Merge tag 'sound-4.16-rc1' of git://git.kernel.org/pub/scm/linux/kern…

…el/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "The major changes in the core API side in this cycle are the still
  on-going ASoC componentization works. Other than that, only few small
  changes such as 20bit PCM format support are found.

  Meanwhile the rest majority of changes are for ASoC drivers:

   - Large cleanups of some of the TI CODEC drivers

   - Continued work on Intel ASoC stuff for new quirks, ACPI GPIO
     handling, Kconfigs and lots of cleanups

   - Refactoring of the Freescale SSI driver, as preliminary work for
     the upcoming changes

   - Work on ST DFSDM driver, including the required IIO patches

   - New drivers for Allwinner A83T, Maxim MAX89373, SocioNext UiniPhier
     EVEA Tempo Semiconductor TSCS42xx and TI PCM816x, TAS5722 and
     TAS6424 devices

   - Removal of dead codes for SN95031 and board drivers

  Last but not least, a few HD-audio and USB-audio quirks are included
  as usual, too"

* tag 'sound-4.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (303 commits)
  ALSA: hda - Reduce the suspend time consumption for ALC256
  ASoC: use seq_file to dump the contents of dai_list,platform_list and codec_list
  ASoC: soc-core: add missing EXPORT_SYMBOL_GPL() for snd_soc_rtdcom_lookup
  IIO: ADC: stm32-dfsdm: remove unused variable again
  ASoC: bcm2835: fix hw_params error when device is in prepared state
  ASoC: mxs-sgtl5000: Do not print error on probe deferral
  ASoC: sgtl5000: Do not print error on probe deferral
  ASoC: Intel: remove select on non-existing SND_SOC_INTEL_COMMON
  ALSA: usb-audio: Support changing input on Sound Blaster E1
  ASoC: Intel: remove second duplicated assignment to pointer 'res'
  ALSA: hda/realtek - update ALC215 depop optimize
  ALSA: hda/realtek - Support headset mode for ALC215/ALC285/ALC289
  ALSA: pcm: Fix trailing semicolon
  ASoC: add Component level .read/.write
  ASoC: cx20442: fix regression by adding back .read/.write
  ASoC: uda1380: fix regression by adding back .read/.write
  ASoC: tlv320dac33: fix regression by adding back .read/.write
  ALSA: hda - Use IS_REACHABLE() for dependency on input
  IIO: ADC: stm32-dfsdm: fix static check warning
  IIO: ADC: stm32-dfsdm: code optimization
  ...

Documentation/ABI/testing/sysfs-bus-iio-dfsdm-adc-stm32

@@ -0,0 +1,16 @@
+What:		/sys/bus/iio/devices/iio:deviceX/in_voltage_spi_clk_freq
+KernelVersion:	4.14
+Contact:	arnaud.pouliquen@st.com
+Description:
+		For audio purpose only.
+		Used by audio driver to set/get the spi input frequency.
+		This is mandatory if DFSDM is slave on SPI bus, to
+		provide information on the SPI clock frequency during runtime
+		Notice that the SPI frequency should be a multiple of sample
+		frequency to ensure the precision.
+		if DFSDM input is SPI master
+			Reading  SPI clkout frequency,
+			error on writing
+		If DFSDM input is SPI Slave:
+			Reading returns value previously set.
+			Writing value before starting conversions.

Documentation/devicetree/bindings/iio/adc/sigma-delta-modulator.txt

@@ -0,0 +1,13 @@
+Device-Tree bindings for sigma delta modulator
+
+Required properties:
+- compatible: should be "ads1201", "sd-modulator". "sd-modulator" can be use
+	as a generic SD modulator if modulator not specified in compatible list.
+- #io-channel-cells = <1>: See the IIO bindings section "IIO consumers".
+
+Example node:
+
+	ads1202: adc@0 {
+		compatible = "sd-modulator";
+		#io-channel-cells = <1>;
+	};

Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.txt

@@ -0,0 +1,128 @@
+STMicroelectronics STM32 DFSDM ADC device driver
+
+
+STM32 DFSDM ADC is a sigma delta analog-to-digital converter dedicated to
+interface external sigma delta modulators to STM32 micro controllers.
+It is mainly targeted for:
+- Sigma delta modulators (motor control, metering...)
+- PDM microphones (audio digital microphone)
+
+It features up to 8 serial digital interfaces (SPI or Manchester) and
+up to 4 filters on stm32h7.
+
+Each child node match with a filter instance.
+
+Contents of a STM32 DFSDM root node:
+------------------------------------
+Required properties:
+- compatible: Should be "st,stm32h7-dfsdm".
+- reg: Offset and length of the DFSDM block register set.
+- clocks: IP and serial interfaces clocking. Should be set according
+		to rcc clock ID and "clock-names".
+- clock-names: Input clock name "dfsdm" must be defined,
+		"audio" is optional. If defined CLKOUT is based on the audio
+		clock, else "dfsdm" is used.
+- #interrupt-cells = <1>;
+- #address-cells = <1>;
+- #size-cells = <0>;
+
+Optional properties:
+- spi-max-frequency: Requested only for SPI master mode.
+		  SPI clock OUT frequency (Hz). This clock must be set according
+		  to "clock" property. Frequency must be a multiple of the rcc
+		  clock frequency. If not, SPI CLKOUT frequency will not be
+		  accurate.
+
+Contents of a STM32 DFSDM child nodes:
+--------------------------------------
+
+Required properties:
+- compatible: Must be:
+	"st,stm32-dfsdm-adc" for sigma delta ADCs
+	"st,stm32-dfsdm-dmic" for audio digital microphone.
+- reg: Specifies the DFSDM filter instance used.
+- interrupts: IRQ lines connected to each DFSDM filter instance.
+- st,adc-channels:	List of single-ended channels muxed for this ADC.
+			valid values:
+				"st,stm32h7-dfsdm" compatibility: 0 to 7.
+- st,adc-channel-names:	List of single-ended channel names.
+- st,filter-order:  SinC filter order from 0 to 5.
+			0: FastSinC
+			[1-5]: order 1 to 5.
+			For audio purpose it is recommended to use order 3 to 5.
+- #io-channel-cells = <1>: See the IIO bindings section "IIO consumers".
+
+Required properties for "st,stm32-dfsdm-adc" compatibility:
+- io-channels: From common IIO binding. Used to pipe external sigma delta
+		modulator or internal ADC output to DFSDM channel.
+		This is not required for "st,stm32-dfsdm-pdm" compatibility as
+		PDM microphone is binded in Audio DT node.
+
+Required properties for "st,stm32-dfsdm-pdm" compatibility:
+- #sound-dai-cells: Must be set to 0.
+- dma: DMA controller phandle and DMA request line associated to the
+		filter instance (specified by the field "reg")
+- dma-names: Must be "rx"
+
+Optional properties:
+- st,adc-channel-types:	Single-ended channel input type.
+			- "SPI_R": SPI with data on rising edge (default)
+			- "SPI_F": SPI with data on falling edge
+			- "MANCH_R": manchester codec, rising edge = logic 0
+			- "MANCH_F": manchester codec, falling edge = logic 1
+- st,adc-channel-clk-src: Conversion clock source.
+			  - "CLKIN": external SPI clock (CLKIN x)
+			  - "CLKOUT": internal SPI clock (CLKOUT) (default)
+			  - "CLKOUT_F": internal SPI clock divided by 2 (falling edge).
+			  - "CLKOUT_R": internal SPI clock divided by 2 (rising edge).
+
+- st,adc-alt-channel: Must be defined if two sigma delta modulator are
+			  connected on same SPI input.
+			  If not set, channel n is connected to SPI input n.
+			  If set, channel n is connected to SPI input n + 1.
+
+- st,filter0-sync: Set to 1 to synchronize with DFSDM filter instance 0.
+		   Used for multi microphones synchronization.
+
+Example of a sigma delta adc connected on DFSDM SPI port 0
+and a pdm microphone connected on DFSDM SPI port 1:
+
+	ads1202: simple_sd_adc@0 {
+		compatible = "ads1202";
+		#io-channel-cells = <1>;
+	};
+
+	dfsdm: dfsdm@40017000 {
+		compatible = "st,stm32h7-dfsdm";
+		reg = <0x40017000 0x400>;
+		clocks = <&rcc DFSDM1_CK>;
+		clock-names = "dfsdm";
+		#interrupt-cells = <1>;
+		#address-cells = <1>;
+		#size-cells = <0>;
+
+		dfsdm_adc0: filter@0 {
+			compatible = "st,stm32-dfsdm-adc";
+			#io-channel-cells = <1>;
+			reg = <0>;
+			interrupts = <110>;
+			st,adc-channels = <0>;
+			st,adc-channel-names = "sd_adc0";
+			st,adc-channel-types = "SPI_F";
+			st,adc-channel-clk-src = "CLKOUT";
+			io-channels = <&ads1202 0>;
+			st,filter-order = <3>;
+		};
+		dfsdm_pdm1: filter@1 {
+			compatible = "st,stm32-dfsdm-dmic";
+			reg = <1>;
+			interrupts = <111>;
+			dmas = <&dmamux1 102 0x400 0x00>;
+			dma-names = "rx";
+			st,adc-channels = <1>;
+			st,adc-channel-names = "dmic1";
+			st,adc-channel-types = "SPI_R";
+			st,adc-channel-clk-src = "CLKOUT";
+			st,filter-order = <5>;
+		};
+	}

Documentation/devicetree/bindings/sound/dmic.txt

@@ -7,10 +7,12 @@ Required properties:
 
 Optional properties:
 	- dmicen-gpios: GPIO specifier for dmic to control start and stop
+	- num-channels: Number of microphones on this DAI
 
 Example node:
 
 	dmic_codec: dmic@0 {
 		compatible = "dmic-codec";
 		dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>;
+		num-channels = <1>;
 	};

Documentation/devicetree/bindings/sound/max98373.txt

@@ -0,0 +1,40 @@
+Maxim Integrated MAX98373 Speaker Amplifier
+
+This device supports I2C.
+
+Required properties:
+
+ - compatible : "maxim,max98373"
+
+ - reg : the I2C address of the device.
+
+Optional properties:
+
+  - maxim,vmon-slot-no : slot number used to send voltage information
+                   or in inteleave mode this will be used as
+                   interleave slot.
+                   slot range : 0 ~ 15,  Default : 0
+
+  - maxim,imon-slot-no : slot number used to send current information
+                   slot range : 0 ~ 15,  Default : 0
+
+  - maxim,spkfb-slot-no : slot number used to send speaker feedback information
+                   slot range : 0 ~ 15,  Default : 0
+
+  - maxim,interleave-mode : For cases where a single combined channel
+		   for the I/V sense data is not sufficient, the device can also be configured
+		   to share a single data output channel on alternating frames.
+		   In this configuration, the current and voltage data will be frame interleaved
+		   on a single output channel.
+                   Boolean, define to enable the interleave mode, Default : false
+
+Example:
+
+codec: max98373@31 {
+   compatible = "maxim,max98373";
+   reg = <0x31>;
+   maxim,vmon-slot-no = <0>;
+   maxim,imon-slot-no = <1>;
+   maxim,spkfb-slot-no = <2>;
+   maxim,interleave-mode;
+};