sound updates for 4.20

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
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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
  ...

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>;
+	};

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";
+};

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";
+};

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";
+};

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";
+        };
+};

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>;
+};

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>;
+	};

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>;
 		};
 	};
 };

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

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

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/

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:

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>;
+	};

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).
 

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>;
+};

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

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

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

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.
+The file tree structure of ALSA driver is depicted below.
-
-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.
 
 ::
 
@@ -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.
+Here contains OSS/Lite codes.
-In the ALSA driver tarball, this directory is empty, of course :)
+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) {
+              if (err < 0)
-                      snd_card_free(card);
+                      goto error;
-                      return err;
-              }
 
               /* (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) {
+              if (err < 0)
-                      snd_card_free(card);
+                      goto error;
-                      return err;
-              }
 
               /* (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) {
+  if (err < 0)
-          snd_card_free(card);
+          goto error;
-          return err;
-  }
 
 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) {
+  if (err < 0)
-          snd_card_free(card);
+          goto error;
-          return err;
-  }
 
 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
+functions.  These resources must be released in the destructor
-these resources must be released in the destructor function (see below).
+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.
 
 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 __maybe_unused snd_my_suspend(struct device *dev)
-  static int snd_my_suspend(struct pci_dev *pci, pm_message_t state)
   {
           .... /* 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
+2. Re-initialize the chip.
-   enable the pci device again by calling
-   :c:func:`pci_enable_device()`. Call
-   :c:func:`pci_set_master()` if necessary, too.
 
-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).
+6. Call :c:func:`snd_power_change_state()` with
-
-7. 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
+          .driver.pm = &snd_my_pm_ops,
-          .suspend = snd_my_suspend,
-          .resume = snd_my_resume,
-  #endif
   };
 
 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
+Also, don't forget to define the module description and the license.
-and devices. Especially, the recent modprobe requires to define the
+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
+alsa-driver tree, ``sound/pci`` directory in the case of PCI
-cards. Then the driver is evaluated, audited and tested by developers
+cards.
-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``).
 
 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
+1. Add a new directory (``sound/pci/xyz``) in ``sound/pci/Makefile``
-   below
+   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 set.
-``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.
 
 :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

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)

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 */

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 */
 

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,		\
+	asoc_simple_card_parse_dai(node, dai_link->codecs,			\
-		&dai_link->codec_dai_name, list_name, cells_name, NULL)
+				   &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);
 

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

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 */

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);

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)

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 */

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)",
+	char node_name[6];
-					"i2sbus: %s (tx)",
+	static const char *rnames[] = { "i2sbus: %pOFn (control)",
-					"i2sbus: %s (rx)" };
+					"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);

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

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,
+static void snd_free_dev_pages(struct snd_dma_buffer *dmab)
-			       dma_addr_t dma)
 {
-	int pg;
+#ifdef CONFIG_X86
-
+	if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC)
-	if (ptr == NULL)
+		set_memory_wb((unsigned long)dmab->area,
-		return;
+			      PAGE_ALIGN(dmab->bytes) >> PAGE_SHIFT);
-	pg = get_order(size);
+#endif
-	dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma);
+	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

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);

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;
 
-	if (!is_playback &&
-	    runtime->status->state == SNDRV_PCM_STATE_PREPARED &&
-	    size >= runtime->start_threshold) {
-		err = snd_pcm_start(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) {
+		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;
+		}
+	}
+
 	avail = snd_pcm_avail(substream);
+
 	while (size > 0) {
 		snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
 		snd_pcm_uframes_t cont;

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,

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;

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;
 

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];
+		snd_rawmidi_proceed(substream);
-		while (snd_rawmidi_transmit(substream, buf, sizeof(buf)) > 0)
-			; /* ignored */
 		return;
 	}
 

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)

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.

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,
+	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
-					 SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32);
+				  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;
 }

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);
+	bebob = devm_kzalloc(&unit->device, sizeof(struct snd_bebob),
-	if (bebob == NULL)
+			     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. */
+		// Block till all of ALSA character devices are released.
-		snd_card_free_when_closed(bebob->card);
+		snd_card_free(bebob->card);
-	} else {
-		/* Don't forget this case. */
-		bebob_free(bebob);
 	}
+
+	mutex_destroy(&bebob->mutex);
+	fw_unit_put(bebob->unit);
 }
 
 static const struct snd_bebob_rate_spec normal_rate_spec = {

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);
+	params = devm_kzalloc(&bebob->card->card_dev,
-	if (params == NULL)
+			      sizeof(struct special_params), GFP_KERNEL);
+	if (!params)
 		return -ENOMEM;
 
 	mutex_lock(&bebob->mutex);

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)
-{
-	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);
+	struct snd_dice *dice = card->private_data;
+
+	snd_dice_stream_destroy_duplex(dice);
+	snd_dice_transaction_destroy(dice);
 }
 
 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);
+	dice = devm_kzalloc(&unit->device, sizeof(struct snd_dice), GFP_KERNEL);
-	if (dice == NULL)
+	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)

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)
-{
-	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);
+	struct snd_dg00x *dg00x = card->private_data;
+
+	snd_dg00x_stream_destroy_duplex(dg00x);
+	snd_dg00x_transaction_unregister(dg00x);
 }
 
 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);
+	dg00x = devm_kzalloc(&unit->device, sizeof(struct snd_dg00x),
-	if (dg00x == NULL)
+			     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. */
+		// Block till all of ALSA character devices are released.
-		snd_card_free_when_closed(dg00x->card);
+		snd_card_free(dg00x->card);
-	} else {
-		/* Don't forget this case. */
-		dg00x_free(dg00x);
 	}
+
+	mutex_destroy(&dg00x->mutex);
+	fw_unit_put(dg00x->unit);
 }
 
 static const struct ieee1394_device_id snd_dg00x_id_table[] = {

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)
-{
-	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);
+	struct snd_ff *ff = card->private_data;
+
+	snd_ff_stream_destroy_duplex(ff);
+	snd_ff_transaction_unregister(ff);
 }
 
 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);
+	ff = devm_kzalloc(&unit->device, sizeof(struct snd_ff), GFP_KERNEL);
-	if (ff == NULL)
+	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. */
+		// Block till all of ALSA character devices are released.
-		snd_card_free_when_closed(ff->card);
+		snd_card_free(ff->card);
-	} else {
-		/* Don't forget this case. */
-		ff_free(ff);
 	}
+
+	mutex_destroy(&ff->mutex);
+	fw_unit_put(ff->unit);
 }
 
 static const struct snd_ff_spec spec_ff400 = {

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);
-		mutex_lock(&devices_mutex);
+	clear_bit(efw->card_index, devices_used);
-		clear_bit(efw->card_index, devices_used);
+	mutex_unlock(&devices_mutex);
-		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);
+	efw->resp_buf = devm_kzalloc(&efw->card->card_dev,
-	if (efw->resp_buf == NULL) {
+				     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. */
+		// Block till all of ALSA character devices are released.
-		snd_card_free_when_closed(efw->card);
+		snd_card_free(efw->card);
-	} else {
-		/* Don't forget this case. */
-		efw_free(efw);
 	}
+
+	mutex_destroy(&efw->mutex);
+	fw_unit_put(efw->unit);
 }
 
 static const struct ieee1394_device_id efw_id_table[] = {

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[] = {

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)
-{
-	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);
+	struct snd_motu *motu = card->private_data;
+
+	snd_motu_transaction_unregister(motu);
+	snd_motu_stream_destroy_duplex(motu);
 }
 
 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);
+	motu = devm_kzalloc(&unit->device, sizeof(struct snd_motu), GFP_KERNEL);
-	if (motu == NULL)
+	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. */
+		// Block till all of ALSA character devices are released.
-		snd_card_free_when_closed(motu->card);
+		snd_card_free(motu->card);
-	} else {
-		/* Don't forget this case. */
-		motu_free(motu);
 	}
+
+	mutex_destroy(&motu->mutex);
+	fw_unit_put(motu->unit);
 }
 
 static void motu_bus_update(struct fw_unit *unit)

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);
+	scs = devm_kzalloc(&oxfw->card->card_dev, sizeof(struct fw_scs1x),
-	if (scs == NULL)
+			   GFP_KERNEL);
+	if (!scs)
 		return -ENOMEM;
 	scs->fw_dev = fw_parent_device(oxfw->unit);
 	oxfw->spec = scs;

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);
+	spkr = devm_kzalloc(&oxfw->card->card_dev, sizeof(struct fw_spkr),
-	if (spkr == NULL)
+			    GFP_KERNEL);
+	if (!spkr)
 		return -ENOMEM;
 	oxfw->spec = spkr;
 

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);
+	formats[eid] = devm_kmemdup(&oxfw->card->card_dev, buf, *len,
-	if (formats[eid] == NULL) {
+				    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);
+		formats[eid] = devm_kmemdup(&oxfw->card->card_dev, buf, len,
-		if (formats[eid] == NULL) {
+					    GFP_KERNEL);
+		if (!formats[eid]) {
 			err = -ENOMEM;
 			break;
 		}

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);
+	oxfw = devm_kzalloc(&unit->device, sizeof(struct snd_oxfw), GFP_KERNEL);
-	if (oxfw == NULL)
+	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. */
+		// Block till all of ALSA character devices are released.
-		snd_card_free_when_closed(oxfw->card);
+		snd_card_free(oxfw->card);
-	} else {
-		/* Don't forget this case. */
-		oxfw_free(oxfw);
 	}
+
+	mutex_destroy(&oxfw->mutex);
+	fw_unit_put(oxfw->unit);
 }
 
 static const struct compat_info griffin_firewave = {

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)
-{
-	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);
+	struct snd_tscm *tscm = card->private_data;
+
+	snd_tscm_transaction_unregister(tscm);
+	snd_tscm_stream_destroy_duplex(tscm);
 }
 
 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);
+	tscm = devm_kzalloc(&unit->device, sizeof(struct snd_tscm), GFP_KERNEL);
-	if (tscm == NULL)
+	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. */
+		// Block till all of ALSA character devices are released.
-		snd_card_free_when_closed(tscm->card);
+		snd_card_free(tscm->card);
-	} else {
-		/* Don't forget this case. */
-		tscm_free(tscm);
 	}
+
+	mutex_destroy(&tscm->mutex);
+	fw_unit_put(tscm->unit);
 }
 
 static const struct ieee1394_device_id snd_tscm_id_table[] = {

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,
+		snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL,
-				AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA);
+				 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;

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))

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 |

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)

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:

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",

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);

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);

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,

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,

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"
 

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

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);

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>

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,
+	ret = snd_pcm_lib_malloc_pages(substream,
-					  params_buffer_bytes(hw_params));
+				       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++;
 		}

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
+static inline bool azx_snoop(struct azx *chip)
-#define azx_snoop(chip)		((chip)->snoop)
+{
-#else
+	return !IS_ENABLED(CONFIG_X86) || chip->snoop;
-#define azx_snoop(chip)		true
+}
-#endif
 
 /*
  * macros for easy use

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 {

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"

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>

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,
+	if (!azx_snoop(chip) && type == SNDRV_DMA_TYPE_DEV)
-				  bus->dev,
+		type = SNDRV_DMA_TYPE_DEV_UC;
-				  size, buf);
+	return snd_dma_alloc_pages(type, bus->dev, size, buf);
-	if (err < 0)
-		return err;
-	mark_pages_wc(chip, buf, true);
-	return 0;
 }
 
 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 */

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"

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;

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>

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 = {
+static const struct hda_controller_ops hda_tegra_ops; /* nothing special */
-	.substream_alloc_pages = substream_alloc_pages,
-	.substream_free_pages = substream_free_pages,
-};
 
 static void hda_tegra_init(struct hda_tegra *hda)
 {

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"

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"

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"

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"

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),

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"
 

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"
@@ -6841,6 +6841,12 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x1a, 0x02a11040},
 		{0x1b, 0x01014020},
 		{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},
@@ -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"},

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 */

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,

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"

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->bdbars.area)
-		if (chip->fix_nocache)
-			fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 0);
 		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];

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;

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))

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) {

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",

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;
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		rtd->capture_channel = pinfo->capture_channel;
+			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);
+		if (bytescount < period_bytes) {
-		runtime->delay = bytes_to_frames(runtime, delay);
+			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);

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;
 };
 

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

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

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);
 }
 
 /**

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");