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Commit ce38207f authored by Linus Torvalds's avatar Linus Torvalds
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Merge tag 'sound-4.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound 

Pull sound updates from Takashi Iwai:
 "No dramatic changes are found in this development cycle, but as usual,
  many commits are applied in a wide range of drivers.

  Most of big changes are in ASoC, where a few bits of framework work
  and quite a lot of cleanups and improvements to existing code have
  been done. The rest are usual stuff, a few HD-audio and USB-audio
  quirks and fixes, as well as the drop of kthread usages in the whole
  subsystem.

  Below are some highlights:

  ASoC:
   - support for stereo DAPM controls
   - some initial work on the of-graph sound card
   - regmap conversions of the remaining AC'97 drivers
   - a new version of the topology ABI; this should be backward
     compatible
   - updates / cleanups of rsnd, sunxi, sti, nau8825, samsung, arizona,
     Intel skylake, atom-sst
   - new drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and
     Realtek RT5665

  USB-audio:
   - yet another race fix at disconnection
   - tolerated packet size calculation for some Android devices
   - quirks for Axe-Fx II, QuickCam, TEAC 501/503

  HD-audio:
   - improvement of Dell pin fixup mapping
   - quirks for HP Z1 Gen3, Alienware 15 R2 2016 and ALC622 headset mic

  Misc:
   - replace all kthread usages with simple works"

* tag 'sound-4.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (296 commits)
  ALSA: hiface: Fix M2Tech hiFace driver sampling rate change
  ALSA: usb-audio: Eliminate noise at the start of DSD playback.
  ALSA: usb-audio: Add native DSD support for TEAC 501/503 DAC
  ASoC: wm_adsp: wm_adsp_buf_alloc should use kfree in error path
  ASoC: topology: avoid uninitialized kcontrol_type
  ALSA: usb-audio: Add QuickCam Communicate Deluxe/S7500 to volume_control_quirks
  ALSA: usb-audio: add implicit fb quirk for Axe-Fx II
  ASoC: zte: spdif: correct ZX_SPDIF_CLK_RAT define
  ASoC: zte: spdif and i2s drivers are not zx296702 specific
  ASoC: rsnd: setup BRGCKR/BRRA/BRRB when starting
  ASoC: rsnd: enable/disable ADG when suspend/resume timing
  ASoC: rsnd: tidyup ssi->usrcnt counter check in hw_params
  ALSA: cs46xx: add a new line
  ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture
  ASoC: nau8825: disable sinc filter for high THD of ADC
  ALSA: usb-audio: more tolerant packetsize
  ALSA: usb-audio: avoid setting of sample rate multiple times on bus
  ASoC: cs35l34: Simplify the logic to set CS35L34_MCLK_CTL setting
  ALSA: hda - Gate the mic jack on HP Z1 Gen3 AiO
  ALSA: hda: when comparing pin configurations, ignore assoc in addition to seq
  ...
parents a9042def 995c6a7f

Documentation/ABI/testing/sysfs-platform-sst-atom

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What:		/sys/devices/platform/8086%x:00/firmware_version

Date:		November 2016

KernelVersion:	4.10

Contact:	"Sebastien Guiriec" <sebastien.guiriec@intel.com>

Description:

		LPE Firmware version for SST driver on all atom

		plaforms (BYT/CHT/Merrifield/BSW).

		If the FW has never been loaded it will display:

			"FW not yet loaded"

		If FW has been loaded it will display:

			"v01.aa.bb.cc"

		aa: Major version is reflecting SoC version:

			0d: BYT FW

			0b: BSW FW

			07: Merrifield FW

		bb: Minor version

		cc: Build version

Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt

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Devicetree bindings for the Axentia TSE-850 audio complex



Required properties:

  - compatible: "axentia,tse850-pcm5142"

  - axentia,ssc-controller: The phandle of the atmel SSC controller used as

    cpu dai.

  - axentia,audio-codec: The phandle of the PCM5142 codec.

  - axentia,add-gpios: gpio specifier that controls the mixer.

  - axentia,loop1-gpios: gpio specifier that controls loop relays on channel 1.

  - axentia,loop2-gpios: gpio specifier that controls loop relays on channel 2.

  - axentia,ana-supply: Regulator that supplies the output amplifier. Must

    support voltages in the 2V - 20V range, in 1V steps.



The schematics explaining the gpios are as follows:



               loop1 relays

   IN1 +---o  +------------+  o---+ OUT1

            \                /

             +              +

             |   /          |

             +--o  +--.     |

             |  add   |     |

             |        V     |

             |      .---.   |

   DAC +----------->|Sum|---+

             |      '---'   |

             |              |

             +              +



   IN2 +---o--+------------+--o---+ OUT2

               loop2 relays



The 'loop1' gpio pin controlls two relays, which are either in loop position,

meaning that input and output are directly connected, or they are in mixer

position, meaning that the signal is passed through the 'Sum' mixer. Similarly

for 'loop2'.



In the above, the 'loop1' relays are inactive, thus feeding IN1 to the mixer

(if 'add' is active) and feeding the mixer output to OUT1. The 'loop2' relays

are active, short-cutting the TSE-850 from channel 2. IN1, IN2, OUT1 and OUT2

are TSE-850 connectors and DAC is the PCB name of the (filtered) output from

the PCM5142 codec.



Example:



	&i2c {

		codec: pcm5142@4c {

			compatible = "ti,pcm5142";



			reg = <0x4c>;



			AVDD-supply = <&reg_3v3>;

			DVDD-supply = <&reg_3v3>;

			CPVDD-supply = <&reg_3v3>;



			clocks = <&sck>;



			pll-in = <3>;

			pll-out = <6>;

		};

	};



	ana: ana-reg {

		compatible = "pwm-regulator";



		regulator-name = "ANA";



		pwms = <&pwm0 2 1000 PWM_POLARITY_INVERTED>;

		pwm-dutycycle-unit = <1000>;

		pwm-dutycycle-range = <100 1000>;



		regulator-min-microvolt = <2000000>;

		regulator-max-microvolt = <20000000>;

		regulator-ramp-delay = <1000>;

	};



	sound {

		compatible = "axentia,tse850-pcm5142";



		axentia,ssc-controller = <&ssc0>;

		axentia,audio-codec = <&codec>;



		axentia,add-gpios = <&pioA 8 GPIO_ACTIVE_LOW>;

		axentia,loop1-gpios = <&pioA 10 GPIO_ACTIVE_LOW>;

		axentia,loop2-gpios = <&pioA 11 GPIO_ACTIVE_LOW>;



		axentia,ana-supply = <&ana>;

	};

Documentation/devicetree/bindings/sound/cs35l34.txt

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CS35L34 Speaker Amplifier



Required properties:



  - compatible : "cirrus,cs35l34"



  - reg : the I2C address of the device for I2C.



  - VA-supply, VP-supply : power supplies for the device,

    as covered in

    Documentation/devicetree/bindings/regulator/regulator.txt.



  - cirrus,boost-vtge-millivolt : Boost Voltage Value.  Configures the boost

    converter's output voltage in mV. The range is from VP to 8V with

    increments of 100mV.



  - cirrus,boost-nanohenry: Inductor value for boost converter. The value is

    in nH and they can be values of 1000nH, 1100nH, 1200nH, 1500nH, and 2200nH.



Optional properties:



  - reset-gpios: GPIO used to reset the amplifier.



  - interrupt-parent : Specifies the phandle of the interrupt controller to

    which the IRQs from CS35L34 are delivered to.

  - interrupts : IRQ line info CS35L34.

    (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt

    for further information relating to interrupt properties)



  - cirrus,boost-peak-milliamp : Boost converter peak current limit in mA. The

    range starts at 1200mA and goes to a maximum of 3840mA with increments of

    80mA. The default value is 2480mA.



  - cirrus,i2s-sdinloc : ADSP SDIN I2S channel location. Indicates whether the

    received mono data is in the left or right portion of the I2S frame

    according to the AD0 pin or directly via this configuration.

    0x0 (Default) = Selected by AD0 input (if AD0 = LOW, use left channel),

    0x2 = Left,

    0x1 = Selected by the inversion of the AD0 input (if AD0 = LOW, use right

    channel),

    0x3 = Right.



  - cirrus,gain-zc-disable: Boolean property. If set, the gain change will take

    effect without waiting for a zero cross.



  - cirrus,tdm-rising-edge: Boolean property. If set, data is on the rising edge of

    SCLK. Otherwise, data is on the falling edge of SCLK.





Example:



cs35l34: cs35l34@40 {

	compatible = "cirrus,cs35l34";

	reg = <0x40>;



	interrupt-parent = <&gpio8>;

	interrupts = <3 IRQ_TYPE_LEVEL_LOW>;



	reset-gpios = <&gpio 10 0>;



	cirrus,boost-vtge-milltvolt = <8000>; /* 8V */

	cirrus,boost-ind-nanohenry = <1000>; /* 1uH */

	cirrus,boost-peak-milliamp = <3000>; /* 3A */

};

Documentation/devicetree/bindings/sound/cs42l42.txt

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CS42L42 audio CODEC



Required properties:



  - compatible : "cirrus,cs42l42"



  - reg : the I2C address of the device for I2C.



  - VP-supply, VCP-supply, VD_FILT-supply, VL-supply, VA-supply :

  power supplies for the device, as covered in

  Documentation/devicetree/bindings/regulator/regulator.txt.



Optional properties:



  - reset-gpios : a GPIO spec for the reset pin. If specified, it will be

  deasserted before communication to the codec starts.



  - interrupt-parent : Specifies the phandle of the interrupt controller to

  which the IRQs from CS42L42 are delivered to.



  - interrupts : IRQ line info CS42L42.

  (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt

  for further information relating to interrupt properties)



  - cirrus,ts-inv : Boolean property. For jacks that invert the tip sense

  polarity. Normal jacks will short tip sense pin to HS1 when headphones are

  plugged in and leave tip sense floating when not plugged in. Inverting jacks

  short tip sense when unplugged and float when plugged in.



  0 = (Default) Non-inverted

  1 = Inverted



  - cirrus,ts-dbnc-rise : Debounce the rising edge of TIP_SENSE_PLUG. With no

  debounce, the tip sense pin might be noisy on a plug event.



  0 - 0ms,

  1 - 125ms,

  2 - 250ms,

  3 - 500ms,

  4 - 750ms,

  5 - (Default) 1s,

  6 - 1.25s,

  7 - 1.5s,



  - cirrus,ts-dbnc-fall : Debounce the falling edge of TIP_SENSE_UNPLUG.

  With no debounce, the tip sense pin might be noisy on an unplug event.



  0 - 0ms,

  1 - 125ms,

  2 - 250ms,

  3 - 500ms,

  4 - 750ms,

  5 - (Default) 1s,

  6 - 1.25s,

  7 - 1.5s,



  - cirrus,btn-det-init-dbnce : This sets how long the driver sleeps after

  enabling button detection interrupts. After auto-detection and before

  servicing button interrupts, the HS bias needs time to settle. If you

  don't wait, there is possibility for erroneous button interrupt.



  0ms - 200ms,

  Default = 100ms



  - cirrus,btn-det-event-dbnce : This sets how long the driver delays after

  receiving a button press interrupt. With level detect interrupts, you want

  to wait a small amount of time to make sure the button press is making a

  clean connection with the bias resistors.



  0ms - 20ms,

  Default = 10ms



  - cirrus,bias-lvls : For a level-detect headset button scheme, each button

  will bias the mic pin to a certain voltage. To determine which button was

  pressed, the driver will compare this biased voltage to sequential,

  decreasing voltages and will stop when a comparator is tripped,

  indicating a comparator voltage < bias voltage. This value represents a

  percentage of the internally generated HS bias voltage. For different

  hardware setups, a designer might want to tweak this. This is an array of

  descending values for the comparator voltage.



  Array of 4 values

  Each 0-63

  < x1 x2 x3 x4 >

  Default = < 15 8 4 1>





Example:



cs42l42: cs42l42@48 {

	compatible = "cirrus,cs42l42";

	reg = <0x48>;

	VA-supply = <&dummy_vreg>;

	VP-supply = <&dummy_vreg>;

	VCP-supply = <&dummy_vreg>;

	VD_FILT-supply = <&dummy_vreg>;

	VL-supply = <&dummy_vreg>;



	reset-gpios = <&axi_gpio_0 1 0>;

	interrupt-parent = <&gpio0>;

	interrupts = <55 8>



	cirrus,ts-inv = <0x00>;

	cirrus,ts-dbnc-rise = <0x05>;

	cirrus,ts-dbnc-fall = <0x00>;

	cirrus,btn-det-init-dbnce = <100>;

	cirrus,btn-det-event-dbnce = <10>;

	cirrus,bias-lvls = <0x0F 0x08 0x04 0x01>;

	cirrus,hs-bias-ramp-rate = <0x02>;

};
 
 No newline at end of file

Documentation/devicetree/bindings/sound/davinci-mcbsp.txt

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@@ -43,7 +43,7 @@ mcbsp0: mcbsp@1d10000 { 
		<0x00310000 0x1000>;

	reg-names = "mpu", "dat";

	interrupts = <97 98>;

	interrupts-names = "rx", "tx";

	interrupt-names = "rx", "tx";

	dmas = <&edma0 3 1

		&edma0 2 1>;

	dma-names = "tx", "rx";