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Commit 1c1f395b authored by Linus Torvalds's avatar Linus Torvalds
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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
  ...
parents 49f9c355 1c9609e3
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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.
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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>;
	};
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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>;
		};
	}
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@@ -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>;
	};
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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;
};
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