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forked fromtorvalds/linux

Commit1c1f395

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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 ...
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What:/sys/bus/iio/devices/iio:deviceX/in_voltage_spi_clk_freq
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KernelVersion:4.14
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Contact:arnaud.pouliquen@st.com
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Description:
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For audio purpose only.
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Used by audio driver to set/get the spi input frequency.
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This is mandatory if DFSDM is slave on SPI bus, to
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provide information on the SPI clock frequency during runtime
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Notice that the SPI frequency should be a multiple of sample
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frequency to ensure the precision.
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if DFSDM input is SPI master
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Reading SPI clkout frequency,
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error on writing
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If DFSDM input is SPI Slave:
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Reading returns value previously set.
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Writing value before starting conversions.
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Device-Tree bindings for sigma delta modulator
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Required properties:
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- compatible: should be "ads1201", "sd-modulator". "sd-modulator" can be use
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as a generic SD modulator if modulator not specified in compatible list.
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- #io-channel-cells = <1>: See the IIO bindings section "IIO consumers".
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Example node:
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ads1202: adc@0 {
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compatible = "sd-modulator";
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#io-channel-cells = <1>;
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};
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STMicroelectronics STM32 DFSDM ADC device driver
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STM32 DFSDM ADC is a sigma delta analog-to-digital converter dedicated to
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interface external sigma delta modulators to STM32 micro controllers.
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It is mainly targeted for:
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- Sigma delta modulators (motor control, metering...)
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- PDM microphones (audio digital microphone)
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It features up to 8 serial digital interfaces (SPI or Manchester) and
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up to 4 filters on stm32h7.
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Each child node match with a filter instance.
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Contents of a STM32 DFSDM root node:
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------------------------------------
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Required properties:
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- compatible: Should be "st,stm32h7-dfsdm".
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- reg: Offset and length of the DFSDM block register set.
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- clocks: IP and serial interfaces clocking. Should be set according
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to rcc clock ID and "clock-names".
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- clock-names: Input clock name "dfsdm" must be defined,
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"audio" is optional. If defined CLKOUT is based on the audio
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clock, else "dfsdm" is used.
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- #interrupt-cells = <1>;
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- #address-cells = <1>;
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- #size-cells = <0>;
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Optional properties:
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- spi-max-frequency: Requested only for SPI master mode.
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SPI clock OUT frequency (Hz). This clock must be set according
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to "clock" property. Frequency must be a multiple of the rcc
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clock frequency. If not, SPI CLKOUT frequency will not be
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accurate.
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Contents of a STM32 DFSDM child nodes:
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--------------------------------------
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Required properties:
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- compatible: Must be:
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"st,stm32-dfsdm-adc" for sigma delta ADCs
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"st,stm32-dfsdm-dmic" for audio digital microphone.
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- reg: Specifies the DFSDM filter instance used.
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- interrupts: IRQ lines connected to each DFSDM filter instance.
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- st,adc-channels:List of single-ended channels muxed for this ADC.
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valid values:
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"st,stm32h7-dfsdm" compatibility: 0 to 7.
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- st,adc-channel-names:List of single-ended channel names.
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- st,filter-order: SinC filter order from 0 to 5.
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0: FastSinC
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[1-5]: order 1 to 5.
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For audio purpose it is recommended to use order 3 to 5.
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- #io-channel-cells = <1>: See the IIO bindings section "IIO consumers".
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Required properties for "st,stm32-dfsdm-adc" compatibility:
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- io-channels: From common IIO binding. Used to pipe external sigma delta
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modulator or internal ADC output to DFSDM channel.
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This is not required for "st,stm32-dfsdm-pdm" compatibility as
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PDM microphone is binded in Audio DT node.
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Required properties for "st,stm32-dfsdm-pdm" compatibility:
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- #sound-dai-cells: Must be set to 0.
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- dma: DMA controller phandle and DMA request line associated to the
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filter instance (specified by the field "reg")
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- dma-names: Must be "rx"
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Optional properties:
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- st,adc-channel-types:Single-ended channel input type.
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- "SPI_R": SPI with data on rising edge (default)
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- "SPI_F": SPI with data on falling edge
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- "MANCH_R": manchester codec, rising edge = logic 0
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- "MANCH_F": manchester codec, falling edge = logic 1
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- st,adc-channel-clk-src: Conversion clock source.
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- "CLKIN": external SPI clock (CLKIN x)
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- "CLKOUT": internal SPI clock (CLKOUT) (default)
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- "CLKOUT_F": internal SPI clock divided by 2 (falling edge).
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- "CLKOUT_R": internal SPI clock divided by 2 (rising edge).
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- st,adc-alt-channel: Must be defined if two sigma delta modulator are
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connected on same SPI input.
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If not set, channel n is connected to SPI input n.
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If set, channel n is connected to SPI input n + 1.
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- st,filter0-sync: Set to 1 to synchronize with DFSDM filter instance 0.
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Used for multi microphones synchronization.
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Example of a sigma delta adc connected on DFSDM SPI port 0
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and a pdm microphone connected on DFSDM SPI port 1:
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ads1202: simple_sd_adc@0 {
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compatible = "ads1202";
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#io-channel-cells = <1>;
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};
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dfsdm: dfsdm@40017000 {
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compatible = "st,stm32h7-dfsdm";
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reg = <0x40017000 0x400>;
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clocks = <&rcc DFSDM1_CK>;
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clock-names = "dfsdm";
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#interrupt-cells = <1>;
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#address-cells = <1>;
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#size-cells = <0>;
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dfsdm_adc0: filter@0 {
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compatible = "st,stm32-dfsdm-adc";
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#io-channel-cells = <1>;
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reg = <0>;
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interrupts = <110>;
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st,adc-channels = <0>;
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st,adc-channel-names = "sd_adc0";
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st,adc-channel-types = "SPI_F";
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st,adc-channel-clk-src = "CLKOUT";
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io-channels = <&ads1202 0>;
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st,filter-order = <3>;
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};
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dfsdm_pdm1: filter@1 {
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compatible = "st,stm32-dfsdm-dmic";
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reg = <1>;
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interrupts = <111>;
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dmas = <&dmamux1 102 0x400 0x00>;
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dma-names = "rx";
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st,adc-channels = <1>;
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st,adc-channel-names = "dmic1";
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st,adc-channel-types = "SPI_R";
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st,adc-channel-clk-src = "CLKOUT";
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st,filter-order = <5>;
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};
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}

‎Documentation/devicetree/bindings/sound/dmic.txt‎

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Optional properties:
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- dmicen-gpios: GPIO specifier for dmic to control start and stop
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- num-channels: Number of microphones on this DAI
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Example node:
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dmic_codec: dmic@0 {
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compatible = "dmic-codec";
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dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>;
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num-channels = <1>;
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};
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Maxim Integrated MAX98373 Speaker Amplifier
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This device supports I2C.
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Required properties:
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- compatible : "maxim,max98373"
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- reg : the I2C address of the device.
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Optional properties:
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- maxim,vmon-slot-no : slot number used to send voltage information
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or in inteleave mode this will be used as
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interleave slot.
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slot range : 0 ~ 15, Default : 0
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- maxim,imon-slot-no : slot number used to send current information
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slot range : 0 ~ 15, Default : 0
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- maxim,spkfb-slot-no : slot number used to send speaker feedback information
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slot range : 0 ~ 15, Default : 0
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- maxim,interleave-mode : For cases where a single combined channel
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for the I/V sense data is not sufficient, the device can also be configured
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to share a single data output channel on alternating frames.
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In this configuration, the current and voltage data will be frame interleaved
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on a single output channel.
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Boolean, define to enable the interleave mode, Default : false
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Example:
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codec: max98373@31 {
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compatible = "maxim,max98373";
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reg = <0x31>;
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maxim,vmon-slot-no = <0>;
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maxim,imon-slot-no = <1>;
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maxim,spkfb-slot-no = <2>;
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maxim,interleave-mode;
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};

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