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authorLinus Torvalds <torvalds@linux-foundation.org>2018-01-29 09:41:47 -0800
committerLinus Torvalds <torvalds@linux-foundation.org>2018-01-29 09:41:47 -0800
commit1c1f395b2873f59830979cf82324fbf00edfb80c (patch)
treee84c9b53a4d4bdb91ec9f4f5c059dc38dad21c76 /Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt
parent49f9c3552ccc30f4f98c45d94d7f9b335596913f (diff)
parent1c9609e3a8cf5997bd35205cfda1ff2218ee793b (diff)
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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+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>;
+ };
+ };
+ };