August 2022 - Alsa-devel - mailman.alsa-project.org

USB-Audio: Add profile for MSI MEG Z690I Unify
by GitHub pull_request - opened 04 Aug '22

04 Aug '22

alsa-project/alsa-ucm-conf pull request #196 was opened from 13r0ck: This is tested and works, though I am not a fan of my `Needle` and `Haystack` solution here, but before also making a PR to alsa-utils I wanted direction. Would it be better to add a `Type Bool` condition, or a `skip` modifier to `SectionDevice`? I prefer the later, because it skips the `If {} Condition {} true {}` lines, but I would like an alsa maintainer's opinion. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/196 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/196.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ALSA: hda/realtek: Add quirk for HP Spectre x360 15-eb0xxx
by Ivan Hasenkampf 03 Aug '22

03 Aug '22

Fixes speaker output on HP Spectre x360 15-eb0xxx Signed-off-by: Ivan Hasenkampf <ivan.hasenkampf(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f55bc43bfa9..fe43e5807b8c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9072,6 +9072,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x86e7, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), + SND_PCI_QUIRK(0x103c, 0x86e8, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), -- 2.37.0

2 1

[PATCH] ALSA: hda/realtek: Add quirk for HP Spectre x360 15-eb0xxx
by Ivan Hasenkampf 03 Aug '22

03 Aug '22

Fixes speaker output on HP Spectre x360 15-eb0xxx Signed-off-by: Ivan Hasenkampf <ivan.hasenkampf(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f55bc43bfa9..fe43e5807b8c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9072,6 +9072,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x86e7, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), + SND_PCI_QUIRK(0x103c, 0x86e8, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), -- 2.37.0

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[PATCH] dt-bindings: sound: tlv320aic3x: Convert to dtschema
by Jai Luthra 03 Aug '22

03 Aug '22

Convert bindings for TI's TLV320AIC3x audio codecs to dtschema. Signed-off-by: Jai Luthra <j-luthra(a)ti.com> --- .../devicetree/bindings/sound/tlv320aic3x.txt | 97 ------------ .../bindings/sound/tlv320aic3x.yaml | 145 ++++++++++++++++++ 2 files changed, 145 insertions(+), 97 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.yaml diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt deleted file mode 100644 index 20931a63fd64..000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ /dev/null @@ -1,97 +0,0 @@ -Texas Instruments - tlv320aic3x Codec module - -The tlv320aic3x serial control bus communicates through both I2C and SPI bus protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tlv320aic3x" - Generic TLV320AIC3x device - "ti,tlv320aic33" - TLV320AIC33 - "ti,tlv320aic3007" - TLV320AIC3007 - "ti,tlv320aic3106" - TLV320AIC3106 - "ti,tlv320aic3104" - TLV320AIC3104 - - -- reg - <int> - I2C slave address - - -Optional properties: - -- 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. - 1 - MICBIAS output is powered to 2.0V, - 2 - MICBIAS output is powered to 2.5V, - 3 - MICBIAS output is connected to AVDD, - If this node is not mentioned or if the value is incorrect, then MicBias - is powered down. -- ai3x-ocmv - Output Common-Mode Voltage selection: - 0 - 1.35V, - 1 - 1.5V, - 2 - 1.65V, - 3 - 1.8V -- 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 - * MONO_LOUT - * HPLOUT - * HPROUT - * HPLCOM - * HPRCOM - -CODEC input pins for TLV320AIC3104: - * MIC2L - * MIC2R - * LINE1L - * LINE1R - -CODEC input pins for other compatible codecs: - * MIC3L - * MIC3R - * LINE1L - * LINE2L - * LINE1R - * LINE2R - -The pins can be used in referring sound node's audio-routing property. - -I2C 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>; - DVDD-supply = <&regulator>; -}; - -SPI example: - -spi0: spi@f0000000 { - tlv320aic3x: codec@0 { - compatible = "ti,tlv320aic3x"; - reg = <0>; /* CS number */ - #sound-dai-cells = <0>; - spi-max-frequency = <1000000>; - - AVDD-supply = <&regulator>; - IOVDD-supply = <&regulator>; - DRVDD-supply = <&regulator>; - DVDD-supply = <&regulator>; - ai3x-ocmv = <0>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.yaml b/Documentation/devicetree/bindings/sound/tlv320aic3x.yaml new file mode 100644 index 000000000000..6efb1d459543 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.yaml @@ -0,0 +1,145 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/tlv320aic3x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320AIC3x Codec Device Tree Bindings + +maintainers: + - Jai Luthra <j-luthra(a)ti.com> + +properties: + compatible: + enum: + - ti,tlv320aic3x + - ti,tlv320aic33 + - ti,tlv320aic3007 + - ti,tlv320aic3106 + - ti,tlv320aic3104 + + reg: + maxItems: 1 + description: i2c slave address + + reset-gpios: + maxItems: 1 + description: + GPIO specification for the active low RESET input. + + ai3x-gpio-func: + description: AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + $ref: /schemas/types.yaml#/definitions/uint32 + minItems: 3 + maxItems: 3 + + ai3x-micbias-vg: + description: MicBias required voltage. If node is omitted then MicBias is powered down. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 1 + description: MICBIAS output is powered to 2.0V. + - const: 2 + description: MICBIAS output is powered to 2.5V. + - const: 3 + description: MICBIAS output is connected to AVDD. + + ai3x-ocmv: + description: Output Common-Mode Voltage selection. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 0 + description: 1.35V + - const: 1 + description: 1.5V + - const: 2 + description: 1.65V + - const: 3 + description: 1.8V + + AVDD-supply: + description: Analog DAC voltage. + + IOVDD-supply: + description: I/O voltage. + + DRVDD-supply: + description: ADC analog and output driver voltage. + + DVDD-supply: + description: Digital core voltage. + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +#The pins can be used in referring sound node's audio-routing property. + +#CODEC output pins: + # LLOUT + # RLOUT + # MONO_LOUT + # HPLOUT + # HPROUT + # HPLCOM + # HPRCOM + +#CODEC input pins for TLV320AIC3104: + # MIC2L + # MIC2R + # LINE1L + # LINE1R + +#CODEC input pins for other compatible codecs: + # MIC3L + # MIC3R + # LINE1L + # LINE2L + # LINE1R + # LINE2R + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_i2c: tlv320aic3x@1b { + compatible = "ti,tlv320aic3x"; + reg = <0x1b>; + + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DRVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + }; + }; + + - | + #include <dt-bindings/gpio/gpio.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_spi: codec@0 { + compatible = "ti,tlv320aic3x"; + reg = <0>; /* CS number */ + #sound-dai-cells = <0>; + + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DRVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + ai3x-ocmv = <0>; + }; + }; + +... -- 2.17.1

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[PATCH] soc: sof: fix clang -Wformat warnings
by Justin Stitt 03 Aug '22

03 Aug '22

When building with Clang we encounter these warnings: | sound/soc/sof/ipc3-topology.c:2343:4: error: format specifies type | 'unsigned char' but the argument has type 'int' [-Werror,-Wformat] | SOF_ABI_MAJOR, SOF_ABI_MINOR, SOF_ABI_PATCH); | ^~~~~~~~~~~~~~~^~~~~~~~~~~~~~~^~~~~~~~~~~~~ Use correct format specifier `%d` since args are of type int. Link: https://github.com/ClangBuiltLinux/linux/issues/378 Reported-by: Nathan Chancellor <nathan(a)kernel.org> Suggested-by: Nathan Chancellor <nathan(a)kernel.org> Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- Reported by Nathan here: https://lore.kernel.org/all/YtmrCJjQrSbv8Aj1@dev-arch.thelio-3990X/ sound/soc/sof/ipc3-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index b2cc046b9f60..65923e7a5976 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -2338,7 +2338,7 @@ static int sof_ipc3_parse_manifest(struct snd_soc_component *scomp, int index, } dev_info(scomp->dev, - "Topology: ABI %d:%d:%d Kernel ABI %hhu:%hhu:%hhu\n", + "Topology: ABI %d:%d:%d Kernel ABI %d:%d:%d\n", man->priv.data[0], man->priv.data[1], man->priv.data[2], SOF_ABI_MAJOR, SOF_ABI_MINOR, SOF_ABI_PATCH); -- 2.37.1.359.gd136c6c3e2-goog

3 3

[PATCH] ASoC: amd: yc: Update DMI table entries for AMD platforms
by Syed Saba Kareem 02 Aug '22

02 Aug '22

From: syed sabakareem <Syed.SabaKareem(a)amd.com> Updated DMI entries 21EM, 21EN, 21J5 and 21J6 for AMD platforms P15v Gen 3 and P14s Gen 3. Link: https://bugzilla.kernel.org/show_bug.cgi?id=216267 Signed-off-by: syed saba kareem <Syed.SabaKareem(a)amd.com> Reported-by: Pananchikkal Renjith <renjith.pananchikkal(a)amd.com> --- sound/soc/amd/yc/acp6x-mach.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index ecfe7a790790..e0b24e1daef3 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -143,6 +143,34 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21CL"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21EM"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21EN"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21J5"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21J6"), + } + }, {} }; -- 2.25.1

2 1

[PATCH v7 2/2] ASoC: sun50i-dmic: dt-bindings: add DT bindings for DMIC controller
by Ban Tao 02 Aug '22

02 Aug '22

DT binding documentation for this new ASoC driver. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Reviewed-by: Maxime Ripard <maxime(a)cerno.tech> --- v1->v2: 1.Fix some build errors. v2->v3: 1.Fix some build errors. v3->v4: 1.None. v4->v5: 1.Add interrupt. 2.Keep clock and reset index. v5->v6: 1.None. v6->v7: 1.None. --- .../bindings/sound/allwinner,sun50i-h6-dmic.yaml | 79 ++++++++++++++++++++++ 1 file changed, 79 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml new file mode 100644 index 0000000..185d3e4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner H6 DMIC Device Tree Bindings + +maintainers: + - Ban Tao <fengzheng923(a)gmail.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: allwinner,sun50i-h6-dmic + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + + dmas: + items: + - description: RX DMA Channel + + dma-names: + items: + - const: rx + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + #include <dt-bindings/clock/sun50i-h6-ccu.h> + #include <dt-bindings/reset/sun50i-h6-ccu.h> + + dmic: dmic@5095000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun50i-h6-dmic"; + reg = <0x05095000 0x400>; + interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&ccu CLK_BUS_DMIC>, <&ccu CLK_DMIC>; + clock-names = "bus", "mod"; + dmas = <&dma 7>; + dma-names = "rx"; + resets = <&ccu RST_BUS_DMIC>; + }; + +... \ No newline at end of file -- 2.7.4

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[PATCH v3] ASoC: dt-bindings: fsl,sai: Convert format to json-schema
by Shengjiu Wang 02 Aug '22

02 Aug '22

Convert the NXP SAI binding to DT schema format using json-schema. The Synchronous Audio Interface (SAI) provides an interface that supports full-duplex serial interfaces with frame synchronization formats such as I2S, AC97, TDM, and codec/DSP interfaces. Beside conversion, 'fsl,shared-interrupt' and '#sound-dai-cells' are added for they are already used by some dts. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v3 - address comments from Krzysztof changes in v2 - fix exclusive property issue - fix order issue of compatible, clock-names, dma-names .../devicetree/bindings/sound/fsl,sai.yaml | 221 ++++++++++++++++++ .../devicetree/bindings/sound/fsl-sai.txt | 95 -------- 2 files changed, 221 insertions(+), 95 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,sai.yaml delete mode 100644 Documentation/devicetree/bindings/sound/fsl-sai.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml new file mode 100644 index 000000000000..86ee46755b0c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -0,0 +1,221 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Audio Interface (SAI). + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: | + The SAI is based on I2S module that used communicating with audio codecs, + which provides a synchronous audio interface that supports fullduplex + serial interfaces with frame synchronization such as I2S, AC97, TDM, and + codec/DSP interfaces. + +properties: + compatible: + oneOf: + - enum: + - fsl,vf610-sai + - fsl,imx6sx-sai + - fsl,imx6ul-sai + - fsl,imx7ulp-sai + - fsl,imx8mq-sai + - fsl,imx8qm-sai + - fsl,imx8ulp-sai + - items: + - enum: + - fsl,imx8mm-sai + - fsl,imx8mn-sai + - fsl,imx8mp-sai + - const: fsl,imx8mq-sai + + reg: + maxItems: 1 + + interrupts: + items: + - description: receive and transmit interrupt + + dmas: + maxItems: 2 + + dma-names: + maxItems: 2 + + clocks: + items: + - description: The ipg clock for register access + - description: master clock source 0 (obsoleted) + - description: master clock source 1 + - description: master clock source 2 + - description: master clock source 3 + - description: PLL clock source for 8kHz series + - description: PLL clock source for 11kHz series + minItems: 4 + + clock-names: + oneOf: + - items: + - const: bus + - const: mclk0 + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + - items: + - const: bus + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + + lsb-first: + $ref: /schemas/types.yaml#/definitions/flag + description: | + Configures whether the LSB or the MSB is transmitted + first for the fifo data. If this property is absent, + the MSB is transmitted first as default, or the LSB + is transmitted first. + type: boolean + + big-endian: + description: | + required if all the SAI registers are big-endian rather than little-endian. + type: boolean + + fsl,sai-synchronous-rx: + $ref: /schemas/types.yaml#/definitions/flag + description: | + SAI will work in the synchronous mode (sync Tx with Rx) which means + both the transmitter and the receiver will send and receive data by + following receiver's bit clocks and frame sync clocks. + type: boolean + + fsl,sai-asynchronous: + $ref: /schemas/types.yaml#/definitions/flag + description: | + SAI will work in the asynchronous mode, which means both transmitter + and receiver will send and receive data by following their own bit clocks + and frame sync clocks separately. + If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the + default synchronous mode (sync Rx with Tx) will be used, which means both + transmitter and receiver will send and receive data by following clocks + of transmitter. + type: boolean + + fsl,dataline: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + description: | + Configure the dataline. It has 3 value for each configuration + maxItems: 16 + items: + items: + - description: format Default(0), I2S(1) or PDM(2) + enum: [0, 1, 2] + - description: dataline mask for 'rx' + - description: dataline mask for 'tx' + + fsl,sai-mclk-direction-output: + $ref: /schemas/types.yaml#/definitions/flag + description: SAI will output the SAI MCLK clock. + type: boolean + + fsl,shared-interrupt: + $ref: /schemas/types.yaml#/definitions/flag + description: Interrupt is shared with other modules. + type: boolean + + "#sound-dai-cells": + const: 0 + description: optional, some dts node didn't add it. + +allOf: + - if: + properties: + compatible: + contains: + const: fsl,vf610-sai + then: + properties: + dmas: + items: + - description: DMA controller phandle and request line for TX + - description: DMA controller phandle and request line for RX + dma-names: + items: + - const: tx + - const: rx + else: + properties: + dmas: + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + dma-names: + items: + - const: rx + - const: tx + - if: + required: + - fsl,sai-asynchronous + then: + properties: + fsl,sai-synchronous-rx: false + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/vf610-clock.h> + sai2: sai@40031000 { + compatible = "fsl,vf610-sai"; + reg = <0x40031000 0x1000>; + interrupts = <86 IRQ_TYPE_LEVEL_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai2_1>; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; + dma-names = "tx", "rx"; + dmas = <&edma0 0 21>, + <&edma0 0 20>; + big-endian; + lsb-first; + }; + + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mm-clock.h> + sai1: sai@30010000 { + compatible = "fsl,imx8mm-sai", "fsl,imx8mq-sai"; + reg = <0x30010000 0x10000>; + interrupts = <GIC_SPI 95 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clk IMX8MM_CLK_SAI1_IPG>, + <&clk IMX8MM_CLK_DUMMY>, + <&clk IMX8MM_CLK_SAI1_ROOT>, + <&clk IMX8MM_CLK_DUMMY>, <&clk IMX8MM_CLK_DUMMY>; + clock-names = "bus", "mclk0", "mclk1", "mclk2", "mclk3"; + dmas = <&sdma2 0 2 0>, <&sdma2 1 2 0>; + dma-names = "rx", "tx"; + fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt deleted file mode 100644 index fbdefc3fade7..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ /dev/null @@ -1,95 +0,0 @@ -Freescale Synchronous Audio Interface (SAI). - -The SAI is based on I2S module that used communicating with audio codecs, -which provides a synchronous audio interface that supports fullduplex -serial interfaces with frame synchronization such as I2S, AC97, TDM, and -codec/DSP interfaces. - -Required properties: - - - compatible : Compatible list, contains "fsl,vf610-sai", - "fsl,imx6sx-sai", "fsl,imx6ul-sai", - "fsl,imx7ulp-sai", "fsl,imx8mq-sai", - "fsl,imx8qm-sai", "fsl,imx8mm-sai", - "fsl,imx8mn-sai", "fsl,imx8mp-sai", or - "fsl,imx8ulp-sai". - - - reg : Offset and length of the register set for the device. - - - clocks : Must contain an entry for each entry in clock-names. - - - clock-names : Must include the "bus" for register access and - "mclk1", "mclk2", "mclk3" for bit clock and frame - clock providing. - "pll8k", "pll11k" are optional, they are the clock - source for root clock, one is for 8kHz series rates - another one is for 11kHz series rates. - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - pinctrl-names : Must contain a "default" entry. - - - pinctrl-NNN : One property must exist for each entry in - pinctrl-names. See ../pinctrl/pinctrl-bindings.txt - for details of the property values. - - - lsb-first : Configures whether the LSB or the MSB is transmitted - first for the fifo data. If this property is absent, - the MSB is transmitted first as default, or the LSB - is transmitted first. - - - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating - that SAI will work in the synchronous mode (sync Tx - with Rx) which means both the transmitter and the - receiver will send and receive data by following - receiver's bit clocks and frame sync clocks. - - - fsl,sai-asynchronous: This is a boolean property. If present, indicating - that SAI will work in the asynchronous mode, which - means both transmitter and receiver will send and - receive data by following their own bit clocks and - frame sync clocks separately. - - - fsl,dataline : configure the dataline. it has 3 value for each configuration - first one means the type: I2S(1) or PDM(2) - second one is dataline mask for 'rx' - third one is dataline mask for 'tx'. - for example: fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; - it means I2S type rx mask is 0xff, tx mask is 0xff, PDM type - rx mask is 0xff, tx mask is 0x11 (dataline 1 and 4 enabled). - -Optional properties: - - - big-endian : Boolean property, required if all the SAI - registers are big-endian rather than little-endian. - -Optional properties (for mx6ul): - - - fsl,sai-mclk-direction-output: This is a boolean property. If present, - indicates that SAI will output the SAI MCLK clock. - -Note: -- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the - default synchronous mode (sync Rx with Tx) will be used, which means both - transmitter and receiver will send and receive data by following clocks - of transmitter. -- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. - -Example: -sai2: sai@40031000 { - compatible = "fsl,vf610-sai"; - reg = <0x40031000 0x1000>; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_PLATFORM_BUS>, - <&clks VF610_CLK_SAI2>, - <&clks 0>, <&clks 0>; - clock-names = "bus", "mclk1", "mclk2", "mclk3"; - dma-names = "tx", "rx"; - dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, - <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; - big-endian; - lsb-first; -}; -- 2.34.1

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[PATCH 0/2] DT binding for sample format conversion
by Sameer Pujar 02 Aug '22

02 Aug '22

DT binding properties are available to fixup rate and channel parameters of a DAI. This series extends this to sample format conversion as well. With this now DAI PCM parameters (channels, sample rate and sample format) can be fixed up as necessary in an audio path. Sameer Pujar (2): ASoC: dt-bindings: Add sample format conversion ASoC: simple-card-utils: Fixup DAI sample format .../devicetree/bindings/sound/audio-graph-port.yaml | 21 +++++++++++++++++++++ .../devicetree/bindings/sound/audio-graph.yaml | 11 +++++++++++ .../devicetree/bindings/sound/simple-card.yaml | 15 +++++++++++++++ include/sound/simple_card_utils.h | 1 + sound/soc/generic/simple-card-utils.c | 12 ++++++++++++ 5 files changed, 60 insertions(+) -- 2.7.4

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Discuss echo reference feature
by Song, Gongjun 02 Aug '22

02 Aug '22

Hi all, Could you please give me some suggestions on how to support loop-back in embedded DSP FW?  ------------------------------------ | CPU: SoC with embedded DSP | ------------------------- | | dai | | | PCM0P --> volume --> demux --> SSP |------> | codec: speaker amplifer | | | | | | | PCM6C <----------------+ | ------------------------- | | ------------------------------------ As the diagram shows, when using the speaker to play audio data (PCM0P), in addition to transferring the audio data to the codec, I also need to transfer this audio data as a reference stream (PCM6C) to the user space for AEC (Automatic Echo Cancellation). The speaker amplifier doesn’t have the capability to send feedback data to SoC over SSP (Synchronous Serial Protocol, I2S/PCM interface) port. So, it’s the demux widget in the embedded DSP firmware (Sound Open Firmware) that sends a copy of playback data to PCM6C as the feedback for user space to implement AEC.  When using the above topology file to verify functionality (playing audio data, and starting to record data at the same time), I get the following error: [  303.956529]  Echo-Ref: ASoC: prepare FE Echo-Ref  [  303.956531]  Echo-Ref: ASoC: no backend DAIs enabled for Echo-Ref  [  303.956532]  Echo-Ref: ASoC: dpcm_fe_dai_prepare() failed (-22)  [  303.957215]  Echo-Ref: ASoC: hw_free FE Echo-Ref  From the log we can see that the above error is caused by PCM6C not having backend dais (Echo-Ref is the name of the PCM6C).  Could you give me some suggestions to implement the above case in the kernel and topology?  Thanks, Gongjun

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