler) == scal_sel)
+                return -EINVAL;
+        pll_param->mclk_scaler = scal_sel;
+        f2 = f2_max;
+
+        /* Calculate the PLL 4-bit integer input and the PLL 24-bit fractional
+         * input; round up the 24+4bit.
+         */
+        pll_ratio = div_u64(f2 << 28, pll_in);
+        pll_param->pre_factor = 0;
+        if (((pll_ratio >> 28) & 0xF) < NAU_PLL_OPTOP_MIN) {
+                pll_ratio <<= 1;
+                pll_param->pre_factor = 1;
+        }
+        pll_param->pll_int = (pll_ratio >> 28) & 0xF;
+        pll_param->pll_frac = ((pll_ratio & 0xFFFFFFF) >> 4);
+
+        return 0;
+}
+
+static int nau8822_config_clkdiv(struct snd_soc_dai *dai, int div, int rate)
+{
+        struct snd_soc_component *component = dai->component;
+        struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+        struct nau8822_pll *pll = &nau8822->pll;
+        int i, sclk, imclk;
+
+        switch (nau8822->div_id) {
+        case NAU8822_CLK_MCLK:
+                /* Configure the master clock prescaler div to make system
+                 * clock to approximate the internal master clock (IMCLK);
+                 * and large or equal to IMCLK.
+                 */
+                div = 0;
+                imclk = rate * 256;
+                for (i = 1; i < ARRAY_SIZE(nau8822_mclk_scaler); i++) {
+                        sclk = (nau8822->sysclk * 10) / nau8822_mclk_scaler[i];
+                        if (sclk < imclk)
+                                break;
+                        div = i;
+                }
+                dev_dbg(component->dev, "master clock prescaler %x for fs %d\n",
+                        div, rate);
+
+                /* master clock from MCLK and disable PLL */
+                snd_soc_component_update_bits(component,
+                        NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK,
+                        (div << NAU8822_MCLKSEL_SFT));
+                snd_soc_component_update_bits(component,
+                        NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK,
+                        NAU8822_CLKM_MCLK);
+                break;
+
+        case NAU8822_CLK_PLL:
+                /* master clock from PLL and enable PLL */
+                if (pll->mclk_scaler != div) {
+                        dev_err(component->dev,
+                        "master clock prescaler not meet PLL parameters\n");
+                        return -EINVAL;
+                }
+                snd_soc_component_update_bits(component,
+                        NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK,
+                        (div << NAU8822_MCLKSEL_SFT));
+                snd_soc_component_update_bits(component,
+                        NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK,
+                        NAU8822_CLKM_PLL);
+                break;
+
+        default:
+                return -EINVAL;
+        }
+
+        return 0;
+}
+
+static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
+                                unsigned int freq_in, unsigned int freq_out)
+{
+        struct snd_soc_component *component = dai->component;
+        struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+        struct nau8822_pll *pll_param = &nau8822->pll;
+        int ret, fs;
+
+        fs = freq_out / 256;
+
+        ret = nau8822_calc_pll(freq_in, fs, pll_param);
+        if (ret < 0) {
+                dev_err(component->dev, "Unsupported input clock %d\n",
+                        freq_in);
+                return ret;
+        }
+
+        dev_info(component->dev,
+                "pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n",
+                pll_param->pll_int, pll_param->pll_frac,
+                pll_param->mclk_scaler, pll_param->pre_factor);
+
+        snd_soc_component_update_bits(component,
+                NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK,
+                (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) |
+                pll_param->pll_int);
+        snd_soc_component_write(component,
+                NAU8822_REG_PLL_K1, (pll_param->pll_frac >> NAU8822_PLLK1_SFT) &
+                NAU8822_PLLK1_MASK);
+        snd_soc_component_write(component,
+                NAU8822_REG_PLL_K2, (pll_param->pll_frac >> NAU8822_PLLK2_SFT) &
+                NAU8822_PLLK2_MASK);
+        snd_soc_component_write(component,
+                NAU8822_REG_PLL_K3, pll_param->pll_frac & NAU8822_PLLK3_MASK);
+        snd_soc_component_update_bits(component,
+                NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK,
+                pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT);
+        snd_soc_component_update_bits(component,
+                NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL);
+
+        return 0;
+}
+
+static int nau8822_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+        struct snd_soc_component *component = dai->component;
+        u16 ctrl1_val = 0, ctrl2_val = 0;
+
+        dev_dbg(component->dev, "%s\n", __func__);
+
+        switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+        case SND_SOC_DAIFMT_CBM_CFM:
+                ctrl2_val |= 1;
+                break;
+        case SND_SOC_DAIFMT_CBS_CFS:
+                ctrl2_val &= ~1;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+        case SND_SOC_DAIFMT_I2S:
+                ctrl1_val |= 0x10;
+                break;
+        case SND_SOC_DAIFMT_RIGHT_J:
+                break;
+        case SND_SOC_DAIFMT_LEFT_J:
+                ctrl1_val |= 0x8;
+                break;
+        case SND_SOC_DAIFMT_DSP_A:
+                ctrl1_val |= 0x18;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+        case SND_SOC_DAIFMT_NB_NF:
+                break;
+        case SND_SOC_DAIFMT_IB_IF:
+                ctrl1_val |= 0x180;
+                break;
+        case SND_SOC_DAIFMT_IB_NF:
+                ctrl1_val |= 0x100;
+                break;
+        case SND_SOC_DAIFMT_NB_IF:
+                ctrl1_val |= 0x80;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        snd_soc_component_update_bits(component,
+                NAU8822_REG_AUDIO_INTERFACE,
+                NAU8822_AIFMT_MASK | NAU8822_LRP_MASK | NAU8822_BCLKP_MASK,
+                ctrl1_val);
+        snd_soc_component_update_bits(component,
+                NAU8822_REG_CLOCKING, NAU8822_CLKIOEN_MASK, ctrl2_val);
+
+        return 0;
+}
+
+static int nau8822_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 nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+        int val_len = 0, val_rate = 0;
+
+        switch (params_format(params)) {
+        case SNDRV_PCM_FORMAT_S16_LE:
+                break;
+        case SNDRV_PCM_FORMAT_S20_3LE:
+                val_len |= NAU8822_WLEN_20;
+                break;
+        case SNDRV_PCM_FORMAT_S24_LE:
+                val_len |= NAU8822_WLEN_24;
+                break;
+        case SNDRV_PCM_FORMAT_S32_LE:
+                val_len |= NAU8822_WLEN_32;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        switch (params_rate(params)) {
+        case 8000:
+                val_rate |= NAU8822_SMPLR_8K;
+                break;
+        case 11025:
+                val_rate |= NAU8822_SMPLR_12K;
+                break;
+        case 16000:
+                val_rate |= NAU8822_SMPLR_16K;
+                break;
+        case 22050:
+                val_rate |= NAU8822_SMPLR_24K;
+                break;
+        case 32000:
+                val_rate |= NAU8822_SMPLR_32K;
+                break;
+        case 44100:
+        case 48000:
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        snd_soc_component_update_bits(component,
+                NAU8822_REG_AUDIO_INTERFACE, NAU8822_WLEN_MASK, val_len);
+        snd_soc_component_update_bits(component,
+                NAU8822_REG_ADDITIONAL_CONTROL, NAU8822_SMPLR_MASK, val_rate);
+
+        /* If the master clock is from MCLK, provide the runtime FS for driver
+         * to get the master clock prescaler configuration.
+         */
+        if (nau8822->div_id == NAU8822_CLK_MCLK)
+                nau8822_config_clkdiv(dai, 0, params_rate(params));
+
+        return 0;
+}
+
+static int nau8822_mute(struct snd_soc_dai *dai, int mute)
+{
+        struct snd_soc_component *component = dai->component;
+
+        dev_dbg(component->dev, "%s: %d\n", __func__, mute);
+
+        if (mute)
+                snd_soc_component_update_bits(component,
+                        NAU8822_REG_DAC_CONTROL, 0x40, 0x40);
+        else
+                snd_soc_component_update_bits(component,
+                        NAU8822_REG_DAC_CONTROL, 0x40, 0);
+
+        return 0;
+}
+
+static int nau8822_set_bias_level(struct snd_soc_component *component,
+                                 enum snd_soc_bias_level level)
+{
+        switch (level) {
+        case SND_SOC_BIAS_ON:
+        case SND_SOC_BIAS_PREPARE:
+                snd_soc_component_update_bits(component,
+                        NAU8822_REG_POWER_MANAGEMENT_1,
+                        NAU8822_REFIMP_MASK, NAU8822_REFIMP_80K);
+                break;
+
+        case SND_SOC_BIAS_STANDBY:
+                snd_soc_component_update_bits(component,
+                        NAU8822_REG_POWER_MANAGEMENT_1,
+                        NAU8822_IOBUF_EN | NAU8822_ABIAS_EN,
+                        NAU8822_IOBUF_EN | NAU8822_ABIAS_EN);
+
+                if (snd_soc_component_get_bias_level(component) ==
+                        SND_SOC_BIAS_OFF) {
+                        snd_soc_component_update_bits(component,
+                                NAU8822_REG_POWER_MANAGEMENT_1,
+                                NAU8822_REFIMP_MASK, NAU8822_REFIMP_3K);
+                        mdelay(100);
+                }
+                snd_soc_component_update_bits(component,
+                        NAU8822_REG_POWER_MANAGEMENT_1,
+                        NAU8822_REFIMP_MASK, NAU8822_REFIMP_300K);
+                break;
+
+        case SND_SOC_BIAS_OFF:
+                snd_soc_component_write(component,
+                        NAU8822_REG_POWER_MANAGEMENT_1, 0);
+                snd_soc_component_write(component,
+                        NAU8822_REG_POWER_MANAGEMENT_2, 0);
+                snd_soc_component_write(component,
+                        NAU8822_REG_POWER_MANAGEMENT_3, 0);
+                break;
+        }
+
+        dev_dbg(component->dev, "%s: %d\n", __func__, level);
+
+        return 0;
+}
+
+#define NAU8822_RATES (SNDRV_PCM_RATE_8000_48000)
+
+#define NAU8822_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+        SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static const struct snd_soc_dai_ops nau8822_dai_ops = {
+        .hw_params      = nau8822_hw_params,
+        .digital_mute   = nau8822_mute,
+        .set_fmt        = nau8822_set_dai_fmt,
+        .set_sysclk     = nau8822_set_dai_sysclk,
+        .set_pll        = nau8822_set_pll,
+};
+
+static struct snd_soc_dai_driver nau8822_dai = {
+        .name = "nau8822-hifi",
+        .playback = {
+                .stream_name = "Playback",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = NAU8822_RATES,
+                .formats = NAU8822_FORMATS,
+        },
+        .capture = {
+                .stream_name = "Capture",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = NAU8822_RATES,
+                .formats = NAU8822_FORMATS,
+        },
+        .ops = &nau8822_dai_ops,
+        .symmetric_rates = 1,
+};
+
+static int nau8822_suspend(struct snd_soc_component *component)
+{
+        struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+
+        snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF);
+
+        regcache_mark_dirty(nau8822->regmap);
+
+        return 0;
+}
+
+static int nau8822_resume(struct snd_soc_component *component)
+{
+        struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+
+        regcache_sync(nau8822->regmap);
+
+        snd_soc_component_force_bias_level(component, SND_SOC_BIAS_STANDBY);
+
+        return 0;
+}
+
+/*
+ * These registers contain an "update" bit - bit 8. This means, for example,
+ * that one can write new DAC digital volume for both channels, but only when
+ * the update bit is set, will also the volume be updated - simultaneously for
+ * both channels.
+ */
+static const int update_reg[] = {
+        NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME,
+        NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME,
+        NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME,
+        NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME,
+        NAU8822_REG_LEFT_INP_PGA_CONTROL,
+        NAU8822_REG_RIGHT_INP_PGA_CONTROL,
+        NAU8822_REG_LHP_VOLUME,
+        NAU8822_REG_RHP_VOLUME,
+        NAU8822_REG_LSPKOUT_VOLUME,
+        NAU8822_REG_RSPKOUT_VOLUME,
+};
+
+static int nau8822_probe(struct snd_soc_component *component)
+{
+        int i;
+
+        /*
+         * Set the update bit in all registers, that have one. This way all
+         * writes to those registers will also cause the update bit to be
+         * written.
+         */
+        for (i = 0; i < ARRAY_SIZE(update_reg); i++)
+                snd_soc_component_update_bits(component,
+                        update_reg[i], 0x100, 0x100);
+
+        return 0;
+}
+
+static const struct snd_soc_component_driver soc_component_dev_nau8822 = {
+        .probe                          = nau8822_probe,
+        .suspend                        = nau8822_suspend,
+        .resume                         = nau8822_resume,
+        .set_bias_level                 = nau8822_set_bias_level,
+        .controls                       = nau8822_snd_controls,
+        .num_controls                   = ARRAY_SIZE(nau8822_snd_controls),
+        .dapm_widgets                   = nau8822_dapm_widgets,
+        .num_dapm_widgets               = ARRAY_SIZE(nau8822_dapm_widgets),
+        .dapm_routes                    = nau8822_dapm_routes,
+        .num_dapm_routes                = ARRAY_SIZE(nau8822_dapm_routes),
+        .idle_bias_on                   = 1,
+        .use_pmdown_time                = 1,
+        .endianness                     = 1,
+        .non_legacy_dai_naming          = 1,
+};
+
+static const struct regmap_config nau8822_regmap_config = {
+        .reg_bits = 7,
+        .val_bits = 9,
+
+        .max_register = NAU8822_REG_MAX_REGISTER,
+        .volatile_reg = nau8822_volatile,
+
+        .readable_reg = nau8822_readable_reg,
+        .writeable_reg = nau8822_writeable_reg,
+
+        .cache_type = REGCACHE_RBTREE,
+        .reg_defaults = nau8822_reg_defaults,
+        .num_reg_defaults = ARRAY_SIZE(nau8822_reg_defaults),
+};
+
+static int nau8822_i2c_probe(struct i2c_client *i2c,
+                            const struct i2c_device_id *id)
+{
+        struct device *dev = &i2c->dev;
+        struct nau8822 *nau8822 = dev_get_platdata(dev);
+        int ret;
+
+        if (!nau8822) {
+                nau8822 = devm_kzalloc(dev, sizeof(*nau8822), GFP_KERNEL);
+                if (nau8822 == NULL)
+                        return -ENOMEM;
+        }
+        i2c_set_clientdata(i2c, nau8822);
+
+        nau8822->regmap = devm_regmap_init_i2c(i2c, &nau8822_regmap_config);
+        if (IS_ERR(nau8822->regmap)) {
+                ret = PTR_ERR(nau8822->regmap);
+                dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
+                return ret;
+        }
+        nau8822->dev = dev;
+
+        /* Reset the codec */
+        ret = regmap_write(nau8822->regmap, NAU8822_REG_RESET, 0x00);
+        if (ret != 0) {
+                dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
+                return ret;
+        }
+
+        ret = devm_snd_soc_register_component(dev, &soc_component_dev_nau8822,
+                                                &nau8822_dai, 1);
+        if (ret != 0) {
+                dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
+                return ret;
+        }
+
+        return 0;
+}
+
+static const struct i2c_device_id nau8822_i2c_id[] = {
+        { "nau8822", 0 },
+        { }
+};
+MODULE_DEVICE_TABLE(i2c, nau8822_i2c_id);
+
+#ifdef CONFIG_OF
+static const struct of_device_id nau8822_of_match[] = {
+        { .compatible = "nuvoton,nau8822", },
+        { }
+};
+MODULE_DEVICE_TABLE(of, nau8822_of_match);
+#endif
+
+static struct i2c_driver nau8822_i2c_driver = {
+        .driver = {
+                .name = "nau8822",
+                .of_match_table = of_match_ptr(nau8822_of_match),
+        },
+        .probe =    nau8822_i2c_probe,
+        .id_table = nau8822_i2c_id,
+};
+module_i2c_driver(nau8822_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC NAU8822 codec driver");
+MODULE_AUTHOR("David Lin <ctlin0@nuvoton.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h
new file mode 100644
index 000000000000..aa79c969cd44
--- /dev/null
+++ b/sound/soc/codecs/nau8822.h
@@ -0,0 +1,204 @@
+/*
+ * nau8822.h  --  NAU8822 Soc Audio Codec driver
+ *
+ * Author: David Lin <ctlin0@nuvoton.com>
+ * Co-author: John Hsu <kchsu0@nuvoton.com>
+ * Co-author: Seven Li <wtli@nuvoton.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 __NAU8822_H__
+#define __NAU8822_H__
+
+#define NAU8822_REG_RESET                       0x00
+#define NAU8822_REG_POWER_MANAGEMENT_1          0x01
+#define NAU8822_REG_POWER_MANAGEMENT_2          0x02
+#define NAU8822_REG_POWER_MANAGEMENT_3          0x03
+#define NAU8822_REG_AUDIO_INTERFACE             0x04
+#define NAU8822_REG_COMPANDING_CONTROL          0x05
+#define NAU8822_REG_CLOCKING                    0x06
+#define NAU8822_REG_ADDITIONAL_CONTROL          0x07
+#define NAU8822_REG_GPIO_CONTROL                0x08
+#define NAU8822_REG_JACK_DETECT_CONTROL_1       0x09
+#define NAU8822_REG_DAC_CONTROL                 0x0A
+#define NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME     0x0B
+#define NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME    0x0C
+#define NAU8822_REG_JACK_DETECT_CONTROL_2       0x0D
+#define NAU8822_REG_ADC_CONTROL                 0x0E
+#define NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME     0x0F
+#define NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME    0x10
+#define NAU8822_REG_EQ1                         0x12
+#define NAU8822_REG_EQ2                         0x13
+#define NAU8822_REG_EQ3                         0x14
+#define NAU8822_REG_EQ4                         0x15
+#define NAU8822_REG_EQ5                         0x16
+#define NAU8822_REG_DAC_LIMITER_1               0x18
+#define NAU8822_REG_DAC_LIMITER_2               0x19
+#define NAU8822_REG_NOTCH_FILTER_1              0x1B
+#define NAU8822_REG_NOTCH_FILTER_2              0x1C
+#define NAU8822_REG_NOTCH_FILTER_3              0x1D
+#define NAU8822_REG_NOTCH_FILTER_4              0x1E
+#define NAU8822_REG_ALC_CONTROL_1               0x20
+#define NAU8822_REG_ALC_CONTROL_2               0x21
+#define NAU8822_REG_ALC_CONTROL_3               0x22
+#define NAU8822_REG_NOISE_GATE                  0x23
+#define NAU8822_REG_PLL_N                       0x24
+#define NAU8822_REG_PLL_K1                      0x25
+#define NAU8822_REG_PLL_K2                      0x26
+#define NAU8822_REG_PLL_K3                      0x27
+#define NAU8822_REG_3D_CONTROL                  0x29
+#define NAU8822_REG_RIGHT_SPEAKER_CONTROL       0x2B
+#define NAU8822_REG_INPUT_CONTROL               0x2C
+#define NAU8822_REG_LEFT_INP_PGA_CONTROL        0x2D
+#define NAU8822_REG_RIGHT_INP_PGA_CONTROL       0x2E
+#define NAU8822_REG_LEFT_ADC_BOOST_CONTROL      0x2F
+#define NAU8822_REG_RIGHT_ADC_BOOST_CONTROL     0x30
+#define NAU8822_REG_OUTPUT_CONTROL              0x31
+#define NAU8822_REG_LEFT_MIXER_CONTROL          0x32
+#define NAU8822_REG_RIGHT_MIXER_CONTROL         0x33
+#define NAU8822_REG_LHP_VOLUME                  0x34
+#define NAU8822_REG_RHP_VOLUME                  0x35
+#define NAU8822_REG_LSPKOUT_VOLUME              0x36
+#define NAU8822_REG_RSPKOUT_VOLUME              0x37
+#define NAU8822_REG_AUX2_MIXER                  0x38
+#define NAU8822_REG_AUX1_MIXER                  0x39
+#define NAU8822_REG_POWER_MANAGEMENT_4          0x3A
+#define NAU8822_REG_LEFT_TIME_SLOT              0x3B
+#define NAU8822_REG_MISC                        0x3C
+#define NAU8822_REG_RIGHT_TIME_SLOT             0x3D
+#define NAU8822_REG_DEVICE_REVISION             0x3E
+#define NAU8822_REG_DEVICE_ID                   0x3F
+#define NAU8822_REG_DAC_DITHER                  0x41
+#define NAU8822_REG_ALC_ENHANCE_1               0x46
+#define NAU8822_REG_ALC_ENHANCE_2               0x47
+#define NAU8822_REG_192KHZ_SAMPLING             0x48
+#define NAU8822_REG_MISC_CONTROL                0x49
+#define NAU8822_REG_INPUT_TIEOFF                0x4A
+#define NAU8822_REG_POWER_REDUCTION             0x4B
+#define NAU8822_REG_AGC_PEAK2PEAK               0x4C
+#define NAU8822_REG_AGC_PEAK_DETECT             0x4D
+#define NAU8822_REG_AUTOMUTE_CONTROL            0x4E
+#define NAU8822_REG_OUTPUT_TIEOFF               0x4F
+#define NAU8822_REG_MAX_REGISTER                NAU8822_REG_OUTPUT_TIEOFF
+
+/* NAU8822_REG_POWER_MANAGEMENT_1 (0x1) */
+#define NAU8822_REFIMP_MASK                     0x3
+#define NAU8822_REFIMP_80K                      0x1
+#define NAU8822_REFIMP_300K                     0x2
+#define NAU8822_REFIMP_3K                       0x3
+#define NAU8822_IOBUF_EN                        (0x1 << 2)
+#define NAU8822_ABIAS_EN                        (0x1 << 3)
+
+/* NAU8822_REG_AUDIO_INTERFACE (0x4) */
+#define NAU8822_AIFMT_MASK                      (0x3 << 3)
+#define NAU8822_WLEN_MASK                       (0x3 << 5)
+#define NAU8822_WLEN_20                         (0x1 << 5)
+#define NAU8822_WLEN_24                         (0x2 << 5)
+#define NAU8822_WLEN_32                         (0x3 << 5)
+#define NAU8822_LRP_MASK                        (0x1 << 7)
+#define NAU8822_BCLKP_MASK                      (0x1 << 8)
+
+/* NAU8822_REG_COMPANDING_CONTROL (0x5) */
+#define NAU8822_ADDAP_SFT                       0
+#define NAU8822_ADCCM_SFT                       1
+#define NAU8822_DACCM_SFT                       3
+
+/* NAU8822_REG_CLOCKING (0x6) */
+#define NAU8822_CLKIOEN_MASK                    0x1
+#define NAU8822_MCLKSEL_SFT                     5
+#define NAU8822_MCLKSEL_MASK                    (0x7 << 5)
+#define NAU8822_BCLKSEL_SFT                     2
+#define NAU8822_BCLKSEL_MASK                    (0x7 << 2)
+#define NAU8822_CLKM_MASK                       (0x1 << 8)
+#define NAU8822_CLKM_MCLK                       (0x0 << 8)
+#define NAU8822_CLKM_PLL                        (0x1 << 8)
+
+/* NAU8822_REG_ADDITIONAL_CONTROL (0x08) */
+#define NAU8822_SMPLR_SFT                       1
+#define NAU8822_SMPLR_MASK                      (0x7 << 1)
+#define NAU8822_SMPLR_48K                       (0x0 << 1)
+#define NAU8822_SMPLR_32K                       (0x1 << 1)
+#define NAU8822_SMPLR_24K                       (0x2 << 1)
+#define NAU8822_SMPLR_16K                       (0x3 << 1)
+#define NAU8822_SMPLR_12K                       (0x4 << 1)
+#define NAU8822_SMPLR_8K                        (0x5 << 1)
+
+/* NAU8822_REG_EQ1 (0x12) */
+#define NAU8822_EQ1GC_SFT                       0
+#define NAU8822_EQ1CF_SFT                       5
+#define NAU8822_EQM_SFT                         8
+
+/* NAU8822_REG_EQ2 (0x13) */
+#define NAU8822_EQ2GC_SFT                       0
+#define NAU8822_EQ2CF_SFT                       5
+#define NAU8822_EQ2BW_SFT                       8
+
+/* NAU8822_REG_EQ3 (0x14) */
+#define NAU8822_EQ3GC_SFT                       0
+#define NAU8822_EQ3CF_SFT                       5
+#define NAU8822_EQ3BW_SFT                       8
+
+/* NAU8822_REG_EQ4 (0x15) */
+#define NAU8822_EQ4GC_SFT                       0
+#define NAU8822_EQ4CF_SFT                       5
+#define NAU8822_EQ4BW_SFT                       8
+
+/* NAU8822_REG_EQ5 (0x16) */
+#define NAU8822_EQ5GC_SFT                       0
+#define NAU8822_EQ5CF_SFT                       5
+
+/* NAU8822_REG_ALC_CONTROL_1 (0x20) */
+#define NAU8822_ALCMINGAIN_SFT                  0
+#define NAU8822_ALCMXGAIN_SFT                   3
+#define NAU8822_ALCEN_SFT                       7
+
+/* NAU8822_REG_ALC_CONTROL_2 (0x21) */
+#define NAU8822_ALCSL_SFT                       0
+#define NAU8822_ALCHT_SFT                       4
+
+/* NAU8822_REG_ALC_CONTROL_3 (0x22) */
+#define NAU8822_ALCATK_SFT                      0
+#define NAU8822_ALCDCY_SFT                      4
+#define NAU8822_ALCM_SFT                        8
+
+/* NAU8822_REG_PLL_N (0x24) */
+#define NAU8822_PLLMCLK_DIV2                    (0x1 << 4)
+#define NAU8822_PLLN_MASK                       0xF
+
+#define NAU8822_PLLK1_SFT                       18
+#define NAU8822_PLLK1_MASK                      0x3F
+
+/* NAU8822_REG_PLL_K2 (0x26) */
+#define NAU8822_PLLK2_SFT                       9
+#define NAU8822_PLLK2_MASK                      0x1FF
+
+/* NAU8822_REG_PLL_K3 (0x27) */
+#define NAU8822_PLLK3_MASK                      0x1FF
+
+/* System Clock Source */
+enum {
+        NAU8822_CLK_MCLK,
+        NAU8822_CLK_PLL,
+};
+
+struct nau8822_pll {
+        int pre_factor;
+        int mclk_scaler;
+        int pll_frac;
+        int pll_int;
+};
+
+/* Codec Private Data */
+struct nau8822 {
+        struct device *dev;
+        struct regmap *regmap;
+        int mclk_idx;
+        struct nau8822_pll pll;
+        int sysclk;
+        int div_id;
+};
+
+#endif  /* __NAU8822_H__ */
diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c
index 690c26e7389e..809b7e9f03ca 100644
--- a/sound/soc/codecs/pcm186x.c
+++ b/sound/soc/codecs/pcm186x.c
@@ -401,7 +401,8 @@ static int pcm186x_set_fmt(struct snd_soc_dai *dai, unsigned int format)
                 break;
         case SND_SOC_DAIFMT_DSP_A:
                 priv->tdm_offset += 1;
-                /* Fall through... DSP_A uses the same basic config as DSP_B
+                /* fall through */
+                /* DSP_A uses the same basic config as DSP_B
                  * except we need to shift the TDM output by one BCK cycle
                  */
         case SND_SOC_DAIFMT_DSP_B:
diff --git a/sound/soc/codecs/pcm3060-i2c.c b/sound/soc/codecs/pcm3060-i2c.c
new file mode 100644
index 000000000000..cdc8314882bc
--- /dev/null
+++ b/sound/soc/codecs/pcm3060-i2c.c
@@ -0,0 +1,60 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 I2C driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <sound/soc.h>
+
+#include "pcm3060.h"
+
+static int pcm3060_i2c_probe(struct i2c_client *i2c,
+                             const struct i2c_device_id *id)
+{
+        struct pcm3060_priv *priv;
+
+        priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+        if (!priv)
+                return -ENOMEM;
+
+        i2c_set_clientdata(i2c, priv);
+
+        priv->regmap = devm_regmap_init_i2c(i2c, &pcm3060_regmap);
+        if (IS_ERR(priv->regmap))
+                return PTR_ERR(priv->regmap);
+
+        return pcm3060_probe(&i2c->dev);
+}
+
+static const struct i2c_device_id pcm3060_i2c_id[] = {
+        { .name = "pcm3060" },
+        { },
+};
+MODULE_DEVICE_TABLE(i2c, pcm3060_i2c_id);
+
+#ifdef CONFIG_OF
+static const struct of_device_id pcm3060_of_match[] = {
+        { .compatible = "ti,pcm3060" },
+        { },
+};
+MODULE_DEVICE_TABLE(of, pcm3060_of_match);
+#endif /* CONFIG_OF */
+
+static struct i2c_driver pcm3060_i2c_driver = {
+        .driver = {
+                .name = "pcm3060",
+#ifdef CONFIG_OF
+                .of_match_table = pcm3060_of_match,
+#endif /* CONFIG_OF */
+        },
+        .id_table = pcm3060_i2c_id,
+        .probe = pcm3060_i2c_probe,
+};
+
+module_i2c_driver(pcm3060_i2c_driver);
+
+MODULE_DESCRIPTION("PCM3060 I2C driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060-spi.c b/sound/soc/codecs/pcm3060-spi.c
new file mode 100644
index 000000000000..f6f19fa80932
--- /dev/null
+++ b/sound/soc/codecs/pcm3060-spi.c
@@ -0,0 +1,59 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 SPI driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+
+#include "pcm3060.h"
+
+static int pcm3060_spi_probe(struct spi_device *spi)
+{
+        struct pcm3060_priv *priv;
+
+        priv = devm_kzalloc(&spi->dev, sizeof(*priv), GFP_KERNEL);
+        if (!priv)
+                return -ENOMEM;
+
+        spi_set_drvdata(spi, priv);
+
+        priv->regmap = devm_regmap_init_spi(spi, &pcm3060_regmap);
+        if (IS_ERR(priv->regmap))
+                return PTR_ERR(priv->regmap);
+
+        return pcm3060_probe(&spi->dev);
+}
+
+static const struct spi_device_id pcm3060_spi_id[] = {
+        { .name = "pcm3060" },
+        { },
+};
+MODULE_DEVICE_TABLE(spi, pcm3060_spi_id);
+
+#ifdef CONFIG_OF
+static const struct of_device_id pcm3060_of_match[] = {
+        { .compatible = "ti,pcm3060" },
+        { },
+};
+MODULE_DEVICE_TABLE(of, pcm3060_of_match);
+#endif /* CONFIG_OF */
+
+static struct spi_driver pcm3060_spi_driver = {
+        .driver = {
+                .name = "pcm3060",
+#ifdef CONFIG_OF
+                .of_match_table = pcm3060_of_match,
+#endif /* CONFIG_OF */
+        },
+        .id_table = pcm3060_spi_id,
+        .probe = pcm3060_spi_probe,
+};
+
+module_spi_driver(pcm3060_spi_driver);
+
+MODULE_DESCRIPTION("PCM3060 SPI driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c
new file mode 100644
index 000000000000..494d9d662be8
--- /dev/null
+++ b/sound/soc/codecs/pcm3060.c
@@ -0,0 +1,295 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 codec driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/module.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "pcm3060.h"
+
+/* dai */
+
+static int pcm3060_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+                              unsigned int freq, int dir)
+{
+        struct snd_soc_component *comp = dai->component;
+        struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+
+        if (dir != SND_SOC_CLOCK_IN) {
+                dev_err(comp->dev, "unsupported sysclock dir: %d\n", dir);
+                return -EINVAL;
+        }
+
+        priv->dai[dai->id].sclk_freq = freq;
+
+        return 0;
+}
+
+static int pcm3060_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+        struct snd_soc_component *comp = dai->component;
+        struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+        unsigned int reg;
+        unsigned int val;
+
+        if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
+                dev_err(comp->dev, "unsupported DAI polarity: 0x%x\n", fmt);
+                return -EINVAL;
+        }
+
+        switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+        case SND_SOC_DAIFMT_CBM_CFM:
+                priv->dai[dai->id].is_master = true;
+                break;
+        case SND_SOC_DAIFMT_CBS_CFS:
+                priv->dai[dai->id].is_master = false;
+                break;
+        default:
+                dev_err(comp->dev, "unsupported DAI master mode: 0x%x\n", fmt);
+                return -EINVAL;
+        }
+
+        switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+        case SND_SOC_DAIFMT_I2S:
+                val = PCM3060_REG_FMT_I2S;
+                break;
+        case SND_SOC_DAIFMT_RIGHT_J:
+                val = PCM3060_REG_FMT_RJ;
+                break;
+        case SND_SOC_DAIFMT_LEFT_J:
+                val = PCM3060_REG_FMT_LJ;
+                break;
+        default:
+                dev_err(comp->dev, "unsupported DAI format: 0x%x\n", fmt);
+                return -EINVAL;
+        }
+
+        if (dai->id == PCM3060_DAI_ID_DAC)
+                reg = PCM3060_REG67;
+        else
+                reg = PCM3060_REG72;
+
+        regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_FMT, val);
+
+        return 0;
+}
+
+static int pcm3060_hw_params(struct snd_pcm_substream *substream,
+                             struct snd_pcm_hw_params *params,
+                             struct snd_soc_dai *dai)
+{
+        struct snd_soc_component *comp = dai->component;
+        struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+        unsigned int rate;
+        unsigned int ratio;
+        unsigned int reg;
+        unsigned int val;
+
+        if (!priv->dai[dai->id].is_master) {
+                val = PCM3060_REG_MS_S;
+                goto val_ready;
+        }
+
+        rate = params_rate(params);
+        if (!rate) {
+                dev_err(comp->dev, "rate is not configured\n");
+                return -EINVAL;
+        }
+
+        ratio = priv->dai[dai->id].sclk_freq / rate;
+
+        switch (ratio) {
+        case 768:
+                val = PCM3060_REG_MS_M768;
+                break;
+        case 512:
+                val = PCM3060_REG_MS_M512;
+                break;
+        case 384:
+                val = PCM3060_REG_MS_M384;
+                break;
+        case 256:
+                val = PCM3060_REG_MS_M256;
+                break;
+        case 192:
+                val = PCM3060_REG_MS_M192;
+                break;
+        case 128:
+                val = PCM3060_REG_MS_M128;
+                break;
+        default:
+                dev_err(comp->dev, "unsupported ratio: %d\n", ratio);
+                return -EINVAL;
+        }
+
+val_ready:
+        if (dai->id == PCM3060_DAI_ID_DAC)
+                reg = PCM3060_REG67;
+        else
+                reg = PCM3060_REG72;
+
+        regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_MS, val);
+
+        return 0;
+}
+
+static const struct snd_soc_dai_ops pcm3060_dai_ops = {
+        .set_sysclk = pcm3060_set_sysclk,
+        .set_fmt = pcm3060_set_fmt,
+        .hw_params = pcm3060_hw_params,
+};
+
+#define PCM3060_DAI_RATES_ADC   (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 | \
+                                 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \
+                                 SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define PCM3060_DAI_RATES_DAC   (PCM3060_DAI_RATES_ADC | \
+                                 SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
+static struct snd_soc_dai_driver pcm3060_dai[] = {
+        {
+                .name = "pcm3060-dac",
+                .id = PCM3060_DAI_ID_DAC,
+                .playback = {
+                        .stream_name = "Playback",
+                        .channels_min = 2,
+                        .channels_max = 2,
+                        .rates = PCM3060_DAI_RATES_DAC,
+                        .formats = SNDRV_PCM_FMTBIT_S24_LE,
+                },
+                .ops = &pcm3060_dai_ops,
+        },
+        {
+                .name = "pcm3060-adc",
+                .id = PCM3060_DAI_ID_ADC,
+                .capture = {
+                        .stream_name = "Capture",
+                        .channels_min = 2,
+                        .channels_max = 2,
+                        .rates = PCM3060_DAI_RATES_ADC,
+                        .formats = SNDRV_PCM_FMTBIT_S24_LE,
+                },
+                .ops = &pcm3060_dai_ops,
+        },
+};
+
+/* dapm */
+
+static DECLARE_TLV_DB_SCALE(pcm3060_dapm_tlv, -10050, 50, 1);
+
+static const struct snd_kcontrol_new pcm3060_dapm_controls[] = {
+        SOC_DOUBLE_R_RANGE_TLV("Master Playback Volume",
+                               PCM3060_REG65, PCM3060_REG66, 0,
+                               PCM3060_REG_AT2_MIN, PCM3060_REG_AT2_MAX,
+                               0, pcm3060_dapm_tlv),
+        SOC_DOUBLE("Master Playback Switch", PCM3060_REG68,
+                   PCM3060_REG_SHIFT_MUT21, PCM3060_REG_SHIFT_MUT22, 1, 1),
+
+        SOC_DOUBLE_R_RANGE_TLV("Master Capture Volume",
+                               PCM3060_REG70, PCM3060_REG71, 0,
+                               PCM3060_REG_AT1_MIN, PCM3060_REG_AT1_MAX,
+                               0, pcm3060_dapm_tlv),
+        SOC_DOUBLE("Master Capture Switch", PCM3060_REG73,
+                   PCM3060_REG_SHIFT_MUT11, PCM3060_REG_SHIFT_MUT12, 1, 1),
+};
+
+static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = {
+        SND_SOC_DAPM_OUTPUT("OUTL+"),
+        SND_SOC_DAPM_OUTPUT("OUTR+"),
+        SND_SOC_DAPM_OUTPUT("OUTL-"),
+        SND_SOC_DAPM_OUTPUT("OUTR-"),
+
+        SND_SOC_DAPM_INPUT("INL"),
+        SND_SOC_DAPM_INPUT("INR"),
+};
+
+static const struct snd_soc_dapm_route pcm3060_dapm_map[] = {
+        { "OUTL+", NULL, "Playback" },
+        { "OUTR+", NULL, "Playback" },
+        { "OUTL-", NULL, "Playback" },
+        { "OUTR-", NULL, "Playback" },
+
+        { "Capture", NULL, "INL" },
+        { "Capture", NULL, "INR" },
+};
+
+/* soc component */
+
+static const struct snd_soc_component_driver pcm3060_soc_comp_driver = {
+        .controls = pcm3060_dapm_controls,
+        .num_controls = ARRAY_SIZE(pcm3060_dapm_controls),
+        .dapm_widgets = pcm3060_dapm_widgets,
+        .num_dapm_widgets = ARRAY_SIZE(pcm3060_dapm_widgets),
+        .dapm_routes = pcm3060_dapm_map,
+        .num_dapm_routes = ARRAY_SIZE(pcm3060_dapm_map),
+};
+
+/* regmap */
+
+static bool pcm3060_reg_writeable(struct device *dev, unsigned int reg)
+{
+        return (reg >= PCM3060_REG64);
+}
+
+static bool pcm3060_reg_readable(struct device *dev, unsigned int reg)
+{
+        return (reg >= PCM3060_REG64);
+}
+
+static bool pcm3060_reg_volatile(struct device *dev, unsigned int reg)
+{
+        /* PCM3060_REG64 is volatile */
+        return (reg == PCM3060_REG64);
+}
+
+static const struct reg_default pcm3060_reg_defaults[] = {
+        { PCM3060_REG64,  0xF0 },
+        { PCM3060_REG65,  0xFF },
+        { PCM3060_REG66,  0xFF },
+        { PCM3060_REG67,  0x00 },
+        { PCM3060_REG68,  0x00 },
+        { PCM3060_REG69,  0x00 },
+        { PCM3060_REG70,  0xD7 },
+        { PCM3060_REG71,  0xD7 },
+        { PCM3060_REG72,  0x00 },
+        { PCM3060_REG73,  0x00 },
+};
+
+const struct regmap_config pcm3060_regmap = {
+        .reg_bits = 8,
+        .val_bits = 8,
+        .writeable_reg = pcm3060_reg_writeable,
+        .readable_reg = pcm3060_reg_readable,
+        .volatile_reg = pcm3060_reg_volatile,
+        .max_register = PCM3060_REG73,
+        .reg_defaults = pcm3060_reg_defaults,
+        .num_reg_defaults = ARRAY_SIZE(pcm3060_reg_defaults),
+        .cache_type = REGCACHE_RBTREE,
+};
+EXPORT_SYMBOL(pcm3060_regmap);
+
+/* device */
+
+int pcm3060_probe(struct device *dev)
+{
+        int rc;
+
+        rc = devm_snd_soc_register_component(dev, &pcm3060_soc_comp_driver,
+                                             pcm3060_dai,
+                                             ARRAY_SIZE(pcm3060_dai));
+        if (rc) {
+                dev_err(dev, "failed to register component, rc=%d\n", rc);
+                return rc;
+        }
+
+        return 0;
+}
+EXPORT_SYMBOL(pcm3060_probe);
+
+MODULE_DESCRIPTION("PCM3060 codec driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h
new file mode 100644
index 000000000000..fd89a68aa8a7
--- /dev/null
+++ b/sound/soc/codecs/pcm3060.h
@@ -0,0 +1,88 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * PCM3060 codec driver
+ *
+ * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+ */
+
+#ifndef _SND_SOC_PCM3060_H
+#define _SND_SOC_PCM3060_H
+
+#include <linux/device.h>
+#include <linux/regmap.h>
+
+extern const struct regmap_config pcm3060_regmap;
+
+#define PCM3060_DAI_ID_DAC      0
+#define PCM3060_DAI_ID_ADC      1
+#define PCM3060_DAI_IDS_NUM     2
+
+struct pcm3060_priv_dai {
+        bool is_master;
+        unsigned int sclk_freq;
+};
+
+struct pcm3060_priv {
+        struct regmap *regmap;
+        struct pcm3060_priv_dai dai[PCM3060_DAI_IDS_NUM];
+};
+
+int pcm3060_probe(struct device *dev);
+int pcm3060_remove(struct device *dev);
+
+/* registers */
+
+#define PCM3060_REG64                   0x40
+#define PCM3060_REG_MRST                0x80
+#define PCM3060_REG_SRST                0x40
+#define PCM3060_REG_ADPSV               0x20
+#define PCM3060_REG_DAPSV               0x10
+#define PCM3060_REG_SE                  0x01
+
+#define PCM3060_REG65                   0x41
+#define PCM3060_REG66                   0x42
+#define PCM3060_REG_AT2_MIN             0x36
+#define PCM3060_REG_AT2_MAX             0xFF
+
+#define PCM3060_REG67                   0x43
+#define PCM3060_REG72                   0x48
+#define PCM3060_REG_CSEL                0x80
+#define PCM3060_REG_MASK_MS             0x70
+#define PCM3060_REG_MS_S                0x00
+#define PCM3060_REG_MS_M768             (0x01 << 4)
+#define PCM3060_REG_MS_M512             (0x02 << 4)
+#define PCM3060_REG_MS_M384             (0x03 << 4)
+#define PCM3060_REG_MS_M256             (0x04 << 4)
+#define PCM3060_REG_MS_M192             (0x05 << 4)
+#define PCM3060_REG_MS_M128             (0x06 << 4)
+#define PCM3060_REG_MASK_FMT            0x03
+#define PCM3060_REG_FMT_I2S             0x00
+#define PCM3060_REG_FMT_LJ              0x01
+#define PCM3060_REG_FMT_RJ              0x02
+
+#define PCM3060_REG68                   0x44
+#define PCM3060_REG_OVER                0x40
+#define PCM3060_REG_DREV2               0x04
+#define PCM3060_REG_SHIFT_MUT21 0x00
+#define PCM3060_REG_SHIFT_MUT22 0x01
+
+#define PCM3060_REG69                   0x45
+#define PCM3060_REG_FLT         0x80
+#define PCM3060_REG_MASK_DMF            0x60
+#define PCM3060_REG_DMC         0x10
+#define PCM3060_REG_ZREV                0x02
+#define PCM3060_REG_AZRO                0x01
+
+#define PCM3060_REG70                   0x46
+#define PCM3060_REG71                   0x47
+#define PCM3060_REG_AT1_MIN             0x0E
+#define PCM3060_REG_AT1_MAX             0xFF
+
+#define PCM3060_REG73                   0x49
+#define PCM3060_REG_ZCDD                0x10
+#define PCM3060_REG_BYP         0x08
+#define PCM3060_REG_DREV1               0x04
+#define PCM3060_REG_SHIFT_MUT11 0x00
+#define PCM3060_REG_SHIFT_MUT12 0x01
+
+#endif /* _SND_SOC_PCM3060_H */
diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 3356c91f55b0..52cc950c9fd1 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -33,6 +33,8 @@
 #define PCM3168A_FMT_RIGHT_J_16         0x3
 #define PCM3168A_FMT_DSP_A              0x4
 #define PCM3168A_FMT_DSP_B              0x5
+#define PCM3168A_FMT_I2S_TDM            0x6
+#define PCM3168A_FMT_LEFT_J_TDM         0x7
 #define PCM3168A_FMT_DSP_MASK           0x4
 
 #define PCM3168A_NUM_SUPPLIES 6
@@ -401,9 +403,11 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
         bool tx, master_mode;
         u32 val, mask, shift, reg;
         unsigned int rate, fmt, ratio, max_ratio;
+        unsigned int chan;
         int i, min_frame_size;
 
         rate = params_rate(params);
+        chan = params_channels(params);
 
         ratio = pcm3168a->sysclk / rate;
 
@@ -456,6 +460,21 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
                 return -EINVAL;
         }
 
+        /* for TDM */
+        if (chan > 2) {
+                switch (fmt) {
+                case PCM3168A_FMT_I2S:
+                        fmt = PCM3168A_FMT_I2S_TDM;
+                        break;
+                case PCM3168A_FMT_LEFT_J:
+                        fmt = PCM3168A_FMT_LEFT_J_TDM;
+                        break;
+                default:
+                        dev_err(component->dev, "TDM is supported under I2S/Left_J only\n");
+                        return -EINVAL;
+                }
+        }
+
         if (master_mode)
                 val = ((i + 1) << shift);
         else
@@ -476,7 +495,69 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
         return 0;
 }
 
+static int pcm3168a_startup(struct snd_pcm_substream *substream,
+                            struct snd_soc_dai *dai)
+{
+        struct snd_soc_component *component = dai->component;
+        struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component);
+        bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+        unsigned int fmt;
+        unsigned int sample_min;
+        unsigned int channel_max;
+
+        if (tx)
+                fmt = pcm3168a->dac_fmt;
+        else
+                fmt = pcm3168a->adc_fmt;
+
+        /*
+         * Available Data Bits
+         *
+         * RIGHT_J : 24 / 16
+         * LEFT_J  : 24
+         * I2S     : 24
+         *
+         * TDM available
+         *
+         * I2S
+         * LEFT_J
+         */
+        switch (fmt) {
+        case PCM3168A_FMT_RIGHT_J:
+                sample_min  = 16;
+                channel_max =  2;
+                break;
+        case PCM3168A_FMT_LEFT_J:
+                sample_min  = 24;
+                if (tx)
+                        channel_max = 8;
+                else
+                        channel_max = 6;
+                break;
+        case PCM3168A_FMT_I2S:
+                sample_min  = 24;
+                if (tx)
+                        channel_max = 8;
+                else
+                        channel_max = 6;
+                break;
+        default:
+                sample_min  = 24;
+                channel_max =  2;
+        }
+
+        snd_pcm_hw_constraint_minmax(substream->runtime,
+                                     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+                                     sample_min, 32);
+
+        snd_pcm_hw_constraint_minmax(substream->runtime,
+                                     SNDRV_PCM_HW_PARAM_CHANNELS,
+                                     2, channel_max);
+
+        return 0;
+}
 static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = {
+        .startup        = pcm3168a_startup,
         .set_fmt        = pcm3168a_set_dai_fmt_dac,
         .set_sysclk     = pcm3168a_set_dai_sysclk,
         .hw_params      = pcm3168a_hw_params,
@@ -484,6 +565,7 @@ static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = {
 };
 
 static const struct snd_soc_dai_ops pcm3168a_adc_dai_ops = {
+        .startup        = pcm3168a_startup,
         .set_fmt        = pcm3168a_set_dai_fmt_adc,
         .set_sysclk     = pcm3168a_set_dai_sysclk,
         .hw_params      = pcm3168a_hw_params
diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c
index d88e67341083..0ef966d56bac 100644
--- a/sound/soc/codecs/rt274.c
+++ b/sound/soc/codecs/rt274.c
@@ -755,6 +755,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
                 break;
         default:
                 dev_warn(component->dev, "invalid pll source, use BCLK\n");
+                /* fall through */
         case RT274_PLL2_S_BCLK:
                 snd_soc_component_update_bits(component, RT274_PLL2_CTRL,
                                 RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_BCLK);
@@ -782,6 +783,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
                         break;
                 default:
                         dev_warn(component->dev, "invalid freq_in, assume 4.8M\n");
+                        /* fall through */
                 case 100:
                         snd_soc_component_write(component, 0x7a, 0xaab6);
                         snd_soc_component_write(component, 0x7b, 0x0301);
diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c
index 6478d10c4f4a..4d46f4567c3a 100644
--- a/sound/soc/codecs/rt5514-spi.c
+++ b/sound/soc/codecs/rt5514-spi.c
@@ -91,6 +91,14 @@ static void rt5514_spi_copy_work(struct work_struct *work)
 
         runtime = rt5514_dsp->substream->runtime;
         period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream);
+        if (!period_bytes) {
+                schedule_delayed_work(&rt5514_dsp->copy_work, 5);
+                goto done;
+        }
+
+        if (rt5514_dsp->buf_size % period_bytes)
+                rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) *
+                        period_bytes;
 
         if (rt5514_dsp->get_size >= rt5514_dsp->buf_size) {
                 rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf,
@@ -149,13 +157,11 @@ done:
 
 static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp)
 {
-        size_t period_bytes;
         u8 buf[8];
 
         if (!rt5514_dsp->substream)
                 return;
 
-        period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream);
         rt5514_dsp->get_size = 0;
 
         /**
@@ -183,10 +189,6 @@ static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp)
 
         rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base;
 
-        if (rt5514_dsp->buf_size % period_bytes)
-                rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) *
-                        period_bytes;
-
         if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit &&
                 rt5514_dsp->buf_rp && rt5514_dsp->buf_size)
                 schedule_delayed_work(&rt5514_dsp->copy_work, 0);
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 5bcedbc7eb4a..b7ba64350a07 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -10,7 +10,6 @@
  */
 
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index 70441661ea4a..7eb2cbd39d6e 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -72,6 +72,7 @@ struct rt5663_priv {
 static const struct reg_sequence rt5663_patch_list[] = {
         { 0x002a, 0x8020 },
         { 0x0086, 0x0028 },
+        { 0x0100, 0xa020 },
         { 0x0117, 0x0f28 },
         { 0x02fb, 0x8089 },
 };
@@ -580,7 +581,7 @@ static const struct reg_default rt5663_reg[] = {
         { 0x00fd, 0x0001 },
         { 0x00fe, 0x10ec },
         { 0x00ff, 0x6406 },
-        { 0x0100, 0xa0a0 },
+        { 0x0100, 0xa020 },
         { 0x0108, 0x4444 },
         { 0x0109, 0x4444 },
         { 0x010a, 0xaaaa },
@@ -2337,6 +2338,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w,
                                 0x8000);
                         snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000,
                                 0x3000);
+                        snd_soc_component_update_bits(component,
+                                RT5663_DIG_VOL_ZCD, 0x00c0, 0x0080);
                 }
                 break;
 
@@ -2351,6 +2354,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w,
                                 RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN);
                         snd_soc_component_update_bits(component,
                                 RT5663_DACREF_LDO, 0x3e0e, 0);
+                        snd_soc_component_update_bits(component,
+                                RT5663_DIG_VOL_ZCD, 0x00c0, 0);
                 }
                 break;
 
diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c
index 3f6046a66b56..230a21c93b6b 100644
--- a/sound/soc/codecs/rt5668.c
+++ b/sound/soc/codecs/rt5668.c
@@ -2588,17 +2588,10 @@ static int rt5668_i2c_probe(struct i2c_client *i2c,
 
         }
 
-        return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668,
+        return devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668,
                         rt5668_dai, ARRAY_SIZE(rt5668_dai));
 }
 
-static int rt5668_i2c_remove(struct i2c_client *i2c)
-{
-        snd_soc_unregister_component(&i2c->dev);
-
-        return 0;
-}
-
 static void rt5668_i2c_shutdown(struct i2c_client *client)
 {
         struct rt5668_priv *rt5668 = i2c_get_clientdata(client);
@@ -2629,7 +2622,6 @@ static struct i2c_driver rt5668_i2c_driver = {
                 .acpi_match_table = ACPI_PTR(rt5668_acpi_match),
         },
         .probe = rt5668_i2c_probe,
-        .remove = rt5668_i2c_remove,
         .shutdown = rt5668_i2c_shutdown,
         .id_table = rt5668_i2c_id,
 };
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index f0f8debc2829..453328c988c0 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -2878,6 +2878,18 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = {
         },
         {
                 .callback = rt5670_quirk_cb,
+                .ident = "Lenovo Thinkpad Tablet 8",
+                .matches = {
+                        DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+                        DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 8"),
+                },
+                .driver_data = (unsigned long *)(RT5670_DMIC_EN |
+                                                 RT5670_DMIC2_INR |
+                                                 RT5670_DEV_GPIO |
+                                                 RT5670_JD_MODE1),
+        },
+        {
+                .callback = rt5670_quirk_cb,
                 .ident = "Lenovo Thinkpad Tablet 10",
                 .matches = {
                         DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c
index bd51f3655ee3..84501c2020c7 100644
--- a/sound/soc/codecs/rt5677-spi.c
+++ b/sound/soc/codecs/rt5677-spi.c
@@ -18,7 +18,6 @@
 #include <linux/interrupt.h>
 #include <linux/irq.h>
 #include <linux/slab.h>
-#include <linux/gpio.h>
 #include <linux/sched.h>
 #include <linux/uaccess.h>
 #include <linux/regulator/consumer.h>
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 8a3052874c29..34cfaf8f6f34 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -67,7 +67,8 @@ struct rt5682_priv {
 };
 
 static const struct reg_sequence patch_list[] = {
-        {0x01c1, 0x1000},
+        {RT5682_HP_IMP_SENS_CTRL_19, 0x1000},
+        {RT5682_DAC_ADC_DIG_VOL1, 0xa020},
 };
 
 static const struct reg_default rt5682_reg[] = {
@@ -749,7 +750,6 @@ static bool rt5682_readable_register(struct device *dev, unsigned int reg)
         }
 }
 
-static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0);
 static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0);
 static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0);
 static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
@@ -1108,10 +1108,6 @@ static void rt5682_jack_detect_handler(struct work_struct *work)
 }
 
 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 + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv),
@@ -1437,6 +1433,28 @@ 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_charge_pump_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_update_bits(component,
+                        RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV);
+                break;
+        case SND_SOC_DAPM_POST_PMD:
+                snd_soc_component_update_bits(component,
+                        RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_LV);
+                break;
+        default:
+                return 0;
+        }
+
+        return 0;
+}
+
 static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
         struct snd_kcontrol *kcontrol, int event)
 {
@@ -1449,10 +1467,10 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
                         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);
+                snd_soc_component_update_bits(component,
+                        RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080);
                 break;
 
         case SND_SOC_DAPM_POST_PMD:
@@ -1460,6 +1478,8 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
                         RT5682_DEPOP_1, 0x60, 0x0);
                 snd_soc_component_write(component,
                         RT5682_HP_CTRL_2, 0x0000);
+                snd_soc_component_update_bits(component,
+                        RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0000);
                 break;
 
         default:
@@ -1723,7 +1743,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
         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),
+                RT5682_PUMP_EN_SFT, 0, rt5682_charge_pump_event,
+                SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
         SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1,
                 RT5682_CAPLESS_EN_SFT, 0, NULL, 0),
 
@@ -1884,6 +1905,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
         {"HP Amp", NULL, "Charge Pump"},
         {"HP Amp", NULL, "CLKDET SYS"},
         {"HP Amp", NULL, "CBJ Power"},
+        {"HP Amp", NULL, "Vref1"},
         {"HP Amp", NULL, "Vref2"},
         {"HPOL Playback", "Switch", "HP Amp"},
         {"HPOR Playback", "Switch", "HP Amp"},
@@ -2452,30 +2474,23 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
         mutex_lock(&rt5682->calibrate_mutex);
 
         rt5682_reset(rt5682->regmap);
-        regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf);
+        regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af);
         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_PWR_ANLG_1, 0xf2af);
+        regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0300);
+        regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x8000);
+        regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100);
+        regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800);
         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_CALIB_ADC_CTRL, 0x7005);
         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, 0x7c00);
 
         regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00);
 
@@ -2491,8 +2506,12 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
                 pr_err("HP Calibration Failure\n");
 
         /* restore settings */
-        regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4);
+        regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0x02af);
+        regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0080);
+        regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000);
         regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000);
+        regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000);
+        regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005);
 
         mutex_unlock(&rt5682->calibrate_mutex);
 
@@ -2566,7 +2585,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 
         rt5682_calibrate(rt5682);
 
-        ret = regmap_register_patch(rt5682->regmap, patch_list,
+        ret = regmap_multi_reg_write(rt5682->regmap, patch_list,
                                     ARRAY_SIZE(patch_list));
         if (ret != 0)
                 dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret);
@@ -2620,6 +2639,10 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
                         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);
+        regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8,
+                        RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA);
+        regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1,
+                        RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ);
 
         INIT_DELAYED_WORK(&rt5682->jack_detect_work,
                                 rt5682_jack_detect_handler);
@@ -2637,11 +2660,17 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 
         }
 
-        return devm_snd_soc_register_component(&i2c->dev,
-                        &soc_component_dev_rt5682,
+        return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682,
                         rt5682_dai, ARRAY_SIZE(rt5682_dai));
 }
 
+static int rt5682_i2c_remove(struct i2c_client *i2c)
+{
+        snd_soc_unregister_component(&i2c->dev);
+
+        return 0;
+}
+
 static void rt5682_i2c_shutdown(struct i2c_client *client)
 {
         struct rt5682_priv *rt5682 = i2c_get_clientdata(client);
@@ -2672,6 +2701,7 @@ static struct i2c_driver rt5682_i2c_driver = {
                 .acpi_match_table = ACPI_PTR(rt5682_acpi_match),
         },
         .probe = rt5682_i2c_probe,
+        .remove = rt5682_i2c_remove,
         .shutdown = rt5682_i2c_shutdown,
         .id_table = rt5682_i2c_id,
 };
diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h
index 8068140ebe3f..d82a8301fd74 100644
--- a/sound/soc/codecs/rt5682.h
+++ b/sound/soc/codecs/rt5682.h
@@ -1214,6 +1214,20 @@
 #define RT5682_JDH_NO_PLUG                      (0x1 << 4)
 #define RT5682_JDH_PLUG                         (0x0 << 4)
 
+/* Bias current control 8 (0x0111) */
+#define RT5682_HPA_CP_BIAS_CTRL_MASK                    (0x3 << 2)
+#define RT5682_HPA_CP_BIAS_2UA                  (0x0 << 2)
+#define RT5682_HPA_CP_BIAS_3UA                  (0x1 << 2)
+#define RT5682_HPA_CP_BIAS_4UA                  (0x2 << 2)
+#define RT5682_HPA_CP_BIAS_6UA                  (0x3 << 2)
+
+/* Charge Pump Internal Register1 (0x0125) */
+#define RT5682_CP_CLK_HP_MASK                   (0x3 << 4)
+#define RT5682_CP_CLK_HP_100KHZ                 (0x0 << 4)
+#define RT5682_CP_CLK_HP_200KHZ                 (0x1 << 4)
+#define RT5682_CP_CLK_HP_300KHZ                 (0x2 << 4)
+#define RT5682_CP_CLK_HP_600KHZ                 (0x3 << 4)
+
 /* Chopper and Clock control for DAC (0x013a)*/
 #define RT5682_CKXEN_DAC1_MASK                  (0x1 << 13)
 #define RT5682_CKXEN_DAC1_SFT                   13
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 60764f6201b1..add18d6d77da 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1218,7 +1218,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_component *component)
          * Searching for a suitable index solving this formula:
          * idx = 40 * log10(vag_val / lo_cagcntrl) + 15
          */
-        vol_quot = (vag * 100) / lo_vag;
+        vol_quot = lo_vag ? (vag * 100) / lo_vag : 0;
         lo_vol = 0;
         for (i = 0; i < ARRAY_SIZE(vol_quot_table); i++) {
                 if (vol_quot >= vol_quot_table[i])
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index d5035f2f2b2b..f753d2db0a5a 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -21,6 +21,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/init.h>
+#include <linux/clk.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
@@ -142,6 +143,7 @@ static const char *sta32x_supply_names[] = {
 /* codec private data */
 struct sta32x_priv {
         struct regmap *regmap;
+        struct clk *xti_clk;
         struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)];
         struct snd_soc_component *component;
         struct sta32x_platform_data *pdata;
@@ -879,6 +881,18 @@ static int sta32x_probe(struct snd_soc_component *component)
         struct sta32x_priv *sta32x = snd_soc_component_get_drvdata(component);
         struct sta32x_platform_data *pdata = sta32x->pdata;
         int i, ret = 0, thermal = 0;
+
+        sta32x->component = component;
+
+        if (sta32x->xti_clk) {
+                ret = clk_prepare_enable(sta32x->xti_clk);
+                if (ret != 0) {
+                        dev_err(component->dev,
+                                "Failed to enable clock: %d\n", ret);
+                        return ret;
+                }
+        }
+
         ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies),
                                     sta32x->supplies);
         if (ret != 0) {
@@ -981,6 +995,9 @@ static void sta32x_remove(struct snd_soc_component *component)
 
         sta32x_watchdog_stop(sta32x);
         regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+
+        if (sta32x->xti_clk)
+                clk_disable_unprepare(sta32x->xti_clk);
 }
 
 static const struct snd_soc_component_driver sta32x_component = {
@@ -1038,6 +1055,8 @@ static int sta32x_probe_dt(struct device *dev, struct sta32x_priv *sta32x)
         of_property_read_u8(np, "st,ch3-output-mapping",
                             &pdata->ch3_output_mapping);
 
+        if (of_get_property(np, "st,fault-detect-recovery", NULL))
+                pdata->fault_detect_recovery = 1;
         if (of_get_property(np, "st,thermal-warning-recovery", NULL))
                 pdata->thermal_warning_recovery = 1;
         if (of_get_property(np, "st,thermal-warning-adjustment", NULL))
@@ -1095,6 +1114,17 @@ static int sta32x_i2c_probe(struct i2c_client *i2c,
         }
 #endif
 
+        /* Clock */
+        sta32x->xti_clk = devm_clk_get(dev, "xti");
+        if (IS_ERR(sta32x->xti_clk)) {
+                ret = PTR_ERR(sta32x->xti_clk);
+
+                if (ret == -EPROBE_DEFER)
+                        return ret;
+
+                sta32x->xti_clk = NULL;
+        }
+
         /* GPIOs */
         sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset",
                                                        GPIOD_OUT_LOW);
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
index ae3d032ac35a..6bd0e5d5347f 100644
--- a/sound/soc/codecs/tas5720.c
+++ b/sound/soc/codecs/tas5720.c
@@ -152,6 +152,7 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai,
                                     int slots, int slot_width)
 {
         struct snd_soc_component *component = dai->component;
+        struct tas5720_data *tas5720 = snd_soc_component_get_drvdata(component);
         unsigned int first_slot;
         int ret;
 
@@ -185,6 +186,20 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai,
         if (ret < 0)
                 goto error_snd_soc_component_update_bits;
 
+        /* Configure TDM slot width. This is only applicable to TAS5722. */
+        switch (tas5720->devtype) {
+        case TAS5722:
+                ret = snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG,
+                                                    TAS5722_TDM_SLOT_16B,
+                                                    slot_width == 16 ?
+                                                    TAS5722_TDM_SLOT_16B : 0);
+                if (ret < 0)
+                        goto error_snd_soc_component_update_bits;
+                break;
+        default:
+                break;
+        }
+
         return 0;
 
 error_snd_soc_component_update_bits:
@@ -485,15 +500,56 @@ static const DECLARE_TLV_DB_RANGE(dac_analog_tlv,
 );
 
 /*
- * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that
- * setting the gain below -100 dB (register value <0x7) is effectively a MUTE
- * as per device datasheet.
+ * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB or 0.25 dB steps
+ * depending on the device. Note that setting the gain below -100 dB
+ * (register value <0x7) is effectively a MUTE as per device datasheet.
+ *
+ * Note that for the TAS5722 the digital volume controls are actually split
+ * over two registers, so we need custom getters/setters for access.
  */
-static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas5720_dac_tlv, -10350, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas5722_dac_tlv, -10350, 25, 0);
+
+static int tas5722_volume_get(struct snd_kcontrol *kcontrol,
+                              struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+        unsigned int val;
+
+        snd_soc_component_read(component, TAS5720_VOLUME_CTRL_REG, &val);
+        ucontrol->value.integer.value[0] = val << 1;
+
+        snd_soc_component_read(component, TAS5722_DIGITAL_CTRL2_REG, &val);
+        ucontrol->value.integer.value[0] |= val & TAS5722_VOL_CONTROL_LSB;
+
+        return 0;
+}
+
+static int tas5722_volume_set(struct snd_kcontrol *kcontrol,
+                              struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+        unsigned int sel = ucontrol->value.integer.value[0];
+
+        snd_soc_component_write(component, TAS5720_VOLUME_CTRL_REG, sel >> 1);
+        snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG,
+                                      TAS5722_VOL_CONTROL_LSB, sel);
+
+        return 0;
+}
 
 static const struct snd_kcontrol_new tas5720_snd_controls[] = {
         SOC_SINGLE_TLV("Speaker Driver Playback Volume",
-                       TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv),
+                       TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, tas5720_dac_tlv),
+        SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG,
+                       TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv),
+};
+
+static const struct snd_kcontrol_new tas5722_snd_controls[] = {
+        SOC_SINGLE_EXT_TLV("Speaker Driver Playback Volume",
+                           0, 0, 511, 0,
+                           tas5722_volume_get, tas5722_volume_set,
+                           tas5722_dac_tlv),
         SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG,
                        TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv),
 };
@@ -527,6 +583,23 @@ static const struct snd_soc_component_driver soc_component_dev_tas5720 = {
         .non_legacy_dai_naming  = 1,
 };
 
+static const struct snd_soc_component_driver soc_component_dev_tas5722 = {
+        .probe = tas5720_codec_probe,
+        .remove = tas5720_codec_remove,
+        .suspend = tas5720_suspend,
+        .resume = tas5720_resume,
+        .controls = tas5722_snd_controls,
+        .num_controls = ARRAY_SIZE(tas5722_snd_controls),
+        .dapm_widgets = tas5720_dapm_widgets,
+        .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets),
+        .dapm_routes = tas5720_audio_map,
+        .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map),
+        .idle_bias_on           = 1,
+        .use_pmdown_time        = 1,
+        .endianness             = 1,
+        .non_legacy_dai_naming  = 1,
+};
+
 /* PCM rates supported by the TAS5720 driver */
 #define TAS5720_RATES   (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
                          SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
@@ -613,9 +686,23 @@ static int tas5720_probe(struct i2c_client *client,
 
         dev_set_drvdata(dev, data);
 
-        ret = devm_snd_soc_register_component(&client->dev,
-                                     &soc_component_dev_tas5720,
-                                     tas5720_dai, ARRAY_SIZE(tas5720_dai));
+        switch (id->driver_data) {
+        case TAS5720:
+                ret = devm_snd_soc_register_component(&client->dev,
+                                        &soc_component_dev_tas5720,
+                                        tas5720_dai,
+                                        ARRAY_SIZE(tas5720_dai));
+                break;
+        case TAS5722:
+                ret = devm_snd_soc_register_component(&client->dev,
+                                        &soc_component_dev_tas5722,
+                                        tas5720_dai,
+                                        ARRAY_SIZE(tas5720_dai));
+                break;
+        default:
+                dev_err(dev, "unexpected private driver data\n");
+                return -EINVAL;
+        }
         if (ret < 0) {
                 dev_err(dev, "failed to register component: %d\n", ret);
                 return ret;
diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c
index 0d6145549a98..36aebdb8f55c 100644
--- a/sound/soc/codecs/tas6424.c
+++ b/sound/soc/codecs/tas6424.c
@@ -41,6 +41,7 @@ struct tas6424_data {
         struct regmap *regmap;
         struct regulator_bulk_data supplies[TAS6424_NUM_SUPPLIES];
         struct delayed_work fault_check_work;
+        unsigned int last_cfault;
         unsigned int last_fault1;
         unsigned int last_fault2;
         unsigned int last_warn;
@@ -406,9 +407,54 @@ static void tas6424_fault_check_work(struct work_struct *work)
         unsigned int reg;
         int ret;
 
+        ret = regmap_read(tas6424->regmap, TAS6424_CHANNEL_FAULT, &reg);
+        if (ret < 0) {
+                dev_err(dev, "failed to read CHANNEL_FAULT register: %d\n", ret);
+                goto out;
+        }
+
+        if (!reg) {
+                tas6424->last_cfault = reg;
+                goto check_global_fault1_reg;
+        }
+
+        /*
+         * Only flag errors once for a given occurrence. This is needed as
+         * the TAS6424 will take time clearing the fault condition internally
+         * during which we don't want to bombard the system with the same
+         * error message over and over.
+         */
+        if ((reg & TAS6424_FAULT_OC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH1))
+                dev_crit(dev, "experienced a channel 1 overcurrent fault\n");
+
+        if ((reg & TAS6424_FAULT_OC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH2))
+                dev_crit(dev, "experienced a channel 2 overcurrent fault\n");
+
+        if ((reg & TAS6424_FAULT_OC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH3))
+                dev_crit(dev, "experienced a channel 3 overcurrent fault\n");
+
+        if ((reg & TAS6424_FAULT_OC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH4))
+                dev_crit(dev, "experienced a channel 4 overcurrent fault\n");
+
+        if ((reg & TAS6424_FAULT_DC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH1))
+                dev_crit(dev, "experienced a channel 1 DC fault\n");
+
+        if ((reg & TAS6424_FAULT_DC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH2))
+                dev_crit(dev, "experienced a channel 2 DC fault\n");
+
+        if ((reg & TAS6424_FAULT_DC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH3))
+                dev_crit(dev, "experienced a channel 3 DC fault\n");
+
+        if ((reg & TAS6424_FAULT_DC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH4))
+                dev_crit(dev, "experienced a channel 4 DC fault\n");
+
+        /* Store current fault1 value so we can detect any changes next time */
+        tas6424->last_cfault = reg;
+
+check_global_fault1_reg:
         ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT1, &reg);
         if (ret < 0) {
-                dev_err(dev, "failed to read FAULT1 register: %d\n", ret);
+                dev_err(dev, "failed to read GLOB_FAULT1 register: %d\n", ret);
                 goto out;
         }
 
@@ -429,12 +475,6 @@ static void tas6424_fault_check_work(struct work_struct *work)
                 goto check_global_fault2_reg;
         }
 
-        /*
-         * Only flag errors once for a given occurrence. This is needed as
-         * the TAS6424 will take time clearing the fault condition internally
-         * during which we don't want to bombard the system with the same
-         * error message over and over.
-         */
         if ((reg & TAS6424_FAULT_PVDD_OV) && !(tas6424->last_fault1 & TAS6424_FAULT_PVDD_OV))
                 dev_crit(dev, "experienced a PVDD overvoltage fault\n");
 
@@ -453,7 +493,7 @@ static void tas6424_fault_check_work(struct work_struct *work)
 check_global_fault2_reg:
         ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT2, &reg);
         if (ret < 0) {
-                dev_err(dev, "failed to read FAULT2 register: %d\n", ret);
+                dev_err(dev, "failed to read GLOB_FAULT2 register: %d\n", ret);
                 goto out;
         }
 
@@ -530,7 +570,7 @@ check_warn_reg:
         /* Store current warn value so we can detect any changes next time */
         tas6424->last_warn = reg;
 
-        /* Clear any faults by toggling the CLEAR_FAULT control bit */
+        /* Clear any warnings by toggling the CLEAR_FAULT control bit */
         ret = regmap_write_bits(tas6424->regmap, TAS6424_MISC_CTRL3,
                                 TAS6424_CLEAR_FAULT, TAS6424_CLEAR_FAULT);
         if (ret < 0)
diff --git a/sound/soc/codecs/tas6424.h b/sound/soc/codecs/tas6424.h
index b5958c45ed0e..c67a7835ca66 100644
--- a/sound/soc/codecs/tas6424.h
+++ b/sound/soc/codecs/tas6424.h
@@ -116,6 +116,16 @@
 #define TAS6424_LDGBYPASS_MASK          BIT(TAS6424_LDGBYPASS_SHIFT)
 
 /* TAS6424_GLOB_FAULT1_REG */
+#define TAS6424_FAULT_OC_CH1            BIT(7)
+#define TAS6424_FAULT_OC_CH2            BIT(6)
+#define TAS6424_FAULT_OC_CH3            BIT(5)
+#define TAS6424_FAULT_OC_CH4            BIT(4)
+#define TAS6424_FAULT_DC_CH1            BIT(3)
+#define TAS6424_FAULT_DC_CH2            BIT(2)
+#define TAS6424_FAULT_DC_CH3            BIT(1)
+#define TAS6424_FAULT_DC_CH4            BIT(0)
+
+/* TAS6424_GLOB_FAULT1_REG */
 #define TAS6424_FAULT_CLOCK             BIT(4)
 #define TAS6424_FAULT_PVDD_OV           BIT(3)
 #define TAS6424_FAULT_VBAT_OV           BIT(2)
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index bf92d36b8f8a..608ad49ad978 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -167,6 +167,7 @@ struct aic31xx_priv {
         u8 p_div;
         int rate_div_line;
         bool master_dapm_route_applied;
+        int irq;
 };
 
 struct aic31xx_rate_divs {
@@ -1391,6 +1392,69 @@ static const struct acpi_device_id aic31xx_acpi_match[] = {
 MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match);
 #endif
 
+static irqreturn_t aic31xx_irq(int irq, void *data)
+{
+        struct aic31xx_priv *aic31xx = data;
+        struct device *dev = aic31xx->dev;
+        unsigned int value;
+        bool handled = false;
+        int ret;
+
+        ret = regmap_read(aic31xx->regmap, AIC31XX_INTRDACFLAG, &value);
+        if (ret) {
+                dev_err(dev, "Failed to read interrupt mask: %d\n", ret);
+                goto exit;
+        }
+
+        if (value)
+                handled = true;
+        else
+                goto read_overflow;
+
+        if (value & AIC31XX_HPLSCDETECT)
+                dev_err(dev, "Short circuit on Left output is detected\n");
+        if (value & AIC31XX_HPRSCDETECT)
+                dev_err(dev, "Short circuit on Right output is detected\n");
+        if (value & ~(AIC31XX_HPLSCDETECT |
+                      AIC31XX_HPRSCDETECT))
+                dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value);
+
+read_overflow:
+        ret = regmap_read(aic31xx->regmap, AIC31XX_OFFLAG, &value);
+        if (ret) {
+                dev_err(dev, "Failed to read overflow flag: %d\n", ret);
+                goto exit;
+        }
+
+        if (value)
+                handled = true;
+        else
+                goto exit;
+
+        if (value & AIC31XX_DAC_OF_LEFT)
+                dev_warn(dev, "Left-channel DAC overflow has occurred\n");
+        if (value & AIC31XX_DAC_OF_RIGHT)
+                dev_warn(dev, "Right-channel DAC overflow has occurred\n");
+        if (value & AIC31XX_DAC_OF_SHIFTER)
+                dev_warn(dev, "DAC barrel shifter overflow has occurred\n");
+        if (value & AIC31XX_ADC_OF)
+                dev_warn(dev, "ADC overflow has occurred\n");
+        if (value & AIC31XX_ADC_OF_SHIFTER)
+                dev_warn(dev, "ADC barrel shifter overflow has occurred\n");
+        if (value & ~(AIC31XX_DAC_OF_LEFT |
+                      AIC31XX_DAC_OF_RIGHT |
+                      AIC31XX_DAC_OF_SHIFTER |
+                      AIC31XX_ADC_OF |
+                      AIC31XX_ADC_OF_SHIFTER))
+                dev_warn(dev, "Unknown overflow interrupt flags: 0x%08x\n", value);
+
+exit:
+        if (handled)
+                return IRQ_HANDLED;
+        else
+                return IRQ_NONE;
+}
+
 static int aic31xx_i2c_probe(struct i2c_client *i2c,
                              const struct i2c_device_id *id)
 {
@@ -1413,6 +1477,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
                 return ret;
         }
         aic31xx->dev = &i2c->dev;
+        aic31xx->irq = i2c->irq;
 
         aic31xx->codec_type = id->driver_data;
 
@@ -1456,6 +1521,26 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
                 return ret;
         }
 
+        if (aic31xx->irq > 0) {
+                regmap_update_bits(aic31xx->regmap, AIC31XX_GPIO1,
+                                   AIC31XX_GPIO1_FUNC_MASK,
+                                   AIC31XX_GPIO1_INT1 <<
+                                   AIC31XX_GPIO1_FUNC_SHIFT);
+
+                regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL,
+                             AIC31XX_SC |
+                             AIC31XX_ENGINE);
+
+                ret = devm_request_threaded_irq(aic31xx->dev, aic31xx->irq,
+                                                NULL, aic31xx_irq,
+                                                IRQF_ONESHOT, "aic31xx-irq",
+                                                aic31xx);
+                if (ret) {
+                        dev_err(aic31xx->dev, "Unable to request IRQ\n");
+                        return ret;
+                }
+        }
+
         if (aic31xx->codec_type & DAC31XX_BIT)
                 return devm_snd_soc_register_component(&i2c->dev,
                                 &soc_codec_driver_aic31xx,
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
index 0b587585b38b..2636f2c6bc79 100644
--- a/sound/soc/codecs/tlv320aic31xx.h
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -173,6 +173,13 @@ struct aic31xx_pdata {
 #define AIC31XX_HPRDRVPWRSTATUS_MASK    BIT(1)
 #define AIC31XX_SPRDRVPWRSTATUS_MASK    BIT(0)
 
+/* AIC31XX_OFFLAG */
+#define AIC31XX_DAC_OF_LEFT             BIT(7)
+#define AIC31XX_DAC_OF_RIGHT            BIT(6)
+#define AIC31XX_DAC_OF_SHIFTER          BIT(5)
+#define AIC31XX_ADC_OF                  BIT(3)
+#define AIC31XX_ADC_OF_SHIFTER          BIT(1)
+
 /* AIC31XX_INTRDACFLAG */
 #define AIC31XX_HPLSCDETECT             BIT(7)
 #define AIC31XX_HPRSCDETECT             BIT(6)
@@ -191,6 +198,22 @@ struct aic31xx_pdata {
 #define AIC31XX_SC                      BIT(3)
 #define AIC31XX_ENGINE                  BIT(2)
 
+/* AIC31XX_GPIO1 */
+#define AIC31XX_GPIO1_FUNC_MASK         GENMASK(5, 2)
+#define AIC31XX_GPIO1_FUNC_SHIFT        2
+#define AIC31XX_GPIO1_DISABLED          0x00
+#define AIC31XX_GPIO1_INPUT             0x01
+#define AIC31XX_GPIO1_GPI               0x02
+#define AIC31XX_GPIO1_GPO               0x03
+#define AIC31XX_GPIO1_CLKOUT            0x04
+#define AIC31XX_GPIO1_INT1              0x05
+#define AIC31XX_GPIO1_INT2              0x06
+#define AIC31XX_GPIO1_ADC_WCLK          0x07
+#define AIC31XX_GPIO1_SBCLK             0x08
+#define AIC31XX_GPIO1_SWCLK             0x09
+#define AIC31XX_GPIO1_ADC_MOD_CLK       0x10
+#define AIC31XX_GPIO1_SDOUT             0x11
+
 /* AIC31XX_DACSETUP */
 #define AIC31XX_SOFTSTEP_MASK           GENMASK(1, 0)
 
diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c
index ff85a0bf6170..93d84e5ae2d5 100644
--- a/sound/soc/codecs/tscs454.c
+++ b/sound/soc/codecs/tscs454.c
@@ -3459,7 +3459,7 @@ static int tscs454_i2c_probe(struct i2c_client *i2c,
         /* Sync pg sel reg with cache */
         regmap_write(tscs454->regmap, R_PAGESEL, 0x00);
 
-        ret = snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454,
+        ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454,
                         tscs454_dais, ARRAY_SIZE(tscs454_dais));
         if (ret) {
                 dev_err(&i2c->dev, "Failed to register component (%d)\n", ret);
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index c5ae07234a00..bba330e30162 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -88,19 +88,6 @@ static int wm2000_write(struct i2c_client *i2c, unsigned int reg,
         return regmap_write(wm2000->regmap, reg, value);
 }
 
-static unsigned int wm2000_read(struct i2c_client *i2c, unsigned int r)
-{
-        struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c);
-        unsigned int val;
-        int ret;
-
-        ret = regmap_read(wm2000->regmap, r, &val);
-        if (ret < 0)
-                return -1;
-
-        return val;
-}
-
 static void wm2000_reset(struct wm2000_priv *wm2000)
 {
         struct i2c_client *i2c = wm2000->i2c;
@@ -115,14 +102,15 @@ static void wm2000_reset(struct wm2000_priv *wm2000)
 static int wm2000_poll_bit(struct i2c_client *i2c,
                            unsigned int reg, u8 mask)
 {
+        struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c);
         int timeout = 4000;
-        int val;
+        unsigned int val;
 
-        val = wm2000_read(i2c, reg);
+        regmap_read(wm2000->regmap, reg, &val);
 
         while (!(val & mask) && --timeout) {
                 msleep(1);
-                val = wm2000_read(i2c, reg);
+                regmap_read(wm2000->regmap, reg, &val);
         }
 
         if (timeout == 0)
@@ -135,6 +123,7 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
 {
         struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
         unsigned long rate;
+        unsigned int val;
         int ret;
 
         if (WARN_ON(wm2000->anc_mode != ANC_OFF))
@@ -213,12 +202,17 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
                              WM2000_MODE_THERMAL_ENABLE);
         }
 
-        ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY);
+        ret = regmap_read(wm2000->regmap, WM2000_REG_SPEECH_CLARITY, &val);
+        if (ret != 0) {
+                dev_err(&i2c->dev, "Unable to read Speech Clarity: %d\n", ret);
+                regulator_bulk_disable(WM2000_NUM_SUPPLIES, wm2000->supplies);
+                return ret;
+        }
         if (wm2000->speech_clarity)
-                ret |= WM2000_SPEECH_CLARITY;
+                val |= WM2000_SPEECH_CLARITY;
         else
-                ret &= ~WM2000_SPEECH_CLARITY;
-        wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret);
+                val &= ~WM2000_SPEECH_CLARITY;
+        wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, val);
 
         wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33);
         wm2000_write(i2c, WM2000_REG_SYS_START1, 0x02);
@@ -824,7 +818,7 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
         const char *filename;
         const struct firmware *fw = NULL;
         int ret, i;
-        int reg;
+        unsigned int reg;
         u16 id;
 
         wm2000 = devm_kzalloc(&i2c->dev, sizeof(*wm2000), GFP_KERNEL);
@@ -860,9 +854,17 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
         }
 
         /* Verify that this is a WM2000 */
-        reg = wm2000_read(i2c, WM2000_REG_ID1);
+        ret = regmap_read(wm2000->regmap, WM2000_REG_ID1, &reg);
+        if (ret != 0) {
+                dev_err(&i2c->dev, "Unable to read ID1: %d\n", ret);
+                return ret;
+        }
         id = reg << 8;
-        reg = wm2000_read(i2c, WM2000_REG_ID2);
+        ret = regmap_read(wm2000->regmap, WM2000_REG_ID2, &reg);
+        if (ret != 0) {
+                dev_err(&i2c->dev, "Unable to read ID2: %d\n", ret);
+                return ret;
+        }
         id |= reg & 0xff;
 
         if (id != 0x2000) {
@@ -871,7 +873,11 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
                 goto err_supplies;
         }
 
-        reg = wm2000_read(i2c, WM2000_REG_REVISON);
+        ret = regmap_read(wm2000->regmap, WM2000_REG_REVISON, &reg);
+        if (ret != 0) {
+                dev_err(&i2c->dev, "Unable to read Revision: %d\n", ret);
+                return ret;
+        }
         dev_info(&i2c->dev, "revision %c\n", reg + 'A');
 
         wm2000->mclk = devm_clk_get(&i2c->dev, "MCLK");
diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c
index 317db9a149a7..cf2cdbece122 100644
--- a/sound/soc/codecs/wm8782.c
+++ b/sound/soc/codecs/wm8782.c
@@ -20,6 +20,7 @@
 #include <linux/module.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
+#include <linux/regulator/consumer.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/ac97_codec.h>
@@ -50,7 +51,51 @@ static struct snd_soc_dai_driver wm8782_dai = {
         },
 };
 
+/* regulator power supply names */
+static const char *supply_names[] = {
+        "Vdda", /* analog supply, 2.7V - 3.6V */
+        "Vdd",  /* digital supply, 2.7V - 5.5V */
+};
+
+struct wm8782_priv {
+        struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
+};
+
+static int wm8782_soc_probe(struct snd_soc_component *component)
+{
+        struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+        return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+
+static void wm8782_soc_remove(struct snd_soc_component *component)
+{
+        struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+        regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+
+#ifdef CONFIG_PM
+static int wm8782_soc_suspend(struct snd_soc_component *component)
+{
+        struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+        regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+        return 0;
+}
+
+static int wm8782_soc_resume(struct snd_soc_component *component)
+{
+        struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+        return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+#else
+#define wm8782_soc_suspend      NULL
+#define wm8782_soc_resume       NULL
+#endif /* CONFIG_PM */
+
 static const struct snd_soc_component_driver soc_component_dev_wm8782 = {
+        .probe                  = wm8782_soc_probe,
+        .remove                 = wm8782_soc_remove,
+        .suspend                = wm8782_soc_suspend,
+        .resume                 = wm8782_soc_resume,
         .dapm_widgets           = wm8782_dapm_widgets,
         .num_dapm_widgets       = ARRAY_SIZE(wm8782_dapm_widgets),
         .dapm_routes            = wm8782_dapm_routes,
@@ -63,6 +108,24 @@ static const struct snd_soc_component_driver soc_component_dev_wm8782 = {
 
 static int wm8782_probe(struct platform_device *pdev)
 {
+        struct device *dev = &pdev->dev;
+        struct wm8782_priv *priv;
+        int ret, i;
+
+        priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+        if (!priv)
+                return -ENOMEM;
+
+        dev_set_drvdata(dev, priv);
+
+        for (i = 0; i < ARRAY_SIZE(supply_names); i++)
+                priv->supplies[i].supply = supply_names[i];
+
+        ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies),
+                                      priv->supplies);
+        if (ret < 0)
+                return ret;
+
         return devm_snd_soc_register_component(&pdev->dev,
                         &soc_component_dev_wm8782, &wm8782_dai, 1);
 }
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 1965635ec07c..2a3e5fbd04e4 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -13,7 +13,6 @@
 
 #include <linux/clk.h>
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 43edaf8cd276..593a11960888 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -11,7 +11,6 @@
  */
 
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index ade34c26ad2f..e873baa9e778 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -638,13 +638,14 @@ static int wm9712_soc_probe(struct snd_soc_component *component)
 {
         struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component);
         struct regmap *regmap;
-        int ret;
 
         if (wm9712->mfd_pdata) {
                 wm9712->ac97 = wm9712->mfd_pdata->ac97;
                 regmap = wm9712->mfd_pdata->regmap;
         } else {
 #ifdef CONFIG_SND_SOC_AC97_BUS
+                int ret;
+
                 wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID,
                                                       WM9712_VENDOR_ID_MASK);
                 if (IS_ERR(wm9712->ac97)) {
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index f61656070225..a53dc174bbf0 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -311,12 +311,12 @@ struct wm_adsp_alg_xm_struct {
 };
 
 struct wm_adsp_buffer {
-        __be32 X_buf_base;              /* XM base addr of first X area */
-        __be32 X_buf_size;              /* Size of 1st X area in words */
-        __be32 X_buf_base2;             /* XM base addr of 2nd X area */
-        __be32 X_buf_brk;               /* Total X size in words */
-        __be32 Y_buf_base;              /* YM base addr of Y area */
-        __be32 wrap;                    /* Total size X and Y in words */
+        __be32 buf1_base;               /* Base addr of first buffer area */
+        __be32 buf1_size;               /* Size of buf1 area in DSP words */
+        __be32 buf2_base;               /* Base addr of 2nd buffer area */
+        __be32 buf1_buf2_size;          /* Size of buf1+buf2 in DSP words */
+        __be32 buf3_base;               /* Base addr of buf3 area */
+        __be32 buf_total_size;          /* Size of buf1+buf2+buf3 in DSP words */
         __be32 high_water_mark;         /* Point at which IRQ is asserted */
         __be32 irq_count;               /* bits 1-31 count IRQ assertions */
         __be32 irq_ack;                 /* acked IRQ count, bit 0 enables IRQ */
@@ -393,18 +393,18 @@ struct wm_adsp_buffer_region_def {
 static const struct wm_adsp_buffer_region_def default_regions[] = {
         {
                 .mem_type = WMFW_ADSP2_XM,
-                .base_offset = HOST_BUFFER_FIELD(X_buf_base),
-                .size_offset = HOST_BUFFER_FIELD(X_buf_size),
+                .base_offset = HOST_BUFFER_FIELD(buf1_base),
+                .size_offset = HOST_BUFFER_FIELD(buf1_size),
         },
         {
                 .mem_type = WMFW_ADSP2_XM,
-                .base_offset = HOST_BUFFER_FIELD(X_buf_base2),
-                .size_offset = HOST_BUFFER_FIELD(X_buf_brk),
+                .base_offset = HOST_BUFFER_FIELD(buf2_base),
+                .size_offset = HOST_BUFFER_FIELD(buf1_buf2_size),
         },
         {
                 .mem_type = WMFW_ADSP2_YM,
-                .base_offset = HOST_BUFFER_FIELD(Y_buf_base),
-                .size_offset = HOST_BUFFER_FIELD(wrap),
+                .base_offset = HOST_BUFFER_FIELD(buf3_base),
+                .size_offset = HOST_BUFFER_FIELD(buf_total_size),
         },
 };
 
@@ -3345,7 +3345,7 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf)
                 region->cumulative_size = offset;
 
                 adsp_dbg(buf->dsp,
-                         "region=%d type=%d base=%04x off=%04x size=%04x\n",
+                         "region=%d type=%d base=%08x off=%08x size=%08x\n",
                          i, region->mem_type, region->base_addr,
                          region->offset, region->cumulative_size);
         }
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index f70db8412c7c..267aee776b2d 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -1041,6 +1041,42 @@ static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp,
         return error_ppm;
 }
 
+static inline u32 davinci_mcasp_tx_delay(struct davinci_mcasp *mcasp)
+{
+        if (!mcasp->txnumevt)
+                return 0;
+
+        return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_WFIFOSTS_OFFSET);
+}
+
+static inline u32 davinci_mcasp_rx_delay(struct davinci_mcasp *mcasp)
+{
+        if (!mcasp->rxnumevt)
+                return 0;
+
+        return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_RFIFOSTS_OFFSET);
+}
+
+static snd_pcm_sframes_t davinci_mcasp_delay(
+                        struct snd_pcm_substream *substream,
+                        struct snd_soc_dai *cpu_dai)
+{
+        struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+        u32 fifo_use;
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                fifo_use = davinci_mcasp_tx_delay(mcasp);
+        else
+                fifo_use = davinci_mcasp_rx_delay(mcasp);
+
+        /*
+         * Divide the used locations with the channel count to get the
+         * FIFO usage in samples (don't care about partial samples in the
+         * buffer).
+         */
+        return fifo_use / substream->runtime->channels;
+}
+
 static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
                                         struct snd_pcm_hw_params *params,
                                         struct snd_soc_dai *cpu_dai)
@@ -1365,6 +1401,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
         .startup        = davinci_mcasp_startup,
         .shutdown       = davinci_mcasp_shutdown,
         .trigger        = davinci_mcasp_trigger,
+        .delay          = davinci_mcasp_delay,
         .hw_params      = davinci_mcasp_hw_params,
         .set_fmt        = davinci_mcasp_set_dai_fmt,
         .set_clkdiv     = davinci_mcasp_set_clkdiv,
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 1033ac6631b0..01052a0808b0 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -151,7 +151,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream,
         int ret;
 
         /* Fetch the Back-End dma_data from DPCM */
-        list_for_each_entry(dpcm, &rtd->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(rtd, stream, dpcm) {
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 struct snd_pcm_substream *substream_be;
                 struct snd_soc_dai *dai = be->cpu_dai;
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index c1d1d06783e5..57b484768a58 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -807,7 +807,7 @@ static int fsl_esai_probe(struct platform_device *pdev)
                 return -ENOMEM;
 
         esai_priv->pdev = pdev;
-        strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1);
+        snprintf(esai_priv->name, sizeof(esai_priv->name), "%pOFn", np);
 
         /* Get the addresses and IRQ */
         res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
index 7f0fa4b52223..9981668ab590 100644
--- a/sound/soc/fsl/fsl_utils.c
+++ b/sound/soc/fsl/fsl_utils.c
@@ -57,8 +57,8 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np,
                 of_node_put(dma_channel_np);
                 return ret;
         }
-        snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
-                 (unsigned long long) res.start, dma_channel_np->name);
+        snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%pOFn",
+                 (unsigned long long) res.start, dma_channel_np);
 
         iprop = of_get_property(dma_channel_np, "cell-index", NULL);
         if (!iprop) {
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index ec731223cab3..e339f36cea95 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -57,6 +57,7 @@ static int pcm030_fabric_probe(struct platform_device *op)
         struct device_node *platform_np;
         struct snd_soc_card *card = &pcm030_card;
         struct pcm030_audio_data *pdata;
+        struct snd_soc_dai_link *dai_link;
         int ret;
         int i;
 
@@ -78,8 +79,8 @@ static int pcm030_fabric_probe(struct platform_device *op)
                 return -ENODEV;
         }
 
-        for (i = 0; i < card->num_links; i++)
-                card->dai_link[i].platform_of_node = platform_np;
+        for_each_card_prelinks(card, i, dai_link)
+                dai_link->platform_of_node = platform_np;
 
         ret = request_module("snd-soc-wm9712");
         if (ret)
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 2094d2c8919f..25c819e402e1 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -25,6 +25,8 @@ struct graph_card_data {
         struct graph_dai_props {
                 struct asoc_simple_dai cpu_dai;
                 struct asoc_simple_dai codec_dai;
+                struct snd_soc_dai_link_component codecs; /* single codec */
+                struct snd_soc_dai_link_component platform;
                 unsigned int mclk_fs;
         } *dai_props;
         unsigned int mclk_fs;
@@ -180,7 +182,8 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port,
         if (ret < 0)
                 goto dai_link_of_err;
 
-        of_property_read_u32(rcpu_ep, "mclk-fs", &dai_props->mclk_fs);
+        of_property_read_u32(cpu_ep,   "mclk-fs", &dai_props->mclk_fs);
+        of_property_read_u32(codec_ep, "mclk-fs", &dai_props->mclk_fs);
 
         ret = asoc_simple_card_parse_graph_cpu(cpu_ep, dai_link);
         if (ret < 0)
@@ -213,7 +216,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port,
         ret = asoc_simple_card_set_dailink_name(dev, dai_link,
                                                 "%s-%s",
                                                 dai_link->cpu_dai_name,
-                                                dai_link->codec_dai_name);
+                                                dai_link->codecs->dai_name);
         if (ret < 0)
                 goto dai_link_of_err;
 
@@ -299,7 +302,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
         struct graph_dai_props *dai_props;
         struct device *dev = &pdev->dev;
         struct snd_soc_card *card;
-        int num, ret;
+        int num, ret, i;
 
         /* Allocate the private data and the DAI link array */
         priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -315,6 +318,18 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
         if (!dai_props || !dai_link)
                 return -ENOMEM;
 
+        /*
+         * Use snd_soc_dai_link_component instead of legacy style
+         * It is codec only. but cpu/platform will be supported in the future.
+         * see
+         *      soc-core.c :: snd_soc_init_multicodec()
+         */
+        for (i = 0; i < num; i++) {
+                dai_link[i].codecs      = &dai_props[i].codecs;
+                dai_link[i].num_codecs  = 1;
+                dai_link[i].platform    = &dai_props[i].platform;
+        }
+
         priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW);
         if (IS_ERR(priv->pa_gpio)) {
                 ret = PTR_ERR(priv->pa_gpio);
diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c
index 92882e392d6c..b83bb31021a9 100644
--- a/sound/soc/generic/audio-graph-scu-card.c
+++ b/sound/soc/generic/audio-graph-scu-card.c
@@ -25,7 +25,11 @@
 struct graph_card_data {
         struct snd_soc_card snd_card;
         struct snd_soc_codec_conf codec_conf;
-        struct asoc_simple_dai *dai_props;
+        struct graph_dai_props {
+                struct asoc_simple_dai dai;
+                struct snd_soc_dai_link_component codecs;
+                struct snd_soc_dai_link_component platform;
+        } *dai_props;
         struct snd_soc_dai_link *dai_link;
         struct asoc_simple_card_data adata;
 };
@@ -39,18 +43,18 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream)
 {
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
-        struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num);
+        struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num);
 
-        return asoc_simple_card_clk_enable(dai_props);
+        return asoc_simple_card_clk_enable(&dai_props->dai);
 }
 
 static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream)
 {
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
-        struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num);
+        struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num);
 
-        asoc_simple_card_clk_disable(dai_props);
+        asoc_simple_card_clk_disable(&dai_props->dai);
 }
 
 static const struct snd_soc_ops asoc_graph_card_ops = {
@@ -63,7 +67,7 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd)
         struct graph_card_data *priv =  snd_soc_card_get_drvdata(rtd->card);
         struct snd_soc_dai *dai;
         struct snd_soc_dai_link *dai_link;
-        struct asoc_simple_dai *dai_props;
+        struct graph_dai_props *dai_props;
         int num = rtd->num;
 
         dai_link        = graph_priv_to_link(priv, num);
@@ -72,7 +76,7 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd)
                                 rtd->cpu_dai :
                                 rtd->codec_dai;
 
-        return asoc_simple_card_init_dai(dai, dai_props);
+        return asoc_simple_card_init_dai(dai, &dai_props->dai);
 }
 
 static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
@@ -92,15 +96,18 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep,
 {
         struct device *dev = graph_priv_to_dev(priv);
         struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, idx);
-        struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, idx);
+        struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx);
         struct snd_soc_card *card = graph_priv_to_card(priv);
         int ret;
 
         if (is_fe) {
+                struct snd_soc_dai_link_component *codecs;
+
                 /* BE is dummy */
-                dai_link->codec_of_node         = NULL;
-                dai_link->codec_dai_name        = "snd-soc-dummy-dai";
-                dai_link->codec_name            = "snd-soc-dummy";
+                codecs                  = dai_link->codecs;
+                codecs->of_node         = NULL;
+                codecs->dai_name        = "snd-soc-dummy-dai";
+                codecs->name            = "snd-soc-dummy";
 
                 /* FE settings */
                 dai_link->dynamic               = 1;
@@ -110,7 +117,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep,
                 if (ret)
                         return ret;
 
-                ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai_props);
+                ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, &dai_props->dai);
                 if (ret < 0)
                         return ret;
 
@@ -137,23 +144,23 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep,
                 if (ret < 0)
                         return ret;
 
-                ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai_props);
+                ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, &dai_props->dai);
                 if (ret < 0)
                         return ret;
 
                 ret = asoc_simple_card_set_dailink_name(dev, dai_link,
                                                         "be.%s",
-                                                        dai_link->codec_dai_name);
+                                                        dai_link->codecs->dai_name);
                 if (ret < 0)
                         return ret;
 
                 snd_soc_of_parse_audio_prefix(card,
                                               &priv->codec_conf,
-                                              dai_link->codec_of_node,
+                                              dai_link->codecs->of_node,
                                               "prefix");
         }
 
-        ret = asoc_simple_card_of_parse_tdm(ep, dai_props);
+        ret = asoc_simple_card_of_parse_tdm(ep, &dai_props->dai);
         if (ret)
                 return ret;
 
@@ -331,10 +338,10 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
 {
         struct graph_card_data *priv;
         struct snd_soc_dai_link *dai_link;
-        struct asoc_simple_dai *dai_props;
+        struct graph_dai_props *dai_props;
         struct device *dev = &pdev->dev;
         struct snd_soc_card *card;
-        int num, ret;
+        int num, ret, i;
 
         /* Allocate the private data and the DAI link array */
         priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -350,6 +357,18 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
         if (!dai_props || !dai_link)
                 return -ENOMEM;
 
+        /*
+         * Use snd_soc_dai_link_component instead of legacy style
+         * It is codec only. but cpu/platform will be supported in the future.
+         * see
+         *      soc-core.c :: snd_soc_init_multicodec()
+         */
+        for (i = 0; i < num; i++) {
+                dai_link[i].codecs      = &dai_props[i].codecs;
+                dai_link[i].num_codecs  = 1;
+                dai_link[i].platform    = &dai_props[i].platform;
+        }
+
         priv->dai_props                 = dai_props;
         priv->dai_link                  = dai_link;
 
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index d3f3f0fec74c..f34cc6cddfa2 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -173,12 +173,24 @@ int asoc_simple_card_parse_clk(struct device *dev,
                                struct device_node *node,
                                struct device_node *dai_of_node,
                                struct asoc_simple_dai *simple_dai,
-                               const char *name)
+                               const char *dai_name,
+                               struct snd_soc_dai_link_component *dlc)
 {
         struct clk *clk;
         u32 val;
 
         /*
+         * Use snd_soc_dai_link_component instead of legacy style.
+         * It is only for codec, but cpu will be supported in the future.
+         * see
+         *      soc-core.c :: snd_soc_init_multicodec()
+         */
+        if (dlc) {
+                dai_of_node     = dlc->of_node;
+                dai_name        = dlc->dai_name;
+        }
+
+        /*
          * Parse dai->sysclk come from "clocks = <&xxx>"
          * (if system has common clock)
          *  or "system-clock-frequency = <xxx>"
@@ -200,7 +212,7 @@ int asoc_simple_card_parse_clk(struct device *dev,
         if (of_property_read_bool(node, "system-clock-direction-out"))
                 simple_dai->clk_direction = SND_SOC_CLOCK_OUT;
 
-        dev_dbg(dev, "%s : sysclk = %d, direction %d\n", name,
+        dev_dbg(dev, "%s : sysclk = %d, direction %d\n", dai_name,
                 simple_dai->sysclk, simple_dai->clk_direction);
 
         return 0;
@@ -208,6 +220,7 @@ int asoc_simple_card_parse_clk(struct device *dev,
 EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk);
 
 int asoc_simple_card_parse_dai(struct device_node *node,
+                                    struct snd_soc_dai_link_component *dlc,
                                     struct device_node **dai_of_node,
                                     const char **dai_name,
                                     const char *list_name,
@@ -221,6 +234,17 @@ int asoc_simple_card_parse_dai(struct device_node *node,
                 return 0;
 
         /*
+         * Use snd_soc_dai_link_component instead of legacy style.
+         * It is only for codec, but cpu will be supported in the future.
+         * see
+         *      soc-core.c :: snd_soc_init_multicodec()
+         */
+        if (dlc) {
+                dai_name        = &dlc->dai_name;
+                dai_of_node     = &dlc->of_node;
+        }
+
+        /*
          * Get node via "sound-dai = <&phandle port>"
          * it will be used as xxx_of_node on soc_bind_dai_link()
          */
@@ -278,6 +302,7 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep)
 }
 
 int asoc_simple_card_parse_graph_dai(struct device_node *ep,
+                                     struct snd_soc_dai_link_component *dlc,
                                      struct device_node **dai_of_node,
                                      const char **dai_name)
 {
@@ -285,6 +310,17 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep,
         struct of_phandle_args args;
         int ret;
 
+        /*
+         * Use snd_soc_dai_link_component instead of legacy style.
+         * It is only for codec, but cpu will be supported in the future.
+         * see
+         *      soc-core.c :: snd_soc_init_multicodec()
+         */
+        if (dlc) {
+                dai_name        = &dlc->dai_name;
+                dai_of_node     = &dlc->of_node;
+        }
+
         if (!ep)
                 return 0;
         if (!dai_name)
@@ -340,10 +376,11 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai);
 int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link)
 {
         /* Assumes platform == cpu */
-        if (!dai_link->platform_of_node)
-                dai_link->platform_of_node = dai_link->cpu_of_node;
+        if (!dai_link->platform->of_node)
+                dai_link->platform->of_node = dai_link->cpu_of_node;
 
         return 0;
+
 }
 EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_dailink);
 
@@ -367,13 +404,11 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_cpu);
 int asoc_simple_card_clean_reference(struct snd_soc_card *card)
 {
         struct snd_soc_dai_link *dai_link;
-        int num_links;
+        int i;
 
-        for (num_links = 0, dai_link = card->dai_link;
-             num_links < card->num_links;
-             num_links++, dai_link++) {
+        for_each_card_prelinks(card, i, dai_link) {
                 of_node_put(dai_link->cpu_of_node);
-                of_node_put(dai_link->codec_of_node);
+                of_node_put(dai_link->codecs->of_node);
         }
         return 0;
 }
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 64bf3560c1d1..5a3f59aa4ba5 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -20,6 +20,8 @@ struct simple_card_data {
         struct simple_dai_props {
                 struct asoc_simple_dai cpu_dai;
                 struct asoc_simple_dai codec_dai;
+                struct snd_soc_dai_link_component codecs; /* single codec */
+                struct snd_soc_dai_link_component platform;
                 unsigned int mclk_fs;
         } *dai_props;
         unsigned int mclk_fs;
@@ -234,7 +236,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
         ret = asoc_simple_card_set_dailink_name(dev, dai_link,
                                                 "%s-%s",
                                                 dai_link->cpu_dai_name,
-                                                dai_link->codec_dai_name);
+                                                dai_link->codecs->dai_name);
         if (ret < 0)
                 goto dai_link_of_err;
 
@@ -363,7 +365,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
         struct device *dev = &pdev->dev;
         struct device_node *np = dev->of_node;
         struct snd_soc_card *card;
-        int num, ret;
+        int num, ret, i;
 
         /* Get the number of DAI links */
         if (np && of_get_child_by_name(np, PREFIX "dai-link"))
@@ -381,6 +383,18 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
         if (!dai_props || !dai_link)
                 return -ENOMEM;
 
+        /*
+         * Use snd_soc_dai_link_component instead of legacy style
+         * It is codec only. but cpu/platform will be supported in the future.
+         * see
+         *      soc-core.c :: snd_soc_init_multicodec()
+         */
+        for (i = 0; i < num; i++) {
+                dai_link[i].codecs      = &dai_props[i].codecs;
+                dai_link[i].num_codecs  = 1;
+                dai_link[i].platform    = &dai_props[i].platform;
+        }
+
         priv->dai_props                 = dai_props;
         priv->dai_link                  = dai_link;
 
@@ -403,6 +417,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 
         } else {
                 struct asoc_simple_card_info *cinfo;
+                struct snd_soc_dai_link_component *codecs;
+                struct snd_soc_dai_link_component *platform;
 
                 cinfo = dev->platform_data;
                 if (!cinfo) {
@@ -419,13 +435,17 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
                         return -EINVAL;
                 }
 
+                codecs                  = dai_link->codecs;
+                codecs->name            = cinfo->codec;
+                codecs->dai_name        = cinfo->codec_dai.name;
+
+                platform                = dai_link->platform;
+                platform->name          = cinfo->platform;
+
                 card->name              = (cinfo->card) ? cinfo->card : cinfo->name;
                 dai_link->name          = cinfo->name;
                 dai_link->stream_name   = cinfo->name;
-                dai_link->platform_name = cinfo->platform;
-                dai_link->codec_name    = cinfo->codec;
                 dai_link->cpu_dai_name  = cinfo->cpu_dai.name;
-                dai_link->codec_dai_name = cinfo->codec_dai.name;
                 dai_link->dai_fmt       = cinfo->daifmt;
                 dai_link->init          = asoc_simple_card_dai_init;
                 memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai,
diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c
index 16a83bc51e0e..85b46f0eae0f 100644
--- a/sound/soc/generic/simple-scu-card.c
+++ b/sound/soc/generic/simple-scu-card.c
@@ -22,7 +22,11 @@
 struct simple_card_data {
         struct snd_soc_card snd_card;
         struct snd_soc_codec_conf codec_conf;
-        struct asoc_simple_dai *dai_props;
+        struct simple_dai_props {
+                struct asoc_simple_dai dai;
+                struct snd_soc_dai_link_component codecs;
+                struct snd_soc_dai_link_component platform;
+        } *dai_props;
         struct snd_soc_dai_link *dai_link;
         struct asoc_simple_card_data adata;
 };
@@ -40,20 +44,20 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream)
 {
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
-        struct asoc_simple_dai *dai_props =
+        struct simple_dai_props *dai_props =
                 simple_priv_to_props(priv, rtd->num);
 
-        return asoc_simple_card_clk_enable(dai_props);
+        return asoc_simple_card_clk_enable(&dai_props->dai);
 }
 
 static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream)
 {
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
-        struct asoc_simple_dai *dai_props =
+        struct simple_dai_props *dai_props =
                 simple_priv_to_props(priv, rtd->num);
 
-        asoc_simple_card_clk_disable(dai_props);
+        asoc_simple_card_clk_disable(&dai_props->dai);
 }
 
 static const struct snd_soc_ops asoc_simple_card_ops = {
@@ -66,7 +70,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
         struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
         struct snd_soc_dai *dai;
         struct snd_soc_dai_link *dai_link;
-        struct asoc_simple_dai *dai_props;
+        struct simple_dai_props *dai_props;
         int num = rtd->num;
 
         dai_link        = simple_priv_to_link(priv, num);
@@ -75,7 +79,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
                                 rtd->cpu_dai :
                                 rtd->codec_dai;
 
-        return asoc_simple_card_init_dai(dai, dai_props);
+        return asoc_simple_card_init_dai(dai, &dai_props->dai);
 }
 
 static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
@@ -95,17 +99,19 @@ static int asoc_simple_card_dai_link_of(struct device_node *np,
 {
         struct device *dev = simple_priv_to_dev(priv);
         struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx);
-        struct asoc_simple_dai *dai_props = simple_priv_to_props(priv, idx);
+        struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx);
         struct snd_soc_card *card = simple_priv_to_card(priv);
         int ret;
 
         if (is_fe) {
                 int is_single_links = 0;
+                struct snd_soc_dai_link_component *codecs;
 
                 /* BE is dummy */
-                dai_link->codec_of_node         = NULL;
-                dai_link->codec_dai_name        = "snd-soc-dummy-dai";
-                dai_link->codec_name            = "snd-soc-dummy";
+                codecs                  = dai_link->codecs;
+                codecs->of_node         = NULL;
+                codecs->dai_name        = "snd-soc-dummy-dai";
+                codecs->name            = "snd-soc-dummy";
 
                 /* FE settings */
                 dai_link->dynamic               = 1;
@@ -116,7 +122,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np,
                 if (ret)
                         return ret;
 
-                ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai_props);
+                ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, &dai_props->dai);
                 if (ret < 0)
                         return ret;
 
@@ -141,23 +147,23 @@ static int asoc_simple_card_dai_link_of(struct device_node *np,
                 if (ret < 0)
                         return ret;
 
-                ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai_props);
+                ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, &dai_props->dai);
                 if (ret < 0)
                         return ret;
 
                 ret = asoc_simple_card_set_dailink_name(dev, dai_link,
                                                         "be.%s",
-                                                        dai_link->codec_dai_name);
+                                                        dai_link->codecs->dai_name);
                 if (ret < 0)
                         return ret;
 
                 snd_soc_of_parse_audio_prefix(card,
                                               &priv->codec_conf,
-                                              dai_link->codec_of_node,
+                                              dai_link->codecs->of_node,
                                               PREFIX "prefix");
         }
 
-        ret = asoc_simple_card_of_parse_tdm(np, dai_props);
+        ret = asoc_simple_card_of_parse_tdm(np, &dai_props->dai);
         if (ret)
                 return ret;
 
@@ -230,11 +236,11 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 {
         struct simple_card_data *priv;
         struct snd_soc_dai_link *dai_link;
-        struct asoc_simple_dai *dai_props;
+        struct simple_dai_props *dai_props;
         struct snd_soc_card *card;
         struct device *dev = &pdev->dev;
         struct device_node *np = dev->of_node;
-        int num, ret;
+        int num, ret, i;
 
         /* Allocate the private data */
         priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -248,6 +254,18 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
         if (!dai_props || !dai_link)
                 return -ENOMEM;
 
+        /*
+         * Use snd_soc_dai_link_component instead of legacy style
+         * It is codec only. but cpu/platform will be supported in the future.
+         * see
+         *      soc-core.c :: snd_soc_init_multicodec()
+         */
+        for (i = 0; i < num; i++) {
+                dai_link[i].codecs      = &dai_props[i].codecs;
+                dai_link[i].num_codecs  = 1;
+                dai_link[i].platform    = &dai_props[i].platform;
+        }
+
         priv->dai_props                         = dai_props;
         priv->dai_link                          = dai_link;
 
diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c
index 53344a3b7a60..a69e5b11b3da 100644
--- a/sound/soc/hisilicon/hi6210-i2s.c
+++ b/sound/soc/hisilicon/hi6210-i2s.c
@@ -269,13 +269,13 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream,
         switch (params_format(params)) {
         case SNDRV_PCM_FORMAT_U16_LE:
                 signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
-                /* fallthru */
+                /* fall through */
         case SNDRV_PCM_FORMAT_S16_LE:
                 bits = HII2S_BITS_16;
                 break;
         case SNDRV_PCM_FORMAT_U24_LE:
                 signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
-                /* fallthru */
+                /* fall through */
         case SNDRV_PCM_FORMAT_S24_LE:
                 bits = HII2S_BITS_24;
                 break;
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 6c36da560877..afc559866095 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -765,7 +765,7 @@ static int sst_soc_prepare(struct device *dev)
         snd_soc_poweroff(drv->soc_card->dev);
 
         /* set the SSPs to idle */
-        list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) {
+        for_each_card_rtds(drv->soc_card, rtd) {
                 struct snd_soc_dai *dai = rtd->cpu_dai;
 
                 if (dai->active) {
@@ -786,7 +786,7 @@ static void sst_soc_complete(struct device *dev)
                 return;
 
         /* restart SSPs */
-        list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) {
+        for_each_card_rtds(drv->soc_card, rtd) {
                 struct snd_soc_dai *dai = rtd->cpu_dai;
 
                 if (dai->active) {
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index cccda87f4b34..73ca1350aa31 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -279,6 +279,28 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH
           This adds support for ASoC Onboard Codec I2S machine driver. This will
           create an alsa sound card for DA7219 + MAX98357A I2S audio codec.
           Say Y if you have such a device.
+
+config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH
+        tristate "KBL with DA7219 and MAX98927 in I2S Mode"
+        depends on MFD_INTEL_LPSS && I2C && ACPI
+        select SND_SOC_DA7219
+        select SND_SOC_MAX98927
+        select SND_SOC_DMIC
+        select SND_SOC_HDAC_HDMI
+        help
+          This adds support for ASoC Onboard Codec I2S machine driver. This will
+          create an alsa sound card for DA7219 + MAX98927 I2S audio codec.
+          Say Y if you have such a device.
+          If unsure select "N".
+
+config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
+        tristate "SKL/KBL/BXT/APL with HDA Codecs"
+        select SND_SOC_HDAC_HDMI
+        select SND_SOC_HDAC_HDA
+        help
+          This adds support for ASoC machine driver for Intel platforms
+          SKL/KBL/BXT/APL with iDisp, HDA audio codecs.
+          Say Y or m if you have such a device. This is a recommended option.
           If unsure select "N".
 
 config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 87ef8b4058e5..5381e27df9cc 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -17,9 +17,11 @@ snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o
 snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o
 snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o
 snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o
+snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o
 snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o
 snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o
 snd-soc-skl_rt286-objs := skl_rt286.o
+snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o
 snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o
 snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o
 
@@ -41,8 +43,10 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o
 obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o
+obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o
 obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o
 obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt5663_rt5514_max98927.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o
+obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 7b0ee67b4fc8..68e6543e6cb0 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -223,7 +223,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
 static int broadwell_suspend(struct snd_soc_card *card){
         struct snd_soc_component *component;
 
-        list_for_each_entry(component, &card->component_dev_list, card_list) {
+        for_each_card_components(card, component) {
                 if (!strcmp(component->name, "i2c-INT343A:00")) {
 
                         dev_dbg(component->dev, "disabling jack detect before going to suspend.\n");
@@ -237,7 +237,7 @@ static int broadwell_suspend(struct snd_soc_card *card){
 static int broadwell_resume(struct snd_soc_card *card){
         struct snd_soc_component *component;
 
-        list_for_each_entry(component, &card->component_dev_list, card_list) {
+        for_each_card_components(card, component) {
                 if (!strcmp(component->name, "i2c-INT343A:00")) {
 
                         dev_dbg(component->dev, "enabling jack detect for resume.\n");
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index b6dc524830b2..8587bd3d1cc1 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -1048,7 +1048,7 @@ static int byt_rt5640_suspend(struct snd_soc_card *card)
         if (!BYT_RT5640_JDSRC(byt_rt5640_quirk))
                 return 0;
 
-        list_for_each_entry(component, &card->component_dev_list, card_list) {
+        for_each_card_components(card, component) {
                 if (!strcmp(component->name, byt_rt5640_codec_name)) {
                         dev_dbg(component->dev, "disabling jack detect before suspend\n");
                         snd_soc_component_set_jack(component, NULL, NULL);
@@ -1067,7 +1067,7 @@ static int byt_rt5640_resume(struct snd_soc_card *card)
         if (!BYT_RT5640_JDSRC(byt_rt5640_quirk))
                 return 0;
 
-        list_for_each_entry(component, &card->component_dev_list, card_list) {
+        for_each_card_components(card, component) {
                 if (!strcmp(component->name, byt_rt5640_codec_name)) {
                         dev_dbg(component->dev, "re-enabling jack detect after resume\n");
                         snd_soc_component_set_jack(component, &priv->jack, NULL);
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 6af02bf879ac..c44298130720 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -742,7 +742,7 @@ static int byt_rt5651_suspend(struct snd_soc_card *card)
         if (!BYT_RT5651_JDSRC(byt_rt5651_quirk))
                 return 0;
 
-        list_for_each_entry(component, &card->component_dev_list, card_list) {
+        for_each_card_components(card, component) {
                 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);
@@ -761,7 +761,7 @@ static int byt_rt5651_resume(struct snd_soc_card *card)
         if (!BYT_RT5651_JDSRC(byt_rt5651_quirk))
                 return 0;
 
-        list_for_each_entry(component, &card->component_dev_list, card_list) {
+        for_each_card_components(card, component) {
                 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);
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
index e5aa13058dd7..51f0d45d6f8f 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5672.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -16,6 +16,7 @@
  *  General Public License for more details.
  */
 
+#include <linux/input.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
@@ -212,6 +213,10 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
         if (ret)
                 return ret;
 
+        snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+        snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+        snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+
         rt5670_set_jack_detect(component, &ctx->headset);
         if (ctx->mclk) {
                 /*
@@ -342,7 +347,7 @@ static int cht_suspend_pre(struct snd_soc_card *card)
         struct snd_soc_component *component;
         struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card);
 
-        list_for_each_entry(component, &card->component_dev_list, card_list) {
+        for_each_card_components(card, component) {
                 if (!strncmp(component->name,
                              ctx->codec_name, sizeof(ctx->codec_name))) {
 
@@ -359,7 +364,7 @@ static int cht_resume_post(struct snd_soc_card *card)
         struct snd_soc_component *component;
         struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card);
 
-        list_for_each_entry(component, &card->component_dev_list, card_list) {
+        for_each_card_components(card, component) {
                 if (!strncmp(component->name,
                              ctx->codec_name, sizeof(ctx->codec_name))) {
 
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
new file mode 100644
index 000000000000..3fa1c3ca6d37
--- /dev/null
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -0,0 +1,983 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2018 Intel Corporation.
+
+/*
+ * Intel Kabylake I2S Machine Driver with MAX98927 & DA7219 Codecs
+ *
+ * Modified from:
+ *   Intel Kabylake I2S Machine driver supporting MAX98927 and
+ *   RT5663 codecs
+ */
+
+#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 "../../codecs/da7219.h"
+#include "../../codecs/hdac_hdmi.h"
+#include "../skylake/skl.h"
+#include "../../codecs/da7219-aad.h"
+
+#define KBL_DIALOG_CODEC_DAI    "da7219-hifi"
+#define MAX98927_CODEC_DAI      "max98927-aif1"
+#define MAXIM_DEV0_NAME         "i2c-MX98927:00"
+#define MAXIM_DEV1_NAME         "i2c-MX98927:01"
+#define DUAL_CHANNEL    2
+#define QUAD_CHANNEL    4
+#define NAME_SIZE       32
+
+static struct snd_soc_card *kabylake_audio_card;
+static struct snd_soc_jack kabylake_hdmi[3];
+
+struct kbl_hdmi_pcm {
+        struct list_head head;
+        struct snd_soc_dai *codec_dai;
+        int device;
+};
+
+struct kbl_codec_private {
+        struct snd_soc_jack kabylake_headset;
+        struct list_head hdmi_pcm_list;
+};
+
+enum {
+        KBL_DPCM_AUDIO_PB = 0,
+        KBL_DPCM_AUDIO_CP,
+        KBL_DPCM_AUDIO_ECHO_REF_CP,
+        KBL_DPCM_AUDIO_REF_CP,
+        KBL_DPCM_AUDIO_DMIC_CP,
+        KBL_DPCM_AUDIO_HDMI1_PB,
+        KBL_DPCM_AUDIO_HDMI2_PB,
+        KBL_DPCM_AUDIO_HDMI3_PB,
+        KBL_DPCM_AUDIO_HS_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, KBL_DIALOG_CODEC_DAI);
+        if (!codec_dai) {
+                dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n");
+                return -EIO;
+        }
+
+        /* Configure sysclk for codec */
+        ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000,
+                                     SND_SOC_CLOCK_IN);
+        if (ret) {
+                dev_err(card->dev, "can't set codec sysclk configuration\n");
+                return ret;
+        }
+
+        if (SND_SOC_DAPM_EVENT_OFF(event)) {
+                ret = snd_soc_dai_set_pll(codec_dai, 0,
+                                     DA7219_SYSCLK_MCLK, 0, 0);
+                if (ret)
+                        dev_err(card->dev, "failed to stop PLL: %d\n", ret);
+        } else if (SND_SOC_DAPM_EVENT_ON(event)) {
+                ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM,
+                                     0, DA7219_PLL_FREQ_OUT_98304);
+                if (ret)
+                        dev_err(card->dev, "failed to start PLL: %d\n", ret);
+        }
+
+        return ret;
+}
+
+static const struct snd_kcontrol_new kabylake_controls[] = {
+        SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+        SOC_DAPM_PIN_SWITCH("Headset Mic"),
+        SOC_DAPM_PIN_SWITCH("Left Spk"),
+        SOC_DAPM_PIN_SWITCH("Right Spk"),
+};
+
+static const struct snd_soc_dapm_widget kabylake_widgets[] = {
+        SND_SOC_DAPM_HP("Headphone Jack", NULL),
+        SND_SOC_DAPM_MIC("Headset Mic", NULL),
+        SND_SOC_DAPM_SPK("Left Spk", NULL),
+        SND_SOC_DAPM_SPK("Right Spk", NULL),
+        SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+        SND_SOC_DAPM_SPK("DP", NULL),
+        SND_SOC_DAPM_SPK("HDMI", 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 kabylake_map[] = {
+        /* speaker */
+        { "Left Spk", NULL, "Left BE_OUT" },
+        { "Right Spk", NULL, "Right BE_OUT" },
+
+        /* other jacks */
+        { "DMic", NULL, "SoC DMIC" },
+
+        { "HDMI", NULL, "hif5 Output" },
+        { "DP", NULL, "hif6 Output" },
+
+        /* CODEC BE connections */
+        { "Left HiFi Playback", NULL, "ssp0 Tx" },
+        { "Right HiFi Playback", NULL, "ssp0 Tx" },
+        { "ssp0 Tx", NULL, "spk_out" },
+
+        /* IV feedback path */
+        { "codec0_fb_in", NULL, "ssp0 Rx"},
+        { "ssp0 Rx", NULL, "Left HiFi Capture" },
+        { "ssp0 Rx", NULL, "Right HiFi Capture" },
+
+        /* AEC capture path */
+        { "echo_ref_out", NULL, "ssp0 Rx" },
+
+        /* DMIC */
+        { "dmic01_hifi", NULL, "DMIC01 Rx" },
+        { "DMIC01 Rx", NULL, "DMIC AIF" },
+
+        { "hifi1", NULL, "iDisp1 Tx" },
+        { "iDisp1 Tx", NULL, "iDisp1_out" },
+        { "hifi2", NULL, "iDisp2 Tx" },
+        { "iDisp2 Tx", NULL, "iDisp2_out" },
+        { "hifi3", NULL, "iDisp3 Tx"},
+        { "iDisp3 Tx", NULL, "iDisp3_out"},
+};
+
+static const struct snd_soc_dapm_route kabylake_ssp1_map[] = {
+        { "Headphone Jack", NULL, "HPL" },
+        { "Headphone Jack", NULL, "HPR" },
+
+        /* other jacks */
+        { "MIC", NULL, "Headset Mic" },
+
+        /* CODEC BE connections */
+        { "Playback", NULL, "ssp1 Tx" },
+        { "ssp1 Tx", NULL, "codec1_out" },
+
+        { "hs_in", NULL, "ssp1 Rx" },
+        { "ssp1 Rx", NULL, "Capture" },
+
+        { "Headphone Jack", NULL, "Platform Clock" },
+        { "Headset Mic", NULL, "Platform Clock" },
+};
+
+static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
+        struct snd_pcm_hw_params *params)
+{
+        struct snd_soc_pcm_runtime *runtime = substream->private_data;
+        int ret = 0, j;
+
+        for (j = 0; j < runtime->num_codecs; j++) {
+                struct snd_soc_dai *codec_dai = runtime->codec_dais[j];
+
+                if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
+                        ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16);
+                        if (ret < 0) {
+                                dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret);
+                                return ret;
+                        }
+                }
+                if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) {
+                        ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16);
+                        if (ret < 0) {
+                                dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret);
+                                return ret;
+                        }
+                }
+        }
+
+        return 0;
+}
+
+static struct snd_soc_ops kabylake_ssp0_ops = {
+        .hw_params = kabylake_ssp0_hw_params,
+};
+
+static int kabylake_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);
+        struct snd_soc_dpcm *dpcm = container_of(
+                        params, struct snd_soc_dpcm, hw_params);
+        struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link;
+        struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link;
+
+        /*
+         * The ADSP will convert the FE rate to 48k, stereo, 24 bit
+         */
+        if (!strcmp(fe_dai_link->name, "Kbl Audio Port") ||
+            !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
+            !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
+                rate->min = rate->max = 48000;
+                channels->min = channels->max = 2;
+                snd_mask_none(fmt);
+                snd_mask_set(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);
+
+        return 0;
+}
+
+static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+        struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+        struct snd_soc_component *component = rtd->codec_dai->component;
+        struct snd_soc_jack *jack;
+        struct snd_soc_card *card = rtd->card;
+        int ret;
+
+
+        ret = snd_soc_dapm_add_routes(&card->dapm,
+                        kabylake_ssp1_map,
+                        ARRAY_SIZE(kabylake_ssp1_map));
+
+        /*
+         * Headset buttons map to the google Reference headset.
+         * These can be configured by userspace.
+         */
+        ret = snd_soc_card_jack_new(kabylake_audio_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->kabylake_headset, NULL, 0);
+        if (ret) {
+                dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
+                return ret;
+        }
+
+        jack = &ctx->kabylake_headset;
+        snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+        snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
+        snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
+        snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
+
+        da7219_aad_jack_det(component, &ctx->kabylake_headset);
+
+        ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC");
+        if (ret)
+                dev_err(rtd->dev, "SoC DMIC - Ignore suspend failed %d\n", ret);
+
+        return ret;
+}
+
+static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
+{
+        struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+        struct snd_soc_dai *dai = rtd->codec_dai;
+        struct kbl_hdmi_pcm *pcm;
+
+        pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+        if (!pcm)
+                return -ENOMEM;
+
+        pcm->device = device;
+        pcm->codec_dai = dai;
+
+        list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+        return 0;
+}
+
+static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
+{
+        return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB);
+}
+
+static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
+{
+        return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB);
+}
+
+static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
+{
+        return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB);
+}
+
+static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd)
+{
+        struct snd_soc_dapm_context *dapm;
+        struct snd_soc_component *component = rtd->cpu_dai->component;
+
+        dapm = snd_soc_component_get_dapm(component);
+        snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
+
+        return 0;
+}
+
+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 kbl_fe_startup(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+
+        /*
+         * On this platform for PCM device we support,
+         * 48Khz
+         * stereo
+         * 16 bit audio
+         */
+
+        runtime->hw.channels_max = DUAL_CHANNEL;
+        snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+                                           &constraints_channels);
+
+        runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+        snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16);
+
+        snd_pcm_hw_constraint_list(runtime, 0,
+                                SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+
+        return 0;
+}
+
+static const struct snd_soc_ops kabylake_da7219_fe_ops = {
+        .startup = kbl_fe_startup,
+};
+
+static int kabylake_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
+         */
+
+        if (params_channels(params) == 2)
+                channels->min = channels->max = 2;
+        else
+                channels->min = channels->max = 4;
+
+        return 0;
+}
+
+static int kabylake_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 struct snd_soc_ops kabylake_dmic_ops = {
+        .startup = kabylake_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 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 kabylake_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);
+}
+
+
+static struct snd_soc_ops skylaye_refcap_ops = {
+        .startup = kabylake_refcap_startup,
+};
+
+static struct snd_soc_codec_conf max98927_codec_conf[] = {
+
+        {
+                .dev_name = MAXIM_DEV0_NAME,
+                .name_prefix = "Right",
+        },
+
+        {
+                .dev_name = MAXIM_DEV1_NAME,
+                .name_prefix = "Left",
+        },
+};
+
+static struct snd_soc_dai_link_component ssp0_codec_components[] = {
+        { /* Left */
+                .name = MAXIM_DEV0_NAME,
+                .dai_name = MAX98927_CODEC_DAI,
+        },
+
+        {  /* For Right */
+                .name = MAXIM_DEV1_NAME,
+                .dai_name = MAX98927_CODEC_DAI,
+        },
+
+};
+
+/* kabylake digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link kabylake_dais[] = {
+        /* Front End DAI links */
+        [KBL_DPCM_AUDIO_PB] = {
+                .name = "Kbl Audio Port",
+                .stream_name = "Audio",
+                .cpu_dai_name = "System Pin",
+                .platform_name = "0000:00:1f.3",
+                .dynamic = 1,
+                .codec_name = "snd-soc-dummy",
+                .codec_dai_name = "snd-soc-dummy-dai",
+                .nonatomic = 1,
+                .init = kabylake_da7219_fe_init,
+                .trigger = {
+                        SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+                .dpcm_playback = 1,
+                .ops = &kabylake_da7219_fe_ops,
+        },
+        [KBL_DPCM_AUDIO_CP] = {
+                .name = "Kbl Audio Capture Port",
+                .stream_name = "Audio Record",
+                .cpu_dai_name = "System Pin",
+                .platform_name = "0000:00:1f.3",
+                .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,
+                .ops = &kabylake_da7219_fe_ops,
+        },
+        [KBL_DPCM_AUDIO_ECHO_REF_CP] = {
+                .name = "Kbl Audio Echo Reference cap",
+                .stream_name = "Echoreference Capture",
+                .cpu_dai_name = "Echoref Pin",
+                .codec_name = "snd-soc-dummy",
+                .codec_dai_name = "snd-soc-dummy-dai",
+                .platform_name = "0000:00:1f.3",
+                .init = NULL,
+                .capture_only = 1,
+                .nonatomic = 1,
+        },
+        [KBL_DPCM_AUDIO_REF_CP] = {
+                .name = "Kbl Audio Reference cap",
+                .stream_name = "Wake on Voice",
+                .cpu_dai_name = "Reference Pin",
+                .codec_name = "snd-soc-dummy",
+                .codec_dai_name = "snd-soc-dummy-dai",
+                .platform_name = "0000:00:1f.3",
+                .init = NULL,
+                .dpcm_capture = 1,
+                .nonatomic = 1,
+                .dynamic = 1,
+                .ops = &skylaye_refcap_ops,
+        },
+        [KBL_DPCM_AUDIO_DMIC_CP] = {
+                .name = "Kbl 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 = "0000:00:1f.3",
+                .init = NULL,
+                .dpcm_capture = 1,
+                .nonatomic = 1,
+                .dynamic = 1,
+                .ops = &kabylake_dmic_ops,
+        },
+        [KBL_DPCM_AUDIO_HDMI1_PB] = {
+                .name = "Kbl HDMI Port1",
+                .stream_name = "Hdmi1",
+                .cpu_dai_name = "HDMI1 Pin",
+                .codec_name = "snd-soc-dummy",
+                .codec_dai_name = "snd-soc-dummy-dai",
+                .platform_name = "0000:00:1f.3",
+                .dpcm_playback = 1,
+                .init = NULL,
+                .trigger = {
+                        SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+                .nonatomic = 1,
+                .dynamic = 1,
+        },
+        [KBL_DPCM_AUDIO_HDMI2_PB] = {
+                .name = "Kbl HDMI Port2",
+                .stream_name = "Hdmi2",
+                .cpu_dai_name = "HDMI2 Pin",
+                .codec_name = "snd-soc-dummy",
+                .codec_dai_name = "snd-soc-dummy-dai",
+                .platform_name = "0000:00:1f.3",
+                .dpcm_playback = 1,
+                .init = NULL,
+                .trigger = {
+                        SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+                .nonatomic = 1,
+                .dynamic = 1,
+        },
+        [KBL_DPCM_AUDIO_HDMI3_PB] = {
+                .name = "Kbl HDMI Port3",
+                .stream_name = "Hdmi3",
+                .cpu_dai_name = "HDMI3 Pin",
+                .codec_name = "snd-soc-dummy",
+                .codec_dai_name = "snd-soc-dummy-dai",
+                .platform_name = "0000:00:1f.3",
+                .trigger = {
+                        SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+                .dpcm_playback = 1,
+                .init = NULL,
+                .nonatomic = 1,
+                .dynamic = 1,
+        },
+        [KBL_DPCM_AUDIO_HS_PB] = {
+                .name = "Kbl Audio Headset Playback",
+                .stream_name = "Headset Audio",
+                .cpu_dai_name = "System Pin2",
+                .codec_name = "snd-soc-dummy",
+                .codec_dai_name = "snd-soc-dummy-dai",
+                .platform_name = "0000:00:1f.3",
+                .dpcm_playback = 1,
+                .nonatomic = 1,
+                .dynamic = 1,
+                .init = kabylake_da7219_fe_init,
+                .trigger = {
+                        SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+                .ops = &kabylake_da7219_fe_ops,
+
+        },
+
+        /* Back End DAI links */
+        {
+                /* SSP0 - Codec */
+                .name = "SSP0-Codec",
+                .id = 0,
+                .cpu_dai_name = "SSP0 Pin",
+                .platform_name = "0000:00:1f.3",
+                .no_pcm = 1,
+                .codecs = ssp0_codec_components,
+                .num_codecs = ARRAY_SIZE(ssp0_codec_components),
+                .dai_fmt = SND_SOC_DAIFMT_DSP_B |
+                        SND_SOC_DAIFMT_NB_NF |
+                        SND_SOC_DAIFMT_CBS_CFS,
+                .dpcm_playback = 1,
+                .dpcm_capture = 1,
+                .ignore_pmdown_time = 1,
+                .be_hw_params_fixup = kabylake_ssp_fixup,
+                .ops = &kabylake_ssp0_ops,
+        },
+        {
+                /* SSP1 - Codec */
+                .name = "SSP1-Codec",
+                .id = 1,
+                .cpu_dai_name = "SSP1 Pin",
+                .platform_name = "0000:00:1f.3",
+                .no_pcm = 1,
+                .codec_name = "i2c-DLGS7219:00",
+                .codec_dai_name = KBL_DIALOG_CODEC_DAI,
+                .init = kabylake_da7219_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 = kabylake_ssp_fixup,
+                .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 = "0000:00:1f.3",
+                .be_hw_params_fixup = kabylake_dmic_fixup,
+                .ignore_suspend = 1,
+                .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 = "0000:00:1f.3",
+                .dpcm_playback = 1,
+                .init = kabylake_hdmi1_init,
+                .no_pcm = 1,
+        },
+        {
+                .name = "iDisp2",
+                .id = 4,
+                .cpu_dai_name = "iDisp2 Pin",
+                .codec_name = "ehdaudio0D2",
+                .codec_dai_name = "intel-hdmi-hifi2",
+                .platform_name = "0000:00:1f.3",
+                .init = kabylake_hdmi2_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 = "0000:00:1f.3",
+                .init = kabylake_hdmi3_init,
+                .dpcm_playback = 1,
+                .no_pcm = 1,
+        },
+};
+
+/* kabylake digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link kabylake_max98927_dais[] = {
+        /* Front End DAI links */
+        [KBL_DPCM_AUDIO_PB] = {
+                .name = "Kbl Audio Port",
+                .stream_name = "Audio",
+                .cpu_dai_name = "System Pin",
+                .platform_name = "0000:00:1f.3",
+                .dynamic = 1,
+                .codec_name = "snd-soc-dummy",
+                .codec_dai_name = "snd-soc-dummy-dai",
+                .nonatomic = 1,
+                .init = kabylake_da7219_fe_init,
+                .trigger = {
+                        SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+                .dpcm_playback = 1,
+                .ops = &kabylake_da7219_fe_ops,
+        },
+        [KBL_DPCM_AUDIO_CP] = {
+                .name = "Kbl Audio Capture Port",
+                .stream_name = "Audio Record",
+                .cpu_dai_name = "System Pin",
+                .platform_name = "0000:00:1f.3",
+                .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,
+                .ops = &kabylake_da7219_fe_ops,
+        },
+        [KBL_DPCM_AUDIO_ECHO_REF_CP] = {
+                .name = "Kbl Audio Echo Reference cap",
+                .stream_name = "Echoreference Capture",
+                .cpu_dai_name = "Echoref Pin",
+                .codec_name = "snd-soc-dummy",
+                .codec_dai_name = "snd-soc-dummy-dai",
+                .platform_name = "0000:00:1f.3",
+                .init = NULL,
+                .capture_only = 1,
+                .nonatomic = 1,
+        },
+        [KBL_DPCM_AUDIO_REF_CP] = {
+                .name = "Kbl Audio Reference cap",
+                .stream_name = "Wake on Voice",
+                .cpu_dai_name = "Reference Pin",
+                .codec_name = "snd-soc-dummy",
+                .codec_dai_name = "snd-soc-dummy-dai",
+                .platform_name = "0000:00:1f.3",
+                .init = NULL,
+                .dpcm_capture = 1,
+                .nonatomic = 1,
+                .dynamic = 1,
+                .ops = &skylaye_refcap_ops,
+        },
+        [KBL_DPCM_AUDIO_DMIC_CP] = {
+                .name = "Kbl 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 = "0000:00:1f.3",
+                .init = NULL,
+                .dpcm_capture = 1,
+                .nonatomic = 1,
+                .dynamic = 1,
+                .ops = &kabylake_dmic_ops,
+        },
+        [KBL_DPCM_AUDIO_HDMI1_PB] = {
+                .name = "Kbl HDMI Port1",
+                .stream_name = "Hdmi1",
+                .cpu_dai_name = "HDMI1 Pin",
+                .codec_name = "snd-soc-dummy",
+                .codec_dai_name = "snd-soc-dummy-dai",
+                .platform_name = "0000:00:1f.3",
+                .dpcm_playback = 1,
+                .init = NULL,
+                .trigger = {
+                        SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+                .nonatomic = 1,
+                .dynamic = 1,
+        },
+        [KBL_DPCM_AUDIO_HDMI2_PB] = {
+                .name = "Kbl HDMI Port2",
+                .stream_name = "Hdmi2",
+                .cpu_dai_name = "HDMI2 Pin",
+                .codec_name = "snd-soc-dummy",
+                .codec_dai_name = "snd-soc-dummy-dai",
+                .platform_name = "0000:00:1f.3",
+                .dpcm_playback = 1,
+                .init = NULL,
+                .trigger = {
+                        SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+                .nonatomic = 1,
+                .dynamic = 1,
+        },
+        [KBL_DPCM_AUDIO_HDMI3_PB] = {
+                .name = "Kbl HDMI Port3",
+                .stream_name = "Hdmi3",
+                .cpu_dai_name = "HDMI3 Pin",
+                .codec_name = "snd-soc-dummy",
+                .codec_dai_name = "snd-soc-dummy-dai",
+                .platform_name = "0000:00:1f.3",
+                .trigger = {
+                        SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+                .dpcm_playback = 1,
+                .init = NULL,
+                .nonatomic = 1,
+                .dynamic = 1,
+        },
+
+        /* Back End DAI links */
+        {
+                /* SSP0 - Codec */
+                .name = "SSP0-Codec",
+                .id = 0,
+                .cpu_dai_name = "SSP0 Pin",
+                .platform_name = "0000:00:1f.3",
+                .no_pcm = 1,
+                .codecs = ssp0_codec_components,
+                .num_codecs = ARRAY_SIZE(ssp0_codec_components),
+                .dai_fmt = SND_SOC_DAIFMT_DSP_B |
+                        SND_SOC_DAIFMT_NB_NF |
+                        SND_SOC_DAIFMT_CBS_CFS,
+                .dpcm_playback = 1,
+                .dpcm_capture = 1,
+                .ignore_pmdown_time = 1,
+                .be_hw_params_fixup = kabylake_ssp_fixup,
+                .ops = &kabylake_ssp0_ops,
+        },
+        {
+                .name = "dmic01",
+                .id = 1,
+                .cpu_dai_name = "DMIC01 Pin",
+                .codec_name = "dmic-codec",
+                .codec_dai_name = "dmic-hifi",
+                .platform_name = "0000:00:1f.3",
+                .be_hw_params_fixup = kabylake_dmic_fixup,
+                .ignore_suspend = 1,
+                .dpcm_capture = 1,
+                .no_pcm = 1,
+        },
+        {
+                .name = "iDisp1",
+                .id = 2,
+                .cpu_dai_name = "iDisp1 Pin",
+                .codec_name = "ehdaudio0D2",
+                .codec_dai_name = "intel-hdmi-hifi1",
+                .platform_name = "0000:00:1f.3",
+                .dpcm_playback = 1,
+                .init = kabylake_hdmi1_init,
+                .no_pcm = 1,
+        },
+        {
+                .name = "iDisp2",
+                .id = 3,
+                .cpu_dai_name = "iDisp2 Pin",
+                .codec_name = "ehdaudio0D2",
+                .codec_dai_name = "intel-hdmi-hifi2",
+                .platform_name = "0000:00:1f.3",
+                .init = kabylake_hdmi2_init,
+                .dpcm_playback = 1,
+                .no_pcm = 1,
+        },
+        {
+                .name = "iDisp3",
+                .id = 4,
+                .cpu_dai_name = "iDisp3 Pin",
+                .codec_name = "ehdaudio0D2",
+                .codec_dai_name = "intel-hdmi-hifi3",
+                .platform_name = "0000:00:1f.3",
+                .init = kabylake_hdmi3_init,
+                .dpcm_playback = 1,
+                .no_pcm = 1,
+        },
+};
+
+static int kabylake_card_late_probe(struct snd_soc_card *card)
+{
+        struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card);
+        struct kbl_hdmi_pcm *pcm;
+        struct snd_soc_component *component = NULL;
+        int err, i = 0;
+        char jack_name[NAME_SIZE];
+
+        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, &kabylake_hdmi[i],
+                                        NULL, 0);
+
+                if (err)
+                        return err;
+
+                err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
+                                                &kabylake_hdmi[i]);
+                if (err < 0)
+                        return err;
+
+                i++;
+        }
+
+        if (!component)
+                return -EINVAL;
+
+        return hdac_hdmi_jack_port_init(component, &card->dapm);
+
+        return 0;
+}
+
+/* kabylake audio machine driver for SPT + DA7219 */
+static struct snd_soc_card kbl_audio_card_da7219_m98927 = {
+        .name = "kblda7219m98927",
+        .owner = THIS_MODULE,
+        .dai_link = kabylake_dais,
+        .num_links = ARRAY_SIZE(kabylake_dais),
+        .controls = kabylake_controls,
+        .num_controls = ARRAY_SIZE(kabylake_controls),
+        .dapm_widgets = kabylake_widgets,
+        .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets),
+        .dapm_routes = kabylake_map,
+        .num_dapm_routes = ARRAY_SIZE(kabylake_map),
+        .codec_conf = max98927_codec_conf,
+        .num_configs = ARRAY_SIZE(max98927_codec_conf),
+        .fully_routed = true,
+        .late_probe = kabylake_card_late_probe,
+};
+
+/* kabylake audio machine driver for Maxim98927 */
+static struct snd_soc_card kbl_audio_card_max98927 = {
+        .name = "kblmax98927",
+        .owner = THIS_MODULE,
+        .dai_link = kabylake_max98927_dais,
+        .num_links = ARRAY_SIZE(kabylake_max98927_dais),
+        .controls = kabylake_controls,
+        .num_controls = ARRAY_SIZE(kabylake_controls),
+        .dapm_widgets = kabylake_widgets,
+        .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets),
+        .dapm_routes = kabylake_map,
+        .num_dapm_routes = ARRAY_SIZE(kabylake_map),
+        .codec_conf = max98927_codec_conf,
+        .num_configs = ARRAY_SIZE(max98927_codec_conf),
+        .fully_routed = true,
+        .late_probe = kabylake_card_late_probe,
+};
+
+static int kabylake_audio_probe(struct platform_device *pdev)
+{
+        struct kbl_codec_private *ctx;
+
+        ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+        if (!ctx)
+                return -ENOMEM;
+
+        INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
+        kabylake_audio_card =
+                (struct snd_soc_card *)pdev->id_entry->driver_data;
+
+        kabylake_audio_card->dev = &pdev->dev;
+        snd_soc_card_set_drvdata(kabylake_audio_card, ctx);
+
+        return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card);
+}
+
+static const struct platform_device_id kbl_board_ids[] = {
+        {
+                .name = "kbl_da7219_max98927",
+                .driver_data =
+                        (kernel_ulong_t)&kbl_audio_card_da7219_m98927,
+        },
+        {
+                .name = "kbl_max98927",
+                .driver_data =
+                        (kernel_ulong_t)&kbl_audio_card_max98927,
+        },
+        { }
+};
+
+static struct platform_driver kabylake_audio = {
+        .probe = kabylake_audio_probe,
+        .driver = {
+                .name = "kbl_da7219_max98927",
+                .pm = &snd_soc_pm_ops,
+        },
+        .id_table = kbl_board_ids,
+};
+
+module_platform_driver(kabylake_audio)
+
+/* Module information */
+MODULE_DESCRIPTION("Audio KabyLake Machine driver for MAX98927 & DA7219");
+MODULE_AUTHOR("Mac Chiang <mac.chiang@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:kbl_da7219_max98927");
+MODULE_ALIAS("platform:kbl_max98927");
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 21a6490746a6..99e1320c485f 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -488,11 +488,10 @@ static int kabylake_ssp0_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;
         int ret = 0, j;
 
-        for (j = 0; j < rtd->num_codecs; j++) {
-                struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
-
+        for_each_rtd_codec_dai(rtd, j, codec_dai) {
                 if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
                         /*
                          * Use channel 4 and 5 for the first amp
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index a892b37eab7c..a737c915d46a 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -353,11 +353,10 @@ static int kabylake_ssp0_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;
         int ret = 0, j;
 
-        for (j = 0; j < rtd->num_codecs; j++) {
-                struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
-
+        for_each_rtd_codec_dai(rtd, j, codec_dai) {
                 if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) {
                         ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16);
                         if (ret < 0) {
diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c
new file mode 100644
index 000000000000..3fdbf239da74
--- /dev/null
+++ b/sound/soc/intel/boards/skl_hda_dsp_common.c
@@ -0,0 +1,127 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2015-18 Intel Corporation.
+
+/*
+ * Common functions used in different Intel machine drivers
+ */
+#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 "../../codecs/hdac_hdmi.h"
+#include "../skylake/skl.h"
+#include "skl_hda_dsp_common.h"
+
+#define NAME_SIZE       32
+
+int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device)
+{
+        struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
+        struct skl_hda_hdmi_pcm *pcm;
+        char dai_name[NAME_SIZE];
+
+        pcm = devm_kzalloc(card->dev, sizeof(*pcm), GFP_KERNEL);
+        if (!pcm)
+                return -ENOMEM;
+
+        snprintf(dai_name, sizeof(dai_name), "intel-hdmi-hifi%d",
+                 ctx->dai_index);
+        pcm->codec_dai = snd_soc_card_get_codec_dai(card, dai_name);
+        if (!pcm->codec_dai)
+                return -EINVAL;
+
+        pcm->device = device;
+        list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+        return 0;
+}
+
+/* skl_hda_digital audio interface glue - connects codec <--> CPU */
+struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = {
+        /* Back End DAI links */
+        {
+                .name = "iDisp1",
+                .id = 1,
+                .cpu_dai_name = "iDisp1 Pin",
+                .codec_name = "ehdaudio0D2",
+                .codec_dai_name = "intel-hdmi-hifi1",
+                .dpcm_playback = 1,
+                .no_pcm = 1,
+        },
+        {
+                .name = "iDisp2",
+                .id = 2,
+                .cpu_dai_name = "iDisp2 Pin",
+                .codec_name = "ehdaudio0D2",
+                .codec_dai_name = "intel-hdmi-hifi2",
+                .dpcm_playback = 1,
+                .no_pcm = 1,
+        },
+        {
+                .name = "iDisp3",
+                .id = 3,
+                .cpu_dai_name = "iDisp3 Pin",
+                .codec_name = "ehdaudio0D2",
+                .codec_dai_name = "intel-hdmi-hifi3",
+                .dpcm_playback = 1,
+                .no_pcm = 1,
+        },
+        {
+                .name = "Analog Playback and Capture",
+                .id = 4,
+                .cpu_dai_name = "Analog CPU DAI",
+                .codec_name = "ehdaudio0D0",
+                .codec_dai_name = "Analog Codec DAI",
+                .platform_name = "0000:00:1f.3",
+                .dpcm_playback = 1,
+                .dpcm_capture = 1,
+                .init = NULL,
+                .no_pcm = 1,
+        },
+        {
+                .name = "Digital Playback and Capture",
+                .id = 5,
+                .cpu_dai_name = "Digital CPU DAI",
+                .codec_name = "ehdaudio0D0",
+                .codec_dai_name = "Digital Codec DAI",
+                .platform_name = "0000:00:1f.3",
+                .dpcm_playback = 1,
+                .dpcm_capture = 1,
+                .init = NULL,
+                .no_pcm = 1,
+        },
+};
+
+int skl_hda_hdmi_jack_init(struct snd_soc_card *card)
+{
+        struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
+        struct snd_soc_component *component = NULL;
+        struct skl_hda_hdmi_pcm *pcm;
+        char jack_name[NAME_SIZE];
+        int err;
+
+        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, &pcm->hdmi_jack,
+                                            NULL, 0);
+
+                if (err)
+                        return err;
+
+                err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
+                                          &pcm->hdmi_jack);
+                if (err < 0)
+                        return err;
+        }
+
+        if (!component)
+                return -EINVAL;
+
+        return hdac_hdmi_jack_port_init(component, &card->dapm);
+}
diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h
new file mode 100644
index 000000000000..87c50aff56cd
--- /dev/null
+++ b/sound/soc/intel/boards/skl_hda_dsp_common.h
@@ -0,0 +1,38 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright(c) 2015-18 Intel Corporation.
+ */
+
+/*
+ * This file defines data structures used in Machine Driver for Intel
+ * platforms with HDA Codecs.
+ */
+
+#ifndef __SOUND_SOC_HDA_DSP_COMMON_H
+#define __SOUND_SOC_HDA_DSP_COMMON_H
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+
+#define HDA_DSP_MAX_BE_DAI_LINKS 5
+
+struct skl_hda_hdmi_pcm {
+        struct list_head head;
+        struct snd_soc_dai *codec_dai;
+        struct snd_soc_jack hdmi_jack;
+        int device;
+};
+
+struct skl_hda_private {
+        struct list_head hdmi_pcm_list;
+        int pcm_count;
+        int dai_index;
+        const char *platform_name;
+};
+
+extern struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS];
+int skl_hda_hdmi_jack_init(struct snd_soc_card *card);
+int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device);
+
+#endif /* __SOUND_SOC_HDA_DSP_COMMON_H */
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
new file mode 100644
index 000000000000..b415dd4c85f5
--- /dev/null
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -0,0 +1,183 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2015-18 Intel Corporation.
+
+/*
+ * Machine Driver for SKL+ platforms with DSP and iDisp, HDA Codecs
+ */
+
+#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 "../../codecs/hdac_hdmi.h"
+#include "../skylake/skl.h"
+#include "skl_hda_dsp_common.h"
+
+static const struct snd_soc_dapm_widget skl_hda_widgets[] = {
+        SND_SOC_DAPM_HP("Analog Out", NULL),
+        SND_SOC_DAPM_MIC("Analog In", NULL),
+        SND_SOC_DAPM_HP("Alt Analog Out", NULL),
+        SND_SOC_DAPM_MIC("Alt Analog In", NULL),
+        SND_SOC_DAPM_SPK("Digital Out", NULL),
+        SND_SOC_DAPM_MIC("Digital In", NULL),
+};
+
+static const struct snd_soc_dapm_route skl_hda_map[] = {
+        { "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"},
+
+        { "Analog Out", NULL, "Codec Output Pin1" },
+        { "Digital Out", NULL, "Codec Output Pin2" },
+        { "Alt Analog Out", NULL, "Codec Output Pin3" },
+
+        { "Codec Input Pin1", NULL, "Analog In" },
+        { "Codec Input Pin2", NULL, "Digital In" },
+        { "Codec Input Pin3", NULL, "Alt Analog In" },
+
+        /* CODEC BE connections */
+        { "Analog Codec Playback", NULL, "Analog CPU Playback" },
+        { "Analog CPU Playback", NULL, "codec0_out" },
+        { "Digital Codec Playback", NULL, "Digital CPU Playback" },
+        { "Digital CPU Playback", NULL, "codec1_out" },
+        { "Alt Analog Codec Playback", NULL, "Alt Analog CPU Playback" },
+        { "Alt Analog CPU Playback", NULL, "codec2_out" },
+
+        { "codec0_in", NULL, "Analog CPU Capture" },
+        { "Analog CPU Capture", NULL, "Analog Codec Capture" },
+        { "codec1_in", NULL, "Digital CPU Capture" },
+        { "Digital CPU Capture", NULL, "Digital Codec Capture" },
+        { "codec2_in", NULL, "Alt Analog CPU Capture" },
+        { "Alt Analog CPU Capture", NULL, "Alt Analog Codec Capture" },
+};
+
+static int skl_hda_card_late_probe(struct snd_soc_card *card)
+{
+        return skl_hda_hdmi_jack_init(card);
+}
+
+static int
+skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link)
+{
+        struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
+        int ret = 0;
+
+        dev_dbg(card->dev, "%s: dai link name - %s\n", __func__, link->name);
+        link->platform_name = ctx->platform_name;
+        link->nonatomic = 1;
+
+        if (strstr(link->name, "HDMI")) {
+                ret = skl_hda_hdmi_add_pcm(card, ctx->pcm_count);
+
+                if (ret < 0)
+                        return ret;
+
+                ctx->dai_index++;
+        }
+
+        ctx->pcm_count++;
+        return ret;
+}
+
+static struct snd_soc_card hda_soc_card = {
+        .name = "skl_hda_card",
+        .owner = THIS_MODULE,
+        .dai_link = skl_hda_be_dai_links,
+        .dapm_widgets = skl_hda_widgets,
+        .dapm_routes = skl_hda_map,
+        .add_dai_link = skl_hda_add_dai_link,
+        .fully_routed = true,
+        .late_probe = skl_hda_card_late_probe,
+};
+
+#define IDISP_DAI_COUNT         3
+/* there are two routes per iDisp output */
+#define IDISP_ROUTE_COUNT       (IDISP_DAI_COUNT * 2)
+#define IDISP_CODEC_MASK        0x4
+
+static int skl_hda_fill_card_info(struct skl_machine_pdata *pdata)
+{
+        struct snd_soc_card *card = &hda_soc_card;
+        struct snd_soc_dai_link *dai_link;
+        u32 codec_count, codec_mask;
+        int i, num_links, num_route;
+
+        codec_mask = pdata->codec_mask;
+        codec_count = hweight_long(codec_mask);
+
+        if (codec_count == 1 && pdata->codec_mask & IDISP_CODEC_MASK) {
+                num_links = IDISP_DAI_COUNT;
+                num_route = IDISP_ROUTE_COUNT;
+        } else if (codec_count == 2 && codec_mask & IDISP_CODEC_MASK) {
+                num_links = ARRAY_SIZE(skl_hda_be_dai_links);
+                num_route = ARRAY_SIZE(skl_hda_map),
+                card->dapm_widgets = skl_hda_widgets;
+                card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets);
+        } else {
+                return -EINVAL;
+        }
+
+        card->num_links = num_links;
+        card->num_dapm_routes = num_route;
+
+        for_each_card_prelinks(card, i, dai_link)
+                dai_link->platform_name = pdata->platform;
+
+        return 0;
+}
+
+static int skl_hda_audio_probe(struct platform_device *pdev)
+{
+        struct skl_machine_pdata *pdata;
+        struct skl_hda_private *ctx;
+        int ret;
+
+        dev_dbg(&pdev->dev, "%s: entry\n", __func__);
+
+        ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+        if (!ctx)
+                return -ENOMEM;
+
+        INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
+        pdata = dev_get_drvdata(&pdev->dev);
+        if (!pdata)
+                return -EINVAL;
+
+        ret = skl_hda_fill_card_info(pdata);
+        if (ret < 0) {
+                dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n");
+                return ret;
+        }
+
+        ctx->pcm_count = hda_soc_card.num_links;
+        ctx->dai_index = 1; /* hdmi codec dai name starts from index 1 */
+        ctx->platform_name = pdata->platform;
+
+        hda_soc_card.dev = &pdev->dev;
+        snd_soc_card_set_drvdata(&hda_soc_card, ctx);
+
+        return devm_snd_soc_register_card(&pdev->dev, &hda_soc_card);
+}
+
+static struct platform_driver skl_hda_audio = {
+        .probe = skl_hda_audio_probe,
+        .driver = {
+                .name = "skl_hda_dsp_generic",
+                .pm = &snd_soc_pm_ops,
+        },
+};
+
+module_platform_driver(skl_hda_audio)
+
+/* Module information */
+MODULE_DESCRIPTION("SKL/KBL/BXT/APL HDA Generic Machine driver");
+MODULE_AUTHOR("Rakesh Ughreja <rakesh.a.ughreja@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:skl_hda_dsp_generic");
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index 915a34cdc8ac..c1f50a079d34 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -7,7 +7,8 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m
         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
+        soc-acpi-intel-cnl-match.o \
+        soc-acpi-intel-hda-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-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
index 4daa8a4f0c0c..097dc06377ba 100644
--- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
@@ -34,6 +34,13 @@ static const struct dmi_system_id byt_table[] = {
                 .callback = byt_thinkpad10_quirk_cb,
                 .matches = {
                         DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+                        DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 8"),
+                },
+        },
+        {
+                .callback = byt_thinkpad10_quirk_cb,
+                .matches = {
+                        DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
                         DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 10"),
                 },
         },
diff --git a/sound/soc/intel/common/soc-acpi-intel-hda-match.c b/sound/soc/intel/common/soc-acpi-intel-hda-match.c
new file mode 100644
index 000000000000..533c1064f84b
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-hda-match.c
@@ -0,0 +1,40 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2018, Intel Corporation.
+
+/*
+ * soc-apci-intel-hda-match.c - tables and support for HDA+ACPI enumeration.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+#include "../skylake/skl.h"
+
+static struct skl_machine_pdata hda_pdata = {
+        .use_tplg_pcm = true,
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[] = {
+        {
+                /* .id is not used in this file */
+                .drv_name = "skl_hda_dsp_generic",
+
+                /* .fw_filename is dynamically set in skylake driver */
+
+                /* .sof_fw_filename is dynamically set in sof/intel driver */
+
+                .sof_tplg_filename = "intel/sof-hda-generic.tplg",
+
+                /*
+                 * .machine_quirk and .quirk_data are not used here but
+                 * can be used if we need a more complicated machine driver
+                 * combining HDA+other device (e.g. DMIC).
+                 */
+                .pdata = &hda_pdata,
+        },
+        {},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_hda_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
index 0ee173ca437d..a317b7790fce 100644
--- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
@@ -32,6 +32,11 @@ static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = {
         .codecs = {"MX98357A"}
 };
 
+static struct snd_soc_acpi_codecs kbl_7219_98927_codecs = {
+        .num_codecs = 1,
+        .codecs = {"MX98927"}
+};
+
 struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = {
         {
                 .id = "INT343A",
@@ -83,6 +88,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = {
                 .quirk_data = &kbl_7219_98357_codecs,
                 .pdata = &skl_dmic_data,
         },
+        {
+                .id = "DLGS7219",
+                .drv_name = "kbl_da7219_max98927",
+                .fw_filename = "intel/dsp_fw_kbl.bin",
+                .machine_quirk = snd_soc_acpi_codec_list,
+                .quirk_data = &kbl_7219_98927_codecs,
+                .pdata = &skl_dmic_data
+        },
         {},
 };
 EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines);
diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c
index 11041aedea31..1e067504b604 100644
--- a/sound/soc/intel/common/sst-firmware.c
+++ b/sound/soc/intel/common/sst-firmware.c
@@ -355,7 +355,7 @@ struct sst_fw *sst_fw_new(struct sst_dsp *dsp,
 
         /* allocate DMA buffer to store FW data */
         sst_fw->dma_buf = dma_alloc_coherent(dsp->dma_dev, sst_fw->size,
-                                &sst_fw->dmable_fw_paddr, GFP_DMA | GFP_KERNEL);
+                                &sst_fw->dmable_fw_paddr, GFP_KERNEL);
         if (!sst_fw->dma_buf) {
                 dev_err(dsp->dev, "error: DMA alloc failed\n");
                 kfree(sst_fw);
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 823e39103edd..557f80c0bfe5 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -32,6 +32,7 @@
 #define HDA_MONO 1
 #define HDA_STEREO 2
 #define HDA_QUAD 4
+#define HDA_MAX 8
 
 static const struct snd_pcm_hardware azx_pcm_hw = {
         .info =                 (SNDRV_PCM_INFO_MMAP |
@@ -494,6 +495,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
                                                         stream->lpib);
                         snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
                 }
+                /* fall through */
 
         case SNDRV_PCM_TRIGGER_START:
         case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
@@ -569,7 +571,10 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream,
         stream_tag = hdac_stream(link_dev)->stream_tag;
 
         /* set the stream tag in the codec dai dma params  */
-        snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0);
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0);
+        else
+                snd_soc_dai_set_tdm_slot(codec_dai, 0, stream_tag, 0, 0);
 
         p_params.s_fmt = snd_pcm_format_width(params_format(params));
         p_params.ch = params_channels(params);
@@ -995,21 +1000,63 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
         },
 },
 {
-        .name = "HD-Codec Pin",
+        .name = "Analog CPU DAI",
         .ops = &skl_link_dai_ops,
         .playback = {
-                .stream_name = "HD-Codec Tx",
-                .channels_min = HDA_STEREO,
-                .channels_max = HDA_STEREO,
-                .rates = SNDRV_PCM_RATE_48000,
-                .formats = SNDRV_PCM_FMTBIT_S16_LE,
+                .stream_name = "Analog CPU Playback",
+                .channels_min = HDA_MONO,
+                .channels_max = HDA_MAX,
+                .rates = SNDRV_PCM_RATE_8000_192000,
+                .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+                        SNDRV_PCM_FMTBIT_S32_LE,
         },
         .capture = {
-                .stream_name = "HD-Codec Rx",
-                .channels_min = HDA_STEREO,
-                .channels_max = HDA_STEREO,
-                .rates = SNDRV_PCM_RATE_48000,
-                .formats = SNDRV_PCM_FMTBIT_S16_LE,
+                .stream_name = "Analog CPU Capture",
+                .channels_min = HDA_MONO,
+                .channels_max = HDA_MAX,
+                .rates = SNDRV_PCM_RATE_8000_192000,
+                .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+                        SNDRV_PCM_FMTBIT_S32_LE,
+        },
+},
+{
+        .name = "Alt Analog CPU DAI",
+        .ops = &skl_link_dai_ops,
+        .playback = {
+                .stream_name = "Alt Analog CPU Playback",
+                .channels_min = HDA_MONO,
+                .channels_max = HDA_MAX,
+                .rates = SNDRV_PCM_RATE_8000_192000,
+                .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+                        SNDRV_PCM_FMTBIT_S32_LE,
+        },
+        .capture = {
+                .stream_name = "Alt Analog CPU Capture",
+                .channels_min = HDA_MONO,
+                .channels_max = HDA_MAX,
+                .rates = SNDRV_PCM_RATE_8000_192000,
+                .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+                        SNDRV_PCM_FMTBIT_S32_LE,
+        },
+},
+{
+        .name = "Digital CPU DAI",
+        .ops = &skl_link_dai_ops,
+        .playback = {
+                .stream_name = "Digital CPU Playback",
+                .channels_min = HDA_MONO,
+                .channels_max = HDA_MAX,
+                .rates = SNDRV_PCM_RATE_8000_192000,
+                .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+                        SNDRV_PCM_FMTBIT_S32_LE,
+        },
+        .capture = {
+                .stream_name = "Digital CPU Capture",
+                .channels_min = HDA_MONO,
+                .channels_max = HDA_MAX,
+                .rates = SNDRV_PCM_RATE_8000_192000,
+                .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+                        SNDRV_PCM_FMTBIT_S32_LE,
         },
 },
 };
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 2620d77729c5..cf8848b779dc 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -898,11 +898,10 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w,
                         bc = (struct skl_algo_data *)sb->dobj.private;
 
                         if (bc->set_params == SKL_PARAM_BIND) {
-                                params = kzalloc(bc->max, GFP_KERNEL);
+                                params = kmemdup(bc->params, bc->max, GFP_KERNEL);
                                 if (!params)
                                         return -ENOMEM;
 
-                                memcpy(params, bc->params, bc->max);
                                 skl_fill_sink_instance_id(ctx, params, bc->max,
                                                                 mconfig);
 
@@ -2461,6 +2460,7 @@ static int skl_tplg_get_token(struct device *dev,
 
         case SKL_TKN_U8_CORE_ID:
                 mconfig->core_id = tkn_elem->value;
+                break;
 
         case SKL_TKN_U8_MOD_TYPE:
                 mconfig->m_type = tkn_elem->value;
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 1d17be0f78a0..29225623b4b4 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -33,9 +33,11 @@
 #include <sound/hda_register.h>
 #include <sound/hdaudio.h>
 #include <sound/hda_i915.h>
+#include <sound/hda_codec.h>
 #include "skl.h"
 #include "skl-sst-dsp.h"
 #include "skl-sst-ipc.h"
+#include "../../../soc/codecs/hdac_hda.h"
 
 /*
  * initialize the PCI registers
@@ -472,6 +474,25 @@ static struct skl_ssp_clk skl_ssp_clks[] = {
                                                 {.name = "ssp5_sclkfs"},
 };
 
+static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl *skl,
+                                        struct snd_soc_acpi_mach *machines)
+{
+        struct hdac_bus *bus = skl_to_bus(skl);
+        struct snd_soc_acpi_mach *mach;
+
+        /* check if we have any codecs detected on bus */
+        if (bus->codec_mask == 0)
+                return NULL;
+
+        /* point to common table */
+        mach = snd_soc_acpi_intel_hda_machines;
+
+        /* all entries in the machine table use the same firmware */
+        mach->fw_filename = machines->fw_filename;
+
+        return mach;
+}
+
 static int skl_find_machine(struct skl *skl, void *driver_data)
 {
         struct hdac_bus *bus = skl_to_bus(skl);
@@ -479,9 +500,13 @@ static int skl_find_machine(struct skl *skl, void *driver_data)
         struct skl_machine_pdata *pdata;
 
         mach = snd_soc_acpi_find_machine(mach);
-        if (mach == NULL) {
-                dev_err(bus->dev, "No matching machine driver found\n");
-                return -ENODEV;
+        if (!mach) {
+                dev_dbg(bus->dev, "No matching I2S machine driver found\n");
+                mach = skl_find_hda_machine(skl, driver_data);
+                if (!mach) {
+                        dev_err(bus->dev, "No matching machine driver found\n");
+                        return -ENODEV;
+                }
         }
 
         skl->mach = mach;
@@ -498,8 +523,9 @@ static int skl_find_machine(struct skl *skl, void *driver_data)
 
 static int skl_machine_device_register(struct skl *skl)
 {
-        struct hdac_bus *bus = skl_to_bus(skl);
         struct snd_soc_acpi_mach *mach = skl->mach;
+        struct hdac_bus *bus = skl_to_bus(skl);
+        struct skl_machine_pdata *pdata;
         struct platform_device *pdev;
         int ret;
 
@@ -516,8 +542,12 @@ static int skl_machine_device_register(struct skl *skl)
                 return -EIO;
         }
 
-        if (mach->pdata)
+        if (mach->pdata) {
+                pdata = (struct skl_machine_pdata *)mach->pdata;
+                pdata->platform = dev_name(bus->dev);
+                pdata->codec_mask = bus->codec_mask;
                 dev_set_drvdata(&pdev->dev, mach->pdata);
+        }
 
         skl->i2s_dev = pdev;
 
@@ -628,6 +658,24 @@ static void skl_clock_device_unregister(struct skl *skl)
                 platform_device_unregister(skl->clk_dev);
 }
 
+#define IDISP_INTEL_VENDOR_ID   0x80860000
+
+/*
+ * load the legacy codec driver
+ */
+static void load_codec_module(struct hda_codec *codec)
+{
+#ifdef MODULE
+        char modalias[MODULE_NAME_LEN];
+        const char *mod = NULL;
+
+        snd_hdac_codec_modalias(&codec->core, modalias, sizeof(modalias));
+        mod = modalias;
+        dev_dbg(&codec->core.dev, "loading %s codec module\n", mod);
+        request_module(mod);
+#endif
+}
+
 /*
  * Probe the given codec address
  */
@@ -637,7 +685,9 @@ static int probe_codec(struct hdac_bus *bus, int addr)
                 (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
         unsigned int res = -1;
         struct skl *skl = bus_to_skl(bus);
+        struct hdac_hda_priv *hda_codec;
         struct hdac_device *hdev;
+        int err;
 
         mutex_lock(&bus->cmd_mutex);
         snd_hdac_bus_send_cmd(bus, cmd);
@@ -645,13 +695,26 @@ static int probe_codec(struct hdac_bus *bus, int addr)
         mutex_unlock(&bus->cmd_mutex);
         if (res == -1)
                 return -EIO;
-        dev_dbg(bus->dev, "codec #%d probed OK\n", addr);
+        dev_dbg(bus->dev, "codec #%d probed OK: %x\n", addr, res);
 
-        hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL);
-        if (!hdev)
+        hda_codec = devm_kzalloc(&skl->pci->dev, sizeof(*hda_codec),
+                                 GFP_KERNEL);
+        if (!hda_codec)
                 return -ENOMEM;
 
-        return snd_hdac_ext_bus_device_init(bus, addr, hdev);
+        hda_codec->codec.bus = skl_to_hbus(skl);
+        hdev = &hda_codec->codec.core;
+
+        err = snd_hdac_ext_bus_device_init(bus, addr, hdev);
+        if (err < 0)
+                return err;
+
+        /* use legacy bus only for HDA codecs, idisp uses ext bus */
+        if ((res & 0xFFFF0000) != IDISP_INTEL_VENDOR_ID) {
+                hdev->type = HDA_DEV_LEGACY;
+                load_codec_module(&hda_codec->codec);
+        }
+        return 0;
 }
 
 /* Codec initialization */
@@ -786,9 +849,10 @@ static int skl_create(struct pci_dev *pci,
                       const struct hdac_io_ops *io_ops,
                       struct skl **rskl)
 {
+        struct hdac_ext_bus_ops *ext_ops = NULL;
         struct skl *skl;
         struct hdac_bus *bus;
-
+        struct hda_bus *hbus;
         int err;
 
         *rskl = NULL;
@@ -803,13 +867,23 @@ static int skl_create(struct pci_dev *pci,
                 return -ENOMEM;
         }
 
+        hbus = skl_to_hbus(skl);
         bus = skl_to_bus(skl);
-        snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, NULL);
+
+#if IS_ENABLED(CONFIG_SND_SOC_HDAC_HDA)
+        ext_ops = snd_soc_hdac_hda_get_ops();
+#endif
+        snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops);
         bus->use_posbuf = 1;
         skl->pci = pci;
         INIT_WORK(&skl->probe_work, skl_probe_work);
         bus->bdl_pos_adj = 0;
 
+        mutex_init(&hbus->prepare_mutex);
+        hbus->pci = pci;
+        hbus->mixer_assigned = -1;
+        hbus->modelname = "sklbus";
+
         *rskl = skl;
 
         return 0;
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index 78aa8bdcb619..8d48cd7c56c8 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -23,6 +23,7 @@
 
 #include <sound/hda_register.h>
 #include <sound/hdaudio_ext.h>
+#include <sound/hda_codec.h>
 #include <sound/soc.h>
 #include "skl-nhlt.h"
 #include "skl-ssp-clk.h"
@@ -71,7 +72,7 @@ struct skl_fw_config {
 };
 
 struct skl {
-        struct hdac_bus hbus;
+        struct hda_bus hbus;
         struct pci_dev *pci;
 
         unsigned int init_done:1; /* delayed init status */
@@ -105,8 +106,11 @@ struct skl {
         struct snd_soc_acpi_mach *mach;
 };
 
-#define skl_to_bus(s)  (&(s)->hbus)
-#define bus_to_skl(bus) container_of(bus, struct skl, hbus)
+#define skl_to_bus(s)  (&(s)->hbus.core)
+#define bus_to_skl(bus) container_of(bus, struct skl, hbus.core)
+
+#define skl_to_hbus(s) (&(s)->hbus)
+#define hbus_to_skl(hbus) container_of((hbus), struct skl, (hbus))
 
 /* to pass dai dma data */
 struct skl_dma_params {
@@ -117,6 +121,8 @@ struct skl_dma_params {
 struct skl_machine_pdata {
         u32 dmic_num;
         bool use_tplg_pcm; /* use dais and dai links from topology */
+        const char *platform;
+        u32 codec_mask;
 };
 
 struct skl_dsp_ops {
diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c
index 666282b865a8..97f9f38ce6b3 100644
--- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c
+++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c
@@ -299,6 +299,7 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev)
                 devm_kzalloc(&pdev->dev, sizeof(struct mt2701_cs42448_private),
                              GFP_KERNEL);
         struct device *dev = &pdev->dev;
+        struct snd_soc_dai_link *dai_link;
 
         if (!priv)
                 return -ENOMEM;
@@ -309,10 +310,10 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev)
                 dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
                 return -EINVAL;
         }
-        for (i = 0; i < card->num_links; i++) {
-                if (mt2701_cs42448_dai_links[i].platform_name)
+        for_each_card_prelinks(card, i, dai_link) {
+                if (dai_link->platform_name)
                         continue;
-                mt2701_cs42448_dai_links[i].platform_of_node = platform_node;
+                dai_link->platform_of_node = platform_node;
         }
 
         card->dev = dev;
@@ -324,10 +325,10 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev)
                         "Property 'audio-codec' missing or invalid\n");
                 return -EINVAL;
         }
-        for (i = 0; i < card->num_links; i++) {
-                if (mt2701_cs42448_dai_links[i].codec_name)
+        for_each_card_prelinks(card, i, dai_link) {
+                if (dai_link->codec_name)
                         continue;
-                mt2701_cs42448_dai_links[i].codec_of_node = codec_node;
+                dai_link->codec_of_node = codec_node;
         }
 
         codec_node_bt_mrg = of_parse_phandle(pdev->dev.of_node,
diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c
index 89f34efd9747..6bc1d3d58e64 100644
--- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c
+++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c
@@ -97,6 +97,7 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
 {
         struct snd_soc_card *card = &mt2701_wm8960_card;
         struct device_node *platform_node, *codec_node;
+        struct snd_soc_dai_link *dai_link;
         int ret, i;
 
         platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -105,10 +106,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
                 dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
                 return -EINVAL;
         }
-        for (i = 0; i < card->num_links; i++) {
-                if (mt2701_wm8960_dai_links[i].platform_name)
+        for_each_card_prelinks(card, i, dai_link) {
+                if (dai_link->platform_name)
                         continue;
-                mt2701_wm8960_dai_links[i].platform_of_node = platform_node;
+                dai_link->platform_of_node = platform_node;
         }
 
         card->dev = &pdev->dev;
@@ -120,10 +121,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
                         "Property 'audio-codec' missing or invalid\n");
                 return -EINVAL;
         }
-        for (i = 0; i < card->num_links; i++) {
-                if (mt2701_wm8960_dai_links[i].codec_name)
+        for_each_card_prelinks(card, i, dai_link) {
+                if (dai_link->codec_name)
                         continue;
-                mt2701_wm8960_dai_links[i].codec_of_node = codec_node;
+                dai_link->codec_of_node = codec_node;
         }
 
         ret = snd_soc_of_parse_audio_routing(card, "audio-routing");
@@ -150,7 +151,6 @@ static const struct of_device_id mt2701_wm8960_machine_dt_match[] = {
 static struct platform_driver mt2701_wm8960_machine = {
         .driver = {
                 .name = "mt2701-wm8960",
-                .owner = THIS_MODULE,
 #ifdef CONFIG_OF
                 .of_match_table = mt2701_wm8960_machine_dt_match,
 #endif
diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c
index b1558c57b9ca..cc41eb531653 100644
--- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c
+++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c
@@ -158,6 +158,7 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev)
 {
         struct snd_soc_card *card = &mt6797_mt6351_card;
         struct device_node *platform_node, *codec_node;
+        struct snd_soc_dai_link *dai_link;
         int ret, i;
 
         card->dev = &pdev->dev;
@@ -168,10 +169,10 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev)
                 dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
                 return -EINVAL;
         }
-        for (i = 0; i < card->num_links; i++) {
-                if (mt6797_mt6351_dai_links[i].platform_name)
+        for_each_card_prelinks(card, i, dai_link) {
+                if (dai_link->platform_name)
                         continue;
-                mt6797_mt6351_dai_links[i].platform_of_node = platform_node;
+                dai_link->platform_of_node = platform_node;
         }
 
         codec_node = of_parse_phandle(pdev->dev.of_node,
@@ -181,10 +182,10 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev)
                         "Property 'audio-codec' missing or invalid\n");
                 return -EINVAL;
         }
-        for (i = 0; i < card->num_links; i++) {
-                if (mt6797_mt6351_dai_links[i].codec_name)
+        for_each_card_prelinks(card, i, dai_link) {
+                if (dai_link->codec_name)
                         continue;
-                mt6797_mt6351_dai_links[i].codec_of_node = codec_node;
+                dai_link->codec_of_node = codec_node;
         }
 
         ret = devm_snd_soc_register_card(&pdev->dev, card);
@@ -205,7 +206,6 @@ static const struct of_device_id mt6797_mt6351_dt_match[] = {
 static struct platform_driver mt6797_mt6351_driver = {
         .driver = {
                 .name = "mt6797-mt6351",
-                .owner = THIS_MODULE,
 #ifdef CONFIG_OF
                 .of_match_table = mt6797_mt6351_dt_match,
 #endif
diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c
index 902d111016d6..4d6596d5cb07 100644
--- a/sound/soc/mediatek/mt8173/mt8173-max98090.c
+++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c
@@ -137,6 +137,7 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
 {
         struct snd_soc_card *card = &mt8173_max98090_card;
         struct device_node *codec_node, *platform_node;
+        struct snd_soc_dai_link *dai_link;
         int ret, i;
 
         platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -145,10 +146,10 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
                 dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
                 return -EINVAL;
         }
-        for (i = 0; i < card->num_links; i++) {
-                if (mt8173_max98090_dais[i].platform_name)
+        for_each_card_prelinks(card, i, dai_link) {
+                if (dai_link->platform_name)
                         continue;
-                mt8173_max98090_dais[i].platform_of_node = platform_node;
+                dai_link->platform_of_node = platform_node;
         }
 
         codec_node = of_parse_phandle(pdev->dev.of_node,
@@ -158,10 +159,10 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
                         "Property 'audio-codec' missing or invalid\n");
                 return -EINVAL;
         }
-        for (i = 0; i < card->num_links; i++) {
-                if (mt8173_max98090_dais[i].codec_name)
+        for_each_card_prelinks(card, i, dai_link) {
+                if (dai_link->codec_name)
                         continue;
-                mt8173_max98090_dais[i].codec_of_node = codec_node;
+                dai_link->codec_of_node = codec_node;
         }
         card->dev = &pdev->dev;
 
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
index 582174d98c6c..da5b58ce791b 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
@@ -44,11 +44,10 @@ static int mt8173_rt5650_rt5514_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;
         int i, ret;
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
-
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 /* pll from mclk 12.288M */
                 ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
                                           params_rate(params) * 512);
@@ -179,6 +178,7 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev)
 {
         struct snd_soc_card *card = &mt8173_rt5650_rt5514_card;
         struct device_node *platform_node;
+        struct snd_soc_dai_link *dai_link;
         int i, ret;
 
         platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -188,10 +188,10 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev)
                 return -EINVAL;
         }
 
-        for (i = 0; i < card->num_links; i++) {
-                if (mt8173_rt5650_rt5514_dais[i].platform_name)
+        for_each_card_prelinks(card, i, dai_link) {
+                if (dai_link->platform_name)
                         continue;
-                mt8173_rt5650_rt5514_dais[i].platform_of_node = platform_node;
+                dai_link->platform_of_node = platform_node;
         }
 
         mt8173_rt5650_rt5514_codecs[0].of_node =
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
index b3670c8a5b8d..d83cd039b413 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
@@ -48,11 +48,10 @@ static int mt8173_rt5650_rt5676_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;
         int i, ret;
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
-
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 /* pll from mclk 12.288M */
                 ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
                                           params_rate(params) * 512);
@@ -225,6 +224,7 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
 {
         struct snd_soc_card *card = &mt8173_rt5650_rt5676_card;
         struct device_node *platform_node;
+        struct snd_soc_dai_link *dai_link;
         int i, ret;
 
         platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -234,10 +234,10 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
                 return -EINVAL;
         }
 
-        for (i = 0; i < card->num_links; i++) {
-                if (mt8173_rt5650_rt5676_dais[i].platform_name)
+        for_each_card_prelinks(card, i, dai_link) {
+                if (dai_link->platform_name)
                         continue;
-                mt8173_rt5650_rt5676_dais[i].platform_of_node = platform_node;
+                dai_link->platform_of_node = platform_node;
         }
 
         mt8173_rt5650_rt5676_codecs[0].of_node =
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
index 7a89b4aad182..7edf250c8fb1 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
@@ -59,6 +59,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream,
 {
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         unsigned int mclk_clock;
+        struct snd_soc_dai *codec_dai;
         int i, ret;
 
         switch (mt8173_rt5650_priv.pll_from) {
@@ -76,9 +77,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream,
                 break;
         }
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
-
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 /* pll from mclk */
                 ret = snd_soc_dai_set_pll(codec_dai, 0, 0, mclk_clock,
                                           params_rate(params) * 512);
@@ -240,6 +239,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
         struct device_node *platform_node;
         struct device_node *np;
         const char *codec_capture_dai;
+        struct snd_soc_dai_link *dai_link;
         int i, ret;
 
         platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -249,10 +249,10 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
                 return -EINVAL;
         }
 
-        for (i = 0; i < card->num_links; i++) {
-                if (mt8173_rt5650_dais[i].platform_name)
+        for_each_card_prelinks(card, i, dai_link) {
+                if (dai_link->platform_name)
                         continue;
-                mt8173_rt5650_dais[i].platform_of_node = platform_node;
+                dai_link->platform_of_node = platform_node;
         }
 
         mt8173_rt5650_codecs[0].of_node =
diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig
index 8af8bc358a90..8b8426ed2363 100644
--- a/sound/soc/meson/Kconfig
+++ b/sound/soc/meson/Kconfig
@@ -4,6 +4,8 @@ menu "ASoC support for Amlogic platforms"
 config SND_MESON_AXG_FIFO
         tristate
         select REGMAP_MMIO
+        imply COMMON_CLK_AXG_AUDIO
+        imply RESET_MESON_AUDIO_ARB
 
 config SND_MESON_AXG_FRDDR
         tristate "Amlogic AXG Playback FIFO support"
@@ -22,6 +24,7 @@ config SND_MESON_AXG_TODDR
 config SND_MESON_AXG_TDM_FORMATTER
         tristate
         select REGMAP_MMIO
+        imply COMMON_CLK_AXG_AUDIO
 
 config SND_MESON_AXG_TDM_INTERFACE
         tristate
@@ -51,6 +54,7 @@ config SND_MESON_AXG_SOUND_CARD
         imply SND_MESON_AXG_TDMIN
         imply SND_MESON_AXG_TDMOUT
         imply SND_MESON_AXG_SPDIFOUT
+        imply SND_MESON_AXG_PDM
         help
           Select Y or M to add support for the AXG SoC sound card
 
@@ -58,8 +62,17 @@ config SND_MESON_AXG_SPDIFOUT
         tristate "Amlogic AXG SPDIF Output Support"
         select SND_PCM_IEC958
         imply SND_SOC_SPDIF
+        imply COMMON_CLK_AXG_AUDIO
         help
           Select Y or M to add support for SPDIF output serializer embedded
           in the Amlogic AXG SoC family
 
+config SND_MESON_AXG_PDM
+        tristate "Amlogic AXG PDM Input Support"
+        imply SND_SOC_DMIC
+        imply COMMON_CLK_AXG_AUDIO
+        help
+          Select Y or M to add support for PDM input embedded
+          in the Amlogic AXG SoC family
+
 endmenu
diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile
index c5e003b093db..4cd25104029d 100644
--- a/sound/soc/meson/Makefile
+++ b/sound/soc/meson/Makefile
@@ -9,6 +9,7 @@ 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
+snd-soc-meson-axg-pdm-objs := axg-pdm.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
@@ -19,3 +20,4 @@ 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
+obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 2914ba0d965b..aa54d2c612c9 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -97,14 +97,14 @@ static void axg_card_clean_references(struct axg_card *priv)
 {
         struct snd_soc_card *card = &priv->card;
         struct snd_soc_dai_link *link;
+        struct snd_soc_dai_link_component *codec;
         int i, j;
 
         if (card->dai_link) {
-                for (i = 0; i < card->num_links; i++) {
-                        link = &card->dai_link[i];
+                for_each_card_prelinks(card, i, link) {
                         of_node_put(link->cpu_of_node);
-                        for (j = 0; j < link->num_codecs; j++)
-                                of_node_put(link->codecs[j].of_node);
+                        for_each_link_codecs(link, j, codec)
+                                of_node_put(codec->of_node);
                 }
         }
 
@@ -167,8 +167,7 @@ static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream,
         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];
+                for_each_rtd_codec_dai(rtd, i, codec_dai) {
                         ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
                                                      SND_SOC_CLOCK_IN);
                         if (ret && ret != -ENOTSUPP)
@@ -196,8 +195,7 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd)
         struct snd_soc_dai *codec_dai;
         int ret, i;
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                codec_dai = rtd->codec_dais[i];
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 ret = snd_soc_dai_set_tdm_slot(codec_dai,
                                                be->codec_masks[i].tx,
                                                be->codec_masks[i].rx,
@@ -478,7 +476,7 @@ static int axg_card_set_be_link(struct snd_soc_card *card,
 
         ret = axg_card_set_link_name(card, link, "be");
         if (ret)
-                dev_err(card->dev, "error setting %s link name\n", np->name);
+                dev_err(card->dev, "error setting %pOFn link name\n", np);
 
         return ret;
 }
diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
index 30262550e37b..0e4f65e654c4 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -203,6 +203,8 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss)
 
         ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0,
                           dev_name(dev), ss);
+        if (ret)
+                return ret;
 
         /* Enable pclk to access registers and clock the fifo ip */
         ret = clk_prepare_enable(fifo->pclk);
diff --git a/sound/soc/meson/axg-pdm.c b/sound/soc/meson/axg-pdm.c
new file mode 100644
index 000000000000..9d5684493ffc
--- /dev/null
+++ b/sound/soc/meson/axg-pdm.c
@@ -0,0 +1,654 @@
+// 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_irq.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/pcm_params.h>
+
+#define PDM_CTRL                        0x00
+#define  PDM_CTRL_EN                    BIT(31)
+#define  PDM_CTRL_OUT_MODE              BIT(29)
+#define  PDM_CTRL_BYPASS_MODE           BIT(28)
+#define  PDM_CTRL_RST_FIFO              BIT(16)
+#define  PDM_CTRL_CHAN_RSTN_MASK        GENMASK(15, 8)
+#define  PDM_CTRL_CHAN_RSTN(x)          ((x) << 8)
+#define  PDM_CTRL_CHAN_EN_MASK          GENMASK(7, 0)
+#define  PDM_CTRL_CHAN_EN(x)            ((x) << 0)
+#define PDM_HCIC_CTRL1                  0x04
+#define  PDM_FILTER_EN                  BIT(31)
+#define  PDM_HCIC_CTRL1_GAIN_SFT_MASK   GENMASK(29, 24)
+#define  PDM_HCIC_CTRL1_GAIN_SFT(x)     ((x) << 24)
+#define  PDM_HCIC_CTRL1_GAIN_MULT_MASK  GENMASK(23, 16)
+#define  PDM_HCIC_CTRL1_GAIN_MULT(x)    ((x) << 16)
+#define  PDM_HCIC_CTRL1_DSR_MASK        GENMASK(8, 4)
+#define  PDM_HCIC_CTRL1_DSR(x)          ((x) << 4)
+#define  PDM_HCIC_CTRL1_STAGE_NUM_MASK  GENMASK(3, 0)
+#define  PDM_HCIC_CTRL1_STAGE_NUM(x)    ((x) << 0)
+#define PDM_HCIC_CTRL2                  0x08
+#define PDM_F1_CTRL                     0x0c
+#define  PDM_LPF_ROUND_MODE_MASK        GENMASK(17, 16)
+#define  PDM_LPF_ROUND_MODE(x)          ((x) << 16)
+#define  PDM_LPF_DSR_MASK               GENMASK(15, 12)
+#define  PDM_LPF_DSR(x)                 ((x) << 12)
+#define  PDM_LPF_STAGE_NUM_MASK         GENMASK(8, 0)
+#define  PDM_LPF_STAGE_NUM(x)           ((x) << 0)
+#define  PDM_LPF_MAX_STAGE              336
+#define  PDM_LPF_NUM                    3
+#define PDM_F2_CTRL                     0x10
+#define PDM_F3_CTRL                     0x14
+#define PDM_HPF_CTRL                    0x18
+#define  PDM_HPF_SFT_STEPS_MASK         GENMASK(20, 16)
+#define  PDM_HPF_SFT_STEPS(x)           ((x) << 16)
+#define  PDM_HPF_OUT_FACTOR_MASK        GENMASK(15, 0)
+#define  PDM_HPF_OUT_FACTOR(x)          ((x) << 0)
+#define PDM_CHAN_CTRL                   0x1c
+#define  PDM_CHAN_CTRL_POINTER_WIDTH    8
+#define  PDM_CHAN_CTRL_POINTER_MAX      ((1 << PDM_CHAN_CTRL_POINTER_WIDTH) - 1)
+#define  PDM_CHAN_CTRL_NUM              4
+#define PDM_CHAN_CTRL1                  0x20
+#define PDM_COEFF_ADDR                  0x24
+#define PDM_COEFF_DATA                  0x28
+#define PDM_CLKG_CTRL                   0x2c
+#define PDM_STS                         0x30
+
+struct axg_pdm_lpf {
+        unsigned int ds;
+        unsigned int round_mode;
+        const unsigned int *tap;
+        unsigned int tap_num;
+};
+
+struct axg_pdm_hcic {
+        unsigned int shift;
+        unsigned int mult;
+        unsigned int steps;
+        unsigned int ds;
+};
+
+struct axg_pdm_hpf {
+        unsigned int out_factor;
+        unsigned int steps;
+};
+
+struct axg_pdm_filters {
+        struct axg_pdm_hcic hcic;
+        struct axg_pdm_hpf hpf;
+        struct axg_pdm_lpf lpf[PDM_LPF_NUM];
+};
+
+struct axg_pdm_cfg {
+        const struct axg_pdm_filters *filters;
+        unsigned int sys_rate;
+};
+
+struct axg_pdm {
+        const struct axg_pdm_cfg *cfg;
+        struct regmap *map;
+        struct clk *dclk;
+        struct clk *sysclk;
+        struct clk *pclk;
+};
+
+static void axg_pdm_enable(struct regmap *map)
+{
+        /* Reset AFIFO */
+        regmap_update_bits(map, PDM_CTRL, PDM_CTRL_RST_FIFO, PDM_CTRL_RST_FIFO);
+        regmap_update_bits(map, PDM_CTRL, PDM_CTRL_RST_FIFO, 0);
+
+        /* Enable PDM */
+        regmap_update_bits(map, PDM_CTRL, PDM_CTRL_EN, PDM_CTRL_EN);
+}
+
+static void axg_pdm_disable(struct regmap *map)
+{
+        regmap_update_bits(map, PDM_CTRL, PDM_CTRL_EN, 0);
+}
+
+static void axg_pdm_filters_enable(struct regmap *map, bool enable)
+{
+        unsigned int val = enable ? PDM_FILTER_EN : 0;
+
+        regmap_update_bits(map, PDM_HCIC_CTRL1, PDM_FILTER_EN, val);
+        regmap_update_bits(map, PDM_F1_CTRL, PDM_FILTER_EN, val);
+        regmap_update_bits(map, PDM_F2_CTRL, PDM_FILTER_EN, val);
+        regmap_update_bits(map, PDM_F3_CTRL, PDM_FILTER_EN, val);
+        regmap_update_bits(map, PDM_HPF_CTRL, PDM_FILTER_EN, val);
+}
+
+static int axg_pdm_trigger(struct snd_pcm_substream *substream, int cmd,
+                           struct snd_soc_dai *dai)
+{
+        struct axg_pdm *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_pdm_enable(priv->map);
+                return 0;
+
+        case SNDRV_PCM_TRIGGER_STOP:
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+        case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+                axg_pdm_disable(priv->map);
+                return 0;
+
+        default:
+                return -EINVAL;
+        }
+}
+
+static unsigned int axg_pdm_get_os(struct axg_pdm *priv)
+{
+        const struct axg_pdm_filters *filters = priv->cfg->filters;
+        unsigned int os = filters->hcic.ds;
+        int i;
+
+        /*
+         * The global oversampling factor is defined by the down sampling
+         * factor applied by each filter (HCIC and LPFs)
+         */
+
+        for (i = 0; i < PDM_LPF_NUM; i++)
+                os *= filters->lpf[i].ds;
+
+        return os;
+}
+
+static int axg_pdm_set_sysclk(struct axg_pdm *priv, unsigned int os,
+                              unsigned int rate)
+{
+        unsigned int sys_rate = os * 2 * rate * PDM_CHAN_CTRL_POINTER_MAX;
+
+        /*
+         * Set the default system clock rate unless it is too fast for
+         * for the requested sample rate. In this case, the sample pointer
+         * counter could overflow so set a lower system clock rate
+         */
+        if (sys_rate < priv->cfg->sys_rate)
+                return clk_set_rate(priv->sysclk, sys_rate);
+
+        return clk_set_rate(priv->sysclk, priv->cfg->sys_rate);
+}
+
+static int axg_pdm_set_sample_pointer(struct axg_pdm *priv)
+{
+        unsigned int spmax, sp, val;
+        int i;
+
+        /* Max sample counter value per half period of dclk */
+        spmax = DIV_ROUND_UP_ULL((u64)clk_get_rate(priv->sysclk),
+                                 clk_get_rate(priv->dclk) * 2);
+
+        /* Check if sysclk is not too fast - should not happen */
+        if (WARN_ON(spmax > PDM_CHAN_CTRL_POINTER_MAX))
+                return -EINVAL;
+
+        /* Capture the data when we are at 75% of the half period */
+        sp = spmax * 3 / 4;
+
+        for (i = 0, val = 0; i < PDM_CHAN_CTRL_NUM; i++)
+                val |= sp << (PDM_CHAN_CTRL_POINTER_WIDTH * i);
+
+        regmap_write(priv->map, PDM_CHAN_CTRL, val);
+        regmap_write(priv->map, PDM_CHAN_CTRL1, val);
+
+        return 0;
+}
+
+static void axg_pdm_set_channel_mask(struct axg_pdm *priv,
+                                     unsigned int channels)
+{
+        unsigned int mask = GENMASK(channels - 1, 0);
+
+        /* Put all channel in reset */
+        regmap_update_bits(priv->map, PDM_CTRL,
+                           PDM_CTRL_CHAN_RSTN_MASK, 0);
+
+        /* Take the necessary channels out of reset and enable them */
+        regmap_update_bits(priv->map, PDM_CTRL,
+                           PDM_CTRL_CHAN_RSTN_MASK |
+                           PDM_CTRL_CHAN_EN_MASK,
+                           PDM_CTRL_CHAN_RSTN(mask) |
+                           PDM_CTRL_CHAN_EN(mask));
+}
+
+static int axg_pdm_hw_params(struct snd_pcm_substream *substream,
+                             struct snd_pcm_hw_params *params,
+                             struct snd_soc_dai *dai)
+{
+        struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+        unsigned int os = axg_pdm_get_os(priv);
+        unsigned int rate = params_rate(params);
+        unsigned int val;
+        int ret;
+
+        switch (params_width(params)) {
+        case 24:
+                val = PDM_CTRL_OUT_MODE;
+                break;
+        case 32:
+                val = 0;
+                break;
+        default:
+                dev_err(dai->dev, "unsupported sample width\n");
+                return -EINVAL;
+        }
+
+        regmap_update_bits(priv->map, PDM_CTRL, PDM_CTRL_OUT_MODE, val);
+
+        ret = axg_pdm_set_sysclk(priv, os, rate);
+        if (ret) {
+                dev_err(dai->dev, "failed to set system clock\n");
+                return ret;
+        }
+
+        ret = clk_set_rate(priv->dclk, rate * os);
+        if (ret) {
+                dev_err(dai->dev, "failed to set dclk\n");
+                return ret;
+        }
+
+        ret = axg_pdm_set_sample_pointer(priv);
+        if (ret) {
+                dev_err(dai->dev, "invalid clock setting\n");
+                return ret;
+        }
+
+        axg_pdm_set_channel_mask(priv, params_channels(params));
+
+        return 0;
+}
+
+static int axg_pdm_startup(struct snd_pcm_substream *substream,
+                           struct snd_soc_dai *dai)
+{
+        struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+        int ret;
+
+        ret = clk_prepare_enable(priv->dclk);
+        if (ret) {
+                dev_err(dai->dev, "enabling dclk failed\n");
+                return ret;
+        }
+
+        /* Enable the filters */
+        axg_pdm_filters_enable(priv->map, true);
+
+        return ret;
+}
+
+static void axg_pdm_shutdown(struct snd_pcm_substream *substream,
+                             struct snd_soc_dai *dai)
+{
+        struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+
+        axg_pdm_filters_enable(priv->map, false);
+        clk_disable_unprepare(priv->dclk);
+}
+
+static const struct snd_soc_dai_ops axg_pdm_dai_ops = {
+        .trigger        = axg_pdm_trigger,
+        .hw_params      = axg_pdm_hw_params,
+        .startup        = axg_pdm_startup,
+        .shutdown       = axg_pdm_shutdown,
+};
+
+static void axg_pdm_set_hcic_ctrl(struct axg_pdm *priv)
+{
+        const struct axg_pdm_hcic *hcic = &priv->cfg->filters->hcic;
+        unsigned int val;
+
+        val = PDM_HCIC_CTRL1_STAGE_NUM(hcic->steps);
+        val |= PDM_HCIC_CTRL1_DSR(hcic->ds);
+        val |= PDM_HCIC_CTRL1_GAIN_MULT(hcic->mult);
+        val |= PDM_HCIC_CTRL1_GAIN_SFT(hcic->shift);
+
+        regmap_update_bits(priv->map, PDM_HCIC_CTRL1,
+                           PDM_HCIC_CTRL1_STAGE_NUM_MASK |
+                           PDM_HCIC_CTRL1_DSR_MASK |
+                           PDM_HCIC_CTRL1_GAIN_MULT_MASK |
+                           PDM_HCIC_CTRL1_GAIN_SFT_MASK,
+                           val);
+}
+
+static void axg_pdm_set_lpf_ctrl(struct axg_pdm *priv, unsigned int index)
+{
+        const struct axg_pdm_lpf *lpf = &priv->cfg->filters->lpf[index];
+        unsigned int offset = index * regmap_get_reg_stride(priv->map)
+                + PDM_F1_CTRL;
+        unsigned int val;
+
+        val = PDM_LPF_STAGE_NUM(lpf->tap_num);
+        val |= PDM_LPF_DSR(lpf->ds);
+        val |= PDM_LPF_ROUND_MODE(lpf->round_mode);
+
+        regmap_update_bits(priv->map, offset,
+                           PDM_LPF_STAGE_NUM_MASK |
+                           PDM_LPF_DSR_MASK |
+                           PDM_LPF_ROUND_MODE_MASK,
+                           val);
+}
+
+static void axg_pdm_set_hpf_ctrl(struct axg_pdm *priv)
+{
+        const struct axg_pdm_hpf *hpf = &priv->cfg->filters->hpf;
+        unsigned int val;
+
+        val = PDM_HPF_OUT_FACTOR(hpf->out_factor);
+        val |= PDM_HPF_SFT_STEPS(hpf->steps);
+
+        regmap_update_bits(priv->map, PDM_HPF_CTRL,
+                           PDM_HPF_OUT_FACTOR_MASK |
+                           PDM_HPF_SFT_STEPS_MASK,
+                           val);
+}
+
+static int axg_pdm_set_lpf_filters(struct axg_pdm *priv)
+{
+        const struct axg_pdm_lpf *lpf = priv->cfg->filters->lpf;
+        unsigned int count = 0;
+        int i, j;
+
+        for (i = 0; i < PDM_LPF_NUM; i++)
+                count += lpf[i].tap_num;
+
+        /* Make sure the coeffs fit in the memory */
+        if (count >= PDM_LPF_MAX_STAGE)
+                return -EINVAL;
+
+        /* Set the initial APB bus register address */
+        regmap_write(priv->map, PDM_COEFF_ADDR, 0);
+
+        /* Set the tap filter values of all 3 filters */
+        for (i = 0; i < PDM_LPF_NUM; i++) {
+                axg_pdm_set_lpf_ctrl(priv, i);
+
+                for (j = 0; j < lpf[i].tap_num; j++)
+                        regmap_write(priv->map, PDM_COEFF_DATA, lpf[i].tap[j]);
+        }
+
+        return 0;
+}
+
+static int axg_pdm_dai_probe(struct snd_soc_dai *dai)
+{
+        struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+        int ret;
+
+        ret = clk_prepare_enable(priv->pclk);
+        if (ret) {
+                dev_err(dai->dev, "enabling pclk failed\n");
+                return ret;
+        }
+
+        /*
+         * sysclk must be set and enabled as well to access the pdm registers
+         * Accessing the register w/o it will give a bus error.
+         */
+        ret = clk_set_rate(priv->sysclk, priv->cfg->sys_rate);
+        if (ret) {
+                dev_err(dai->dev, "setting sysclk failed\n");
+                goto err_pclk;
+        }
+
+        ret = clk_prepare_enable(priv->sysclk);
+        if (ret) {
+                dev_err(dai->dev, "enabling sysclk failed\n");
+                goto err_pclk;
+        }
+
+        /* Make sure the device is initially disabled */
+        axg_pdm_disable(priv->map);
+
+        /* Make sure filter bypass is disabled */
+        regmap_update_bits(priv->map, PDM_CTRL, PDM_CTRL_BYPASS_MODE, 0);
+
+        /* Load filter settings */
+        axg_pdm_set_hcic_ctrl(priv);
+        axg_pdm_set_hpf_ctrl(priv);
+
+        ret = axg_pdm_set_lpf_filters(priv);
+        if (ret) {
+                dev_err(dai->dev, "invalid filter configuration\n");
+                goto err_sysclk;
+        }
+
+        return 0;
+
+err_sysclk:
+        clk_disable_unprepare(priv->sysclk);
+err_pclk:
+        clk_disable_unprepare(priv->pclk);
+        return ret;
+}
+
+static int axg_pdm_dai_remove(struct snd_soc_dai *dai)
+{
+        struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+
+        clk_disable_unprepare(priv->sysclk);
+        clk_disable_unprepare(priv->pclk);
+
+        return 0;
+}
+
+static struct snd_soc_dai_driver axg_pdm_dai_drv = {
+        .name = "PDM",
+        .capture = {
+                .stream_name    = "Capture",
+                .channels_min   = 1,
+                .channels_max   = 8,
+                .rates          = SNDRV_PCM_RATE_CONTINUOUS,
+                .rate_min       = 5512,
+                .rate_max       = 48000,
+                .formats        = (SNDRV_PCM_FMTBIT_S24_LE |
+                                   SNDRV_PCM_FMTBIT_S32_LE),
+        },
+        .ops            = &axg_pdm_dai_ops,
+        .probe          = axg_pdm_dai_probe,
+        .remove         = axg_pdm_dai_remove,
+};
+
+static const struct snd_soc_component_driver axg_pdm_component_drv = {};
+
+static const struct regmap_config axg_pdm_regmap_cfg = {
+        .reg_bits       = 32,
+        .val_bits       = 32,
+        .reg_stride     = 4,
+        .max_register   = PDM_STS,
+};
+
+static const unsigned int lpf1_default_tap[] = {
+        0x000014, 0xffffb2, 0xfffed9, 0xfffdce, 0xfffd45,
+        0xfffe32, 0x000147, 0x000645, 0x000b86, 0x000e21,
+        0x000ae3, 0x000000, 0xffeece, 0xffdca8, 0xffd212,
+        0xffd7d1, 0xfff2a7, 0x001f4c, 0x0050c2, 0x0072aa,
+        0x006ff1, 0x003c32, 0xffdc4e, 0xff6a18, 0xff0fef,
+        0xfefbaf, 0xff4c40, 0x000000, 0x00ebc8, 0x01c077,
+        0x02209e, 0x01c1a4, 0x008e60, 0xfebe52, 0xfcd690,
+        0xfb8fa5, 0xfba498, 0xfd9812, 0x0181ce, 0x06f5f3,
+        0x0d112f, 0x12a958, 0x169686, 0x18000e, 0x169686,
+        0x12a958, 0x0d112f, 0x06f5f3, 0x0181ce, 0xfd9812,
+        0xfba498, 0xfb8fa5, 0xfcd690, 0xfebe52, 0x008e60,
+        0x01c1a4, 0x02209e, 0x01c077, 0x00ebc8, 0x000000,
+        0xff4c40, 0xfefbaf, 0xff0fef, 0xff6a18, 0xffdc4e,
+        0x003c32, 0x006ff1, 0x0072aa, 0x0050c2, 0x001f4c,
+        0xfff2a7, 0xffd7d1, 0xffd212, 0xffdca8, 0xffeece,
+        0x000000, 0x000ae3, 0x000e21, 0x000b86, 0x000645,
+        0x000147, 0xfffe32, 0xfffd45, 0xfffdce, 0xfffed9,
+        0xffffb2, 0x000014,
+};
+
+static const unsigned int lpf2_default_tap[] = {
+        0x00050a, 0xfff004, 0x0002c1, 0x003c12, 0xffa818,
+        0xffc87d, 0x010aef, 0xff5223, 0xfebd93, 0x028f41,
+        0xff5c0e, 0xfc63f8, 0x055f81, 0x000000, 0xf478a0,
+        0x11c5e3, 0x2ea74d, 0x11c5e3, 0xf478a0, 0x000000,
+        0x055f81, 0xfc63f8, 0xff5c0e, 0x028f41, 0xfebd93,
+        0xff5223, 0x010aef, 0xffc87d, 0xffa818, 0x003c12,
+        0x0002c1, 0xfff004, 0x00050a,
+};
+
+static const unsigned int lpf3_default_tap[] = {
+        0x000000, 0x000081, 0x000000, 0xfffedb, 0x000000,
+        0x00022d, 0x000000, 0xfffc46, 0x000000, 0x0005f7,
+        0x000000, 0xfff6eb, 0x000000, 0x000d4e, 0x000000,
+        0xffed1e, 0x000000, 0x001a1c, 0x000000, 0xffdcb0,
+        0x000000, 0x002ede, 0x000000, 0xffc2d1, 0x000000,
+        0x004ebe, 0x000000, 0xff9beb, 0x000000, 0x007dd7,
+        0x000000, 0xff633a, 0x000000, 0x00c1d2, 0x000000,
+        0xff11d5, 0x000000, 0x012368, 0x000000, 0xfe9c45,
+        0x000000, 0x01b252, 0x000000, 0xfdebf6, 0x000000,
+        0x0290b8, 0x000000, 0xfcca0d, 0x000000, 0x041d7c,
+        0x000000, 0xfa8152, 0x000000, 0x07e9c6, 0x000000,
+        0xf28fb5, 0x000000, 0x28b216, 0x3fffde, 0x28b216,
+        0x000000, 0xf28fb5, 0x000000, 0x07e9c6, 0x000000,
+        0xfa8152, 0x000000, 0x041d7c, 0x000000, 0xfcca0d,
+        0x000000, 0x0290b8, 0x000000, 0xfdebf6, 0x000000,
+        0x01b252, 0x000000, 0xfe9c45, 0x000000, 0x012368,
+        0x000000, 0xff11d5, 0x000000, 0x00c1d2, 0x000000,
+        0xff633a, 0x000000, 0x007dd7, 0x000000, 0xff9beb,
+        0x000000, 0x004ebe, 0x000000, 0xffc2d1, 0x000000,
+        0x002ede, 0x000000, 0xffdcb0, 0x000000, 0x001a1c,
+        0x000000, 0xffed1e, 0x000000, 0x000d4e, 0x000000,
+        0xfff6eb, 0x000000, 0x0005f7, 0x000000, 0xfffc46,
+        0x000000, 0x00022d, 0x000000, 0xfffedb, 0x000000,
+        0x000081, 0x000000,
+};
+
+/*
+ * These values are sane defaults for the axg platform:
+ * - OS = 64
+ * - Latency = 38700 (?)
+ *
+ * TODO: There is a lot of different HCIC, LPFs and HPF configurations possible.
+ *       the configuration may depend on the dmic used by the platform, the
+ *       expected tradeoff between latency and quality, etc ... If/When other
+ *       settings are required, we should add a fw interface to this driver to
+ *       load new filter settings.
+ */
+static const struct axg_pdm_filters axg_default_filters = {
+        .hcic = {
+                .shift = 0x15,
+                .mult = 0x80,
+                .steps = 7,
+                .ds = 8,
+        },
+        .hpf = {
+                .out_factor = 0x8000,
+                .steps = 13,
+        },
+        .lpf = {
+                [0] = {
+                        .ds = 2,
+                        .round_mode = 1,
+                        .tap = lpf1_default_tap,
+                        .tap_num = ARRAY_SIZE(lpf1_default_tap),
+                },
+                [1] = {
+                        .ds = 2,
+                        .round_mode = 0,
+                        .tap = lpf2_default_tap,
+                        .tap_num = ARRAY_SIZE(lpf2_default_tap),
+                },
+                [2] = {
+                        .ds = 2,
+                        .round_mode = 1,
+                        .tap = lpf3_default_tap,
+                        .tap_num = ARRAY_SIZE(lpf3_default_tap)
+                },
+        },
+};
+
+static const struct axg_pdm_cfg axg_pdm_config = {
+        .filters = &axg_default_filters,
+        .sys_rate = 250000000,
+};
+
+static const struct of_device_id axg_pdm_of_match[] = {
+        {
+                .compatible = "amlogic,axg-pdm",
+                .data = &axg_pdm_config,
+        }, {}
+};
+MODULE_DEVICE_TABLE(of, axg_pdm_of_match);
+
+static int axg_pdm_probe(struct platform_device *pdev)
+{
+        struct device *dev = &pdev->dev;
+        struct axg_pdm *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);
+
+        priv->cfg = of_device_get_match_data(dev);
+        if (!priv->cfg) {
+                dev_err(dev, "failed to match device\n");
+                return -ENODEV;
+        }
+
+        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_pdm_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->dclk = devm_clk_get(dev, "dclk");
+        if (IS_ERR(priv->dclk)) {
+                ret = PTR_ERR(priv->dclk);
+                if (ret != -EPROBE_DEFER)
+                        dev_err(dev, "failed to get dclk: %d\n", ret);
+                return ret;
+        }
+
+        priv->sysclk = devm_clk_get(dev, "sysclk");
+        if (IS_ERR(priv->sysclk)) {
+                ret = PTR_ERR(priv->sysclk);
+                if (ret != -EPROBE_DEFER)
+                        dev_err(dev, "failed to get dclk: %d\n", ret);
+                return ret;
+        }
+
+        return devm_snd_soc_register_component(dev, &axg_pdm_component_drv,
+                                               &axg_pdm_dai_drv, 1);
+}
+
+static struct platform_driver axg_pdm_pdrv = {
+        .probe = axg_pdm_probe,
+        .driver = {
+                .name = "axg-pdm",
+                .of_match_table = axg_pdm_of_match,
+        },
+};
+module_platform_driver(axg_pdm_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG PDM Input driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index 7b8baf46d968..585ce030b79b 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -42,6 +42,7 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
         struct axg_tdm_stream *rx = (struct axg_tdm_stream *)
                 dai->capture_dma_data;
         unsigned int tx_slots, rx_slots;
+        unsigned int fmt = 0;
 
         tx_slots = axg_tdm_slots_total(tx_mask);
         rx_slots = axg_tdm_slots_total(rx_mask);
@@ -52,38 +53,45 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
                 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:
+                slot_width = 32;
+                /* Fall-through */
         case 32:
-                iface->slot_width = slot_width;
+                fmt |= SNDRV_PCM_FMTBIT_S32_LE;
+                /* Fall-through */
+        case 24:
+                fmt |= SNDRV_PCM_FMTBIT_S24_LE;
+                fmt |= SNDRV_PCM_FMTBIT_S20_LE;
+                /* Fall-through */
+        case 16:
+                fmt |= SNDRV_PCM_FMTBIT_S16_LE;
+                /* Fall-through */
+        case 8:
+                fmt |= SNDRV_PCM_FMTBIT_S8;
                 break;
         default:
                 dev_err(dai->dev, "unsupported slot width: %d\n", slot_width);
                 return -EINVAL;
         }
 
+        iface->slot_width = slot_width;
+
+        /* Amend the dai driver and let dpcm merge do its job */
+        if (tx) {
+                tx->mask = tx_mask;
+                dai->driver->playback.channels_max = tx_slots;
+                dai->driver->playback.formats = fmt;
+        }
+
+        if (rx) {
+                rx->mask = rx_mask;
+                dai->driver->capture.channels_max = rx_slots;
+                dai->driver->capture.formats = fmt;
+        }
+
         return 0;
 }
 EXPORT_SYMBOL_GPL(axg_tdm_set_tdm_slots);
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index 81b09d740ed9..6384bb6dacfd 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -356,7 +356,7 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev)
         if (ret)
                 goto out;
 
-        ret = snd_soc_register_component(&pdev->dev, &nuc900_ac97_component,
+        ret = devm_snd_soc_register_component(&pdev->dev, &nuc900_ac97_component,
                                          &nuc900_ac97_dai, 1);
         if (ret)
                 goto out;
@@ -373,8 +373,6 @@ out:
 
 static int nuc900_ac97_drvremove(struct platform_device *pdev)
 {
-        snd_soc_unregister_component(&pdev->dev);
-
         nuc900_ac97_data = NULL;
         snd_soc_set_ac97_ops(NULL);
 
diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c
index 8a99a8837dc9..673a9eb153b2 100644
--- a/sound/soc/omap/omap-hdmi-audio.c
+++ b/sound/soc/omap/omap-hdmi-audio.c
@@ -348,7 +348,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
         default:
                 return -EINVAL;
         }
-        ret = snd_soc_register_component(ad->dssdev, &omap_hdmi_component,
+        ret = devm_snd_soc_register_component(ad->dssdev, &omap_hdmi_component,
                                          dai_drv, 1);
         if (ret)
                 return ret;
@@ -383,7 +383,6 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
         ret = snd_soc_register_card(card);
         if (ret) {
                 dev_err(dev, "snd_soc_register_card failed (%d)\n", ret);
-                snd_soc_unregister_component(ad->dssdev);
                 return ret;
         }
 
@@ -400,7 +399,6 @@ static int omap_hdmi_audio_remove(struct platform_device *pdev)
         struct hdmi_audio_data *ad = platform_get_drvdata(pdev);
 
         snd_soc_unregister_card(ad->card);
-        snd_soc_unregister_component(ad->dssdev);
         return 0;
 }
 
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 776e148b0aa2..943b44de1464 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -19,14 +19,13 @@ config SND_MMP_SOC
 
 config SND_PXA2XX_AC97
         tristate
-        select SND_AC97_CODEC
 
 config SND_PXA2XX_SOC_AC97
         tristate
-        select AC97_BUS
+        select AC97_BUS_NEW
         select SND_PXA2XX_LIB
         select SND_PXA2XX_LIB_AC97
-        select SND_SOC_AC97_BUS
+        select SND_SOC_AC97_BUS_NEW
 
 config SND_PXA2XX_SOC_I2S
         select SND_PXA2XX_LIB
@@ -80,6 +79,7 @@ config SND_PXA2XX_SOC_TOSA
         tristate "SoC AC97 Audio support for Tosa"
         depends on SND_PXA2XX_SOC && MACH_TOSA
         depends on MFD_TC6393XB
+        depends on !AC97_BUS
         select SND_PXA2XX_SOC_AC97
         select SND_SOC_WM9712
         help
@@ -89,6 +89,7 @@ config SND_PXA2XX_SOC_TOSA
 config SND_PXA2XX_SOC_E740
         tristate "SoC AC97 Audio support for e740"
         depends on SND_PXA2XX_SOC && MACH_E740
+        depends on !AC97_BUS
         select SND_SOC_WM9705
         select SND_PXA2XX_SOC_AC97
         help
@@ -98,6 +99,7 @@ config SND_PXA2XX_SOC_E740
 config SND_PXA2XX_SOC_E750
         tristate "SoC AC97 Audio support for e750"
         depends on SND_PXA2XX_SOC && MACH_E750
+        depends on !AC97_BUS
         select SND_SOC_WM9705
         select SND_PXA2XX_SOC_AC97
         help
@@ -107,6 +109,7 @@ config SND_PXA2XX_SOC_E750
 config SND_PXA2XX_SOC_E800
         tristate "SoC AC97 Audio support for e800"
         depends on SND_PXA2XX_SOC && MACH_E800
+        depends on !AC97_BUS
         select SND_SOC_WM9712
         select SND_PXA2XX_SOC_AC97
         help
@@ -117,6 +120,7 @@ config SND_PXA2XX_SOC_EM_X270
         tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300"
         depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \
                         MACH_CM_X300)
+        depends on !AC97_BUS
         select SND_PXA2XX_SOC_AC97
         select SND_SOC_WM9712
         help
@@ -127,6 +131,7 @@ config SND_PXA2XX_SOC_PALM27X
         bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive"
         depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \
                         MACH_PALMT5 || MACH_PALMTE2)
+        depends on !AC97_BUS
         select SND_PXA2XX_SOC_AC97
         select SND_SOC_WM9712
         help
@@ -156,6 +161,7 @@ config SND_SOC_TTC_DKB
 config SND_SOC_ZYLONITE
         tristate "SoC Audio support for Marvell Zylonite"
         depends on SND_PXA2XX_SOC && MACH_ZYLONITE
+        depends on !AC97_BUS
         select SND_PXA2XX_SOC_AC97
         select SND_PXA_SOC_SSP
         select SND_SOC_WM9713
@@ -195,6 +201,7 @@ config SND_PXA2XX_SOC_MAGICIAN
 config SND_PXA2XX_SOC_MIOA701
         tristate "SoC Audio support for MIO A701"
         depends on SND_PXA2XX_SOC && MACH_MIOA701
+        depends on !AC97_BUS
         select SND_PXA2XX_SOC_AC97
         select SND_SOC_WM9713
         help
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 69033e1a84e6..adcf8ba9d287 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -103,6 +103,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
                 pxa_ssp_disable(ssp);
         }
 
+        if (priv->extclk)
+                clk_prepare_enable(priv->extclk);
+
         dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL);
         if (!dma)
                 return -ENOMEM;
@@ -125,6 +128,9 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
                 clk_disable_unprepare(ssp->clk);
         }
 
+        if (priv->extclk)
+                clk_disable_unprepare(priv->extclk);
+
         kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
         snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
 }
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 9f779657bc86..f8a3aa6c6d4e 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -17,6 +17,7 @@
 #include <linux/dmaengine.h>
 #include <linux/dma/pxa-dma.h>
 
+#include <sound/ac97/controller.h>
 #include <sound/core.h>
 #include <sound/ac97_codec.h>
 #include <sound/soc.h>
@@ -27,43 +28,35 @@
 #include <mach/regs-ac97.h>
 #include <mach/audio.h>
 
-static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
+static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv)
 {
         pxa2xx_ac97_try_warm_reset();
 
         pxa2xx_ac97_finish_reset();
 }
 
-static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97)
+static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv)
 {
         pxa2xx_ac97_try_cold_reset();
 
         pxa2xx_ac97_finish_reset();
 }
 
-static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97,
-                                              unsigned short reg)
+static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot,
+                                  unsigned short reg)
 {
-        int ret;
-
-        ret = pxa2xx_ac97_read(ac97->num, reg);
-        if (ret < 0)
-                return 0;
-        else
-                return (unsigned short)(ret & 0xffff);
+        return pxa2xx_ac97_read(slot, reg);
 }
 
-static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97,
-                                     unsigned short reg, unsigned short val)
+static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot,
+                                   unsigned short reg, unsigned short val)
 {
-        int ret;
-
-        ret = pxa2xx_ac97_write(ac97->num, reg, val);
+        return pxa2xx_ac97_write(slot, reg, val);
 }
 
-static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
-        .read   = pxa2xx_ac97_legacy_read,
-        .write  = pxa2xx_ac97_legacy_write,
+static struct ac97_controller_ops pxa2xx_ac97_ops = {
+        .read   = pxa2xx_ac97_read_actrl,
+        .write  = pxa2xx_ac97_write_actrl,
         .warm_reset     = pxa2xx_ac97_warm_reset,
         .reset  = pxa2xx_ac97_cold_reset,
 };
@@ -233,6 +226,9 @@ MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids);
 static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 {
         int ret;
+        struct ac97_controller *ctrl;
+        pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data;
+        void **codecs_pdata;
 
         if (pdev->id != -1) {
                 dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n");
@@ -245,10 +241,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
                 return ret;
         }
 
-        ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops);
-        if (ret != 0)
-                return ret;
+        codecs_pdata = pdata ? pdata->codec_pdata : NULL;
+        ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev,
+                                            AC97_SLOTS_AVAILABLE_ALL,
+                                            codecs_pdata);
+        if (IS_ERR(ctrl))
+                return PTR_ERR(ctrl);
 
+        platform_set_drvdata(pdev, ctrl);
         /* Punt most of the init to the SoC probe; we may need the machine
          * driver to do interesting things with the clocking to get us up
          * and running.
@@ -259,8 +259,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 
 static int pxa2xx_ac97_dev_remove(struct platform_device *pdev)
 {
+        struct ac97_controller *ctrl = platform_get_drvdata(pdev);
+
         snd_soc_unregister_component(&pdev->dev);
-        snd_soc_set_ac97_ops(NULL);
+        snd_ac97_controller_unregister(ctrl);
         pxa2xx_ac97_hw_remove(pdev);
         return 0;
 }
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 1543e85629f8..fb45f396ab4a 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -25,13 +25,12 @@ static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 
 static void apq8096_add_be_ops(struct snd_soc_card *card)
 {
-        struct snd_soc_dai_link *link = card->dai_link;
-        int i, num_links = card->num_links;
+        struct snd_soc_dai_link *link;
+        int i;
 
-        for (i = 0; i < num_links; i++) {
+        for_each_card_prelinks(card, i, link) {
                 if (link->no_pcm == 1)
                         link->be_hw_params_fixup = apq8096_be_hw_params_fixup;
-                link++;
         }
 }
 
diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c
index 932c3ebfd252..da242515e146 100644
--- a/sound/soc/qcom/qdsp6/q6adm.c
+++ b/sound/soc/qcom/qdsp6/q6adm.c
@@ -2,25 +2,24 @@
 // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
 // Copyright (c) 2018, Linaro Limited
 
-#include <linux/slab.h>
-#include <linux/wait.h>
-#include <linux/kernel.h>
 #include <linux/device.h>
-#include <linux/module.h>
-#include <linux/sched.h>
 #include <linux/jiffies.h>
+#include <linux/kernel.h>
+#include <linux/kref.h>
+#include <linux/module.h>
 #include <linux/of.h>
 #include <linux/of_platform.h>
-#include <linux/kref.h>
-#include <linux/wait.h>
-#include <linux/soc/qcom/apr.h>
 #include <linux/platform_device.h>
+#include <linux/sched.h>
+#include <linux/slab.h>
+#include <linux/soc/qcom/apr.h>
+#include <linux/wait.h>
 #include <sound/asound.h>
 #include "q6adm.h"
 #include "q6afe.h"
 #include "q6core.h"
-#include "q6dsp-errno.h"
 #include "q6dsp-common.h"
+#include "q6dsp-errno.h"
 
 #define ADM_CMD_DEVICE_OPEN_V5          0x00010326
 #define ADM_CMDRSP_DEVICE_OPEN_V5       0x00010329
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 9db9a2944ef2..a16c71c03058 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -8,7 +8,6 @@
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <sound/soc.h>
-#include <sound/soc.h>
 #include <sound/soc-dapm.h>
 #include <sound/pcm.h>
 #include <asm/dma.h>
@@ -319,10 +318,11 @@ static int q6asm_dai_open(struct snd_pcm_substream *substream)
         prtd->audio_client = q6asm_audio_client_alloc(dev,
                                 (q6asm_cb)event_handler, prtd, stream_id,
                                 LEGACY_PCM_MODE);
-        if (!prtd->audio_client) {
+        if (IS_ERR(prtd->audio_client)) {
                 pr_info("%s: Could not allocate memory\n", __func__);
+                ret = PTR_ERR(prtd->audio_client);
                 kfree(prtd);
-                return -ENOMEM;
+                return ret;
         }
 
         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -493,7 +493,7 @@ static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)
                 }
         }
 
-        return ret;
+        return 0;
 }
 
 static void q6asm_dai_pcm_free(struct snd_pcm *pcm)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 2b2c7233bb5f..e1cfa846a1dc 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -11,7 +11,6 @@
 #include <linux/spinlock.h>
 #include <linux/kref.h>
 #include <linux/of.h>
-#include <linux/of_platform.h>
 #include <uapi/sound/asound.h>
 #include <linux/delay.h>
 #include <linux/slab.h>
diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c
index 06f03a5fe9bd..cdfc8ab6cfc0 100644
--- a/sound/soc/qcom/qdsp6/q6core.c
+++ b/sound/soc/qcom/qdsp6/q6core.c
@@ -10,7 +10,6 @@
 #include <linux/of.h>
 #include <linux/of_platform.h>
 #include <linux/jiffies.h>
-#include <linux/wait.h>
 #include <linux/soc/qcom/apr.h>
 #include "q6core.h"
 #include "q6dsp-errno.h"
@@ -105,12 +104,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data)
                 bytes = sizeof(*fwk) + fwk->num_services *
                                 sizeof(fwk->svc_api_info[0]);
 
-                core->fwk_version = kzalloc(bytes, GFP_ATOMIC);
+                core->fwk_version = kmemdup(data->payload, bytes, GFP_ATOMIC);
                 if (!core->fwk_version)
                         return -ENOMEM;
 
-                memcpy(core->fwk_version, data->payload, bytes);
-
                 core->fwk_version_supported = true;
                 core->resp_received = true;
 
@@ -124,12 +121,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data)
 
                 len = sizeof(*v) + v->num_services * sizeof(v->svc_api_info[0]);
 
-                core->svc_version = kzalloc(len, GFP_ATOMIC);
+                core->svc_version = kmemdup(data->payload, len, GFP_ATOMIC);
                 if (!core->svc_version)
                         return -ENOMEM;
 
-                memcpy(core->svc_version, data->payload, len);
-
                 core->get_version_supported = true;
                 core->resp_received = true;
 
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 2a781d87ee65..9effbecc571f 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -195,15 +195,14 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 
 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;
+        struct snd_soc_dai_link *link;
+        int i;
 
-        for (i = 0; i < num_links; i++) {
+        for_each_card_prelinks(card, i, link) {
                 if (link->no_pcm == 1) {
                         link->ops = &sdm845_be_ops;
                         link->be_hw_params_fixup = sdm845_be_hw_params_fixup;
                 }
-                link++;
         }
 }
 
diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c
index 929b3fe289b0..a472d5eb2950 100644
--- a/sound/soc/rockchip/rk3288_hdmi_analog.c
+++ b/sound/soc/rockchip/rk3288_hdmi_analog.c
@@ -286,7 +286,6 @@ static struct platform_driver rockchip_sound_driver = {
         .probe = snd_rk_mc_probe,
         .driver = {
                 .name = DRV_NAME,
-                .owner = THIS_MODULE,
                 .pm = &snd_soc_pm_ops,
                 .of_match_table = rockchip_sound_of_match,
         },
diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c
index f77538319221..9e7b5fa4cf59 100644
--- a/sound/soc/rockchip/rockchip_pcm.c
+++ b/sound/soc/rockchip/rockchip_pcm.c
@@ -21,7 +21,8 @@ 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,
+                                  SNDRV_PCM_INFO_RESUME |
+                                  SNDRV_PCM_INFO_INTERLEAVED,
         .period_bytes_min       = 32,
         .period_bytes_max       = 8192,
         .periods_min            = 1,
diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c
index 43332c32d7e9..dc93941e01c3 100644
--- a/sound/soc/samsung/tm2_wm5110.c
+++ b/sound/soc/samsung/tm2_wm5110.c
@@ -491,6 +491,7 @@ static int tm2_probe(struct platform_device *pdev)
         struct snd_soc_card *card = &tm2_card;
         struct tm2_machine_priv *priv;
         struct of_phandle_args args;
+        struct snd_soc_dai_link *dai_link;
         int num_codecs, ret, i;
 
         priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -558,18 +559,18 @@ static int tm2_probe(struct platform_device *pdev)
         }
 
         /* Initialize WM5110 - I2S and HDMI - I2S1 DAI links */
-        for (i = 0; i < card->num_links; i++) {
+        for_each_card_prelinks(card, i, dai_link) {
                 unsigned int dai_index = 0; /* WM5110 */
 
-                card->dai_link[i].cpu_name = NULL;
-                card->dai_link[i].platform_name = NULL;
+                dai_link->cpu_name = NULL;
+                dai_link->platform_name = NULL;
 
                 if (num_codecs > 1 && i == card->num_links - 1)
                         dai_index = 1; /* HDMI */
 
-                card->dai_link[i].codec_of_node = codec_dai_node[dai_index];
-                card->dai_link[i].cpu_of_node = cpu_dai_node[dai_index];
-                card->dai_link[i].platform_of_node = cpu_dai_node[dai_index];
+                dai_link->codec_of_node = codec_dai_node[dai_index];
+                dai_link->cpu_of_node = cpu_dai_node[dai_index];
+                dai_link->platform_of_node = cpu_dai_node[dai_index];
         }
 
         if (num_codecs > 1) {
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index c2b496398e6b..17622ceb98c0 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -319,13 +319,12 @@ static int hac_soc_platform_probe(struct platform_device *pdev)
         if (ret != 0)
                 return ret;
 
-        return snd_soc_register_component(&pdev->dev, &sh4_hac_component,
+        return devm_snd_soc_register_component(&pdev->dev, &sh4_hac_component,
                                           sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
 }
 
 static int hac_soc_platform_remove(struct platform_device *pdev)
 {
-        snd_soc_unregister_component(&pdev->dev);
         snd_soc_set_ac97_ops(NULL);
         return 0;
 }
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 051f96405346..28327dd2c6cb 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -582,7 +582,7 @@ static void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct rsnd_adg *adg)
         int i;
 
         for_each_rsnd_clk(clk, adg, i)
-                dev_dbg(dev, "%s    : %p : %ld\n",
+                dev_dbg(dev, "%s    : %pa : %ld\n",
                         clk_name[i], clk, clk_get_rate(clk));
 
         dev_dbg(dev, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n",
@@ -595,7 +595,7 @@ static void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct rsnd_adg *adg)
          * by BRGCKR::BRGCKR_31
          */
         for_each_rsnd_clkout(clk, adg, i)
-                dev_dbg(dev, "clkout %d : %p : %ld\n", i,
+                dev_dbg(dev, "clkout %d : %pa : %ld\n", i,
                         clk, clk_get_rate(clk));
 }
 #else
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index d23c2bbff0cf..f930f51b686f 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -102,7 +102,9 @@
 #include "rsnd.h"
 
 #define RSND_RATES SNDRV_PCM_RATE_8000_192000
-#define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
+#define RSND_FMTS (SNDRV_PCM_FMTBIT_S8 |\
+                   SNDRV_PCM_FMTBIT_S16_LE |\
+                   SNDRV_PCM_FMTBIT_S24_LE)
 
 static const struct of_device_id rsnd_of_match[] = {
         { .compatible = "renesas,rcar_sound-gen1", .data = (void *)RSND_GEN1 },
@@ -280,6 +282,8 @@ u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io)
         struct device *dev = rsnd_priv_to_dev(priv);
 
         switch (snd_pcm_format_width(runtime->format)) {
+        case 8:
+                return 16 << 16;
         case 16:
                 return 8 << 16;
         case 24:
@@ -331,7 +335,7 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io)
                 target = cmd ? cmd : ssiu;
         }
 
-        /* Non target mod or 24bit data needs normal DALIGN */
+        /* Non target mod or non 16bit needs normal DALIGN */
         if ((snd_pcm_format_width(runtime->format) != 16) ||
             (mod != target))
                 return 0x76543210;
@@ -367,7 +371,7 @@ u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod)
          * HW    24bit data is located as 0x******00
          *
          */
-        if (snd_pcm_format_width(runtime->format) == 16)
+        if (snd_pcm_format_width(runtime->format) != 24)
                 return 0;
 
         for (i = 0; i < ARRAY_SIZE(playback_mods); i++) {
@@ -540,6 +544,14 @@ int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai,
         return rdai->ssi_lane;
 }
 
+int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width)
+{
+        if (width > 0)
+                rdai->chan_width = width;
+
+        return rdai->chan_width;
+}
+
 struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id)
 {
         if ((id < 0) || (id >= rsnd_rdai_nr(priv)))
@@ -681,6 +693,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
                 rdai->frm_clk_inv = 0;
                 break;
         case SND_SOC_DAIFMT_LEFT_J:
+        case SND_SOC_DAIFMT_DSP_B:
                 rdai->sys_delay = 1;
                 rdai->data_alignment = 0;
                 rdai->frm_clk_inv = 1;
@@ -690,6 +703,11 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
                 rdai->data_alignment = 1;
                 rdai->frm_clk_inv = 1;
                 break;
+        case SND_SOC_DAIFMT_DSP_A:
+                rdai->sys_delay = 0;
+                rdai->data_alignment = 0;
+                rdai->frm_clk_inv = 1;
+                break;
         }
 
         /* set clock inversion */
@@ -720,6 +738,16 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai,
         struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
         struct device *dev = rsnd_priv_to_dev(priv);
 
+        switch (slot_width) {
+        case 16:
+        case 24:
+        case 32:
+                break;
+        default:
+                /* use default */
+                slot_width = 32;
+        }
+
         switch (slots) {
         case 2:
         case 6:
@@ -727,6 +755,7 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai,
                 /* TDM Extend Mode */
                 rsnd_rdai_channels_set(rdai, slots);
                 rsnd_rdai_ssi_lane_set(rdai, 1);
+                rsnd_rdai_width_set(rdai, slot_width);
                 break;
         default:
                 dev_err(dev, "unsupported TDM slots (%d)\n", slots);
@@ -755,7 +784,7 @@ static unsigned int rsnd_soc_hw_rate_list[] = {
         192000,
 };
 
-static int rsnd_soc_hw_rule(struct rsnd_priv *priv,
+static int rsnd_soc_hw_rule(struct rsnd_dai *rdai,
                             unsigned int *list, int list_num,
                             struct snd_interval *baseline, struct snd_interval *iv)
 {
@@ -772,14 +801,14 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv,
                 if (!snd_interval_test(iv, list[i]))
                         continue;
 
-                rate = rsnd_ssi_clk_query(priv,
+                rate = rsnd_ssi_clk_query(rdai,
                                           baseline->min, list[i], NULL);
                 if (rate > 0) {
                         p.min = min(p.min, list[i]);
                         p.max = max(p.max, list[i]);
                 }
 
-                rate = rsnd_ssi_clk_query(priv,
+                rate = rsnd_ssi_clk_query(rdai,
                                           baseline->max, list[i], NULL);
                 if (rate > 0) {
                         p.min = min(p.min, list[i]);
@@ -790,17 +819,14 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv,
         return snd_interval_refine(iv, &p);
 }
 
-static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params,
-                                   struct snd_pcm_hw_rule *rule,
-                                   int is_play)
+static int rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params,
+                                 struct snd_pcm_hw_rule *rule)
 {
         struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
         struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
         struct snd_interval ic;
-        struct snd_soc_dai *dai = rule->private;
-        struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
-        struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
-        struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture;
+        struct rsnd_dai_stream *io = rule->private;
+        struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
 
         /*
          * possible sampling rate limitation is same as
@@ -811,34 +837,19 @@ static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params,
         ic.min =
         ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params);
 
-        return rsnd_soc_hw_rule(priv, rsnd_soc_hw_rate_list,
+        return rsnd_soc_hw_rule(rdai, rsnd_soc_hw_rate_list,
                                 ARRAY_SIZE(rsnd_soc_hw_rate_list),
                                 &ic, ir);
 }
 
-static int rsnd_soc_hw_rule_rate_playback(struct snd_pcm_hw_params *params,
-                                 struct snd_pcm_hw_rule *rule)
-{
-        return __rsnd_soc_hw_rule_rate(params, rule, 1);
-}
-
-static int rsnd_soc_hw_rule_rate_capture(struct snd_pcm_hw_params *params,
-                                          struct snd_pcm_hw_rule *rule)
-{
-        return __rsnd_soc_hw_rule_rate(params, rule, 0);
-}
-
-static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params,
-                                       struct snd_pcm_hw_rule *rule,
-                                       int is_play)
+static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params,
+                                     struct snd_pcm_hw_rule *rule)
 {
         struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
         struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
         struct snd_interval ic;
-        struct snd_soc_dai *dai = rule->private;
-        struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
-        struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
-        struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture;
+        struct rsnd_dai_stream *io = rule->private;
+        struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
 
         /*
          * possible sampling rate limitation is same as
@@ -849,23 +860,11 @@ static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params,
         ic.min =
         ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params);
 
-        return rsnd_soc_hw_rule(priv, rsnd_soc_hw_channels_list,
+        return rsnd_soc_hw_rule(rdai, rsnd_soc_hw_channels_list,
                                 ARRAY_SIZE(rsnd_soc_hw_channels_list),
                                 ir, &ic);
 }
 
-static int rsnd_soc_hw_rule_channels_playback(struct snd_pcm_hw_params *params,
-                                              struct snd_pcm_hw_rule *rule)
-{
-        return __rsnd_soc_hw_rule_channels(params, rule, 1);
-}
-
-static int rsnd_soc_hw_rule_channels_capture(struct snd_pcm_hw_params *params,
-                                             struct snd_pcm_hw_rule *rule)
-{
-        return __rsnd_soc_hw_rule_channels(params, rule, 0);
-}
-
 static const struct snd_pcm_hardware rsnd_pcm_hardware = {
         .info =         SNDRV_PCM_INFO_INTERLEAVED      |
                         SNDRV_PCM_INFO_MMAP             |
@@ -882,12 +881,10 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream,
                                 struct snd_soc_dai *dai)
 {
         struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
-        struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
         struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream);
         struct snd_pcm_hw_constraint_list *constraint = &rdai->constraint;
         struct snd_pcm_runtime *runtime = substream->runtime;
         unsigned int max_channels = rsnd_rdai_channels_get(rdai);
-        int ret;
         int i;
 
         rsnd_dai_stream_init(io, substream);
@@ -922,25 +919,16 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream,
                 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 
                 snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
-                                    is_play ? rsnd_soc_hw_rule_rate_playback :
-                                              rsnd_soc_hw_rule_rate_capture,
-                                    dai,
+                                    rsnd_soc_hw_rule_rate,
+                                    is_play ? &rdai->playback : &rdai->capture,
                                     SNDRV_PCM_HW_PARAM_CHANNELS, -1);
                 snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-                                    is_play ? rsnd_soc_hw_rule_channels_playback :
-                                              rsnd_soc_hw_rule_channels_capture,
-                                    dai,
+                                    rsnd_soc_hw_rule_channels,
+                                    is_play ? &rdai->playback : &rdai->capture,
                                     SNDRV_PCM_HW_PARAM_RATE, -1);
         }
 
-        /*
-         * call rsnd_dai_call without spinlock
-         */
-        ret = rsnd_dai_call(nolock_start, io, priv);
-        if (ret < 0)
-                rsnd_dai_call(nolock_stop, io, priv);
-
-        return ret;
+        return 0;
 }
 
 static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream,
@@ -953,7 +941,7 @@ static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream,
         /*
          * call rsnd_dai_call without spinlock
          */
-        rsnd_dai_call(nolock_stop, io, priv);
+        rsnd_dai_call(cleanup, io, priv);
 
         rsnd_dai_stream_quit(io);
 }
@@ -1083,6 +1071,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv,
         rdai->capture.rdai              = rdai;
         rsnd_rdai_channels_set(rdai, 2); /* default 2ch */
         rsnd_rdai_ssi_lane_set(rdai, 1); /* default 1lane */
+        rsnd_rdai_width_set(rdai, 32);   /* default 32bit width */
 
         for (io_i = 0;; io_i++) {
                 playback = of_parse_phandle(dai_np, "playback", io_i);
@@ -1274,8 +1263,15 @@ int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io)
 int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io)
 {
         struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+        struct rsnd_priv *priv = rsnd_io_to_priv(io);
+        struct device *dev = rsnd_priv_to_dev(priv);
+
+        if (!runtime) {
+                dev_warn(dev, "Can't update kctrl when idle\n");
+                return 0;
+        }
 
-        return !!runtime;
+        return 1;
 }
 
 struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg)
diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c
index 6a55aa753003..ad702377a6c3 100644
--- a/sound/soc/sh/rcar/ctu.c
+++ b/sound/soc/sh/rcar/ctu.c
@@ -258,7 +258,7 @@ static int rsnd_ctu_hw_params(struct rsnd_mod *mod,
                 struct snd_pcm_hw_params *be_params;
                 int stream = substream->stream;
 
-                list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+                for_each_dpcm_be(fe, stream, dpcm) {
                         be_params = &dpcm->hw_params;
                         if (params_channels(fe_params) != params_channels(be_params))
                                 ctu->channels = params_channels(be_params);
diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index d65ea7bc4dac..6d1947515dc8 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -106,9 +106,9 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod,
         return 0;
 }
 
-static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod,
-                                   struct rsnd_dai_stream *io,
-                                   struct rsnd_priv *priv)
+static int rsnd_dmaen_cleanup(struct rsnd_mod *mod,
+                              struct rsnd_dai_stream *io,
+                              struct rsnd_priv *priv)
 {
         struct rsnd_dma *dma = rsnd_mod_to_dma(mod);
         struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma);
@@ -116,7 +116,7 @@ static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod,
         /*
          * DMAEngine release uses mutex lock.
          * Thus, it shouldn't be called under spinlock.
-         * Let's call it under nolock_start
+         * Let's call it under prepare
          */
         if (dmaen->chan)
                 dma_release_channel(dmaen->chan);
@@ -126,23 +126,22 @@ static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod,
         return 0;
 }
 
-static int rsnd_dmaen_nolock_start(struct rsnd_mod *mod,
-                            struct rsnd_dai_stream *io,
-                            struct rsnd_priv *priv)
+static int rsnd_dmaen_prepare(struct rsnd_mod *mod,
+                              struct rsnd_dai_stream *io,
+                              struct rsnd_priv *priv)
 {
         struct rsnd_dma *dma = rsnd_mod_to_dma(mod);
         struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma);
         struct device *dev = rsnd_priv_to_dev(priv);
 
-        if (dmaen->chan) {
-                dev_err(dev, "it already has dma channel\n");
-                return -EIO;
-        }
+        /* maybe suspended */
+        if (dmaen->chan)
+                return 0;
 
         /*
          * DMAEngine request uses mutex lock.
          * Thus, it shouldn't be called under spinlock.
-         * Let's call it under nolock_start
+         * Let's call it under prepare
          */
         dmaen->chan = rsnd_dmaen_request_channel(io,
                                                  dma->mod_from,
@@ -291,8 +290,8 @@ static int rsnd_dmaen_pointer(struct rsnd_mod *mod,
 
 static struct rsnd_mod_ops rsnd_dmaen_ops = {
         .name   = "audmac",
-        .nolock_start = rsnd_dmaen_nolock_start,
-        .nolock_stop  = rsnd_dmaen_nolock_stop,
+        .prepare = rsnd_dmaen_prepare,
+        .cleanup = rsnd_dmaen_cleanup,
         .start  = rsnd_dmaen_start,
         .stop   = rsnd_dmaen_stop,
         .pointer= rsnd_dmaen_pointer,
@@ -302,16 +301,26 @@ static struct rsnd_mod_ops rsnd_dmaen_ops = {
  *              Audio DMAC peri peri
  */
 static const u8 gen2_id_table_ssiu[] = {
-        0x00, /* SSI00 */
-        0x04, /* SSI10 */
-        0x08, /* SSI20 */
-        0x0c, /* SSI3  */
-        0x0d, /* SSI4  */
-        0x0e, /* SSI5  */
-        0x0f, /* SSI6  */
-        0x10, /* SSI7  */
-        0x11, /* SSI8  */
-        0x12, /* SSI90 */
+        /* SSI00 ~ SSI07 */
+        0x00, 0x01, 0x02, 0x03, 0x39, 0x3a, 0x3b, 0x3c,
+        /* SSI10 ~ SSI17 */
+        0x04, 0x05, 0x06, 0x07, 0x3d, 0x3e, 0x3f, 0x40,
+        /* SSI20 ~ SSI27 */
+        0x08, 0x09, 0x0a, 0x0b, 0x41, 0x42, 0x43, 0x44,
+        /* SSI30 ~ SSI37 */
+        0x0c, 0x45, 0x46, 0x47, 0x48, 0x49, 0x4a, 0x4b,
+        /* SSI40 ~ SSI47 */
+        0x0d, 0x4c, 0x4d, 0x4e, 0x4f, 0x50, 0x51, 0x52,
+        /* SSI5 */
+        0x0e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+        /* SSI6 */
+        0x0f, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+        /* SSI7 */
+        0x10, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+        /* SSI8 */
+        0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+        /* SSI90 ~ SSI97 */
+        0x12, 0x13, 0x14, 0x15, 0x53, 0x54, 0x55, 0x56,
 };
 static const u8 gen2_id_table_scu[] = {
         0x2d, /* SCU_SRCI0 */
@@ -337,18 +346,23 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io,
         struct rsnd_mod *src = rsnd_io_to_mod_src(io);
         struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io);
         const u8 *entry = NULL;
-        int id = rsnd_mod_id(mod);
+        int id = 255;
         int size = 0;
 
         if (mod == ssi) {
+                int busif = rsnd_ssi_get_busif(io);
+
                 entry = gen2_id_table_ssiu;
                 size = ARRAY_SIZE(gen2_id_table_ssiu);
+                id = (rsnd_mod_id(mod) * 8) + busif;
         } else if (mod == src) {
                 entry = gen2_id_table_scu;
                 size = ARRAY_SIZE(gen2_id_table_scu);
+                id = rsnd_mod_id(mod);
         } else if (mod == dvc) {
                 entry = gen2_id_table_cmd;
                 size = ARRAY_SIZE(gen2_id_table_cmd);
+                id = rsnd_mod_id(mod);
         }
 
         if ((!entry) || (size <= id)) {
@@ -382,7 +396,7 @@ static void rsnd_dmapp_write(struct rsnd_dma *dma, u32 data, u32 reg)
         struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv);
         struct device *dev = rsnd_priv_to_dev(priv);
 
-        dev_dbg(dev, "w %p : %08x\n", rsnd_dmapp_addr(dmac, dma, reg), data);
+        dev_dbg(dev, "w 0x%px : %08x\n", rsnd_dmapp_addr(dmac, dma, reg), data);
 
         iowrite32(data, rsnd_dmapp_addr(dmac, dma, reg));
 }
@@ -491,11 +505,11 @@ static struct rsnd_mod_ops rsnd_dmapp_ops = {
 #define RDMA_SSI_I_N(addr, i)   (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8)
 #define RDMA_SSI_O_N(addr, i)   (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc)
 
-#define RDMA_SSIU_I_N(addr, i)  (addr ##_reg - 0x00441000 + (0x1000 * i))
-#define RDMA_SSIU_O_N(addr, i)  (addr ##_reg - 0x00441000 + (0x1000 * i))
+#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400))
+#define RDMA_SSIU_O_N(addr, i, j) RDMA_SSIU_I_N(addr, i, j)
 
-#define RDMA_SSIU_I_P(addr, i)  (addr ##_reg - 0x00141000 + (0x1000 * i))
-#define RDMA_SSIU_O_P(addr, i)  (addr ##_reg - 0x00141000 + (0x1000 * i))
+#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400))
+#define RDMA_SSIU_O_P(addr, i, j) RDMA_SSIU_I_P(addr, i, j)
 
 #define RDMA_SRC_I_N(addr, i)   (addr ##_reg - 0x00500000 + (0x400 * i))
 #define RDMA_SRC_O_N(addr, i)   (addr ##_reg - 0x004fc000 + (0x400 * i))
@@ -521,6 +535,7 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io,
                       !!rsnd_io_to_mod_mix(io) ||
                       !!rsnd_io_to_mod_ctu(io);
         int id = rsnd_mod_id(mod);
+        int busif = rsnd_ssi_get_busif(io);
         struct dma_addr {
                 dma_addr_t out_addr;
                 dma_addr_t in_addr;
@@ -537,25 +552,35 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io,
                 },
                 /* SSI */
                 /* Capture */
-                {{{ RDMA_SSI_O_N(ssi, id),      0 },
-                  { RDMA_SSIU_O_P(ssi, id),     0 },
-                  { RDMA_SSIU_O_P(ssi, id),     0 } },
+                {{{ RDMA_SSI_O_N(ssi, id),              0 },
+                  { RDMA_SSIU_O_P(ssi, id, busif),      0 },
+                  { RDMA_SSIU_O_P(ssi, id, busif),      0 } },
                  /* Playback */
-                 {{ 0,                          RDMA_SSI_I_N(ssi, id) },
-                  { 0,                          RDMA_SSIU_I_P(ssi, id) },
-                  { 0,                          RDMA_SSIU_I_P(ssi, id) } }
+                 {{ 0,                  RDMA_SSI_I_N(ssi, id) },
+                  { 0,                  RDMA_SSIU_I_P(ssi, id, busif) },
+                  { 0,                  RDMA_SSIU_I_P(ssi, id, busif) } }
                 },
                 /* SSIU */
                 /* Capture */
-                {{{ RDMA_SSIU_O_N(ssi, id),     0 },
-                  { RDMA_SSIU_O_P(ssi, id),     0 },
-                  { RDMA_SSIU_O_P(ssi, id),     0 } },
+                {{{ RDMA_SSIU_O_N(ssi, id, busif),      0 },
+                  { RDMA_SSIU_O_P(ssi, id, busif),      0 },
+                  { RDMA_SSIU_O_P(ssi, id, busif),      0 } },
                  /* Playback */
-                 {{ 0,                          RDMA_SSIU_I_N(ssi, id) },
-                  { 0,                          RDMA_SSIU_I_P(ssi, id) },
-                  { 0,                          RDMA_SSIU_I_P(ssi, id) } } },
+                 {{ 0,                  RDMA_SSIU_I_N(ssi, id, busif) },
+                  { 0,                  RDMA_SSIU_I_P(ssi, id, busif) },
+                  { 0,                  RDMA_SSIU_I_P(ssi, id, busif) } } },
         };
 
+        /*
+         * FIXME
+         *
+         * We can't support SSI9-4/5/6/7, because its address is
+         * out of calculation rule
+         */
+        if ((id == 9) && (busif >= 4))
+                dev_err(dev, "This driver doesn't support SSI%d-%d, so far",
+                        id, busif);
+
         /* it shouldn't happen */
         if (use_cmd && !use_src)
                 dev_err(dev, "DVC is selected without SRC\n");
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 0230301fe078..1f7881cc16b2 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -219,12 +219,33 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv)
                 RSND_GEN_S_REG(HDMI1_SEL,       0x9e4),
 
                 /* FIXME: it needs SSI_MODE2/3 in the future */
-                RSND_GEN_M_REG(SSI_BUSIF_MODE,  0x0,    0x80),
-                RSND_GEN_M_REG(SSI_BUSIF_ADINR, 0x4,    0x80),
-                RSND_GEN_M_REG(SSI_BUSIF_DALIGN,0x8,    0x80),
-                RSND_GEN_M_REG(SSI_MODE,        0xc,    0x80),
-                RSND_GEN_M_REG(SSI_CTRL,        0x10,   0x80),
-                RSND_GEN_M_REG(SSI_INT_ENABLE,  0x18,   0x80),
+                RSND_GEN_M_REG(SSI_BUSIF0_MODE,         0x0,    0x80),
+                RSND_GEN_M_REG(SSI_BUSIF0_ADINR,        0x4,    0x80),
+                RSND_GEN_M_REG(SSI_BUSIF0_DALIGN,       0x8,    0x80),
+                RSND_GEN_M_REG(SSI_BUSIF1_MODE,         0x20,   0x80),
+                RSND_GEN_M_REG(SSI_BUSIF1_ADINR,        0x24,   0x80),
+                RSND_GEN_M_REG(SSI_BUSIF1_DALIGN,       0x28,   0x80),
+                RSND_GEN_M_REG(SSI_BUSIF2_MODE,         0x40,   0x80),
+                RSND_GEN_M_REG(SSI_BUSIF2_ADINR,        0x44,   0x80),
+                RSND_GEN_M_REG(SSI_BUSIF2_DALIGN,       0x48,   0x80),
+                RSND_GEN_M_REG(SSI_BUSIF3_MODE,         0x60,   0x80),
+                RSND_GEN_M_REG(SSI_BUSIF3_ADINR,        0x64,   0x80),
+                RSND_GEN_M_REG(SSI_BUSIF3_DALIGN,       0x68,   0x80),
+                RSND_GEN_M_REG(SSI_BUSIF4_MODE,         0x500,  0x80),
+                RSND_GEN_M_REG(SSI_BUSIF4_ADINR,        0x504,  0x80),
+                RSND_GEN_M_REG(SSI_BUSIF4_DALIGN,       0x508,  0x80),
+                RSND_GEN_M_REG(SSI_BUSIF5_MODE,         0x520,  0x80),
+                RSND_GEN_M_REG(SSI_BUSIF5_ADINR,        0x524,  0x80),
+                RSND_GEN_M_REG(SSI_BUSIF5_DALIGN,       0x528,  0x80),
+                RSND_GEN_M_REG(SSI_BUSIF6_MODE,         0x540,  0x80),
+                RSND_GEN_M_REG(SSI_BUSIF6_ADINR,        0x544,  0x80),
+                RSND_GEN_M_REG(SSI_BUSIF6_DALIGN,       0x548,  0x80),
+                RSND_GEN_M_REG(SSI_BUSIF7_MODE,         0x560,  0x80),
+                RSND_GEN_M_REG(SSI_BUSIF7_ADINR,        0x564,  0x80),
+                RSND_GEN_M_REG(SSI_BUSIF7_DALIGN,       0x568,  0x80),
+                RSND_GEN_M_REG(SSI_MODE,                0xc,    0x80),
+                RSND_GEN_M_REG(SSI_CTRL,                0x10,   0x80),
+                RSND_GEN_M_REG(SSI_INT_ENABLE,          0x18,   0x80),
         };
 
         static const struct rsnd_regmap_field_conf conf_scu[] = {
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 8f7a0abfa751..4464d1d0a042 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -156,9 +156,30 @@ enum rsnd_reg {
         RSND_REG_SSI_MODE2,
         RSND_REG_SSI_CONTROL,
         RSND_REG_SSI_CTRL,
-        RSND_REG_SSI_BUSIF_MODE,
-        RSND_REG_SSI_BUSIF_ADINR,
-        RSND_REG_SSI_BUSIF_DALIGN,
+        RSND_REG_SSI_BUSIF0_MODE,
+        RSND_REG_SSI_BUSIF0_ADINR,
+        RSND_REG_SSI_BUSIF0_DALIGN,
+        RSND_REG_SSI_BUSIF1_MODE,
+        RSND_REG_SSI_BUSIF1_ADINR,
+        RSND_REG_SSI_BUSIF1_DALIGN,
+        RSND_REG_SSI_BUSIF2_MODE,
+        RSND_REG_SSI_BUSIF2_ADINR,
+        RSND_REG_SSI_BUSIF2_DALIGN,
+        RSND_REG_SSI_BUSIF3_MODE,
+        RSND_REG_SSI_BUSIF3_ADINR,
+        RSND_REG_SSI_BUSIF3_DALIGN,
+        RSND_REG_SSI_BUSIF4_MODE,
+        RSND_REG_SSI_BUSIF4_ADINR,
+        RSND_REG_SSI_BUSIF4_DALIGN,
+        RSND_REG_SSI_BUSIF5_MODE,
+        RSND_REG_SSI_BUSIF5_ADINR,
+        RSND_REG_SSI_BUSIF5_DALIGN,
+        RSND_REG_SSI_BUSIF6_MODE,
+        RSND_REG_SSI_BUSIF6_ADINR,
+        RSND_REG_SSI_BUSIF6_DALIGN,
+        RSND_REG_SSI_BUSIF7_MODE,
+        RSND_REG_SSI_BUSIF7_ADINR,
+        RSND_REG_SSI_BUSIF7_DALIGN,
         RSND_REG_SSI_INT_ENABLE,
         RSND_REG_SSI_SYS_STATUS0,
         RSND_REG_SSI_SYS_STATUS1,
@@ -274,15 +295,12 @@ struct rsnd_mod_ops {
         int (*fallback)(struct rsnd_mod *mod,
                         struct rsnd_dai_stream *io,
                         struct rsnd_priv *priv);
-        int (*nolock_start)(struct rsnd_mod *mod,
-                    struct rsnd_dai_stream *io,
-                    struct rsnd_priv *priv);
-        int (*nolock_stop)(struct rsnd_mod *mod,
-                    struct rsnd_dai_stream *io,
-                    struct rsnd_priv *priv);
         int (*prepare)(struct rsnd_mod *mod,
                        struct rsnd_dai_stream *io,
                        struct rsnd_priv *priv);
+        int (*cleanup)(struct rsnd_mod *mod,
+                       struct rsnd_dai_stream *io,
+                       struct rsnd_priv *priv);
 };
 
 struct rsnd_dai_stream;
@@ -300,9 +318,8 @@ struct rsnd_mod {
 /*
  * status
  *
- * 0xH0000CBA
+ * 0xH0000CB0
  *
- * A    0: nolock_start 1: nolock_stop
  * B    0: init         1: quit
  * C    0: start        1: stop
  *
@@ -313,9 +330,8 @@ struct rsnd_mod {
  * H    0: hw_params
  * H    0: pointer
  * H    0: prepare
+ * H    0: cleanup
  */
-#define __rsnd_mod_shift_nolock_start   0
-#define __rsnd_mod_shift_nolock_stop    0
 #define __rsnd_mod_shift_init           4
 #define __rsnd_mod_shift_quit           4
 #define __rsnd_mod_shift_start          8
@@ -328,11 +344,12 @@ struct rsnd_mod {
 #define __rsnd_mod_shift_hw_params      28 /* always called */
 #define __rsnd_mod_shift_pointer        28 /* always called */
 #define __rsnd_mod_shift_prepare        28 /* always called */
+#define __rsnd_mod_shift_cleanup        28 /* always called */
 
 #define __rsnd_mod_add_probe            0
 #define __rsnd_mod_add_remove           0
-#define __rsnd_mod_add_nolock_start      1
-#define __rsnd_mod_add_nolock_stop      -1
+#define __rsnd_mod_add_prepare          0
+#define __rsnd_mod_add_cleanup          0
 #define __rsnd_mod_add_init              1
 #define __rsnd_mod_add_quit             -1
 #define __rsnd_mod_add_start             1
@@ -342,10 +359,11 @@ struct rsnd_mod {
 #define __rsnd_mod_add_fallback         0
 #define __rsnd_mod_add_hw_params        0
 #define __rsnd_mod_add_pointer          0
-#define __rsnd_mod_add_prepare          0
 
 #define __rsnd_mod_call_probe           0
 #define __rsnd_mod_call_remove          0
+#define __rsnd_mod_call_prepare         0
+#define __rsnd_mod_call_cleanup         0
 #define __rsnd_mod_call_init            0
 #define __rsnd_mod_call_quit            1
 #define __rsnd_mod_call_start           0
@@ -355,9 +373,6 @@ struct rsnd_mod {
 #define __rsnd_mod_call_fallback        0
 #define __rsnd_mod_call_hw_params       0
 #define __rsnd_mod_call_pointer         0
-#define __rsnd_mod_call_nolock_start    0
-#define __rsnd_mod_call_nolock_stop     1
-#define __rsnd_mod_call_prepare         0
 
 #define rsnd_mod_to_priv(mod)   ((mod)->priv)
 #define rsnd_mod_name(mod)      ((mod)->ops->name)
@@ -438,6 +453,7 @@ struct rsnd_dai_stream {
         char name[RSND_DAI_NAME_SIZE];
         struct snd_pcm_substream *substream;
         struct rsnd_mod *mod[RSND_MOD_MAX];
+        struct rsnd_mod *dma;
         struct rsnd_dai *rdai;
         struct device *dmac_dev; /* for IPMMU */
         u32 parent_ssi_status;
@@ -467,6 +483,7 @@ struct rsnd_dai {
 
         int max_channels;       /* 2ch - 16ch */
         int ssi_lane;           /* 1lane - 4lane */
+        int chan_width;         /* 16/24/32 bit width */
 
         unsigned int clk_master:1;
         unsigned int bit_clk_inv:1;
@@ -500,6 +517,11 @@ int rsnd_rdai_channels_ctrl(struct rsnd_dai *rdai,
 int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai,
                             int ssi_lane);
 
+#define rsnd_rdai_width_set(rdai, width) \
+        rsnd_rdai_width_ctrl(rdai, width)
+#define rsnd_rdai_width_get(rdai) \
+        rsnd_rdai_width_ctrl(rdai, 0)
+int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width);
 void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io);
 int rsnd_dai_connect(struct rsnd_mod *mod,
                      struct rsnd_dai_stream *io,
@@ -692,6 +714,7 @@ void rsnd_ssi_remove(struct rsnd_priv *priv);
 struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id);
 int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod);
 int rsnd_ssi_use_busif(struct rsnd_dai_stream *io);
+int rsnd_ssi_get_busif(struct rsnd_dai_stream *io);
 u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io);
 
 #define RSND_SSI_HDMI_PORT0     0xf0
@@ -709,7 +732,7 @@ int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod);
 void rsnd_parse_connect_ssi(struct rsnd_dai *rdai,
                             struct device_node *playback,
                             struct device_node *capture);
-unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv,
+unsigned int rsnd_ssi_clk_query(struct rsnd_dai *rdai,
                        int param1, int param2, int *idx);
 
 /*
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index beccfbac7581..cd38a43b976f 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -158,7 +158,7 @@ static int rsnd_src_hw_params(struct rsnd_mod *mod,
                 struct snd_soc_dpcm *dpcm;
                 struct snd_pcm_hw_params *be_params;
 
-                list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+                for_each_dpcm_be(fe, stream, dpcm) {
                         be_params = &dpcm->hw_params;
 
                         if (params_rate(fe_params) != params_rate(be_params))
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 3f880ec66459..fcb4df23248c 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -42,7 +42,13 @@
 #define DWL_24          (5 << 19)       /* Data Word Length */
 #define DWL_32          (6 << 19)       /* Data Word Length */
 
+/*
+ * System word length
+ */
+#define SWL_16          (1 << 16)       /* R/W System Word Length */
+#define SWL_24          (2 << 16)       /* R/W System Word Length */
 #define SWL_32          (3 << 16)       /* R/W System Word Length */
+
 #define SCKD            (1 << 15)       /* Serial Bit Clock Direction */
 #define SWSD            (1 << 14)       /* Serial WS Direction */
 #define SCKP            (1 << 13)       /* Serial Bit Clock Polarity */
@@ -72,7 +78,6 @@
 
 struct rsnd_ssi {
         struct rsnd_mod mod;
-        struct rsnd_mod *dma;
 
         u32 flags;
         u32 cr_own;
@@ -145,6 +150,11 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io)
         return use_busif;
 }
 
+int rsnd_ssi_get_busif(struct rsnd_dai_stream *io)
+{
+        return 0; /* BUSIF0 only for now */
+}
+
 static void rsnd_ssi_status_clear(struct rsnd_mod *mod)
 {
         rsnd_mod_write(mod, SSISR, 0);
@@ -220,14 +230,32 @@ u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io)
         return 0;
 }
 
-unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv,
+static u32 rsnd_rdai_width_to_swl(struct rsnd_dai *rdai)
+{
+        struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
+        struct device *dev = rsnd_priv_to_dev(priv);
+        int width = rsnd_rdai_width_get(rdai);
+
+        switch (width) {
+        case 32: return SWL_32;
+        case 24: return SWL_24;
+        case 16: return SWL_16;
+        }
+
+        dev_err(dev, "unsupported slot width value: %d\n", width);
+        return 0;
+}
+
+unsigned int rsnd_ssi_clk_query(struct rsnd_dai *rdai,
                        int param1, int param2, int *idx)
 {
+        struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
         int ssi_clk_mul_table[] = {
                 1, 2, 4, 8, 16, 6, 12,
         };
         int j, ret;
         unsigned int main_rate;
+        int width = rsnd_rdai_width_get(rdai);
 
         for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) {
 
@@ -240,12 +268,7 @@ unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv,
                 if (j == 0)
                         continue;
 
-                /*
-                 * this driver is assuming that
-                 * system word is 32bit x chan
-                 * see rsnd_ssi_init()
-                 */
-                main_rate = 32 * param1 * param2 * ssi_clk_mul_table[j];
+                main_rate = width * param1 * param2 * ssi_clk_mul_table[j];
 
                 ret = rsnd_adg_clk_query(priv, main_rate);
                 if (ret < 0)
@@ -289,10 +312,15 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
                         return -EINVAL;
                 }
 
+                if (ssi->chan != chan) {
+                        dev_err(dev, "SSI parent/child should use same chan\n");
+                        return -EINVAL;
+                }
+
                 return 0;
         }
 
-        main_rate = rsnd_ssi_clk_query(priv, rate, chan, &idx);
+        main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx);
         if (!main_rate) {
                 dev_err(dev, "unsupported clock rate\n");
                 return -EIO;
@@ -312,9 +340,11 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
          * SSICR  : FORCE, SCKD, SWSD
          * SSIWSR : CONT
          */
-        ssi->cr_clk = FORCE | SWL_32 | SCKD | SWSD | CKDV(idx);
+        ssi->cr_clk = FORCE | rsnd_rdai_width_to_swl(rdai) |
+                        SCKD | SWSD | CKDV(idx);
         ssi->wsr = CONT;
         ssi->rate = rate;
+        ssi->chan = chan;
 
         dev_dbg(dev, "%s[%d] outputs %u Hz\n",
                 rsnd_mod_name(mod),
@@ -340,6 +370,7 @@ static void rsnd_ssi_master_clk_stop(struct rsnd_mod *mod,
 
         ssi->cr_clk     = 0;
         ssi->rate       = 0;
+        ssi->chan       = 0;
 
         rsnd_adg_ssi_clk_stop(mod);
 }
@@ -357,15 +388,11 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 
         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 | rsnd_rdai_width_to_swl(rdai);
 
         if (rdai->bit_clk_inv)
                 cr_own |= SCKP;
-        if (rdai->frm_clk_inv ^ is_tdm)
+        if (rdai->frm_clk_inv && !is_tdm)
                 cr_own |= SWSP;
         if (rdai->data_alignment)
                 cr_own |= SDTA;
@@ -373,6 +400,17 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
                 cr_own |= DEL;
 
         /*
+         * TDM Mode
+         * see
+         *      rsnd_ssiu_init_gen2()
+         */
+        wsr = ssi->wsr;
+        if (is_tdm) {
+                wsr     |= WS_MODE;
+                cr_own  |= CHNL_8;
+        }
+
+        /*
          * We shouldn't exchange SWSP after running.
          * This means, parent needs to care it.
          */
@@ -384,6 +422,9 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 
         cr_own &= ~DWL_MASK;
         switch (snd_pcm_format_width(runtime->format)) {
+        case 8:
+                cr_own |= DWL_8;
+                break;
         case 16:
                 cr_own |= DWL_16;
                 break;
@@ -399,16 +440,6 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
                 cr_mode = DIEN;         /* PIO : enable Data interrupt */
         }
 
-        /*
-         * TDM Extend Mode
-         * see
-         *      rsnd_ssiu_init_gen2()
-         */
-        wsr = ssi->wsr;
-        if (is_tdm) {
-                wsr     |= WS_MODE;
-                cr_own  |= CHNL_8;
-        }
 init_end:
         ssi->cr_own     = cr_own;
         ssi->cr_mode    = cr_mode;
@@ -488,26 +519,16 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod,
                               struct snd_pcm_substream *substream,
                               struct snd_pcm_hw_params *params)
 {
-        struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
-        int chan = params_channels(params);
+        struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
+        unsigned int fmt_width = snd_pcm_format_width(params_format(params));
 
-        /*
-         * snd_pcm_ops::hw_params will be called *before*
-         * snd_soc_dai_ops::trigger. Thus, ssi->usrcnt is 0
-         * in 1st call.
-         */
-        if (ssi->usrcnt) {
-                /*
-                 * Already working.
-                 * It will happen if SSI has parent/child connection.
-                 * it is error if child <-> parent SSI uses
-                 * different channels.
-                 */
-                if (ssi->chan != chan)
-                        return -EIO;
-        }
+        if (fmt_width > rdai->chan_width) {
+                struct rsnd_priv *priv = rsnd_io_to_priv(io);
+                struct device *dev = rsnd_priv_to_dev(priv);
 
-        ssi->chan = chan;
+                dev_err(dev, "invalid combination of slot-width and format-data-width\n");
+                return -EINVAL;
+        }
 
         return 0;
 }
@@ -873,7 +894,6 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
                               struct rsnd_dai_stream *io,
                               struct rsnd_priv *priv)
 {
-        struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
         int ret;
 
         /*
@@ -888,7 +908,7 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
                 return ret;
 
         /* SSI probe might be called many times in MUX multi path */
-        ret = rsnd_dma_attach(io, mod, &ssi->dma);
+        ret = rsnd_dma_attach(io, mod, &io->dma);
 
         return ret;
 }
diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index 016fbf5ac242..39b67643b5dc 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -10,9 +10,12 @@
 
 struct rsnd_ssiu {
         struct rsnd_mod mod;
+        u32 busif_status[8]; /* for BUSIF0 - BUSIF7 */
+        unsigned int usrcnt;
 };
 
 #define rsnd_ssiu_nr(priv) ((priv)->ssiu_nr)
+#define rsnd_mod_to_ssiu(_mod) container_of((_mod), struct rsnd_ssiu, mod)
 #define for_each_rsnd_ssiu(pos, priv, i)                                \
         for (i = 0;                                                     \
              (i < rsnd_ssiu_nr(priv)) &&                                \
@@ -120,6 +123,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
                                struct rsnd_dai_stream *io,
                                struct rsnd_priv *priv)
 {
+        struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod);
         int hdmi = rsnd_ssi_hdmi_port(io);
         int ret;
         u32 mode = 0;
@@ -128,6 +132,8 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
         if (ret < 0)
                 return ret;
 
+        ssiu->usrcnt++;
+
         if (rsnd_runtime_is_ssi_tdm(io)) {
                 /*
                  * TDM Extend Mode
@@ -140,15 +146,59 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
         rsnd_mod_write(mod, SSI_MODE, mode);
 
         if (rsnd_ssi_use_busif(io)) {
-                rsnd_mod_write(mod, SSI_BUSIF_ADINR,
-                               rsnd_get_adinr_bit(mod, io) |
-                               (rsnd_io_is_play(io) ?
-                                rsnd_runtime_channel_after_ctu(io) :
-                                rsnd_runtime_channel_original(io)));
-                rsnd_mod_write(mod, SSI_BUSIF_MODE,
-                               rsnd_get_busif_shift(io, mod) | 1);
-                rsnd_mod_write(mod, SSI_BUSIF_DALIGN,
-                               rsnd_get_dalign(mod, io));
+                int id = rsnd_mod_id(mod);
+                int busif = rsnd_ssi_get_busif(io);
+
+                /*
+                 * FIXME
+                 *
+                 * We can't support SSI9-4/5/6/7, because its address is
+                 * out of calculation rule
+                 */
+                if ((id == 9) && (busif >= 4)) {
+                        struct device *dev = rsnd_priv_to_dev(priv);
+
+                        dev_err(dev, "This driver doesn't support SSI%d-%d, so far",
+                                id, busif);
+                }
+
+#define RSND_WRITE_BUSIF(i)                                             \
+                rsnd_mod_write(mod, SSI_BUSIF##i##_ADINR,               \
+                               rsnd_get_adinr_bit(mod, io) |            \
+                               (rsnd_io_is_play(io) ?                   \
+                                rsnd_runtime_channel_after_ctu(io) :    \
+                                rsnd_runtime_channel_original(io)));    \
+                rsnd_mod_write(mod, SSI_BUSIF##i##_MODE,                \
+                               rsnd_get_busif_shift(io, mod) | 1);      \
+                rsnd_mod_write(mod, SSI_BUSIF##i##_DALIGN,              \
+                               rsnd_get_dalign(mod, io))
+
+                switch (busif) {
+                case 0:
+                        RSND_WRITE_BUSIF(0);
+                        break;
+                case 1:
+                        RSND_WRITE_BUSIF(1);
+                        break;
+                case 2:
+                        RSND_WRITE_BUSIF(2);
+                        break;
+                case 3:
+                        RSND_WRITE_BUSIF(3);
+                        break;
+                case 4:
+                        RSND_WRITE_BUSIF(4);
+                        break;
+                case 5:
+                        RSND_WRITE_BUSIF(5);
+                        break;
+                case 6:
+                        RSND_WRITE_BUSIF(6);
+                        break;
+                case 7:
+                        RSND_WRITE_BUSIF(7);
+                        break;
+                }
         }
 
         if (hdmi) {
@@ -194,10 +244,12 @@ static int rsnd_ssiu_start_gen2(struct rsnd_mod *mod,
                                 struct rsnd_dai_stream *io,
                                 struct rsnd_priv *priv)
 {
+        int busif = rsnd_ssi_get_busif(io);
+
         if (!rsnd_ssi_use_busif(io))
                 return 0;
 
-        rsnd_mod_write(mod, SSI_CTRL, 0x1);
+        rsnd_mod_bset(mod, SSI_CTRL, 1 << (busif * 4), 1 << (busif * 4));
 
         if (rsnd_ssi_multi_slaves_runtime(io))
                 rsnd_mod_write(mod, SSI_CONTROL, 0x1);
@@ -209,10 +261,16 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod,
                                struct rsnd_dai_stream *io,
                                struct rsnd_priv *priv)
 {
+        struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod);
+        int busif = rsnd_ssi_get_busif(io);
+
         if (!rsnd_ssi_use_busif(io))
                 return 0;
 
-        rsnd_mod_write(mod, SSI_CTRL, 0);
+        rsnd_mod_bset(mod, SSI_CTRL, 1 << (busif * 4), 0);
+
+        if (--ssiu->usrcnt)
+                return 0;
 
         if (rsnd_ssi_multi_slaves_runtime(io))
                 rsnd_mod_write(mod, SSI_CONTROL, 0);
@@ -246,6 +304,16 @@ int rsnd_ssiu_attach(struct rsnd_dai_stream *io,
         return rsnd_dai_connect(mod, io, mod->type);
 }
 
+static u32 *rsnd_ssiu_get_status(struct rsnd_dai_stream *io,
+                                 struct rsnd_mod *mod,
+                                 enum rsnd_mod_type type)
+{
+        struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod);
+        int busif = rsnd_ssi_get_busif(io);
+
+        return &ssiu->busif_status[busif];
+}
+
 int rsnd_ssiu_probe(struct rsnd_priv *priv)
 {
         struct device *dev = rsnd_priv_to_dev(priv);
@@ -269,7 +337,7 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv)
 
         for_each_rsnd_ssiu(ssiu, priv, i) {
                 ret = rsnd_mod_init(priv, rsnd_mod_get(ssiu),
-                                    ops, NULL, rsnd_mod_get_status,
+                                    ops, NULL, rsnd_ssiu_get_status,
                                     RSND_MOD_SSIU, i);
                 if (ret)
                         return ret;
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 409d082e80d1..699397a09167 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -157,7 +157,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
         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)
+                for_each_dpcm_be(fe, stream, dpcm)
                         dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
 
                 dpcm_be_disconnect(fe, stream);
@@ -321,7 +321,7 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream)
         ret = dpcm_be_dai_shutdown(fe, stream);
 
         /* mark FE's links ready to prune */
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+        for_each_dpcm_be(fe, stream, dpcm)
                 dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
 
         dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 473eefe8658e..6ddcf12bc030 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -52,6 +52,10 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root);
 
 static DEFINE_MUTEX(client_mutex);
 static LIST_HEAD(component_list);
+static LIST_HEAD(unbind_card_list);
+
+#define for_each_component(component)                   \
+        list_for_each_entry(component, &component_list, list)
 
 /*
  * This is a timeout to do a DAPM powerdown after a stream is closed().
@@ -62,8 +66,9 @@ static int pmdown_time = 5000;
 module_param(pmdown_time, int, 0);
 MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
 
-/* If a DMI filed contain strings in this blacklist (e.g.
- * "Type2 - Board Manufacturer" or  "Type1 - TBD by OEM"), it will be taken
+/*
+ * If a DMI filed contain strings in this blacklist (e.g.
+ * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken
  * as invalid and dropped when setting the card long name from DMI info.
  */
 static const char * const dmi_blacklist[] = {
@@ -175,8 +180,8 @@ static int dai_list_show(struct seq_file *m, void *v)
 
         mutex_lock(&client_mutex);
 
-        list_for_each_entry(component, &component_list, list)
-                list_for_each_entry(dai, &component->dai_list, list)
+        for_each_component(component)
+                for_each_component_dais(component, dai)
                         seq_printf(m, "%s\n", dai->name);
 
         mutex_unlock(&client_mutex);
@@ -191,7 +196,7 @@ static int component_list_show(struct seq_file *m, void *v)
 
         mutex_lock(&client_mutex);
 
-        list_for_each_entry(component, &component_list, list)
+        for_each_component(component)
                 seq_printf(m, "%s\n", component->name);
 
         mutex_unlock(&client_mutex);
@@ -218,7 +223,7 @@ static void soc_init_card_debugfs(struct snd_soc_card *card)
                                                     &card->pop_time);
         if (!card->debugfs_pop_time)
                 dev_warn(card->dev,
-                       "ASoC: Failed to create pop time debugfs file\n");
+                         "ASoC: Failed to create pop time debugfs file\n");
 }
 
 static void soc_cleanup_card_debugfs(struct snd_soc_card *card)
@@ -341,7 +346,7 @@ struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
 {
         struct snd_soc_pcm_runtime *rtd;
 
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
                 if (rtd->dai_link->no_pcm &&
                         !strcmp(rtd->dai_link->name, dai_link))
                         return rtd->pcm->streams[stream].substream;
@@ -398,7 +403,7 @@ static void soc_remove_pcm_runtimes(struct snd_soc_card *card)
 {
         struct snd_soc_pcm_runtime *rtd, *_rtd;
 
-        list_for_each_entry_safe(rtd, _rtd, &card->rtd_list, list) {
+        for_each_card_rtds_safe(card, rtd, _rtd) {
                 list_del(&rtd->list);
                 soc_free_pcm_runtime(rtd);
         }
@@ -411,7 +416,7 @@ struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card,
 {
         struct snd_soc_pcm_runtime *rtd;
 
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
                 if (!strcmp(rtd->dai_link->name, dai_link))
                         return rtd;
         }
@@ -422,7 +427,8 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime);
 
 static void codec2codec_close_delayed_work(struct work_struct *work)
 {
-        /* Currently nothing to do for c2c links
+        /*
+         * Currently nothing to do for c2c links
          * Since c2c links are internal nodes in the DAPM graph and
          * don't interface with the outside world or application layer
          * we don't have to do any special handling on close.
@@ -442,8 +448,9 @@ int snd_soc_suspend(struct device *dev)
         if (!card->instantiated)
                 return 0;
 
-        /* Due to the resume being scheduled into a workqueue we could
-        * suspend before that's finished - wait for it to complete.
+        /*
+         * Due to the resume being scheduled into a workqueue we could
+         * suspend before that's finished - wait for it to complete.
          */
         snd_power_wait(card->snd_card, SNDRV_CTL_POWER_D0);
 
@@ -451,13 +458,13 @@ int snd_soc_suspend(struct device *dev)
         snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot);
 
         /* mute any active DACs */
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
+                struct snd_soc_dai *dai;
 
                 if (rtd->dai_link->ignore_suspend)
                         continue;
 
-                for (i = 0; i < rtd->num_codecs; i++) {
-                        struct snd_soc_dai *dai = rtd->codec_dais[i];
+                for_each_rtd_codec_dai(rtd, i, dai) {
                         struct snd_soc_dai_driver *drv = dai->driver;
 
                         if (drv->ops->digital_mute && dai->playback_active)
@@ -466,7 +473,7 @@ int snd_soc_suspend(struct device *dev)
         }
 
         /* suspend all pcms */
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
                 if (rtd->dai_link->ignore_suspend)
                         continue;
 
@@ -476,7 +483,7 @@ int snd_soc_suspend(struct device *dev)
         if (card->suspend_pre)
                 card->suspend_pre(card);
 
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
                 struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 
                 if (rtd->dai_link->ignore_suspend)
@@ -487,10 +494,10 @@ int snd_soc_suspend(struct device *dev)
         }
 
         /* close any waiting streams */
-        list_for_each_entry(rtd, &card->rtd_list, list)
+        for_each_card_rtds(card, rtd)
                 flush_delayed_work(&rtd->delayed_work);
 
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
 
                 if (rtd->dai_link->ignore_suspend)
                         continue;
@@ -509,11 +516,14 @@ int snd_soc_suspend(struct device *dev)
         snd_soc_dapm_sync(&card->dapm);
 
         /* suspend all COMPONENTs */
-        list_for_each_entry(component, &card->component_dev_list, card_list) {
-                struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+        for_each_card_components(card, component) {
+                struct snd_soc_dapm_context *dapm =
+                                snd_soc_component_get_dapm(component);
 
-                /* If there are paths active then the COMPONENT will be held with
-                 * bias _ON and should not be suspended. */
+                /*
+                 * If there are paths active then the COMPONENT will be held
+                 * with bias _ON and should not be suspended.
+                 */
                 if (!component->suspended) {
                         switch (snd_soc_dapm_get_bias_level(dapm)) {
                         case SND_SOC_BIAS_STANDBY:
@@ -547,7 +557,7 @@ int snd_soc_suspend(struct device *dev)
                 }
         }
 
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
                 struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 
                 if (rtd->dai_link->ignore_suspend)
@@ -567,18 +577,21 @@ int snd_soc_suspend(struct device *dev)
 }
 EXPORT_SYMBOL_GPL(snd_soc_suspend);
 
-/* deferred resume work, so resume can complete before we finished
+/*
+ * deferred resume work, so resume can complete before we finished
  * setting our codec back up, which can be very slow on I2C
  */
 static void soc_resume_deferred(struct work_struct *work)
 {
         struct snd_soc_card *card =
-                        container_of(work, struct snd_soc_card, deferred_resume_work);
+                        container_of(work, struct snd_soc_card,
+                                     deferred_resume_work);
         struct snd_soc_pcm_runtime *rtd;
         struct snd_soc_component *component;
         int i;
 
-        /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
+        /*
+         * our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
          * so userspace apps are blocked from touching us
          */
 
@@ -591,7 +604,7 @@ static void soc_resume_deferred(struct work_struct *work)
                 card->resume_pre(card);
 
         /* resume control bus DAIs */
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
                 struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 
                 if (rtd->dai_link->ignore_suspend)
@@ -601,7 +614,7 @@ static void soc_resume_deferred(struct work_struct *work)
                         cpu_dai->driver->resume(cpu_dai);
         }
 
-        list_for_each_entry(component, &card->component_dev_list, card_list) {
+        for_each_card_components(card, component) {
                 if (component->suspended) {
                         if (component->driver->resume)
                                 component->driver->resume(component);
@@ -609,7 +622,7 @@ static void soc_resume_deferred(struct work_struct *work)
                 }
         }
 
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
 
                 if (rtd->dai_link->ignore_suspend)
                         continue;
@@ -624,13 +637,13 @@ static void soc_resume_deferred(struct work_struct *work)
         }
 
         /* unmute any active DACs */
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
+                struct snd_soc_dai *dai;
 
                 if (rtd->dai_link->ignore_suspend)
                         continue;
 
-                for (i = 0; i < rtd->num_codecs; i++) {
-                        struct snd_soc_dai *dai = rtd->codec_dais[i];
+                for_each_rtd_codec_dai(rtd, i, dai) {
                         struct snd_soc_dai_driver *drv = dai->driver;
 
                         if (drv->ops->digital_mute && dai->playback_active)
@@ -638,7 +651,7 @@ static void soc_resume_deferred(struct work_struct *work)
                 }
         }
 
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
                 struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 
                 if (rtd->dai_link->ignore_suspend)
@@ -673,16 +686,15 @@ int snd_soc_resume(struct device *dev)
                 return 0;
 
         /* activate pins from sleep state */
-        list_for_each_entry(rtd, &card->rtd_list, list) {
-                struct snd_soc_dai **codec_dais = rtd->codec_dais;
+        for_each_card_rtds(card, rtd) {
+                struct snd_soc_dai *codec_dai;
                 struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
                 int j;
 
                 if (cpu_dai->active)
                         pinctrl_pm_select_default_state(cpu_dai->dev);
 
-                for (j = 0; j < rtd->num_codecs; j++) {
-                        struct snd_soc_dai *codec_dai = codec_dais[j];
+                for_each_rtd_codec_dai(rtd, j, codec_dai) {
                         if (codec_dai->active)
                                 pinctrl_pm_select_default_state(codec_dai->dev);
                 }
@@ -694,8 +706,9 @@ int snd_soc_resume(struct device *dev)
          * have that problem and may take a substantial amount of time to resume
          * due to I/O costs and anti-pop so handle them out of line.
          */
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
                 struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
                 bus_control |= cpu_dai->driver->bus_control;
         }
         if (bus_control) {
@@ -725,7 +738,7 @@ static struct snd_soc_component *soc_find_component(
 
         lockdep_assert_held(&client_mutex);
 
-        list_for_each_entry(component, &component_list, list) {
+        for_each_component(component) {
                 if (of_node) {
                         if (component->dev->of_node == of_node)
                                 return component;
@@ -737,6 +750,24 @@ static struct snd_soc_component *soc_find_component(
         return NULL;
 }
 
+static int snd_soc_is_matching_component(
+        const struct snd_soc_dai_link_component *dlc,
+        struct snd_soc_component *component)
+{
+        struct device_node *component_of_node;
+
+        component_of_node = component->dev->of_node;
+        if (!component_of_node && component->dev->parent)
+                component_of_node = component->dev->parent->of_node;
+
+        if (dlc->of_node && component_of_node != dlc->of_node)
+                return 0;
+        if (dlc->name && strcmp(component->name, dlc->name))
+                return 0;
+
+        return 1;
+}
+
 /**
  * snd_soc_find_dai - Find a registered DAI
  *
@@ -753,21 +784,14 @@ struct snd_soc_dai *snd_soc_find_dai(
 {
         struct snd_soc_component *component;
         struct snd_soc_dai *dai;
-        struct device_node *component_of_node;
 
         lockdep_assert_held(&client_mutex);
 
-        /* Find CPU DAI from registered DAIs*/
-        list_for_each_entry(component, &component_list, list) {
-                component_of_node = component->dev->of_node;
-                if (!component_of_node && component->dev->parent)
-                        component_of_node = component->dev->parent->of_node;
-
-                if (dlc->of_node && component_of_node != dlc->of_node)
-                        continue;
-                if (dlc->name && strcmp(component->name, dlc->name))
+        /* Find CPU DAI from registered DAIs */
+        for_each_component(component) {
+                if (!snd_soc_is_matching_component(dlc, component))
                         continue;
-                list_for_each_entry(dai, &component->dai_list, list) {
+                for_each_component_dais(component, dai) {
                         if (dlc->dai_name && strcmp(dai->name, dlc->dai_name)
                             && (!dai->driver->name
                                 || strcmp(dai->driver->name, dlc->dai_name)))
@@ -781,7 +805,6 @@ struct snd_soc_dai *snd_soc_find_dai(
 }
 EXPORT_SYMBOL_GPL(snd_soc_find_dai);
 
-
 /**
  * snd_soc_find_dai_link - Find a DAI link
  *
@@ -805,7 +828,7 @@ struct snd_soc_dai_link *snd_soc_find_dai_link(struct snd_soc_card *card,
 
         lockdep_assert_held(&client_mutex);
 
-        list_for_each_entry_safe(link, _link, &card->dai_link_list, list) {
+        for_each_card_links_safe(card, link, _link) {
                 if (link->id != id)
                         continue;
 
@@ -828,7 +851,7 @@ static bool soc_is_dai_link_bound(struct snd_soc_card *card,
 {
         struct snd_soc_pcm_runtime *rtd;
 
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
                 if (rtd->dai_link == dai_link)
                         return true;
         }
@@ -844,8 +867,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
         struct snd_soc_dai_link_component cpu_dai_component;
         struct snd_soc_component *component;
         struct snd_soc_dai **codec_dais;
-        struct device_node *platform_of_node;
-        const char *platform_name;
         int i;
 
         if (dai_link->ignore)
@@ -877,6 +898,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
         rtd->num_codecs = dai_link->num_codecs;
 
         /* Find CODEC from registered CODECs */
+        /* we can use for_each_rtd_codec_dai() after this */
         codec_dais = rtd->codec_dais;
         for (i = 0; i < rtd->num_codecs; i++) {
                 codec_dais[i] = snd_soc_find_dai(&codecs[i]);
@@ -891,24 +913,11 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
         /* Single codec links expect codec and codec_dai in runtime data */
         rtd->codec_dai = codec_dais[0];
 
-        /* if there's no platform we match on the empty platform */
-        platform_name = dai_link->platform_name;
-        if (!platform_name && !dai_link->platform_of_node)
-                platform_name = "snd-soc-dummy";
-
         /* find one from the set of registered platforms */
-        list_for_each_entry(component, &component_list, list) {
-                platform_of_node = component->dev->of_node;
-                if (!platform_of_node && component->dev->parent->of_node)
-                        platform_of_node = component->dev->parent->of_node;
-
-                if (dai_link->platform_of_node) {
-                        if (platform_of_node != dai_link->platform_of_node)
-                                continue;
-                } else {
-                        if (strcmp(component->name, platform_name))
-                                continue;
-                }
+        for_each_component(component) {
+                if (!snd_soc_is_matching_component(dai_link->platform,
+                                                   component))
+                        continue;
 
                 snd_soc_rtdcom_add(rtd, component);
         }
@@ -918,7 +927,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 
 _err_defer:
         soc_free_pcm_runtime(rtd);
-        return  -EPROBE_DEFER;
+        return -EPROBE_DEFER;
 }
 
 static void soc_remove_component(struct snd_soc_component *component)
@@ -942,23 +951,25 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order)
 {
         int err;
 
-        if (dai && dai->probed &&
-                        dai->driver->remove_order == order) {
-                if (dai->driver->remove) {
-                        err = dai->driver->remove(dai);
-                        if (err < 0)
-                                dev_err(dai->dev,
-                                        "ASoC: failed to remove %s: %d\n",
-                                        dai->name, err);
-                }
-                dai->probed = 0;
+        if (!dai || !dai->probed ||
+            dai->driver->remove_order != order)
+                return;
+
+        if (dai->driver->remove) {
+                err = dai->driver->remove(dai);
+                if (err < 0)
+                        dev_err(dai->dev,
+                                "ASoC: failed to remove %s: %d\n",
+                                dai->name, err);
         }
+        dai->probed = 0;
 }
 
 static void soc_remove_link_dais(struct snd_soc_card *card,
                 struct snd_soc_pcm_runtime *rtd, int order)
 {
         int i;
+        struct snd_soc_dai *codec_dai;
 
         /* unregister the rtd device */
         if (rtd->dev_registered) {
@@ -967,8 +978,8 @@ static void soc_remove_link_dais(struct snd_soc_card *card,
         }
 
         /* remove the CODEC DAI */
-        for (i = 0; i < rtd->num_codecs; i++)
-                soc_remove_dai(rtd->codec_dais[i], order);
+        for_each_rtd_codec_dai(rtd, i, codec_dai)
+                soc_remove_dai(codec_dai, order);
 
         soc_remove_dai(rtd->cpu_dai, order);
 }
@@ -993,28 +1004,57 @@ static void soc_remove_dai_links(struct snd_soc_card *card)
         struct snd_soc_pcm_runtime *rtd;
         struct snd_soc_dai_link *link, *_link;
 
-        for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-                        order++) {
-                list_for_each_entry(rtd, &card->rtd_list, list)
+        for_each_comp_order(order) {
+                for_each_card_rtds(card, rtd)
                         soc_remove_link_dais(card, rtd, order);
         }
 
-        for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-                        order++) {
-                list_for_each_entry(rtd, &card->rtd_list, list)
+        for_each_comp_order(order) {
+                for_each_card_rtds(card, rtd)
                         soc_remove_link_components(card, rtd, order);
         }
 
-        list_for_each_entry_safe(link, _link, &card->dai_link_list, list) {
+        for_each_card_links_safe(card, link, _link) {
                 if (link->dobj.type == SND_SOC_DOBJ_DAI_LINK)
                         dev_warn(card->dev, "Topology forgot to remove link %s?\n",
                                 link->name);
 
                 list_del(&link->list);
-                card->num_dai_links--;
         }
 }
 
+static int snd_soc_init_platform(struct snd_soc_card *card,
+                                 struct snd_soc_dai_link *dai_link)
+{
+        struct snd_soc_dai_link_component *platform = dai_link->platform;
+
+        /*
+         * FIXME
+         *
+         * this function should be removed in the future
+         */
+        /* convert Legacy platform link */
+        if (!platform) {
+                platform = devm_kzalloc(card->dev,
+                                sizeof(struct snd_soc_dai_link_component),
+                                GFP_KERNEL);
+                if (!platform)
+                        return -ENOMEM;
+
+                dai_link->platform      = platform;
+                platform->name          = dai_link->platform_name;
+                platform->of_node       = dai_link->platform_of_node;
+                platform->dai_name      = NULL;
+        }
+
+        /* if there's no platform we match on the empty platform */
+        if (!platform->name &&
+            !platform->of_node)
+                platform->name = "snd-soc-dummy";
+
+        return 0;
+}
+
 static int snd_soc_init_multicodec(struct snd_soc_card *card,
                                    struct snd_soc_dai_link *dai_link)
 {
@@ -1043,9 +1083,16 @@ static int snd_soc_init_multicodec(struct snd_soc_card *card,
 }
 
 static int soc_init_dai_link(struct snd_soc_card *card,
-                                   struct snd_soc_dai_link *link)
+                             struct snd_soc_dai_link *link)
 {
         int i, ret;
+        struct snd_soc_dai_link_component *codec;
+
+        ret = snd_soc_init_platform(card, link);
+        if (ret) {
+                dev_err(card->dev, "ASoC: failed to init multiplatform\n");
+                return ret;
+        }
 
         ret = snd_soc_init_multicodec(card, link);
         if (ret) {
@@ -1053,19 +1100,19 @@ static int soc_init_dai_link(struct snd_soc_card *card,
                 return ret;
         }
 
-        for (i = 0; i < link->num_codecs; i++) {
+        for_each_link_codecs(link, i, codec) {
                 /*
                  * Codec must be specified by 1 of name or OF node,
                  * not both or neither.
                  */
-                if (!!link->codecs[i].name ==
-                    !!link->codecs[i].of_node) {
+                if (!!codec->name ==
+                    !!codec->of_node) {
                         dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n",
                                 link->name);
                         return -EINVAL;
                 }
                 /* Codec DAI name must be specified */
-                if (!link->codecs[i].dai_name) {
+                if (!codec->dai_name) {
                         dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n",
                                 link->name);
                         return -EINVAL;
@@ -1076,13 +1123,12 @@ static int soc_init_dai_link(struct snd_soc_card *card,
          * Platform may be specified by either name or OF node, but
          * can be left unspecified, and a dummy platform will be used.
          */
-        if (link->platform_name && link->platform_of_node) {
+        if (link->platform->name && link->platform->of_node) {
                 dev_err(card->dev,
                         "ASoC: Both platform name/of_node are set for %s\n",
                         link->name);
                 return -EINVAL;
         }
-
         /*
          * CPU device may be specified by either name or OF node, but
          * can be left unspecified, and will be matched based on DAI
@@ -1111,7 +1157,8 @@ static int soc_init_dai_link(struct snd_soc_card *card,
 
 void snd_soc_disconnect_sync(struct device *dev)
 {
-        struct snd_soc_component *component = snd_soc_lookup_component(dev, NULL);
+        struct snd_soc_component *component =
+                        snd_soc_lookup_component(dev, NULL);
 
         if (!component || !component->card)
                 return;
@@ -1142,14 +1189,14 @@ int snd_soc_add_dai_link(struct snd_soc_card *card,
         }
 
         lockdep_assert_held(&client_mutex);
-        /* Notify the machine driver for extra initialization
+        /*
+         * Notify the machine driver for extra initialization
          * on the link created by topology.
          */
         if (dai_link->dobj.type && card->add_dai_link)
                 card->add_dai_link(card, dai_link);
 
         list_add_tail(&dai_link->list, &card->dai_link_list);
-        card->num_dai_links++;
 
         return 0;
 }
@@ -1178,16 +1225,16 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card,
         }
 
         lockdep_assert_held(&client_mutex);
-        /* Notify the machine driver for extra destruction
+        /*
+         * Notify the machine driver for extra destruction
          * on the link created by topology.
          */
         if (dai_link->dobj.type && card->remove_dai_link)
                 card->remove_dai_link(card, dai_link);
 
-        list_for_each_entry_safe(link, _link, &card->dai_link_list, list) {
+        for_each_card_links_safe(card, link, _link) {
                 if (link == dai_link) {
                         list_del(&link->list);
-                        card->num_dai_links--;
                         return;
                 }
         }
@@ -1239,7 +1286,8 @@ static void soc_set_name_prefix(struct snd_soc_card *card,
 static int soc_probe_component(struct snd_soc_card *card,
         struct snd_soc_component *component)
 {
-        struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+        struct snd_soc_dapm_context *dapm =
+                        snd_soc_component_get_dapm(component);
         struct snd_soc_dai *dai;
         int ret;
 
@@ -1277,7 +1325,7 @@ static int soc_probe_component(struct snd_soc_card *card,
                 }
         }
 
-        list_for_each_entry(dai, &component->dai_list, list) {
+        for_each_component_dais(component, dai) {
                 ret = snd_soc_dapm_new_dai_widgets(dapm, dai);
                 if (ret != 0) {
                         dev_err(component->dev,
@@ -1320,6 +1368,7 @@ static int soc_probe_component(struct snd_soc_card *card,
                                         component->driver->num_dapm_routes);
 
         list_add(&dapm->list, &card->dapm_list);
+        /* see for_each_card_components */
         list_add(&component->card_list, &card->component_dev_list);
 
         return 0;
@@ -1370,8 +1419,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
 }
 
 static int soc_probe_link_components(struct snd_soc_card *card,
-                        struct snd_soc_pcm_runtime *rtd,
-                                     int order)
+                                     struct snd_soc_pcm_runtime *rtd, int order)
 {
         struct snd_soc_component *component;
         struct snd_soc_rtdcom_list *rtdcom;
@@ -1398,6 +1446,7 @@ static int soc_probe_dai(struct snd_soc_dai *dai, int order)
 
         if (dai->driver->probe) {
                 int ret = dai->driver->probe(dai);
+
                 if (ret < 0) {
                         dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n",
                                 dai->name, ret);
@@ -1431,48 +1480,6 @@ static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais,
         return 0;
 }
 
-static int soc_link_dai_widgets(struct snd_soc_card *card,
-                                struct snd_soc_dai_link *dai_link,
-                                struct snd_soc_pcm_runtime *rtd)
-{
-        struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-        struct snd_soc_dai *codec_dai = rtd->codec_dai;
-        struct snd_soc_dapm_widget *sink, *source;
-        int ret;
-
-        if (rtd->num_codecs > 1)
-                dev_warn(card->dev, "ASoC: Multiple codecs not supported yet\n");
-
-        /* link the DAI widgets */
-        sink = codec_dai->playback_widget;
-        source = cpu_dai->capture_widget;
-        if (sink && source) {
-                ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params,
-                                           dai_link->num_params,
-                                           source, sink);
-                if (ret != 0) {
-                        dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n",
-                                sink->name, source->name, ret);
-                        return ret;
-                }
-        }
-
-        sink = cpu_dai->playback_widget;
-        source = codec_dai->capture_widget;
-        if (sink && source) {
-                ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params,
-                                           dai_link->num_params,
-                                           source, sink);
-                if (ret != 0) {
-                        dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n",
-                                sink->name, source->name, ret);
-                        return ret;
-                }
-        }
-
-        return 0;
-}
-
 static int soc_probe_link_dais(struct snd_soc_card *card,
                 struct snd_soc_pcm_runtime *rtd, int order)
 {
@@ -1480,6 +1487,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
         struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
         struct snd_soc_rtdcom_list *rtdcom;
         struct snd_soc_component *component;
+        struct snd_soc_dai *codec_dai;
         int i, ret, num;
 
         dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n",
@@ -1493,8 +1501,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
                 return ret;
 
         /* probe the CODEC DAI */
-        for (i = 0; i < rtd->num_codecs; i++) {
-                ret = soc_probe_dai(rtd->codec_dais[i], order);
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
+                ret = soc_probe_dai(codec_dai, order);
                 if (ret)
                         return ret;
         }
@@ -1546,7 +1554,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
         }
 
         if (cpu_dai->driver->compress_new) {
-                /*create compress_device"*/
+                /* create compress_device" */
                 ret = cpu_dai->driver->compress_new(rtd, num);
                 if (ret < 0) {
                         dev_err(card->dev, "ASoC: can't create compress %s\n",
@@ -1560,7 +1568,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
                         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);
+                                        dai_link->stream_name, ret);
                                 return ret;
                         }
                         ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd);
@@ -1573,11 +1581,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
                 } else {
                         INIT_DELAYED_WORK(&rtd->delayed_work,
                                                 codec2codec_close_delayed_work);
-
-                        /* link the DAI widgets */
-                        ret = soc_link_dai_widgets(card, dai_link, rtd);
-                        if (ret)
-                                return ret;
                 }
         }
 
@@ -1628,8 +1631,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card)
         int order;
         int ret;
 
-        for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-                order++) {
+        for_each_comp_order(order) {
                 list_for_each_entry(comp, &card->aux_comp_list, card_aux_list) {
                         if (comp->driver->probe_order == order) {
                                 ret = soc_probe_component(card, comp);
@@ -1651,8 +1653,7 @@ static void soc_remove_aux_devices(struct snd_soc_card *card)
         struct snd_soc_component *comp, *_comp;
         int order;
 
-        for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-                order++) {
+        for_each_comp_order(order) {
                 list_for_each_entry_safe(comp, _comp,
                         &card->aux_comp_list, card_aux_list) {
 
@@ -1681,14 +1682,12 @@ static void soc_remove_aux_devices(struct snd_soc_card *card)
 int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
         unsigned int dai_fmt)
 {
-        struct snd_soc_dai **codec_dais = rtd->codec_dais;
         struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+        struct snd_soc_dai *codec_dai;
         unsigned int i;
         int ret;
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                struct snd_soc_dai *codec_dai = codec_dais[i];
-
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt);
                 if (ret != 0 && ret != -ENOTSUPP) {
                         dev_warn(codec_dai->dev,
@@ -1697,8 +1696,10 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
                 }
         }
 
-        /* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */
-        /* the component which has non_legacy_dai_naming is Codec */
+        /*
+         * Flip the polarity for the "CPU" end of a CODEC<->CODEC link
+         * the component which has non_legacy_dai_naming is Codec
+         */
         if (cpu_dai->component->driver->non_legacy_dai_naming) {
                 unsigned int inv_dai_fmt;
 
@@ -1732,9 +1733,9 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
 }
 EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt);
 
-
 #ifdef CONFIG_DMI
-/* Trim special characters, and replace '-' with '_' since '-' is used to
+/*
+ * Trim special characters, and replace '-' with '_' since '-' is used to
  * separate different DMI fields in the card long name. Only number and
  * alphabet characters and a few separator characters are kept.
  */
@@ -1753,7 +1754,8 @@ static void cleanup_dmi_name(char *name)
         name[j] = '\0';
 }
 
-/* Check if a DMI field is valid, i.e. not containing any string
+/*
+ * Check if a DMI field is valid, i.e. not containing any string
  * in the black list.
  */
 static int is_dmi_valid(const char *field)
@@ -1816,7 +1818,6 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
                 return 0;
         }
 
-
         snprintf(card->dmi_longname, sizeof(card->snd_card->longname),
                          "%s", vendor);
         cleanup_dmi_name(card->dmi_longname);
@@ -1832,7 +1833,8 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
                 if (len < longname_buf_size)
                         cleanup_dmi_name(card->dmi_longname + len);
 
-                /* some vendors like Lenovo may only put a self-explanatory
+                /*
+                 * some vendors like Lenovo may only put a self-explanatory
                  * name in the product version field
                  */
                 product_version = dmi_get_system_info(DMI_PRODUCT_VERSION);
@@ -1891,7 +1893,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
         struct snd_soc_dai_link *dai_link;
         int i;
 
-        list_for_each_entry(component, &component_list, list) {
+        for_each_component(component) {
 
                 /* does this component override FEs ? */
                 if (!component->driver->ignore_machine)
@@ -1903,9 +1905,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
                         continue;
 
                 /* machine matches, so override the rtd data */
-                for (i = 0; i < card->num_links; i++) {
-
-                        dai_link = &card->dai_link[i];
+                for_each_card_prelinks(card, i, dai_link) {
 
                         /* ignore this FE */
                         if (dai_link->dynamic) {
@@ -1917,7 +1917,11 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
                                  card->dai_link[i].name);
 
                         /* override platform component */
-                        dai_link->platform_name = component->name;
+                        if (snd_soc_init_platform(card, dai_link) < 0) {
+                                dev_err(card->dev, "init platform error");
+                                continue;
+                        }
+                        dai_link->platform->name = component->name;
 
                         /* convert non BE into BE */
                         dai_link->no_pcm = 1;
@@ -1926,7 +1930,8 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
                         dai_link->be_hw_params_fixup =
                                 component->driver->be_hw_params_fixup;
 
-                        /* most BE links don't set stream name, so set it to
+                        /*
+                         * 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)
@@ -1936,7 +1941,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
                 /* Inform userspace we are using alternate topology */
                 if (component->driver->topology_name_prefix) {
 
-                        /* topology shortname created ? */
+                        /* topology shortname created? */
                         if (!card->topology_shortname_created) {
                                 comp_drv = component->driver;
 
@@ -1965,8 +1970,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
         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]);
+        for_each_card_prelinks(card, i, dai_link) {
+                ret = soc_bind_dai_link(card, dai_link);
                 if (ret != 0)
                         goto base_error;
         }
@@ -1979,8 +1984,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
         }
 
         /* add predefined DAI links to the list */
-        for (i = 0; i < card->num_links; i++)
-                snd_soc_add_dai_link(card, card->dai_link+i);
+        for_each_card_prelinks(card, i, dai_link)
+                snd_soc_add_dai_link(card, dai_link);
 
         /* card bind complete so register a sound card */
         ret = snd_card_new(card->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
@@ -2024,9 +2029,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
         }
 
         /* probe all components used by DAI links on this card */
-        for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-                        order++) {
-                list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_comp_order(order) {
+                for_each_card_rtds(card, rtd) {
                         ret = soc_probe_link_components(card, rtd, order);
                         if (ret < 0) {
                                 dev_err(card->dev,
@@ -2042,10 +2046,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
         if (ret < 0)
                 goto probe_dai_err;
 
-        /* Find new DAI links added during probing components and bind them.
+        /*
+         * Find new DAI links added during probing components and bind them.
          * Components with topology may bring new DAIs and DAI links.
          */
-        list_for_each_entry(dai_link, &card->dai_link_list, list) {
+        for_each_card_links(card, dai_link) {
                 if (soc_is_dai_link_bound(card, dai_link))
                         continue;
 
@@ -2058,9 +2063,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
         }
 
         /* probe all DAI links on this card */
-        for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-                        order++) {
-                list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_comp_order(order) {
+                for_each_card_rtds(card, rtd) {
                         ret = soc_probe_link_dais(card, rtd, order);
                         if (ret < 0) {
                                 dev_err(card->dev,
@@ -2075,7 +2079,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
         snd_soc_dapm_connect_dai_link_widgets(card);
 
         if (card->controls)
-                snd_soc_add_card_controls(card, card->controls, card->num_controls);
+                snd_soc_add_card_controls(card, card->controls,
+                                          card->num_controls);
 
         if (card->dapm_routes)
                 snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
@@ -2181,7 +2186,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card)
         struct snd_soc_pcm_runtime *rtd;
 
         /* make sure any delayed work runs */
-        list_for_each_entry(rtd, &card->rtd_list, list)
+        for_each_card_rtds(card, rtd)
                 flush_delayed_work(&rtd->delayed_work);
 
         /* free the ALSA card at first; this syncs with pending operations */
@@ -2221,21 +2226,23 @@ int snd_soc_poweroff(struct device *dev)
         if (!card->instantiated)
                 return 0;
 
-        /* Flush out pmdown_time work - we actually do want to run it
-         * now, we're shutting down so no imminent restart. */
-        list_for_each_entry(rtd, &card->rtd_list, list)
+        /*
+         * Flush out pmdown_time work - we actually do want to run it
+         * now, we're shutting down so no imminent restart.
+         */
+        for_each_card_rtds(card, rtd)
                 flush_delayed_work(&rtd->delayed_work);
 
         snd_soc_dapm_shutdown(card);
 
         /* deactivate pins to sleep state */
-        list_for_each_entry(rtd, &card->rtd_list, list) {
+        for_each_card_rtds(card, rtd) {
                 struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+                struct snd_soc_dai *codec_dai;
                 int i;
 
                 pinctrl_pm_select_sleep_state(cpu_dai->dev);
-                for (i = 0; i < rtd->num_codecs; i++) {
-                        struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+                for_each_rtd_codec_dai(rtd, i, codec_dai) {
                         pinctrl_pm_select_sleep_state(codec_dai->dev);
                 }
         }
@@ -2315,6 +2322,7 @@ static int snd_soc_add_controls(struct snd_card *card, struct device *dev,
 
         for (i = 0; i < num_controls; i++) {
                 const struct snd_kcontrol_new *control = &controls[i];
+
                 err = snd_ctl_add(card, snd_soc_cnew(control, data,
                                                      control->name, prefix));
                 if (err < 0) {
@@ -2432,8 +2440,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
  *
  * Configures the CODEC master (MCLK) or system (SYSCLK) clocking.
  */
-int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id,
-                             int source, unsigned int freq, int dir)
+int snd_soc_component_set_sysclk(struct snd_soc_component *component,
+                                 int clk_id, int source, unsigned int freq,
+                                 int dir)
 {
         if (component->driver->set_sysclk)
                 return component->driver->set_sysclk(component, clk_id, source,
@@ -2501,7 +2510,7 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id,
 {
         if (component->driver->set_pll)
                 return component->driver->set_pll(component, pll_id, source,
-                                              freq_in, freq_out);
+                                                  freq_in, freq_out);
 
         return -EINVAL;
 }
@@ -2532,8 +2541,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio);
  */
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
-        if (dai->driver == NULL)
-                return -EINVAL;
         if (dai->driver->ops->set_fmt == NULL)
                 return -ENOTSUPP;
         return dai->driver->ops->set_fmt(dai, fmt);
@@ -2549,8 +2556,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
  * Generates the TDM tx and rx slot default masks for DAI.
  */
 static int snd_soc_xlate_tdm_slot_mask(unsigned int slots,
-                                          unsigned int *tx_mask,
-                                          unsigned int *rx_mask)
+                                       unsigned int *tx_mask,
+                                       unsigned int *rx_mask)
 {
         if (*tx_mask || *rx_mask)
                 return 0;
@@ -2680,9 +2687,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
                              int direction)
 {
-        if (!dai->driver)
-                return -ENOTSUPP;
-
         if (dai->driver->ops->mute_stream)
                 return dai->driver->ops->mute_stream(dai, mute, direction);
         else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
@@ -2693,6 +2697,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
 
+static int snd_soc_bind_card(struct snd_soc_card *card)
+{
+        struct snd_soc_pcm_runtime *rtd;
+        int ret;
+
+        ret = snd_soc_instantiate_card(card);
+        if (ret != 0)
+                return ret;
+
+        /* deactivate pins to sleep state */
+        for_each_card_rtds(card, rtd) {
+                struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+                struct snd_soc_dai *codec_dai;
+                int j;
+
+                for_each_rtd_codec_dai(rtd, j, codec_dai) {
+                        if (!codec_dai->active)
+                                pinctrl_pm_select_sleep_state(codec_dai->dev);
+                }
+
+                if (!cpu_dai->active)
+                        pinctrl_pm_select_sleep_state(cpu_dai->dev);
+        }
+
+        return ret;
+}
+
 /**
  * snd_soc_register_card - Register a card with the ASoC core
  *
@@ -2702,13 +2733,12 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
 int snd_soc_register_card(struct snd_soc_card *card)
 {
         int i, ret;
-        struct snd_soc_pcm_runtime *rtd;
+        struct snd_soc_dai_link *link;
 
         if (!card->name || !card->dev)
                 return -EINVAL;
 
-        for (i = 0; i < card->num_links; i++) {
-                struct snd_soc_dai_link *link = &card->dai_link[i];
+        for_each_card_prelinks(card, i, link) {
 
                 ret = soc_init_dai_link(card, link);
                 if (ret) {
@@ -2723,7 +2753,6 @@ int snd_soc_register_card(struct snd_soc_card *card)
         snd_soc_initialize_card_lists(card);
 
         INIT_LIST_HEAD(&card->dai_link_list);
-        card->num_dai_links = 0;
 
         INIT_LIST_HEAD(&card->rtd_list);
         card->num_rtd = 0;
@@ -2734,28 +2763,23 @@ int snd_soc_register_card(struct snd_soc_card *card)
         mutex_init(&card->mutex);
         mutex_init(&card->dapm_mutex);
 
-        ret = snd_soc_instantiate_card(card);
-        if (ret != 0)
-                return ret;
-
-        /* deactivate pins to sleep state */
-        list_for_each_entry(rtd, &card->rtd_list, list)  {
-                struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-                int j;
-
-                for (j = 0; j < rtd->num_codecs; j++) {
-                        struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
-                        if (!codec_dai->active)
-                                pinctrl_pm_select_sleep_state(codec_dai->dev);
-                }
+        return snd_soc_bind_card(card);
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_card);
 
-                if (!cpu_dai->active)
-                        pinctrl_pm_select_sleep_state(cpu_dai->dev);
+static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister)
+{
+        if (card->instantiated) {
+                card->instantiated = false;
+                snd_soc_dapm_shutdown(card);
+                soc_cleanup_card_resources(card);
+                if (!unregister)
+                        list_add(&card->list, &unbind_card_list);
+        } else {
+                if (unregister)
+                        list_del(&card->list);
         }
-
-        return ret;
 }
-EXPORT_SYMBOL_GPL(snd_soc_register_card);
 
 /**
  * snd_soc_unregister_card - Unregister a card with the ASoC core
@@ -2765,12 +2789,8 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card);
  */
 int snd_soc_unregister_card(struct snd_soc_card *card)
 {
-        if (card->instantiated) {
-                card->instantiated = false;
-                snd_soc_dapm_shutdown(card);
-                soc_cleanup_card_resources(card);
-                dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
-        }
+        snd_soc_unbind_card(card, true);
+        dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
 
         return 0;
 }
@@ -2802,7 +2822,7 @@ static char *fmt_single_name(struct device *dev, int *id)
                 }
 
         } else {
-                /* I2C component devices are named "bus-addr"  */
+                /* I2C component devices are named "bus-addr" */
                 if (sscanf(name, "%x-%x", &id1, &id2) == 2) {
                         char tmp[NAME_SIZE];
 
@@ -2810,7 +2830,8 @@ static char *fmt_single_name(struct device *dev, int *id)
                         *id = ((id1 & 0xffff) << 16) + id2;
 
                         /* sanitize component name for DAI link creation */
-                        snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name, name);
+                        snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name,
+                                 name);
                         strlcpy(name, tmp, NAME_SIZE);
                 } else
                         *id = 0;
@@ -2845,7 +2866,7 @@ static void snd_soc_unregister_dais(struct snd_soc_component *component)
 {
         struct snd_soc_dai *dai, *_dai;
 
-        list_for_each_entry_safe(dai, _dai, &component->dai_list, list) {
+        for_each_component_dais_safe(component, dai, _dai) {
                 dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n",
                         dai->name);
                 list_del(&dai->list);
@@ -2877,7 +2898,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component,
          * component-less anymore.
          */
         if (legacy_dai_naming &&
-           (dai_drv->id == 0 || dai_drv->name == NULL)) {
+            (dai_drv->id == 0 || dai_drv->name == NULL)) {
                 dai->name = fmt_single_name(dev, &dai->id);
         } else {
                 dai->name = fmt_multiple_name(dev, dai_drv);
@@ -2897,6 +2918,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component,
         if (!dai->driver->ops)
                 dai->driver->ops = &null_dai_ops;
 
+        /* see for_each_component_dais */
         list_add_tail(&dai->list, &component->dai_list);
         component->num_dai++;
 
@@ -2910,11 +2932,10 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component,
  * @component: The component the DAIs are registered for
  * @dai_drv: DAI driver to use for the DAIs
  * @count: Number of DAIs
- * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the
- *                     parent's name.
  */
 static int snd_soc_register_dais(struct snd_soc_component *component,
-                                 struct snd_soc_dai_driver *dai_drv, size_t count)
+                                 struct snd_soc_dai_driver *dai_drv,
+                                 size_t count)
 {
         struct device *dev = component->dev;
         struct snd_soc_dai *dai;
@@ -2925,8 +2946,8 @@ static int snd_soc_register_dais(struct snd_soc_component *component,
 
         for (i = 0; i < count; i++) {
 
-                dai = soc_add_dai(component, dai_drv + i,
-                                  count == 1 && !component->driver->non_legacy_dai_naming);
+                dai = soc_add_dai(component, dai_drv + i, count == 1 &&
+                                  !component->driver->non_legacy_dai_naming);
                 if (dai == NULL) {
                         ret = -ENOMEM;
                         goto err;
@@ -2970,7 +2991,8 @@ int snd_soc_register_dai(struct snd_soc_component *component,
         if (!dai)
                 return -ENOMEM;
 
-        /* Create the DAI widgets here. After adding DAIs, topology may
+        /*
+         * Create the DAI widgets here. After adding DAIs, topology may
          * also add routes that need these widgets as source or sink.
          */
         ret = snd_soc_dapm_new_dai_widgets(dapm, dai);
@@ -3052,7 +3074,8 @@ static void snd_soc_component_setup_regmap(struct snd_soc_component *component)
 #ifdef CONFIG_REGMAP
 
 /**
- * snd_soc_component_init_regmap() - Initialize regmap instance for the component
+ * snd_soc_component_init_regmap() - Initialize regmap instance for the
+ *                                   component
  * @component: The component for which to initialize the regmap instance
  * @regmap: The regmap instance that should be used by the component
  *
@@ -3070,7 +3093,8 @@ void snd_soc_component_init_regmap(struct snd_soc_component *component,
 EXPORT_SYMBOL_GPL(snd_soc_component_init_regmap);
 
 /**
- * snd_soc_component_exit_regmap() - De-initialize regmap instance for the component
+ * snd_soc_component_exit_regmap() - De-initialize regmap instance for the
+ *                                   component
  * @component: The component for which to de-initialize the regmap instance
  *
  * Calls regmap_exit() on the regmap instance associated to the component and
@@ -3094,11 +3118,13 @@ static void snd_soc_component_add(struct snd_soc_component *component)
 
         if (!component->driver->write && !component->driver->read) {
                 if (!component->regmap)
-                        component->regmap = dev_get_regmap(component->dev, NULL);
+                        component->regmap = dev_get_regmap(component->dev,
+                                                           NULL);
                 if (component->regmap)
                         snd_soc_component_setup_regmap(component);
         }
 
+        /* see for_each_component */
         list_add(&component->list, &component_list);
         INIT_LIST_HEAD(&component->dobj_list);
 
@@ -3116,7 +3142,7 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component)
         struct snd_soc_card *card = component->card;
 
         if (card)
-                snd_soc_unregister_card(card);
+                snd_soc_unbind_card(card, false);
 
         list_del(&component->list);
 }
@@ -3156,6 +3182,18 @@ static void convert_endianness_formats(struct snd_soc_pcm_stream *stream)
                         stream->formats |= endianness_format_map[i];
 }
 
+static void snd_soc_try_rebind_card(void)
+{
+        struct snd_soc_card *card, *c;
+
+        if (!list_empty(&unbind_card_list)) {
+                list_for_each_entry_safe(card, c, &unbind_card_list, list) {
+                        if (!snd_soc_bind_card(card))
+                                list_del(&card->list);
+                }
+        }
+}
+
 int snd_soc_add_component(struct device *dev,
                         struct snd_soc_component *component,
                         const struct snd_soc_component_driver *component_driver,
@@ -3183,6 +3221,7 @@ int snd_soc_add_component(struct device *dev,
         }
 
         snd_soc_component_add(component);
+        snd_soc_try_rebind_card();
 
         return 0;
 
@@ -3221,27 +3260,28 @@ static int __snd_soc_unregister_component(struct device *dev)
         int found = 0;
 
         mutex_lock(&client_mutex);
-        list_for_each_entry(component, &component_list, list) {
+        for_each_component(component) {
                 if (dev != component->dev)
                         continue;
 
-                snd_soc_tplg_component_remove(component, SND_SOC_TPLG_INDEX_ALL);
+                snd_soc_tplg_component_remove(component,
+                                              SND_SOC_TPLG_INDEX_ALL);
                 snd_soc_component_del_unlocked(component);
                 found = 1;
                 break;
         }
         mutex_unlock(&client_mutex);
 
-        if (found) {
+        if (found)
                 snd_soc_component_cleanup(component);
-        }
 
         return found;
 }
 
 void snd_soc_unregister_component(struct device *dev)
 {
-        while (__snd_soc_unregister_component(dev));
+        while (__snd_soc_unregister_component(dev))
+                ;
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_component);
 
@@ -3253,7 +3293,7 @@ struct snd_soc_component *snd_soc_lookup_component(struct device *dev,
 
         ret = NULL;
         mutex_lock(&client_mutex);
-        list_for_each_entry(component, &component_list, list) {
+        for_each_component(component) {
                 if (dev != component->dev)
                         continue;
 
@@ -3653,7 +3693,7 @@ int snd_soc_get_dai_id(struct device_node *ep)
          */
         ret = -ENOTSUPP;
         mutex_lock(&client_mutex);
-        list_for_each_entry(pos, &component_list, list) {
+        for_each_component(pos) {
                 struct device_node *component_of_node = pos->dev->of_node;
 
                 if (!component_of_node && pos->dev->parent)
@@ -3683,7 +3723,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args,
         int ret = -EPROBE_DEFER;
 
         mutex_lock(&client_mutex);
-        list_for_each_entry(pos, &component_list, list) {
+        for_each_component(pos) {
                 component_of_node = pos->dev->of_node;
                 if (!component_of_node && pos->dev->parent)
                         component_of_node = pos->dev->parent->of_node;
@@ -3719,7 +3759,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args,
                         ret = 0;
 
                         /* find target DAI */
-                        list_for_each_entry(dai, &pos->dai_list, list) {
+                        for_each_component_dais(pos, dai) {
                                 if (id == 0)
                                         break;
                                 id--;
@@ -3764,10 +3804,10 @@ EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name);
  */
 void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link)
 {
-        struct snd_soc_dai_link_component *component = dai_link->codecs;
+        struct snd_soc_dai_link_component *component;
         int index;
 
-        for (index = 0; index < dai_link->num_codecs; index++, component++) {
+        for_each_link_codecs(dai_link, index, component) {
                 if (!component->of_node)
                         break;
                 of_node_put(component->of_node);
@@ -3819,12 +3859,10 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev,
         dai_link->num_codecs = num_codecs;
 
         /* Parse the list */
-        for (index = 0, component = dai_link->codecs;
-             index < dai_link->num_codecs;
-             index++, component++) {
+        for_each_link_codecs(dai_link, index, component) {
                 ret = of_parse_phandle_with_args(of_node, name,
                                                  "#sound-dai-cells",
-                                                  index, &args);
+                                                 index, &args);
                 if (ret)
                         goto err;
                 component->of_node = args.np;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 461d951917c0..a5178845065b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -18,7 +18,6 @@
 //      device reopen.
 
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/async.h>
 #include <linux/delay.h>
@@ -364,10 +363,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
                                 ret = PTR_ERR(data->widget);
                                 goto err_data;
                         }
-                        if (!data->widget) {
-                                ret = -ENOMEM;
-                                goto err_data;
-                        }
                 }
                 break;
         case snd_soc_dapm_demux:
@@ -402,10 +397,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
                                 ret = PTR_ERR(data->widget);
                                 goto err_data;
                         }
-                        if (!data->widget) {
-                                ret = -ENOMEM;
-                                goto err_data;
-                        }
 
                         snd_soc_dapm_add_path(widget->dapm, data->widget,
                                               widget, NULL, NULL);
@@ -1026,9 +1017,10 @@ static int dapm_new_dai_link(struct snd_soc_dapm_widget *w)
         struct snd_kcontrol *kcontrol;
         struct snd_soc_dapm_context *dapm = w->dapm;
         struct snd_card *card = dapm->card->snd_card;
+        struct snd_soc_pcm_runtime *rtd = w->priv;
 
         /* create control for links with > 1 config */
-        if (w->num_params <= 1)
+        if (rtd->dai_link->num_params <= 1)
                 return 0;
 
         /* add kcontrol */
@@ -1320,14 +1312,13 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w,
 
         soc_dapm_async_complete(w->dapm);
 
-#ifdef CONFIG_HAVE_CLK
         if (SND_SOC_DAPM_EVENT_ON(event)) {
                 return clk_prepare_enable(w->clk);
         } else {
                 clk_disable_unprepare(w->clk);
                 return 0;
         }
-#endif
+
         return 0;
 }
 EXPORT_SYMBOL_GPL(dapm_clock_event);
@@ -1953,7 +1944,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
         dapm_pre_sequence_async(&card->dapm, 0);
         /* Run other bias changes in parallel */
         list_for_each_entry(d, &card->dapm_list, list) {
-                if (d != &card->dapm)
+                if (d != &card->dapm && d->bias_level != d->target_bias_level)
                         async_schedule_domain(dapm_pre_sequence_async, d,
                                                 &async_domain);
         }
@@ -1977,7 +1968,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
 
         /* Run all the bias changes in parallel */
         list_for_each_entry(d, &card->dapm_list, list) {
-                if (d != &card->dapm)
+                if (d != &card->dapm && d->bias_level != d->target_bias_level)
                         async_schedule_domain(dapm_post_sequence_async, d,
                                                 &async_domain);
         }
@@ -2371,12 +2362,13 @@ static ssize_t dapm_widget_show(struct device *dev,
         struct device_attribute *attr, char *buf)
 {
         struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
+        struct snd_soc_dai *codec_dai;
         int i, count = 0;
 
         mutex_lock(&rtd->card->dapm_mutex);
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                struct snd_soc_component *cmpnt = rtd->codec_dais[i]->component;
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
+                struct snd_soc_component *cmpnt = codec_dai->component;
 
                 count += dapm_widget_show_component(cmpnt, buf + count);
         }
@@ -3426,35 +3418,6 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch);
 
 struct snd_soc_dapm_widget *
-snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
-        const struct snd_soc_dapm_widget *widget)
-{
-        struct snd_soc_dapm_widget *w;
-
-        mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
-        w = snd_soc_dapm_new_control_unlocked(dapm, widget);
-        /* Do not nag about probe deferrals */
-        if (IS_ERR(w)) {
-                int ret = PTR_ERR(w);
-
-                if (ret != -EPROBE_DEFER)
-                        dev_err(dapm->dev,
-                                "ASoC: Failed to create DAPM control %s (%d)\n",
-                                widget->name, ret);
-                goto out_unlock;
-        }
-        if (!w)
-                dev_err(dapm->dev,
-                        "ASoC: Failed to create DAPM control %s\n",
-                        widget->name);
-
-out_unlock:
-        mutex_unlock(&dapm->card->dapm_mutex);
-        return w;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
-
-struct snd_soc_dapm_widget *
 snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
                          const struct snd_soc_dapm_widget *widget)
 {
@@ -3464,53 +3427,37 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
         int ret;
 
         if ((w = dapm_cnew_widget(widget)) == NULL)
-                return NULL;
+                return ERR_PTR(-ENOMEM);
 
         switch (w->id) {
         case snd_soc_dapm_regulator_supply:
                 w->regulator = devm_regulator_get(dapm->dev, w->name);
                 if (IS_ERR(w->regulator)) {
                         ret = PTR_ERR(w->regulator);
-                        if (ret == -EPROBE_DEFER)
-                                return ERR_PTR(ret);
-                        dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
-                                w->name, ret);
-                        return NULL;
+                        goto request_failed;
                 }
 
                 if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) {
                         ret = regulator_allow_bypass(w->regulator, true);
                         if (ret != 0)
-                                dev_warn(w->dapm->dev,
+                                dev_warn(dapm->dev,
                                          "ASoC: Failed to bypass %s: %d\n",
                                          w->name, ret);
                 }
                 break;
         case snd_soc_dapm_pinctrl:
                 w->pinctrl = devm_pinctrl_get(dapm->dev);
-                if (IS_ERR_OR_NULL(w->pinctrl)) {
+                if (IS_ERR(w->pinctrl)) {
                         ret = PTR_ERR(w->pinctrl);
-                        if (ret == -EPROBE_DEFER)
-                                return ERR_PTR(ret);
-                        dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
-                                w->name, ret);
-                        return NULL;
+                        goto request_failed;
                 }
                 break;
         case snd_soc_dapm_clock_supply:
-#ifdef CONFIG_CLKDEV_LOOKUP
                 w->clk = devm_clk_get(dapm->dev, w->name);
                 if (IS_ERR(w->clk)) {
                         ret = PTR_ERR(w->clk);
-                        if (ret == -EPROBE_DEFER)
-                                return ERR_PTR(ret);
-                        dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
-                                w->name, ret);
-                        return NULL;
+                        goto request_failed;
                 }
-#else
-                return NULL;
-#endif
                 break;
         default:
                 break;
@@ -3523,7 +3470,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
                 w->name = kstrdup_const(widget->name, GFP_KERNEL);
         if (w->name == NULL) {
                 kfree(w);
-                return NULL;
+                return ERR_PTR(-ENOMEM);
         }
 
         switch (w->id) {
@@ -3600,7 +3547,37 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
         /* machine layer sets up unconnected pins and insertions */
         w->connected = 1;
         return w;
+
+request_failed:
+        if (ret != -EPROBE_DEFER)
+                dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
+                        w->name, ret);
+
+        return ERR_PTR(ret);
+}
+
+/**
+ * snd_soc_dapm_new_control - create new dapm control
+ * @dapm: DAPM context
+ * @widget: widget template
+ *
+ * Creates new DAPM control based upon a template.
+ *
+ * Returns a widget pointer on success or an error pointer on failure
+ */
+struct snd_soc_dapm_widget *
+snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
+                         const struct snd_soc_dapm_widget *widget)
+{
+        struct snd_soc_dapm_widget *w;
+
+        mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+        w = snd_soc_dapm_new_control_unlocked(dapm, widget);
+        mutex_unlock(&dapm->card->dapm_mutex);
+
+        return w;
 }
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
 
 /**
  * snd_soc_dapm_new_controls - create new dapm controls
@@ -3625,19 +3602,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
                 w = snd_soc_dapm_new_control_unlocked(dapm, widget);
                 if (IS_ERR(w)) {
                         ret = PTR_ERR(w);
-                        /* Do not nag about probe deferrals */
-                        if (ret == -EPROBE_DEFER)
-                                break;
-                        dev_err(dapm->dev,
-                                "ASoC: Failed to create DAPM control %s (%d)\n",
-                                widget->name, ret);
-                        break;
-                }
-                if (!w) {
-                        dev_err(dapm->dev,
-                                "ASoC: Failed to create DAPM control %s\n",
-                                widget->name);
-                        ret = -ENOMEM;
                         break;
                 }
                 widget++;
@@ -3650,32 +3614,23 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);
 static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
                                   struct snd_kcontrol *kcontrol, int event)
 {
-        struct snd_soc_dapm_path *source_p, *sink_p;
+        struct snd_soc_dapm_path *path;
         struct snd_soc_dai *source, *sink;
         struct snd_soc_pcm_runtime *rtd = w->priv;
-        const struct snd_soc_pcm_stream *config = w->params + w->params_select;
+        const struct snd_soc_pcm_stream *config;
         struct snd_pcm_substream substream;
         struct snd_pcm_hw_params *params = NULL;
         struct snd_pcm_runtime *runtime = NULL;
         unsigned int fmt;
-        int ret;
+        int ret = 0;
+
+        config = rtd->dai_link->params + rtd->params_select;
 
         if (WARN_ON(!config) ||
             WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) ||
                     list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])))
                 return -EINVAL;
 
-        /* We only support a single source and sink, pick the first */
-        source_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_OUT],
-                                    struct snd_soc_dapm_path,
-                                    list_node[SND_SOC_DAPM_DIR_OUT]);
-        sink_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_IN],
-                                    struct snd_soc_dapm_path,
-                                    list_node[SND_SOC_DAPM_DIR_IN]);
-
-        source = source_p->source->priv;
-        sink = sink_p->sink->priv;
-
         /* Be a little careful as we don't want to overflow the mask array */
         if (config->formats) {
                 fmt = ffs(config->formats) - 1;
@@ -3717,59 +3672,95 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
         switch (event) {
         case SND_SOC_DAPM_PRE_PMU:
                 substream.stream = SNDRV_PCM_STREAM_CAPTURE;
-                if (source->driver->ops->startup) {
-                        ret = source->driver->ops->startup(&substream, source);
-                        if (ret < 0) {
-                                dev_err(source->dev,
-                                        "ASoC: startup() failed: %d\n", ret);
-                                goto out;
+                snd_soc_dapm_widget_for_each_source_path(w, path) {
+                        source = path->source->priv;
+
+                        if (source->driver->ops->startup) {
+                                ret = source->driver->ops->startup(&substream,
+                                                                   source);
+                                if (ret < 0) {
+                                        dev_err(source->dev,
+                                                "ASoC: startup() failed: %d\n",
+                                                ret);
+                                        goto out;
+                                }
+                                source->active++;
                         }
-                        source->active++;
+                        ret = soc_dai_hw_params(&substream, params, source);
+                        if (ret < 0)
+                                goto out;
                 }
-                ret = soc_dai_hw_params(&substream, params, source);
-                if (ret < 0)
-                        goto out;
 
                 substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
-                if (sink->driver->ops->startup) {
-                        ret = sink->driver->ops->startup(&substream, sink);
-                        if (ret < 0) {
-                                dev_err(sink->dev,
-                                        "ASoC: startup() failed: %d\n", ret);
-                                goto out;
+                snd_soc_dapm_widget_for_each_sink_path(w, path) {
+                        sink = path->sink->priv;
+
+                        if (sink->driver->ops->startup) {
+                                ret = sink->driver->ops->startup(&substream,
+                                                                 sink);
+                                if (ret < 0) {
+                                        dev_err(sink->dev,
+                                                "ASoC: startup() failed: %d\n",
+                                                ret);
+                                        goto out;
+                                }
+                                sink->active++;
                         }
-                        sink->active++;
+                        ret = soc_dai_hw_params(&substream, params, sink);
+                        if (ret < 0)
+                                goto out;
                 }
-                ret = soc_dai_hw_params(&substream, params, sink);
-                if (ret < 0)
-                        goto out;
                 break;
 
         case SND_SOC_DAPM_POST_PMU:
-                ret = snd_soc_dai_digital_mute(sink, 0,
-                                               SNDRV_PCM_STREAM_PLAYBACK);
-                if (ret != 0 && ret != -ENOTSUPP)
-                        dev_warn(sink->dev, "ASoC: Failed to unmute: %d\n", ret);
-                ret = 0;
+                snd_soc_dapm_widget_for_each_sink_path(w, path) {
+                        sink = path->sink->priv;
+
+                        ret = snd_soc_dai_digital_mute(sink, 0,
+                                                       SNDRV_PCM_STREAM_PLAYBACK);
+                        if (ret != 0 && ret != -ENOTSUPP)
+                                dev_warn(sink->dev,
+                                         "ASoC: Failed to unmute: %d\n", ret);
+                        ret = 0;
+                }
                 break;
 
         case SND_SOC_DAPM_PRE_PMD:
-                ret = snd_soc_dai_digital_mute(sink, 1,
-                                               SNDRV_PCM_STREAM_PLAYBACK);
-                if (ret != 0 && ret != -ENOTSUPP)
-                        dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret);
-                ret = 0;
+                snd_soc_dapm_widget_for_each_sink_path(w, path) {
+                        sink = path->sink->priv;
+
+                        ret = snd_soc_dai_digital_mute(sink, 1,
+                                                       SNDRV_PCM_STREAM_PLAYBACK);
+                        if (ret != 0 && ret != -ENOTSUPP)
+                                dev_warn(sink->dev,
+                                         "ASoC: Failed to mute: %d\n", ret);
+                        ret = 0;
+                }
 
-                source->active--;
-                if (source->driver->ops->shutdown) {
-                        substream.stream = SNDRV_PCM_STREAM_CAPTURE;
-                        source->driver->ops->shutdown(&substream, source);
+                substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+                snd_soc_dapm_widget_for_each_source_path(w, path) {
+                        source = path->source->priv;
+
+                        if (source->driver->ops->hw_free)
+                                source->driver->ops->hw_free(&substream,
+                                                             source);
+
+                        source->active--;
+                        if (source->driver->ops->shutdown)
+                                source->driver->ops->shutdown(&substream,
+                                                              source);
                 }
 
-                sink->active--;
-                if (sink->driver->ops->shutdown) {
-                        substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
-                        sink->driver->ops->shutdown(&substream, sink);
+                substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+                snd_soc_dapm_widget_for_each_sink_path(w, path) {
+                        sink = path->sink->priv;
+
+                        if (sink->driver->ops->hw_free)
+                                sink->driver->ops->hw_free(&substream, sink);
+
+                        sink->active--;
+                        if (sink->driver->ops->shutdown)
+                                sink->driver->ops->shutdown(&substream, sink);
                 }
                 break;
 
@@ -3788,8 +3779,9 @@ static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol,
                           struct snd_ctl_elem_value *ucontrol)
 {
         struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol);
+        struct snd_soc_pcm_runtime *rtd = w->priv;
 
-        ucontrol->value.enumerated.item[0] = w->params_select;
+        ucontrol->value.enumerated.item[0] = rtd->params_select;
 
         return 0;
 }
@@ -3798,18 +3790,19 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol,
                           struct snd_ctl_elem_value *ucontrol)
 {
         struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol);
+        struct snd_soc_pcm_runtime *rtd = w->priv;
 
         /* Can't change the config when widget is already powered */
         if (w->power)
                 return -EBUSY;
 
-        if (ucontrol->value.enumerated.item[0] == w->params_select)
+        if (ucontrol->value.enumerated.item[0] == rtd->params_select)
                 return 0;
 
-        if (ucontrol->value.enumerated.item[0] >= w->num_params)
+        if (ucontrol->value.enumerated.item[0] >= rtd->dai_link->num_params)
                 return -EINVAL;
 
-        w->params_select = ucontrol->value.enumerated.item[0];
+        rtd->params_select = ucontrol->value.enumerated.item[0];
 
         return 0;
 }
@@ -3896,12 +3889,10 @@ outfree_w_param:
         return NULL;
 }
 
-int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
-                         struct snd_soc_pcm_runtime *rtd,
-                         const struct snd_soc_pcm_stream *params,
-                         unsigned int num_params,
-                         struct snd_soc_dapm_widget *source,
-                         struct snd_soc_dapm_widget *sink)
+static struct snd_soc_dapm_widget *
+snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
+                     struct snd_soc_dapm_widget *source,
+                     struct snd_soc_dapm_widget *sink)
 {
         struct snd_soc_dapm_widget template;
         struct snd_soc_dapm_widget *w;
@@ -3913,7 +3904,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
         link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s",
                                    source->name, sink->name);
         if (!link_name)
-                return -ENOMEM;
+                return ERR_PTR(-ENOMEM);
 
         memset(&template, 0, sizeof(template));
         template.reg = SND_SOC_NOPM;
@@ -3925,9 +3916,10 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
         template.kcontrol_news = NULL;
 
         /* allocate memory for control, only in case of multiple configs */
-        if (num_params > 1) {
-                w_param_text = devm_kcalloc(card->dev, num_params,
-                                        sizeof(char *), GFP_KERNEL);
+        if (rtd->dai_link->num_params > 1) {
+                w_param_text = devm_kcalloc(card->dev,
+                                            rtd->dai_link->num_params,
+                                            sizeof(char *), GFP_KERNEL);
                 if (!w_param_text) {
                         ret = -ENOMEM;
                         goto param_fail;
@@ -3936,7 +3928,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
                 template.num_kcontrols = 1;
                 template.kcontrol_news =
                                         snd_soc_dapm_alloc_kcontrol(card,
-                                                link_name, params, num_params,
+                                                link_name,
+                                                rtd->dai_link->params,
+                                                rtd->dai_link->num_params,
                                                 w_param_text, &private_value);
                 if (!template.kcontrol_news) {
                         ret = -ENOMEM;
@@ -3950,37 +3944,20 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
         w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template);
         if (IS_ERR(w)) {
                 ret = PTR_ERR(w);
-                /* Do not nag about probe deferrals */
-                if (ret != -EPROBE_DEFER)
-                        dev_err(card->dev,
-                                "ASoC: Failed to create %s widget (%d)\n",
-                                link_name, ret);
-                goto outfree_kcontrol_news;
-        }
-        if (!w) {
-                dev_err(card->dev, "ASoC: Failed to create %s widget\n",
-                        link_name);
-                ret = -ENOMEM;
                 goto outfree_kcontrol_news;
         }
 
-        w->params = params;
-        w->num_params = num_params;
         w->priv = rtd;
 
-        ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL);
-        if (ret)
-                goto outfree_w;
-        return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL);
+        return w;
 
-outfree_w:
-        devm_kfree(card->dev, w);
 outfree_kcontrol_news:
         devm_kfree(card->dev, (void *)template.kcontrol_news);
-        snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text);
+        snd_soc_dapm_free_kcontrol(card, &private_value,
+                                   rtd->dai_link->num_params, w_param_text);
 param_fail:
         devm_kfree(card->dev, link_name);
-        return ret;
+        return ERR_PTR(ret);
 }
 
 int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
@@ -4003,21 +3980,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
                         template.name);
 
                 w = snd_soc_dapm_new_control_unlocked(dapm, &template);
-                if (IS_ERR(w)) {
-                        int ret = PTR_ERR(w);
-
-                        /* Do not nag about probe deferrals */
-                        if (ret != -EPROBE_DEFER)
-                                dev_err(dapm->dev,
-                                "ASoC: Failed to create %s widget (%d)\n",
-                                dai->driver->playback.stream_name, ret);
-                        return ret;
-                }
-                if (!w) {
-                        dev_err(dapm->dev, "ASoC: Failed to create %s widget\n",
-                                dai->driver->playback.stream_name);
-                        return -ENOMEM;
-                }
+                if (IS_ERR(w))
+                        return PTR_ERR(w);
 
                 w->priv = dai;
                 dai->playback_widget = w;
@@ -4032,21 +3996,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
                         template.name);
 
                 w = snd_soc_dapm_new_control_unlocked(dapm, &template);
-                if (IS_ERR(w)) {
-                        int ret = PTR_ERR(w);
-
-                        /* Do not nag about probe deferrals */
-                        if (ret != -EPROBE_DEFER)
-                                dev_err(dapm->dev,
-                                "ASoC: Failed to create %s widget (%d)\n",
-                                dai->driver->playback.stream_name, ret);
-                        return ret;
-                }
-                if (!w) {
-                        dev_err(dapm->dev, "ASoC: Failed to create %s widget\n",
-                                dai->driver->capture.stream_name);
-                        return -ENOMEM;
-                }
+                if (IS_ERR(w))
+                        return PTR_ERR(w);
 
                 w->priv = dai;
                 dai->capture_widget = w;
@@ -4115,34 +4066,79 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
                                           struct snd_soc_pcm_runtime *rtd)
 {
         struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-        struct snd_soc_dapm_widget *sink, *source;
+        struct snd_soc_dai *codec_dai;
+        struct snd_soc_dapm_widget *playback = NULL, *capture = NULL;
+        struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu;
         int i;
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+        if (rtd->dai_link->params) {
+                playback_cpu = cpu_dai->capture_widget;
+                capture_cpu = cpu_dai->playback_widget;
+        } else {
+                playback = cpu_dai->playback_widget;
+                capture = cpu_dai->capture_widget;
+                playback_cpu = playback;
+                capture_cpu = capture;
+        }
+
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
 
                 /* connect BE DAI playback if widgets are valid */
-                if (codec_dai->playback_widget && cpu_dai->playback_widget) {
-                        source = cpu_dai->playback_widget;
-                        sink = codec_dai->playback_widget;
+                codec = codec_dai->playback_widget;
+
+                if (playback_cpu && codec) {
+                        if (!playback) {
+                                playback = snd_soc_dapm_new_dai(card, rtd,
+                                                                playback_cpu,
+                                                                codec);
+                                if (IS_ERR(playback)) {
+                                        dev_err(rtd->dev,
+                                                "ASoC: Failed to create DAI %s: %ld\n",
+                                                codec_dai->name,
+                                                PTR_ERR(playback));
+                                        continue;
+                                }
+
+                                snd_soc_dapm_add_path(&card->dapm, playback_cpu,
+                                                      playback, NULL, NULL);
+                        }
+
                         dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
-                                cpu_dai->component->name, source->name,
-                                codec_dai->component->name, sink->name);
+                                cpu_dai->component->name, playback_cpu->name,
+                                codec_dai->component->name, codec->name);
 
-                        snd_soc_dapm_add_path(&card->dapm, source, sink,
-                                NULL, NULL);
+                        snd_soc_dapm_add_path(&card->dapm, playback, codec,
+                                              NULL, NULL);
                 }
+        }
 
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 /* connect BE DAI capture if widgets are valid */
-                if (codec_dai->capture_widget && cpu_dai->capture_widget) {
-                        source = codec_dai->capture_widget;
-                        sink = cpu_dai->capture_widget;
+                codec = codec_dai->capture_widget;
+
+                if (codec && capture_cpu) {
+                        if (!capture) {
+                                capture = snd_soc_dapm_new_dai(card, rtd,
+                                                               codec,
+                                                               capture_cpu);
+                                if (IS_ERR(capture)) {
+                                        dev_err(rtd->dev,
+                                                "ASoC: Failed to create DAI %s: %ld\n",
+                                                codec_dai->name,
+                                                PTR_ERR(capture));
+                                        continue;
+                                }
+
+                                snd_soc_dapm_add_path(&card->dapm, capture,
+                                                      capture_cpu, NULL, NULL);
+                        }
+
                         dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
-                                codec_dai->component->name, source->name,
-                                cpu_dai->component->name, sink->name);
+                                codec_dai->component->name, codec->name,
+                                cpu_dai->component->name, capture_cpu->name);
 
-                        snd_soc_dapm_add_path(&card->dapm, source, sink,
-                                NULL, NULL);
+                        snd_soc_dapm_add_path(&card->dapm, codec, capture,
+                                              NULL, NULL);
                 }
         }
 }
@@ -4192,12 +4188,12 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
         struct snd_soc_pcm_runtime *rtd;
 
         /* for each BE DAI link... */
-        list_for_each_entry(rtd, &card->rtd_list, list)  {
+        for_each_card_rtds(card, rtd)  {
                 /*
                  * dynamic FE links have no fixed DAI mapping.
                  * CODEC<->CODEC links have no direct connection.
                  */
-                if (rtd->dai_link->dynamic || rtd->dai_link->params)
+                if (rtd->dai_link->dynamic)
                         continue;
 
                 dapm_connect_dai_link_widgets(card, rtd);
@@ -4207,11 +4203,12 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
 static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
         int event)
 {
+        struct snd_soc_dai *codec_dai;
         int i;
 
         soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event);
-        for (i = 0; i < rtd->num_codecs; i++)
-                soc_dapm_dai_stream_event(rtd->codec_dais[i], stream, event);
+        for_each_rtd_codec_dai(rtd, i, codec_dai)
+                soc_dapm_dai_stream_event(codec_dai, stream, event);
 
         dapm_power_widgets(rtd->card, event);
 }
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 592efb370c44..f4dc3d445aae 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -373,7 +373,7 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol,
         unsigned int rshift = mc->rshift;
         int max = mc->max;
         int min = mc->min;
-        unsigned int mask = (1 << (fls(min + max) - 1)) - 1;
+        unsigned int mask = (1U << (fls(min + max) - 1)) - 1;
         unsigned int val;
         int ret;
 
@@ -418,7 +418,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol,
         unsigned int rshift = mc->rshift;
         int max = mc->max;
         int min = mc->min;
-        unsigned int mask = (1 << (fls(min + max) - 1)) - 1;
+        unsigned int mask = (1U << (fls(min + max) - 1)) - 1;
         int err = 0;
         unsigned int val, val_mask, val2 = 0;
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e8b98bfd4cf1..03f36e534050 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -59,25 +59,26 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream)
 void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
 {
         struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+        struct snd_soc_dai *codec_dai;
         int i;
 
         lockdep_assert_held(&rtd->pcm_mutex);
 
         if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
                 cpu_dai->playback_active++;
-                for (i = 0; i < rtd->num_codecs; i++)
-                        rtd->codec_dais[i]->playback_active++;
+                for_each_rtd_codec_dai(rtd, i, codec_dai)
+                        codec_dai->playback_active++;
         } else {
                 cpu_dai->capture_active++;
-                for (i = 0; i < rtd->num_codecs; i++)
-                        rtd->codec_dais[i]->capture_active++;
+                for_each_rtd_codec_dai(rtd, i, codec_dai)
+                        codec_dai->capture_active++;
         }
 
         cpu_dai->active++;
         cpu_dai->component->active++;
-        for (i = 0; i < rtd->num_codecs; i++) {
-                rtd->codec_dais[i]->active++;
-                rtd->codec_dais[i]->component->active++;
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
+                codec_dai->active++;
+                codec_dai->component->active++;
         }
 }
 
@@ -94,25 +95,26 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
 void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream)
 {
         struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+        struct snd_soc_dai *codec_dai;
         int i;
 
         lockdep_assert_held(&rtd->pcm_mutex);
 
         if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
                 cpu_dai->playback_active--;
-                for (i = 0; i < rtd->num_codecs; i++)
-                        rtd->codec_dais[i]->playback_active--;
+                for_each_rtd_codec_dai(rtd, i, codec_dai)
+                        codec_dai->playback_active--;
         } else {
                 cpu_dai->capture_active--;
-                for (i = 0; i < rtd->num_codecs; i++)
-                        rtd->codec_dais[i]->capture_active--;
+                for_each_rtd_codec_dai(rtd, i, codec_dai)
+                        codec_dai->capture_active--;
         }
 
         cpu_dai->active--;
         cpu_dai->component->active--;
-        for (i = 0; i < rtd->num_codecs; i++) {
-                rtd->codec_dais[i]->component->active--;
-                rtd->codec_dais[i]->active--;
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
+                codec_dai->component->active--;
+                codec_dai->active--;
         }
 }
 
@@ -172,7 +174,7 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir,
 {
         struct snd_soc_dpcm *dpcm;
 
-        list_for_each_entry(dpcm, &fe->dpcm[dir].be_clients, list_be) {
+        for_each_dpcm_be(fe, dir, dpcm) {
 
                 struct snd_soc_pcm_runtime *be = dpcm->be;
 
@@ -253,6 +255,7 @@ static int soc_pcm_params_symmetry(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 *codec_dai;
         unsigned int rate, channels, sample_bits, symmetry, i;
 
         rate = params_rate(params);
@@ -263,8 +266,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
         symmetry = cpu_dai->driver->symmetric_rates ||
                 rtd->dai_link->symmetric_rates;
 
-        for (i = 0; i < rtd->num_codecs; i++)
-                symmetry |= rtd->codec_dais[i]->driver->symmetric_rates;
+        for_each_rtd_codec_dai(rtd, i, codec_dai)
+                symmetry |= codec_dai->driver->symmetric_rates;
 
         if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) {
                 dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n",
@@ -275,8 +278,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
         symmetry = cpu_dai->driver->symmetric_channels ||
                 rtd->dai_link->symmetric_channels;
 
-        for (i = 0; i < rtd->num_codecs; i++)
-                symmetry |= rtd->codec_dais[i]->driver->symmetric_channels;
+        for_each_rtd_codec_dai(rtd, i, codec_dai)
+                symmetry |= codec_dai->driver->symmetric_channels;
 
         if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) {
                 dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n",
@@ -287,8 +290,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
         symmetry = cpu_dai->driver->symmetric_samplebits ||
                 rtd->dai_link->symmetric_samplebits;
 
-        for (i = 0; i < rtd->num_codecs; i++)
-                symmetry |= rtd->codec_dais[i]->driver->symmetric_samplebits;
+        for_each_rtd_codec_dai(rtd, i, codec_dai)
+                symmetry |= codec_dai->driver->symmetric_samplebits;
 
         if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) {
                 dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n",
@@ -304,17 +307,18 @@ static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream)
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver;
         struct snd_soc_dai_link *link = rtd->dai_link;
+        struct snd_soc_dai *codec_dai;
         unsigned int symmetry, i;
 
         symmetry = cpu_driver->symmetric_rates || link->symmetric_rates ||
                 cpu_driver->symmetric_channels || link->symmetric_channels ||
                 cpu_driver->symmetric_samplebits || link->symmetric_samplebits;
 
-        for (i = 0; i < rtd->num_codecs; i++)
+        for_each_rtd_codec_dai(rtd, i, codec_dai)
                 symmetry = symmetry ||
-                        rtd->codec_dais[i]->driver->symmetric_rates ||
-                        rtd->codec_dais[i]->driver->symmetric_channels ||
-                        rtd->codec_dais[i]->driver->symmetric_samplebits;
+                        codec_dai->driver->symmetric_rates ||
+                        codec_dai->driver->symmetric_channels ||
+                        codec_dai->driver->symmetric_samplebits;
 
         return symmetry;
 }
@@ -342,8 +346,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream)
         unsigned int bits = 0, cpu_bits;
 
         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-                for (i = 0; i < rtd->num_codecs; i++) {
-                        codec_dai = rtd->codec_dais[i];
+                for_each_rtd_codec_dai(rtd, i, codec_dai) {
                         if (codec_dai->driver->playback.sig_bits == 0) {
                                 bits = 0;
                                 break;
@@ -352,8 +355,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream)
                 }
                 cpu_bits = cpu_dai->driver->playback.sig_bits;
         } else {
-                for (i = 0; i < rtd->num_codecs; i++) {
-                        codec_dai = rtd->codec_dais[i];
+                for_each_rtd_codec_dai(rtd, i, codec_dai) {
                         if (codec_dai->driver->capture.sig_bits == 0) {
                                 bits = 0;
                                 break;
@@ -372,6 +374,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
         struct snd_pcm_runtime *runtime = substream->runtime;
         struct snd_pcm_hardware *hw = &runtime->hw;
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_dai *codec_dai;
         struct snd_soc_dai_driver *cpu_dai_drv = rtd->cpu_dai->driver;
         struct snd_soc_dai_driver *codec_dai_drv;
         struct snd_soc_pcm_stream *codec_stream;
@@ -388,7 +391,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
                 cpu_stream = &cpu_dai_drv->capture;
 
         /* first calculate min/max only for CODECs in the DAI link */
-        for (i = 0; i < rtd->num_codecs; i++) {
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
 
                 /*
                  * Skip CODECs which don't support the current stream type.
@@ -399,11 +402,11 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
                  * bailed out on a higher level, since there would be no
                  * CODEC to support the transfer direction in that case.
                  */
-                if (!snd_soc_dai_stream_valid(rtd->codec_dais[i],
+                if (!snd_soc_dai_stream_valid(codec_dai,
                                               substream->stream))
                         continue;
 
-                codec_dai_drv = rtd->codec_dais[i]->driver;
+                codec_dai_drv = codec_dai->driver;
                 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
                         codec_stream = &codec_dai_drv->playback;
                 else
@@ -482,8 +485,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
         int i, ret = 0;
 
         pinctrl_pm_select_default_state(cpu_dai->dev);
-        for (i = 0; i < rtd->num_codecs; i++)
-                pinctrl_pm_select_default_state(rtd->codec_dais[i]->dev);
+        for_each_rtd_codec_dai(rtd, i, codec_dai)
+                pinctrl_pm_select_default_state(codec_dai->dev);
 
         for_each_rtdcom(rtd, rtdcom) {
                 component = rtdcom->component;
@@ -520,8 +523,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
         }
         component = NULL;
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                codec_dai = rtd->codec_dais[i];
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 if (codec_dai->driver->ops->startup) {
                         ret = codec_dai->driver->ops->startup(substream,
                                                               codec_dai);
@@ -588,10 +590,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
                         goto config_err;
         }
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                if (rtd->codec_dais[i]->active) {
-                        ret = soc_pcm_apply_symmetry(substream,
-                                                     rtd->codec_dais[i]);
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
+                if (codec_dai->active) {
+                        ret = soc_pcm_apply_symmetry(substream, codec_dai);
                         if (ret != 0)
                                 goto config_err;
                 }
@@ -620,8 +621,7 @@ machine_err:
         i = rtd->num_codecs;
 
 codec_dai_err:
-        while (--i >= 0) {
-                codec_dai = rtd->codec_dais[i];
+        for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) {
                 if (codec_dai->driver->ops->shutdown)
                         codec_dai->driver->ops->shutdown(substream, codec_dai);
         }
@@ -641,9 +641,9 @@ out:
                 pm_runtime_put_autosuspend(component->dev);
         }
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                if (!rtd->codec_dais[i]->active)
-                        pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev);
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
+                if (!codec_dai->active)
+                        pinctrl_pm_select_sleep_state(codec_dai->dev);
         }
         if (!cpu_dai->active)
                 pinctrl_pm_select_sleep_state(cpu_dai->dev);
@@ -701,8 +701,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
         if (!cpu_dai->active)
                 cpu_dai->rate = 0;
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                codec_dai = rtd->codec_dais[i];
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 if (!codec_dai->active)
                         codec_dai->rate = 0;
         }
@@ -712,8 +711,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
         if (cpu_dai->driver->ops->shutdown)
                 cpu_dai->driver->ops->shutdown(substream, cpu_dai);
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                codec_dai = rtd->codec_dais[i];
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 if (codec_dai->driver->ops->shutdown)
                         codec_dai->driver->ops->shutdown(substream, codec_dai);
         }
@@ -751,9 +749,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
                 pm_runtime_put_autosuspend(component->dev);
         }
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                if (!rtd->codec_dais[i]->active)
-                        pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev);
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
+                if (!codec_dai->active)
+                        pinctrl_pm_select_sleep_state(codec_dai->dev);
         }
         if (!cpu_dai->active)
                 pinctrl_pm_select_sleep_state(cpu_dai->dev);
@@ -801,8 +799,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
                 }
         }
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                codec_dai = rtd->codec_dais[i];
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 if (codec_dai->driver->ops->prepare) {
                         ret = codec_dai->driver->ops->prepare(substream,
                                                               codec_dai);
@@ -834,8 +831,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
         snd_soc_dapm_stream_event(rtd, substream->stream,
                         SND_SOC_DAPM_STREAM_START);
 
-        for (i = 0; i < rtd->num_codecs; i++)
-                snd_soc_dai_digital_mute(rtd->codec_dais[i], 0,
+        for_each_rtd_codec_dai(rtd, i, codec_dai)
+                snd_soc_dai_digital_mute(codec_dai, 0,
                                          substream->stream);
         snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream);
 
@@ -920,6 +917,7 @@ 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;
+        struct snd_soc_dai *codec_dai;
         int i, ret = 0;
 
         mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
@@ -932,8 +930,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
                 }
         }
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 struct snd_pcm_hw_params codec_params;
 
                 /*
@@ -1018,8 +1015,7 @@ interface_err:
         i = rtd->num_codecs;
 
 codec_err:
-        while (--i >= 0) {
-                struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+        for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) {
                 if (codec_dai->driver->ops->hw_free)
                         codec_dai->driver->ops->hw_free(substream, codec_dai);
                 codec_dai->rate = 0;
@@ -1052,8 +1048,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
                 cpu_dai->sample_bits = 0;
         }
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                codec_dai = rtd->codec_dais[i];
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 if (codec_dai->active == 1) {
                         codec_dai->rate = 0;
                         codec_dai->channels = 0;
@@ -1062,10 +1057,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
         }
 
         /* apply codec digital mute */
-        for (i = 0; i < rtd->num_codecs; i++) {
-                if ((playback && rtd->codec_dais[i]->playback_active == 1) ||
-                    (!playback && rtd->codec_dais[i]->capture_active == 1))
-                        snd_soc_dai_digital_mute(rtd->codec_dais[i], 1,
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
+                if ((playback && codec_dai->playback_active == 1) ||
+                    (!playback && codec_dai->capture_active == 1))
+                        snd_soc_dai_digital_mute(codec_dai, 1,
                                                  substream->stream);
         }
 
@@ -1077,8 +1072,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
         soc_pcm_components_hw_free(substream, NULL);
 
         /* now free hw params for the DAIs  */
-        for (i = 0; i < rtd->num_codecs; i++) {
-                codec_dai = rtd->codec_dais[i];
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 if (codec_dai->driver->ops->hw_free)
                         codec_dai->driver->ops->hw_free(substream, codec_dai);
         }
@@ -1099,8 +1093,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
         struct snd_soc_dai *codec_dai;
         int i, ret;
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                codec_dai = rtd->codec_dais[i];
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 if (codec_dai->driver->ops->trigger) {
                         ret = codec_dai->driver->ops->trigger(substream,
                                                               cmd, codec_dai);
@@ -1144,8 +1137,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
         struct snd_soc_dai *codec_dai;
         int i, ret;
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                codec_dai = rtd->codec_dais[i];
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 if (codec_dai->driver->ops->bespoke_trigger) {
                         ret = codec_dai->driver->ops->bespoke_trigger(substream,
                                                                 cmd, codec_dai);
@@ -1199,8 +1191,7 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
         if (cpu_dai->driver->ops->delay)
                 delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
 
-        for (i = 0; i < rtd->num_codecs; i++) {
-                codec_dai = rtd->codec_dais[i];
+        for_each_rtd_codec_dai(rtd, i, codec_dai) {
                 if (codec_dai->driver->ops->delay)
                         codec_delay = max(codec_delay,
                                         codec_dai->driver->ops->delay(substream,
@@ -1220,7 +1211,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe,
         struct snd_soc_dpcm *dpcm;
 
         /* only add new dpcms */
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
                 if (dpcm->be == be && dpcm->fe == fe)
                         return 0;
         }
@@ -1261,7 +1252,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe,
 
         be_substream = snd_soc_dpcm_get_substream(be, stream);
 
-        list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+        for_each_dpcm_fe(be, stream, dpcm) {
                 if (dpcm->fe == fe)
                         continue;
 
@@ -1281,7 +1272,7 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream)
 {
         struct snd_soc_dpcm *dpcm, *d;
 
-        list_for_each_entry_safe(dpcm, d, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be_safe(fe, stream, dpcm, d) {
                 dev_dbg(fe->dev, "ASoC: BE %s disconnect check for %s\n",
                                 stream ? "capture" : "playback",
                                 dpcm->be->dai_link->name);
@@ -1310,12 +1301,13 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
                 struct snd_soc_dapm_widget *widget, int stream)
 {
         struct snd_soc_pcm_runtime *be;
+        struct snd_soc_dai *dai;
         int i;
 
         dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name);
 
         if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-                list_for_each_entry(be, &card->rtd_list, list) {
+                for_each_card_rtds(card, be) {
 
                         if (!be->dai_link->no_pcm)
                                 continue;
@@ -1327,15 +1319,14 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
                         if (be->cpu_dai->playback_widget == widget)
                                 return be;
 
-                        for (i = 0; i < be->num_codecs; i++) {
-                                struct snd_soc_dai *dai = be->codec_dais[i];
+                        for_each_rtd_codec_dai(be, i, dai) {
                                 if (dai->playback_widget == widget)
                                         return be;
                         }
                 }
         } else {
 
-                list_for_each_entry(be, &card->rtd_list, list) {
+                for_each_card_rtds(card, be) {
 
                         if (!be->dai_link->no_pcm)
                                 continue;
@@ -1347,8 +1338,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
                         if (be->cpu_dai->capture_widget == widget)
                                 return be;
 
-                        for (i = 0; i < be->num_codecs; i++) {
-                                struct snd_soc_dai *dai = be->codec_dais[i];
+                        for_each_rtd_codec_dai(be, i, dai) {
                                 if (dai->capture_widget == widget)
                                         return be;
                         }
@@ -1388,32 +1378,31 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget,
 {
         struct snd_soc_card *card = widget->dapm->card;
         struct snd_soc_pcm_runtime *rtd;
+        struct snd_soc_dai *dai;
         int i;
 
         if (dir == SND_SOC_DAPM_DIR_OUT) {
-                list_for_each_entry(rtd, &card->rtd_list, list) {
+                for_each_card_rtds(card, rtd) {
                         if (!rtd->dai_link->no_pcm)
                                 continue;
 
                         if (rtd->cpu_dai->playback_widget == widget)
                                 return true;
 
-                        for (i = 0; i < rtd->num_codecs; ++i) {
-                                struct snd_soc_dai *dai = rtd->codec_dais[i];
+                        for_each_rtd_codec_dai(rtd, i, dai) {
                                 if (dai->playback_widget == widget)
                                         return true;
                         }
                 }
         } else { /* SND_SOC_DAPM_DIR_IN */
-                list_for_each_entry(rtd, &card->rtd_list, list) {
+                for_each_card_rtds(card, rtd) {
                         if (!rtd->dai_link->no_pcm)
                                 continue;
 
                         if (rtd->cpu_dai->capture_widget == widget)
                                 return true;
 
-                        for (i = 0; i < rtd->num_codecs; ++i) {
-                                struct snd_soc_dai *dai = rtd->codec_dais[i];
+                        for_each_rtd_codec_dai(rtd, i, dai) {
                                 if (dai->capture_widget == widget)
                                         return true;
                         }
@@ -1445,10 +1434,11 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
         struct snd_soc_dpcm *dpcm;
         struct snd_soc_dapm_widget_list *list = *list_;
         struct snd_soc_dapm_widget *widget;
+        struct snd_soc_dai *dai;
         int prune = 0;
 
         /* Destroy any old FE <--> BE connections */
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
                 unsigned int i;
 
                 /* is there a valid CPU DAI widget for this BE */
@@ -1459,8 +1449,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
                         continue;
 
                 /* is there a valid CODEC DAI widget for this BE */
-                for (i = 0; i < dpcm->be->num_codecs; i++) {
-                        struct snd_soc_dai *dai = dpcm->be->codec_dais[i];
+                for_each_rtd_codec_dai(dpcm->be, i, dai) {
                         widget = dai_get_widget(dai, stream);
 
                         /* prune the BE if it's no longer in our active list */
@@ -1555,7 +1544,7 @@ void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream)
 {
         struct snd_soc_dpcm *dpcm;
 
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+        for_each_dpcm_be(fe, stream, dpcm)
                 dpcm->be->dpcm[stream].runtime_update =
                                                 SND_SOC_DPCM_UPDATE_NO;
 }
@@ -1566,7 +1555,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe,
         struct snd_soc_dpcm *dpcm;
 
         /* disable any enabled and non active backends */
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
 
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 struct snd_pcm_substream *be_substream =
@@ -1595,7 +1584,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream)
         int err, count = 0;
 
         /* only startup BE DAIs that are either sinks or sources to this FE DAI */
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
 
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 struct snd_pcm_substream *be_substream =
@@ -1649,7 +1638,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream)
 
 unwind:
         /* disable any enabled and non active backends */
-        list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be_rollback(fe, stream, dpcm) {
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 struct snd_pcm_substream *be_substream =
                         snd_soc_dpcm_get_substream(be, stream);
@@ -1680,7 +1669,7 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
                                  struct snd_soc_pcm_stream *stream)
 {
         runtime->hw.rate_min = stream->rate_min;
-        runtime->hw.rate_max = stream->rate_max;
+        runtime->hw.rate_max = min_not_zero(stream->rate_max, UINT_MAX);
         runtime->hw.channels_min = stream->channels_min;
         runtime->hw.channels_max = stream->channels_max;
         if (runtime->hw.formats)
@@ -1695,6 +1684,7 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream,
 {
         struct snd_soc_pcm_runtime *fe = substream->private_data;
         struct snd_soc_dpcm *dpcm;
+        struct snd_soc_dai *dai;
         int stream = substream->stream;
 
         if (!fe->dai_link->dpcm_merged_format)
@@ -1705,22 +1695,21 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream,
          * if FE want to use it (= dpcm_merged_format)
          */
 
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 struct snd_soc_dai_driver *codec_dai_drv;
                 struct snd_soc_pcm_stream *codec_stream;
                 int i;
 
-                for (i = 0; i < be->num_codecs; i++) {
+                for_each_rtd_codec_dai(be, i, dai) {
                         /*
                          * 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))
+                        if (!snd_soc_dai_stream_valid(dai, stream))
                                 continue;
 
-                        codec_dai_drv = be->codec_dais[i]->driver;
+                        codec_dai_drv = dai->driver;
                         if (stream == SNDRV_PCM_STREAM_PLAYBACK)
                                 codec_stream = &codec_dai_drv->playback;
                         else
@@ -1747,7 +1736,7 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream,
          * if FE want to use it (= dpcm_merged_chan)
          */
 
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
                 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;
@@ -1799,12 +1788,13 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream,
          * if FE want to use it (= dpcm_merged_chan)
          */
 
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
                 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;
+                struct snd_soc_dai *dai;
                 int i;
 
                 if (stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -1816,16 +1806,15 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream,
                 *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++) {
+                for_each_rtd_codec_dai(be, i, dai) {
                         /*
                          * 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))
+                        if (!snd_soc_dai_stream_valid(dai, stream))
                                 continue;
 
-                        codec_dai_drv = be->codec_dais[i]->driver;
+                        codec_dai_drv = dai->driver;
                         if (stream == SNDRV_PCM_STREAM_PLAYBACK)
                                 codec_stream = &codec_dai_drv->playback;
                         else
@@ -1902,11 +1891,12 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream,
         }
 
         /* apply symmetry for BE */
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 struct snd_pcm_substream *be_substream =
                         snd_soc_dpcm_get_substream(be, stream);
                 struct snd_soc_pcm_runtime *rtd = be_substream->private_data;
+                struct snd_soc_dai *codec_dai;
                 int i;
 
                 if (rtd->dai_link->be_hw_params_fixup)
@@ -1923,10 +1913,10 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream,
                                 return err;
                 }
 
-                for (i = 0; i < rtd->num_codecs; i++) {
-                        if (rtd->codec_dais[i]->active) {
+                for_each_rtd_codec_dai(rtd, i, codec_dai) {
+                        if (codec_dai->active) {
                                 err = soc_pcm_apply_symmetry(fe_substream,
-                                                             rtd->codec_dais[i]);
+                                                             codec_dai);
                                 if (err < 0)
                                         return err;
                         }
@@ -1986,7 +1976,7 @@ int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream)
         struct snd_soc_dpcm *dpcm;
 
         /* only shutdown BEs that are either sinks or sources to this FE DAI */
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
 
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 struct snd_pcm_substream *be_substream =
@@ -2050,7 +2040,7 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
 
         /* only hw_params backends that are either sinks or sources
          * to this frontend DAI */
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
 
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 struct snd_pcm_substream *be_substream =
@@ -2119,7 +2109,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream)
         struct snd_soc_dpcm *dpcm;
         int ret;
 
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
 
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 struct snd_pcm_substream *be_substream =
@@ -2170,7 +2160,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream)
 
 unwind:
         /* disable any enabled and non active backends */
-        list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be_rollback(fe, stream, dpcm) {
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 struct snd_pcm_substream *be_substream =
                         snd_soc_dpcm_get_substream(be, stream);
@@ -2250,7 +2240,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
         struct snd_soc_dpcm *dpcm;
         int ret = 0;
 
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
 
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 struct snd_pcm_substream *be_substream =
@@ -2436,7 +2426,7 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream)
         struct snd_soc_dpcm *dpcm;
         int ret = 0;
 
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
 
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 struct snd_pcm_substream *be_substream =
@@ -2646,7 +2636,7 @@ close:
         dpcm_be_dai_shutdown(fe, stream);
 disconnect:
         /* disconnect any non started BEs */
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
                                 dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
@@ -2771,14 +2761,14 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
 
         mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
         /* shutdown all old paths first */
-        list_for_each_entry(fe, &card->rtd_list, list) {
+        for_each_card_rtds(card, fe) {
                 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) {
+        for_each_card_rtds(card, fe) {
                 ret = soc_dpcm_fe_runtime_update(fe, 1);
                 if (ret)
                         goto out;
@@ -2791,10 +2781,9 @@ out:
 int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
 {
         struct snd_soc_dpcm *dpcm;
-        struct list_head *clients =
-                &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients;
+        struct snd_soc_dai *dai;
 
-        list_for_each_entry(dpcm, clients, list_be) {
+        for_each_dpcm_be(fe, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
 
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 int i;
@@ -2802,8 +2791,7 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
                 if (be->dai_link->ignore_suspend)
                         continue;
 
-                for (i = 0; i < be->num_codecs; i++) {
-                        struct snd_soc_dai *dai = be->codec_dais[i];
+                for_each_rtd_codec_dai(be, i, dai) {
                         struct snd_soc_dai_driver *drv = dai->driver;
 
                         dev_dbg(be->dev, "ASoC: BE digital mute %s\n",
@@ -2844,7 +2832,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
         ret = dpcm_fe_dai_startup(fe_substream);
         if (ret < 0) {
                 /* clean up all links */
-                list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+                for_each_dpcm_be(fe, stream, dpcm)
                         dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
 
                 dpcm_be_disconnect(fe, stream);
@@ -2867,7 +2855,7 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
         ret = dpcm_fe_dai_shutdown(fe_substream);
 
         /* mark FE's links ready to prune */
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+        for_each_dpcm_be(fe, stream, dpcm)
                 dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
 
         dpcm_be_disconnect(fe, stream);
@@ -3041,8 +3029,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
                 playback = rtd->dai_link->dpcm_playback;
                 capture = rtd->dai_link->dpcm_capture;
         } else {
-                for (i = 0; i < rtd->num_codecs; i++) {
-                        codec_dai = rtd->codec_dais[i];
+                for_each_rtd_codec_dai(rtd, i, codec_dai) {
                         if (codec_dai->driver->playback.channels_min)
                                 playback = 1;
                         if (codec_dai->driver->capture.channels_min)
@@ -3230,7 +3217,7 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
         struct snd_soc_dpcm *dpcm;
         int state;
 
-        list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+        for_each_dpcm_fe(be, stream, dpcm) {
 
                 if (dpcm->fe == fe)
                         continue;
@@ -3257,7 +3244,7 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
         struct snd_soc_dpcm *dpcm;
         int state;
 
-        list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+        for_each_dpcm_fe(be, stream, dpcm) {
 
                 if (dpcm->fe == fe)
                         continue;
@@ -3337,7 +3324,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe,
                 goto out;
         }
 
-        list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+        for_each_dpcm_be(fe, stream, dpcm) {
                 struct snd_soc_pcm_runtime *be = dpcm->be;
                 params = &dpcm->hw_params;
 
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 66e77e020745..045ef136903d 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -993,7 +993,7 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count,
                                 kfree(se);
                                 continue;
                         }
-                        /* fall through and create texts */
+                        /* fall through */
                 case SND_SOC_TPLG_CTL_ENUM:
                 case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
                 case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
@@ -1310,7 +1310,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create(
                                         ec->hdr.name);
                                 goto err_se;
                         }
-                        /* fall through to create texts */
+                        /* fall through */
                 case SND_SOC_TPLG_CTL_ENUM:
                 case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
                 case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
@@ -1565,17 +1565,6 @@ widget:
                 widget = snd_soc_dapm_new_control_unlocked(dapm, &template);
         if (IS_ERR(widget)) {
                 ret = PTR_ERR(widget);
-                /* Do not nag about probe deferrals */
-                if (ret != -EPROBE_DEFER)
-                        dev_err(tplg->dev,
-                                "ASoC: failed to create widget %s controls (%d)\n",
-                                w->name, ret);
-                goto hdr_err;
-        }
-        if (widget == NULL) {
-                dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n",
-                        w->name);
-                ret = -ENOMEM;
                 goto hdr_err;
         }
 
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index e0c93496c0cd..e3b9dd634c6d 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -273,13 +273,13 @@ static int dummy_dma_open(struct snd_pcm_substream *substream)
         return 0;
 }
 
-static const struct snd_pcm_ops dummy_dma_ops = {
+static const struct snd_pcm_ops snd_dummy_dma_ops = {
         .open           = dummy_dma_open,
         .ioctl          = snd_pcm_lib_ioctl,
 };
 
 static const struct snd_soc_component_driver dummy_platform = {
-        .ops = &dummy_dma_ops,
+        .ops = &snd_dummy_dma_ops,
 };
 
 static const struct snd_soc_component_driver dummy_codec = {
diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig
index 9b2681397dba..c66ffa72057e 100644
--- a/sound/soc/stm/Kconfig
+++ b/sound/soc/stm/Kconfig
@@ -3,6 +3,7 @@ menu "STMicroelectronics STM32 SOC audio support"
 config SND_SOC_STM32_SAI
         tristate "STM32 SAI interface (Serial Audio Interface) support"
         depends on (ARCH_STM32 && OF) || COMPILE_TEST
+        depends on COMMON_CLK
         depends on SND_SOC
         select SND_SOC_GENERIC_DMAENGINE_PCM
         select REGMAP_MMIO
diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c
index f22654253c43..d597eba61992 100644
--- a/sound/soc/stm/stm32_sai.c
+++ b/sound/soc/stm/stm32_sai.c
@@ -104,7 +104,7 @@ static int stm32_sai_set_sync(struct stm32_sai_data *sai_client,
 
         if (!pdev) {
                 dev_err(&sai_client->pdev->dev,
-                        "Device not found for node %s\n", np_provider->name);
+                        "Device not found for node %pOFn\n", np_provider);
                 return -ENODEV;
         }
 
diff --git a/sound/soc/stm/stm32_sai.h b/sound/soc/stm/stm32_sai.h
index f25422174909..08de899c766b 100644
--- a/sound/soc/stm/stm32_sai.h
+++ b/sound/soc/stm/stm32_sai.h
@@ -91,6 +91,9 @@
 #define SAI_XCR1_OSR_SHIFT      26
 #define SAI_XCR1_OSR            BIT(SAI_XCR1_OSR_SHIFT)
 
+#define SAI_XCR1_MCKEN_SHIFT    27
+#define SAI_XCR1_MCKEN          BIT(SAI_XCR1_MCKEN_SHIFT)
+
 /******************* Bit definition for SAI_XCR2 register *******************/
 #define SAI_XCR2_FTH_SHIFT      0
 #define SAI_XCR2_FTH_MASK       GENMASK(2, SAI_XCR2_FTH_SHIFT)
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index 06fba9650ac4..ea05cc91aa05 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -17,6 +17,7 @@
  */
 
 #include <linux/clk.h>
+#include <linux/clk-provider.h>
 #include <linux/kernel.h>
 #include <linux/module.h>
 #include <linux/of_irq.h>
@@ -68,6 +69,8 @@
 #define SAI_IEC60958_BLOCK_FRAMES       192
 #define SAI_IEC60958_STATUS_BYTES       24
 
+#define SAI_MCLK_NAME_LEN               32
+
 /**
  * struct stm32_sai_sub_data - private data of SAI sub block (block A or B)
  * @pdev: device data pointer
@@ -80,6 +83,7 @@
  * @pdata: SAI block parent data pointer
  * @np_sync_provider: synchronization provider node
  * @sai_ck: kernel clock feeding the SAI clock generator
+ * @sai_mclk: master clock from SAI mclk provider
  * @phys_addr: SAI registers physical base address
  * @mclk_rate: SAI block master clock frequency (Hz). set at init
  * @id: SAI sub block id corresponding to sub-block A or B
@@ -110,6 +114,7 @@ struct stm32_sai_sub_data {
         struct stm32_sai_data *pdata;
         struct device_node *np_sync_provider;
         struct clk *sai_ck;
+        struct clk *sai_mclk;
         dma_addr_t phys_addr;
         unsigned int mclk_rate;
         unsigned int id;
@@ -251,6 +256,176 @@ static const struct snd_kcontrol_new iec958_ctls = {
         .put = snd_pcm_iec958_put,
 };
 
+struct stm32_sai_mclk_data {
+        struct clk_hw hw;
+        unsigned long freq;
+        struct stm32_sai_sub_data *sai_data;
+};
+
+#define to_mclk_data(_hw) container_of(_hw, struct stm32_sai_mclk_data, hw)
+#define STM32_SAI_MAX_CLKS 1
+
+static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai,
+                                 unsigned long input_rate,
+                                 unsigned long output_rate)
+{
+        int version = sai->pdata->conf->version;
+        int div;
+
+        div = DIV_ROUND_CLOSEST(input_rate, output_rate);
+        if (div > SAI_XCR1_MCKDIV_MAX(version)) {
+                dev_err(&sai->pdev->dev, "Divider %d out of range\n", div);
+                return -EINVAL;
+        }
+        dev_dbg(&sai->pdev->dev, "SAI divider %d\n", div);
+
+        if (input_rate % div)
+                dev_dbg(&sai->pdev->dev,
+                        "Rate not accurate. requested (%ld), actual (%ld)\n",
+                        output_rate, input_rate / div);
+
+        return div;
+}
+
+static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai,
+                                 unsigned int div)
+{
+        int version = sai->pdata->conf->version;
+        int ret, cr1, mask;
+
+        if (div > SAI_XCR1_MCKDIV_MAX(version)) {
+                dev_err(&sai->pdev->dev, "Divider %d out of range\n", div);
+                return -EINVAL;
+        }
+
+        mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version));
+        cr1 = SAI_XCR1_MCKDIV_SET(div);
+        ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1);
+        if (ret < 0)
+                dev_err(&sai->pdev->dev, "Failed to update CR1 register\n");
+
+        return ret;
+}
+
+static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate,
+                                      unsigned long *prate)
+{
+        struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+        struct stm32_sai_sub_data *sai = mclk->sai_data;
+        int div;
+
+        div = stm32_sai_get_clk_div(sai, *prate, rate);
+        if (div < 0)
+                return div;
+
+        mclk->freq = *prate / div;
+
+        return mclk->freq;
+}
+
+static unsigned long stm32_sai_mclk_recalc_rate(struct clk_hw *hw,
+                                                unsigned long parent_rate)
+{
+        struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+
+        return mclk->freq;
+}
+
+static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate,
+                                   unsigned long parent_rate)
+{
+        struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+        struct stm32_sai_sub_data *sai = mclk->sai_data;
+        unsigned int div;
+        int ret;
+
+        div = stm32_sai_get_clk_div(sai, parent_rate, rate);
+        if (div < 0)
+                return div;
+
+        ret = stm32_sai_set_clk_div(sai, div);
+        if (ret)
+                return ret;
+
+        mclk->freq = rate;
+
+        return 0;
+}
+
+static int stm32_sai_mclk_enable(struct clk_hw *hw)
+{
+        struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+        struct stm32_sai_sub_data *sai = mclk->sai_data;
+
+        dev_dbg(&sai->pdev->dev, "Enable master clock\n");
+
+        return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
+                                  SAI_XCR1_MCKEN, SAI_XCR1_MCKEN);
+}
+
+static void stm32_sai_mclk_disable(struct clk_hw *hw)
+{
+        struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+        struct stm32_sai_sub_data *sai = mclk->sai_data;
+
+        dev_dbg(&sai->pdev->dev, "Disable master clock\n");
+
+        regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_MCKEN, 0);
+}
+
+static const struct clk_ops mclk_ops = {
+        .enable = stm32_sai_mclk_enable,
+        .disable = stm32_sai_mclk_disable,
+        .recalc_rate = stm32_sai_mclk_recalc_rate,
+        .round_rate = stm32_sai_mclk_round_rate,
+        .set_rate = stm32_sai_mclk_set_rate,
+};
+
+static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai)
+{
+        struct clk_hw *hw;
+        struct stm32_sai_mclk_data *mclk;
+        struct device *dev = &sai->pdev->dev;
+        const char *pname = __clk_get_name(sai->sai_ck);
+        char *mclk_name, *p, *s = (char *)pname;
+        int ret, i = 0;
+
+        mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL);
+        if (!mclk)
+                return -ENOMEM;
+
+        mclk_name = devm_kcalloc(dev, sizeof(char),
+                                 SAI_MCLK_NAME_LEN, GFP_KERNEL);
+        if (!mclk_name)
+                return -ENOMEM;
+
+        /*
+         * Forge mclk clock name from parent clock name and suffix.
+         * String after "_" char is stripped in parent name.
+         */
+        p = mclk_name;
+        while (*s && *s != '_' && (i < (SAI_MCLK_NAME_LEN - 7))) {
+                *p++ = *s++;
+                i++;
+        }
+        STM_SAI_IS_SUB_A(sai) ? strcat(p, "a_mclk") : strcat(p, "b_mclk");
+
+        mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0);
+        mclk->sai_data = sai;
+        hw = &mclk->hw;
+
+        dev_dbg(dev, "Register master clock %s\n", mclk_name);
+        ret = devm_clk_hw_register(&sai->pdev->dev, hw);
+        if (ret) {
+                dev_err(dev, "mclk register returned %d\n", ret);
+                return ret;
+        }
+        sai->sai_mclk = hw->clk;
+
+        /* register mclk provider */
+        return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw);
+}
+
 static irqreturn_t stm32_sai_isr(int irq, void *devid)
 {
         struct stm32_sai_sub_data *sai = (struct stm32_sai_sub_data *)devid;
@@ -312,15 +487,25 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai,
         struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai);
         int ret;
 
-        if ((dir == SND_SOC_CLOCK_OUT) && sai->master) {
+        if (dir == SND_SOC_CLOCK_OUT) {
                 ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
                                          SAI_XCR1_NODIV,
                                          (unsigned int)~SAI_XCR1_NODIV);
                 if (ret < 0)
                         return ret;
 
-                sai->mclk_rate = freq;
                 dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq);
+                sai->mclk_rate = freq;
+
+                if (sai->sai_mclk) {
+                        ret = clk_set_rate_exclusive(sai->sai_mclk,
+                                                     sai->mclk_rate);
+                        if (ret) {
+                                dev_err(cpu_dai->dev,
+                                        "Could not set mclk rate\n");
+                                return ret;
+                        }
+                }
         }
 
         return 0;
@@ -715,15 +900,9 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
 {
         struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai);
         int cr1, mask, div = 0;
-        int sai_clk_rate, mclk_ratio, den, ret;
-        int version = sai->pdata->conf->version;
+        int sai_clk_rate, mclk_ratio, den;
         unsigned int rate = params_rate(params);
 
-        if (!sai->mclk_rate) {
-                dev_err(cpu_dai->dev, "Mclk rate is null\n");
-                return -EINVAL;
-        }
-
         if (!(rate % 11025))
                 clk_set_parent(sai->sai_ck, sai->pdata->clk_x11k);
         else
@@ -731,14 +910,22 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
         sai_clk_rate = clk_get_rate(sai->sai_ck);
 
         if (STM_SAI_IS_F4(sai->pdata)) {
-                /*
-                 * mclk_rate = 256 * fs
-                 * MCKDIV = 0 if sai_ck < 3/2 * mclk_rate
-                 * MCKDIV = sai_ck / (2 * mclk_rate) otherwise
+                /* mclk on (NODIV=0)
+                 *   mclk_rate = 256 * fs
+                 *   MCKDIV = 0 if sai_ck < 3/2 * mclk_rate
+                 *   MCKDIV = sai_ck / (2 * mclk_rate) otherwise
+                 * mclk off (NODIV=1)
+                 *   MCKDIV ignored. sck = sai_ck
                  */
-                if (2 * sai_clk_rate >= 3 * sai->mclk_rate)
-                        div = DIV_ROUND_CLOSEST(sai_clk_rate,
-                                                2 * sai->mclk_rate);
+                if (!sai->mclk_rate)
+                        return 0;
+
+                if (2 * sai_clk_rate >= 3 * sai->mclk_rate) {
+                        div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+                                                    2 * sai->mclk_rate);
+                        if (div < 0)
+                                return div;
+                }
         } else {
                 /*
                  * TDM mode :
@@ -750,8 +937,10 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
                  * Note: NOMCK/NODIV correspond to same bit.
                  */
                 if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) {
-                        div = DIV_ROUND_CLOSEST(sai_clk_rate,
-                                                (params_rate(params) * 128));
+                        div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+                                                    rate * 128);
+                        if (div < 0)
+                                return div;
                 } else {
                         if (sai->mclk_rate) {
                                 mclk_ratio = sai->mclk_rate / rate;
@@ -764,31 +953,22 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
                                                 mclk_ratio);
                                         return -EINVAL;
                                 }
-                                div = DIV_ROUND_CLOSEST(sai_clk_rate,
-                                                        sai->mclk_rate);
+                                div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+                                                            sai->mclk_rate);
+                                if (div < 0)
+                                        return div;
                         } else {
                                 /* mclk-fs not set, master clock not active */
                                 den = sai->fs_length * params_rate(params);
-                                div = DIV_ROUND_CLOSEST(sai_clk_rate, den);
+                                div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+                                                            den);
+                                if (div < 0)
+                                        return div;
                         }
                 }
         }
 
-        if (div > SAI_XCR1_MCKDIV_MAX(version)) {
-                dev_err(cpu_dai->dev, "Divider %d out of range\n", div);
-                return -EINVAL;
-        }
-        dev_dbg(cpu_dai->dev, "SAI clock %d, divider %d\n", sai_clk_rate, div);
-
-        mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version));
-        cr1 = SAI_XCR1_MCKDIV_SET(div);
-        ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1);
-        if (ret < 0) {
-                dev_err(cpu_dai->dev, "Failed to update CR1 register\n");
-                return ret;
-        }
-
-        return 0;
+        return stm32_sai_set_clk_div(sai, div);
 }
 
 static int stm32_sai_hw_params(struct snd_pcm_substream *substream,
@@ -881,6 +1061,9 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream,
                            SAI_XCR1_NODIV);
 
         clk_disable_unprepare(sai->sai_ck);
+
+        clk_rate_exclusive_put(sai->sai_mclk);
+
         sai->substream = NULL;
 }
 
@@ -903,6 +1086,8 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai)
         struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev);
         int cr1 = 0, cr1_mask;
 
+        sai->cpu_dai = cpu_dai;
+
         sai->dma_params.addr = (dma_addr_t)(sai->phys_addr + STM_SAI_DR_REGX);
         /*
          * DMA supports 4, 8 or 16 burst sizes. Burst size 4 is the best choice,
@@ -1124,16 +1309,15 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev,
         sai->sync = SAI_SYNC_NONE;
         if (args.np) {
                 if (args.np == np) {
-                        dev_err(&pdev->dev, "%s sync own reference\n",
-                                np->name);
+                        dev_err(&pdev->dev, "%pOFn sync own reference\n", np);
                         of_node_put(args.np);
                         return -EINVAL;
                 }
 
                 sai->np_sync_provider  = of_get_parent(args.np);
                 if (!sai->np_sync_provider) {
-                        dev_err(&pdev->dev, "%s parent node not found\n",
-                                np->name);
+                        dev_err(&pdev->dev, "%pOFn parent node not found\n",
+                                np);
                         of_node_put(args.np);
                         return -ENODEV;
                 }
@@ -1182,6 +1366,23 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev,
                 return PTR_ERR(sai->sai_ck);
         }
 
+        if (STM_SAI_IS_F4(sai->pdata))
+                return 0;
+
+        /* Register mclk provider if requested */
+        if (of_find_property(np, "#clock-cells", NULL)) {
+                ret = stm32_sai_add_mclk_provider(sai);
+                if (ret < 0)
+                        return ret;
+        } else {
+                sai->sai_mclk = devm_clk_get(&pdev->dev, "MCLK");
+                if (IS_ERR(sai->sai_mclk)) {
+                        if (PTR_ERR(sai->sai_mclk) != -ENOENT)
+                                return PTR_ERR(sai->sai_mclk);
+                        sai->sai_mclk = NULL;
+                }
+        }
+
         return 0;
 }
 
diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig
index 22408bc2d6ec..66aad0d3f9c7 100644
--- a/sound/soc/sunxi/Kconfig
+++ b/sound/soc/sunxi/Kconfig
@@ -12,7 +12,7 @@ config SND_SUN4I_CODEC
 config SND_SUN8I_CODEC
         tristate "Allwinner SUN8I audio codec"
         depends on OF
-        depends on MACH_SUN8I || COMPILE_TEST
+        depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST
         select REGMAP_MMIO
         help
           This option enables the digital part of the internal audio codec for
@@ -23,11 +23,19 @@ config SND_SUN8I_CODEC
 config SND_SUN8I_CODEC_ANALOG
         tristate "Allwinner sun8i Codec Analog Controls Support"
         depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST
-        select REGMAP
+        select SND_SUN8I_ADDA_PR_REGMAP
         help
           Say Y or M if you want to add support for the analog controls for
           the codec embedded in newer Allwinner SoCs.
 
+config SND_SUN50I_CODEC_ANALOG
+        tristate "Allwinner sun50i Codec Analog Controls Support"
+        depends on (ARM64 && ARCH_SUNXI) || COMPILE_TEST
+        select SND_SUNXI_ADDA_PR_REGMAP
+        help
+          Say Y or M if you want to add support for the analog controls for
+          the codec embedded in Allwinner A64 SoC.
+
 config SND_SUN4I_I2S
         tristate "Allwinner A10 I2S Support"
         select SND_SOC_GENERIC_DMAENGINE_PCM
@@ -45,4 +53,9 @@ config SND_SUN4I_SPDIF
         help
           Say Y or M to add support for the S/PDIF audio block in the Allwinner
           A10 and affiliated SoCs.
+
+config SND_SUN8I_ADDA_PR_REGMAP
+        tristate
+        select REGMAP
+
 endmenu
diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile
index 4a9ef67386ca..a86be340a076 100644
--- a/sound/soc/sunxi/Makefile
+++ b/sound/soc/sunxi/Makefile
@@ -3,4 +3,6 @@ obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o
 obj-$(CONFIG_SND_SUN4I_I2S) += sun4i-i2s.o
 obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o
 obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o
+obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o
 obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o
+obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index a4aa931ebfae..d5ec1a20499d 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -644,40 +644,6 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
         return 0;
 }
 
-static int sun4i_i2s_startup(struct snd_pcm_substream *substream,
-                             struct snd_soc_dai *dai)
-{
-        struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-
-        /* Enable the whole hardware block */
-        regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-                           SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN);
-
-        /* Enable the first output line */
-        regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-                           SUN4I_I2S_CTRL_SDO_EN_MASK,
-                           SUN4I_I2S_CTRL_SDO_EN(0));
-
-
-        return clk_prepare_enable(i2s->mod_clk);
-}
-
-static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream,
-                               struct snd_soc_dai *dai)
-{
-        struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-
-        clk_disable_unprepare(i2s->mod_clk);
-
-        /* Disable our output lines */
-        regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-                           SUN4I_I2S_CTRL_SDO_EN_MASK, 0);
-
-        /* Disable the whole hardware block */
-        regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-                           SUN4I_I2S_CTRL_GL_EN, 0);
-}
-
 static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id,
                                 unsigned int freq, int dir)
 {
@@ -695,8 +661,6 @@ static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = {
         .hw_params      = sun4i_i2s_hw_params,
         .set_fmt        = sun4i_i2s_set_fmt,
         .set_sysclk     = sun4i_i2s_set_sysclk,
-        .shutdown       = sun4i_i2s_shutdown,
-        .startup        = sun4i_i2s_startup,
         .trigger        = sun4i_i2s_trigger,
 };
 
@@ -869,6 +833,21 @@ static int sun4i_i2s_runtime_resume(struct device *dev)
                 goto err_disable_clk;
         }
 
+        /* Enable the whole hardware block */
+        regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+                           SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN);
+
+        /* Enable the first output line */
+        regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+                           SUN4I_I2S_CTRL_SDO_EN_MASK,
+                           SUN4I_I2S_CTRL_SDO_EN(0));
+
+        ret = clk_prepare_enable(i2s->mod_clk);
+        if (ret) {
+                dev_err(dev, "Failed to enable module clock\n");
+                goto err_disable_clk;
+        }
+
         return 0;
 
 err_disable_clk:
@@ -880,6 +859,16 @@ static int sun4i_i2s_runtime_suspend(struct device *dev)
 {
         struct sun4i_i2s *i2s = dev_get_drvdata(dev);
 
+        clk_disable_unprepare(i2s->mod_clk);
+
+        /* Disable our output lines */
+        regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+                           SUN4I_I2S_CTRL_SDO_EN_MASK, 0);
+
+        /* Disable the whole hardware block */
+        regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+                           SUN4I_I2S_CTRL_GL_EN, 0);
+
         regcache_cache_only(i2s->regmap, true);
 
         clk_disable_unprepare(i2s->bus_clk);
@@ -961,6 +950,23 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
         .field_rxchansel        = REG_FIELD(SUN8I_I2S_RX_CHAN_SEL_REG, 0, 2),
 };
 
+static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
+        .has_reset              = true,
+        .reg_offset_txdata      = SUN8I_I2S_FIFO_TX_REG,
+        .sun4i_i2s_regmap       = &sun4i_i2s_regmap_config,
+        .has_slave_select_bit   = true,
+        .field_clkdiv_mclk_en   = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
+        .field_fmt_wss          = REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
+        .field_fmt_sr           = REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
+        .field_fmt_bclk         = REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
+        .field_fmt_lrclk        = REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+        .field_fmt_mode         = REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
+        .field_txchanmap        = REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
+        .field_rxchanmap        = REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
+        .field_txchansel        = REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
+        .field_rxchansel        = REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
+};
+
 static int sun4i_i2s_init_regmap_fields(struct device *dev,
                                         struct sun4i_i2s *i2s)
 {
@@ -1169,6 +1175,10 @@ static const struct of_device_id sun4i_i2s_match[] = {
                 .compatible = "allwinner,sun8i-h3-i2s",
                 .data = &sun8i_h3_i2s_quirks,
         },
+        {
+                .compatible = "allwinner,sun50i-a64-codec-i2s",
+                .data = &sun50i_a64_codec_i2s_quirks,
+        },
         {}
 };
 MODULE_DEVICE_TABLE(of, sun4i_i2s_match);
diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c
new file mode 100644
index 000000000000..8f5f999df631
--- /dev/null
+++ b/sound/soc/sunxi/sun50i-codec-analog.c
@@ -0,0 +1,444 @@
+// SPDX-License-Identifier: GPL-2.0+
+/*
+ * This driver supports the analog controls for the internal codec
+ * found in Allwinner's A64 SoC.
+ *
+ * Copyright (C) 2016 Chen-Yu Tsai <wens@csie.org>
+ * Copyright (C) 2017 Marcus Cooper <codekipper@gmail.com>
+ * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com>
+ *
+ * Based on sun8i-codec-analog.c
+ *
+ */
+
+#include <linux/io.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "sun8i-adda-pr-regmap.h"
+
+/* Codec analog control register offsets and bit fields */
+#define SUN50I_ADDA_HP_CTRL             0x00
+#define SUN50I_ADDA_HP_CTRL_PA_CLK_GATE         7
+#define SUN50I_ADDA_HP_CTRL_HPPA_EN             6
+#define SUN50I_ADDA_HP_CTRL_HPVOL               0
+
+#define SUN50I_ADDA_OL_MIX_CTRL         0x01
+#define SUN50I_ADDA_OL_MIX_CTRL_MIC1            6
+#define SUN50I_ADDA_OL_MIX_CTRL_MIC2            5
+#define SUN50I_ADDA_OL_MIX_CTRL_PHONE           4
+#define SUN50I_ADDA_OL_MIX_CTRL_PHONEN          3
+#define SUN50I_ADDA_OL_MIX_CTRL_LINEINL         2
+#define SUN50I_ADDA_OL_MIX_CTRL_DACL            1
+#define SUN50I_ADDA_OL_MIX_CTRL_DACR            0
+
+#define SUN50I_ADDA_OR_MIX_CTRL         0x02
+#define SUN50I_ADDA_OR_MIX_CTRL_MIC1            6
+#define SUN50I_ADDA_OR_MIX_CTRL_MIC2            5
+#define SUN50I_ADDA_OR_MIX_CTRL_PHONE           4
+#define SUN50I_ADDA_OR_MIX_CTRL_PHONEP          3
+#define SUN50I_ADDA_OR_MIX_CTRL_LINEINR         2
+#define SUN50I_ADDA_OR_MIX_CTRL_DACR            1
+#define SUN50I_ADDA_OR_MIX_CTRL_DACL            0
+
+#define SUN50I_ADDA_LINEOUT_CTRL0       0x05
+#define SUN50I_ADDA_LINEOUT_CTRL0_LEN           7
+#define SUN50I_ADDA_LINEOUT_CTRL0_REN           6
+#define SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL      5
+#define SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL      4
+
+#define SUN50I_ADDA_LINEOUT_CTRL1       0x06
+#define SUN50I_ADDA_LINEOUT_CTRL1_VOL           0
+
+#define SUN50I_ADDA_MIC1_CTRL           0x07
+#define SUN50I_ADDA_MIC1_CTRL_MIC1G             4
+#define SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN         3
+#define SUN50I_ADDA_MIC1_CTRL_MIC1BOOST         0
+
+#define SUN50I_ADDA_MIC2_CTRL           0x08
+#define SUN50I_ADDA_MIC2_CTRL_MIC2G             4
+#define SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN         3
+#define SUN50I_ADDA_MIC2_CTRL_MIC2BOOST         0
+
+#define SUN50I_ADDA_LINEIN_CTRL         0x09
+#define SUN50I_ADDA_LINEIN_CTRL_LINEING         0
+
+#define SUN50I_ADDA_MIX_DAC_CTRL        0x0a
+#define SUN50I_ADDA_MIX_DAC_CTRL_DACAREN        7
+#define SUN50I_ADDA_MIX_DAC_CTRL_DACALEN        6
+#define SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN         5
+#define SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN         4
+#define SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE      3
+#define SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE      2
+#define SUN50I_ADDA_MIX_DAC_CTRL_RHPIS          1
+#define SUN50I_ADDA_MIX_DAC_CTRL_LHPIS          0
+
+#define SUN50I_ADDA_L_ADCMIX_SRC        0x0b
+#define SUN50I_ADDA_L_ADCMIX_SRC_MIC1           6
+#define SUN50I_ADDA_L_ADCMIX_SRC_MIC2           5
+#define SUN50I_ADDA_L_ADCMIX_SRC_PHONE          4
+#define SUN50I_ADDA_L_ADCMIX_SRC_PHONEN         3
+#define SUN50I_ADDA_L_ADCMIX_SRC_LINEINL        2
+#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL         1
+#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR         0
+
+#define SUN50I_ADDA_R_ADCMIX_SRC        0x0c
+#define SUN50I_ADDA_R_ADCMIX_SRC_MIC1           6
+#define SUN50I_ADDA_R_ADCMIX_SRC_MIC2           5
+#define SUN50I_ADDA_R_ADCMIX_SRC_PHONE          4
+#define SUN50I_ADDA_R_ADCMIX_SRC_PHONEP         3
+#define SUN50I_ADDA_R_ADCMIX_SRC_LINEINR        2
+#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXR          1
+#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXL          0
+
+#define SUN50I_ADDA_ADC_CTRL            0x0d
+#define SUN50I_ADDA_ADC_CTRL_ADCREN             7
+#define SUN50I_ADDA_ADC_CTRL_ADCLEN             6
+#define SUN50I_ADDA_ADC_CTRL_ADCG               0
+
+#define SUN50I_ADDA_HS_MBIAS_CTRL       0x0e
+#define SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN    7
+
+#define SUN50I_ADDA_JACK_MIC_CTRL       0x1d
+#define SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN    5
+
+/* mixer controls */
+static const struct snd_kcontrol_new sun50i_a64_codec_mixer_controls[] = {
+        SOC_DAPM_DOUBLE_R("DAC Playback Switch",
+                          SUN50I_ADDA_OL_MIX_CTRL,
+                          SUN50I_ADDA_OR_MIX_CTRL,
+                          SUN50I_ADDA_OL_MIX_CTRL_DACL, 1, 0),
+        SOC_DAPM_DOUBLE_R("DAC Reversed Playback Switch",
+                          SUN50I_ADDA_OL_MIX_CTRL,
+                          SUN50I_ADDA_OR_MIX_CTRL,
+                          SUN50I_ADDA_OL_MIX_CTRL_DACR, 1, 0),
+        SOC_DAPM_DOUBLE_R("Line In Playback Switch",
+                          SUN50I_ADDA_OL_MIX_CTRL,
+                          SUN50I_ADDA_OR_MIX_CTRL,
+                          SUN50I_ADDA_OL_MIX_CTRL_LINEINL, 1, 0),
+        SOC_DAPM_DOUBLE_R("Mic1 Playback Switch",
+                          SUN50I_ADDA_OL_MIX_CTRL,
+                          SUN50I_ADDA_OR_MIX_CTRL,
+                          SUN50I_ADDA_OL_MIX_CTRL_MIC1, 1, 0),
+        SOC_DAPM_DOUBLE_R("Mic2 Playback Switch",
+                          SUN50I_ADDA_OL_MIX_CTRL,
+                          SUN50I_ADDA_OR_MIX_CTRL,
+                          SUN50I_ADDA_OL_MIX_CTRL_MIC2, 1, 0),
+};
+
+/* ADC mixer controls */
+static const struct snd_kcontrol_new sun50i_codec_adc_mixer_controls[] = {
+        SOC_DAPM_DOUBLE_R("Mixer Capture Switch",
+                          SUN50I_ADDA_L_ADCMIX_SRC,
+                          SUN50I_ADDA_R_ADCMIX_SRC,
+                          SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL, 1, 0),
+        SOC_DAPM_DOUBLE_R("Mixer Reversed Capture Switch",
+                          SUN50I_ADDA_L_ADCMIX_SRC,
+                          SUN50I_ADDA_R_ADCMIX_SRC,
+                          SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR, 1, 0),
+        SOC_DAPM_DOUBLE_R("Line In Capture Switch",
+                          SUN50I_ADDA_L_ADCMIX_SRC,
+                          SUN50I_ADDA_R_ADCMIX_SRC,
+                          SUN50I_ADDA_L_ADCMIX_SRC_LINEINL, 1, 0),
+        SOC_DAPM_DOUBLE_R("Mic1 Capture Switch",
+                          SUN50I_ADDA_L_ADCMIX_SRC,
+                          SUN50I_ADDA_R_ADCMIX_SRC,
+                          SUN50I_ADDA_L_ADCMIX_SRC_MIC1, 1, 0),
+        SOC_DAPM_DOUBLE_R("Mic2 Capture Switch",
+                          SUN50I_ADDA_L_ADCMIX_SRC,
+                          SUN50I_ADDA_R_ADCMIX_SRC,
+                          SUN50I_ADDA_L_ADCMIX_SRC_MIC2, 1, 0),
+};
+
+static const DECLARE_TLV_DB_SCALE(sun50i_codec_out_mixer_pregain_scale,
+                                  -450, 150, 0);
+static const DECLARE_TLV_DB_RANGE(sun50i_codec_mic_gain_scale,
+        0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+        1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0),
+);
+
+static const DECLARE_TLV_DB_SCALE(sun50i_codec_hp_vol_scale, -6300, 100, 1);
+
+static const DECLARE_TLV_DB_RANGE(sun50i_codec_lineout_vol_scale,
+        0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1),
+        2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0),
+);
+
+
+/* volume / mute controls */
+static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = {
+        SOC_SINGLE_TLV("Headphone Playback Volume",
+                       SUN50I_ADDA_HP_CTRL,
+                       SUN50I_ADDA_HP_CTRL_HPVOL, 0x3f, 0,
+                       sun50i_codec_hp_vol_scale),
+
+        SOC_DOUBLE("Headphone Playback Switch",
+                   SUN50I_ADDA_MIX_DAC_CTRL,
+                   SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE,
+                   SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE, 1, 0),
+
+        /* Mixer pre-gain */
+        SOC_SINGLE_TLV("Mic1 Playback Volume", SUN50I_ADDA_MIC1_CTRL,
+                       SUN50I_ADDA_MIC1_CTRL_MIC1G,
+                       0x7, 0, sun50i_codec_out_mixer_pregain_scale),
+
+        /* Microphone Amp boost gain */
+        SOC_SINGLE_TLV("Mic1 Boost Volume", SUN50I_ADDA_MIC1_CTRL,
+                       SUN50I_ADDA_MIC1_CTRL_MIC1BOOST, 0x7, 0,
+                       sun50i_codec_mic_gain_scale),
+
+        /* Mixer pre-gain */
+        SOC_SINGLE_TLV("Mic2 Playback Volume",
+                       SUN50I_ADDA_MIC2_CTRL, SUN50I_ADDA_MIC2_CTRL_MIC2G,
+                       0x7, 0, sun50i_codec_out_mixer_pregain_scale),
+
+        /* Microphone Amp boost gain */
+        SOC_SINGLE_TLV("Mic2 Boost Volume", SUN50I_ADDA_MIC2_CTRL,
+                       SUN50I_ADDA_MIC2_CTRL_MIC2BOOST, 0x7, 0,
+                       sun50i_codec_mic_gain_scale),
+
+        /* ADC */
+        SOC_SINGLE_TLV("ADC Gain Capture Volume", SUN50I_ADDA_ADC_CTRL,
+                       SUN50I_ADDA_ADC_CTRL_ADCG, 0x7, 0,
+                       sun50i_codec_out_mixer_pregain_scale),
+
+        /* Mixer pre-gain */
+        SOC_SINGLE_TLV("Line In Playback Volume", SUN50I_ADDA_LINEIN_CTRL,
+                       SUN50I_ADDA_LINEIN_CTRL_LINEING,
+                       0x7, 0, sun50i_codec_out_mixer_pregain_scale),
+
+        SOC_SINGLE_TLV("Line Out Playback Volume",
+                       SUN50I_ADDA_LINEOUT_CTRL1,
+                       SUN50I_ADDA_LINEOUT_CTRL1_VOL, 0x1f, 0,
+                       sun50i_codec_lineout_vol_scale),
+
+        SOC_DOUBLE("Line Out Playback Switch",
+                   SUN50I_ADDA_LINEOUT_CTRL0,
+                   SUN50I_ADDA_LINEOUT_CTRL0_LEN,
+                   SUN50I_ADDA_LINEOUT_CTRL0_REN, 1, 0),
+
+};
+
+static const char * const sun50i_codec_hp_src_enum_text[] = {
+        "DAC", "Mixer",
+};
+
+static SOC_ENUM_DOUBLE_DECL(sun50i_codec_hp_src_enum,
+                            SUN50I_ADDA_MIX_DAC_CTRL,
+                            SUN50I_ADDA_MIX_DAC_CTRL_LHPIS,
+                            SUN50I_ADDA_MIX_DAC_CTRL_RHPIS,
+                            sun50i_codec_hp_src_enum_text);
+
+static const struct snd_kcontrol_new sun50i_codec_hp_src[] = {
+        SOC_DAPM_ENUM("Headphone Source Playback Route",
+                      sun50i_codec_hp_src_enum),
+};
+
+static const char * const sun50i_codec_lineout_src_enum_text[] = {
+        "Stereo", "Mono Differential",
+};
+
+static SOC_ENUM_DOUBLE_DECL(sun50i_codec_lineout_src_enum,
+                            SUN50I_ADDA_LINEOUT_CTRL0,
+                            SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL,
+                            SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL,
+                            sun50i_codec_lineout_src_enum_text);
+
+static const struct snd_kcontrol_new sun50i_codec_lineout_src[] = {
+        SOC_DAPM_ENUM("Line Out Source Playback Route",
+                      sun50i_codec_lineout_src_enum),
+};
+
+static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = {
+        /* DAC */
+        SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL,
+                         SUN50I_ADDA_MIX_DAC_CTRL_DACALEN, 0),
+        SND_SOC_DAPM_DAC("Right DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL,
+                         SUN50I_ADDA_MIX_DAC_CTRL_DACAREN, 0),
+        /* ADC */
+        SND_SOC_DAPM_ADC("Left ADC", NULL, SUN50I_ADDA_ADC_CTRL,
+                         SUN50I_ADDA_ADC_CTRL_ADCLEN, 0),
+        SND_SOC_DAPM_ADC("Right ADC", NULL, SUN50I_ADDA_ADC_CTRL,
+                         SUN50I_ADDA_ADC_CTRL_ADCREN, 0),
+        /*
+         * Due to this component and the codec belonging to separate DAPM
+         * contexts, we need to manually link the above widgets to their
+         * stream widgets at the card level.
+         */
+
+        SND_SOC_DAPM_MUX("Headphone Source Playback Route",
+                         SND_SOC_NOPM, 0, 0, sun50i_codec_hp_src),
+        SND_SOC_DAPM_OUT_DRV("Headphone Amp", SUN50I_ADDA_HP_CTRL,
+                             SUN50I_ADDA_HP_CTRL_HPPA_EN, 0, NULL, 0),
+        SND_SOC_DAPM_OUTPUT("HP"),
+
+        SND_SOC_DAPM_MUX("Line Out Source Playback Route",
+                         SND_SOC_NOPM, 0, 0, sun50i_codec_lineout_src),
+        SND_SOC_DAPM_OUTPUT("LINEOUT"),
+
+        /* Microphone inputs */
+        SND_SOC_DAPM_INPUT("MIC1"),
+
+        /* Microphone Bias */
+        SND_SOC_DAPM_SUPPLY("MBIAS", SUN50I_ADDA_HS_MBIAS_CTRL,
+                            SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN,
+                            0, NULL, 0),
+
+        /* Mic input path */
+        SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN50I_ADDA_MIC1_CTRL,
+                         SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN, 0, NULL, 0),
+
+        /* Microphone input */
+        SND_SOC_DAPM_INPUT("MIC2"),
+
+        /* Microphone Bias */
+        SND_SOC_DAPM_SUPPLY("HBIAS", SUN50I_ADDA_JACK_MIC_CTRL,
+                            SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN,
+                            0, NULL, 0),
+
+        /* Mic input path */
+        SND_SOC_DAPM_PGA("Mic2 Amplifier", SUN50I_ADDA_MIC2_CTRL,
+                         SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN, 0, NULL, 0),
+
+        /* Line input */
+        SND_SOC_DAPM_INPUT("LINEIN"),
+
+        /* Mixers */
+        SND_SOC_DAPM_MIXER("Left Mixer", SUN50I_ADDA_MIX_DAC_CTRL,
+                           SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN, 0,
+                           sun50i_a64_codec_mixer_controls,
+                           ARRAY_SIZE(sun50i_a64_codec_mixer_controls)),
+        SND_SOC_DAPM_MIXER("Right Mixer", SUN50I_ADDA_MIX_DAC_CTRL,
+                           SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN, 0,
+                           sun50i_a64_codec_mixer_controls,
+                           ARRAY_SIZE(sun50i_a64_codec_mixer_controls)),
+        SND_SOC_DAPM_MIXER("Left ADC Mixer", SUN50I_ADDA_ADC_CTRL,
+                           SUN50I_ADDA_ADC_CTRL_ADCLEN, 0,
+                           sun50i_codec_adc_mixer_controls,
+                           ARRAY_SIZE(sun50i_codec_adc_mixer_controls)),
+        SND_SOC_DAPM_MIXER("Right ADC Mixer", SUN50I_ADDA_ADC_CTRL,
+                           SUN50I_ADDA_ADC_CTRL_ADCREN, 0,
+                           sun50i_codec_adc_mixer_controls,
+                           ARRAY_SIZE(sun50i_codec_adc_mixer_controls)),
+};
+
+static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = {
+        /* Left Mixer Routes */
+        { "Left Mixer", "DAC Playback Switch", "Left DAC" },
+        { "Left Mixer", "DAC Reversed Playback Switch", "Right DAC" },
+        { "Left Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" },
+
+        /* Right Mixer Routes */
+        { "Right Mixer", "DAC Playback Switch", "Right DAC" },
+        { "Right Mixer", "DAC Reversed Playback Switch", "Left DAC" },
+        { "Right Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" },
+
+        /* Left ADC Mixer Routes */
+        { "Left ADC Mixer", "Mixer Capture Switch", "Left Mixer" },
+        { "Left ADC Mixer", "Mixer Reversed Capture Switch", "Right Mixer" },
+        { "Left ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" },
+
+        /* Right ADC Mixer Routes */
+        { "Right ADC Mixer", "Mixer Capture Switch", "Right Mixer" },
+        { "Right ADC Mixer", "Mixer Reversed Capture Switch", "Left Mixer" },
+        { "Right ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" },
+
+        /* ADC Routes */
+        { "Left ADC", NULL, "Left ADC Mixer" },
+        { "Right ADC", NULL, "Right ADC Mixer" },
+
+        /* Headphone Routes */
+        { "Headphone Source Playback Route", "DAC", "Left DAC" },
+        { "Headphone Source Playback Route", "DAC", "Right DAC" },
+        { "Headphone Source Playback Route", "Mixer", "Left Mixer" },
+        { "Headphone Source Playback Route", "Mixer", "Right Mixer" },
+        { "Headphone Amp", NULL, "Headphone Source Playback Route" },
+        { "HP", NULL, "Headphone Amp" },
+
+        /* Microphone Routes */
+        { "Mic1 Amplifier", NULL, "MIC1"},
+
+        /* Microphone Routes */
+        { "Mic2 Amplifier", NULL, "MIC2"},
+        { "Left Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" },
+        { "Right Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" },
+        { "Left ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" },
+        { "Right ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" },
+
+        /* Line-in Routes */
+        { "Left Mixer", "Line In Playback Switch", "LINEIN" },
+        { "Right Mixer", "Line In Playback Switch", "LINEIN" },
+        { "Left ADC Mixer", "Line In Capture Switch", "LINEIN" },
+        { "Right ADC Mixer", "Line In Capture Switch", "LINEIN" },
+
+        /* Line-out Routes */
+        { "Line Out Source Playback Route", "Stereo", "Left Mixer" },
+        { "Line Out Source Playback Route", "Stereo", "Right Mixer" },
+        { "Line Out Source Playback Route", "Mono Differential", "Left Mixer" },
+        { "Line Out Source Playback Route", "Mono Differential",
+                "Right Mixer" },
+        { "LINEOUT", NULL, "Line Out Source Playback Route" },
+};
+
+static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = {
+        .controls               = sun50i_a64_codec_controls,
+        .num_controls           = ARRAY_SIZE(sun50i_a64_codec_controls),
+        .dapm_widgets           = sun50i_a64_codec_widgets,
+        .num_dapm_widgets       = ARRAY_SIZE(sun50i_a64_codec_widgets),
+        .dapm_routes            = sun50i_a64_codec_routes,
+        .num_dapm_routes        = ARRAY_SIZE(sun50i_a64_codec_routes),
+};
+
+static const struct of_device_id sun50i_codec_analog_of_match[] = {
+        {
+                .compatible = "allwinner,sun50i-a64-codec-analog",
+        },
+        {}
+};
+MODULE_DEVICE_TABLE(of, sun50i_codec_analog_of_match);
+
+static int sun50i_codec_analog_probe(struct platform_device *pdev)
+{
+        struct resource *res;
+        struct regmap *regmap;
+        void __iomem *base;
+
+        res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+        base = devm_ioremap_resource(&pdev->dev, res);
+        if (IS_ERR(base)) {
+                dev_err(&pdev->dev, "Failed to map the registers\n");
+                return PTR_ERR(base);
+        }
+
+        regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base);
+        if (IS_ERR(regmap)) {
+                dev_err(&pdev->dev, "Failed to create regmap\n");
+                return PTR_ERR(regmap);
+        }
+
+        return devm_snd_soc_register_component(&pdev->dev,
+                                               &sun50i_codec_analog_cmpnt_drv,
+                                               NULL, 0);
+}
+
+static struct platform_driver sun50i_codec_analog_driver = {
+        .driver = {
+                .name = "sun50i-codec-analog",
+                .of_match_table = sun50i_codec_analog_of_match,
+        },
+        .probe = sun50i_codec_analog_probe,
+};
+module_platform_driver(sun50i_codec_analog_driver);
+
+MODULE_DESCRIPTION("Allwinner internal codec analog controls driver for A64");
+MODULE_AUTHOR("Vasily Khoruzhick <anarsoul@gmail.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:sun50i-codec-analog");
diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.c b/sound/soc/sunxi/sun8i-adda-pr-regmap.c
new file mode 100644
index 000000000000..e68ce9d2884d
--- /dev/null
+++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.c
@@ -0,0 +1,102 @@
+// SPDX-License-Identifier: GPL-2.0+
+/*
+ * This driver provides regmap to access to analog part of audio codec
+ * found on Allwinner A23, A31s, A33, H3 and A64 Socs
+ *
+ * Copyright 2016 Chen-Yu Tsai <wens@csie.org>
+ * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com>
+ */
+
+#include <linux/io.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+
+#include "sun8i-adda-pr-regmap.h"
+
+/* Analog control register access bits */
+#define ADDA_PR                 0x0             /* PRCM base + 0x1c0 */
+#define ADDA_PR_RESET                   BIT(28)
+#define ADDA_PR_WRITE                   BIT(24)
+#define ADDA_PR_ADDR_SHIFT              16
+#define ADDA_PR_ADDR_MASK               GENMASK(4, 0)
+#define ADDA_PR_DATA_IN_SHIFT           8
+#define ADDA_PR_DATA_IN_MASK            GENMASK(7, 0)
+#define ADDA_PR_DATA_OUT_SHIFT          0
+#define ADDA_PR_DATA_OUT_MASK           GENMASK(7, 0)
+
+/* regmap access bits */
+static int adda_reg_read(void *context, unsigned int reg, unsigned int *val)
+{
+        void __iomem *base = (void __iomem *)context;
+        u32 tmp;
+
+        /* De-assert reset */
+        writel(readl(base) | ADDA_PR_RESET, base);
+
+        /* Clear write bit */
+        writel(readl(base) & ~ADDA_PR_WRITE, base);
+
+        /* Set register address */
+        tmp = readl(base);
+        tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
+        tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
+        writel(tmp, base);
+
+        /* Read back value */
+        *val = readl(base) & ADDA_PR_DATA_OUT_MASK;
+
+        return 0;
+}
+
+static int adda_reg_write(void *context, unsigned int reg, unsigned int val)
+{
+        void __iomem *base = (void __iomem *)context;
+        u32 tmp;
+
+        /* De-assert reset */
+        writel(readl(base) | ADDA_PR_RESET, base);
+
+        /* Set register address */
+        tmp = readl(base);
+        tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
+        tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
+        writel(tmp, base);
+
+        /* Set data to write */
+        tmp = readl(base);
+        tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT);
+        tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT;
+        writel(tmp, base);
+
+        /* Set write bit to signal a write */
+        writel(readl(base) | ADDA_PR_WRITE, base);
+
+        /* Clear write bit */
+        writel(readl(base) & ~ADDA_PR_WRITE, base);
+
+        return 0;
+}
+
+static const struct regmap_config adda_pr_regmap_cfg = {
+        .name           = "adda-pr",
+        .reg_bits       = 5,
+        .reg_stride     = 1,
+        .val_bits       = 8,
+        .reg_read       = adda_reg_read,
+        .reg_write      = adda_reg_write,
+        .fast_io        = true,
+        .max_register   = 31,
+};
+
+struct regmap *sun8i_adda_pr_regmap_init(struct device *dev,
+                                         void __iomem *base)
+{
+        return devm_regmap_init(dev, NULL, base, &adda_pr_regmap_cfg);
+}
+EXPORT_SYMBOL_GPL(sun8i_adda_pr_regmap_init);
+
+MODULE_DESCRIPTION("Allwinner analog audio codec regmap driver");
+MODULE_AUTHOR("Vasily Khoruzhick <anarsoul@gmail.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:sunxi-adda-pr");
diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.h b/sound/soc/sunxi/sun8i-adda-pr-regmap.h
new file mode 100644
index 000000000000..a5ae95dfebc1
--- /dev/null
+++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.h
@@ -0,0 +1,7 @@
+/* SPDX-License-Identifier: GPL-2.0+ */
+/*
+ * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com>
+ */
+
+struct regmap *sun8i_adda_pr_regmap_init(struct device *dev,
+                                         void __iomem *base);
diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c
index 485e79f292c4..916a46bbc1c8 100644
--- a/sound/soc/sunxi/sun8i-codec-analog.c
+++ b/sound/soc/sunxi/sun8i-codec-analog.c
@@ -27,6 +27,8 @@
 #include <sound/soc-dapm.h>
 #include <sound/tlv.h>
 
+#include "sun8i-adda-pr-regmap.h"
+
 /* Codec analog control register offsets and bit fields */
 #define SUN8I_ADDA_HP_VOLC              0x00
 #define SUN8I_ADDA_HP_VOLC_PA_CLK_GATE          7
@@ -120,81 +122,6 @@
 #define SUN8I_ADDA_ADC_AP_EN_ADCLEN             6
 #define SUN8I_ADDA_ADC_AP_EN_ADCG               0
 
-/* Analog control register access bits */
-#define ADDA_PR                 0x0             /* PRCM base + 0x1c0 */
-#define ADDA_PR_RESET                   BIT(28)
-#define ADDA_PR_WRITE                   BIT(24)
-#define ADDA_PR_ADDR_SHIFT              16
-#define ADDA_PR_ADDR_MASK               GENMASK(4, 0)
-#define ADDA_PR_DATA_IN_SHIFT           8
-#define ADDA_PR_DATA_IN_MASK            GENMASK(7, 0)
-#define ADDA_PR_DATA_OUT_SHIFT          0
-#define ADDA_PR_DATA_OUT_MASK           GENMASK(7, 0)
-
-/* regmap access bits */
-static int adda_reg_read(void *context, unsigned int reg, unsigned int *val)
-{
-        void __iomem *base = (void __iomem *)context;
-        u32 tmp;
-
-        /* De-assert reset */
-        writel(readl(base) | ADDA_PR_RESET, base);
-
-        /* Clear write bit */
-        writel(readl(base) & ~ADDA_PR_WRITE, base);
-
-        /* Set register address */
-        tmp = readl(base);
-        tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
-        tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
-        writel(tmp, base);
-
-        /* Read back value */
-        *val = readl(base) & ADDA_PR_DATA_OUT_MASK;
-
-        return 0;
-}
-
-static int adda_reg_write(void *context, unsigned int reg, unsigned int val)
-{
-        void __iomem *base = (void __iomem *)context;
-        u32 tmp;
-
-        /* De-assert reset */
-        writel(readl(base) | ADDA_PR_RESET, base);
-
-        /* Set register address */
-        tmp = readl(base);
-        tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
-        tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
-        writel(tmp, base);
-
-        /* Set data to write */
-        tmp = readl(base);
-        tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT);
-        tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT;
-        writel(tmp, base);
-
-        /* Set write bit to signal a write */
-        writel(readl(base) | ADDA_PR_WRITE, base);
-
-        /* Clear write bit */
-        writel(readl(base) & ~ADDA_PR_WRITE, base);
-
-        return 0;
-}
-
-static const struct regmap_config adda_pr_regmap_cfg = {
-        .name           = "adda-pr",
-        .reg_bits       = 5,
-        .reg_stride     = 1,
-        .val_bits       = 8,
-        .reg_read       = adda_reg_read,
-        .reg_write      = adda_reg_write,
-        .fast_io        = true,
-        .max_register   = 24,
-};
-
 /* mixer controls */
 static const struct snd_kcontrol_new sun8i_codec_mixer_controls[] = {
         SOC_DAPM_DOUBLE_R("DAC Playback Switch",
@@ -912,7 +839,7 @@ static int sun8i_codec_analog_probe(struct platform_device *pdev)
                 return PTR_ERR(base);
         }
 
-        regmap = devm_regmap_init(&pdev->dev, NULL, base, &adda_pr_regmap_cfg);
+        regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base);
         if (IS_ERR(regmap)) {
                 dev_err(&pdev->dev, "Failed to create regmap\n");
                 return PTR_ERR(regmap);
diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c
index fb37dd927e33..522a72fde78d 100644
--- a/sound/soc/sunxi/sun8i-codec.c
+++ b/sound/soc/sunxi/sun8i-codec.c
@@ -24,6 +24,7 @@
 #include <linux/io.h>
 #include <linux/pm_runtime.h>
 #include <linux/regmap.h>
+#include <linux/log2.h>
 
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
@@ -52,7 +53,6 @@
 #define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_INV                13
 #define SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV                9
 #define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV                6
-#define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_16             (1 << 6)
 #define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ                4
 #define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ_16             (1 << 4)
 #define SUN8I_AIF1CLK_CTRL_AIF1_DATA_FMT                2
@@ -300,12 +300,23 @@ static u8 sun8i_codec_get_bclk_div(struct sun8i_codec *scodec,
         return best_val;
 }
 
+static int sun8i_codec_get_lrck_div(unsigned int channels,
+                                    unsigned int word_size)
+{
+        unsigned int div = word_size * channels;
+
+        if (div < 16 || div > 256)
+                return -EINVAL;
+
+        return ilog2(div) - 4;
+}
+
 static int sun8i_codec_hw_params(struct snd_pcm_substream *substream,
                                  struct snd_pcm_hw_params *params,
                                  struct snd_soc_dai *dai)
 {
         struct sun8i_codec *scodec = snd_soc_component_get_drvdata(dai->component);
-        int sample_rate;
+        int sample_rate, lrck_div;
         u8 bclk_div;
 
         /*
@@ -321,9 +332,14 @@ static int sun8i_codec_hw_params(struct snd_pcm_substream *substream,
                            SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK,
                            bclk_div << SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV);
 
+        lrck_div = sun8i_codec_get_lrck_div(params_channels(params),
+                                            params_physical_width(params));
+        if (lrck_div < 0)
+                return lrck_div;
+
         regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL,
                            SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_MASK,
-                           SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_16);
+                           lrck_div << SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV);
 
         sample_rate = sun8i_codec_get_hw_rate(params);
         if (sample_rate < 0)
diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c
index 45a4aa9d2a47..901457da25ec 100644
--- a/sound/soc/tegra/tegra_sgtl5000.c
+++ b/sound/soc/tegra/tegra_sgtl5000.c
@@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev)
                 dev_err(&pdev->dev,
                         "Property 'nvidia,i2s-controller' missing/invalid\n");
                 ret = -EINVAL;
-                goto err;
+                goto err_put_codec_of_node;
         }
 
         tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_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) {
@@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev)
 
 err_fini_utils:
         tegra_asoc_utils_fini(&machine->util_data);
+err_put_cpu_of_node:
+        of_node_put(tegra_sgtl5000_dai.cpu_of_node);
+        tegra_sgtl5000_dai.cpu_of_node = NULL;
+        tegra_sgtl5000_dai.platform_of_node = NULL;
+err_put_codec_of_node:
+        of_node_put(tegra_sgtl5000_dai.codec_of_node);
+        tegra_sgtl5000_dai.codec_of_node = NULL;
 err:
         return ret;
 }
@@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev)
 
         tegra_asoc_utils_fini(&machine->util_data);
 
+        of_node_put(tegra_sgtl5000_dai.cpu_of_node);
+        tegra_sgtl5000_dai.cpu_of_node = NULL;
+        tegra_sgtl5000_dai.platform_of_node = NULL;
+        of_node_put(tegra_sgtl5000_dai.codec_of_node);
+        tegra_sgtl5000_dai.codec_of_node = NULL;
+
         return ret;
 }
 
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
index e2ad00e3cae1..1cfca698ae4b 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -208,13 +208,12 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev)
         if (err < 0)
                 return err;
 
-        return snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component,
+        return devm_snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component,
                                           &txx9aclc_ac97_dai, 1);
 }
 
 static int txx9aclc_ac97_dev_remove(struct platform_device *pdev)
 {
-        snd_soc_unregister_component(&pdev->dev);
         snd_soc_set_ac97_ops(NULL);
         return 0;
 }
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index d55ca48de3ea..f4a72e39ffa9 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -200,6 +200,7 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb)
                         break;
                 }
 #ifdef CONFIG_SND_USB_CAIAQ_INPUT
+                /* fall through */
         case EP1_CMD_READ_ERP:
         case EP1_CMD_READ_ANALOG:
                 snd_usb_caiaq_input_dispatch(cdev, buf, urb->actual_length);
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index dcfc546d81b9..b737f0ec77d0 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1175,8 +1175,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream,
                 if (port->ep->umidi->disconnected) {
                         /* gobble up remaining bytes to prevent wait in
                          * snd_rawmidi_drain_output */
-                        while (!snd_rawmidi_transmit_empty(substream))
-                                snd_rawmidi_transmit_ack(substream, 1);
+                        snd_rawmidi_proceed(substream);
                         return;
                 }
                 tasklet_schedule(&port->ep->tasklet);
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index cbfb48bdea51..85ae0ff2382a 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -29,6 +29,7 @@
 
 #include <linux/hid.h>
 #include <linux/init.h>
+#include <linux/math64.h>
 #include <linux/slab.h>
 #include <linux/usb.h>
 #include <linux/usb/audio.h>
@@ -1817,6 +1818,380 @@ static int dell_dock_mixer_init(struct usb_mixer_interface *mixer)
         return 0;
 }
 
+/* RME Class Compliant device quirks */
+
+#define SND_RME_GET_STATUS1                     23
+#define SND_RME_GET_CURRENT_FREQ                17
+#define SND_RME_CLK_SYSTEM_SHIFT                16
+#define SND_RME_CLK_SYSTEM_MASK                 0x1f
+#define SND_RME_CLK_AES_SHIFT                   8
+#define SND_RME_CLK_SPDIF_SHIFT                 12
+#define SND_RME_CLK_AES_SPDIF_MASK              0xf
+#define SND_RME_CLK_SYNC_SHIFT                  6
+#define SND_RME_CLK_SYNC_MASK                   0x3
+#define SND_RME_CLK_FREQMUL_SHIFT               18
+#define SND_RME_CLK_FREQMUL_MASK                0x7
+#define SND_RME_CLK_SYSTEM(x) \
+        ((x >> SND_RME_CLK_SYSTEM_SHIFT) & SND_RME_CLK_SYSTEM_MASK)
+#define SND_RME_CLK_AES(x) \
+        ((x >> SND_RME_CLK_AES_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK)
+#define SND_RME_CLK_SPDIF(x) \
+        ((x >> SND_RME_CLK_SPDIF_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK)
+#define SND_RME_CLK_SYNC(x) \
+        ((x >> SND_RME_CLK_SYNC_SHIFT) & SND_RME_CLK_SYNC_MASK)
+#define SND_RME_CLK_FREQMUL(x) \
+        ((x >> SND_RME_CLK_FREQMUL_SHIFT) & SND_RME_CLK_FREQMUL_MASK)
+#define SND_RME_CLK_AES_LOCK                    0x1
+#define SND_RME_CLK_AES_SYNC                    0x4
+#define SND_RME_CLK_SPDIF_LOCK                  0x2
+#define SND_RME_CLK_SPDIF_SYNC                  0x8
+#define SND_RME_SPDIF_IF_SHIFT                  4
+#define SND_RME_SPDIF_FORMAT_SHIFT              5
+#define SND_RME_BINARY_MASK                     0x1
+#define SND_RME_SPDIF_IF(x) \
+        ((x >> SND_RME_SPDIF_IF_SHIFT) & SND_RME_BINARY_MASK)
+#define SND_RME_SPDIF_FORMAT(x) \
+        ((x >> SND_RME_SPDIF_FORMAT_SHIFT) & SND_RME_BINARY_MASK)
+
+static const u32 snd_rme_rate_table[] = {
+        32000, 44100, 48000, 50000,
+        64000, 88200, 96000, 100000,
+        128000, 176400, 192000, 200000,
+        256000, 352800, 384000, 400000,
+        512000, 705600, 768000, 800000
+};
+/* maximum number of items for AES and S/PDIF rates for above table */
+#define SND_RME_RATE_IDX_AES_SPDIF_NUM          12
+
+enum snd_rme_domain {
+        SND_RME_DOMAIN_SYSTEM,
+        SND_RME_DOMAIN_AES,
+        SND_RME_DOMAIN_SPDIF
+};
+
+enum snd_rme_clock_status {
+        SND_RME_CLOCK_NOLOCK,
+        SND_RME_CLOCK_LOCK,
+        SND_RME_CLOCK_SYNC
+};
+
+static int snd_rme_read_value(struct snd_usb_audio *chip,
+                              unsigned int item,
+                              u32 *value)
+{
+        struct usb_device *dev = chip->dev;
+        int err;
+
+        err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
+                              item,
+                              USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE,
+                              0, 0,
+                              value, sizeof(*value));
+        if (err < 0)
+                dev_err(&dev->dev,
+                        "unable to issue vendor read request %d (ret = %d)",
+                        item, err);
+        return err;
+}
+
+static int snd_rme_get_status1(struct snd_kcontrol *kcontrol,
+                               u32 *status1)
+{
+        struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol);
+        struct snd_usb_audio *chip = list->mixer->chip;
+        int err;
+
+        err = snd_usb_lock_shutdown(chip);
+        if (err < 0)
+                return err;
+        err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, status1);
+        snd_usb_unlock_shutdown(chip);
+        return err;
+}
+
+static int snd_rme_rate_get(struct snd_kcontrol *kcontrol,
+                            struct snd_ctl_elem_value *ucontrol)
+{
+        u32 status1;
+        u32 rate = 0;
+        int idx;
+        int err;
+
+        err = snd_rme_get_status1(kcontrol, &status1);
+        if (err < 0)
+                return err;
+        switch (kcontrol->private_value) {
+        case SND_RME_DOMAIN_SYSTEM:
+                idx = SND_RME_CLK_SYSTEM(status1);
+                if (idx < ARRAY_SIZE(snd_rme_rate_table))
+                        rate = snd_rme_rate_table[idx];
+                break;
+        case SND_RME_DOMAIN_AES:
+                idx = SND_RME_CLK_AES(status1);
+                if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM)
+                        rate = snd_rme_rate_table[idx];
+                break;
+        case SND_RME_DOMAIN_SPDIF:
+                idx = SND_RME_CLK_SPDIF(status1);
+                if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM)
+                        rate = snd_rme_rate_table[idx];
+                break;
+        default:
+                return -EINVAL;
+        }
+        ucontrol->value.integer.value[0] = rate;
+        return 0;
+}
+
+static int snd_rme_sync_state_get(struct snd_kcontrol *kcontrol,
+                                  struct snd_ctl_elem_value *ucontrol)
+{
+        u32 status1;
+        int idx = SND_RME_CLOCK_NOLOCK;
+        int err;
+
+        err = snd_rme_get_status1(kcontrol, &status1);
+        if (err < 0)
+                return err;
+        switch (kcontrol->private_value) {
+        case SND_RME_DOMAIN_AES:  /* AES */
+                if (status1 & SND_RME_CLK_AES_SYNC)
+                        idx = SND_RME_CLOCK_SYNC;
+                else if (status1 & SND_RME_CLK_AES_LOCK)
+                        idx = SND_RME_CLOCK_LOCK;
+                break;
+        case SND_RME_DOMAIN_SPDIF:  /* SPDIF */
+                if (status1 & SND_RME_CLK_SPDIF_SYNC)
+                        idx = SND_RME_CLOCK_SYNC;
+                else if (status1 & SND_RME_CLK_SPDIF_LOCK)
+                        idx = SND_RME_CLOCK_LOCK;
+                break;
+        default:
+                return -EINVAL;
+        }
+        ucontrol->value.enumerated.item[0] = idx;
+        return 0;
+}
+
+static int snd_rme_spdif_if_get(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_value *ucontrol)
+{
+        u32 status1;
+        int err;
+
+        err = snd_rme_get_status1(kcontrol, &status1);
+        if (err < 0)
+                return err;
+        ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_IF(status1);
+        return 0;
+}
+
+static int snd_rme_spdif_format_get(struct snd_kcontrol *kcontrol,
+                                    struct snd_ctl_elem_value *ucontrol)
+{
+        u32 status1;
+        int err;
+
+        err = snd_rme_get_status1(kcontrol, &status1);
+        if (err < 0)
+                return err;
+        ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_FORMAT(status1);
+        return 0;
+}
+
+static int snd_rme_sync_source_get(struct snd_kcontrol *kcontrol,
+                                   struct snd_ctl_elem_value *ucontrol)
+{
+        u32 status1;
+        int err;
+
+        err = snd_rme_get_status1(kcontrol, &status1);
+        if (err < 0)
+                return err;
+        ucontrol->value.enumerated.item[0] = SND_RME_CLK_SYNC(status1);
+        return 0;
+}
+
+static int snd_rme_current_freq_get(struct snd_kcontrol *kcontrol,
+                                    struct snd_ctl_elem_value *ucontrol)
+{
+        struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol);
+        struct snd_usb_audio *chip = list->mixer->chip;
+        u32 status1;
+        const u64 num = 104857600000000ULL;
+        u32 den;
+        unsigned int freq;
+        int err;
+
+        err = snd_usb_lock_shutdown(chip);
+        if (err < 0)
+                return err;
+        err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, &status1);
+        if (err < 0)
+                goto end;
+        err = snd_rme_read_value(chip, SND_RME_GET_CURRENT_FREQ, &den);
+        if (err < 0)
+                goto end;
+        freq = (den == 0) ? 0 : div64_u64(num, den);
+        freq <<= SND_RME_CLK_FREQMUL(status1);
+        ucontrol->value.integer.value[0] = freq;
+
+end:
+        snd_usb_unlock_shutdown(chip);
+        return err;
+}
+
+static int snd_rme_rate_info(struct snd_kcontrol *kcontrol,
+                             struct snd_ctl_elem_info *uinfo)
+{
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->count = 1;
+        switch (kcontrol->private_value) {
+        case SND_RME_DOMAIN_SYSTEM:
+                uinfo->value.integer.min = 32000;
+                uinfo->value.integer.max = 800000;
+                break;
+        case SND_RME_DOMAIN_AES:
+        case SND_RME_DOMAIN_SPDIF:
+        default:
+                uinfo->value.integer.min = 0;
+                uinfo->value.integer.max = 200000;
+        }
+        uinfo->value.integer.step = 0;
+        return 0;
+}
+
+static int snd_rme_sync_state_info(struct snd_kcontrol *kcontrol,
+                                   struct snd_ctl_elem_info *uinfo)
+{
+        static const char *const sync_states[] = {
+                "No Lock", "Lock", "Sync"
+        };
+
+        return snd_ctl_enum_info(uinfo, 1,
+                                 ARRAY_SIZE(sync_states), sync_states);
+}
+
+static int snd_rme_spdif_if_info(struct snd_kcontrol *kcontrol,
+                                 struct snd_ctl_elem_info *uinfo)
+{
+        static const char *const spdif_if[] = {
+                "Coaxial", "Optical"
+        };
+
+        return snd_ctl_enum_info(uinfo, 1,
+                                 ARRAY_SIZE(spdif_if), spdif_if);
+}
+
+static int snd_rme_spdif_format_info(struct snd_kcontrol *kcontrol,
+                                     struct snd_ctl_elem_info *uinfo)
+{
+        static const char *const optical_type[] = {
+                "Consumer", "Professional"
+        };
+
+        return snd_ctl_enum_info(uinfo, 1,
+                                 ARRAY_SIZE(optical_type), optical_type);
+}
+
+static int snd_rme_sync_source_info(struct snd_kcontrol *kcontrol,
+                                    struct snd_ctl_elem_info *uinfo)
+{
+        static const char *const sync_sources[] = {
+                "Internal", "AES", "SPDIF", "Internal"
+        };
+
+        return snd_ctl_enum_info(uinfo, 1,
+                                 ARRAY_SIZE(sync_sources), sync_sources);
+}
+
+static struct snd_kcontrol_new snd_rme_controls[] = {
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                .name = "AES Rate",
+                .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+                .info = snd_rme_rate_info,
+                .get = snd_rme_rate_get,
+                .private_value = SND_RME_DOMAIN_AES
+        },
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                .name = "AES Sync",
+                .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+                .info = snd_rme_sync_state_info,
+                .get = snd_rme_sync_state_get,
+                .private_value = SND_RME_DOMAIN_AES
+        },
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                .name = "SPDIF Rate",
+                .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+                .info = snd_rme_rate_info,
+                .get = snd_rme_rate_get,
+                .private_value = SND_RME_DOMAIN_SPDIF
+        },
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                .name = "SPDIF Sync",
+                .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+                .info = snd_rme_sync_state_info,
+                .get = snd_rme_sync_state_get,
+                .private_value = SND_RME_DOMAIN_SPDIF
+        },
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                .name = "SPDIF Interface",
+                .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+                .info = snd_rme_spdif_if_info,
+                .get = snd_rme_spdif_if_get,
+        },
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                .name = "SPDIF Format",
+                .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+                .info = snd_rme_spdif_format_info,
+                .get = snd_rme_spdif_format_get,
+        },
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                .name = "Sync Source",
+                .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+                .info = snd_rme_sync_source_info,
+                .get = snd_rme_sync_source_get
+        },
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                .name = "System Rate",
+                .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+                .info = snd_rme_rate_info,
+                .get = snd_rme_rate_get,
+                .private_value = SND_RME_DOMAIN_SYSTEM
+        },
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                .name = "Current Frequency",
+                .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+                .info = snd_rme_rate_info,
+                .get = snd_rme_current_freq_get
+        }
+};
+
+static int snd_rme_controls_create(struct usb_mixer_interface *mixer)
+{
+        int err, i;
+
+        for (i = 0; i < ARRAY_SIZE(snd_rme_controls); ++i) {
+                err = add_single_ctl_with_resume(mixer, 0,
+                                                 NULL,
+                                                 &snd_rme_controls[i],
+                                                 NULL);
+                if (err < 0)
+                        return err;
+        }
+
+        return 0;
+}
+
 int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 {
         int err = 0;
@@ -1904,6 +2279,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
         case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */
                 err = dell_dock_mixer_init(mixer);
                 break;
+
+        case USB_ID(0x2a39, 0x3fd2): /* RME ADI-2 Pro */
+        case USB_ID(0x2a39, 0x3fd3): /* RME ADI-2 DAC */
+        case USB_ID(0x2a39, 0x3fd4): /* RME */
+                err = snd_rme_controls_create(mixer);
+                break;
         }
 
         return err;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 08aa78007020..849953e5775c 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3346,19 +3346,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
                                 .ifnum = 0,
                                 .type = QUIRK_AUDIO_STANDARD_MIXER,
                         },
-                        /* Capture */
-                        {
-                                .ifnum = 1,
-                                .type = QUIRK_IGNORE_INTERFACE,
-                        },
                         /* Playback */
                         {
-                                .ifnum = 2,
+                                .ifnum = 1,
                                 .type = QUIRK_AUDIO_FIXED_ENDPOINT,
                                 .data = &(const struct audioformat) {
                                         .formats = SNDRV_PCM_FMTBIT_S16_LE,
                                         .channels = 2,
-                                        .iface = 2,
+                                        .iface = 1,
                                         .altsetting = 1,
                                         .altset_idx = 1,
                                         .attributes = UAC_EP_CS_ATTR_FILL_MAX |
diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c
index fa7dca5a68c8..83d76c345940 100644
--- a/sound/x86/intel_hdmi_audio.c
+++ b/sound/x86/intel_hdmi_audio.c
@@ -30,7 +30,6 @@
 #include <linux/pm_runtime.h>
 #include <linux/dma-mapping.h>
 #include <linux/delay.h>
-#include <asm/set_memory.h>
 #include <sound/core.h>
 #include <sound/asoundef.h>
 #include <sound/pcm.h>
@@ -1141,8 +1140,7 @@ static int had_pcm_hw_params(struct snd_pcm_substream *substream,
                              struct snd_pcm_hw_params *hw_params)
 {
         struct snd_intelhad *intelhaddata;
-        unsigned long addr;
-        int pages, buf_size, retval;
+        int buf_size, retval;
 
         intelhaddata = snd_pcm_substream_chip(substream);
         buf_size = params_buffer_bytes(hw_params);
@@ -1151,17 +1149,6 @@ static int had_pcm_hw_params(struct snd_pcm_substream *substream,
                 return retval;
         dev_dbg(intelhaddata->dev, "%s:allocated memory = %d\n",
                 __func__, buf_size);
-        /* mark the pages as uncached region */
-        addr = (unsigned long) substream->runtime->dma_area;
-        pages = (substream->runtime->dma_bytes + PAGE_SIZE - 1) / PAGE_SIZE;
-        retval = set_memory_uc(addr, pages);
-        if (retval) {
-                dev_err(intelhaddata->dev, "set_memory_uc failed.Error:%d\n",
-                        retval);
-                return retval;
-        }
-        memset(substream->runtime->dma_area, 0, buf_size);
-
         return retval;
 }
 
@@ -1171,21 +1158,11 @@ static int had_pcm_hw_params(struct snd_pcm_substream *substream,
 static int had_pcm_hw_free(struct snd_pcm_substream *substream)
 {
         struct snd_intelhad *intelhaddata;
-        unsigned long addr;
-        u32 pages;
 
         intelhaddata = snd_pcm_substream_chip(substream);
         had_do_reset(intelhaddata);
 
-        /* mark back the pages as cached/writeback region before the free */
-        if (substream->runtime->dma_area != NULL) {
-                addr = (unsigned long) substream->runtime->dma_area;
-                pages = (substream->runtime->dma_bytes + PAGE_SIZE - 1) /
-                                                                PAGE_SIZE;
-                set_memory_wb(addr, pages);
-                return snd_pcm_lib_free_pages(substream);
-        }
-        return 0;
+        return snd_pcm_lib_free_pages(substream);
 }
 
 /*
@@ -1860,7 +1837,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev)
                  * try to allocate 600k buffer as default which is large enough
                  */
                 snd_pcm_lib_preallocate_pages_for_all(pcm,
-                                                      SNDRV_DMA_TYPE_DEV, NULL,
+                                                      SNDRV_DMA_TYPE_DEV_UC, NULL,
                                                       HAD_DEFAULT_BUFFER, HAD_MAX_BUFFER);
 
                 /* create controls */
diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c
index 129180e17db1..2cbd9679aca1 100644
--- a/sound/xen/xen_snd_front_alsa.c
+++ b/sound/xen/xen_snd_front_alsa.c
@@ -637,31 +637,31 @@ static int alsa_pb_fill_silence(struct snd_pcm_substream *substream,
  * to know when the buffer can be transferred to the backend.
  */
 
-static struct snd_pcm_ops snd_drv_alsa_playback_ops = {
-        .open = alsa_open,
-        .close = alsa_close,
-        .ioctl = snd_pcm_lib_ioctl,
-        .hw_params = alsa_hw_params,
-        .hw_free = alsa_hw_free,
-        .prepare = alsa_prepare,
-        .trigger = alsa_trigger,
-        .pointer = alsa_pointer,
-        .copy_user = alsa_pb_copy_user,
-        .copy_kernel = alsa_pb_copy_kernel,
-        .fill_silence = alsa_pb_fill_silence,
+static const struct snd_pcm_ops snd_drv_alsa_playback_ops = {
+        .open           = alsa_open,
+        .close          = alsa_close,
+        .ioctl          = snd_pcm_lib_ioctl,
+        .hw_params      = alsa_hw_params,
+        .hw_free        = alsa_hw_free,
+        .prepare        = alsa_prepare,
+        .trigger        = alsa_trigger,
+        .pointer        = alsa_pointer,
+        .copy_user      = alsa_pb_copy_user,
+        .copy_kernel    = alsa_pb_copy_kernel,
+        .fill_silence   = alsa_pb_fill_silence,
 };
 
-static struct snd_pcm_ops snd_drv_alsa_capture_ops = {
-        .open = alsa_open,
-        .close = alsa_close,
-        .ioctl = snd_pcm_lib_ioctl,
-        .hw_params = alsa_hw_params,
-        .hw_free = alsa_hw_free,
-        .prepare = alsa_prepare,
-        .trigger = alsa_trigger,
-        .pointer = alsa_pointer,
-        .copy_user = alsa_cap_copy_user,
-        .copy_kernel = alsa_cap_copy_kernel,
+static const struct snd_pcm_ops snd_drv_alsa_capture_ops = {
+        .open           = alsa_open,
+        .close          = alsa_close,
+        .ioctl          = snd_pcm_lib_ioctl,
+        .hw_params      = alsa_hw_params,
+        .hw_free        = alsa_hw_free,
+        .prepare        = alsa_prepare,
+        .trigger        = alsa_trigger,
+        .pointer        = alsa_pointer,
+        .copy_user      = alsa_cap_copy_user,
+        .copy_kernel    = alsa_cap_copy_kernel,
 };
 
 static int new_pcm_instance(struct xen_snd_front_card_info *card_info,