Merge tag 'sound-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into next

Pull sound updates from Takashi Iwai:
 "At this time, majority of changes come from ASoC world while we got a
  few new drivers in other places for FireWire and USB.  There have been
  lots of ASoC core cleanups / refactoring, but very little visible to
  external users.

  ASoC:
   - Support for specifying aux CODECs in DT
   - Removal of the deprecated mux and enum macros
   - More moves towards full componentisation
   - Removal of some unused I/O code
   - Lots of cleanups, fixes and enhancements to the davinci, Freescale,
     Haswell and Realtek drivers
   - Several drivers exposed directly in Kconfig for use with
     simple-card
   - GPIO descriptor support for jacks
   - More updates and fixes to the Freescale SSI, Intel and rsnd drivers
   - New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651
     and ST STA350, Analog Devices ADAU1361, ADAU1381, ADAU1761 and
     ADAU1781, and Realtek RT5677

  HD-audio:
   - Clean up Dell headset quirks
   - Noise fixes for Dell and Sony laptops
   - Thinkpad T440 dock fix
   - Realtek codec updates (ALC293,ALC233,ALC3235)
   - Tegra HD-audio HDMI support

  FireWire-audio:
   - FireWire audio stack enhancement (AMDTP, MIDI), support for
     incoming isochronous stream and duplex streams with timestamp
     synchronization
   - BeBoB-based devices support
   - Fireworks-based device support

  USB-audio:
   - Behringer BCD2000 USB device support

  Misc:
   - Clean up of a few old drivers, atmel, fm801, etc"

* tag 'sound-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (480 commits)
  ASoC: Fix wrong argument for card remove callbacks
  ASoC: free jack GPIOs before the sound card is freed
  ALSA: firewire-lib: Remove a comment about restriction of asynchronous operation
  ASoC: cache: Fix error code when not using ASoC level cache
  ALSA: hda/realtek - Fix COEF widget NID for ALC260 replacer fixup
  ALSA: hda/realtek - Correction of fixup codes for PB V7900 laptop
  ALSA: firewire-lib: Use IEC 61883-6 compliant labels for Raw Audio data
  ASoC: add RT5677 CODEC driver
  ASoC: intel: The Baytrail/MAX98090 driver depends on I2C
  ASoC: rt5640: Add the function "get_clk_info" to RL6231 shared support
  ASoC: rt5640: Add the function of the PLL clock calculation to RL6231 shared support
  ASoC: rt5640: Add RL6231 class device shared support for RT5640, RT5645 and RT5651
  ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error.
  ASoC: Add helper functions to cast from DAPM context to CODEC/platform
  ALSA: bebob: sizeof() vs ARRAY_SIZE() typo
  ASoC: wm9713: correct mono out PGA sources
  ALSA: synth: emux: soundfont.c: Cleaning up memory leak
  ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320
  ASoC: fsl-ssi: Use regmap
  ASoC: fsl-ssi: reorder and document fsl_ssi_private
  ...

Documentation/devicetree/bindings/mfd/mc13xxx.txt

@@ -10,6 +10,9 @@ Optional properties:
 - fsl,mc13xxx-uses-touch : Indicate the touchscreen controller is being used
 
 Sub-nodes:
+- codec: Contain the Audio Codec node.
+  - adc-port: Contain PMIC SSI port number used for ADC.
+  - dac-port: Contain PMIC SSI port number used for DAC.
 - leds : Contain the led nodes and initial register values in property
   "led-control". Number of register depends of used IC, for MC13783 is 6,
   for MC13892 is 4, for MC34708 is 1. See datasheet for bits definitions of

Documentation/devicetree/bindings/sound/ak4104.txt

@@ -8,6 +8,8 @@ Required properties:
 
   - reg : The chip select number on the SPI bus
 
+  - vdd-supply : A regulator node, providing 2.7V - 3.6V
+
 Optional properties:
 
   - reset-gpio : a GPIO spec for the reset pin. If specified, it will be
@@ -19,4 +21,5 @@ spdif: ak4104@0 {
 	compatible = "asahi-kasei,ak4104";
 	reg = <0>;
 	spi-max-frequency = <5000000>;
+	vdd-supply = <&vdd_3v3_reg>;
 };

Documentation/devicetree/bindings/sound/alc5623.txt

+ALC5621/ALC5622/ALC5623 audio Codec
+
+Required properties:
+
+ - compatible:	"realtek,alc5623"
+ - reg:		the I2C address of the device.
+
+Optional properties:
+
+ - add-ctrl:	  Default register value for Reg-40h, Additional Control
+		  Register. If absent or has the value of 0, the
+		  register is untouched.
+
+ - jack-det-ctrl: Default register value for Reg-5Ah, Jack Detect
+		  Control Register. If absent or has value 0, the
+		  register is untouched.
+
+Example:
+
+	alc5621: alc5621@1a {
+		compatible = "alc5621";
+		reg = <0x1a>;
+		add-ctrl = <0x3700>;
+		jack-det-ctrl = <0x4810>;
+	};

Documentation/devicetree/bindings/sound/cs42l56.txt

+CS42L52 audio CODEC
+
+Required properties:
+
+  - compatible : "cirrus,cs42l56"
+
+  - reg : the I2C address of the device for I2C
+
+  - VA-supply, VCP-supply, VLDO-supply : power supplies for the device,
+  as covered in Documentation/devicetree/bindings/regulator/regulator.txt.
+
+Optional properties:
+
+  - cirrus,gpio-nreset : GPIO controller's phandle and the number
+  of the GPIO used to reset the codec.
+
+  - cirrus,chgfreq-divisor : Values used to set the Charge Pump Frequency.
+  Allowable values of 0x00 through 0x0F. These are raw values written to the
+  register, not the actual frequency. The frequency is determined by the following.
+  Frequency = MCLK / 4 * (N+2)
+  N = chgfreq_val
+  MCLK = Where MCLK is the frequency of the mclk signal after the MCLKDIV2 circuit.
+
+  - cirrus,ain1a-ref-cfg, ain1b-ref-cfg : boolean, If present, AIN1A or AIN1B are configured
+  as a pseudo-differential input referenced to AIN1REF/AIN3A.
+
+  - cirrus,ain2a-ref-cfg, ain2b-ref-cfg : boolean, If present, AIN2A or AIN2B are configured
+  as a pseudo-differential input referenced to AIN2REF/AIN3B.
+
+  - cirrus,micbias-lvl: Set the output voltage level on the MICBIAS Pin.
+  0 = 0.5 x VA
+  1 = 0.6 x VA
+  2 = 0.7 x VA
+  3 = 0.8 x VA
+  4 = 0.83 x VA
+  5 = 0.91 x VA
+
+  - cirrus,adaptive-pwr-cfg : Configures how the power to the Headphone and Lineout
+  Amplifiers adapt to the output signal levels.
+  0 = Adapt to Volume Mode. Voltage level determined by the sum of the relevant volume settings.
+  1 = Fixed - Headphone and Line Amp supply = + or - VCP/2.
+  2 = Fixed - Headphone and Line Amp supply = + or - VCP.
+  3 = Adapted to Signal; Voltage level is dynamically determined by the output signal.
+
+  - cirrus,hpf-left-freq, hpf-right-freq : Sets the corner frequency (-3dB point) for the internal High-Pass
+  Filter.
+  0 = 1.8Hz
+  1 = 119Hz
+  2 = 236Hz
+  3 = 464Hz
+
+
+Example:
+
+codec: codec@4b {
+	compatible = "cirrus,cs42l56";
+	reg = <0x4b>;
+	gpio-reset = <&gpio 10 0>;
+	cirrus,chgfreq-divisor = <0x05>;
+	cirrus.ain1_ref_cfg;
+	cirrus,micbias-lvl = <5>;
+	VA-supply = <&reg_audio>;
+};

Documentation/devicetree/bindings/sound/fsl-sai.txt

@@ -7,10 +7,11 @@ codec/DSP interfaces.
 
 
 Required properties:
-- compatible: Compatible list, contains "fsl,vf610-sai".
+- compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai".
 - reg: Offset and length of the register set for the device.
 - clocks: Must contain an entry for each entry in clock-names.
-- clock-names : Must include the "sai" entry.
+- clock-names : Must include the "bus" for register access and "mclk1" "mclk2"
+  "mclk3" for bit clock and frame clock providing.
 - dmas : Generic dma devicetree binding as described in
   Documentation/devicetree/bindings/dma/dma.txt.
 - dma-names : Two dmas have to be defined, "tx" and "rx".
@@ -30,8 +31,10 @@ sai2: sai@40031000 {
 	      reg = <0x40031000 0x1000>;
 	      pinctrl-names = "default";
 	      pinctrl-0 = <&pinctrl_sai2_1>;
-	      clocks = <&clks VF610_CLK_SAI2>;
-	      clock-names = "sai";
+	      clocks = <&clks VF610_CLK_PLATFORM_BUS>,
+		     <&clks VF610_CLK_SAI2>,
+		     <&clks 0>, <&clks 0>;
+	      clock-names = "bus", "mclk1", "mclk2", "mclk3";
 	      dma-names = "tx", "rx";
 	      dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>,
 		   <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>;

Documentation/devicetree/bindings/sound/max98090.txt

@@ -10,6 +10,12 @@ Required properties:
 
 - interrupts : The CODEC's interrupt output.
 
+Optional properties:
+
+- clocks: The phandle of the master clock to the CODEC
+
+- clock-names: Should be "mclk"
+
 Pins on the device (for linking into audio routes):
 
   * MIC1

Documentation/devicetree/bindings/sound/max98095.txt

+MAX98095 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "maxim,max98095".
+
+- reg : The I2C address of the device.
+
+Optional properties:
+
+- clocks: The phandle of the master clock to the CODEC
+
+- clock-names: Should be "mclk"
+
+Example:
+
+max98095: codec@11 {
+	compatible = "maxim,max98095";
+	reg = <0x11>;
+};

Documentation/devicetree/bindings/sound/nokia,rx51.txt

+* Nokia N900 audio setup
+
+Required properties:
+- compatible: Should contain "nokia,n900-audio"
+- nokia,cpu-dai: phandle for the McBSP node
+- nokia,audio-codec: phandles for the main TLV320AIC3X node and the
+                     auxiliary TLV320AIC3X node (in this order)
+- nokia,headphone-amplifier: phandle for the TPA6130A2 node
+- tvout-selection-gpios: GPIO for tvout selection
+- jack-detection-gpios: GPIO for jack detection
+- eci-switch-gpios: GPIO for ECI (Enhancement Control Interface) switch
+- speaker-amplifier-gpios: GPIO for speaker amplifier
+
+Example:
+
+sound {
+	compatible = "nokia,n900-audio";
+
+	nokia,cpu-dai = <&mcbsp2>;
+	nokia,audio-codec = <&tlv320aic3x>, <&tlv320aic3x_aux>;
+	nokia,headphone-amplifier = <&tpa6130a2>;
+
+	tvout-selection-gpios = <&gpio2 8 GPIO_ACTIVE_HIGH>; /* 40 */
+	jack-detection-gpios = <&gpio6 17 GPIO_ACTIVE_HIGH>; /* 177 */
+	eci-switch-gpios = <&gpio6 22 GPIO_ACTIVE_HIGH>; /* 182 */
+	speaker-amplifier-gpios = <&twl_gpio 7 GPIO_ACTIVE_HIGH>;
+};

Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt

+NVIDIA Tegra30 HDA controller
+
+Required properties:
+- compatible : "nvidia,tegra30-hda"
+- reg : Should contain the HDA registers location and length.
+- interrupts : The interrupt from the HDA controller.
+- clocks : Must contain an entry for each required entry in clock-names.
+  See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries: hda, hdacodec_2x, hda2hdmi
+- resets : Must contain an entry for each entry in reset-names.
+  See ../reset/reset.txt for details.
+- reset-names : Must include the following entries: hda, hdacodec_2x, hda2hdmi
+
+Example:
+
+hda@0,70030000 {
+	compatible = "nvidia,tegra124-hda", "nvidia,tegra30-hda";
+	reg = <0x0 0x70030000 0x0 0x10000>;
+	interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>;
+	clocks = <&tegra_car TEGRA124_CLK_HDA>,
+		 <&tegra_car TEGRA124_CLK_HDA2HDMI>,
+		 <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>;
+	clock-names = "hda", "hda2hdmi", "hda2codec_2x";
+	resets = <&tegra_car 125>, /* hda */
+		 <&tegra_car 128>; /* hda2hdmi */
+		 <&tegra_car 111>, /* hda2codec_2x */
+	reset-names = "hda", "hda2hdmi", "hda2codec_2x";
+};

Documentation/devicetree/bindings/sound/renesas,rsnd.txt

@@ -20,6 +20,7 @@ Required properties:
 SSI subnode properties:
 - interrupts			: Should contain SSI interrupt for PIO transfer
 - shared-pin			: if shared clock pin
+- pio-transfer			: use PIO transfer mode
 
 SRC subnode properties:
 no properties at this point

Documentation/devicetree/bindings/sound/rt5640.txt

-RT5640 audio CODEC
+RT5640/RT5639 audio CODEC
 
 This device supports I2C only.
 
 Required properties:
 
-- compatible : "realtek,rt5640".
+- compatible : One of "realtek,rt5640" or "realtek,rt5639".
 
 - reg : The I2C address of the device.
 
@@ -18,7 +18,7 @@ Optional properties:
 
 - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
 
-Pins on the device (for linking into audio routes):
+Pins on the device (for linking into audio routes) for RT5639/RT5640:
 
   * DMIC1
   * DMIC2
@@ -31,13 +31,16 @@ Pins on the device (for linking into audio routes):
   * HPOR
   * LOUTL
   * LOUTR
-  * MONOP
-  * MONON
   * SPOLP
   * SPOLN
   * SPORP
   * SPORN
 
+Additional pins on the device for RT5640:
+
+  * MONOP
+  * MONON
+
 Example:
 
 rt5640 {

Documentation/devicetree/bindings/sound/simple-card.txt

 Simple-Card:
 
-Simple-Card specifies audio DAI connection of SoC <-> codec.
+Simple-Card specifies audio DAI connections of SoC <-> codec.
 
 Required properties:
 
@@ -10,26 +10,54 @@ Optional properties:
 
 - simple-audio-card,name		: User specified audio sound card name, one string
 					  property.
-- simple-audio-card,format		: CPU/CODEC common audio format.
-					  "i2s", "right_j", "left_j" , "dsp_a"
-					  "dsp_b", "ac97", "pdm", "msb", "lsb"
 - simple-audio-card,widgets		: Please refer to widgets.txt.
 - simple-audio-card,routing		: A list of the connections between audio components.
 					  Each entry is a pair of strings, the first being the
 					  connection's sink, the second being the connection's
 					  source.
-- dai-tdm-slot-num			: Please refer to tdm-slot.txt.
-- dai-tdm-slot-width			: Please refer to tdm-slot.txt.
+- simple-audio-card,mclk-fs             : Multiplication factor between stream rate and codec
+  					  mclk.
+
+Optional subnodes:
+
+- simple-audio-card,dai-link		: Container for dai-link level
+					  properties and the CPU and CODEC
+					  sub-nodes. This container may be
+					  omitted when the card has only one
+					  DAI link. See the examples and the
+					  section bellow.
+
+Dai-link subnode properties and subnodes:
+
+If dai-link subnode is omitted and the subnode properties are directly
+under "sound"-node the subnode property and subnode names have to be
+prefixed with "simple-audio-card,"-prefix.
 
-Required subnodes:
+Required dai-link subnodes:
 
-- simple-audio-card,dai-link		: container for the CPU and CODEC sub-nodes
-					  This container may be omitted when the
-					  card has only one DAI link.
-					  See the examples.
+- cpu					: CPU   sub-node
+- codec					: CODEC sub-node
 
-- simple-audio-card,cpu			: CPU   sub-node
-- simple-audio-card,codec		: CODEC sub-node
+Optional dai-link subnode properties:
+
+- format				: CPU/CODEC common audio format.
+					  "i2s", "right_j", "left_j" , "dsp_a"
+					  "dsp_b", "ac97", "pdm", "msb", "lsb"
+- frame-master				: Indicates dai-link frame master.
+					  phandle to a cpu or codec subnode.
+- bitclock-master			: Indicates dai-link bit clock master.
+					  phandle to a cpu or codec subnode.
+- bitclock-inversion			: bool property. Add this if the
+					  dai-link uses bit clock inversion.
+- frame-inversion			: bool property. Add this if the
+					  dai-link uses frame clock inversion.
+
+For backward compatibility the frame-master and bitclock-master
+properties can be used as booleans in codec subnode to indicate if the
+codec is the dai-link frame or bit clock master. In this case there
+should be no dai-link node, the same properties should not be present
+at sound-node level, and the bitclock-inversion and frame-inversion
+properties should also be placed in the codec node if needed.
 
 Required CPU/CODEC subnodes properties:
 
@@ -37,29 +65,21 @@ Required CPU/CODEC subnodes properties:
 
 Optional CPU/CODEC subnodes properties:
 
-- format				: CPU/CODEC specific audio format if needed.
-					  see simple-audio-card,format
-- frame-master				: bool property. add this if subnode is frame master
-- bitclock-master			: bool property. add this if subnode is bitclock master
-- bitclock-inversion			: bool property. add this if subnode has clock inversion
-- frame-inversion			: bool property. add this if subnode has frame inversion
+- dai-tdm-slot-num			: Please refer to tdm-slot.txt.
+- dai-tdm-slot-width			: Please refer to tdm-slot.txt.
 - clocks / system-clock-frequency	: specify subnode's clock if needed.
 					  it can be specified via "clocks" if system has
 					  clock node (= common clock), or "system-clock-frequency"
 					  (if system doens't support common clock)
 
-Note:
- * For 'format', 'frame-master', 'bitclock-master', 'bitclock-inversion' and
-   'frame-inversion', the simple card will use the settings of CODEC for both
-   CPU and CODEC sides as we need to keep the settings identical for both ends
-   of the link.
-
 Example 1 - single DAI link:
 
 sound {
 	compatible = "simple-audio-card";
 	simple-audio-card,name = "VF610-Tower-Sound-Card";
 	simple-audio-card,format = "left_j";
+	simple-audio-card,bitclock-master = <&dailink0_master>;
+	simple-audio-card,frame-master = <&dailink0_master>;
 	simple-audio-card,widgets =
 		"Microphone", "Microphone Jack",
 		"Headphone", "Headphone Jack",
@@ -69,17 +89,12 @@ sound {
 		"Headphone Jack", "HP_OUT",
 		"External Speaker", "LINE_OUT";
 
-	dai-tdm-slot-num = <2>;
-	dai-tdm-slot-width = <8>;
-
 	simple-audio-card,cpu {
 		sound-dai = <&sh_fsi2 0>;
 	};
 
-	simple-audio-card,codec {
+	dailink0_master: simple-audio-card,codec {
 		sound-dai = <&ak4648>;
-		bitclock-master;
-		frame-master;
 		clocks = <&osc>;
 	};
 };
@@ -105,31 +120,31 @@ Example 2 - many DAI links:
 sound {
 	compatible = "simple-audio-card";
 	simple-audio-card,name = "Cubox Audio";
-	simple-audio-card,format = "i2s";
 
 	simple-audio-card,dai-link@0 {		/* I2S - HDMI */
-		simple-audio-card,cpu {
+		format = "i2s";
+		cpu {
 			sound-dai = <&audio1 0>;
 		};
-		simple-audio-card,codec {
+		codec {
 			sound-dai = <&tda998x 0>;
 		};
 	};
 
 	simple-audio-card,dai-link@1 {		/* S/PDIF - HDMI */
-		simple-audio-card,cpu {
+		cpu {
 			sound-dai = <&audio1 1>;
 		};
-		simple-audio-card,codec {
+		codec {
 			sound-dai = <&tda998x 1>;
 		};
 	};
 
 	simple-audio-card,dai-link@2 {		/* S/PDIF - S/PDIF */
-		simple-audio-card,cpu {
+		cpu {
 			sound-dai = <&audio1 1>;
 		};
-		simple-audio-card,codec {
+		codec {
 			sound-dai = <&spdif_codec>;
 		};
 	};

Documentation/devicetree/bindings/sound/snow.txt

+Audio Binding for Snow boards
+
+Required properties:
+- compatible : Can be one of the following,
+			"google,snow-audio-max98090" or
+			"google,snow-audio-max98095"
+- samsung,i2s-controller: The phandle of the Samsung I2S controller
+- samsung,audio-codec: The phandle of the audio codec
+
+Example:
+
+sound {
+		compatible = "google,snow-audio-max98095";
+
+		samsung,i2s-controller = <&i2s0>;
+		samsung,audio-codec = <&max98095>;
+};

Documentation/devicetree/bindings/sound/st,sta350.txt

+STA350 audio CODEC
+
+The driver for this device only supports I2C.
+
+Required properties:
+
+  - compatible: "st,sta350"
+  - reg: the I2C address of the device for I2C
+  - reset-gpios: a GPIO spec for the reset pin. If specified, it will be
+		 deasserted before communication to the codec starts.
+
+  - power-down-gpios: a GPIO spec for the power down pin. If specified,
+		      it will be deasserted before communication to the codec
+		      starts.
+
+  - vdd-dig-supply: regulator spec, providing 3.3V
+  - vdd-pll-supply: regulator spec, providing 3.3V
+  - vcc-supply: regulator spec, providing 5V - 26V
+
+Optional properties:
+
+  -  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
+	2: 2 Channel (Full-Bridge) Power, 1 Channel FFX
+	3: 1 Channel Mono-Parallel
+	If parameter is missing, mode 0 will be enabled.
+	This property has to be specified as '/bits/ 8' value.
+
+  -  st,ch1-output-mapping: Channel 1 output mapping
+  -  st,ch2-output-mapping: Channel 2 output mapping
+  -  st,ch3-output-mapping: Channel 3 output mapping
+	0: Channel 1
+	1: Channel 2
+	2: Channel 3
+	If parameter is missing, channel 1 is choosen.
+	This properties have to be specified as '/bits/ 8' values.
+
+  -  st,thermal-warning-recover:
+	If present, thermal warning recovery is enabled.
+
+  -  st,thermal-warning-adjustment:
+	If present, thermal warning adjustment is enabled.
+
+  -  st,fault-detect-recovery:
+	If present, then fault recovery will be enabled.
+
+  -  st,ffx-power-output-mode: string
+	The FFX power output mode selects how the FFX output timing is
+	configured. Must be one of these values:
+	  -  "drop-compensation"
+	  -  "tapered-compensation"
+	  -  "full-power-mode"
+	  -  "variable-drop-compensation" (default)
+
+  -  st,drop-compensation-ns: number
+	Only required for "st,ffx-power-output-mode" ==
+	"variable-drop-compensation".
+	Specifies the drop compensation in nanoseconds.
+	The value must be in the range of 0..300, and only
+	multiples of 20 are allowed. Default is 140ns.
+
+  -  st,overcurrent-warning-adjustment:
+	If present, overcurrent warning adjustment is enabled.
+
+  -  st,max-power-use-mpcc:
+	If present, then MPCC bits are used for MPC coefficients,
+	otherwise standard MPC coefficients are used.
+
+  -  st,max-power-corr:
+	If present, power bridge correction for THD reduction near maximum
+	power output is enabled.
+
+  -  st,am-reduction-mode:
+	If present, FFX mode runs in AM reduction mode, otherwise normal
+	FFX mode is used.
+
+  -  st,odd-pwm-speed-mode:
+	If present, PWM speed mode run on odd speed mode (341.3 kHz) on all
+	channels. If not present, normal PWM spped mode (384 kHz) will be used.
+
+  -  st,distortion-compensation:
+	If present, distortion compensation variable uses DCC coefficient.
+	If not present, preset DC coefficient is used.
+
+  -  st,invalid-input-detect-mute:
+	If present, automatic invalid input detect mute is enabled.
+
+  -  st,activate-mute-output:
+	If present, a mute output will be activated in ase the volume will
+	reach a value lower than -76 dBFS.
+
+  -  st,bridge-immediate-off:
+	If present, the bridge will be switched off immediately after the
+	power-down-gpio goes low. Otherwise, the bridge will wait for 13
+	million clock cycles to pass before shutting down.
+
+  -  st,noise-shape-dc-cut:
+	If present, the noise-shaping technique on the DC cutoff filter are
+	enabled.
+
+  -  st,powerdown-master-volume:
+	If present, the power-down pin and I2C power-down functions will
+	act on the master volume. Otherwise, the functions will act on the
+	mute commands.
+
+  -  st,powerdown-delay-divider:
+	If present, the bridge power-down time will be divided by the provided
+	value. If not specified, a divider of 1 will be used. Allowed values
+	are 1, 2, 4, 8, 16, 32, 64 and 128.
+	This property has to be specified as '/bits/ 8' value.
+
+Example:
+
+codec: sta350@38 {
+	compatible = "st,sta350";
+	reg = <0x1c>;
+	reset-gpios = <&gpio1 19 0>;
+	power-down-gpios = <&gpio1 16 0>;
+	st,output-conf = /bits/ 8  <0x3>;	// set output to 2-channel
+						// (full-bridge) power,
+						// 2-channel data-out
+	st,ch1-output-mapping = /bits/ 8 <0>;	// set channel 1 output ch 1
+	st,ch2-output-mapping = /bits/ 8 <0>;	// set channel 2 output ch 1
+	st,ch3-output-mapping = /bits/ 8 <0>;	// set channel 3 output ch 1
+	st,max-power-correction;		// enables power bridge
+						// correction for THD reduction
+						// near maximum power output
+	st,invalid-input-detect-mute;		// mute if no valid digital
+						// audio signal is provided.
+};

MAINTAINERS

@@ -8270,6 +8270,7 @@ L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 W:	http://www.alsa-project.org/
 T:	git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
 T:	git git://git.alsa-project.org/alsa-kernel.git
+Q:	http://patchwork.kernel.org/project/alsa-devel/list/
 S:	Maintained
 F:	Documentation/sound/
 F:	include/sound/

arch/mips/include/asm/mach-jz4740/dma.h

@@ -22,8 +22,6 @@ enum jz4740_dma_request_type {
 	JZ4740_DMA_TYPE_UART_RECEIVE	= 21,
 	JZ4740_DMA_TYPE_SPI_TRANSMIT	= 22,
 	JZ4740_DMA_TYPE_SPI_RECEIVE	= 23,
-	JZ4740_DMA_TYPE_AIC_TRANSMIT	= 24,
-	JZ4740_DMA_TYPE_AIC_RECEIVE	= 25,
 	JZ4740_DMA_TYPE_MMC_TRANSMIT	= 26,
 	JZ4740_DMA_TYPE_MMC_RECEIVE	= 27,
 	JZ4740_DMA_TYPE_TCU		= 28,

arch/mips/jz4740/board-qi_lb60.c

@@ -425,6 +425,15 @@ static struct platform_device qi_lb60_audio_device = {
 	.id = -1,
 };
 
+static struct gpiod_lookup_table qi_lb60_audio_gpio_table = {
+	.dev_id = "qi-lb60-audio",
+	.table = {
+		GPIO_LOOKUP("Bank B", 29, "snd", 0),
+		GPIO_LOOKUP("Bank D", 4, "amp", 0),
+		{ },
+	},
+};
+
 static struct platform_device *jz_platform_devices[] __initdata = {
 	&jz4740_udc_device,
 	&jz4740_udc_xceiv_device,
@@ -461,6 +470,8 @@ static int __init qi_lb60_init_platform_devices(void)
 	jz4740_adc_device.dev.platform_data = &qi_lb60_battery_pdata;
 	jz4740_mmc_device.dev.platform_data = &qi_lb60_mmc_pdata;
 
+	gpiod_add_lookup_table(&qi_lb60_audio_gpio_table);
+
 	jz4740_serial_device_register();
 
 	spi_register_board_info(qi_lb60_spi_board_info,

drivers/firewire/core.h

@@ -118,7 +118,6 @@ int fw_card_add(struct fw_card *card,
 		u32 max_receive, u32 link_speed, u64 guid);
 void fw_core_remove_card(struct fw_card *card);
 int fw_compute_block_crc(__be32 *block);
-void fw_schedule_bus_reset(struct fw_card *card, bool delayed, bool short_reset);
 void fw_schedule_bm_work(struct fw_card *card, unsigned long delay);
 
 /* -cdev */

drivers/mfd/mc13xxx-core.c

@@ -673,9 +673,13 @@ int mc13xxx_common_init(struct device *dev)
 	if (mc13xxx->flags & MC13XXX_USE_ADC)
 		mc13xxx_add_subdevice(mc13xxx, "%s-adc");
 
-	if (mc13xxx->flags & MC13XXX_USE_CODEC)
-		mc13xxx_add_subdevice_pdata(mc13xxx, "%s-codec",
-					pdata->codec, sizeof(*pdata->codec));
+	if (mc13xxx->flags & MC13XXX_USE_CODEC) {
+		if (pdata)
+			mc13xxx_add_subdevice_pdata(mc13xxx, "%s-codec",
+				pdata->codec, sizeof(*pdata->codec));
+		else
+			mc13xxx_add_subdevice(mc13xxx, "%s-codec");
+	}
 
 	if (mc13xxx->flags & MC13XXX_USE_RTC)
 		mc13xxx_add_subdevice(mc13xxx, "%s-rtc");

include/linux/firewire.h

@@ -367,6 +367,9 @@ static inline int fw_stream_packet_destination_id(int tag, int channel, int sy)
 	return tag << 14 | channel << 8 | sy;
 }
 
+void fw_schedule_bus_reset(struct fw_card *card, bool delayed,
+			   bool short_reset);
+
 struct fw_descriptor {
 	struct list_head link;
 	size_t length;

include/linux/platform_data/adau17x1.h

+/*
+ * Driver for ADAU1761/ADAU1461/ADAU1761/ADAU1961/ADAU1781/ADAU1781 codecs
+ *
+ * Copyright 2011-2014 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef __LINUX_PLATFORM_DATA_ADAU17X1_H__
+#define __LINUX_PLATFORM_DATA_ADAU17X1_H__
+
+/**
+ * enum adau17x1_micbias_voltage - Microphone bias voltage
+ * @ADAU17X1_MICBIAS_0_90_AVDD: 0.9 * AVDD
+ * @ADAU17X1_MICBIAS_0_65_AVDD: 0.65 * AVDD
+ */
+enum adau17x1_micbias_voltage {
+	ADAU17X1_MICBIAS_0_90_AVDD = 0,
+	ADAU17X1_MICBIAS_0_65_AVDD = 1,
+};
+
+/**
+ * enum adau1761_digmic_jackdet_pin_mode - Configuration of the JACKDET/MICIN pin
+ * @ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: Disable the pin
+ * @ADAU1761_DIGMIC_JACKDET_PIN_MODE_DIGMIC: Configure the pin for usage as
+ *   digital microphone input.
+ * @ADAU1761_DIGMIC_JACKDET_PIN_MODE_JACKDETECT: Configure the pin for jack
+ *  insertion detection.
+ */
+enum adau1761_digmic_jackdet_pin_mode {
+	ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE,
+	ADAU1761_DIGMIC_JACKDET_PIN_MODE_DIGMIC,
+	ADAU1761_DIGMIC_JACKDET_PIN_MODE_JACKDETECT,
+};
+
+/**
+ * adau1761_jackdetect_debounce_time - Jack insertion detection debounce time
+ * @ADAU1761_JACKDETECT_DEBOUNCE_5MS: 5 milliseconds
+ * @ADAU1761_JACKDETECT_DEBOUNCE_10MS: 10 milliseconds
+ * @ADAU1761_JACKDETECT_DEBOUNCE_20MS: 20 milliseconds
+ * @ADAU1761_JACKDETECT_DEBOUNCE_40MS: 40 milliseconds
+ */
+enum adau1761_jackdetect_debounce_time {
+	ADAU1761_JACKDETECT_DEBOUNCE_5MS = 0,
+	ADAU1761_JACKDETECT_DEBOUNCE_10MS = 1,
+	ADAU1761_JACKDETECT_DEBOUNCE_20MS = 2,
+	ADAU1761_JACKDETECT_DEBOUNCE_40MS = 3,
+};
+
+/**
+ * enum adau1761_output_mode - Output mode configuration
+ * @ADAU1761_OUTPUT_MODE_HEADPHONE: Headphone output
+ * @ADAU1761_OUTPUT_MODE_HEADPHONE_CAPLESS: Capless headphone output
+ * @ADAU1761_OUTPUT_MODE_LINE: Line output
+ */
+enum adau1761_output_mode {
+	ADAU1761_OUTPUT_MODE_HEADPHONE,
+	ADAU1761_OUTPUT_MODE_HEADPHONE_CAPLESS,
+	ADAU1761_OUTPUT_MODE_LINE,
+};
+
+/**
+ * struct adau1761_platform_data - ADAU1761 Codec driver platform data
+ * @input_differential: If true the input pins will be configured in
+ *  differential mode.
+ * @lineout_mode: Output mode for the LOUT/ROUT pins
+ * @headphone_mode: Output mode for the LHP/RHP pins
+ * @digmic_jackdetect_pin_mode: JACKDET/MICIN pin configuration
+ * @jackdetect_debounce_time: Jack insertion detection debounce time.
+ *  Note: This value will only be used, if the JACKDET/MICIN pin is configured
+ *  for jack insertion detection.
+ * @jackdetect_active_low: If true the jack insertion detection is active low.
+ *  Othwise it will be active high.
+ * @micbias_voltage: Microphone voltage bias
+ */
+struct adau1761_platform_data {
+	bool input_differential;
+	enum adau1761_output_mode lineout_mode;
+	enum adau1761_output_mode headphone_mode;
+
+	enum adau1761_digmic_jackdet_pin_mode digmic_jackdetect_pin_mode;
+
+	enum adau1761_jackdetect_debounce_time jackdetect_debounce_time;
+	bool jackdetect_active_low;
+
+	enum adau17x1_micbias_voltage micbias_voltage;
+};
+
+/**
+ * struct adau1781_platform_data - ADAU1781 Codec driver platform data
+ * @left_input_differential: If true configure the left input as
+ * differential input.
+ * @right_input_differential: If true configure the right input as differntial
+ *  input.
+ * @use_dmic: If true configure the MIC pins as digital microphone pins instead
+ *  of analog microphone pins.
+ * @micbias_voltage: Microphone voltage bias
+ */
+struct adau1781_platform_data {
+	bool left_input_differential;
+	bool right_input_differential;
+
+	bool use_dmic;
+
+	enum adau17x1_micbias_voltage micbias_voltage;
+};
+
+#endif

include/sound/atmel-ac97c.h

@@ -23,7 +23,6 @@
  * @reset_pin: GPIO pin wired to the reset input on the external AC97 codec,
  *             optional to use, set to -ENODEV if not in use. AC97 layer will
  *             try to do a software reset of the external codec anyway.
- * @flags: Flags for which directions should be enabled.
  *
  * If the user do not want to use a DMA channel for playback or capture, i.e.
  * only one feature is required on the board. The slave for playback or capture
@@ -33,7 +32,6 @@
 struct ac97c_platform_data {
 	struct dw_dma_slave	rx_dws;
 	struct dw_dma_slave	tx_dws;
-	unsigned int 		flags;
 	int			reset_pin;
 };
 

include/sound/core.h

@@ -282,13 +282,6 @@ int snd_card_new(struct device *parent, int idx, const char *xid,
 		 struct module *module, int extra_size,
 		 struct snd_card **card_ret);
 
-static inline int __deprecated
-snd_card_create(int idx, const char *id, struct module *module, int extra_size,
-		struct snd_card **ret)
-{
-	return snd_card_new(NULL, idx, id, module, extra_size, ret);
-}
-
 int snd_card_disconnect(struct snd_card *card);
 int snd_card_free(struct snd_card *card);
 int snd_card_free_when_closed(struct snd_card *card);

include/sound/cs42l56.h

+/*
+ * linux/sound/cs42l56.h -- Platform data for CS42L56
+ *
+ * Copyright (c) 2014 Cirrus Logic Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __CS42L56_H
+#define __CS42L56_H
+
+struct cs42l56_platform_data {
+
+	/* GPIO for Reset */
+	unsigned int gpio_nreset;
+
+	/* MICBIAS Level. Check datasheet Pg48 */
+	unsigned int micbias_lvl;
+
+	/* Analog Input 1A Reference 0=Single 1=Pseudo-Differential */
+	unsigned int ain1a_ref_cfg;
+
+	/* Analog Input 2A Reference 0=Single 1=Pseudo-Differential */
+	unsigned int ain2a_ref_cfg;
+
+	/* Analog Input 1B Reference 0=Single 1=Pseudo-Differential */
+	unsigned int ain1b_ref_cfg;
+
+	/* Analog Input 2B Reference 0=Single 1=Pseudo-Differential */
+	unsigned int ain2b_ref_cfg;
+
+	/* Charge Pump Freq. Check datasheet Pg62 */
+	unsigned int chgfreq;
+
+	/* HighPass Filter Right Channel Corner Frequency */
+	unsigned int hpfb_freq;
+
+	/* HighPass Filter Left Channel Corner Frequency */
+	unsigned int hpfa_freq;
+
+	/* Adaptive Power Control for LO/HP */
+	unsigned int adaptive_pwr;
+
+};
+
+#endif /* __CS42L56_H */

include/sound/omap-pcm.h

+/*
+ * omap-pcm.h - OMAP PCM driver
+ *
+ * Copyright (C) 2014 Texas Instruments, Inc.
+ *
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ */
+
+#ifndef __OMAP_PCM_H__
+#define __OMAP_PCM_H__
+
+#if IS_ENABLED(CONFIG_SND_OMAP_SOC)
+int omap_pcm_platform_register(struct device *dev);
+#else
+static inline int omap_pcm_platform_register(struct device *dev)
+{
+	return 0;
+}
+#endif /* CONFIG_SND_OMAP_SOC */
+
+#endif /* __OMAP_PCM_H__ */

include/sound/rcar_snd.h

@@ -34,47 +34,39 @@
  * B : SSI direction
  */
 #define RSND_SSI_CLK_PIN_SHARE		(1 << 31)
-#define RSND_SSI_PLAY			(1 << 24)
 
 #define RSND_SSI(_dma_id, _pio_irq, _flags)		\
 { .dma_id = _dma_id, .pio_irq = _pio_irq, .flags = _flags }
-#define RSND_SSI_SET(_dai_id, _dma_id, _pio_irq, _flags)	\
-{ .dai_id = _dai_id, .dma_id = _dma_id, .pio_irq = _pio_irq, .flags = _flags }
 #define RSND_SSI_UNUSED \
-{ .dai_id = -1, .dma_id = -1, .pio_irq = -1, .flags = 0 }
+{ .dma_id = -1, .pio_irq = -1, .flags = 0 }
 
 struct rsnd_ssi_platform_info {
-	int dai_id;	/* will be removed */
 	int dma_id;
 	int pio_irq;
 	u32 flags;
 };
 
-/*
- * flags
- */
-#define RSND_SCU_USE_HPBIF		(1 << 31) /* it needs RSND_SSI_DEPENDENT */
-
 #define RSND_SRC(rate, _dma_id)						\
-{ .flags = RSND_SCU_USE_HPBIF, .convert_rate = rate, .dma_id = _dma_id, }
-#define RSND_SRC_SET(rate, _dma_id)		\
-	{ .flags = RSND_SCU_USE_HPBIF, .convert_rate = rate, .dma_id = _dma_id, }
+{ .convert_rate = rate, .dma_id = _dma_id, }
 #define RSND_SRC_UNUSED				\
-	{ .flags = 0, .convert_rate = 0, .dma_id = 0, }
-
-#define rsnd_scu_platform_info	rsnd_src_platform_info
-#define src_info		scu_info
-#define src_info_nr		scu_info_nr
+{ .convert_rate = 0, .dma_id = -1, }
 
 struct rsnd_src_platform_info {
-	u32 flags;
 	u32 convert_rate; /* sampling rate convert */
 	int dma_id; /* for Gen2 SCU */
 };
 
+/*
+ * flags
+ */
+struct rsnd_dvc_platform_info {
+	u32 flags;
+};
+
 struct rsnd_dai_path_info {
 	struct rsnd_ssi_platform_info *ssi;
 	struct rsnd_src_platform_info *src;
+	struct rsnd_dvc_platform_info *dvc;
 };
 
 struct rsnd_dai_platform_info {
@@ -99,6 +91,8 @@ struct rcar_snd_info {
 	int ssi_info_nr;
 	struct rsnd_src_platform_info *src_info;
 	int src_info_nr;
+	struct rsnd_dvc_platform_info *dvc_info;
+	int dvc_info_nr;
 	struct rsnd_dai_platform_info *dai_info;
 	int dai_info_nr;
 	int (*start)(int id);

include/sound/rt5640.h

@@ -16,6 +16,10 @@ struct rt5640_platform_data {
 	bool in1_diff;
 	bool in2_diff;
 
+	bool dmic_en;
+	bool dmic1_data_pin; /* 0 = IN1P; 1 = GPIO3 */
+	bool dmic2_data_pin; /* 0 = IN1N; 1 = GPIO4 */
+
 	int ldo1_en; /* GPIO for LDO1_EN */
 };
 

include/sound/rt5645.h

+/*
+ * linux/sound/rt5645.h -- Platform data for RT5645
+ *
+ * Copyright 2013 Realtek Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_RT5645_H
+#define __LINUX_SND_RT5645_H
+
+struct rt5645_platform_data {
+	/* IN2 can optionally be differential */
+	bool in2_diff;
+
+	bool dmic_en;
+	unsigned int dmic1_data_pin;
+	/* 0 = IN2N; 1 = GPIO5; 2 = GPIO11 */
+	unsigned int dmic2_data_pin;
+	/* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */
+};
+
+#endif

include/sound/rt5651.h

+/*
+ * linux/sound/rt286.h -- Platform data for RT286
+ *
+ * Copyright 2013 Realtek Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_RT5651_H
+#define __LINUX_SND_RT5651_H
+
+struct rt5651_platform_data {
+	/* IN2 can optionally be differential */
+	bool in2_diff;
+
+	bool dmic_en;
+};
+
+#endif

include/sound/rt5677.h

+/*
+ * linux/sound/rt5677.h -- Platform data for RT5677
+ *
+ * Copyright 2013 Realtek Semiconductor Corp.
+ * Author: Oder Chiou <oder_chiou@realtek.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_RT5677_H
+#define __LINUX_SND_RT5677_H
+
+struct rt5677_platform_data {
+	/* IN1 IN2 can optionally be differential */
+	bool in1_diff;
+	bool in2_diff;
+};
+
+#endif

include/sound/soc-dai.h

@@ -252,7 +252,6 @@ struct snd_soc_dai {
 	unsigned int symmetric_rates:1;
 	unsigned int symmetric_channels:1;
 	unsigned int symmetric_samplebits:1;
-	struct snd_pcm_runtime *runtime;
 	unsigned int active;
 	unsigned char probed:1;
 
@@ -277,7 +276,6 @@ struct snd_soc_dai {
 	struct snd_soc_card *card;
 
 	struct list_head list;
-	struct list_head card_list;
 };
 
 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,

include/sound/soc-dapm.h

@@ -107,10 +107,6 @@ struct device;
 {	.id = snd_soc_dapm_mux, .name = wname, \
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.kcontrol_news = wcontrols, .num_kcontrols = 1}
-#define SND_SOC_DAPM_VIRT_MUX(wname, wreg, wshift, winvert, wcontrols) \
-	SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols)
-#define SND_SOC_DAPM_VALUE_MUX(wname, wreg, wshift, winvert, wcontrols) \
-	SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols)
 
 /* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
 #define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\
@@ -166,10 +162,6 @@ struct device;
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.kcontrol_news = wcontrols, .num_kcontrols = 1, \
 	.event = wevent, .event_flags = wflags}
-#define SND_SOC_DAPM_VIRT_MUX_E(wname, wreg, wshift, winvert, wcontrols, \
-	wevent, wflags) \
-	SND_SOC_DAPM_MUX_E(wname, wreg, wshift, winvert, wcontrols, wevent, \
-		wflags)
 
 /* additional sequencing control within an event type */
 #define SND_SOC_DAPM_PGA_S(wname, wsubseq, wreg, wshift, winvert, \
@@ -256,9 +248,8 @@ struct device;
 /* generic widgets */
 #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \
 {	.id = wid, .name = wname, .kcontrol_news = NULL, .num_kcontrols = 0, \
-	.reg = -((wreg) + 1), .shift = wshift, .mask = wmask, \
-	.on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \
-	.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD}
+	.reg = wreg, .shift = wshift, .mask = wmask, \
+	.on_val = won_val, .off_val = woff_val, }
 #define SND_SOC_DAPM_SUPPLY(wname, wreg, wshift, winvert, wevent, wflags) \
 {	.id = snd_soc_dapm_supply, .name = wname, \
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
@@ -305,16 +296,12 @@ struct device;
  	.get = snd_soc_dapm_get_enum_double, \
  	.put = snd_soc_dapm_put_enum_double, \
   	.private_value = (unsigned long)&xenum }
-#define SOC_DAPM_ENUM_VIRT(xname, xenum) \
-	SOC_DAPM_ENUM(xname, xenum)
 #define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_enum_double, \
 	.get = xget, \
 	.put = xput, \
 	.private_value = (unsigned long)&xenum }
-#define SOC_DAPM_VALUE_ENUM(xname, xenum) \
-	SOC_DAPM_ENUM(xname, xenum)
 #define SOC_DAPM_PIN_SWITCH(xname) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname " Switch", \
 	.info = snd_soc_dapm_info_pin_switch, \
@@ -362,8 +349,6 @@ struct regulator;
 struct snd_soc_dapm_widget_list;
 struct snd_soc_dapm_update;
 
-int dapm_reg_event(struct snd_soc_dapm_widget *w,
-		   struct snd_kcontrol *kcontrol, int event);
 int dapm_regulator_event(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *kcontrol, int event);
 int dapm_clock_event(struct snd_soc_dapm_widget *w,
@@ -606,6 +591,7 @@ struct snd_soc_dapm_context {
 			     enum snd_soc_dapm_type, int);
 
 	struct device *dev; /* from parent - for debug */
+	struct snd_soc_component *component; /* parent component */
 	struct snd_soc_codec *codec; /* parent codec */
 	struct snd_soc_platform *platform; /* parent platform */
 	struct snd_soc_card *card; /* parent card */

include/sound/sta350.h

+/*
+ * Platform data for ST STA350 ASoC codec driver.
+ *
+ * Copyright: 2014 Raumfeld GmbH
+ * Author: Sven Brandau <info@brandau.biz>
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+#ifndef __LINUX_SND__STA350_H
+#define __LINUX_SND__STA350_H
+
+#define STA350_OCFG_2CH		0
+#define STA350_OCFG_2_1CH	1
+#define STA350_OCFG_1CH		3
+
+#define STA350_OM_CH1		0
+#define STA350_OM_CH2		1
+#define STA350_OM_CH3		2
+
+#define STA350_THERMAL_ADJUSTMENT_ENABLE	1
+#define STA350_THERMAL_RECOVERY_ENABLE		2
+#define STA350_FAULT_DETECT_RECOVERY_BYPASS	1
+
+#define STA350_FFX_PM_DROP_COMP			0
+#define STA350_FFX_PM_TAPERED_COMP		1
+#define STA350_FFX_PM_FULL_POWER		2
+#define STA350_FFX_PM_VARIABLE_DROP_COMP	3
+
+
+struct sta350_platform_data {
+	u8 output_conf;
+	u8 ch1_output_mapping;
+	u8 ch2_output_mapping;
+	u8 ch3_output_mapping;
+	u8 ffx_power_output_mode;
+	u8 drop_compensation_ns;
+	u8 powerdown_delay_divider;
+	unsigned int thermal_warning_recovery:1;
+	unsigned int thermal_warning_adjustment:1;
+	unsigned int fault_detect_recovery:1;
+	unsigned int oc_warning_adjustment:1;
+	unsigned int max_power_use_mpcc:1;
+	unsigned int max_power_correction:1;
+	unsigned int am_reduction_mode:1;
+	unsigned int odd_pwm_speed_mode:1;
+	unsigned int distortion_compensation:1;
+	unsigned int invalid_input_detect_mute:1;
+	unsigned int activate_mute_output:1;
+	unsigned int bridge_immediate_off:1;
+	unsigned int noise_shape_dc_cut:1;
+	unsigned int powerdown_master_vol:1;
+};
+
+#endif /* __LINUX_SND__STA350_H */

include/trace/events/asoc.h

@@ -11,102 +11,10 @@
 
 struct snd_soc_jack;
 struct snd_soc_codec;
-struct snd_soc_platform;
 struct snd_soc_card;
 struct snd_soc_dapm_widget;
 struct snd_soc_dapm_path;
 
-/*
- * Log register events
- */
-DECLARE_EVENT_CLASS(snd_soc_reg,
-
-	TP_PROTO(struct snd_soc_codec *codec, unsigned int reg,
-		 unsigned int val),
-
-	TP_ARGS(codec, reg, val),
-
-	TP_STRUCT__entry(
-		__string(	name,		codec->name	)
-		__field(	int,		id		)
-		__field(	unsigned int,	reg		)
-		__field(	unsigned int,	val		)
-	),
-
-	TP_fast_assign(
-		__assign_str(name, codec->name);
-		__entry->id = codec->id;
-		__entry->reg = reg;
-		__entry->val = val;
-	),
-
-	TP_printk("codec=%s.%d reg=%x val=%x", __get_str(name),
-		  (int)__entry->id, (unsigned int)__entry->reg,
-		  (unsigned int)__entry->val)
-);
-
-DEFINE_EVENT(snd_soc_reg, snd_soc_reg_write,
-
-	TP_PROTO(struct snd_soc_codec *codec, unsigned int reg,
-		 unsigned int val),
-
-	TP_ARGS(codec, reg, val)
-
-);
-
-DEFINE_EVENT(snd_soc_reg, snd_soc_reg_read,
-
-	TP_PROTO(struct snd_soc_codec *codec, unsigned int reg,
-		 unsigned int val),
-
-	TP_ARGS(codec, reg, val)
-
-);
-
-DECLARE_EVENT_CLASS(snd_soc_preg,
-
-	TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
-		 unsigned int val),
-
-	TP_ARGS(platform, reg, val),
-
-	TP_STRUCT__entry(
-		__string(	name,		platform->name	)
-		__field(	int,		id		)
-		__field(	unsigned int,	reg		)
-		__field(	unsigned int,	val		)
-	),
-
-	TP_fast_assign(
-		__assign_str(name, platform->name);
-		__entry->id = platform->id;
-		__entry->reg = reg;
-		__entry->val = val;
-	),
-
-	TP_printk("platform=%s.%d reg=%x val=%x", __get_str(name),
-		  (int)__entry->id, (unsigned int)__entry->reg,
-		  (unsigned int)__entry->val)
-);
-
-DEFINE_EVENT(snd_soc_preg, snd_soc_preg_write,
-
-	TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
-		 unsigned int val),
-
-	TP_ARGS(platform, reg, val)
-
-);
-
-DEFINE_EVENT(snd_soc_preg, snd_soc_preg_read,
-
-	TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
-		 unsigned int val),
-
-	TP_ARGS(platform, reg, val)
-
-);
-
 DECLARE_EVENT_CLASS(snd_soc_card,
 
 	TP_PROTO(struct snd_soc_card *card, int val),

include/uapi/sound/asound.h

@@ -94,9 +94,11 @@ enum {
 	SNDRV_HWDEP_IFACE_HDA,		/* HD-audio */
 	SNDRV_HWDEP_IFACE_USB_STREAM,	/* direct access to usb stream */
 	SNDRV_HWDEP_IFACE_FW_DICE,	/* TC DICE FireWire device */
+	SNDRV_HWDEP_IFACE_FW_FIREWORKS,	/* Echo Audio Fireworks based device */
+	SNDRV_HWDEP_IFACE_FW_BEBOB,	/* BridgeCo BeBoB based device */
 
 	/* Don't forget to change the following: */
-	SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_DICE
+	SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_BEBOB
 };
 
 struct snd_hwdep_info {

include/uapi/sound/firewire.h

@@ -2,11 +2,13 @@
 #define _UAPI_SOUND_FIREWIRE_H_INCLUDED
 
 #include <linux/ioctl.h>
+#include <linux/types.h>
 
 /* events can be read() from the hwdep device */
 
 #define SNDRV_FIREWIRE_EVENT_LOCK_STATUS	0x000010cc
 #define SNDRV_FIREWIRE_EVENT_DICE_NOTIFICATION	0xd1ce004e
+#define SNDRV_FIREWIRE_EVENT_EFW_RESPONSE	0x4e617475
 
 struct snd_firewire_event_common {
 	unsigned int type; /* SNDRV_FIREWIRE_EVENT_xxx */
@@ -22,10 +24,27 @@ struct snd_firewire_event_dice_notification {
 	unsigned int notification; /* DICE-specific bits */
 };
 
+#define SND_EFW_TRANSACTION_USER_SEQNUM_MAX	((__u32)((__u16)~0) - 1)
+/* each field should be in big endian */
+struct snd_efw_transaction {
+	__be32 length;
+	__be32 version;
+	__be32 seqnum;
+	__be32 category;
+	__be32 command;
+	__be32 status;
+	__be32 params[0];
+};
+struct snd_firewire_event_efw_response {
+	unsigned int type;
+	__be32 response[0];	/* some responses */
+};
+
 union snd_firewire_event {
 	struct snd_firewire_event_common            common;
 	struct snd_firewire_event_lock_status       lock_status;
 	struct snd_firewire_event_dice_notification dice_notification;
+	struct snd_firewire_event_efw_response      efw_response;
 };
 
 
@@ -34,7 +53,9 @@ union snd_firewire_event {
 #define SNDRV_FIREWIRE_IOCTL_UNLOCK    _IO('H', 0xfa)
 
 #define SNDRV_FIREWIRE_TYPE_DICE	1
-/* Fireworks, AV/C, RME, MOTU, ... */
+#define SNDRV_FIREWIRE_TYPE_FIREWORKS	2
+#define SNDRV_FIREWIRE_TYPE_BEBOB	3
+/* AV/C, RME, MOTU, ... */
 
 struct snd_firewire_get_info {
 	unsigned int type; /* SNDRV_FIREWIRE_TYPE_xxx */

sound/aoa/codecs/onyx.c

@@ -241,7 +241,7 @@ static struct snd_kcontrol_new inputgain_control = {
 static int onyx_snd_capture_source_info(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "Line-In", "Microphone" };
+	static const char * const texts[] = { "Line-In", "Microphone" };
 
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;

sound/arm/pxa2xx-pcm.c

@@ -14,6 +14,8 @@
 #include <linux/dma-mapping.h>
 #include <linux/dmaengine.h>
 
+#include <mach/dma.h>
+
 #include <sound/core.h>
 #include <sound/pxa2xx-lib.h>
 #include <sound/dmaengine_pcm.h>

sound/arm/pxa2xx-pcm.h

@@ -9,12 +9,11 @@
  * it under the terms of the GNU General Public License version 2 as
  * published by the Free Software Foundation.
  */
-#include <mach/dma.h>
 
 struct pxa2xx_runtime_data {
 	int dma_ch;
 	struct snd_dmaengine_dai_dma_data *params;
-	pxa_dma_desc *dma_desc_array;
+	struct pxa_dma_desc *dma_desc_array;
 	dma_addr_t dma_desc_array_phys;
 };
 

sound/atmel/ac97c.c

@@ -1198,6 +1198,7 @@ static int atmel_ac97c_remove(struct platform_device *pdev)
 }
 
 static struct platform_driver atmel_ac97c_driver = {
+	.probe		= atmel_ac97c_probe,
 	.remove		= atmel_ac97c_remove,
 	.driver		= {
 		.name	= "atmel_ac97c",
@@ -1205,19 +1206,7 @@ static struct platform_driver atmel_ac97c_driver = {
 		.pm	= ATMEL_AC97C_PM_OPS,
 	},
 };
-
-static int __init atmel_ac97c_init(void)
-{
-	return platform_driver_probe(&atmel_ac97c_driver,
-			atmel_ac97c_probe);
-}
-module_init(atmel_ac97c_init);
-
-static void __exit atmel_ac97c_exit(void)
-{
-	platform_driver_unregister(&atmel_ac97c_driver);
-}
-module_exit(atmel_ac97c_exit);
+module_platform_driver(atmel_ac97c_driver);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Driver for Atmel AC97 controller");

sound/core/pcm_lib.c

@@ -345,7 +345,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 			snd_pcm_debug_name(substream, name, sizeof(name));
 			xrun_log_show(substream);
 			pcm_err(substream->pcm,
-				"BUG: %s, pos = %ld, buffer size = %ld, period size = %ld\n",
+				"XRUN: %s, pos = %ld, buffer size = %ld, period size = %ld\n",
 				name, pos, runtime->buffer_size,
 				runtime->period_size);
 		}

sound/core/seq/seq_midi.c

@@ -362,13 +362,13 @@ snd_seq_midisynth_register_port(struct snd_seq_device *dev)
 		if (! port->name[0]) {
 			if (info->name[0]) {
 				if (ports > 1)
-					snprintf(port->name, sizeof(port->name), "%s-%d", info->name, p);
+					snprintf(port->name, sizeof(port->name), "%s-%u", info->name, p);
 				else
 					snprintf(port->name, sizeof(port->name), "%s", info->name);
 			} else {
 				/* last resort */
 				if (ports > 1)
-					sprintf(port->name, "MIDI %d-%d-%d", card->number, device, p);
+					sprintf(port->name, "MIDI %d-%d-%u", card->number, device, p);
 				else
 					sprintf(port->name, "MIDI %d-%d", card->number, device);
 			}

sound/firewire/Kconfig

@@ -9,12 +9,12 @@ if SND_FIREWIRE && FIREWIRE
 
 config SND_FIREWIRE_LIB
 	tristate
-	depends on SND_PCM
+	select SND_PCM
+	select SND_RAWMIDI
 
 config SND_DICE
 	tristate "DICE-based DACs (EXPERIMENTAL)"
 	select SND_HWDEP
-	select SND_PCM
 	select SND_FIREWIRE_LIB
 	help
 	  Say Y here to include support for many DACs based on the DICE
@@ -28,7 +28,6 @@ config SND_DICE
 
 config SND_FIREWIRE_SPEAKERS
 	tristate "FireWire speakers"
-	select SND_PCM
 	select SND_FIREWIRE_LIB
 	help
 	  Say Y here to include support for the Griffin FireWave Surround
@@ -39,7 +38,6 @@ config SND_FIREWIRE_SPEAKERS
 
 config SND_ISIGHT
 	tristate "Apple iSight microphone"
-	select SND_PCM
 	select SND_FIREWIRE_LIB
 	help
 	  Say Y here to include support for the front and rear microphones
@@ -50,8 +48,6 @@ config SND_ISIGHT
 
 config SND_SCS1X
 	tristate "Stanton Control System 1 MIDI"
-	select SND_PCM
-	select SND_RAWMIDI
 	select SND_FIREWIRE_LIB
 	help
 	  Say Y here to include support for the MIDI ports of the Stanton
@@ -61,4 +57,59 @@ config SND_SCS1X
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-scs1x.
 
+config SND_FIREWORKS
+	tristate "Echo Fireworks board module support"
+	select SND_FIREWIRE_LIB
+	select SND_HWDEP
+	help
+	  Say Y here to include support for FireWire devices based
+	  on Echo Digital Audio Fireworks board:
+	   * Mackie Onyx 400F/1200F
+	   * Echo AudioFire12/8(until 2009 July)
+	   * Echo AudioFire2/4/Pre8/8(since 2009 July)
+	   * Echo Fireworks 8/HDMI
+	   * Gibson Robot Interface Pack/GoldTop
+
+	  To compile this driver as a module, choose M here: the module
+	  will be called snd-fireworks.
+
+config SND_BEBOB
+	tristate "BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware"
+	select SND_FIREWIRE_LIB
+	select SND_HWDEP
+        help
+	 Say Y here to include support for FireWire devices based
+	 on BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware:
+	  * Edirol FA-66/FA-101
+	  * PreSonus FIREBOX/FIREPOD/FP10/Inspire1394
+	  * BridgeCo RDAudio1/Audio5
+	  * Mackie Onyx 1220/1620/1640 (Firewire I/O Card)
+	  * Mackie d.2 (Firewire Option)
+	  * Stanton FinalScratch 2 (ScratchAmp)
+	  * Tascam IF-FW/DM
+	  * Behringer XENIX UFX 1204/1604
+	  * Behringer Digital Mixer X32 series (X-UF Card)
+	  * Apogee Rosetta 200/400 (X-FireWire card)
+	  * Apogee DA/AD/DD-16X (X-FireWire card)
+	  * Apogee Ensemble
+	  * ESI Quotafire610
+	  * AcousticReality eARMasterOne
+	  * CME MatrixKFW
+	  * Phonic Helix Board 12 MkII/18 MkII/24 MkII
+	  * Phonic Helix Board 12 Universal/18 Universal/24 Universal
+	  * Lynx Aurora 8/16 (LT-FW)
+	  * ICON FireXon
+	  * PrismSound Orpheus/ADA-8XR
+	  * TerraTec PHASE 24 FW/PHASE X24 FW/PHASE 88 Rack FW
+	  * Terratec EWS MIC2/EWS MIC4
+	  * Terratec Aureon 7.1 Firewire
+	  * Yamaha GO44/GO46
+	  * Focusrite Saffire/Saffire LE/SaffirePro10 IO/SaffirePro26 IO
+	  * M-Audio Firewire410/AudioPhile/Solo
+	  * M-Audio Ozonic/NRV10/ProfireLightBridge
+	  * M-Audio Firewire 1814/ProjectMix IO
+
+          To compile this driver as a module, choose M here: the module
+          will be called snd-bebob.
+
 endif # SND_FIREWIRE

sound/firewire/Makefile

@@ -10,3 +10,5 @@ obj-$(CONFIG_SND_DICE) += snd-dice.o
 obj-$(CONFIG_SND_FIREWIRE_SPEAKERS) += snd-firewire-speakers.o
 obj-$(CONFIG_SND_ISIGHT) += snd-isight.o
 obj-$(CONFIG_SND_SCS1X) += snd-scs1x.o
+obj-$(CONFIG_SND_FIREWORKS) += fireworks/
+obj-$(CONFIG_SND_BEBOB) += bebob/

sound/firewire/amdtp.h

@@ -8,7 +8,7 @@
 #include "packets-buffer.h"
 
 /**
- * enum cip_out_flags - describes details of the streaming protocol
+ * enum cip_flags - describes details of the streaming protocol
  * @CIP_NONBLOCKING: In non-blocking mode, each packet contains
  *	sample_rate/8000 samples, with rounding up or down to adjust
  *	for clock skew and left-over fractional samples.  This should
@@ -16,15 +16,30 @@
  * @CIP_BLOCKING: In blocking mode, each packet contains either zero or
  *	SYT_INTERVAL samples, with these two types alternating so that
  *	the overall sample rate comes out right.
- * @CIP_HI_DUALWIRE: At rates above 96 kHz, pretend that the stream runs
- *	at half the actual sample rate with twice the number of channels;
- *	two samples of a channel are stored consecutively in the packet.
- *	Requires blocking mode and SYT_INTERVAL-aligned PCM buffer size.
+ * @CIP_SYNC_TO_DEVICE: In sync to device mode, time stamp in out packets is
+ *	generated by in packets. Defaultly this driver generates timestamp.
+ * @CIP_EMPTY_WITH_TAG0: Only for in-stream. Empty in-packets have TAG0.
+ * @CIP_DBC_IS_END_EVENT: Only for in-stream. The value of dbc in an in-packet
+ *	corresponds to the end of event in the packet. Out of IEC 61883.
+ * @CIP_WRONG_DBS: Only for in-stream. The value of dbs is wrong in in-packets.
+ *	The value of data_block_quadlets is used instead of reported value.
+ * @SKIP_DBC_ZERO_CHECK: Only for in-stream.  Packets with zero in dbc is
+ *	skipped for detecting discontinuity.
+ * @CIP_SKIP_INIT_DBC_CHECK: Only for in-stream. The value of dbc in first
+ *	packet is not continuous from an initial value.
+ * @CIP_EMPTY_HAS_WRONG_DBC: Only for in-stream. The value of dbc in empty
+ *	packet is wrong but the others are correct.
  */
-enum cip_out_flags {
-	CIP_NONBLOCKING	= 0x00,
-	CIP_BLOCKING	= 0x01,
-	CIP_HI_DUALWIRE	= 0x02,
+enum cip_flags {
+	CIP_NONBLOCKING		= 0x00,
+	CIP_BLOCKING		= 0x01,
+	CIP_SYNC_TO_DEVICE	= 0x02,
+	CIP_EMPTY_WITH_TAG0	= 0x04,
+	CIP_DBC_IS_END_EVENT	= 0x08,
+	CIP_WRONG_DBS		= 0x10,
+	CIP_SKIP_DBC_ZERO_CHECK	= 0x20,
+	CIP_SKIP_INIT_DBC_CHECK	= 0x40,
+	CIP_EMPTY_HAS_WRONG_DBC	= 0x80,
 };
 
 /**
@@ -41,27 +56,55 @@ enum cip_sfc {
 	CIP_SFC_COUNT
 };
 
+#define AMDTP_IN_PCM_FORMAT_BITS	SNDRV_PCM_FMTBIT_S32
+
 #define AMDTP_OUT_PCM_FORMAT_BITS	(SNDRV_PCM_FMTBIT_S16 | \
 					 SNDRV_PCM_FMTBIT_S32)
 
+
+/*
+ * This module supports maximum 64 PCM channels for one PCM stream
+ * This is for our convenience.
+ */
+#define AMDTP_MAX_CHANNELS_FOR_PCM	64
+
+/*
+ * AMDTP packet can include channels for MIDI conformant data.
+ * Each MIDI conformant data channel includes 8 MPX-MIDI data stream.
+ * Each MPX-MIDI data stream includes one data stream from/to MIDI ports.
+ *
+ * This module supports maximum 1 MIDI conformant data channels.
+ * Then this AMDTP packets can transfer maximum 8 MIDI data streams.
+ */
+#define AMDTP_MAX_CHANNELS_FOR_MIDI	1
+
 struct fw_unit;
 struct fw_iso_context;
 struct snd_pcm_substream;
+struct snd_pcm_runtime;
+struct snd_rawmidi_substream;
 
-struct amdtp_out_stream {
+enum amdtp_stream_direction {
+	AMDTP_OUT_STREAM = 0,
+	AMDTP_IN_STREAM
+};
+
+struct amdtp_stream {
 	struct fw_unit *unit;
-	enum cip_out_flags flags;
+	enum cip_flags flags;
+	enum amdtp_stream_direction direction;
 	struct fw_iso_context *context;
 	struct mutex mutex;
 
 	enum cip_sfc sfc;
-	bool dual_wire;
 	unsigned int data_block_quadlets;
 	unsigned int pcm_channels;
 	unsigned int midi_ports;
-	void (*transfer_samples)(struct amdtp_out_stream *s,
+	void (*transfer_samples)(struct amdtp_stream *s,
 				 struct snd_pcm_substream *pcm,
 				 __be32 *buffer, unsigned int frames);
+	u8 pcm_positions[AMDTP_MAX_CHANNELS_FOR_PCM];
+	u8 midi_position;
 
 	unsigned int syt_interval;
 	unsigned int transfer_delay;
@@ -82,65 +125,148 @@ struct amdtp_out_stream {
 	unsigned int pcm_buffer_pointer;
 	unsigned int pcm_period_pointer;
 	bool pointer_flush;
+
+	struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
+
+	/* quirk: fixed interval of dbc between previos/current packets. */
+	unsigned int tx_dbc_interval;
+
+	/* quirk: the first count of data blocks in an rx packet for MIDI */
+	unsigned int rx_blocks_for_midi;
+
+	bool callbacked;
+	wait_queue_head_t callback_wait;
+	struct amdtp_stream *sync_slave;
 };
 
-int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit,
-			  enum cip_out_flags flags);
-void amdtp_out_stream_destroy(struct amdtp_out_stream *s);
+int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
+		      enum amdtp_stream_direction dir,
+		      enum cip_flags flags);
+void amdtp_stream_destroy(struct amdtp_stream *s);
 
-void amdtp_out_stream_set_parameters(struct amdtp_out_stream *s,
-				     unsigned int rate,
-				     unsigned int pcm_channels,
-				     unsigned int midi_ports);
-unsigned int amdtp_out_stream_get_max_payload(struct amdtp_out_stream *s);
+void amdtp_stream_set_parameters(struct amdtp_stream *s,
+				 unsigned int rate,
+				 unsigned int pcm_channels,
+				 unsigned int midi_ports);
+unsigned int amdtp_stream_get_max_payload(struct amdtp_stream *s);
 
-int amdtp_out_stream_start(struct amdtp_out_stream *s, int channel, int speed);
-void amdtp_out_stream_update(struct amdtp_out_stream *s);
-void amdtp_out_stream_stop(struct amdtp_out_stream *s);
+int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed);
+void amdtp_stream_update(struct amdtp_stream *s);
+void amdtp_stream_stop(struct amdtp_stream *s);
 
-void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s,
-				     snd_pcm_format_t format);
-void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s);
-unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s);
-void amdtp_out_stream_pcm_abort(struct amdtp_out_stream *s);
+int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
+					struct snd_pcm_runtime *runtime);
+void amdtp_stream_set_pcm_format(struct amdtp_stream *s,
+				 snd_pcm_format_t format);
+void amdtp_stream_pcm_prepare(struct amdtp_stream *s);
+unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s);
+void amdtp_stream_pcm_abort(struct amdtp_stream *s);
 
 extern const unsigned int amdtp_syt_intervals[CIP_SFC_COUNT];
+extern const unsigned int amdtp_rate_table[CIP_SFC_COUNT];
 
-static inline bool amdtp_out_stream_running(struct amdtp_out_stream *s)
+/**
+ * amdtp_stream_running - check stream is running or not
+ * @s: the AMDTP stream
+ *
+ * If this function returns true, the stream is running.
+ */
+static inline bool amdtp_stream_running(struct amdtp_stream *s)
 {
 	return !IS_ERR(s->context);
 }
 
 /**
- * amdtp_out_streaming_error - check for streaming error
- * @s: the AMDTP output stream
+ * amdtp_streaming_error - check for streaming error
+ * @s: the AMDTP stream
  *
  * If this function returns true, the stream's packet queue has stopped due to
  * an asynchronous error.
  */
-static inline bool amdtp_out_streaming_error(struct amdtp_out_stream *s)
+static inline bool amdtp_streaming_error(struct amdtp_stream *s)
 {
 	return s->packet_index < 0;
 }
 
 /**
- * amdtp_out_stream_pcm_trigger - start/stop playback from a PCM device
- * @s: the AMDTP output stream
+ * amdtp_stream_pcm_running - check PCM substream is running or not
+ * @s: the AMDTP stream
+ *
+ * If this function returns true, PCM substream in the AMDTP stream is running.
+ */
+static inline bool amdtp_stream_pcm_running(struct amdtp_stream *s)
+{
+	return !!s->pcm;
+}
+
+/**
+ * amdtp_stream_pcm_trigger - start/stop playback from a PCM device
+ * @s: the AMDTP stream
  * @pcm: the PCM device to be started, or %NULL to stop the current device
  *
  * Call this function on a running isochronous stream to enable the actual
  * transmission of PCM data.  This function should be called from the PCM
  * device's .trigger callback.
  */
-static inline void amdtp_out_stream_pcm_trigger(struct amdtp_out_stream *s,
-						struct snd_pcm_substream *pcm)
+static inline void amdtp_stream_pcm_trigger(struct amdtp_stream *s,
+					    struct snd_pcm_substream *pcm)
 {
 	ACCESS_ONCE(s->pcm) = pcm;
 }
 
+/**
+ * amdtp_stream_midi_trigger - start/stop playback/capture with a MIDI device
+ * @s: the AMDTP stream
+ * @port: index of MIDI port
+ * @midi: the MIDI device to be started, or %NULL to stop the current device
+ *
+ * Call this function on a running isochronous stream to enable the actual
+ * transmission of MIDI data.  This function should be called from the MIDI
+ * device's .trigger callback.
+ */
+static inline void amdtp_stream_midi_trigger(struct amdtp_stream *s,
+					     unsigned int port,
+					     struct snd_rawmidi_substream *midi)
+{
+	if (port < s->midi_ports)
+		ACCESS_ONCE(s->midi[port]) = midi;
+}
+
 static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc)
 {
 	return sfc & 1;
 }
 
+static inline void amdtp_stream_set_sync(enum cip_flags sync_mode,
+					 struct amdtp_stream *master,
+					 struct amdtp_stream *slave)
+{
+	if (sync_mode == CIP_SYNC_TO_DEVICE) {
+		master->flags |= CIP_SYNC_TO_DEVICE;
+		slave->flags |= CIP_SYNC_TO_DEVICE;
+		master->sync_slave = slave;
+	} else {
+		master->flags &= ~CIP_SYNC_TO_DEVICE;
+		slave->flags &= ~CIP_SYNC_TO_DEVICE;
+		master->sync_slave = NULL;
+	}
+
+	slave->sync_slave = NULL;
+}
+
+/**
+ * amdtp_stream_wait_callback - sleep till callbacked or timeout
+ * @s: the AMDTP stream
+ * @timeout: msec till timeout
+ *
+ * If this function return false, the AMDTP stream should be stopped.
+ */
+static inline bool amdtp_stream_wait_callback(struct amdtp_stream *s,
+					      unsigned int timeout)
+{
+	return wait_event_timeout(s->callback_wait,
+				  s->callbacked == true,
+				  msecs_to_jiffies(timeout)) > 0;
+}
+
 #endif

sound/firewire/bebob/Makefile

+snd-bebob-objs := bebob_command.o bebob_stream.o bebob_proc.o bebob_midi.o \
+		  bebob_pcm.o bebob_hwdep.o bebob_terratec.o bebob_yamaha.o \
+		  bebob_focusrite.o bebob_maudio.o bebob.o
+obj-m += snd-bebob.o

sound/firewire/bebob/bebob.h

+/*
+ * bebob.h - a part of driver for BeBoB based devices
+ *
+ * Copyright (c) 2013-2014 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#ifndef SOUND_BEBOB_H_INCLUDED
+#define SOUND_BEBOB_H_INCLUDED
+
+#include <linux/compat.h>
+#include <linux/device.h>
+#include <linux/firewire.h>
+#include <linux/firewire-constants.h>
+#include <linux/module.h>
+#include <linux/mod_devicetable.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/rawmidi.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/firewire.h>
+#include <sound/hwdep.h>
+
+#include "../lib.h"
+#include "../fcp.h"
+#include "../packets-buffer.h"
+#include "../iso-resources.h"
+#include "../amdtp.h"
+#include "../cmp.h"
+
+/* basic register addresses on DM1000/DM1100/DM1500 */
+#define BEBOB_ADDR_REG_INFO	0xffffc8020000ULL
+#define BEBOB_ADDR_REG_REQ	0xffffc8021000ULL
+
+struct snd_bebob;
+
+#define SND_BEBOB_STRM_FMT_ENTRIES	7
+struct snd_bebob_stream_formation {
+	unsigned int pcm;
+	unsigned int midi;
+};
+/* this is a lookup table for index of stream formations */
+extern const unsigned int snd_bebob_rate_table[SND_BEBOB_STRM_FMT_ENTRIES];
+
+/* device specific operations */
+#define SND_BEBOB_CLOCK_INTERNAL	"Internal"
+struct snd_bebob_clock_spec {
+	unsigned int num;
+	char *const *labels;
+	int (*get)(struct snd_bebob *bebob, unsigned int *id);
+};
+struct snd_bebob_rate_spec {
+	int (*get)(struct snd_bebob *bebob, unsigned int *rate);
+	int (*set)(struct snd_bebob *bebob, unsigned int rate);
+};
+struct snd_bebob_meter_spec {
+	unsigned int num;
+	char *const *labels;
+	int (*get)(struct snd_bebob *bebob, u32 *target, unsigned int size);
+};
+struct snd_bebob_spec {
+	struct snd_bebob_clock_spec *clock;
+	struct snd_bebob_rate_spec *rate;
+	struct snd_bebob_meter_spec *meter;
+};
+
+struct snd_bebob {
+	struct snd_card *card;
+	struct fw_unit *unit;
+	int card_index;
+
+	struct mutex mutex;
+	spinlock_t lock;
+
+	const struct snd_bebob_spec *spec;
+
+	unsigned int midi_input_ports;
+	unsigned int midi_output_ports;
+
+	/* for bus reset quirk */
+	struct completion bus_reset;
+	bool connected;
+
+	struct amdtp_stream *master;
+	struct amdtp_stream tx_stream;
+	struct amdtp_stream rx_stream;
+	struct cmp_connection out_conn;
+	struct cmp_connection in_conn;
+	atomic_t capture_substreams;
+	atomic_t playback_substreams;
+
+	struct snd_bebob_stream_formation
+		tx_stream_formations[SND_BEBOB_STRM_FMT_ENTRIES];
+	struct snd_bebob_stream_formation
+		rx_stream_formations[SND_BEBOB_STRM_FMT_ENTRIES];
+
+	int sync_input_plug;
+
+	/* for uapi */
+	int dev_lock_count;
+	bool dev_lock_changed;
+	wait_queue_head_t hwdep_wait;
+
+	/* for M-Audio special devices */
+	void *maudio_special_quirk;
+	bool deferred_registration;
+};
+
+static inline int
+snd_bebob_read_block(struct fw_unit *unit, u64 addr, void *buf, int size)
+{
+	return snd_fw_transaction(unit, TCODE_READ_BLOCK_REQUEST,
+				  BEBOB_ADDR_REG_INFO + addr,
+				  buf, size, 0);
+}
+
+static inline int
+snd_bebob_read_quad(struct fw_unit *unit, u64 addr, u32 *buf)
+{
+	return snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST,
+				  BEBOB_ADDR_REG_INFO + addr,
+				  (void *)buf, sizeof(u32), 0);
+}
+
+/* AV/C Audio Subunit Specification 1.0 (Oct 2000, 1394TA) */
+int avc_audio_set_selector(struct fw_unit *unit, unsigned int subunit_id,
+			   unsigned int fb_id, unsigned int num);
+int avc_audio_get_selector(struct fw_unit *unit, unsigned  int subunit_id,
+			   unsigned int fb_id, unsigned int *num);
+
+/*
+ * AVC command extensions, AV/C Unit and Subunit, Revision 17
+ * (Nov 2003, BridgeCo)
+ */
+#define	AVC_BRIDGECO_ADDR_BYTES	6
+enum avc_bridgeco_plug_dir {
+	AVC_BRIDGECO_PLUG_DIR_IN	= 0x00,
+	AVC_BRIDGECO_PLUG_DIR_OUT	= 0x01
+};
+enum avc_bridgeco_plug_mode {
+	AVC_BRIDGECO_PLUG_MODE_UNIT		= 0x00,
+	AVC_BRIDGECO_PLUG_MODE_SUBUNIT		= 0x01,
+	AVC_BRIDGECO_PLUG_MODE_FUNCTION_BLOCK	= 0x02
+};
+enum avc_bridgeco_plug_unit {
+	AVC_BRIDGECO_PLUG_UNIT_ISOC	= 0x00,
+	AVC_BRIDGECO_PLUG_UNIT_EXT	= 0x01,
+	AVC_BRIDGECO_PLUG_UNIT_ASYNC	= 0x02
+};
+enum avc_bridgeco_plug_type {
+	AVC_BRIDGECO_PLUG_TYPE_ISOC	= 0x00,
+	AVC_BRIDGECO_PLUG_TYPE_ASYNC	= 0x01,
+	AVC_BRIDGECO_PLUG_TYPE_MIDI	= 0x02,
+	AVC_BRIDGECO_PLUG_TYPE_SYNC	= 0x03,
+	AVC_BRIDGECO_PLUG_TYPE_ANA	= 0x04,
+	AVC_BRIDGECO_PLUG_TYPE_DIG	= 0x05
+};
+static inline void
+avc_bridgeco_fill_unit_addr(u8 buf[AVC_BRIDGECO_ADDR_BYTES],
+			    enum avc_bridgeco_plug_dir dir,
+			    enum avc_bridgeco_plug_unit unit,
+			    unsigned int pid)
+{
+	buf[0] = 0xff;	/* Unit */
+	buf[1] = dir;
+	buf[2] = AVC_BRIDGECO_PLUG_MODE_UNIT;
+	buf[3] = unit;
+	buf[4] = 0xff & pid;
+	buf[5] = 0xff;	/* reserved */
+}
+static inline void
+avc_bridgeco_fill_msu_addr(u8 buf[AVC_BRIDGECO_ADDR_BYTES],
+			   enum avc_bridgeco_plug_dir dir,
+			   unsigned int pid)
+{
+	buf[0] = 0x60;	/* Music subunit */
+	buf[1] = dir;
+	buf[2] = AVC_BRIDGECO_PLUG_MODE_SUBUNIT;
+	buf[3] = 0xff & pid;
+	buf[4] = 0xff;	/* reserved */
+	buf[5] = 0xff;	/* reserved */
+}
+int avc_bridgeco_get_plug_ch_pos(struct fw_unit *unit,
+				 u8 addr[AVC_BRIDGECO_ADDR_BYTES],
+				 u8 *buf, unsigned int len);
+int avc_bridgeco_get_plug_type(struct fw_unit *unit,
+			       u8 addr[AVC_BRIDGECO_ADDR_BYTES],
+			       enum avc_bridgeco_plug_type *type);
+int avc_bridgeco_get_plug_section_type(struct fw_unit *unit,
+				       u8 addr[AVC_BRIDGECO_ADDR_BYTES],
+				       unsigned int id, u8 *type);
+int avc_bridgeco_get_plug_input(struct fw_unit *unit,
+				u8 addr[AVC_BRIDGECO_ADDR_BYTES],
+				u8 input[7]);
+int avc_bridgeco_get_plug_strm_fmt(struct fw_unit *unit,
+				   u8 addr[AVC_BRIDGECO_ADDR_BYTES], u8 *buf,
+				   unsigned int *len, unsigned int eid);
+
+/* for AMDTP streaming */
+int snd_bebob_stream_get_rate(struct snd_bebob *bebob, unsigned int *rate);
+int snd_bebob_stream_set_rate(struct snd_bebob *bebob, unsigned int rate);
+int snd_bebob_stream_check_internal_clock(struct snd_bebob *bebob,
+					  bool *internal);
+int snd_bebob_stream_discover(struct snd_bebob *bebob);
+int snd_bebob_stream_map(struct snd_bebob *bebob,
+			 struct amdtp_stream *stream);
+int snd_bebob_stream_init_duplex(struct snd_bebob *bebob);
+int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate);
+void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob);
+void snd_bebob_stream_update_duplex(struct snd_bebob *bebob);
+void snd_bebob_stream_destroy_duplex(struct snd_bebob *bebob);
+
+void snd_bebob_stream_lock_changed(struct snd_bebob *bebob);
+int snd_bebob_stream_lock_try(struct snd_bebob *bebob);
+void snd_bebob_stream_lock_release(struct snd_bebob *bebob);
+
+void snd_bebob_proc_init(struct snd_bebob *bebob);
+
+int snd_bebob_create_midi_devices(struct snd_bebob *bebob);
+
+int snd_bebob_create_pcm_devices(struct snd_bebob *bebob);
+
+int snd_bebob_create_hwdep_device(struct snd_bebob *bebob);
+
+/* model specific operations */
+extern struct snd_bebob_spec phase88_rack_spec;
+extern struct snd_bebob_spec phase24_series_spec;
+extern struct snd_bebob_spec yamaha_go_spec;
+extern struct snd_bebob_spec saffirepro_26_spec;
+extern struct snd_bebob_spec saffirepro_10_spec;
+extern struct snd_bebob_spec saffire_le_spec;
+extern struct snd_bebob_spec saffire_spec;
+extern struct snd_bebob_spec maudio_fw410_spec;
+extern struct snd_bebob_spec maudio_audiophile_spec;
+extern struct snd_bebob_spec maudio_solo_spec;
+extern struct snd_bebob_spec maudio_ozonic_spec;
+extern struct snd_bebob_spec maudio_nrv10_spec;
+extern struct snd_bebob_spec maudio_special_spec;
+int snd_bebob_maudio_special_discover(struct snd_bebob *bebob, bool is1814);
+int snd_bebob_maudio_load_firmware(struct fw_unit *unit);
+
+#define SND_BEBOB_DEV_ENTRY(vendor, model, data) \
+{ \
+	.match_flags	= IEEE1394_MATCH_VENDOR_ID | \
+			  IEEE1394_MATCH_MODEL_ID, \
+	.vendor_id	= vendor, \
+	.model_id	= model, \
+	.driver_data	= (kernel_ulong_t)data \
+}
+
+#endif

sound/firewire/bebob/bebob_command.c

+/*
+ * bebob_command.c - driver for BeBoB based devices
+ *
+ * Copyright (c) 2013-2014 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "./bebob.h"
+
+int avc_audio_set_selector(struct fw_unit *unit, unsigned int subunit_id,
+			   unsigned int fb_id, unsigned int num)
+{
+	u8 *buf;
+	int err;
+
+	buf = kzalloc(12, GFP_KERNEL);
+	if (buf == NULL)
+		return -ENOMEM;
+
+	buf[0]  = 0x00;		/* AV/C CONTROL */
+	buf[1]  = 0x08 | (0x07 & subunit_id);	/* AUDIO SUBUNIT ID */
+	buf[2]  = 0xb8;		/* FUNCTION BLOCK  */
+	buf[3]  = 0x80;		/* type is 'selector'*/
+	buf[4]  = 0xff & fb_id;	/* function block id */
+	buf[5]  = 0x10;		/* control attribute is CURRENT */
+	buf[6]  = 0x02;		/* selector length is 2 */
+	buf[7]  = 0xff & num;	/* input function block plug number */
+	buf[8]  = 0x01;		/* control selector is SELECTOR_CONTROL */
+
+	err = fcp_avc_transaction(unit, buf, 12, buf, 12,
+				  BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) |
+				  BIT(6) | BIT(7) | BIT(8));
+	if (err > 0 && err < 9)
+		err = -EIO;
+	else if (buf[0] == 0x08) /* NOT IMPLEMENTED */
+		err = -ENOSYS;
+	else if (buf[0] == 0x0a) /* REJECTED */
+		err = -EINVAL;
+	else if (err > 0)
+		err = 0;
+
+	kfree(buf);
+	return err;
+}
+
+int avc_audio_get_selector(struct fw_unit *unit, unsigned int subunit_id,
+			   unsigned int fb_id, unsigned int *num)
+{
+	u8 *buf;
+	int err;
+
+	buf = kzalloc(12, GFP_KERNEL);
+	if (buf == NULL)
+		return -ENOMEM;
+
+	buf[0]  = 0x01;		/* AV/C STATUS */
+	buf[1]  = 0x08 | (0x07 & subunit_id);	/* AUDIO SUBUNIT ID */
+	buf[2]  = 0xb8;		/* FUNCTION BLOCK */
+	buf[3]  = 0x80;		/* type is 'selector'*/
+	buf[4]  = 0xff & fb_id;	/* function block id */
+	buf[5]  = 0x10;		/* control attribute is CURRENT */
+	buf[6]  = 0x02;		/* selector length is 2 */
+	buf[7]  = 0xff;		/* input function block plug number */
+	buf[8]  = 0x01;		/* control selector is SELECTOR_CONTROL */
+
+	err = fcp_avc_transaction(unit, buf, 12, buf, 12,
+				  BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) |
+				  BIT(6) | BIT(8));
+	if (err > 0 && err < 9)
+		err = -EIO;
+	else if (buf[0] == 0x08) /* NOT IMPLEMENTED */
+		err = -ENOSYS;
+	else if (buf[0] == 0x0a) /* REJECTED */
+		err = -EINVAL;
+	else if (buf[0] == 0x0b) /* IN TRANSITION */
+		err = -EAGAIN;
+	if (err < 0)
+		goto end;
+
+	*num = buf[7];
+	err = 0;
+end:
+	kfree(buf);
+	return err;
+}
+
+static inline void
+avc_bridgeco_fill_extension_addr(u8 *buf, u8 *addr)
+{
+	buf[1] = addr[0];
+	memcpy(buf + 4, addr + 1, 5);
+}
+
+static inline void
+avc_bridgeco_fill_plug_info_extension_command(u8 *buf, u8 *addr,
+					      unsigned int itype)
+{
+	buf[0] = 0x01;	/* AV/C STATUS */
+	buf[2] = 0x02;	/* AV/C GENERAL PLUG INFO */
+	buf[3] = 0xc0;	/* BridgeCo extension */
+	avc_bridgeco_fill_extension_addr(buf, addr);
+	buf[9] = itype;	/* info type */
+}
+
+int avc_bridgeco_get_plug_type(struct fw_unit *unit,
+			       u8 addr[AVC_BRIDGECO_ADDR_BYTES],
+			       enum avc_bridgeco_plug_type *type)
+{
+	u8 *buf;
+	int err;
+
+	buf = kzalloc(12, GFP_KERNEL);
+	if (buf == NULL)
+		return -ENOMEM;
+
+	/* Info type is 'plug type'. */
+	avc_bridgeco_fill_plug_info_extension_command(buf, addr, 0x00);
+
+	err = fcp_avc_transaction(unit, buf, 12, buf, 12,
+				  BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) |
+				  BIT(6) | BIT(7) | BIT(9));
+	if ((err >= 0) && (err < 8))
+		err = -EIO;
+	else if (buf[0] == 0x08) /* NOT IMPLEMENTED */
+		err = -ENOSYS;
+	else if (buf[0] == 0x0a) /* REJECTED */
+		err = -EINVAL;
+	else if (buf[0] == 0x0b) /* IN TRANSITION */
+		err = -EAGAIN;
+	if (err < 0)
+		goto end;
+
+	*type = buf[10];
+	err = 0;
+end:
+	kfree(buf);
+	return err;
+}
+
+int avc_bridgeco_get_plug_ch_pos(struct fw_unit *unit,
+				 u8 addr[AVC_BRIDGECO_ADDR_BYTES],
+				 u8 *buf, unsigned int len)
+{
+	int err;
+
+	/* Info type is 'channel position'. */
+	avc_bridgeco_fill_plug_info_extension_command(buf, addr, 0x03);
+
+	err = fcp_avc_transaction(unit, buf, 12, buf, 256,
+				  BIT(1) | BIT(2) | BIT(3) | BIT(4) |
+				  BIT(5) | BIT(6) | BIT(7) | BIT(9));
+	if ((err >= 0) && (err < 8))
+		err = -EIO;
+	else if (buf[0] == 0x08) /* NOT IMPLEMENTED */
+		err = -ENOSYS;
+	else if (buf[0] == 0x0a) /* REJECTED */
+		err = -EINVAL;
+	else if (buf[0] == 0x0b) /* IN TRANSITION */
+		err = -EAGAIN;
+	if (err < 0)
+		goto end;
+
+	/* Pick up specific data. */
+	memmove(buf, buf + 10, err - 10);
+	err = 0;
+end:
+	return err;
+}
+
+int avc_bridgeco_get_plug_section_type(struct fw_unit *unit,
+				       u8 addr[AVC_BRIDGECO_ADDR_BYTES],
+				       unsigned int id, u8 *type)
+{
+	u8 *buf;
+	int err;
+
+	/* section info includes charactors but this module don't need it */
+	buf = kzalloc(12, GFP_KERNEL);
+	if (buf == NULL)
+		return -ENOMEM;
+
+	/* Info type is 'section info'. */
+	avc_bridgeco_fill_plug_info_extension_command(buf, addr, 0x07);
+	buf[10] = 0xff & ++id;	/* section id */
+
+	err = fcp_avc_transaction(unit, buf, 12, buf, 12,
+				  BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) |
+				  BIT(6) | BIT(7) | BIT(9) | BIT(10));
+	if ((err >= 0) && (err < 8))
+		err = -EIO;
+	else if (buf[0] == 0x08) /* NOT IMPLEMENTED */
+		err = -ENOSYS;
+	else if (buf[0] == 0x0a) /* REJECTED */
+		err = -EINVAL;
+	else if (buf[0] == 0x0b) /* IN TRANSITION */
+		err = -EAGAIN;
+	if (err < 0)
+		goto end;
+
+	*type = buf[11];
+	err = 0;
+end:
+	kfree(buf);
+	return err;
+}
+
+int avc_bridgeco_get_plug_input(struct fw_unit *unit,
+				u8 addr[AVC_BRIDGECO_ADDR_BYTES], u8 input[7])
+{
+	int err;
+	u8 *buf;
+
+	buf = kzalloc(18, GFP_KERNEL);
+	if (buf == NULL)
+		return -ENOMEM;
+
+	/* Info type is 'plug input'. */
+	avc_bridgeco_fill_plug_info_extension_command(buf, addr, 0x05);
+
+	err = fcp_avc_transaction(unit, buf, 16, buf, 16,
+				  BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) |
+				  BIT(6) | BIT(7));
+	if ((err >= 0) && (err < 8))
+		err = -EIO;
+	else if (buf[0] == 0x08) /* NOT IMPLEMENTED */
+		err = -ENOSYS;
+	else if (buf[0] == 0x0a) /* REJECTED */
+		err = -EINVAL;
+	else if (buf[0] == 0x0b) /* IN TRANSITION */
+		err = -EAGAIN;
+	if (err < 0)
+		goto end;
+
+	memcpy(input, buf + 10, 5);
+	err = 0;
+end:
+	kfree(buf);
+	return err;
+}
+
+int avc_bridgeco_get_plug_strm_fmt(struct fw_unit *unit,
+				   u8 addr[AVC_BRIDGECO_ADDR_BYTES], u8 *buf,
+				   unsigned int *len, unsigned int eid)
+{
+	int err;
+
+	/* check given buffer */
+	if ((buf == NULL) || (*len < 12)) {
+		err = -EINVAL;
+		goto end;
+	}
+
+	buf[0] = 0x01;	/* AV/C STATUS */
+	buf[2] = 0x2f;	/* AV/C STREAM FORMAT SUPPORT */
+	buf[3] = 0xc1;	/* Bridgeco extension - List Request */
+	avc_bridgeco_fill_extension_addr(buf, addr);
+	buf[10] = 0xff & eid;	/* Entry ID */
+
+	err = fcp_avc_transaction(unit, buf, 12, buf, *len,
+				  BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) |
+				  BIT(6) | BIT(7) | BIT(10));
+	if ((err >= 0) && (err < 12))
+		err = -EIO;
+	else if (buf[0] == 0x08)        /* NOT IMPLEMENTED */
+		err = -ENOSYS;
+	else if (buf[0] == 0x0a)        /* REJECTED */
+		err = -EINVAL;
+	else if (buf[0] == 0x0b)        /* IN TRANSITION */
+		err = -EAGAIN;
+	else if (buf[10] != eid)
+		err = -EIO;
+	if (err < 0)
+		goto end;
+
+	/* Pick up 'stream format info'. */
+	memmove(buf, buf + 11, err - 11);
+	*len = err - 11;
+	err = 0;
+end:
+	return err;
+}

sound/firewire/bebob/bebob_focusrite.c

+/*
+ * bebob_focusrite.c - a part of driver for BeBoB based devices
+ *
+ * Copyright (c) 2013-2014 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "./bebob.h"
+
+#define ANA_IN	"Analog In"
+#define DIG_IN	"Digital In"
+#define ANA_OUT	"Analog Out"
+#define DIG_OUT	"Digital Out"
+#define STM_IN	"Stream In"
+
+#define SAFFIRE_ADDRESS_BASE			0x000100000000ULL
+
+#define SAFFIRE_OFFSET_CLOCK_SOURCE		0x00f8
+#define SAFFIREPRO_OFFSET_CLOCK_SOURCE		0x0174
+
+/* whether sync to external device or not */
+#define SAFFIRE_OFFSET_CLOCK_SYNC_EXT		0x013c
+#define SAFFIRE_LE_OFFSET_CLOCK_SYNC_EXT	0x0432
+#define SAFFIREPRO_OFFSET_CLOCK_SYNC_EXT	0x0164
+
+#define SAFFIRE_CLOCK_SOURCE_INTERNAL		0
+#define SAFFIRE_CLOCK_SOURCE_SPDIF		1
+
+/* '1' is absent, why... */
+#define SAFFIREPRO_CLOCK_SOURCE_INTERNAL	0
+#define SAFFIREPRO_CLOCK_SOURCE_SPDIF		2
+#define SAFFIREPRO_CLOCK_SOURCE_ADAT1		3
+#define SAFFIREPRO_CLOCK_SOURCE_ADAT2		4
+#define SAFFIREPRO_CLOCK_SOURCE_WORDCLOCK	5
+
+/* S/PDIF, ADAT1, ADAT2 is enabled or not. three quadlets */
+#define SAFFIREPRO_ENABLE_DIG_IFACES		0x01a4
+
+/* saffirepro has its own parameter for sampling frequency */
+#define SAFFIREPRO_RATE_NOREBOOT		0x01cc
+/* index is the value for this register */
+static const unsigned int rates[] = {
+	[0] = 0,
+	[1] = 44100,
+	[2] = 48000,
+	[3] = 88200,
+	[4] = 96000,
+	[5] = 176400,
+	[6] = 192000
+};
+
+/* saffire(no label)/saffire LE has metering */
+#define SAFFIRE_OFFSET_METER			0x0100
+#define SAFFIRE_LE_OFFSET_METER			0x0168
+
+static inline int
+saffire_read_block(struct snd_bebob *bebob, u64 offset,
+		   u32 *buf, unsigned int size)
+{
+	unsigned int i;
+	int err;
+	__be32 *tmp = (__be32 *)buf;
+
+	err =  snd_fw_transaction(bebob->unit, TCODE_READ_BLOCK_REQUEST,
+				  SAFFIRE_ADDRESS_BASE + offset,
+				  tmp, size, 0);
+	if (err < 0)
+		goto end;
+
+	for (i = 0; i < size / sizeof(u32); i++)
+		buf[i] = be32_to_cpu(tmp[i]);
+end:
+	return err;
+}
+
+static inline int
+saffire_read_quad(struct snd_bebob *bebob, u64 offset, u32 *value)
+{
+	int err;
+	__be32 tmp;
+
+	err = snd_fw_transaction(bebob->unit, TCODE_READ_QUADLET_REQUEST,
+				 SAFFIRE_ADDRESS_BASE + offset,
+				 &tmp, sizeof(__be32), 0);
+	if (err < 0)
+		goto end;
+
+	*value = be32_to_cpu(tmp);
+end:
+	return err;
+}
+
+static inline int
+saffire_write_quad(struct snd_bebob *bebob, u64 offset, u32 value)
+{
+	__be32 data = cpu_to_be32(value);
+
+	return snd_fw_transaction(bebob->unit, TCODE_WRITE_QUADLET_REQUEST,
+				  SAFFIRE_ADDRESS_BASE + offset,
+				  &data, sizeof(__be32), 0);
+}
+
+static char *const saffirepro_26_clk_src_labels[] = {
+	SND_BEBOB_CLOCK_INTERNAL, "S/PDIF", "ADAT1", "ADAT2", "Word Clock"
+};
+
+static char *const saffirepro_10_clk_src_labels[] = {
+	SND_BEBOB_CLOCK_INTERNAL, "S/PDIF", "Word Clock"
+};
+static int
+saffirepro_both_clk_freq_get(struct snd_bebob *bebob, unsigned int *rate)
+{
+	u32 id;
+	int err;
+
+	err = saffire_read_quad(bebob, SAFFIREPRO_RATE_NOREBOOT, &id);
+	if (err < 0)
+		goto end;
+	if (id >= ARRAY_SIZE(rates))
+		err = -EIO;
+	else
+		*rate = rates[id];
+end:
+	return err;
+}
+static int
+saffirepro_both_clk_freq_set(struct snd_bebob *bebob, unsigned int rate)
+{
+	u32 id;
+
+	for (id = 0; id < ARRAY_SIZE(rates); id++) {
+		if (rates[id] == rate)
+			break;
+	}
+	if (id == ARRAY_SIZE(rates))
+		return -EINVAL;
+
+	return saffire_write_quad(bebob, SAFFIREPRO_RATE_NOREBOOT, id);
+}
+static int
+saffirepro_both_clk_src_get(struct snd_bebob *bebob, unsigned int *id)
+{
+	int err;
+	u32 value;
+
+	err = saffire_read_quad(bebob, SAFFIREPRO_OFFSET_CLOCK_SOURCE, &value);
+	if (err < 0)
+		goto end;
+
+	if (bebob->spec->clock->labels == saffirepro_10_clk_src_labels) {
+		if (value == SAFFIREPRO_CLOCK_SOURCE_WORDCLOCK)
+			*id = 2;
+		else if (value == SAFFIREPRO_CLOCK_SOURCE_SPDIF)
+			*id = 1;
+	} else if (value > 1) {
+		*id = value - 1;
+	}
+end:
+	return err;
+}
+
+struct snd_bebob_spec saffire_le_spec;
+static char *const saffire_both_clk_src_labels[] = {
+	SND_BEBOB_CLOCK_INTERNAL, "S/PDIF"
+};
+static int
+saffire_both_clk_src_get(struct snd_bebob *bebob, unsigned int *id)
+{
+	int err;
+	u32 value;
+
+	err = saffire_read_quad(bebob, SAFFIRE_OFFSET_CLOCK_SOURCE, &value);
+	if (err >= 0)
+		*id = 0xff & value;
+
+	return err;
+};
+static char *const saffire_le_meter_labels[] = {
+	ANA_IN, ANA_IN, DIG_IN,
+	ANA_OUT, ANA_OUT, ANA_OUT, ANA_OUT,
+	STM_IN, STM_IN
+};
+static char *const saffire_meter_labels[] = {
+	ANA_IN, ANA_IN,
+	STM_IN, STM_IN, STM_IN, STM_IN, STM_IN,
+};
+static int
+saffire_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size)
+{
+	struct snd_bebob_meter_spec *spec = bebob->spec->meter;
+	unsigned int channels;
+	u64 offset;
+	int err;
+
+	if (spec->labels == saffire_le_meter_labels)
+		offset = SAFFIRE_LE_OFFSET_METER;
+	else
+		offset = SAFFIRE_OFFSET_METER;
+
+	channels = spec->num * 2;
+	if (size < channels * sizeof(u32))
+		return -EIO;
+
+	err = saffire_read_block(bebob, offset, buf, size);
+	if (err >= 0 && spec->labels == saffire_le_meter_labels) {
+		swap(buf[1], buf[3]);
+		swap(buf[2], buf[3]);
+		swap(buf[3], buf[4]);
+
+		swap(buf[7], buf[10]);
+		swap(buf[8], buf[10]);
+		swap(buf[9], buf[11]);
+		swap(buf[11], buf[12]);
+
+		swap(buf[15], buf[16]);
+	}
+
+	return err;
+}
+
+static struct snd_bebob_rate_spec saffirepro_both_rate_spec = {
+	.get	= &saffirepro_both_clk_freq_get,
+	.set	= &saffirepro_both_clk_freq_set,
+};
+/* Saffire Pro 26 I/O  */
+static struct snd_bebob_clock_spec saffirepro_26_clk_spec = {
+	.num	= ARRAY_SIZE(saffirepro_26_clk_src_labels),
+	.labels	= saffirepro_26_clk_src_labels,
+	.get	= &saffirepro_both_clk_src_get,
+};
+struct snd_bebob_spec saffirepro_26_spec = {
+	.clock	= &saffirepro_26_clk_spec,
+	.rate	= &saffirepro_both_rate_spec,
+	.meter	= NULL
+};
+/* Saffire Pro 10 I/O */
+static struct snd_bebob_clock_spec saffirepro_10_clk_spec = {
+	.num	= ARRAY_SIZE(saffirepro_10_clk_src_labels),
+	.labels	= saffirepro_10_clk_src_labels,
+	.get	= &saffirepro_both_clk_src_get,
+};
+struct snd_bebob_spec saffirepro_10_spec = {
+	.clock	= &saffirepro_10_clk_spec,
+	.rate	= &saffirepro_both_rate_spec,
+	.meter	= NULL
+};
+
+static struct snd_bebob_rate_spec saffire_both_rate_spec = {
+	.get	= &snd_bebob_stream_get_rate,
+	.set	= &snd_bebob_stream_set_rate,
+};
+static struct snd_bebob_clock_spec saffire_both_clk_spec = {
+	.num	= ARRAY_SIZE(saffire_both_clk_src_labels),
+	.labels	= saffire_both_clk_src_labels,
+	.get	= &saffire_both_clk_src_get,
+};
+/* Saffire LE */
+static struct snd_bebob_meter_spec saffire_le_meter_spec = {
+	.num	= ARRAY_SIZE(saffire_le_meter_labels),
+	.labels	= saffire_le_meter_labels,
+	.get	= &saffire_meter_get,
+};
+struct snd_bebob_spec saffire_le_spec = {
+	.clock	= &saffire_both_clk_spec,
+	.rate	= &saffire_both_rate_spec,
+	.meter	= &saffire_le_meter_spec
+};
+/* Saffire */
+static struct snd_bebob_meter_spec saffire_meter_spec = {
+	.num	= ARRAY_SIZE(saffire_meter_labels),
+	.labels	= saffire_meter_labels,
+	.get	= &saffire_meter_get,
+};
+struct snd_bebob_spec saffire_spec = {
+	.clock	= &saffire_both_clk_spec,
+	.rate	= &saffire_both_rate_spec,
+	.meter	= &saffire_meter_spec
+};

sound/firewire/bebob/bebob_hwdep.c

+/*
+ * bebob_hwdep.c - a part of driver for BeBoB based devices
+ *
+ * Copyright (c) 2013-2014 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+/*
+ * This codes give three functionality.
+ *
+ * 1.get firewire node infomation
+ * 2.get notification about starting/stopping stream
+ * 3.lock/unlock stream
+ */
+
+#include "bebob.h"
+
+static long
+hwdep_read(struct snd_hwdep *hwdep, char __user *buf,  long count,
+	   loff_t *offset)
+{
+	struct snd_bebob *bebob = hwdep->private_data;
+	DEFINE_WAIT(wait);
+	union snd_firewire_event event;
+
+	spin_lock_irq(&bebob->lock);
+
+	while (!bebob->dev_lock_changed) {
+		prepare_to_wait(&bebob->hwdep_wait, &wait, TASK_INTERRUPTIBLE);
+		spin_unlock_irq(&bebob->lock);
+		schedule();
+		finish_wait(&bebob->hwdep_wait, &wait);
+		if (signal_pending(current))
+			return -ERESTARTSYS;
+		spin_lock_irq(&bebob->lock);
+	}
+
+	memset(&event, 0, sizeof(event));
+	if (bebob->dev_lock_changed) {
+		event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS;
+		event.lock_status.status = (bebob->dev_lock_count > 0);
+		bebob->dev_lock_changed = false;
+
+		count = min_t(long, count, sizeof(event.lock_status));
+	}
+
+	spin_unlock_irq(&bebob->lock);
+
+	if (copy_to_user(buf, &event, count))
+		return -EFAULT;
+
+	return count;
+}
+
+static unsigned int
+hwdep_poll(struct snd_hwdep *hwdep, struct file *file, poll_table *wait)
+{
+	struct snd_bebob *bebob = hwdep->private_data;
+	unsigned int events;
+
+	poll_wait(file, &bebob->hwdep_wait, wait);
+
+	spin_lock_irq(&bebob->lock);
+	if (bebob->dev_lock_changed)
+		events = POLLIN | POLLRDNORM;
+	else
+		events = 0;
+	spin_unlock_irq(&bebob->lock);
+
+	return events;
+}
+
+static int
+hwdep_get_info(struct snd_bebob *bebob, void __user *arg)
+{
+	struct fw_device *dev = fw_parent_device(bebob->unit);
+	struct snd_firewire_get_info info;
+
+	memset(&info, 0, sizeof(info));
+	info.type = SNDRV_FIREWIRE_TYPE_BEBOB;
+	info.card = dev->card->index;
+	*(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]);
+	*(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]);
+	strlcpy(info.device_name, dev_name(&dev->device),
+		sizeof(info.device_name));
+
+	if (copy_to_user(arg, &info, sizeof(info)))
+		return -EFAULT;
+
+	return 0;
+}
+
+static int
+hwdep_lock(struct snd_bebob *bebob)
+{
+	int err;
+
+	spin_lock_irq(&bebob->lock);
+
+	if (bebob->dev_lock_count == 0) {
+		bebob->dev_lock_count = -1;
+		err = 0;
+	} else {
+		err = -EBUSY;
+	}
+
+	spin_unlock_irq(&bebob->lock);
+
+	return err;
+}
+
+static int
+hwdep_unlock(struct snd_bebob *bebob)
+{
+	int err;
+
+	spin_lock_irq(&bebob->lock);
+
+	if (bebob->dev_lock_count == -1) {
+		bebob->dev_lock_count = 0;
+		err = 0;
+	} else {
+		err = -EBADFD;
+	}
+
+	spin_unlock_irq(&bebob->lock);
+
+	return err;
+}
+
+static int
+hwdep_release(struct snd_hwdep *hwdep, struct file *file)
+{
+	struct snd_bebob *bebob = hwdep->private_data;
+
+	spin_lock_irq(&bebob->lock);
+	if (bebob->dev_lock_count == -1)
+		bebob->dev_lock_count = 0;
+	spin_unlock_irq(&bebob->lock);
+
+	return 0;
+}
+
+static int
+hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file,
+	    unsigned int cmd, unsigned long arg)
+{
+	struct snd_bebob *bebob = hwdep->private_data;
+
+	switch (cmd) {
+	case SNDRV_FIREWIRE_IOCTL_GET_INFO:
+		return hwdep_get_info(bebob, (void __user *)arg);
+	case SNDRV_FIREWIRE_IOCTL_LOCK:
+		return hwdep_lock(bebob);
+	case SNDRV_FIREWIRE_IOCTL_UNLOCK:
+		return hwdep_unlock(bebob);
+	default:
+		return -ENOIOCTLCMD;
+	}
+}
+
+#ifdef CONFIG_COMPAT
+static int
+hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file,
+		   unsigned int cmd, unsigned long arg)
+{
+	return hwdep_ioctl(hwdep, file, cmd,
+			   (unsigned long)compat_ptr(arg));
+}
+#else
+#define hwdep_compat_ioctl NULL
+#endif
+
+static const struct snd_hwdep_ops hwdep_ops = {
+	.read		= hwdep_read,
+	.release	= hwdep_release,
+	.poll		= hwdep_poll,
+	.ioctl		= hwdep_ioctl,
+	.ioctl_compat	= hwdep_compat_ioctl,
+};
+
+int snd_bebob_create_hwdep_device(struct snd_bebob *bebob)
+{
+	struct snd_hwdep *hwdep;
+	int err;
+
+	err = snd_hwdep_new(bebob->card, "BeBoB", 0, &hwdep);
+	if (err < 0)
+		goto end;
+	strcpy(hwdep->name, "BeBoB");
+	hwdep->iface = SNDRV_HWDEP_IFACE_FW_BEBOB;
+	hwdep->ops = hwdep_ops;
+	hwdep->private_data = bebob;
+	hwdep->exclusive = true;
+end:
+	return err;
+}
+

sound/firewire/bebob/bebob_midi.c

+/*
+ * bebob_midi.c - a part of driver for BeBoB based devices
+ *
+ * Copyright (c) 2013-2014 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "bebob.h"
+
+static int midi_capture_open(struct snd_rawmidi_substream *substream)
+{
+	struct snd_bebob *bebob = substream->rmidi->private_data;
+	int err;
+
+	err = snd_bebob_stream_lock_try(bebob);
+	if (err < 0)
+		goto end;
+
+	atomic_inc(&bebob->capture_substreams);
+	err = snd_bebob_stream_start_duplex(bebob, 0);
+	if (err < 0)
+		snd_bebob_stream_lock_release(bebob);
+end:
+	return err;
+}
+
+static int midi_playback_open(struct snd_rawmidi_substream *substream)
+{
+	struct snd_bebob *bebob = substream->rmidi->private_data;
+	int err;
+
+	err = snd_bebob_stream_lock_try(bebob);
+	if (err < 0)
+		goto end;
+
+	atomic_inc(&bebob->playback_substreams);
+	err = snd_bebob_stream_start_duplex(bebob, 0);
+	if (err < 0)
+		snd_bebob_stream_lock_release(bebob);
+end:
+	return err;
+}
+
+static int midi_capture_close(struct snd_rawmidi_substream *substream)
+{
+	struct snd_bebob *bebob = substream->rmidi->private_data;
+
+	atomic_dec(&bebob->capture_substreams);
+	snd_bebob_stream_stop_duplex(bebob);
+
+	snd_bebob_stream_lock_release(bebob);
+	return 0;
+}
+
+static int midi_playback_close(struct snd_rawmidi_substream *substream)
+{
+	struct snd_bebob *bebob = substream->rmidi->private_data;
+
+	atomic_dec(&bebob->playback_substreams);
+	snd_bebob_stream_stop_duplex(bebob);
+
+	snd_bebob_stream_lock_release(bebob);
+	return 0;
+}
+
+static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
+{
+	struct snd_bebob *bebob = substrm->rmidi->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&bebob->lock, flags);
+
+	if (up)
+		amdtp_stream_midi_trigger(&bebob->tx_stream,
+					  substrm->number, substrm);
+	else
+		amdtp_stream_midi_trigger(&bebob->tx_stream,
+					  substrm->number, NULL);
+
+	spin_unlock_irqrestore(&bebob->lock, flags);
+}
+
+static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
+{
+	struct snd_bebob *bebob = substrm->rmidi->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&bebob->lock, flags);
+
+	if (up)
+		amdtp_stream_midi_trigger(&bebob->rx_stream,
+					  substrm->number, substrm);
+	else
+		amdtp_stream_midi_trigger(&bebob->rx_stream,
+					  substrm->number, NULL);
+
+	spin_unlock_irqrestore(&bebob->lock, flags);
+}
+
+static struct snd_rawmidi_ops midi_capture_ops = {
+	.open		= midi_capture_open,
+	.close		= midi_capture_close,
+	.trigger	= midi_capture_trigger,
+};
+
+static struct snd_rawmidi_ops midi_playback_ops = {
+	.open		= midi_playback_open,
+	.close		= midi_playback_close,
+	.trigger	= midi_playback_trigger,
+};
+
+static void set_midi_substream_names(struct snd_bebob *bebob,
+				     struct snd_rawmidi_str *str)
+{
+	struct snd_rawmidi_substream *subs;
+
+	list_for_each_entry(subs, &str->substreams, list) {
+		snprintf(subs->name, sizeof(subs->name),
+			 "%s MIDI %d",
+			 bebob->card->shortname, subs->number + 1);
+	}
+}
+
+int snd_bebob_create_midi_devices(struct snd_bebob *bebob)
+{
+	struct snd_rawmidi *rmidi;
+	struct snd_rawmidi_str *str;
+	int err;
+
+	/* create midi ports */
+	err = snd_rawmidi_new(bebob->card, bebob->card->driver, 0,
+			      bebob->midi_output_ports, bebob->midi_input_ports,
+			      &rmidi);
+	if (err < 0)
+		return err;
+
+	snprintf(rmidi->name, sizeof(rmidi->name),
+		 "%s MIDI", bebob->card->shortname);
+	rmidi->private_data = bebob;
+
+	if (bebob->midi_input_ports > 0) {
+		rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
+
+		snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+				    &midi_capture_ops);
+
+		str = &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT];
+
+		set_midi_substream_names(bebob, str);
+	}
+
+	if (bebob->midi_output_ports > 0) {
+		rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
+
+		snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+				    &midi_playback_ops);
+
+		str = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
+
+		set_midi_substream_names(bebob, str);
+	}
+
+	if ((bebob->midi_output_ports > 0) && (bebob->midi_input_ports > 0))
+		rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX;
+
+	return 0;
+}

sound/firewire/cmp.h

@@ -7,12 +7,17 @@
 
 struct fw_unit;
 
+enum cmp_direction {
+	CMP_INPUT = 0,
+	CMP_OUTPUT,
+};
+
 /**
  * struct cmp_connection - manages an isochronous connection to a device
  * @speed: the connection's actual speed
  *
- * This structure manages (using CMP) an isochronous stream from the local
- * computer to a device's input plug (iPCR).
+ * This structure manages (using CMP) an isochronous stream between the local
+ * computer and a device's input plug (iPCR) and output plug (oPCR).
  *
  * There is no corresponding oPCR created on the local computer, so it is not
  * possible to overlay connections on top of this one.
@@ -26,11 +31,14 @@ struct cmp_connection {
 	__be32 last_pcr_value;
 	unsigned int pcr_index;
 	unsigned int max_speed;
+	enum cmp_direction direction;
 };
 
 int cmp_connection_init(struct cmp_connection *connection,
 			struct fw_unit *unit,
-			unsigned int ipcr_index);
+			enum cmp_direction direction,
+			unsigned int pcr_index);
+int cmp_connection_check_used(struct cmp_connection *connection, bool *used);
 void cmp_connection_destroy(struct cmp_connection *connection);
 
 int cmp_connection_establish(struct cmp_connection *connection,

sound/firewire/fireworks/Makefile

+snd-fireworks-objs := fireworks_transaction.o fireworks_command.o \
+		      fireworks_stream.o fireworks_proc.o fireworks_midi.o \
+		      fireworks_pcm.o fireworks_hwdep.o fireworks.o
+obj-m += snd-fireworks.o

sound/pci/hda/Makefile

 snd-hda-intel-objs := hda_intel.o
 snd-hda-controller-objs := hda_controller.o
+snd-hda-tegra-objs := hda_tegra.o
 # for haswell power well
 snd-hda-intel-$(CONFIG_SND_HDA_I915) +=	hda_i915.o
 
@@ -47,3 +48,4 @@ obj-$(CONFIG_SND_HDA_CODEC_HDMI) += snd-hda-codec-hdmi.o
 # otherwise the codec patches won't be hooked before the PCI probe
 # when built in kernel
 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o
+obj-$(CONFIG_SND_HDA_TEGRA) += snd-hda-tegra.o