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

24 files changed, 625 insertions, 160 deletions

diff --git a/Documentation/ABI/testing/sysfs-bus-iio-dfsdm-adc-stm32 b/Documentation/ABI/testing/sysfs-bus-iio-dfsdm-adc-stm32
new file mode 100644
index 000000000000..da9822309f07
--- /dev/null
+++ b/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.
\ No newline at end of file
diff --git a/Documentation/devicetree/bindings/iio/adc/sigma-delta-modulator.txt b/Documentation/devicetree/bindings/iio/adc/sigma-delta-modulator.txt
new file mode 100644
index 000000000000..e9ebb8a20e0d
--- /dev/null
+++ b/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>;
+	};
diff --git a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.txt b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.txt
new file mode 100644
index 000000000000..911492da48f3
--- /dev/null
+++ b/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>;
+		};
+	}
diff --git a/Documentation/devicetree/bindings/sound/dmic.txt b/Documentation/devicetree/bindings/sound/dmic.txt
index 54c8ef6498a8..f7bf65611453 100644
--- a/Documentation/devicetree/bindings/sound/dmic.txt
+++ b/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>;
 	};
diff --git a/Documentation/devicetree/bindings/sound/max98373.txt b/Documentation/devicetree/bindings/sound/max98373.txt
new file mode 100644
index 000000000000..456cb1c59353
--- /dev/null
+++ b/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;
+};
diff --git a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt
index 77a57f84bed4..6df87b97f7cb 100644
--- a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt
+++ b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt
@@ -2,153 +2,143 @@ Mediatek AFE PCM controller for mt2701
 
 Required properties:
 - compatible = "mediatek,mt2701-audio";
-- reg: register location and size
 - interrupts: should contain AFE and ASYS interrupts
 - interrupt-names: should be "afe" and "asys"
 - power-domains: should define the power domain
+- clocks: Must contain an entry for each entry in clock-names
+  See ../clocks/clock-bindings.txt for details
 - clock-names: should have these clock names:
 		"infra_sys_audio_clk",
 		"top_audio_mux1_sel",
 		"top_audio_mux2_sel",
-		"top_audio_mux1_div",
-		"top_audio_mux2_div",
-		"top_audio_48k_timing",
-		"top_audio_44k_timing",
-		"top_audpll_mux_sel",
-		"top_apll_sel",
-		"top_aud1_pll_98M",
-		"top_aud2_pll_90M",
-		"top_hadds2_pll_98M",
-		"top_hadds2_pll_294M",
-		"top_audpll",
-		"top_audpll_d4",
-		"top_audpll_d8",
-		"top_audpll_d16",
-		"top_audpll_d24",
-		"top_audintbus_sel",
-		"clk_26m",
-		"top_syspll1_d4",
-		"top_aud_k1_src_sel",
-		"top_aud_k2_src_sel",
-		"top_aud_k3_src_sel",
-		"top_aud_k4_src_sel",
-		"top_aud_k5_src_sel",
-		"top_aud_k6_src_sel",
-		"top_aud_k1_src_div",
-		"top_aud_k2_src_div",
-		"top_aud_k3_src_div",
-		"top_aud_k4_src_div",
-		"top_aud_k5_src_div",
-		"top_aud_k6_src_div",
-		"top_aud_i2s1_mclk",
-		"top_aud_i2s2_mclk",
-		"top_aud_i2s3_mclk",
-		"top_aud_i2s4_mclk",
-		"top_aud_i2s5_mclk",
-		"top_aud_i2s6_mclk",
-		"top_asm_m_sel",
-		"top_asm_h_sel",
-		"top_univpll2_d4",
-		"top_univpll2_d2",
-		"top_syspll_d5";
+		"top_audio_a1sys_hp",
+		"top_audio_a2sys_hp",
+		"i2s0_src_sel",
+		"i2s1_src_sel",
+		"i2s2_src_sel",
+		"i2s3_src_sel",
+		"i2s0_src_div",
+		"i2s1_src_div",
+		"i2s2_src_div",
+		"i2s3_src_div",
+		"i2s0_mclk_en",
+		"i2s1_mclk_en",
+		"i2s2_mclk_en",
+		"i2s3_mclk_en",
+		"i2so0_hop_ck",
+		"i2so1_hop_ck",
+		"i2so2_hop_ck",
+		"i2so3_hop_ck",
+		"i2si0_hop_ck",
+		"i2si1_hop_ck",
+		"i2si2_hop_ck",
+		"i2si3_hop_ck",
+		"asrc0_out_ck",
+		"asrc1_out_ck",
+		"asrc2_out_ck",
+		"asrc3_out_ck",
+		"audio_afe_pd",
+		"audio_afe_conn_pd",
+		"audio_a1sys_pd",
+		"audio_a2sys_pd",
+		"audio_mrgif_pd";
+- assigned-clocks: list of input clocks and dividers for the audio system.
+		   See ../clocks/clock-bindings.txt for details.
+- assigned-clocks-parents: parent of input clocks of assigned clocks.
+- assigned-clock-rates: list of clock frequencies of assigned clocks.
+
+Must be a subnode of MediaTek audsys device tree node.
+See ../arm/mediatek/mediatek,audsys.txt for details about the parent node.
 
 Example:
 
-	afe: mt2701-afe-pcm@11220000 {
-		compatible = "mediatek,mt2701-audio";
-		reg = <0 0x11220000 0 0x2000>,
-		      <0 0x112A0000 0 0x20000>;
-		interrupts = <GIC_SPI 104 IRQ_TYPE_LEVEL_LOW>,
-			     <GIC_SPI 132 IRQ_TYPE_LEVEL_LOW>;
-		interrupt-names	= "afe", "asys";
-		power-domains = <&scpsys MT2701_POWER_DOMAIN_IFR_MSC>;
-		clocks = <&infracfg CLK_INFRA_AUDIO>,
-			 <&topckgen CLK_TOP_AUD_MUX1_SEL>,
-			 <&topckgen CLK_TOP_AUD_MUX2_SEL>,
-			 <&topckgen CLK_TOP_AUD_MUX1_DIV>,
-			 <&topckgen CLK_TOP_AUD_MUX2_DIV>,
-			 <&topckgen CLK_TOP_AUD_48K_TIMING>,
-			 <&topckgen CLK_TOP_AUD_44K_TIMING>,
-			 <&topckgen CLK_TOP_AUDPLL_MUX_SEL>,
-			 <&topckgen CLK_TOP_APLL_SEL>,
-			 <&topckgen CLK_TOP_AUD1PLL_98M>,
-			 <&topckgen CLK_TOP_AUD2PLL_90M>,
-			 <&topckgen CLK_TOP_HADDS2PLL_98M>,
-			 <&topckgen CLK_TOP_HADDS2PLL_294M>,
-			 <&topckgen CLK_TOP_AUDPLL>,
-			 <&topckgen CLK_TOP_AUDPLL_D4>,
-			 <&topckgen CLK_TOP_AUDPLL_D8>,
-			 <&topckgen CLK_TOP_AUDPLL_D16>,
-			 <&topckgen CLK_TOP_AUDPLL_D24>,
-			 <&topckgen CLK_TOP_AUDINTBUS_SEL>,
-			 <&clk26m>,
-			 <&topckgen CLK_TOP_SYSPLL1_D4>,
-			 <&topckgen CLK_TOP_AUD_K1_SRC_SEL>,
-			 <&topckgen CLK_TOP_AUD_K2_SRC_SEL>,
-			 <&topckgen CLK_TOP_AUD_K3_SRC_SEL>,
-			 <&topckgen CLK_TOP_AUD_K4_SRC_SEL>,
-			 <&topckgen CLK_TOP_AUD_K5_SRC_SEL>,
-			 <&topckgen CLK_TOP_AUD_K6_SRC_SEL>,
-			 <&topckgen CLK_TOP_AUD_K1_SRC_DIV>,
-			 <&topckgen CLK_TOP_AUD_K2_SRC_DIV>,
-			 <&topckgen CLK_TOP_AUD_K3_SRC_DIV>,
-			 <&topckgen CLK_TOP_AUD_K4_SRC_DIV>,
-			 <&topckgen CLK_TOP_AUD_K5_SRC_DIV>,
-			 <&topckgen CLK_TOP_AUD_K6_SRC_DIV>,
-			 <&topckgen CLK_TOP_AUD_I2S1_MCLK>,
-			 <&topckgen CLK_TOP_AUD_I2S2_MCLK>,
-			 <&topckgen CLK_TOP_AUD_I2S3_MCLK>,
-			 <&topckgen CLK_TOP_AUD_I2S4_MCLK>,
-			 <&topckgen CLK_TOP_AUD_I2S5_MCLK>,
-			 <&topckgen CLK_TOP_AUD_I2S6_MCLK>,
-			 <&topckgen CLK_TOP_ASM_M_SEL>,
-			 <&topckgen CLK_TOP_ASM_H_SEL>,
-			 <&topckgen CLK_TOP_UNIVPLL2_D4>,
-			 <&topckgen CLK_TOP_UNIVPLL2_D2>,
-			 <&topckgen CLK_TOP_SYSPLL_D5>;
+	audsys: audio-subsystem@11220000 {
+		compatible = "mediatek,mt2701-audsys", "syscon", "simple-mfd";
+		...
+
+		afe: audio-controller {
+			compatible = "mediatek,mt2701-audio";
+			interrupts =  <GIC_SPI 104 IRQ_TYPE_LEVEL_LOW>,
+				      <GIC_SPI 132 IRQ_TYPE_LEVEL_LOW>;
+			interrupt-names	= "afe", "asys";
+			power-domains = <&scpsys MT2701_POWER_DOMAIN_IFR_MSC>;
+
+			clocks = <&infracfg CLK_INFRA_AUDIO>,
+				 <&topckgen CLK_TOP_AUD_MUX1_SEL>,
+				 <&topckgen CLK_TOP_AUD_MUX2_SEL>,
+				 <&topckgen CLK_TOP_AUD_48K_TIMING>,
+				 <&topckgen CLK_TOP_AUD_44K_TIMING>,
+				 <&topckgen CLK_TOP_AUD_K1_SRC_SEL>,
+				 <&topckgen CLK_TOP_AUD_K2_SRC_SEL>,
+				 <&topckgen CLK_TOP_AUD_K3_SRC_SEL>,
+				 <&topckgen CLK_TOP_AUD_K4_SRC_SEL>,
+				 <&topckgen CLK_TOP_AUD_K1_SRC_DIV>,
+				 <&topckgen CLK_TOP_AUD_K2_SRC_DIV>,
+				 <&topckgen CLK_TOP_AUD_K3_SRC_DIV>,
+				 <&topckgen CLK_TOP_AUD_K4_SRC_DIV>,
+				 <&topckgen CLK_TOP_AUD_I2S1_MCLK>,
+				 <&topckgen CLK_TOP_AUD_I2S2_MCLK>,
+				 <&topckgen CLK_TOP_AUD_I2S3_MCLK>,
+				 <&topckgen CLK_TOP_AUD_I2S4_MCLK>,
+				 <&audsys CLK_AUD_I2SO1>,
+				 <&audsys CLK_AUD_I2SO2>,
+				 <&audsys CLK_AUD_I2SO3>,
+				 <&audsys CLK_AUD_I2SO4>,
+				 <&audsys CLK_AUD_I2SIN1>,
+				 <&audsys CLK_AUD_I2SIN2>,
+				 <&audsys CLK_AUD_I2SIN3>,
+				 <&audsys CLK_AUD_I2SIN4>,
+				 <&audsys CLK_AUD_ASRCO1>,
+				 <&audsys CLK_AUD_ASRCO2>,
+				 <&audsys CLK_AUD_ASRCO3>,
+				 <&audsys CLK_AUD_ASRCO4>,
+				 <&audsys CLK_AUD_AFE>,
+				 <&audsys CLK_AUD_AFE_CONN>,
+				 <&audsys CLK_AUD_A1SYS>,
+				 <&audsys CLK_AUD_A2SYS>,
+				 <&audsys CLK_AUD_AFE_MRGIF>;
+
+			clock-names = "infra_sys_audio_clk",
+				      "top_audio_mux1_sel",
+				      "top_audio_mux2_sel",
+				      "top_audio_a1sys_hp",
+				      "top_audio_a2sys_hp",
+				      "i2s0_src_sel",
+				      "i2s1_src_sel",
+				      "i2s2_src_sel",
+				      "i2s3_src_sel",
+				      "i2s0_src_div",
+				      "i2s1_src_div",
+				      "i2s2_src_div",
+				      "i2s3_src_div",
+				      "i2s0_mclk_en",
+				      "i2s1_mclk_en",
+				      "i2s2_mclk_en",
+				      "i2s3_mclk_en",
+				      "i2so0_hop_ck",
+				      "i2so1_hop_ck",
+				      "i2so2_hop_ck",
+				      "i2so3_hop_ck",
+				      "i2si0_hop_ck",
+				      "i2si1_hop_ck",
+				      "i2si2_hop_ck",
+				      "i2si3_hop_ck",
+				      "asrc0_out_ck",
+				      "asrc1_out_ck",
+				      "asrc2_out_ck",
+				      "asrc3_out_ck",
+				      "audio_afe_pd",
+				      "audio_afe_conn_pd",
+				      "audio_a1sys_pd",
+				      "audio_a2sys_pd",
+				      "audio_mrgif_pd";
 
-		clock-names = "infra_sys_audio_clk",
-			      "top_audio_mux1_sel",
-			      "top_audio_mux2_sel",
-			      "top_audio_mux1_div",
-			      "top_audio_mux2_div",
-			      "top_audio_48k_timing",
-			      "top_audio_44k_timing",
-			      "top_audpll_mux_sel",
-			      "top_apll_sel",
-			      "top_aud1_pll_98M",
-			      "top_aud2_pll_90M",
-			      "top_hadds2_pll_98M",
-			      "top_hadds2_pll_294M",
-			      "top_audpll",
-			      "top_audpll_d4",
-			      "top_audpll_d8",
-			      "top_audpll_d16",
-			      "top_audpll_d24",
-			      "top_audintbus_sel",
-			      "clk_26m",
-			      "top_syspll1_d4",
-			      "top_aud_k1_src_sel",
-			      "top_aud_k2_src_sel",
-			      "top_aud_k3_src_sel",
-			      "top_aud_k4_src_sel",
-			      "top_aud_k5_src_sel",
-			      "top_aud_k6_src_sel",
-			      "top_aud_k1_src_div",
-			      "top_aud_k2_src_div",
-			      "top_aud_k3_src_div",
-			      "top_aud_k4_src_div",
-			      "top_aud_k5_src_div",
-			      "top_aud_k6_src_div",
-			      "top_aud_i2s1_mclk",
-			      "top_aud_i2s2_mclk",
-			      "top_aud_i2s3_mclk",
-			      "top_aud_i2s4_mclk",
-			      "top_aud_i2s5_mclk",
-			      "top_aud_i2s6_mclk",
-			      "top_asm_m_sel",
-			      "top_asm_h_sel",
-			      "top_univpll2_d4",
-			      "top_univpll2_d2",
-			      "top_syspll_d5";
+			assigned-clocks = <&topckgen CLK_TOP_AUD_MUX1_SEL>,
+					  <&topckgen CLK_TOP_AUD_MUX2_SEL>,
+					  <&topckgen CLK_TOP_AUD_MUX1_DIV>,
+					  <&topckgen CLK_TOP_AUD_MUX2_DIV>;
+			assigned-clock-parents = <&topckgen CLK_TOP_AUD1PLL_98M>,
+						 <&topckgen CLK_TOP_AUD2PLL_90M>;
+			assigned-clock-rates = <0>, <0>, <49152000>, <45158400>;
+		};
 	};
diff --git a/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt
index 601c518eddaa..4eb980bd0287 100644
--- a/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt
+++ b/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt
@@ -1,10 +1,31 @@
 * Freescale MXS audio complex with SGTL5000 codec
 
 Required properties:
-- compatible: "fsl,mxs-audio-sgtl5000"
-- model: The user-visible name of this sound complex
-- saif-controllers: The phandle list of the MXS SAIF controller
-- audio-codec: The phandle of the SGTL5000 audio codec
+- compatible		: "fsl,mxs-audio-sgtl5000"
+- model			: The user-visible name of this sound complex
+- saif-controllers	: The phandle list of the MXS SAIF controller
+- audio-codec		: The phandle of the SGTL5000 audio codec
+- audio-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. Valid names could be power supplies, SGTL5000
+			  pins, and the jacks on the board:
+
+			  Power supplies:
+			   * Mic Bias
+
+			  SGTL5000 pins:
+			   * MIC_IN
+			   * LINE_IN
+			   * HP_OUT
+			   * LINE_OUT
+
+			  Board connectors:
+			   * Mic Jack
+			   * Line In Jack
+			   * Headphone Jack
+			   * Line Out Jack
+			   * Ext Spk
 
 Example:
 
@@ -14,4 +35,8 @@ sound {
 	model = "imx28-evk-sgtl5000";
 	saif-controllers = <&saif0 &saif1>;
 	audio-codec = <&sgtl5000>;
+	audio-routing =
+		"MIC_IN", "Mic Jack",
+		"Mic Jack", "Mic Bias",
+		"Headphone Jack", "HP_OUT";
 };
diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt
index 2f5e973285a6..d16d96839bcb 100644
--- a/Documentation/devicetree/bindings/sound/nau8825.txt
+++ b/Documentation/devicetree/bindings/sound/nau8825.txt
@@ -69,7 +69,7 @@ Optional properties:
   - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms
   - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms
 
-  - nuvoton,crosstalk-bypass: make crosstalk function bypass if set.
+  - nuvoton,crosstalk-enable: make crosstalk function enable if set.
 
   - clocks: list of phandle and clock specifier pairs according to common clock bindings for the
       clocks described in clock-names
@@ -98,7 +98,7 @@ Example:
       nuvoton,short-key-debounce = <2>;
       nuvoton,jack-insert-debounce = <7>;
       nuvoton,jack-eject-debounce = <7>;
-      nuvoton,crosstalk-bypass;
+      nuvoton,crosstalk-enable;
 
       clock-names = "mclk";
       clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>;
diff --git a/Documentation/devicetree/bindings/sound/pcm186x.txt b/Documentation/devicetree/bindings/sound/pcm186x.txt
new file mode 100644
index 000000000000..1087f4855980
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/pcm186x.txt
@@ -0,0 +1,42 @@
+Texas Instruments PCM186x Universal Audio ADC
+
+These devices support both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "ti,pcm1862",
+                "ti,pcm1863",
+                "ti,pcm1864",
+                "ti,pcm1865"
+
+ - reg : The I2C address of the device for I2C, the chip select
+         number for SPI.
+
+ - avdd-supply: Analog core power supply (3.3v)
+ - dvdd-supply: Digital core power supply
+ - iovdd-supply: Digital IO power supply
+        See regulator/regulator.txt for more information
+
+CODEC input pins:
+ * VINL1
+ * VINR1
+ * VINL2
+ * VINR2
+ * VINL3
+ * VINR3
+ * VINL4
+ * VINR4
+
+The pins can be used in referring sound node's audio-routing property.
+
+Example:
+
+	pcm186x: audio-codec@4a {
+		compatible = "ti,pcm1865";
+		reg = <0x4a>;
+
+		avdd-supply = <&reg_3v3_analog>;
+		dvdd-supply = <&reg_3v3>;
+		iovdd-supply = <&reg_1v8>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index 085bec364caf..5bed9a595772 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -4,7 +4,7 @@ Renesas R-Car sound
 * Modules
 =============================================
 
-Renesas R-Car sound is constructed from below modules
+Renesas R-Car and RZ/G sound is constructed from below modules
 (for Gen2 or later)
 
  SCU		: Sampling Rate Converter Unit
@@ -197,12 +197,17 @@ Ex)
 	[MEM] -> [SRC2] -> [CTU03] -+
 
 	sound {
+		#address-cells = <1>;
+		#size-cells = <0>;
+
 		compatible = "simple-scu-audio-card";
 		...
-		simple-audio-card,cpu-0 {
+		simple-audio-card,cpu@0 {
+			reg = <0>;
 			sound-dai = <&rcar_sound 0>;
 		};
-		simple-audio-card,cpu-1 {
+		simple-audio-card,cpu@1 {
+			reg = <1>;
 			sound-dai = <&rcar_sound 1>;
 		};
 		simple-audio-card,codec {
@@ -334,9 +339,11 @@ Required properties:
 
 - compatible			: "renesas,rcar_sound-<soctype>", fallbacks
 				  "renesas,rcar_sound-gen1" if generation1, and
-				  "renesas,rcar_sound-gen2" if generation2
+				  "renesas,rcar_sound-gen2" if generation2 (or RZ/G1)
 				  "renesas,rcar_sound-gen3" if generation3
 				  Examples with soctypes are:
+				    - "renesas,rcar_sound-r8a7743" (RZ/G1M)
+				    - "renesas,rcar_sound-r8a7745" (RZ/G1E)
 				    - "renesas,rcar_sound-r8a7778" (R-Car M1A)
 				    - "renesas,rcar_sound-r8a7779" (R-Car H1)
 				    - "renesas,rcar_sound-r8a7790" (R-Car H2)
diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt
index 166f2290233b..17c13e74667d 100644
--- a/Documentation/devicetree/bindings/sound/simple-card.txt
+++ b/Documentation/devicetree/bindings/sound/simple-card.txt
@@ -140,6 +140,7 @@ sound {
 	simple-audio-card,name = "Cubox Audio";
 
 	simple-audio-card,dai-link@0 {		/* I2S - HDMI */
+		reg = <0>;
 		format = "i2s";
 		cpu {
 			sound-dai = <&audio1 0>;
@@ -150,6 +151,7 @@ sound {
 	};
 
 	simple-audio-card,dai-link@1 {		/* S/PDIF - HDMI */
+		reg = <1>;
 		cpu {
 			sound-dai = <&audio1 1>;
 		};
@@ -159,6 +161,7 @@ sound {
 	};
 
 	simple-audio-card,dai-link@2 {		/* S/PDIF - S/PDIF */
+		reg = <2>;
 		cpu {
 			sound-dai = <&audio1 1>;
 		};
diff --git a/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt b/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt
new file mode 100644
index 000000000000..864f5b00b031
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt
@@ -0,0 +1,63 @@
+STMicroelectronics Audio Digital Filter Sigma Delta modulators(DFSDM)
+
+The DFSDM allows PDM microphones capture through SPI interface. The Audio
+interface is seems as a sub block of the DFSDM device.
+For details on DFSDM bindings refer to ../iio/adc/st,stm32-dfsdm-adc.txt
+
+Required properties:
+  - compatible: "st,stm32h7-dfsdm-dai".
+
+  - #sound-dai-cells : Must be equal to 0
+
+  - io-channels : phandle to iio dfsdm instance node.
+
+Example of a sound card using audio DFSDM node.
+
+	sound_card {
+		compatible = "audio-graph-card";
+
+		dais = <&cpu_port>;
+	};
+
+	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-dmic";
+			reg = <0>;
+			interrupts = <110>;
+			dmas = <&dmamux1 101 0x400 0x00>;
+			dma-names = "rx";
+			st,adc-channels = <1>;
+			st,adc-channel-names = "dmic0";
+			st,adc-channel-types = "SPI_R";
+			st,adc-channel-clk-src = "CLKOUT";
+			st,filter-order = <5>;
+
+			dfsdm_dai0: dfsdm-dai {
+				compatible = "st,stm32h7-dfsdm-dai";
+				#sound-dai-cells = <0>;
+				io-channels = <&dfsdm_adc0 0>;
+				cpu_port: port {
+				dfsdm_endpoint: endpoint {
+					remote-endpoint = <&dmic0_endpoint>;
+				};
+			};
+		};
+	};
+
+	dmic0: dmic@0 {
+		compatible = "dmic-codec";
+		#sound-dai-cells = <0>;
+		port {
+			dmic0_endpoint: endpoint {
+				remote-endpoint = <&dfsdm_endpoint>;
+			};
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt
index 1f9cd7095337..b1acc1a256ba 100644
--- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt
+++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt
@@ -20,11 +20,6 @@ Required properties:
 
 Optional properties:
   - resets: Reference to a reset controller asserting the SAI
-  - st,sync: specify synchronization mode.
-	By default SAI sub-block is in asynchronous mode.
-	This property sets SAI sub-block as slave of another SAI sub-block.
-	Must contain the phandle and index of the sai sub-block providing
-	the synchronization.
 
 SAI subnodes:
 Two subnodes corresponding to SAI sub-block instances A et B can be defined.
@@ -44,6 +39,13 @@ SAI subnodes required properties:
   - pinctrl-names: should contain only value "default"
   - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/pinctrl-stm32.txt
 
+SAI subnodes Optional properties:
+  - st,sync: specify synchronization mode.
+	By default SAI sub-block is in asynchronous mode.
+	This property sets SAI sub-block as slave of another SAI sub-block.
+	Must contain the phandle and index of the sai sub-block providing
+	the synchronization.
+
 The device node should contain one 'port' child node with one child 'endpoint'
 node, according to the bindings defined in Documentation/devicetree/bindings/
 graph.txt.
diff --git a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
index 05d7135a8d2f..b9d50d6cdef3 100644
--- a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
+++ b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
@@ -8,6 +8,7 @@ Required properties:
 - compatible: should be one of the following:
    - "allwinner,sun4i-a10-i2s"
    - "allwinner,sun6i-a31-i2s"
+   - "allwinner,sun8i-a83t-i2s"
    - "allwinner,sun8i-h3-i2s"
 - reg: physical base address of the controller and length of memory mapped
   region.
@@ -23,6 +24,7 @@ Required properties:
 
 Required properties for the following compatibles:
 	- "allwinner,sun6i-a31-i2s"
+	- "allwinner,sun8i-a83t-i2s"
 	- "allwinner,sun8i-h3-i2s"
 - resets: phandle to the reset line for this codec
 
diff --git a/Documentation/devicetree/bindings/sound/tas5720.txt b/Documentation/devicetree/bindings/sound/tas5720.txt
index 40d94f82beb3..7481653fe8e3 100644
--- a/Documentation/devicetree/bindings/sound/tas5720.txt
+++ b/Documentation/devicetree/bindings/sound/tas5720.txt
@@ -6,10 +6,12 @@ audio playback. For more product information please see the links below:
 
 http://www.ti.com/product/TAS5720L
 http://www.ti.com/product/TAS5720M
+http://www.ti.com/product/TAS5722L
 
 Required properties:
 
-- compatible : "ti,tas5720"
+- compatible : "ti,tas5720",
+               "ti,tas5722"
 - reg : I2C slave address
 - dvdd-supply : phandle to a 3.3-V supply for the digital circuitry
 - pvdd-supply : phandle to a supply used for the Class-D amp and the analog
diff --git a/Documentation/devicetree/bindings/sound/tfa9879.txt b/Documentation/devicetree/bindings/sound/tfa9879.txt
index 23ba522d9e2b..1620e6848436 100644
--- a/Documentation/devicetree/bindings/sound/tfa9879.txt
+++ b/Documentation/devicetree/bindings/sound/tfa9879.txt
@@ -6,18 +6,18 @@ Required properties:
 
 - reg : the I2C address of the device
 
+- #sound-dai-cells : must be 0.
+
 Example:
 
 &i2c1 {
-	clock-frequency = <100000>;
 	pinctrl-names = "default";
 	pinctrl-0 = <&pinctrl_i2c1>;
-	status = "okay";
 
-	codec: tfa9879@6c {
+	amp: amp@6c {
 		#sound-dai-cells = <0>;
 		compatible = "nxp,tfa9879";
 		reg = <0x6c>;
-        };
+	};
 };
 
diff --git a/Documentation/devicetree/bindings/sound/ti,tas6424.txt b/Documentation/devicetree/bindings/sound/ti,tas6424.txt
new file mode 100644
index 000000000000..1c4ada0eef4e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ti,tas6424.txt
@@ -0,0 +1,20 @@
+Texas Instruments TAS6424 Quad-Channel Audio amplifier
+
+The TAS6424 serial control bus communicates through I2C protocols.
+
+Required properties:
+	- compatible: "ti,tas6424" - TAS6424
+	- reg: I2C slave address
+	- sound-dai-cells: must be equal to 0
+
+Example:
+
+tas6424: tas6424@6a {
+	compatible = "ti,tas6424";
+	reg = <0x6a>;
+
+	#sound-dai-cells = <0>;
+};
+
+For more product information please see the link below:
+http://www.ti.com/product/TAS6424-Q1
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
index 6fbba562eaa7..5b3c33bb99e5 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
@@ -22,7 +22,7 @@ Required properties:
 
 Optional properties:
 
-- gpio-reset - gpio pin number used for codec reset
+- reset-gpios - GPIO specification for the active low RESET input.
 - ai31xx-micbias-vg - MicBias Voltage setting
         1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V
         2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V
@@ -30,6 +30,10 @@ Optional properties:
 	If this node is not mentioned or if the value is unknown, then
 	micbias	is set to 2.0V.
 
+Deprecated properties:
+
+- gpio-reset - gpio pin number used for codec reset
+
 CODEC output pins:
   * HPL
   * HPR
@@ -48,6 +52,7 @@ CODEC input pins:
 The pins can be used in referring sound node's audio-routing property.
 
 Example:
+#include <dt-bindings/gpio/gpio.h>
 #include <dt-bindings/sound/tlv320aic31xx-micbias.h>
 
 tlv320aic31xx: tlv320aic31xx@18 {
@@ -56,6 +61,8 @@ tlv320aic31xx: tlv320aic31xx@18 {
 
 	ai31xx-micbias-vg = <MICBIAS_OFF>;
 
+	reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>;
+
 	HPVDD-supply = <&regulator>;
 	SPRVDD-supply = <&regulator>;
 	SPLVDD-supply = <&regulator>;
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
index ba5b45c483f5..9796c4639262 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
@@ -17,7 +17,7 @@ Required properties:
 
 Optional properties:
 
-- gpio-reset - gpio pin number used for codec reset
+- reset-gpios - GPIO specification for the active low RESET input.
 - ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality
 				    - Not supported on tlv320aic3104
 - ai3x-micbias-vg - MicBias Voltage required.
@@ -34,6 +34,10 @@ Optional properties:
 - AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the
   device as covered in Documentation/devicetree/bindings/regulator/regulator.txt
 
+Deprecated properties:
+
+- gpio-reset - gpio pin number used for codec reset
+
 CODEC output pins:
   * LLOUT
   * RLOUT
@@ -61,10 +65,14 @@ The pins can be used in referring sound node's audio-routing property.
 
 Example:
 
+#include <dt-bindings/gpio/gpio.h>
+
 tlv320aic3x: tlv320aic3x@1b {
 	compatible = "ti,tlv320aic3x";
 	reg = <0x1b>;
 
+	reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>;
+
 	AVDD-supply = <&regulator>;
 	IOVDD-supply = <&regulator>;
 	DRVDD-supply = <&regulator>;
diff --git a/Documentation/devicetree/bindings/sound/tscs42xx.txt b/Documentation/devicetree/bindings/sound/tscs42xx.txt
new file mode 100644
index 000000000000..2ac2f0996697
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tscs42xx.txt
@@ -0,0 +1,16 @@
+TSCS42XX Audio CODEC
+
+Required Properties:
+
+	- compatible :	"tempo,tscs42A1" for analog mic
+			"tempo,tscs42A2" for digital mic
+
+	- reg : 	<0x71> for analog mic
+			<0x69> for digital mic
+
+Example:
+
+wookie: codec@69 {
+	compatible = "tempo,tscs42A2";
+	reg = <0x69>;
+};
diff --git a/Documentation/devicetree/bindings/sound/uniphier,evea.txt b/Documentation/devicetree/bindings/sound/uniphier,evea.txt
new file mode 100644
index 000000000000..3f31b235f18b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/uniphier,evea.txt
@@ -0,0 +1,26 @@
+Socionext EVEA - UniPhier SoC internal codec driver
+
+Required properties:
+- compatible      : should be "socionext,uniphier-evea".
+- reg             : offset and length of the register set for the device.
+- clock-names     : should include following entries:
+                    "evea", "exiv"
+- clocks          : a list of phandle, should contain an entry for each
+                    entries in clock-names.
+- reset-names     : should include following entries:
+                    "evea", "exiv", "adamv"
+- resets          : a list of phandle, should contain reset entries of
+                    reset-names.
+- #sound-dai-cells: should be 1.
+
+Example:
+
+	codec {
+		compatible = "socionext,uniphier-evea";
+		reg = <0x57900000 0x1000>;
+		clock-names = "evea", "exiv";
+		clocks = <&sys_clk 41>, <&sys_clk 42>;
+		reset-names = "evea", "exiv", "adamv";
+		resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>;
+		#sound-dai-cells = <1>;
+	};
diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt
index 0994bdd82cd3..f776fb804a8c 100644
--- a/Documentation/devicetree/bindings/vendor-prefixes.txt
+++ b/Documentation/devicetree/bindings/vendor-prefixes.txt
@@ -347,6 +347,7 @@ tcg	Trusted Computing Group
 tcl	Toby Churchill Ltd.
 technexion	TechNexion
 technologic	Technologic Systems
+tempo	Tempo Semiconductor
 terasic	Terasic Inc.
 thine	THine Electronics, Inc.
 ti	Texas Instruments
diff --git a/Documentation/driver-api/iio/hw-consumer.rst b/Documentation/driver-api/iio/hw-consumer.rst
new file mode 100644
index 000000000000..8facce6a6733
--- /dev/null
+++ b/Documentation/driver-api/iio/hw-consumer.rst
@@ -0,0 +1,51 @@
+===========
+HW consumer
+===========
+An IIO device can be directly connected to another device in hardware. in this
+case the buffers between IIO provider and IIO consumer are handled by hardware.
+The Industrial I/O HW consumer offers a way to bond these IIO devices without
+software buffer for data. The implementation can be found under
+:file:`drivers/iio/buffer/hw-consumer.c`
+
+
+* struct :c:type:`iio_hw_consumer` — Hardware consumer structure
+* :c:func:`iio_hw_consumer_alloc` — Allocate IIO hardware consumer
+* :c:func:`iio_hw_consumer_free` — Free IIO hardware consumer
+* :c:func:`iio_hw_consumer_enable` — Enable IIO hardware consumer
+* :c:func:`iio_hw_consumer_disable` — Disable IIO hardware consumer
+
+
+HW consumer setup
+=================
+
+As standard IIO device the implementation is based on IIO provider/consumer.
+A typical IIO HW consumer setup looks like this::
+
+	static struct iio_hw_consumer *hwc;
+
+	static const struct iio_info adc_info = {
+		.read_raw = adc_read_raw,
+	};
+
+	static int adc_read_raw(struct iio_dev *indio_dev,
+				struct iio_chan_spec const *chan, int *val,
+				int *val2, long mask)
+	{
+		ret = iio_hw_consumer_enable(hwc);
+
+		/* Acquire data */
+
+		ret = iio_hw_consumer_disable(hwc);
+	}
+
+	static int adc_probe(struct platform_device *pdev)
+	{
+		hwc = devm_iio_hw_consumer_alloc(&iio->dev);
+	}
+
+More details
+============
+.. kernel-doc:: include/linux/iio/hw-consumer.h
+.. kernel-doc:: drivers/iio/buffer/industrialio-hw-consumer.c
+   :export:
+
diff --git a/Documentation/driver-api/iio/index.rst b/Documentation/driver-api/iio/index.rst
index e5c3922d1b6f..7fba341bd8b2 100644
--- a/Documentation/driver-api/iio/index.rst
+++ b/Documentation/driver-api/iio/index.rst
@@ -15,3 +15,4 @@ Contents:
    buffers
    triggers
    triggered-buffers
+   hw-consumer