diff options
130 files changed, 5633 insertions, 530 deletions
diff --git a/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml b/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml index e66ef2da7879..9af40da5688e 100644 --- a/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml +++ b/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml @@ -20,6 +20,7 @@ properties: - fsl,imx8qxp-dsp - fsl,imx8qm-dsp - fsl,imx8mp-dsp + - fsl,imx8ulp-dsp - fsl,imx8qxp-hifi4 - fsl,imx8qm-hifi4 - fsl,imx8mp-hifi4 diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index b4b35edcb493..5b8d59245f82 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -40,6 +40,7 @@ properties: patternProperties: "^dai-link-[0-9]+$": type: object + additionalProperties: false description: |- dai-link child nodes: Container for dai-link level properties and the CODEC sub-nodes. @@ -63,6 +64,7 @@ patternProperties: patternProperties: "^codec-[0-9]+$": type: object + additionalProperties: false description: |- Codecs: dai-link representing backend links should have at least one subnode. diff --git a/Documentation/devicetree/bindings/sound/apple,mca.yaml b/Documentation/devicetree/bindings/sound/apple,mca.yaml new file mode 100644 index 000000000000..d5dc92b5b654 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/apple,mca.yaml @@ -0,0 +1,131 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/apple,mca.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Apple MCA I2S transceiver + +description: | + MCA is an I2S transceiver peripheral found on M1 and other Apple chips. It is + composed of a number of identical clusters which can operate independently + or in an interlinked fashion. Up to 6 clusters have been seen on an MCA. + +maintainers: + - Martin PoviÅ¡er <povik+lin@cutebit.org> + +properties: + compatible: + items: + - enum: + - apple,t6000-mca + - apple,t8103-mca + - const: apple,mca + + reg: + items: + - description: Register region of the MCA clusters proper + - description: Register region of the DMA glue and its FIFOs + + interrupts: + minItems: 4 + maxItems: 6 + description: + One interrupt per each cluster + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 + + dmas: + minItems: 16 + maxItems: 24 + description: + DMA channels corresponding to the SERDES units in the peripheral. They are + listed in groups of four per cluster, and within the group they are given + as associated to the TXA, RXA, TXB, RXB units. + + dma-names: + minItems: 16 + items: + - const: tx0a + - const: rx0a + - const: tx0b + - const: rx0b + - const: tx1a + - const: rx1a + - const: tx1b + - const: rx1b + - const: tx2a + - const: rx2a + - const: tx2b + - const: rx2b + - const: tx3a + - const: rx3a + - const: tx3b + - const: rx3b + - const: tx4a + - const: rx4a + - const: tx4b + - const: rx4b + - const: tx5a + - const: rx5a + - const: tx5b + - const: rx5b + description: | + Names for the DMA channels: 'tx'/'rx', then cluster number, then 'a'/'b' + based on the associated SERDES unit. + + clocks: + minItems: 4 + maxItems: 6 + description: + Clusters' input reference clock. + + resets: + maxItems: 1 + + power-domains: + minItems: 5 + maxItems: 7 + description: + First a general power domain for register access, then the power + domains of individual clusters for their operation. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - power-domains + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + mca: i2s@9b600000 { + compatible = "apple,t6000-mca", "apple,mca"; + reg = <0x9b600000 0x10000>, + <0x9b200000 0x20000>; + + clocks = <&nco 0>, <&nco 1>, <&nco 2>, <&nco 3>; + power-domains = <&ps_audio_p>, <&ps_mca0>, <&ps_mca1>, + <&ps_mca2>, <&ps_mca3>; + dmas = <&admac 0>, <&admac 1>, <&admac 2>, <&admac 3>, + <&admac 4>, <&admac 5>, <&admac 6>, <&admac 7>, + <&admac 8>, <&admac 9>, <&admac 10>, <&admac 11>, + <&admac 12>, <&admac 13>, <&admac 14>, <&admac 15>; + dma-names = "tx0a", "rx0a", "tx0b", "rx0b", + "tx1a", "rx1a", "tx1b", "rx1b", + "tx2a", "rx2a", "tx2b", "rx2b", + "tx3a", "rx3a", "tx3b", "rx3b"; + + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index 5c368674d11a..bc46a95ed840 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -19,14 +19,17 @@ properties: description: "device name prefix" $ref: /schemas/types.yaml#/definitions/string convert-rate: - description: CPU to Codec rate convert. - $ref: /schemas/types.yaml#/definitions/uint32 + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-rate" convert-channels: - description: CPU to Codec rate channels. - $ref: /schemas/types.yaml#/definitions/uint32 + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-channels" + convert-sample-format: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format" + patternProperties: "^endpoint(@[0-9a-f]+)?": $ref: /schemas/graph.yaml#/$defs/endpoint-base + unevaluatedProperties: false + properties: mclk-fs: description: | @@ -65,11 +68,11 @@ patternProperties: - msb - lsb convert-rate: - description: CPU to Codec rate convert. - $ref: /schemas/types.yaml#/definitions/uint32 + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-rate" convert-channels: - description: CPU to Codec rate channels. - $ref: /schemas/types.yaml#/definitions/uint32 + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-channels" + convert-sample-format: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format" dai-tdm-slot-width-map: description: Mapping of sample widths to slot widths. For hardware diff --git a/Documentation/devicetree/bindings/sound/audio-graph.yaml b/Documentation/devicetree/bindings/sound/audio-graph.yaml index 4b46794e5153..aaa99c2deda0 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph.yaml @@ -27,11 +27,12 @@ properties: description: User specified audio sound widgets. $ref: /schemas/types.yaml#/definitions/non-unique-string-array convert-rate: - description: CPU to Codec rate convert. - $ref: /schemas/types.yaml#/definitions/uint32 + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-rate" convert-channels: - description: CPU to Codec rate channels. - $ref: /schemas/types.yaml#/definitions/uint32 + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-channels" + convert-sample-format: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format" + pa-gpios: maxItems: 1 hp-det-gpio: diff --git a/Documentation/devicetree/bindings/sound/dai-params.yaml b/Documentation/devicetree/bindings/sound/dai-params.yaml new file mode 100644 index 000000000000..f5fb71f9b603 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dai-params.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/dai-params.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Digital Audio Interface (DAI) Stream Parameters + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +select: false + +$defs: + + dai-channels: + description: Number of audio channels used by DAI + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 32 + + dai-sample-format: + description: Audio sample format used by DAI + $ref: /schemas/types.yaml#/definitions/string + enum: + - s8 + - s16_le + - s24_le + - s24_3le + - s32_le + + dai-sample-rate: + description: Audio sample rate used by DAI + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 8000 + maximum: 192000 + +properties: {} + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/everest,es8326.yaml b/Documentation/devicetree/bindings/sound/everest,es8326.yaml new file mode 100755 index 000000000000..07781408e788 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8326.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8326.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8326 audio CODEC + +maintainers: + - David Yang <yangxiaohua@everest-semi.com> + +properties: + compatible: + const: everest,es8326 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + + everest,jack-pol: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + just the value of reg 57. Bit(3) decides whether the jack polarity is inverted. + Bit(2) decides whether the button on the headset is inverted. + Bit(1)/(0) decides the mic properity to be OMTP/CTIA or auto. + minimum: 0x00 + maximum: 0x0f + default: 0x0f + + everest,mic1-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset plugged. + minimum: 0x00 + maximum: 0x77 + default: 0x22 + + everest,mic2-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset unplugged. + minimum: 0x00 + maximum: 0x77 + default: 0x44 + + everest,jack-detect-inverted: + $ref: /schemas/types.yaml#/definitions/flag + description: + Defined to invert the jack detection. + + everest,interrupt-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + value of reg 0x58, Defines the interrupt source. + Bit(2) 1 means button press triggers irq, 0 means not. + Bit(3) 1 means PIN9 is the irq source for jack detection. When set to 0, + bias change on PIN9 do not triggers irq. + Bit(4) 1 means PIN27 is the irq source for jack detection. + Bit(5) 1 means PIN9 is the irq source after MIC detect. + Bit(6) 1 means PIN27 is the irq source after MIC detect. + minimum: 0 + maximum: 0x3c + default: 0x08 + + everest,interrupt-clk: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + value of reg 0x59, Defines the interrupt output behavior. + Bit(0-3) 0 means irq pulse equals 512*internal clock + 1 means irq pulse equals 1024*internal clock + 2 means ... + 7 means irq pulse equals 65536*internal clock + 8 means irq mutes PA + 9 means irq mutes PA and DAC output + Bit(4) 1 means we invert the interrupt output. + Bit(6) 1 means the chip do not detect jack type after button released. + 0 means the chip detect jack type again after button released. + minimum: 0 + maximum: 0x7f + default: 0x45 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + es8326: codec@19 { + compatible = "everest,es8326"; + reg = <0x19>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + everest,mic1-src = [22]; + everest,mic2-src = [44]; + everest,jack-pol = [0e]; + everest,interrupt-src = [08]; + everest,interrupt-clk = [45]; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml new file mode 100644 index 000000000000..70c4111d59c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -0,0 +1,216 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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@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: + 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: + 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: + 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: + description: SAI will output the SAI MCLK clock. + type: boolean + + fsl,shared-interrupt: + 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; -}; diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml index 233caa0ade87..67ccddd44489 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -61,6 +61,8 @@ patternProperties: cpu: description: Holds subnode which indicates cpu dai. type: object + additionalProperties: false + properties: sound-dai: maxItems: 1 @@ -68,6 +70,8 @@ patternProperties: codec: description: Holds subnode which indicates codec dai. type: object + additionalProperties: false + properties: sound-dai: maxItems: 1 diff --git a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml index bb3a435722c7..b6f5d486600e 100644 --- a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml +++ b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml @@ -58,6 +58,7 @@ patternProperties: cpu: description: Holds subnode which indicates cpu dai. type: object + additionalProperties: false properties: sound-dai: maxItems: 1 @@ -65,6 +66,7 @@ patternProperties: codec: description: Holds subnode which indicates codec dai. type: object + additionalProperties: false properties: sound-dai: minItems: 1 diff --git a/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml b/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml index 513cd28b2027..d427f7f623db 100644 --- a/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml +++ b/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml @@ -43,6 +43,16 @@ properties: required: - sound-dai + mediatek,adsp: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8186 ADSP platform. + + mediatek,dai-link: + $ref: /schemas/types.yaml#/definitions/string-array + description: + A list of the desired dai-links in the sound card. Each entry is a + name defined in the machine driver. + additionalProperties: false required: diff --git a/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml b/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml index 059a7629b2d3..4fc5b045d3cf 100644 --- a/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml +++ b/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml @@ -43,6 +43,16 @@ properties: required: - sound-dai + mediatek,adsp: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8186 ADSP platform. + + mediatek,dai-link: + $ref: /schemas/types.yaml#/definitions/string-array + description: + A list of the desired dai-links in the sound card. Each entry is a + name defined in the machine driver. + additionalProperties: false required: diff --git a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml index 4fa179909c62..478be7e3fa29 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml @@ -30,6 +30,8 @@ properties: headset-codec: type: object + additionalProperties: false + properties: sound-dai: $ref: /schemas/types.yaml#/definitions/phandle @@ -38,6 +40,8 @@ properties: speaker-codecs: type: object + additionalProperties: false + properties: sound-dai: minItems: 1 diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index e6e27d09783e..a3a4289f713e 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -71,6 +71,8 @@ patternProperties: cpu: description: Holds subnode which indicates cpu dai. type: object + additionalProperties: false + properties: sound-dai: maxItems: 1 @@ -78,6 +80,8 @@ patternProperties: platform: description: Holds subnode which indicates platform dai. type: object + additionalProperties: false + properties: sound-dai: maxItems: 1 @@ -85,6 +89,8 @@ patternProperties: codec: description: Holds subnode which indicates codec dai. type: object + additionalProperties: false + properties: sound-dai: minItems: 1 diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index e17c0245f77a..268895c90bd5 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -129,6 +129,8 @@ properties: patternProperties: "^dvc-[0-1]$": type: object + additionalProperties: false + properties: dmas: maxItems: 1 @@ -145,7 +147,7 @@ properties: patternProperties: "^mix-[0-1]$": type: object - # no properties + additionalProperties: false additionalProperties: false rcar_sound,ctu: @@ -154,7 +156,7 @@ properties: patternProperties: "^ctu-[0-7]$": type: object - # no properties + additionalProperties: false additionalProperties: false rcar_sound,src: @@ -163,6 +165,8 @@ properties: patternProperties: "^src-[0-9]$": type: object + additionalProperties: false + properties: interrupts: maxItems: 1 @@ -186,6 +190,8 @@ properties: patternProperties: "^ssiu-[0-9]+$": type: object + additionalProperties: false + properties: dmas: maxItems: 2 @@ -206,6 +212,8 @@ properties: patternProperties: "^ssi-[0-9]$": type: object + additionalProperties: false + properties: interrupts: maxItems: 1 @@ -243,6 +251,8 @@ properties: patternProperties: "^dai([0-9]+)?$": type: object + additionalProperties: false + properties: playback: $ref: /schemas/types.yaml#/definitions/phandle-array diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml index a01c4ad929b8..447e013f6e17 100644 --- a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml @@ -23,6 +23,7 @@ properties: cpu: type: object + additionalProperties: false properties: sound-dai: minItems: 2 @@ -34,6 +35,7 @@ properties: - sound-dai codec: + additionalProperties: false type: object properties: sound-dai: diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml index ec50bcb4af5f..31095913e330 100644 --- a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -19,6 +19,7 @@ properties: cpu: type: object + additionalProperties: false properties: sound-dai: maxItems: 1 @@ -28,6 +29,7 @@ properties: codec: type: object + additionalProperties: false properties: sound-dai: maxItems: 1 diff --git a/Documentation/devicetree/bindings/sound/samsung,snow.yaml b/Documentation/devicetree/bindings/sound/samsung,snow.yaml index 51a83d3c7274..3d49aa4c9be2 100644 --- a/Documentation/devicetree/bindings/sound/samsung,snow.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,snow.yaml @@ -19,6 +19,7 @@ properties: codec: type: object + additionalProperties: false properties: sound-dai: description: List of phandles to the CODEC and HDMI IP nodes. @@ -30,6 +31,7 @@ properties: cpu: type: object + additionalProperties: false properties: sound-dai: description: Phandle to the Samsung I2S controller. diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml index fe2e15504ebc..1a3abc949505 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -60,6 +60,7 @@ required: patternProperties: "^audio-controller@[0-9a-f]+$": type: object + additionalProperties: false description: Two subnodes corresponding to SAI sub-block instances A et B can be defined. Subnode can be omitted for unsused sub-block. diff --git a/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml new file mode 100644 index 000000000000..9681b72b4918 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,src4xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments SRC4392 Device Tree Bindings + +description: | + The SRC4392 is a digital audio codec that can be connected via + I2C or SPI. Currently, only I2C bus is supported. + +maintainers: + - Matt Flax <flatmax@flatmax.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + compatible: + const: ti,src4392 + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@70 { + #sound-dai-cells = <0>; + compatible = "ti,src4392"; + reg = <0x70>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/spi/mediatek,spi-mtk-nor.yaml b/Documentation/devicetree/bindings/spi/mediatek,spi-mtk-nor.yaml index 970b1119898b..a453996c13f2 100644 --- a/Documentation/devicetree/bindings/spi/mediatek,spi-mtk-nor.yaml +++ b/Documentation/devicetree/bindings/spi/mediatek,spi-mtk-nor.yaml @@ -85,8 +85,9 @@ examples: compatible = "mediatek,mt8173-nor"; reg = <0 0x1100d000 0 0xe0>; interrupts = <1>; - clocks = <&pericfg CLK_PERI_SPI>, <&topckgen CLK_TOP_SPINFI_IFR_SEL>; - clock-names = "spi", "sf"; + clocks = <&pericfg CLK_PERI_SPI>, <&topckgen CLK_TOP_SPINFI_IFR_SEL>, + <&pericfg CLK_PERI_NFI>; + clock-names = "spi", "sf", "axi"; #address-cells = <1>; #size-cells = <0>; diff --git a/MAINTAINERS b/MAINTAINERS index 8a5012ba6ff9..5f91a6b62f2f 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1899,6 +1899,14 @@ F: include/dt-bindings/pinctrl/apple.h F: include/linux/apple-mailbox.h F: include/linux/soc/apple/* +ARM/APPLE MACHINE SOUND DRIVERS +M: Martin PoviÅ¡er <povik+lin@cutebit.org> +L: asahi@lists.linux.dev +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/apple,* +F: drivers/sound/apple/* + ARM/ARTPEC MACHINE SUPPORT M: Jesper Nilsson <jesper.nilsson@axis.com> M: Lars Persson <lars.persson@axis.com> diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 8d4000664fa3..d95b07896a3e 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -298,6 +298,38 @@ int sdw_transfer(struct sdw_bus *bus, struct sdw_msg *msg) } /** + * sdw_show_ping_status() - Direct report of PING status, to be used by Peripheral drivers + * @bus: SDW bus + * @sync_delay: Delay before reading status + */ +void sdw_show_ping_status(struct sdw_bus *bus, bool sync_delay) +{ + u32 status; + + if (!bus->ops->read_ping_status) + return; + + /* + * wait for peripheral to sync if desired. 10-15ms should be more than + * enough in most cases. + */ + if (sync_delay) + usleep_range(10000, 15000); + + mutex_lock(&bus->msg_lock); + + status = bus->ops->read_ping_status(bus); + + mutex_unlock(&bus->msg_lock); + + if (!status) + dev_warn(bus->dev, "%s: no peripherals attached\n", __func__); + else + dev_dbg(bus->dev, "PING status: %#x\n", status); +} +EXPORT_SYMBOL(sdw_show_ping_status); + +/** * sdw_transfer_defer() - Asynchronously transfer message to a SDW Slave device * @bus: SDW bus * @msg: SDW message to be xfered diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index 4fbb19557f5e..615b0b63a3e1 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -756,6 +756,14 @@ cdns_reset_page_addr(struct sdw_bus *bus, unsigned int dev_num) } EXPORT_SYMBOL(cdns_reset_page_addr); +u32 cdns_read_ping_status(struct sdw_bus *bus) +{ + struct sdw_cdns *cdns = bus_to_cdns(bus); + + return cdns_readl(cdns, CDNS_MCP_SLAVE_STAT); +} +EXPORT_SYMBOL(cdns_read_ping_status); + /* * IRQ handling */ diff --git a/drivers/soundwire/cadence_master.h b/drivers/soundwire/cadence_master.h index 595d72c15d97..ca9e805bab88 100644 --- a/drivers/soundwire/cadence_master.h +++ b/drivers/soundwire/cadence_master.h @@ -177,6 +177,8 @@ enum sdw_command_response cdns_xfer_msg_defer(struct sdw_bus *bus, struct sdw_msg *msg, struct sdw_defer *defer); +u32 cdns_read_ping_status(struct sdw_bus *bus); + int cdns_bus_conf(struct sdw_bus *bus, struct sdw_bus_params *params); int cdns_set_sdw_stream(struct snd_soc_dai *dai, diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 89d1d0d021fc..a5965e8827b9 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1262,6 +1262,7 @@ static struct sdw_master_ops sdw_intel_ops = { .set_bus_conf = cdns_bus_conf, .pre_bank_switch = intel_pre_bank_switch, .post_bank_switch = intel_post_bank_switch, + .read_ping_status = cdns_read_ping_status, }; static int intel_init(struct sdw_intel *sdw) diff --git a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h index 0d3276c8fc11..9f7c5103bc82 100644 --- a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h +++ b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h @@ -193,6 +193,24 @@ #define LPASS_CLK_ID_RX_CORE_MCLK 59 #define LPASS_CLK_ID_RX_CORE_NPL_MCLK 60 #define LPASS_CLK_ID_VA_CORE_2X_MCLK 61 +/* Clock ID for MCLK for WSA2 core */ +#define LPASS_CLK_ID_WSA2_CORE_MCLK 62 +/* Clock ID for NPL MCLK for WSA2 core */ +#define LPASS_CLK_ID_WSA2_CORE_2X_MCLK 63 +/* Clock ID for RX Core TX MCLK */ +#define LPASS_CLK_ID_RX_CORE_TX_MCLK 64 +/* Clock ID for RX CORE TX 2X MCLK */ +#define LPASS_CLK_ID_RX_CORE_TX_2X_MCLK 65 +/* Clock ID for WSA core TX MCLK */ +#define LPASS_CLK_ID_WSA_CORE_TX_MCLK 66 +/* Clock ID for WSA core TX 2X MCLK */ +#define LPASS_CLK_ID_WSA_CORE_TX_2X_MCLK 67 +/* Clock ID for WSA2 core TX MCLK */ +#define LPASS_CLK_ID_WSA2_CORE_TX_MCLK 68 +/* Clock ID for WSA2 core TX 2X MCLK */ +#define LPASS_CLK_ID_WSA2_CORE_TX_2X_MCLK 69 +/* Clock ID for RX CORE MCLK2 2X MCLK */ +#define LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK 70 #define LPASS_HW_AVTIMER_VOTE 101 #define LPASS_HW_MACRO_VOTE 102 diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index 39058c841469..822599957b35 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -839,6 +839,8 @@ struct sdw_defer { * @set_bus_conf: Set the bus configuration * @pre_bank_switch: Callback for pre bank switch * @post_bank_switch: Callback for post bank switch + * @read_ping_status: Read status from PING frames, reported with two bits per Device. + * Bits 31:24 are reserved. */ struct sdw_master_ops { int (*read_prop)(struct sdw_bus *bus); @@ -855,6 +857,7 @@ struct sdw_master_ops { struct sdw_bus_params *params); int (*pre_bank_switch)(struct sdw_bus *bus); int (*post_bank_switch)(struct sdw_bus *bus); + u32 (*read_ping_status)(struct sdw_bus *bus); }; @@ -919,6 +922,8 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, struct fwnode_handle *fwnode); void sdw_bus_master_delete(struct sdw_bus *bus); +void sdw_show_ping_status(struct sdw_bus *bus, bool sync_delay); + /** * sdw_port_config: Master or Slave Port configuration * diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index ab55f40896e0..a0b827f0c2f6 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -39,6 +39,7 @@ struct asoc_simple_dai { struct asoc_simple_data { u32 convert_rate; u32 convert_channels; + const char *convert_sample_format; }; struct asoc_simple_jack { diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h index bc7fd46ec2bc..82a7db23db69 100644 --- a/include/sound/soc-acpi-intel-match.h +++ b/include/sound/soc-acpi-intel-match.h @@ -14,7 +14,6 @@ * these tables are not constants, some fields can be used for * pdata or machine ops */ -extern struct snd_soc_acpi_mach snd_soc_acpi_intel_haswell_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[]; @@ -30,6 +29,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_ehl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_sdw_machines[]; @@ -38,6 +38,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[]; /* diff --git a/include/sound/soc.h b/include/sound/soc.h index aad24a1d3276..4351d86eedf6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -31,31 +31,31 @@ #define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert, xautodisable) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .rreg = xreg, .shift = shift_left, \ - .rshift = shift_right, .max = xmax, .platform_max = xmax, \ + .rshift = shift_right, .max = xmax, \ .invert = xinvert, .autodisable = xautodisable}) #define SOC_DOUBLE_S_VALUE(xreg, shift_left, shift_right, xmin, xmax, xsign_bit, xinvert, xautodisable) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .rreg = xreg, .shift = shift_left, \ - .rshift = shift_right, .min = xmin, .max = xmax, .platform_max = xmax, \ + .rshift = shift_right, .min = xmin, .max = xmax, \ .sign_bit = xsign_bit, .invert = xinvert, .autodisable = xautodisable}) #define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, xautodisable) \ SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert, xautodisable) #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \ ((unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .max = xmax, .platform_max = xmax, .invert = xinvert}) + {.reg = xreg, .max = xmax, .invert = xinvert}) #define SOC_DOUBLE_R_VALUE(xlreg, xrreg, xshift, xmax, xinvert) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ - .max = xmax, .platform_max = xmax, .invert = xinvert}) + .max = xmax, .invert = xinvert}) #define SOC_DOUBLE_R_S_VALUE(xlreg, xrreg, xshift, xmin, xmax, xsign_bit, xinvert) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ - .max = xmax, .min = xmin, .platform_max = xmax, .sign_bit = xsign_bit, \ + .max = xmax, .min = xmin, .sign_bit = xsign_bit, \ .invert = xinvert}) #define SOC_DOUBLE_R_RANGE_VALUE(xlreg, xrreg, xshift, xmin, xmax, xinvert) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ - .min = xmin, .max = xmax, .platform_max = xmax, .invert = xinvert}) + .min = xmin, .max = xmax, .invert = xinvert}) #define SOC_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ @@ -68,7 +68,7 @@ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .rreg = xreg, .shift = xshift, \ .rshift = xshift, .min = xmin, .max = xmax, \ - .platform_max = xmax, .invert = xinvert} } + .invert = xinvert} } #define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -99,7 +99,7 @@ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .rreg = xreg, .shift = xshift, \ .rshift = xshift, .min = xmin, .max = xmax, \ - .platform_max = xmax, .invert = xinvert} } + .invert = xinvert} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ @@ -199,7 +199,7 @@ .put = snd_soc_put_volsw, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .rreg = xreg, \ - .min = xmin, .max = xmax, .platform_max = xmax, \ + .min = xmin, .max = xmax, \ .sign_bit = 7,} } #define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -273,7 +273,7 @@ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .rreg = xreg, .shift = xshift, \ .rshift = xshift, .min = xmin, .max = xmax, \ - .platform_max = xmax, .invert = xinvert} } + .invert = xinvert} } #define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ diff --git a/include/sound/sof.h b/include/sound/sof.h index 367dccfea7ad..341fef19e612 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -89,6 +89,7 @@ struct snd_sof_pdata { /* machine */ struct platform_device *pdev_mach; const struct snd_soc_acpi_mach *machine; + const struct snd_sof_of_mach *of_machine; void *hw_pdata; @@ -102,6 +103,7 @@ struct snd_sof_pdata { struct sof_dev_desc { /* list of machines using this configuration */ struct snd_soc_acpi_mach *machines; + struct snd_sof_of_mach *of_machines; /* alternate list of machines using this configuration */ struct snd_soc_acpi_mach *alt_machines; diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 7d4747b6bab2..848fbae26c3b 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -68,6 +68,7 @@ config SND_SOC_ACPI # All the supported SoCs source "sound/soc/adi/Kconfig" source "sound/soc/amd/Kconfig" +source "sound/soc/apple/Kconfig" source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/bcm/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 453181ef6c94..507eaed1d6a1 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -34,6 +34,7 @@ obj-$(CONFIG_SND_SOC_ACPI) += snd-soc-acpi.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ +obj-$(CONFIG_SND_SOC) += apple/ obj-$(CONFIG_SND_SOC) += adi/ obj-$(CONFIG_SND_SOC) += amd/ obj-$(CONFIG_SND_SOC) += atmel/ diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index 393f729ef561..ac416572db0d 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -25,6 +25,65 @@ #define DRV_NAME "acp_i2s_playcap" +static int acp_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct acp_dev_data *adata = snd_soc_dai_get_drvdata(cpu_dai); + int mode; + + mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + switch (mode) { + case SND_SOC_DAIFMT_I2S: + adata->tdm_mode = TDM_DISABLE; + break; + case SND_SOC_DAIFMT_DSP_A: + adata->tdm_mode = TDM_ENABLE; + break; + default: + return -EINVAL; + } + return 0; +} + +static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mask, + int slots, int slot_width) +{ + struct device *dev = dai->component->dev; + struct acp_dev_data *adata = snd_soc_dai_get_drvdata(dai); + struct acp_stream *stream; + int slot_len; + + switch (slot_width) { + case SLOT_WIDTH_8: + slot_len = 8; + break; + case SLOT_WIDTH_16: + slot_len = 16; + break; + case SLOT_WIDTH_24: + slot_len = 24; + break; + case SLOT_WIDTH_32: + slot_len = 0; + break; + default: + dev_err(dev, "Unsupported bitdepth %d\n", slot_width); + return -EINVAL; + } + + spin_lock_irq(&adata->acp_lock); + list_for_each_entry(stream, &adata->stream_list, list) { + if (tx_mask && stream->dir == SNDRV_PCM_STREAM_PLAYBACK) + adata->tdm_tx_fmt[stream->dai_id - 1] = + FRM_LEN | (slots << 15) | (slot_len << 18); + else if (rx_mask && stream->dir == SNDRV_PCM_STREAM_CAPTURE) + adata->tdm_rx_fmt[stream->dai_id - 1] = + FRM_LEN | (slots << 15) | (slot_len << 18); + } + spin_unlock_irq(&adata->acp_lock); + return 0; +} + static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -33,7 +92,7 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ struct acp_resource *rsrc; u32 val; u32 xfer_resolution; - u32 reg_val; + u32 reg_val, fmt_reg, tdm_fmt; u32 lrclk_div_val, bclk_div_val; adata = snd_soc_dai_get_drvdata(dai); @@ -62,12 +121,15 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ switch (dai->driver->id) { case I2S_BT_INSTANCE: reg_val = ACP_BTTDM_ITER; + fmt_reg = ACP_BTTDM_TXFRMT; break; case I2S_SP_INSTANCE: reg_val = ACP_I2STDM_ITER; + fmt_reg = ACP_I2STDM_TXFRMT; break; case I2S_HS_INSTANCE: reg_val = ACP_HSTDM_ITER; + fmt_reg = ACP_HSTDM_TXFRMT; break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); @@ -77,12 +139,15 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ switch (dai->driver->id) { case I2S_BT_INSTANCE: reg_val = ACP_BTTDM_IRER; + fmt_reg = ACP_BTTDM_RXFRMT; break; case I2S_SP_INSTANCE: reg_val = ACP_I2STDM_IRER; + fmt_reg = ACP_I2STDM_RXFRMT; break; case I2S_HS_INSTANCE: reg_val = ACP_HSTDM_IRER; + fmt_reg = ACP_HSTDM_RXFRMT; break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); @@ -95,6 +160,16 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ val = val | (xfer_resolution << 3); writel(val, adata->acp_base + reg_val); + if (adata->tdm_mode) { + val = readl(adata->acp_base + reg_val); + writel(val | BIT(1), adata->acp_base + reg_val); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tdm_fmt = adata->tdm_tx_fmt[dai->driver->id - 1]; + else + tdm_fmt = adata->tdm_rx_fmt[dai->driver->id - 1]; + writel(tdm_fmt, adata->acp_base + fmt_reg); + } + if (rsrc->soc_mclk) { switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -443,6 +518,7 @@ static int acp_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_d stream->id = dai->driver->id + dir; stream->dai_id = dai->driver->id; stream->irq_bit = irq_bit; + stream->dir = substream->stream; return 0; } @@ -452,6 +528,8 @@ const struct snd_soc_dai_ops asoc_acp_cpu_dai_ops = { .hw_params = acp_i2s_hwparams, .prepare = acp_i2s_prepare, .trigger = acp_i2s_trigger, + .set_fmt = acp_i2s_set_fmt, + .set_tdm_slot = acp_i2s_set_tdm_slot, }; EXPORT_SYMBOL_NS_GPL(asoc_acp_cpu_dai_ops, SND_SOC_ACP_COMMON); diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 2c8e960cc9a6..ef2ce083521e 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -62,10 +62,9 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id if (!chip) return -ENOMEM; - if (pci_enable_device(pci)) { - dev_err(&pci->dev, "pci_enable_device failed\n"); - return -ENODEV; - } + if (pci_enable_device(pci)) + return dev_err_probe(&pci->dev, -ENODEV, + "pci_enable_device failed\n"); ret = pci_request_regions(pci, "AMD ACP3x audio"); if (ret < 0) { @@ -105,14 +104,13 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id chip->base = devm_ioremap(&pci->dev, addr, pci_resource_len(pci, 0)); if (!chip->base) { ret = -ENOMEM; - goto release_regions; + goto unregister_dmic_dev; } res = devm_kzalloc(&pci->dev, sizeof(struct resource) * num_res, GFP_KERNEL); if (!res) { - platform_device_unregister(dmic_dev); ret = -ENOMEM; - goto release_regions; + goto unregister_dmic_dev; } for (i = 0; i < num_res; i++, res_acp++) { @@ -139,13 +137,14 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id pdev = platform_device_register_full(&pdevinfo); if (IS_ERR(pdev)) { dev_err(&pci->dev, "cannot register %s device\n", pdevinfo.name); - platform_device_unregister(dmic_dev); ret = PTR_ERR(pdev); - goto release_regions; + goto unregister_dmic_dev; } return ret; +unregister_dmic_dev: + platform_device_unregister(dmic_dev); release_regions: pci_release_regions(pci); disable_pci: diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index f561d39b33e2..85a81add4ef9 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -94,7 +94,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) struct acp_resource *rsrc = adata->rsrc; struct acp_stream *stream; u16 i2s_flag = 0; - u32 ext_intr_stat, ext_intr_stat1, i; + u32 ext_intr_stat, ext_intr_stat1; if (!adata) return IRQ_NONE; @@ -104,25 +104,24 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); - for (i = 0; i < ACP_MAX_STREAM; i++) { - stream = adata->stream[i]; - if (stream && (ext_intr_stat & stream->irq_bit)) { + spin_lock(&adata->acp_lock); + list_for_each_entry(stream, &adata->stream_list, list) { + if (ext_intr_stat & stream->irq_bit) { writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); snd_pcm_period_elapsed(stream->substream); i2s_flag = 1; - break; } if (adata->rsrc->no_of_ctrls == 2) { - if (stream && (ext_intr_stat1 & stream->irq_bit)) { + if (ext_intr_stat1 & stream->irq_bit) { writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); snd_pcm_period_elapsed(stream->substream); i2s_flag = 1; - break; } } } + spin_unlock(&adata->acp_lock); if (i2s_flag) return IRQ_HANDLED; @@ -146,9 +145,8 @@ static void config_pte_for_stream(struct acp_dev_data *adata, struct acp_stream writel(0x01, adata->acp_base + ACPAXI2AXI_ATU_CTRL); } -static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size) +static void config_acp_dma(struct acp_dev_data *adata, struct acp_stream *stream, int size) { - struct acp_stream *stream = adata->stream[cpu_id]; struct snd_pcm_substream *substream = stream->substream; struct acp_resource *rsrc = adata->rsrc; dma_addr_t addr = substream->dma_buffer.addr; @@ -174,13 +172,10 @@ static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size) static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct snd_pcm_runtime *runtime = substream->runtime; struct device *dev = component->dev; struct acp_dev_data *adata = dev_get_drvdata(dev); struct acp_stream *stream; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; int ret; stream = kzalloc(sizeof(*stream), GFP_KERNEL); @@ -188,7 +183,10 @@ static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_subs return -ENOMEM; stream->substream = substream; - adata->stream[stream_id] = stream; + + spin_lock_irq(&adata->acp_lock); + list_add_tail(&stream->list, &adata->stream_list); + spin_unlock_irq(&adata->acp_lock); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) runtime->hw = acp_pcm_hardware_playback; @@ -212,16 +210,13 @@ static int acp_dma_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); struct acp_dev_data *adata = snd_soc_component_get_drvdata(component); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct acp_stream *stream = substream->runtime->private_data; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; u64 size = params_buffer_bytes(params); /* Configure ACP DMA block with params */ config_pte_for_stream(adata, stream); - config_acp_dma(adata, stream_id, size); + config_acp_dma(adata, stream, size); return 0; } @@ -261,16 +256,15 @@ static int acp_dma_new(struct snd_soc_component *component, static int acp_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct device *dev = component->dev; struct acp_dev_data *adata = dev_get_drvdata(dev); - struct acp_stream *stream; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; + struct acp_stream *stream = substream->runtime->private_data; - stream = adata->stream[stream_id]; + /* Remove entry from list */ + spin_lock_irq(&adata->acp_lock); + list_del(&stream->list); + spin_unlock_irq(&adata->acp_lock); kfree(stream); - adata->stream[stream_id] = NULL; return 0; } @@ -305,6 +299,10 @@ int acp_platform_register(struct device *dev) dev_err(dev, "Fail to register acp i2s component\n"); return status; } + + INIT_LIST_HEAD(&adata->stream_list); + spin_lock_init(&adata->acp_lock); + return 0; } EXPORT_SYMBOL_NS_GPL(acp_platform_register, SND_SOC_ACP_COMMON); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index af9603724a68..5f2119f42271 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -21,9 +21,9 @@ #define ACP3X_DEV 3 #define ACP6X_DEV 6 -#define I2S_SP_INSTANCE 0x00 -#define I2S_BT_INSTANCE 0x01 -#define DMIC_INSTANCE 0x02 +#define DMIC_INSTANCE 0x00 +#define I2S_SP_INSTANCE 0x01 +#define I2S_BT_INSTANCE 0x02 #define I2S_HS_INSTANCE 0x03 #define MEM_WINDOW_START 0x4080000 @@ -84,6 +84,14 @@ #define ACP_MAX_STREAM 8 +#define TDM_ENABLE 1 +#define TDM_DISABLE 0 + +#define SLOT_WIDTH_8 0x8 +#define SLOT_WIDTH_16 0x10 +#define SLOT_WIDTH_24 0x18 +#define SLOT_WIDTH_32 0x20 + struct acp_chip_info { char *name; /* Platform name */ unsigned int acp_rev; /* ACP Revision id */ @@ -91,10 +99,12 @@ struct acp_chip_info { }; struct acp_stream { + struct list_head list; struct snd_pcm_substream *substream; int irq_bit; int dai_id; int id; + int dir; u64 bytescount; u32 reg_offset; u32 pte_offset; @@ -119,11 +129,13 @@ struct acp_dev_data { void __iomem *acp_base; unsigned int i2s_irq; + bool tdm_mode; /* SOC specific dais */ struct snd_soc_dai_driver *dai_driver; int num_dai; - struct acp_stream *stream[ACP_MAX_STREAM]; + struct list_head stream_list; + spinlock_t acp_lock; struct snd_soc_acpi_mach *machines; struct platform_device *mach_dev; @@ -132,6 +144,8 @@ struct acp_dev_data { u32 lrclk_div; struct acp_resource *rsrc; + u32 tdm_tx_fmt[3]; + u32 tdm_rx_fmt[3]; }; union acp_i2stdm_mstrclkgen { 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"), + } + }, {} }; diff --git a/sound/soc/apple/Kconfig b/sound/soc/apple/Kconfig new file mode 100644 index 000000000000..0ba955657e98 --- /dev/null +++ b/sound/soc/apple/Kconfig @@ -0,0 +1,9 @@ +config SND_SOC_APPLE_MCA + tristate "Apple Silicon MCA driver" + depends on ARCH_APPLE || COMPILE_TEST + select SND_DMAENGINE_PCM + select COMMON_CLK + default ARCH_APPLE + help + This option enables an ASoC platform driver for MCA peripherals found + on Apple Silicon SoCs. diff --git a/sound/soc/apple/Makefile b/sound/soc/apple/Makefile new file mode 100644 index 000000000000..7a30bf452817 --- /dev/null +++ b/sound/soc/apple/Makefile @@ -0,0 +1,3 @@ +snd-soc-apple-mca-objs := mca.o + +obj-$(CONFIG_SND_SOC_APPLE_MCA) += snd-soc-apple-mca.o diff --git a/sound/soc/apple/mca.c b/sound/soc/apple/mca.c new file mode 100644 index 000000000000..aa67d57c9a9b --- /dev/null +++ b/sound/soc/apple/mca.c @@ -0,0 +1,1167 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Apple SoCs MCA driver +// +// Copyright (C) The Asahi Linux Contributors +// +// The MCA peripheral is made up of a number of identical units called clusters. +// Each cluster has its separate clock parent, SYNC signal generator, carries +// four SERDES units and has a dedicated I2S port on the SoC's periphery. +// +// The clusters can operate independently, or can be combined together in a +// configurable manner. We mostly treat them as self-contained independent +// units and don't configure any cross-cluster connections except for the I2S +// ports. The I2S ports can be routed to any of the clusters (irrespective +// of their native cluster). We map this onto ASoC's (DPCM) notion of backend +// and frontend DAIs. The 'cluster guts' are frontends which are dynamically +// routed to backend I2S ports. +// +// DAI references in devicetree are resolved to backends. The routing between +// frontends and backends is determined by the machine driver in the DAPM paths +// it supplies. + +#include <linux/bitfield.h> +#include <linux/clk.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_clk.h> +#include <linux/of_dma.h> +#include <linux/platform_device.h> +#include <linux/pm_domain.h> +#include <linux/regmap.h> +#include <linux/reset.h> +#include <linux/slab.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#define USE_RXB_FOR_CAPTURE + +/* Relative to cluster base */ +#define REG_STATUS 0x0 +#define STATUS_MCLK_EN BIT(0) +#define REG_MCLK_CONF 0x4 +#define MCLK_CONF_DIV GENMASK(11, 8) + +#define REG_SYNCGEN_STATUS 0x100 +#define SYNCGEN_STATUS_EN BIT(0) +#define REG_SYNCGEN_MCLK_SEL 0x104 +#define SYNCGEN_MCLK_SEL GENMASK(3, 0) +#define REG_SYNCGEN_HI_PERIOD 0x108 +#define REG_SYNCGEN_LO_PERIOD 0x10c + +#define REG_PORT_ENABLES 0x600 +#define PORT_ENABLES_CLOCKS GENMASK(2, 1) +#define PORT_ENABLES_TX_DATA BIT(3) +#define REG_PORT_CLOCK_SEL 0x604 +#define PORT_CLOCK_SEL GENMASK(11, 8) +#define REG_PORT_DATA_SEL 0x608 +#define PORT_DATA_SEL_TXA(cl) (1 << ((cl)*2)) +#define PORT_DATA_SEL_TXB(cl) (2 << ((cl)*2)) + +#define REG_INTSTATE 0x700 +#define REG_INTMASK 0x704 + +/* Bases of serdes units (relative to cluster) */ +#define CLUSTER_RXA_OFF 0x200 +#define CLUSTER_TXA_OFF 0x300 +#define CLUSTER_RXB_OFF 0x400 +#define CLUSTER_TXB_OFF 0x500 + +#define CLUSTER_TX_OFF CLUSTER_TXA_OFF + +#ifndef USE_RXB_FOR_CAPTURE +#define CLUSTER_RX_OFF CLUSTER_RXA_OFF +#else +#define CLUSTER_RX_OFF CLUSTER_RXB_OFF +#endif + +/* Relative to serdes unit base */ +#define REG_SERDES_STATUS 0x00 +#define SERDES_STATUS_EN BIT(0) +#define SERDES_STATUS_RST BIT(1) +#define REG_TX_SERDES_CONF 0x04 +#define REG_RX_SERDES_CONF 0x08 +#define SERDES_CONF_NCHANS GENMASK(3, 0) +#define SERDES_CONF_WIDTH_MASK GENMASK(8, 4) +#define SERDES_CONF_WIDTH_16BIT 0x40 +#define SERDES_CONF_WIDTH_20BIT 0x80 +#define SERDES_CONF_WIDTH_24BIT 0xc0 +#define SERDES_CONF_WIDTH_32BIT 0x100 +#define SERDES_CONF_BCLK_POL 0x400 +#define SERDES_CONF_LSB_FIRST 0x800 +#define SERDES_CONF_UNK1 BIT(12) +#define SERDES_CONF_UNK2 BIT(13) +#define SERDES_CONF_UNK3 BIT(14) +#define SERDES_CONF_NO_DATA_FEEDBACK BIT(15) +#define SERDES_CONF_SYNC_SEL GENMASK(18, 16) +#define SERDES_CONF_SOME_RST BIT(19) +#define REG_TX_SERDES_BITSTART 0x08 +#define REG_RX_SERDES_BITSTART 0x0c +#define REG_TX_SERDES_SLOTMASK 0x0c +#define REG_RX_SERDES_SLOTMASK 0x10 +#define REG_RX_SERDES_PORT 0x04 + +/* Relative to switch base */ +#define REG_DMA_ADAPTER_A(cl) (0x8000 * (cl)) +#define REG_DMA_ADAPTER_B(cl) (0x8000 * (cl) + 0x4000) +#define DMA_ADAPTER_TX_LSB_PAD GENMASK(4, 0) +#define DMA_ADAPTER_TX_NCHANS GENMASK(6, 5) +#define DMA_ADAPTER_RX_MSB_PAD GENMASK(12, 8) +#define DMA_ADAPTER_RX_NCHANS GENMASK(14, 13) +#define DMA_ADAPTER_NCHANS GENMASK(22, 20) + +#define SWITCH_STRIDE 0x8000 +#define CLUSTER_STRIDE 0x4000 + +#define MAX_NCLUSTERS 6 + +#define APPLE_MCA_FMTBITS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +struct mca_cluster { + int no; + __iomem void *base; + struct mca_data *host; + struct device *pd_dev; + struct clk *clk_parent; + struct dma_chan *dma_chans[SNDRV_PCM_STREAM_LAST + 1]; + + bool port_started[SNDRV_PCM_STREAM_LAST + 1]; + int port_driver; /* The cluster driving this cluster's port */ + + bool clocks_in_use[SNDRV_PCM_STREAM_LAST + 1]; + struct device_link *pd_link; + + unsigned int bclk_ratio; + + /* Masks etc. picked up via the set_tdm_slot method */ + int tdm_slots; + int tdm_slot_width; + unsigned int tdm_tx_mask; + unsigned int tdm_rx_mask; +}; + +struct mca_data { + struct device *dev; + + __iomem void *switch_base; + + struct device *pd_dev; + struct reset_control *rstc; + struct device_link *pd_link; + + /* Mutex for accessing port_driver of foreign clusters */ + struct mutex port_mutex; + + int nclusters; + struct mca_cluster clusters[]; +}; + +static void mca_modify(struct mca_cluster *cl, int regoffset, u32 mask, u32 val) +{ + __iomem void *ptr = cl->base + regoffset; + u32 newval; + + newval = (val & mask) | (readl_relaxed(ptr) & ~mask); + writel_relaxed(newval, ptr); +} + +/* + * Get the cluster of FE or BE DAI + */ +static struct mca_cluster *mca_dai_to_cluster(struct snd_soc_dai *dai) +{ + struct mca_data *mca = snd_soc_dai_get_drvdata(dai); + /* + * FE DAIs are 0 ... nclusters - 1 + * BE DAIs are nclusters ... 2*nclusters - 1 + */ + int cluster_no = dai->id % mca->nclusters; + + return &mca->clusters[cluster_no]; +} + +/* called before PCM trigger */ +static void mca_fe_early_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct mca_cluster *cl = mca_dai_to_cluster(dai); + bool is_tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int serdes_unit = is_tx ? CLUSTER_TX_OFF : CLUSTER_RX_OFF; + int serdes_conf = + serdes_unit + (is_tx ? REG_TX_SERDES_CONF : REG_RX_SERDES_CONF); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + mca_modify(cl, serdes_unit + REG_SERDES_STATUS, + SERDES_STATUS_EN | SERDES_STATUS_RST, + SERDES_STATUS_RST); + mca_modify(cl, serdes_conf, SERDES_CONF_SOME_RST, + SERDES_CONF_SOME_RST); + readl_relaxed(cl->base + serdes_conf); + mca_modify(cl, serdes_conf, SERDES_STATUS_RST, 0); + WARN_ON(readl_relaxed(cl->base + REG_SERDES_STATUS) & + SERDES_STATUS_RST); + break; + default: + break; + } +} + +static int mca_fe_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct mca_cluster *cl = mca_dai_to_cluster(dai); + bool is_tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int serdes_unit = is_tx ? CLUSTER_TX_OFF : CLUSTER_RX_OFF; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + mca_modify(cl, serdes_unit + REG_SERDES_STATUS, + SERDES_STATUS_EN | SERDES_STATUS_RST, + SERDES_STATUS_EN); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + mca_modify(cl, serdes_unit + REG_SERDES_STATUS, + SERDES_STATUS_EN, 0); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int mca_fe_enable_clocks(struct mca_cluster *cl) +{ + struct mca_data *mca = cl->host; + int ret; + + ret = clk_prepare_enable(cl->clk_parent); + if (ret) { + dev_err(mca->dev, + "cluster %d: unable to enable clock parent: %d\n", + cl->no, ret); + return ret; + } + + /* + * We can't power up the device earlier than this because + * the power state driver would error out on seeing the device + * as clock-gated. + */ + cl->pd_link = device_link_add(mca->dev, cl->pd_dev, + DL_FLAG_STATELESS | DL_FLAG_PM_RUNTIME | + DL_FLAG_RPM_ACTIVE); + if (!cl->pd_link) { + dev_err(mca->dev, + "cluster %d: unable to prop-up power domain\n", cl->no); + clk_disable_unprepare(cl->clk_parent); + return -EINVAL; + } + + writel_relaxed(cl->no + 1, cl->base + REG_SYNCGEN_MCLK_SEL); + mca_modify(cl, REG_SYNCGEN_STATUS, SYNCGEN_STATUS_EN, + SYNCGEN_STATUS_EN); + mca_modify(cl, REG_STATUS, STATUS_MCLK_EN, STATUS_MCLK_EN); + + return 0; +} + +static void mca_fe_disable_clocks(struct mca_cluster *cl) +{ + mca_modify(cl, REG_SYNCGEN_STATUS, SYNCGEN_STATUS_EN, 0); + mca_modify(cl, REG_STATUS, STATUS_MCLK_EN, 0); + + device_link_del(cl->pd_link); + clk_disable_unprepare(cl->clk_parent); +} + +static bool mca_fe_clocks_in_use(struct mca_cluster *cl) +{ + struct mca_data *mca = cl->host; + struct mca_cluster *be_cl; + int stream, i; + + mutex_lock(&mca->port_mutex); + for (i = 0; i < mca->nclusters; i++) { + be_cl = &mca->clusters[i]; + + if (be_cl->port_driver != cl->no) + continue; + + for_each_pcm_streams(stream) { + if (be_cl->clocks_in_use[stream]) { + mutex_unlock(&mca->port_mutex); + return true; + } + } + } + mutex_unlock(&mca->port_mutex); + return false; +} + +static int mca_be_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mca_cluster *cl = mca_dai_to_cluster(dai); + struct mca_data *mca = cl->host; + struct mca_cluster *fe_cl; + int ret; + + if (cl->port_driver < 0) + return -EINVAL; + + fe_cl = &mca->clusters[cl->port_driver]; + + /* + * Typically the CODECs we are paired with will require clocks + * to be present at time of unmute with the 'mute_stream' op + * or at time of DAPM widget power-up. We need to enable clocks + * here at the latest (frontend prepare would be too late). + */ + if (!mca_fe_clocks_in_use(fe_cl)) { + ret = mca_fe_enable_clocks(fe_cl); + if (ret < 0) + return ret; + } + + cl->clocks_in_use[substream->stream] = true; + + return 0; +} + +static int mca_be_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mca_cluster *cl = mca_dai_to_cluster(dai); + struct mca_data *mca = cl->host; + struct mca_cluster *fe_cl; + + if (cl->port_driver < 0) + return -EINVAL; + + /* + * We are operating on a foreign cluster here, but since we + * belong to the same PCM, accesses should have been + * synchronized at ASoC level. + */ + fe_cl = &mca->clusters[cl->port_driver]; + if (!mca_fe_clocks_in_use(fe_cl)) + return 0; /* Nothing to do */ + + cl->clocks_in_use[substream->stream] = false; + + if (!mca_fe_clocks_in_use(fe_cl)) + mca_fe_disable_clocks(fe_cl); + + return 0; +} + +static unsigned int mca_crop_mask(unsigned int mask, int nchans) +{ + while (hweight32(mask) > nchans) + mask &= ~(1 << __fls(mask)); + + return mask; +} + +static int mca_configure_serdes(struct mca_cluster *cl, int serdes_unit, + unsigned int mask, int slots, int nchans, + int slot_width, bool is_tx, int port) +{ + __iomem void *serdes_base = cl->base + serdes_unit; + u32 serdes_conf, serdes_conf_mask; + + serdes_conf_mask = SERDES_CONF_WIDTH_MASK | SERDES_CONF_NCHANS; + serdes_conf = FIELD_PREP(SERDES_CONF_NCHANS, max(slots, 1) - 1); + switch (slot_width) { + case 16: + serdes_conf |= SERDES_CONF_WIDTH_16BIT; + break; + case 20: + serdes_conf |= SERDES_CONF_WIDTH_20BIT; + break; + case 24: + serdes_conf |= SERDES_CONF_WIDTH_24BIT; + break; + case 32: + serdes_conf |= SERDES_CONF_WIDTH_32BIT; + break; + default: + goto err; + } + + serdes_conf_mask |= SERDES_CONF_SYNC_SEL; + serdes_conf |= FIELD_PREP(SERDES_CONF_SYNC_SEL, cl->no + 1); + + if (is_tx) { + serdes_conf_mask |= SERDES_CONF_UNK1 | SERDES_CONF_UNK2 | + SERDES_CONF_UNK3; + serdes_conf |= SERDES_CONF_UNK1 | SERDES_CONF_UNK2 | + SERDES_CONF_UNK3; + } else { + serdes_conf_mask |= SERDES_CONF_UNK1 | SERDES_CONF_UNK2 | + SERDES_CONF_UNK3 | + SERDES_CONF_NO_DATA_FEEDBACK; + serdes_conf |= SERDES_CONF_UNK1 | SERDES_CONF_UNK2 | + SERDES_CONF_NO_DATA_FEEDBACK; + } + + mca_modify(cl, + serdes_unit + + (is_tx ? REG_TX_SERDES_CONF : REG_RX_SERDES_CONF), + serdes_conf_mask, serdes_conf); + + if (is_tx) { + writel_relaxed(0xffffffff, + serdes_base + REG_TX_SERDES_SLOTMASK); + writel_relaxed(~((u32)mca_crop_mask(mask, nchans)), + serdes_base + REG_TX_SERDES_SLOTMASK + 0x4); + writel_relaxed(0xffffffff, + serdes_base + REG_TX_SERDES_SLOTMASK + 0x8); + writel_relaxed(~((u32)mask), + serdes_base + REG_TX_SERDES_SLOTMASK + 0xc); + } else { + writel_relaxed(0xffffffff, + serdes_base + REG_RX_SERDES_SLOTMASK); + writel_relaxed(~((u32)mca_crop_mask(mask, nchans)), + serdes_base + REG_RX_SERDES_SLOTMASK + 0x4); + writel_relaxed(1 << port, + serdes_base + REG_RX_SERDES_PORT); + } + + return 0; + +err: + dev_err(cl->host->dev, + "unsupported SERDES configuration requested (mask=0x%x slots=%d slot_width=%d)\n", + mask, slots, slot_width); + return -EINVAL; +} + +static int mca_fe_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct mca_cluster *cl = mca_dai_to_cluster(dai); + + cl->tdm_slots = slots; + cl->tdm_slot_width = slot_width; + cl->tdm_tx_mask = tx_mask; + cl->tdm_rx_mask = rx_mask; + + return 0; +} + +static int mca_fe_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct mca_cluster *cl = mca_dai_to_cluster(dai); + struct mca_data *mca = cl->host; + bool fpol_inv = false; + u32 serdes_conf = 0; + u32 bitstart; + + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != + SND_SOC_DAIFMT_BP_FP) + goto err; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + fpol_inv = 0; + bitstart = 1; + break; + case SND_SOC_DAIFMT_LEFT_J: + fpol_inv = 1; + bitstart = 0; + break; + default: + goto err; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + case SND_SOC_DAIFMT_IB_IF: + fpol_inv ^= 1; + break; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + case SND_SOC_DAIFMT_NB_IF: + serdes_conf |= SERDES_CONF_BCLK_POL; + break; + } + + if (!fpol_inv) + goto err; + + mca_modify(cl, CLUSTER_TX_OFF + REG_TX_SERDES_CONF, + SERDES_CONF_BCLK_POL, serdes_conf); + mca_modify(cl, CLUSTER_RX_OFF + REG_RX_SERDES_CONF, + SERDES_CONF_BCLK_POL, serdes_conf); + writel_relaxed(bitstart, + cl->base + CLUSTER_TX_OFF + REG_TX_SERDES_BITSTART); + writel_relaxed(bitstart, + cl->base + CLUSTER_RX_OFF + REG_RX_SERDES_BITSTART); + + return 0; + +err: + dev_err(mca->dev, "unsupported DAI format (0x%x) requested\n", fmt); + return -EINVAL; +} + +static int mca_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct mca_cluster *cl = mca_dai_to_cluster(dai); + + cl->bclk_ratio = ratio; + + return 0; +} + +static int mca_fe_get_port(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(substream); + struct snd_soc_pcm_runtime *be; + struct snd_soc_dpcm *dpcm; + + be = NULL; + for_each_dpcm_be(fe, substream->stream, dpcm) { + be = dpcm->be; + break; + } + + if (!be) + return -EINVAL; + + return mca_dai_to_cluster(asoc_rtd_to_cpu(be, 0))->no; +} + +static int mca_fe_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct mca_cluster *cl = mca_dai_to_cluster(dai); + struct mca_data *mca = cl->host; + struct device *dev = mca->dev; + unsigned int samp_rate = params_rate(params); + bool is_tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool refine_tdm = false; + unsigned long bclk_ratio; + unsigned int tdm_slots, tdm_slot_width, tdm_mask; + u32 regval, pad; + int ret, port, nchans_ceiled; + + if (!cl->tdm_slot_width) { + /* + * We were not given TDM settings from above, set initial + * guesses which will later be refined. + */ + tdm_slot_width = params_width(params); + tdm_slots = params_channels(params); + refine_tdm = true; + } else { + tdm_slot_width = cl->tdm_slot_width; + tdm_slots = cl->tdm_slots; + tdm_mask = is_tx ? cl->tdm_tx_mask : cl->tdm_rx_mask; + } + + if (cl->bclk_ratio) + bclk_ratio = cl->bclk_ratio; + else + bclk_ratio = tdm_slot_width * tdm_slots; + + if (refine_tdm) { + int nchannels = params_channels(params); + + if (nchannels > 2) { + dev_err(dev, "missing TDM for stream with two or more channels\n"); + return -EINVAL; + } + + if ((bclk_ratio % nchannels) != 0) { + dev_err(dev, "BCLK ratio (%ld) not divisible by no. of channels (%d)\n", + bclk_ratio, nchannels); + return -EINVAL; + } + + tdm_slot_width = bclk_ratio / nchannels; + + if (tdm_slot_width > 32 && nchannels == 1) + tdm_slot_width = 32; + + if (tdm_slot_width < params_width(params)) { + dev_err(dev, "TDM slots too narrow (tdm=%d params=%d)\n", + tdm_slot_width, params_width(params)); + return -EINVAL; + } + + tdm_mask = (1 << tdm_slots) - 1; + } + + port = mca_fe_get_port(substream); + if (port < 0) + return port; + + ret = mca_configure_serdes(cl, is_tx ? CLUSTER_TX_OFF : CLUSTER_RX_OFF, + tdm_mask, tdm_slots, params_channels(params), + tdm_slot_width, is_tx, port); + if (ret) + return ret; + + pad = 32 - params_width(params); + + /* + * TODO: Here the register semantics aren't clear. + */ + nchans_ceiled = min_t(int, params_channels(params), 4); + regval = FIELD_PREP(DMA_ADAPTER_NCHANS, nchans_ceiled) | + FIELD_PREP(DMA_ADAPTER_TX_NCHANS, 0x2) | + FIELD_PREP(DMA_ADAPTER_RX_NCHANS, 0x2) | + FIELD_PREP(DMA_ADAPTER_TX_LSB_PAD, pad) | + FIELD_PREP(DMA_ADAPTER_RX_MSB_PAD, pad); + +#ifndef USE_RXB_FOR_CAPTURE + writel_relaxed(regval, mca->switch_base + REG_DMA_ADAPTER_A(cl->no)); +#else + if (is_tx) + writel_relaxed(regval, + mca->switch_base + REG_DMA_ADAPTER_A(cl->no)); + else + writel_relaxed(regval, + mca->switch_base + REG_DMA_ADAPTER_B(cl->no)); +#endif + + if (!mca_fe_clocks_in_use(cl)) { + /* + * Set up FSYNC duty cycle as even as possible. + */ + writel_relaxed((bclk_ratio / 2) - 1, + cl->base + REG_SYNCGEN_HI_PERIOD); + writel_relaxed(((bclk_ratio + 1) / 2) - 1, + cl->base + REG_SYNCGEN_LO_PERIOD); + writel_relaxed(FIELD_PREP(MCLK_CONF_DIV, 0x1), + cl->base + REG_MCLK_CONF); + + ret = clk_set_rate(cl->clk_parent, bclk_ratio * samp_rate); + if (ret) { + dev_err(mca->dev, "cluster %d: unable to set clock parent: %d\n", + cl->no, ret); + return ret; + } + } + + return 0; +} + +static const struct snd_soc_dai_ops mca_fe_ops = { + .set_fmt = mca_fe_set_fmt, + .set_bclk_ratio = mca_set_bclk_ratio, + .set_tdm_slot = mca_fe_set_tdm_slot, + .hw_params = mca_fe_hw_params, + .trigger = mca_fe_trigger, +}; + +static bool mca_be_started(struct mca_cluster *cl) +{ + int stream; + + for_each_pcm_streams(stream) + if (cl->port_started[stream]) + return true; + return false; +} + +static int mca_be_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *be = asoc_substream_to_rtd(substream); + struct snd_soc_pcm_runtime *fe; + struct mca_cluster *cl = mca_dai_to_cluster(dai); + struct mca_cluster *fe_cl; + struct mca_data *mca = cl->host; + struct snd_soc_dpcm *dpcm; + + fe = NULL; + + for_each_dpcm_fe(be, substream->stream, dpcm) { + if (fe && dpcm->fe != fe) { + dev_err(mca->dev, "many FE per one BE unsupported\n"); + return -EINVAL; + } + + fe = dpcm->fe; + } + + if (!fe) + return -EINVAL; + + fe_cl = mca_dai_to_cluster(asoc_rtd_to_cpu(fe, 0)); + + if (mca_be_started(cl)) { + /* + * Port is already started in the other direction. + * Make sure there isn't a conflict with another cluster + * driving the port. + */ + if (cl->port_driver != fe_cl->no) + return -EINVAL; + + cl->port_started[substream->stream] = true; + return 0; + } + + writel_relaxed(PORT_ENABLES_CLOCKS | PORT_ENABLES_TX_DATA, + cl->base + REG_PORT_ENABLES); + writel_relaxed(FIELD_PREP(PORT_CLOCK_SEL, fe_cl->no + 1), + cl->base + REG_PORT_CLOCK_SEL); + writel_relaxed(PORT_DATA_SEL_TXA(fe_cl->no), + cl->base + REG_PORT_DATA_SEL); + mutex_lock(&mca->port_mutex); + cl->port_driver = fe_cl->no; + mutex_unlock(&mca->port_mutex); + cl->port_started[substream->stream] = true; + + return 0; +} + +static void mca_be_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mca_cluster *cl = mca_dai_to_cluster(dai); + struct mca_data *mca = cl->host; + + cl->port_started[substream->stream] = false; + + if (!mca_be_started(cl)) { + /* + * Were we the last direction to shutdown? + * Turn off the lights. + */ + writel_relaxed(0, cl->base + REG_PORT_ENABLES); + writel_relaxed(0, cl->base + REG_PORT_DATA_SEL); + mutex_lock(&mca->port_mutex); + cl->port_driver = -1; + mutex_unlock(&mca->port_mutex); + } +} + +static const struct snd_soc_dai_ops mca_be_ops = { + .prepare = mca_be_prepare, + .hw_free = mca_be_hw_free, + .startup = mca_be_startup, + .shutdown = mca_be_shutdown, +}; + +static int mca_set_runtime_hwparams(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct dma_chan *chan) +{ + struct device *dma_dev = chan->device->dev; + struct snd_dmaengine_dai_dma_data dma_data = {}; + int ret; + + struct snd_pcm_hardware hw; + + memset(&hw, 0, sizeof(hw)); + + hw.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED; + hw.periods_min = 2; + hw.periods_max = UINT_MAX; + hw.period_bytes_min = 256; + hw.period_bytes_max = dma_get_max_seg_size(dma_dev); + hw.buffer_bytes_max = SIZE_MAX; + hw.fifo_size = 16; + + ret = snd_dmaengine_pcm_refine_runtime_hwparams(substream, &dma_data, + &hw, chan); + + if (ret) + return ret; + + return snd_soc_set_runtime_hwparams(substream, &hw); +} + +static int mca_pcm_open(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct mca_cluster *cl = mca_dai_to_cluster(asoc_rtd_to_cpu(rtd, 0)); + struct dma_chan *chan = cl->dma_chans[substream->stream]; + int ret; + + if (rtd->dai_link->no_pcm) + return 0; + + ret = mca_set_runtime_hwparams(component, substream, chan); + if (ret) + return ret; + + return snd_dmaengine_pcm_open(substream, chan); +} + +static int mca_hw_params(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct dma_slave_config slave_config; + int ret; + + if (rtd->dai_link->no_pcm) + return 0; + + memset(&slave_config, 0, sizeof(slave_config)); + ret = snd_hwparams_to_dma_slave_config(substream, params, + &slave_config); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + slave_config.dst_port_window_size = + min_t(u32, params_channels(params), 4); + else + slave_config.src_port_window_size = + min_t(u32, params_channels(params), 4); + + return dmaengine_slave_config(chan, &slave_config); +} + +static int mca_close(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + + if (rtd->dai_link->no_pcm) + return 0; + + return snd_dmaengine_pcm_close(substream); +} + +static int mca_trigger(struct snd_soc_component *component, + struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + + if (rtd->dai_link->no_pcm) + return 0; + + /* + * Before we do the PCM trigger proper, insert an opportunity + * to reset the frontend's SERDES. + */ + mca_fe_early_trigger(substream, cmd, asoc_rtd_to_cpu(rtd, 0)); + + return snd_dmaengine_pcm_trigger(substream, cmd); +} + +static snd_pcm_uframes_t mca_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + + if (rtd->dai_link->no_pcm) + return -ENOTSUPP; + + return snd_dmaengine_pcm_pointer(substream); +} + +static int mca_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) +{ + struct mca_cluster *cl = mca_dai_to_cluster(asoc_rtd_to_cpu(rtd, 0)); + unsigned int i; + + if (rtd->dai_link->no_pcm) + return 0; + + for_each_pcm_streams(i) { + struct snd_pcm_substream *substream = + rtd->pcm->streams[i].substream; + struct dma_chan *chan = cl->dma_chans[i]; + + if (!substream) + continue; + + if (!chan) { + dev_err(component->dev, "missing DMA channel for stream %d on SERDES %d\n", + i, cl->no); + return -EINVAL; + } + + snd_pcm_set_managed_buffer(substream, SNDRV_DMA_TYPE_DEV_IRAM, + chan->device->dev, 512 * 1024 * 6, + SIZE_MAX); + } + + return 0; +} + +static const struct snd_soc_component_driver mca_component = { + .name = "apple-mca", + .open = mca_pcm_open, + .close = mca_close, + .hw_params = mca_hw_params, + .trigger = mca_trigger, + .pointer = mca_pointer, + .pcm_construct = mca_pcm_new, +}; + +static void apple_mca_release(struct mca_data *mca) +{ + int i, stream; + + for (i = 0; i < mca->nclusters; i++) { + struct mca_cluster *cl = &mca->clusters[i]; + + for_each_pcm_streams(stream) { + if (IS_ERR_OR_NULL(cl->dma_chans[stream])) + continue; + + dma_release_channel(cl->dma_chans[stream]); + } + + if (!IS_ERR_OR_NULL(cl->clk_parent)) + clk_put(cl->clk_parent); + + if (!IS_ERR_OR_NULL(cl->pd_dev)) + dev_pm_domain_detach(cl->pd_dev, true); + } + + if (mca->pd_link) + device_link_del(mca->pd_link); + + if (!IS_ERR_OR_NULL(mca->pd_dev)) + dev_pm_domain_detach(mca->pd_dev, true); + + reset_control_assert(mca->rstc); +} + +static int apple_mca_probe(struct platform_device *pdev) +{ + struct mca_data *mca; + struct mca_cluster *clusters; + struct snd_soc_dai_driver *dai_drivers; + struct resource *res; + void __iomem *base; + int nclusters; + int ret, i; + + base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); + if (IS_ERR(base)) + return PTR_ERR(base); + + if (resource_size(res) < CLUSTER_STRIDE) + return -EINVAL; + nclusters = (resource_size(res) - CLUSTER_STRIDE) / CLUSTER_STRIDE + 1; + + mca = devm_kzalloc(&pdev->dev, struct_size(mca, clusters, nclusters), + GFP_KERNEL); + if (!mca) + return -ENOMEM; + mca->dev = &pdev->dev; + mca->nclusters = nclusters; + mutex_init(&mca->port_mutex); + platform_set_drvdata(pdev, mca); + clusters = mca->clusters; + + mca->switch_base = + devm_platform_ioremap_resource(pdev, 1); + if (IS_ERR(mca->switch_base)) + return PTR_ERR(mca->switch_base); + + mca->rstc = devm_reset_control_get_optional_shared(&pdev->dev, NULL); + if (IS_ERR(mca->rstc)) + return PTR_ERR(mca->rstc); + + dai_drivers = devm_kzalloc( + &pdev->dev, sizeof(*dai_drivers) * 2 * nclusters, GFP_KERNEL); + if (!dai_drivers) + return -ENOMEM; + + mca->pd_dev = dev_pm_domain_attach_by_id(&pdev->dev, 0); + if (IS_ERR(mca->pd_dev)) + return -EINVAL; + + mca->pd_link = device_link_add(&pdev->dev, mca->pd_dev, + DL_FLAG_STATELESS | DL_FLAG_PM_RUNTIME | + DL_FLAG_RPM_ACTIVE); + if (!mca->pd_link) { + ret = -EINVAL; + /* Prevent an unbalanced reset assert */ + mca->rstc = NULL; + goto err_release; + } + + reset_control_deassert(mca->rstc); + + for (i = 0; i < nclusters; i++) { + struct mca_cluster *cl = &clusters[i]; + struct snd_soc_dai_driver *fe = + &dai_drivers[mca->nclusters + i]; + struct snd_soc_dai_driver *be = &dai_drivers[i]; + int stream; + + cl->host = mca; + cl->no = i; + cl->base = base + CLUSTER_STRIDE * i; + cl->port_driver = -1; + cl->clk_parent = of_clk_get(pdev->dev.of_node, i); + if (IS_ERR(cl->clk_parent)) { + dev_err(&pdev->dev, "unable to obtain clock %d: %ld\n", + i, PTR_ERR(cl->clk_parent)); + ret = PTR_ERR(cl->clk_parent); + goto err_release; + } + cl->pd_dev = dev_pm_domain_attach_by_id(&pdev->dev, i + 1); + if (IS_ERR(cl->pd_dev)) { + dev_err(&pdev->dev, + "unable to obtain cluster %d PD: %ld\n", i, + PTR_ERR(cl->pd_dev)); + ret = PTR_ERR(cl->pd_dev); + goto err_release; + } + + for_each_pcm_streams(stream) { + struct dma_chan *chan; + bool is_tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); +#ifndef USE_RXB_FOR_CAPTURE + char *name = devm_kasprintf(&pdev->dev, GFP_KERNEL, + is_tx ? "tx%da" : "rx%da", + i); +#else + char *name = devm_kasprintf(&pdev->dev, GFP_KERNEL, + is_tx ? "tx%da" : "rx%db", + i); +#endif + + chan = of_dma_request_slave_channel(pdev->dev.of_node, + name); + if (IS_ERR(chan)) { + if (PTR_ERR(chan) != -EPROBE_DEFER) + dev_err(&pdev->dev, + "no %s DMA channel: %ld\n", + name, PTR_ERR(chan)); + + ret = PTR_ERR(chan); + goto err_release; + } + + cl->dma_chans[stream] = chan; + } + + fe->id = i; + fe->name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "mca-pcm-%d", i); + if (!fe->name) { + ret = -ENOMEM; + goto err_release; + } + fe->ops = &mca_fe_ops; + fe->playback.channels_min = 1; + fe->playback.channels_max = 32; + fe->playback.rates = SNDRV_PCM_RATE_8000_192000; + fe->playback.formats = APPLE_MCA_FMTBITS; + fe->capture.channels_min = 1; + fe->capture.channels_max = 32; + fe->capture.rates = SNDRV_PCM_RATE_8000_192000; + fe->capture.formats = APPLE_MCA_FMTBITS; + fe->symmetric_rate = 1; + + fe->playback.stream_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "PCM%d TX", i); + fe->capture.stream_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "PCM%d RX", i); + + if (!fe->playback.stream_name || !fe->capture.stream_name) { + ret = -ENOMEM; + goto err_release; + } + + be->id = i + nclusters; + be->name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "mca-i2s-%d", i); + if (!be->name) { + ret = -ENOMEM; + goto err_release; + } + be->ops = &mca_be_ops; + be->playback.channels_min = 1; + be->playback.channels_max = 32; + be->playback.rates = SNDRV_PCM_RATE_8000_192000; + be->playback.formats = APPLE_MCA_FMTBITS; + be->capture.channels_min = 1; + be->capture.channels_max = 32; + be->capture.rates = SNDRV_PCM_RATE_8000_192000; + be->capture.formats = APPLE_MCA_FMTBITS; + + be->playback.stream_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "I2S%d TX", i); + be->capture.stream_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "I2S%d RX", i); + if (!be->playback.stream_name || !be->capture.stream_name) { + ret = -ENOMEM; + goto err_release; + } + } + + ret = devm_snd_soc_register_component(&pdev->dev, &mca_component, + dai_drivers, nclusters * 2); + if (ret) { + dev_err(&pdev->dev, "unable to register ASoC component: %d\n", + ret); + goto err_release; + } + + return 0; + +err_release: + apple_mca_release(mca); + return ret; +} + +static int apple_mca_remove(struct platform_device *pdev) +{ + struct mca_data *mca = platform_get_drvdata(pdev); + + apple_mca_release(mca); + return 0; +} + +static const struct of_device_id apple_mca_of_match[] = { + { .compatible = "apple,mca", }, + {} +}; +MODULE_DEVICE_TABLE(of, apple_mca_of_match); + +static struct platform_driver apple_mca_driver = { + .driver = { + .name = "apple-mca", + .of_match_table = apple_mca_of_match, + }, + .probe = apple_mca_probe, + .remove = apple_mca_remove, +}; +module_platform_driver(apple_mca_driver); + +MODULE_AUTHOR("Martin PoviÅ¡er <povik+lin@cutebit.org>"); +MODULE_DESCRIPTION("ASoC Apple MCA driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index e868b7e028d6..3763454436c1 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -891,7 +891,6 @@ static int asoc_ssc_init(struct device *dev) int atmel_ssc_set_audio(int ssc_id) { struct ssc_device *ssc; - int ret; /* If we can grab the SSC briefly to parent the DAI device off it */ ssc = ssc_request(ssc_id); @@ -903,9 +902,7 @@ int atmel_ssc_set_audio(int ssc_id) ssc_info[ssc_id].ssc = ssc; } - ret = asoc_ssc_init(&ssc->pdev->dev); - - return ret; + return asoc_ssc_init(&ssc->pdev->dev); } EXPORT_SYMBOL_GPL(atmel_ssc_set_audio); diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index 4850a177803d..ab2d7a791f39 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -196,7 +196,7 @@ struct mchp_spdiftx_dev { struct clk *pclk; struct clk *gclk; unsigned int fmt; - int gclk_enabled:1; + unsigned int gclk_enabled:1; }; static inline int mchp_spdiftx_is_running(struct mchp_spdiftx_dev *dev) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 4d25fb61c652..1430642c8433 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -172,7 +172,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { dev_err_probe(&pdev->dev, ret, - "snd_soc_register_card() failed: %d\n", ret); + "snd_soc_register_card() failed\n"); goto err; } diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d16b4efb88a7..968d0701f2e8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_DA9055 imply SND_SOC_DMIC imply SND_SOC_ES8316 + imply SND_SOC_ES8326 imply SND_SOC_ES8328_SPI imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 @@ -205,6 +206,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_SIMPLE_AMPLIFIER imply SND_SOC_SIMPLE_MUX imply SND_SOC_SPDIF + imply SND_SOC_SRC4XXX_I2C imply SND_SOC_SSM2305 imply SND_SOC_SSM2518 imply SND_SOC_SSM2602_SPI @@ -608,7 +610,7 @@ config SND_SOC_BT_SCO config SND_SOC_CPCAP tristate "Motorola CPCAP codec" - depends on MFD_CPCAP + depends on MFD_CPCAP || COMPILE_TEST config SND_SOC_CQ0093VC tristate @@ -913,6 +915,10 @@ config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C +config SND_SOC_ES8326 + tristate "Everest Semi ES8326 CODEC" + depends on I2C + config SND_SOC_ES8328 tristate @@ -966,7 +972,7 @@ config SND_SOC_LM49453 config SND_SOC_LOCHNAGAR_SC tristate "Lochnagar Sound Card" - depends on MFD_LOCHNAGAR + depends on MFD_LOCHNAGAR || COMPILE_TEST help This driver support the sound card functionality of the Cirrus Logic Lochnagar audio development board. @@ -1191,7 +1197,7 @@ config SND_SOC_RK3328 config SND_SOC_RK817 tristate "Rockchip RK817 audio CODEC" - depends on MFD_RK808 + depends on MFD_RK808 || COMPILE_TEST && I2C select REGMAP_I2C config SND_SOC_RL6231 @@ -1471,6 +1477,18 @@ config SND_SOC_SIMPLE_MUX config SND_SOC_SPDIF tristate "S/PDIF CODEC" +config SND_SOC_SRC4XXX_I2C + tristate "Texas Instruments SRC4XXX DIR/DIT and SRC codecs" + depends on I2C + select SND_SOC_SRC4XXX + help + Enable support for the TI SRC4XXX family of codecs. These include the + scr4392 which has digital receivers, transmitters, and + a sample rate converter, including numerous ports. + +config SND_SOC_SRC4XXX + tristate + config SND_SOC_SSM2305 tristate "Analog Devices SSM2305 Class-D Amplifier" help @@ -1727,7 +1745,7 @@ config SND_SOC_WCD934X tristate "WCD9340/WCD9341 Codec" depends on COMMON_CLK select SND_SOC_WCD_MBHC - depends on MFD_WCD934X + depends on MFD_WCD934X || COMPILE_TEST help The WCD9340/9341 is a audio codec IC Integrated in Qualcomm SoCs like SDM845. diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 92fd441d426a..16a01635dd04 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -100,6 +100,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o +snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -231,6 +232,8 @@ snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o snd-soc-si476x-objs := si476x.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-src4xxx-objs := src4xxx.o +snd-soc-src4xxx-i2c-objs := src4xxx-i2c.o snd-soc-ssm2305-objs := ssm2305.o snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o @@ -455,6 +458,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o +obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o @@ -579,6 +583,8 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SRC4XXX) += snd-soc-src4xxx.o +obj-$(CONFIG_SND_SOC_SRC4XXX_I2C) += snd-soc-src4xxx-i2c.o obj-$(CONFIG_SND_SOC_SSM2305) += snd-soc-ssm2305.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d545a593a251..de1e276bdf7d 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -12,7 +12,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/version.h> -#include <linux/kernel.h> +#include <linux/types.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/i2c.h> @@ -37,6 +37,14 @@ #include "cs42l42.h" #include "cirrus_legacy.h" +static const char * const cs42l42_supply_names[] = { + "VA", + "VP", + "VCP", + "VD_FILT", + "VL", +}; + static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_FRZ_CTL, 0x00 }, { CS42L42_SRC_CTL, 0x10 }, @@ -395,7 +403,7 @@ static int cs42l42_slow_start_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); u8 val; - /* all bits of SLOW_START_EN much change together */ + /* all bits of SLOW_START_EN must change together */ switch (ucontrol->value.integer.value[0]) { case 0: val = 0; @@ -885,22 +893,21 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); unsigned int channels = params_channels(params); unsigned int width = (params_width(params) / 8) - 1; + unsigned int slot_width = 0; unsigned int val = 0; int ret; cs42l42->srate = params_rate(params); - cs42l42->bclk = snd_soc_params_to_bclk(params); - - /* I2S frame always has 2 channels even for mono audio */ - if (channels == 1) - cs42l42->bclk *= 2; /* * Assume 24-bit samples are in 32-bit slots, to prevent SCLK being * more than assumed (which would result in overclocking). */ if (params_width(params) == 24) - cs42l42->bclk = (cs42l42->bclk / 3) * 4; + slot_width = 32; + + /* I2S frame always has multiple of 2 channels */ + cs42l42->bclk = snd_soc_tdm_params_to_bclk(params, slot_width, 0, 2); switch (substream->stream) { case SNDRV_PCM_STREAM_CAPTURE: @@ -2214,6 +2221,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client) return ret; } + BUILD_BUG_ON(ARRAY_SIZE(cs42l42_supply_names) != ARRAY_SIZE(cs42l42->supplies)); for (i = 0; i < ARRAY_SIZE(cs42l42->supplies); i++) cs42l42->supplies[i].supply = cs42l42_supply_names[i]; diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 5f50970375d4..50299c9f283a 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -12,18 +12,15 @@ #ifndef __CS42L42_H__ #define __CS42L42_H__ +#include <dt-bindings/sound/cs42l42.h> +#include <linux/device.h> +#include <linux/gpio.h> #include <linux/mutex.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> #include <sound/jack.h> #include <sound/cs42l42.h> -static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = { - "VA", - "VP", - "VCP", - "VD_FILT", - "VL", -}; - struct cs42l42_private { struct regmap *regmap; struct device *dev; diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index ca4d47cc9c91..06c6ad3ca2b7 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1666,10 +1666,9 @@ static int cs43130_show_dc(struct device *dev, char *buf, u8 ch) struct cs43130_private *cs43130 = i2c_get_clientdata(client); if (!cs43130->hpload_done) - return scnprintf(buf, PAGE_SIZE, "NO_HPLOAD\n"); + return sysfs_emit(buf, "NO_HPLOAD\n"); else - return scnprintf(buf, PAGE_SIZE, "%u\n", - cs43130->hpload_dc[ch]); + return sysfs_emit(buf, "%u\n", cs43130->hpload_dc[ch]); } static ssize_t hpload_dc_l_show(struct device *dev, @@ -1705,8 +1704,8 @@ static int cs43130_show_ac(struct device *dev, char *buf, u8 ch) if (cs43130->hpload_done && cs43130->ac_meas) { for (i = 0; i < ARRAY_SIZE(cs43130_ac_freq); i++) { - tmp = scnprintf(buf + j, PAGE_SIZE - j, "%u\n", - cs43130->hpload_ac[i][ch]); + tmp = sysfs_emit_at(buf, j, "%u\n", + cs43130->hpload_ac[i][ch]); if (!tmp) break; @@ -1715,7 +1714,7 @@ static int cs43130_show_ac(struct device *dev, char *buf, u8 ch) return j; } else { - return scnprintf(buf, PAGE_SIZE, "NO_HPLOAD\n"); + return sysfs_emit(buf, "NO_HPLOAD\n"); } } diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index de7185f73e1e..8643014472ae 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -767,9 +767,31 @@ static void es8316_remove(struct snd_soc_component *component) clk_disable_unprepare(es8316->mclk); } +static int es8316_resume(struct snd_soc_component *component) +{ + struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(es8316->regmap, false); + regcache_sync(es8316->regmap); + + return 0; +} + +static int es8316_suspend(struct snd_soc_component *component) +{ + struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(es8316->regmap, true); + regcache_mark_dirty(es8316->regmap); + + return 0; +} + static const struct snd_soc_component_driver soc_component_dev_es8316 = { .probe = es8316_probe, .remove = es8316_remove, + .resume = es8316_resume, + .suspend = es8316_suspend, .set_jack = es8316_set_jack, .controls = es8316_snd_controls, .num_controls = ARRAY_SIZE(es8316_snd_controls), diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c new file mode 100755 index 000000000000..87c1cc16592b --- /dev/null +++ b/sound/soc/codecs/es8326.c @@ -0,0 +1,905 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// es8326.c -- es8326 ALSA SoC audio driver +// Copyright Everest Semiconductor Co., Ltd +// +// Authors: David Yang <yangxiaohua@everest-semi.com> +// + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8326.h" + +struct es8326_priv { + struct clk *mclk; + struct i2c_client *i2c; + struct regmap *regmap; + struct snd_soc_component *component; + struct delayed_work jack_detect_work; + struct delayed_work button_press_work; + struct snd_soc_jack *jack; + int irq; + /* The lock protects the situation that an irq is generated + * while enabling or disabling or during an irq. + */ + struct mutex lock; + u8 mic1_src; + u8 mic2_src; + u8 jack_pol; + u8 interrupt_src; + u8 interrupt_clk; + bool jd_inverted; + unsigned int sysclk; +}; + +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_analog_pga_tlv, 0, 300, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_pga_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(softramp_rate, 0, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_target_tlv, -3200, 200, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_recovery_tlv, -125, 250, 0); + +static const char *const winsize[] = { + "0.25db/2 LRCK", + "0.25db/4 LRCK", + "0.25db/8 LRCK", + "0.25db/16 LRCK", + "0.25db/32 LRCK", + "0.25db/64 LRCK", + "0.25db/128 LRCK", + "0.25db/256 LRCK", + "0.25db/512 LRCK", + "0.25db/1024 LRCK", + "0.25db/2048 LRCK", + "0.25db/4096 LRCK", + "0.25db/8192 LRCK", + "0.25db/16384 LRCK", + "0.25db/32768 LRCK", + "0.25db/65536 LRCK", +}; + +static const char *const dacpol_txt[] = { + "Normal", "R Invert", "L Invert", "L + R Invert" }; + +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8326_DAC_DSM, 4, 4, dacpol_txt); +static const struct soc_enum alc_winsize = + SOC_ENUM_SINGLE(ES8326_ADC_RAMPRATE, 4, 16, winsize); +static const struct soc_enum drc_winsize = + SOC_ENUM_SINGLE(ES8326_DRC_WINSIZE, 4, 16, winsize); + +static const struct snd_kcontrol_new es8326_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", ES8326_DAC_VOL, 0, 0xbf, 0, dac_vol_tlv), + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE_TLV("DAC Ramp Rate", ES8326_DAC_RAMPRATE, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE_TLV("DRC Recovery Level", ES8326_DRC_RECOVERY, 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("DRC Winsize", drc_winsize), + SOC_SINGLE_TLV("DRC Target Level", ES8326_DRC_WINSIZE, 0, 0x0f, 0, drc_target_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ES8326_ADC1_VOL, ES8326_ADC2_VOL, 0, 0xff, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC PGA Volume", ES8326_ADC_SCALE, 4, 0, 5, 0, adc_pga_tlv), + SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8326_PGAGAIN, 0, 10, 0, adc_analog_pga_tlv), + SOC_SINGLE_TLV("ADC Ramp Rate", ES8326_ADC_RAMPRATE, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE("ALC Capture Switch", ES8326_ALC_RECOVERY, 3, 1, 0), + SOC_SINGLE_TLV("ALC Capture Recovery Level", ES8326_ALC_LEVEL, + 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("ALC Capture Winsize", alc_winsize), + SOC_SINGLE_TLV("ALC Capture Target Level", ES8326_ALC_LEVEL, + 0, 0x0f, 0, drc_target_tlv), + +}; + +static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* ADC Digital Mute */ + SND_SOC_DAPM_PGA("ADC L1", ES8326_ADC_MUTE, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R1", ES8326_ADC_MUTE, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC L2", ES8326_ADC_MUTE, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R2", ES8326_ADC_MUTE, 3, 1, NULL, 0), + + /* Analog Power Supply*/ + SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN, 1, 1), + SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", ES8326_ANA_MICBIAS, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", ES8326_ANA_MICBIAS, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX, 3, 0, NULL, 0), + + /* Headphone Charge Pump and Output */ + SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRIVER, + 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRIVER, + 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRIVER, + 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRIVER, + 0, 1, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL, + ES8326_HPOR_SHIFT, 7, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL, + 0, 7, 7, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8326_dapm_routes[] = { + {"ADC L1", NULL, "MIC1"}, + {"ADC R1", NULL, "MIC2"}, + {"ADC L2", NULL, "MIC3"}, + {"ADC R2", NULL, "MIC4"}, + + {"ADC L", NULL, "ADC L1"}, + {"ADC R", NULL, "ADC R1"}, + {"ADC L", NULL, "ADC L2"}, + {"ADC R", NULL, "ADC R2"}, + + {"I2S OUT", NULL, "ADC L"}, + {"I2S OUT", NULL, "ADC R"}, + + {"I2S OUT", NULL, "Analog Power"}, + {"I2S OUT", NULL, "ADC Vref"}, + {"I2S OUT", NULL, "Vref Power"}, + {"I2S OUT", NULL, "IBias Power"}, + {"I2S IN", NULL, "Analog Power"}, + {"I2S IN", NULL, "DAC Vref"}, + {"I2S IN", NULL, "Vref Power"}, + {"I2S IN", NULL, "IBias Power"}, + + {"Right DAC", NULL, "I2S IN"}, + {"Left DAC", NULL, "I2S IN"}, + + {"LHPMIX", NULL, "Left DAC"}, + {"RHPMIX", NULL, "Right DAC"}, + + {"HPOR", NULL, "HPOR Cal"}, + {"HPOL", NULL, "HPOL Cal"}, + {"HPOR", NULL, "HPOR Supply"}, + {"HPOL", NULL, "HPOL Supply"}, + {"HPOL", NULL, "Headphone Charge Pump"}, + {"HPOR", NULL, "Headphone Charge Pump"}, + {"HPOL", NULL, "Headphone Driver Bias"}, + {"HPOR", NULL, "Headphone Driver Bias"}, + {"HPOL", NULL, "Headphone LDO"}, + {"HPOR", NULL, "Headphone LDO"}, + {"HPOL", NULL, "Headphone Reference"}, + {"HPOR", NULL, "Headphone Reference"}, + + {"HPOL", NULL, "LHPMIX"}, + {"HPOR", NULL, "RHPMIX"}, +}; + +static const struct regmap_range es8326_volatile_ranges[] = { + regmap_reg_range(ES8326_HP_DETECT, ES8326_HP_DETECT), +}; + +static const struct regmap_access_table es8326_volatile_table = { + .yes_ranges = es8326_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), +}; + +static const struct regmap_config es8326_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0xff, + .volatile_table = &es8326_volatile_table, + .cache_type = REGCACHE_RBTREE, +}; + +struct _coeff_div { + u16 fs; + u32 rate; + u32 mclk; + u8 reg4; + u8 reg5; + u8 reg6; + u8 reg7; + u8 reg8; + u8 reg9; + u8 rega; + u8 regb; +}; + +/* codec hifi mclk clock divider coefficients */ +/* {ratio, LRCK, MCLK, REG04, REG05, REG06, REG07, REG08, REG09, REG10, REG11} */ +static const struct _coeff_div coeff_div[] = { + {32, 8000, 256000, 0x60, 0x00, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {32, 16000, 512000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {32, 44100, 1411200, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {32, 48000, 1536000, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {36, 8000, 288000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {36, 16000, 576000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {48, 8000, 384000, 0x60, 0x02, 0x1F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {48, 16000, 768000, 0x20, 0x02, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {48, 48000, 2304000, 0x00, 0x02, 0x0D, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 8000, 512000, 0x60, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {64, 16000, 1024000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + + {64, 44100, 2822400, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 48000, 3072000, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {72, 8000, 576000, 0x20, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {72, 16000, 1152000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {96, 8000, 768000, 0x60, 0x02, 0x1D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {96, 16000, 1536000, 0x20, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {100, 48000, 4800000, 0x04, 0x04, 0x3F, 0x6D, 0x38, 0x08, 0x4f, 0x1f}, + {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {128, 8000, 1024000, 0x60, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {128, 16000, 2048000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {128, 44100, 5644800, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {128, 48000, 6144000, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {192, 16000, 3072000, 0x20, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {256, 8000, 2048000, 0x60, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {256, 44100, 11289600, 0x00, 0x00, 0x10, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {256, 48000, 12288000, 0x00, 0x00, 0x30, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {384, 48000, 18432000, 0x00, 0x02, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {400, 48000, 19200000, 0x09, 0x04, 0x0f, 0x6d, 0x3a, 0x0A, 0x4F, 0x1F}, + {500, 48000, 24000000, 0x18, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {512, 44100, 22579200, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 48000, 24576000, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {800, 48000, 38400000, 0x00, 0x18, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {1024, 8000, 8192000, 0x60, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {1024, 16000, 16384000, 0x20, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1152, 16000, 18432000, 0x20, 0x08, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1536, 8000, 12288000, 0x60, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + + {1536, 16000, 24576000, 0x20, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1625, 8000, 13000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {1625, 16000, 26000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {2048, 8000, 16384000, 0x60, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {2304, 8000, 18432000, 0x40, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x5F}, + {3072, 8000, 24576000, 0x60, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {3250, 8000, 26000000, 0x0C, 0x18, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +static int es8326_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *codec = codec_dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + + es8326->sysclk = freq; + + return 0; +} + +static int es8326_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFP: + snd_soc_component_update_bits(component, ES8326_RESET, + ES8326_MASTER_MODE_EN, ES8326_MASTER_MODE_EN); + break; + case SND_SOC_DAIFMT_CBC_CFC: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + dev_err(component->dev, "Codec driver does not support right justified\n"); + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ES8326_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ES8326_DAIFMT_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ES8326_DAIFMT_DSP_B; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, ES8326_FMT, ES8326_DAIFMT_MASK, iface); + + return 0; +} + +static int es8326_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + u8 srate = 0; + int coeff; + + coeff = get_coeff(es8326->sysclk, params_rate(params)); + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srate |= ES8326_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srate |= ES8326_S20_3_LE; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + srate |= ES8326_S18_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + srate |= ES8326_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + srate |= ES8326_S32_LE; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_component_update_bits(component, ES8326_FMT, ES8326_DATA_LEN_MASK, srate); + + if (coeff >= 0) { + regmap_write(es8326->regmap, ES8326_CLK_DIV1, + coeff_div[coeff].reg4); + regmap_write(es8326->regmap, ES8326_CLK_DIV2, + coeff_div[coeff].reg5); + regmap_write(es8326->regmap, ES8326_CLK_DLL, + coeff_div[coeff].reg6); + regmap_write(es8326->regmap, ES8326_CLK_MUX, + coeff_div[coeff].reg7); + regmap_write(es8326->regmap, ES8326_CLK_ADC_SEL, + coeff_div[coeff].reg8); + regmap_write(es8326->regmap, ES8326_CLK_DAC_SEL, + coeff_div[coeff].reg9); + regmap_write(es8326->regmap, ES8326_CLK_ADC_OSR, + coeff_div[coeff].rega); + regmap_write(es8326->regmap, ES8326_CLK_DAC_OSR, + coeff_div[coeff].regb); + } else { + dev_warn(component->dev, "Clock coefficients do not match"); + } + + return 0; +} + +static int es8326_set_bias_level(struct snd_soc_component *codec, + enum snd_soc_bias_level level) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + ret = clk_prepare_enable(es8326->mclk); + if (ret) + return ret; + regmap_write(es8326->regmap, ES8326_RESET, ES8326_PWRUP_SEQ_EN); + regmap_write(es8326->regmap, ES8326_INTOUT_IO, 0x45); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, + (ES8326_IO_DMIC_CLK << ES8326_SDINOUT1_SHIFT)); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE, 0x05); + regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x02); + regmap_write(es8326->regmap, ES8326_PGA_PDN, 0x40); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0xAA); + regmap_write(es8326->regmap, ES8326_RESET, ES8326_CSM_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + clk_disable_unprepare(es8326->mclk); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0x11); + regmap_write(es8326->regmap, ES8326_RESET, ES8326_CSM_OFF); + regmap_write(es8326->regmap, ES8326_PGA_PDN, 0xF8); + regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x00); + regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x08); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_RESET, + ES8326_CODEC_RESET | ES8326_PWRUP_SEQ_EN); + break; + } + + return 0; +} + +#define es8326_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops es8326_ops = { + .hw_params = es8326_pcm_hw_params, + .set_fmt = es8326_set_dai_fmt, + .set_sysclk = es8326_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver es8326_dai = { + .name = "ES8326 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = es8326_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = es8326_FORMATS, + }, + .ops = &es8326_ops, + .symmetric_rate = 1, +}; + +static void es8326_enable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void es8326_disable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +/* + * For button detection, set the following in soundcard + * snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + * snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + * snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + */ +static void es8326_jack_button_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, button_press_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + static int button_to_report, press_count; + static int prev_button, cur_button; + + if (!(es8326->jack->status & SND_JACK_HEADSET)) /* Jack unplugged */ + return; + + mutex_lock(&es8326->lock); + iface = snd_soc_component_read(comp, ES8326_HP_DETECT); + switch (iface) { + case 0x93: + /* pause button detected */ + cur_button = SND_JACK_BTN_0; + break; + case 0x6f: + /* button volume up */ + cur_button = SND_JACK_BTN_1; + break; + case 0x27: + /* button volume down */ + cur_button = SND_JACK_BTN_2; + break; + case 0x1e: + /* button released or not pressed */ + cur_button = 0; + break; + default: + break; + } + + if ((prev_button == cur_button) && (cur_button != 0)) { + press_count++; + if (press_count > 10) { + /* report a press every 500ms */ + snd_soc_jack_report(es8326->jack, cur_button, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + press_count = 0; + } + button_to_report = cur_button; + queue_delayed_work(system_wq, &es8326->button_press_work, + msecs_to_jiffies(50)); + } else if (prev_button != cur_button) { + /* mismatch, detect again */ + prev_button = cur_button; + queue_delayed_work(system_wq, &es8326->button_press_work, + msecs_to_jiffies(50)); + } else { + /* released or no pressed */ + if (button_to_report != 0) { + snd_soc_jack_report(es8326->jack, button_to_report, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + snd_soc_jack_report(es8326->jack, 0, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + button_to_report = 0; + } + } + mutex_unlock(&es8326->lock); +} + +static void es8326_jack_detect_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, jack_detect_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + + mutex_lock(&es8326->lock); + iface = snd_soc_component_read(comp, ES8326_HP_DETECT); + dev_dbg(comp->dev, "gpio flag %#04x", iface); + if ((iface & ES8326_HPINSERT_FLAG) == 0) { + /* Jack unplugged or spurious IRQ */ + dev_dbg(comp->dev, "No headset detected"); + if (es8326->jack->status & SND_JACK_HEADPHONE) { + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC, es8326->mic2_src); + es8326_disable_micbias(comp); + } + } else if ((iface & ES8326_HPINSERT_FLAG) == ES8326_HPINSERT_FLAG) { + if (es8326->jack->status & SND_JACK_HEADSET) { + /* detect button */ + queue_delayed_work(system_wq, &es8326->button_press_work, 10); + } else { + if ((iface & ES8326_HPBUTTON_FLAG) == 0x00) { + dev_dbg(comp->dev, "Headset detected"); + snd_soc_jack_report(es8326->jack, + SND_JACK_HEADSET, SND_JACK_HEADSET); + snd_soc_component_write(comp, + ES8326_ADC1_SRC, es8326->mic1_src); + } else { + dev_dbg(comp->dev, "Headphone detected"); + snd_soc_jack_report(es8326->jack, + SND_JACK_HEADPHONE, SND_JACK_HEADSET); + } + } + } + mutex_unlock(&es8326->lock); +} + +static irqreturn_t es8326_irq(int irq, void *dev_id) +{ + struct es8326_priv *es8326 = dev_id; + struct snd_soc_component *comp = es8326->component; + + if (!es8326->jack) + goto out; + + es8326_enable_micbias(comp); + + if (es8326->jack->status & SND_JACK_HEADSET) + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(10)); + else + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(300)); + +out: + return IRQ_HANDLED; +} + +static int es8326_resume(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + unsigned int reg; + + regcache_cache_only(es8326->regmap, false); + regcache_sync(es8326->regmap); + + regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_ON); + /* Two channel ADC */ + regmap_write(es8326->regmap, ES8326_PULLUP_CTL, 0x02); + regmap_write(es8326->regmap, ES8326_CLK_INV, 0x00); + regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC, 0x1F); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS1, 0xC8); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS2, 0x88); + regmap_write(es8326->regmap, ES8326_CLK_CAL_TIME, 0x20); + regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x08); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0x22); + regmap_write(es8326->regmap, ES8326_ADC1_SRC, es8326->mic1_src); + regmap_write(es8326->regmap, ES8326_ADC2_SRC, es8326->mic2_src); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0x88); + regmap_write(es8326->regmap, ES8326_HP_DET, + ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); + regmap_write(es8326->regmap, ES8326_INT_SOURCE, es8326->interrupt_src); + regmap_write(es8326->regmap, ES8326_INTOUT_IO, es8326->interrupt_clk); + regmap_write(es8326->regmap, ES8326_RESET, ES8326_CSM_ON); + snd_soc_component_update_bits(component, ES8326_PGAGAIN, + ES8326_MIC_SEL_MASK, ES8326_MIC1_SEL); + + regmap_read(es8326->regmap, ES8326_CHIP_VERSION, ®); + if ((reg & ES8326_VERSION_B) == 1) { + regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xDD); + regmap_write(es8326->regmap, ES8326_ANA_VSEL, 0x7F); + regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x0F); + /* enable button detect */ + regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xA0); + } + + es8326_irq(es8326->irq, es8326); + return 0; +} + +static int es8326_suspend(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + cancel_delayed_work_sync(&es8326->jack_detect_work); + es8326_disable_micbias(component); + + regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF); + regcache_cache_only(es8326->regmap, true); + regcache_mark_dirty(es8326->regmap); + + return 0; +} + +static int es8326_probe(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + int ret; + + es8326->component = component; + es8326->jd_inverted = device_property_read_bool(component->dev, + "everest,jack-detect-inverted"); + + ret = device_property_read_u8(component->dev, "everest,mic1-src", &es8326->mic1_src); + if (ret != 0) { + dev_dbg(component->dev, "mic1-src return %d", ret); + es8326->mic1_src = ES8326_ADC_AMIC; + } + dev_dbg(component->dev, "mic1-src %x", es8326->mic1_src); + + ret = device_property_read_u8(component->dev, "everest,mic2-src", &es8326->mic2_src); + if (ret != 0) { + dev_dbg(component->dev, "mic2-src return %d", ret); + es8326->mic2_src = ES8326_ADC_DMIC; + } + dev_dbg(component->dev, "mic2-src %x", es8326->mic2_src); + + ret = device_property_read_u8(component->dev, "everest,jack-pol", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "jack-pol return %d", ret); + es8326->jack_pol = ES8326_HP_DET_BUTTON_POL | ES8326_HP_TYPE_OMTP; + } + dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); + + ret = device_property_read_u8(component->dev, "everest,interrupt-src", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "interrupt-src return %d", ret); + es8326->interrupt_src = ES8326_HP_DET_SRC_PIN9; + } + dev_dbg(component->dev, "interrupt-src %x", es8326->interrupt_src); + + ret = device_property_read_u8(component->dev, "everest,interrupt-clk", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "interrupt-clk return %d", ret); + es8326->interrupt_clk = 0x45; + } + dev_dbg(component->dev, "interrupt-clk %x", es8326->interrupt_clk); + + es8326_resume(component); + return 0; +} + +static void es8326_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + mutex_lock(&es8326->lock); + if (es8326->jd_inverted) + snd_soc_component_update_bits(component, ES8326_HP_DET, + ES8326_HP_DET_JACK_POL, ~es8326->jack_pol); + es8326->jack = jack; + + mutex_unlock(&es8326->lock); + es8326_irq(es8326->irq, es8326); +} + +static void es8326_disable_jack_detect(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "Enter into %s\n", __func__); + if (!es8326->jack) + return; /* Already disabled (or never enabled) */ + cancel_delayed_work_sync(&es8326->jack_detect_work); + + mutex_lock(&es8326->lock); + if (es8326->jack->status & SND_JACK_MICROPHONE) { + es8326_disable_micbias(component); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + } + es8326->jack = NULL; + mutex_unlock(&es8326->lock); +} + +static int es8326_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + es8326_enable_jack_detect(component, jack); + else + es8326_disable_jack_detect(component); + + return 0; +} + +static void es8326_remove(struct snd_soc_component *component) +{ + es8326_disable_jack_detect(component); + es8326_set_bias_level(component, SND_SOC_BIAS_OFF); +} + +static const struct snd_soc_component_driver soc_component_dev_es8326 = { + .probe = es8326_probe, + .remove = es8326_remove, + .resume = es8326_resume, + .suspend = es8326_suspend, + .set_bias_level = es8326_set_bias_level, + .set_jack = es8326_set_jack, + .dapm_widgets = es8326_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8326_dapm_widgets), + .dapm_routes = es8326_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8326_dapm_routes), + .controls = es8326_snd_controls, + .num_controls = ARRAY_SIZE(es8326_snd_controls), + .use_pmdown_time = 1, + .endianness = 1, +}; + +static int es8326_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct es8326_priv *es8326; + int ret; + + es8326 = devm_kzalloc(&i2c->dev, sizeof(struct es8326_priv), GFP_KERNEL); + if (!es8326) + return -ENOMEM; + + i2c_set_clientdata(i2c, es8326); + es8326->i2c = i2c; + mutex_init(&es8326->lock); + es8326->regmap = devm_regmap_init_i2c(i2c, &es8326_regmap_config); + if (IS_ERR(es8326->regmap)) { + ret = PTR_ERR(es8326->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + es8326->irq = i2c->irq; + INIT_DELAYED_WORK(&es8326->jack_detect_work, + es8326_jack_detect_handler); + INIT_DELAYED_WORK(&es8326->button_press_work, + es8326_jack_button_handler); + /* ES8316 is level-based while ES8326 is edge-based */ + ret = devm_request_threaded_irq(&i2c->dev, es8326->irq, NULL, es8326_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "es8326", es8326); + if (ret) { + dev_warn(&i2c->dev, "Failed to request IRQ: %d: %d\n", + es8326->irq, ret); + es8326->irq = -ENXIO; + } + + es8326->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(es8326->mclk)) { + dev_err(&i2c->dev, "unable to get mclk\n"); + return PTR_ERR(es8326->mclk); + } + if (!es8326->mclk) + dev_warn(&i2c->dev, "assuming static mclk\n"); + + ret = clk_prepare_enable(es8326->mclk); + if (ret) { + dev_err(&i2c->dev, "unable to enable mclk\n"); + return ret; + } + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_es8326, + &es8326_dai, 1); +} + +static const struct i2c_device_id es8326_i2c_id[] = { + {"es8326", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8326_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id es8326_of_match[] = { + { .compatible = "everest,es8326", }, + {} +}; +MODULE_DEVICE_TABLE(of, es8326_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id es8326_acpi_match[] = { + {"ESSX8326", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8326_acpi_match); +#endif + +static struct i2c_driver es8326_i2c_driver = { + .driver = { + .name = "es8326", + .acpi_match_table = ACPI_PTR(es8326_acpi_match), + .of_match_table = of_match_ptr(es8326_of_match), + }, + .probe = es8326_i2c_probe, + .id_table = es8326_i2c_id, +}; +module_i2c_driver(es8326_i2c_driver); + +MODULE_DESCRIPTION("ASoC es8326 driver"); +MODULE_AUTHOR("David Yang <yangxiaohua@everest-semi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h new file mode 100755 index 000000000000..8e5ffe5ee10d --- /dev/null +++ b/sound/soc/codecs/es8326.h @@ -0,0 +1,182 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * es8326.h -- es8326 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua@everest-semi.com> + */ + +#ifndef _ES8326_H +#define _ES8326_H + +#define CONFIG_HHTECH_MINIPMP 1 + +/* ES8326 register space */ +#define ES8326_RESET 0x00 +#define ES8326_CLK_CTL 0x01 +#define ES8326_CLK_INV 0x02 +#define ES8326_CLK_RESAMPLE 0x03 +#define ES8326_CLK_DIV1 0x04 +#define ES8326_CLK_DIV2 0x05 +#define ES8326_CLK_DLL 0x06 +#define ES8326_CLK_MUX 0x07 +#define ES8326_CLK_ADC_SEL 0x08 +#define ES8326_CLK_DAC_SEL 0x09 +#define ES8326_CLK_ADC_OSR 0x0a +#define ES8326_CLK_DAC_OSR 0x0b +#define ES8326_CLK_DIV_CPC 0x0c +#define ES8326_CLK_DIV_BCLK 0x0d +#define ES8326_CLK_TRI 0x0e +#define ES8326_CLK_DIV_LRCK 0x0f +#define ES8326_CLK_VMIDS1 0x10 +#define ES8326_CLK_VMIDS2 0x11 +#define ES8326_CLK_CAL_TIME 0x12 +#define ES8326_FMT 0x13 + +#define ES8326_DAC_MUTE 0x14 +#define ES8326_ADC_MUTE 0x15 +#define ES8326_ANA_PDN 0x16 +#define ES8326_PGA_PDN 0x17 +#define ES8326_VMIDSEL 0x18 +#define ES8326_ANA_LP 0x19 +#define ES8326_ANA_DMS 0x1a +#define ES8326_ANA_MICBIAS 0x1b +#define ES8326_ANA_VSEL 0x1c +#define ES8326_SYS_BIAS 0x1d +#define ES8326_BIAS_SW1 0x1e +#define ES8326_BIAS_SW2 0x1f +#define ES8326_BIAS_SW3 0x20 +#define ES8326_BIAS_SW4 0x21 +#define ES8326_VMIDLOW 0x22 +#define ES8326_PGAGAIN 0x23 +#define ES8326_HP_DRIVER 0x24 +#define ES8326_DAC2HPMIX 0x25 +#define ES8326_HP_VOL 0x26 +#define ES8326_HP_CAL 0x27 +#define ES8326_HP_DRIVER_REF 0x28 +#define ES8326_ADC_SCALE 0x29 +#define ES8326_ADC1_SRC 0x2a +#define ES8326_ADC2_SRC 0x2b +#define ES8326_ADC1_VOL 0x2c +#define ES8326_ADC2_VOL 0x2d +#define ES8326_ADC_RAMPRATE 0x2e +#define ES8326_ALC_RECOVERY 0x32 +#define ES8326_ALC_LEVEL 0x33 +#define ES8326_ADC_HPFS1 0x34 +#define ES8326_ADC_HPFS2 0x35 +#define ES8326_ADC_EQ 0x36 +#define ES8326_HP_OFFSET_CAL 0x4A +#define ES8326_HPL_OFFSET_INI 0x4B +#define ES8326_HPR_OFFSET_INI 0x4C +#define ES8326_DAC_DSM 0x4D +#define ES8326_DAC_RAMPRATE 0x4E +#define ES8326_DAC_VPPSCALE 0x4F +#define ES8326_DAC_VOL 0x50 +#define ES8326_DRC_RECOVERY 0x53 +#define ES8326_DRC_WINSIZE 0x54 +#define ES8326_HPJACK_TIMER 0x56 +#define ES8326_HP_DET 0x57 +#define ES8326_INT_SOURCE 0x58 +#define ES8326_INTOUT_IO 0x59 +#define ES8326_SDINOUT1_IO 0x5A +#define ES8326_SDINOUT23_IO 0x5B +#define ES8326_JACK_PULSE 0x5C + +#define ES8326_PULLUP_CTL 0xF9 +#define ES8326_HP_DETECT 0xFB +#define ES8326_CHIP_ID1 0xFD +#define ES8326_CHIP_ID2 0xFE +#define ES8326_CHIP_VERSION 0xFF + +/* ES8326_RESET */ +#define ES8326_CSM_ON (1 << 7) +#define ES8326_MASTER_MODE_EN (1 << 6) +#define ES8326_PWRUP_SEQ_EN (1 << 5) +#define ES8326_CODEC_RESET (0x0f << 0) +#define ES8326_CSM_OFF (0 << 7) + +/* ES8326_CLK_CTL */ +#define ES8326_CLK_ON (0x7f << 0) +#define ES8326_CLK_OFF (0 << 0) + +/* ES8326_CLK_INV */ +#define ES8326_BCLK_AS_MCLK (1 << 3) + +/* ES8326_FMT */ +#define ES8326_S24_LE (0 << 2) +#define ES8326_S20_3_LE (1 << 2) +#define ES8326_S18_LE (2 << 2) +#define ES8326_S16_LE (3 << 2) +#define ES8326_S32_LE (4 << 2) +#define ES8326_DATA_LEN_MASK (7 << 2) + +#define ES8326_DAIFMT_MASK ((1 << 5) | (3 << 0)) +#define ES8326_DAIFMT_I2S 0 +#define ES8326_DAIFMT_LEFT_J (1 << 0) +#define ES8326_DAIFMT_DSP_A (3 << 0) +#define ES8326_DAIFMT_DSP_B ((1 << 5) | (3 << 0)) + +/* ES8326_PGAGAIN */ +#define ES8326_MIC_SEL_MASK (3 << 4) +#define ES8326_MIC1_SEL (1 << 4) +#define ES8326_MIC2_SEL (1 << 5) + +/* ES8326_HP_CAL */ +#define ES8326_HPOR_SHIFT 4 + +/* ES8326_ADC1_SRC */ +#define ES8326_ADC1_SHIFT 0 +#define ES8326_ADC2_SHIFT 4 +#define ES8326_ADC_SRC_ANA 0 +#define ES8326_ADC_SRC_ANA_INV_SW0 1 +#define ES8326_ADC_SRC_ANA_INV_SW1 2 +#define ES8326_ADC_SRC_DMIC_MCLK 3 +#define ES8326_ADC_SRC_DMIC_SDIN2 4 +#define ES8326_ADC_SRC_DMIC_SDIN2_INV 5 +#define ES8326_ADC_SRC_DMIC_SDIN3 6 +#define ES8326_ADC_SRC_DMIC_SDIN3_INV 7 + +#define ES8326_ADC_AMIC ((ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC1_SHIFT)) +#define ES8326_ADC_DMIC ((ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC1_SHIFT)) +/* ES8326_ADC2_SRC */ +#define ES8326_ADC3_SHIFT 0 +#define ES8326_ADC4_SHIFT 3 + +/* ES8326_HP_DET */ +#define ES8326_HP_DET_SRC_PIN27 (1 << 5) +#define ES8326_HP_DET_SRC_PIN9 (1 << 4) +#define ES8326_HP_DET_JACK_POL (1 << 3) +#define ES8326_HP_DET_BUTTON_POL (1 << 2) +#define ES8326_HP_TYPE_OMTP (3 << 0) +#define ES8326_HP_TYPE_CTIA (2 << 0) +#define ES8326_HP_TYPE_AUTO (1 << 0) +#define ES8326_HP_TYPE_AUTO_INV (0 << 0) + +/* ES8326_SDINOUT1_IO */ +#define ES8326_IO_INPUT (0 << 0) +#define ES8326_IO_SDIN_SLOT0 (1 << 0) +#define ES8326_IO_SDIN_SLOT1 (2 << 0) +#define ES8326_IO_SDIN_SLOT2 (3 << 0) +#define ES8326_IO_SDIN_SLOT7 (8 << 0) +#define ES8326_IO_DMIC_CLK (9 << 0) +#define ES8326_IO_DMIC_CLK_INV (0x0a << 0) +#define ES8326_IO_SDOUT2 (0x0b << 0) +#define ES8326_IO_LOW (0x0e << 0) +#define ES8326_IO_HIGH (0x0f << 0) +#define ES8326_ADC2DAC (1 << 3) +#define ES8326_SDINOUT1_SHIFT 4 + +/* ES8326_SDINOUT23_IO */ +#define ES8326_SDINOUT2_SHIFT 4 +#define ES8326_SDINOUT3_SHIFT 0 + +/* ES8326_HP_DETECT */ +#define ES8326_HPINSERT_FLAG (1 << 1) +#define ES8326_HPBUTTON_FLAG (1 << 0) + +/* ES8326_CHIP_VERSION 0xFF */ +#define ES8326_VERSION_B (1 << 0) + +#endif diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 5435a49604cf..405ec16be2b6 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -474,6 +474,9 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = { max98088_mic2pre_get, max98088_mic2pre_set, max98088_micboost_tlv), + SOC_SINGLE("Noise Gate Threshold", M98088_REG_40_MICAGC_THRESH, + 4, 15, 0), + SOC_SINGLE("INA Volume", M98088_REG_37_LVL_INA, 0, 7, 1), SOC_SINGLE("INB Volume", M98088_REG_38_LVL_INB, 0, 7, 1), @@ -1746,7 +1749,6 @@ MODULE_DEVICE_TABLE(i2c, max98088_i2c_id); static int max98088_i2c_probe(struct i2c_client *i2c) { struct max98088_priv *max98088; - int ret; const struct i2c_device_id *id; max98088 = devm_kzalloc(&i2c->dev, sizeof(struct max98088_priv), @@ -1769,9 +1771,8 @@ static int max98088_i2c_probe(struct i2c_client *i2c) i2c_set_clientdata(i2c, max98088); max98088->pdata = i2c->dev.platform_data; - ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_max98088, + return devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_max98088, &max98088_dai[0], 2); - return ret; } #if defined(CONFIG_OF) diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index 97b64477dde6..899965b19d12 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -281,6 +281,8 @@ static __maybe_unused int max98373_resume(struct device *dev) msecs_to_jiffies(MAX98373_PROBE_TIMEOUT)); if (!time) { dev_err(dev, "Initialization not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + return -ETIMEDOUT; } diff --git a/sound/soc/codecs/mt6359-accdet.c b/sound/soc/codecs/mt6359-accdet.c index c190628e2905..7f624854948c 100644 --- a/sound/soc/codecs/mt6359-accdet.c +++ b/sound/soc/codecs/mt6359-accdet.c @@ -965,7 +965,7 @@ static int mt6359_accdet_probe(struct platform_device *pdev) mutex_init(&priv->res_lock); priv->accdet_irq = platform_get_irq(pdev, 0); - if (priv->accdet_irq) { + if (priv->accdet_irq >= 0) { ret = devm_request_threaded_irq(&pdev->dev, priv->accdet_irq, NULL, mt6359_accdet_irq, IRQF_TRIGGER_HIGH | IRQF_ONESHOT, @@ -979,7 +979,7 @@ static int mt6359_accdet_probe(struct platform_device *pdev) if (priv->caps & ACCDET_PMIC_EINT0) { priv->accdet_eint0 = platform_get_irq(pdev, 1); - if (priv->accdet_eint0) { + if (priv->accdet_eint0 >= 0) { ret = devm_request_threaded_irq(&pdev->dev, priv->accdet_eint0, NULL, mt6359_accdet_irq, @@ -994,7 +994,7 @@ static int mt6359_accdet_probe(struct platform_device *pdev) } } else if (priv->caps & ACCDET_PMIC_EINT1) { priv->accdet_eint1 = platform_get_irq(pdev, 2); - if (priv->accdet_eint1) { + if (priv->accdet_eint1 >= 0) { ret = devm_request_threaded_irq(&pdev->dev, priv->accdet_eint1, NULL, mt6359_accdet_irq, diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 0be6e72ff5a9..5c29416aa781 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -749,6 +749,8 @@ static int __maybe_unused rt1308_dev_resume(struct device *dev) msecs_to_jiffies(RT1308_PROBE_TIMEOUT)); if (!time) { dev_err(&slave->dev, "Initialization not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + return -ETIMEDOUT; } diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index e53396606a1c..ed0a11436362 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -734,6 +734,8 @@ static int __maybe_unused rt1316_dev_resume(struct device *dev) msecs_to_jiffies(RT1316_PROBE_TIMEOUT)); if (!time) { dev_err(&slave->dev, "Initialization not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + return -ETIMEDOUT; } diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 38ab8d4291c2..5a844329800f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1986,7 +1986,7 @@ static int rt5640_set_bias_level(struct snd_soc_component *component, snd_soc_component_write(component, RT5640_PWR_MIXER, 0x0000); if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) snd_soc_component_write(component, RT5640_PWR_ANLG1, - 0x0018); + 0x2818); else snd_soc_component_write(component, RT5640_PWR_ANLG1, 0x0000); @@ -2600,7 +2600,8 @@ static void rt5640_enable_hda_jack_detect( snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0); snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, - RT5640_PWR_VREF2, RT5640_PWR_VREF2); + RT5640_PWR_VREF2 | RT5640_PWR_MB | RT5640_PWR_BG, + RT5640_PWR_VREF2 | RT5640_PWR_MB | RT5640_PWR_BG); usleep_range(10000, 15000); snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, RT5640_PWR_FV2, RT5640_PWR_FV2); diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index f04e18c32489..c1a94229dc7e 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -793,6 +793,8 @@ static int __maybe_unused rt5682_dev_resume(struct device *dev) msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); if (!time) { dev_err(&slave->dev, "Initialization not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + return -ETIMEDOUT; } diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c index f7439e40ca8b..96fc5f36d0d0 100644 --- a/sound/soc/codecs/rt700-sdw.c +++ b/sound/soc/codecs/rt700-sdw.c @@ -542,6 +542,8 @@ static int __maybe_unused rt700_dev_resume(struct device *dev) msecs_to_jiffies(RT700_PROBE_TIMEOUT)); if (!time) { dev_err(&slave->dev, "Initialization not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + return -ETIMEDOUT; } diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index a085b2f530aa..4120842fe699 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -449,6 +449,8 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) msecs_to_jiffies(RT711_PROBE_TIMEOUT)); if (!time) { dev_err(&slave->dev, "Initialization not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + return -ETIMEDOUT; } diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index 13e731d16675..3f981a9e7fb6 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -244,6 +244,8 @@ static int __maybe_unused rt715_dev_resume(struct device *dev) msecs_to_jiffies(RT715_PROBE_TIMEOUT)); if (!time) { dev_err(&slave->dev, "Enumeration not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + return -ETIMEDOUT; } diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index b047bf87a100..4e61e16470ed 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -562,6 +562,8 @@ static int __maybe_unused rt715_dev_resume(struct device *dev) msecs_to_jiffies(RT715_PROBE_TIMEOUT)); if (!time) { dev_err(&slave->dev, "Initialization not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + return -ETIMEDOUT; } diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index b992216aee55..3047a6fbb380 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -227,13 +227,11 @@ static int sigma_fw_load_control(struct sigmadsp *sigmadsp, if (!ctrl) return -ENOMEM; - name = kzalloc(name_len + 1, GFP_KERNEL); + name = kmemdup_nul(ctrl_chunk->name, name_len, GFP_KERNEL); if (!name) { ret = -ENOMEM; goto err_free_ctrl; } - memcpy(name, ctrl_chunk->name, name_len); - name[name_len] = '\0'; ctrl->name = name; /* diff --git a/sound/soc/codecs/src4xxx-i2c.c b/sound/soc/codecs/src4xxx-i2c.c new file mode 100644 index 000000000000..43daa9dc8ab5 --- /dev/null +++ b/sound/soc/codecs/src4xxx-i2c.c @@ -0,0 +1,47 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Driver for SRC4XXX codecs +// +// Copyright 2021-2022 Deqx Pty Ltd +// Author: Matt Flax <flatmax@flatmax.com> + +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "src4xxx.h" + +static int src4xxx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + return src4xxx_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &src4xxx_regmap_config), NULL); +} + +static const struct i2c_device_id src4xxx_i2c_ids[] = { + { "src4392", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, src4xxx_i2c_ids); + +static const struct of_device_id src4xxx_of_match[] = { + { .compatible = "ti,src4392", }, + { } +}; +MODULE_DEVICE_TABLE(of, src4xxx_of_match); + + +static struct i2c_driver src4xxx_i2c_driver = { + .driver = { + .name = "src4xxx", + .of_match_table = of_match_ptr(src4xxx_of_match), + }, + .probe = src4xxx_i2c_probe, + .id_table = src4xxx_i2c_ids, +}; +module_i2c_driver(src4xxx_i2c_driver); + +MODULE_DESCRIPTION("ASoC SRC4392 CODEC I2C driver"); +MODULE_AUTHOR("Matt Flax <flatmax@flatmax.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/src4xxx.c b/sound/soc/codecs/src4xxx.c new file mode 100644 index 000000000000..db4e280dd055 --- /dev/null +++ b/sound/soc/codecs/src4xxx.c @@ -0,0 +1,518 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TI SRC4xxx Audio Codec driver +// +// Copyright 2021-2022 Deqx Pty Ltd +// Author: Matt Flax <flatmax@flatmax.com> + +#include <linux/module.h> + +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "src4xxx.h" + +struct src4xxx { + struct regmap *regmap; + bool master[2]; + int mclk_hz; + struct device *dev; +}; + +enum {SRC4XXX_PORTA, SRC4XXX_PORTB}; + +/* SRC attenuation */ +static const DECLARE_TLV_DB_SCALE(src_tlv, -12750, 50, 0); + +static const struct snd_kcontrol_new src4xxx_controls[] = { + SOC_DOUBLE_R_TLV("SRC Volume", + SRC4XXX_SCR_CTL_30, SRC4XXX_SCR_CTL_31, 0, 255, 1, src_tlv), +}; + +/* I2S port control */ +static const char * const port_out_src_text[] = { + "loopback", "other_port", "DIR", "SRC" +}; +static SOC_ENUM_SINGLE_DECL(porta_out_src_enum, SRC4XXX_PORTA_CTL_03, 4, + port_out_src_text); +static SOC_ENUM_SINGLE_DECL(portb_out_src_enum, SRC4XXX_PORTB_CTL_05, 4, + port_out_src_text); +static const struct snd_kcontrol_new porta_out_control = + SOC_DAPM_ENUM("Port A source select", porta_out_src_enum); +static const struct snd_kcontrol_new portb_out_control = + SOC_DAPM_ENUM("Port B source select", portb_out_src_enum); + +/* Digital audio transmitter control */ +static const char * const dit_mux_text[] = {"Port A", "Port B", "DIR", "SRC"}; +static SOC_ENUM_SINGLE_DECL(dit_mux_enum, SRC4XXX_TX_CTL_07, 3, dit_mux_text); +static const struct snd_kcontrol_new dit_mux_control = + SOC_DAPM_ENUM("DIT source", dit_mux_enum); + +/* SRC control */ +static const char * const src_in_text[] = {"Port A", "Port B", "DIR"}; +static SOC_ENUM_SINGLE_DECL(src_in_enum, SRC4XXX_SCR_CTL_2D, 0, src_in_text); +static const struct snd_kcontrol_new src_in_control = + SOC_DAPM_ENUM("SRC source select", src_in_enum); + +/* DIR control */ +static const char * const dir_in_text[] = {"Ch 1", "Ch 2", "Ch 3", "Ch 4"}; +static SOC_ENUM_SINGLE_DECL(dir_in_enum, SRC4XXX_RCV_CTL_0D, 0, dir_in_text); +static const struct snd_kcontrol_new dir_in_control = + SOC_DAPM_ENUM("Digital Input", dir_in_enum); + +static const struct snd_soc_dapm_widget src4xxx_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("loopback_A"), + SND_SOC_DAPM_INPUT("other_port_A"), + SND_SOC_DAPM_INPUT("DIR_A"), + SND_SOC_DAPM_INPUT("SRC_A"), + SND_SOC_DAPM_MUX("Port A source", + SND_SOC_NOPM, 0, 0, &porta_out_control), + + SND_SOC_DAPM_INPUT("loopback_B"), + SND_SOC_DAPM_INPUT("other_port_B"), + SND_SOC_DAPM_INPUT("DIR_B"), + SND_SOC_DAPM_INPUT("SRC_B"), + SND_SOC_DAPM_MUX("Port B source", + SND_SOC_NOPM, 0, 0, &portb_out_control), + + SND_SOC_DAPM_INPUT("Port_A"), + SND_SOC_DAPM_INPUT("Port_B"), + SND_SOC_DAPM_INPUT("DIR_"), + + /* Digital audio receivers and transmitters */ + SND_SOC_DAPM_OUTPUT("DIR_OUT"), + SND_SOC_DAPM_OUTPUT("SRC_OUT"), + SND_SOC_DAPM_MUX("DIT Out Src", SRC4XXX_PWR_RST_01, + SRC4XXX_ENABLE_DIT_SHIFT, 1, &dit_mux_control), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF_A_RX", "Playback A", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_A_SHIFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF_A_TX", "Capture A", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_A_SHIFT, 1), + SND_SOC_DAPM_AIF_IN("AIF_B_RX", "Playback B", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_B_SHIFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF_B_TX", "Capture B", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_B_SHIFT, 1), + + SND_SOC_DAPM_MUX("SRC source", SND_SOC_NOPM, 0, 0, &src_in_control), + + SND_SOC_DAPM_INPUT("MCLK"), + SND_SOC_DAPM_INPUT("RXMCLKI"), + SND_SOC_DAPM_INPUT("RXMCLKO"), + + SND_SOC_DAPM_INPUT("RX1"), + SND_SOC_DAPM_INPUT("RX2"), + SND_SOC_DAPM_INPUT("RX3"), + SND_SOC_DAPM_INPUT("RX4"), + SND_SOC_DAPM_MUX("Digital Input", SRC4XXX_PWR_RST_01, + SRC4XXX_ENABLE_DIR_SHIFT, 1, &dir_in_control), +}; + +static const struct snd_soc_dapm_route src4xxx_audio_routes[] = { + /* I2S Input to Output Routing */ + {"Port A source", "loopback", "loopback_A"}, + {"Port A source", "other_port", "other_port_A"}, + {"Port A source", "DIR", "DIR_A"}, + {"Port A source", "SRC", "SRC_A"}, + {"Port B source", "loopback", "loopback_B"}, + {"Port B source", "other_port", "other_port_B"}, + {"Port B source", "DIR", "DIR_B"}, + {"Port B source", "SRC", "SRC_B"}, + /* DIT muxing */ + {"DIT Out Src", "Port A", "Capture A"}, + {"DIT Out Src", "Port B", "Capture B"}, + {"DIT Out Src", "DIR", "DIR_OUT"}, + {"DIT Out Src", "SRC", "SRC_OUT"}, + + /* SRC input selection */ + {"SRC source", "Port A", "Port_A"}, + {"SRC source", "Port B", "Port_B"}, + {"SRC source", "DIR", "DIR_"}, + /* SRC mclk selection */ + {"SRC mclk source", "Master (MCLK)", "MCLK"}, + {"SRC mclk source", "Master (RXCLKI)", "RXMCLKI"}, + {"SRC mclk source", "Recovered receiver clk", "RXMCLKO"}, + /* DIR input selection */ + {"Digital Input", "Ch 1", "RX1"}, + {"Digital Input", "Ch 2", "RX2"}, + {"Digital Input", "Ch 3", "RX3"}, + {"Digital Input", "Ch 4", "RX4"}, +}; + + +static const struct snd_soc_component_driver src4xxx_driver = { + .controls = src4xxx_controls, + .num_controls = ARRAY_SIZE(src4xxx_controls), + + .dapm_widgets = src4xxx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(src4xxx_dapm_widgets), + .dapm_routes = src4xxx_audio_routes, + .num_dapm_routes = ARRAY_SIZE(src4xxx_audio_routes), +}; + +static int src4xxx_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + unsigned int ctrl; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl = SRC4XXX_BUS_MASTER; + src4xxx->master[dai->id] = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl = 0; + src4xxx->master[dai->id] = false; + break; + default: + return -EINVAL; + break; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl |= SRC4XXX_BUS_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl |= SRC4XXX_BUS_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl |= SRC4XXX_BUS_RIGHT_J_24; + break; + default: + return -EINVAL; + break; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + break; + } + + regmap_update_bits(src4xxx->regmap, SRC4XXX_BUS_FMT(dai->id), + SRC4XXX_BUS_FMT_MS_MASK, ctrl); + + return 0; +} + +static int src4xxx_set_mclk_hz(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + + dev_info(component->dev, "changing mclk rate from %d to %d Hz\n", + src4xxx->mclk_hz, freq); + src4xxx->mclk_hz = freq; + + return 0; +} + +static int src4xxx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + unsigned int mclk_div; + int val, pj, jd, d; + int reg; + int ret; + + switch (dai->id) { + case SRC4XXX_PORTB: + reg = SRC4XXX_PORTB_CTL_06; + break; + default: + reg = SRC4XXX_PORTA_CTL_04; + break; + } + + if (src4xxx->master[dai->id]) { + mclk_div = src4xxx->mclk_hz/params_rate(params); + if (src4xxx->mclk_hz != mclk_div*params_rate(params)) { + dev_err(component->dev, + "mclk %d / rate %d has a remainder.\n", + src4xxx->mclk_hz, params_rate(params)); + return -EINVAL; + } + + val = ((int)mclk_div - 128) / 128; + if ((val < 0) | (val > 3)) { + dev_err(component->dev, + "div register setting %d is out of range\n", + val); + dev_err(component->dev, + "unsupported sample rate %d Hz for the master clock of %d Hz\n", + params_rate(params), src4xxx->mclk_hz); + return -EINVAL; + } + + /* set the TX DIV */ + ret = regmap_update_bits(src4xxx->regmap, + SRC4XXX_TX_CTL_07, SRC4XXX_TX_MCLK_DIV_MASK, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + if (ret) { + dev_err(component->dev, + "Couldn't set the TX's div register to %d << %d = 0x%x\n", + val, SRC4XXX_TX_MCLK_DIV_SHIFT, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + return ret; + } + + /* set the PLL for the digital receiver */ + switch (src4xxx->mclk_hz) { + case 24576000: + pj = 0x22; + jd = 0x00; + d = 0x00; + break; + case 22579200: + pj = 0x22; + jd = 0x1b; + d = 0xa3; + break; + default: + /* don't error out here, + * other parts of the chip are still functional + * Dummy initialize variables to avoid + * -Wsometimes-uninitialized from clang. + */ + dev_info(component->dev, + "Couldn't set the RCV PLL as this master clock rate is unknown. Chosen regmap values may not match real world values.\n"); + pj = 0x0; + jd = 0xff; + d = 0xff; + break; + } + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_0F); + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_10, jd); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_10); + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_11, d); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_11); + + ret = regmap_update_bits(src4xxx->regmap, + SRC4XXX_TX_CTL_07, SRC4XXX_TX_MCLK_DIV_MASK, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + if (ret < 0) { + dev_err(component->dev, + "Couldn't set the TX's div register to %d << %d = 0x%x\n", + val, SRC4XXX_TX_MCLK_DIV_SHIFT, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + return ret; + } + + return regmap_update_bits(src4xxx->regmap, reg, + SRC4XXX_MCLK_DIV_MASK, val); + } else { + dev_info(dai->dev, "not setting up MCLK as not master\n"); + } + + return 0; +}; + +static const struct snd_soc_dai_ops src4xxx_dai_ops = { + .hw_params = src4xxx_hw_params, + .set_sysclk = src4xxx_set_mclk_hz, + .set_fmt = src4xxx_set_dai_fmt, +}; + +#define SRC4XXX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) +#define SRC4XXX_RATES (SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000|\ + SNDRV_PCM_RATE_88200|\ + SNDRV_PCM_RATE_96000|\ + SNDRV_PCM_RATE_176400|\ + SNDRV_PCM_RATE_192000) + +static struct snd_soc_dai_driver src4xxx_dai_driver[] = { + { + .id = SRC4XXX_PORTA, + .name = "src4xxx-portA", + .playback = { + .stream_name = "Playback A", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .capture = { + .stream_name = "Capture A", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .ops = &src4xxx_dai_ops, + }, + { + .id = SRC4XXX_PORTB, + .name = "src4xxx-portB", + .playback = { + .stream_name = "Playback B", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .capture = { + .stream_name = "Capture B", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .ops = &src4xxx_dai_ops, + }, +}; + +static const struct reg_default src4xxx_reg_defaults[] = { + { SRC4XXX_PWR_RST_01, 0x00 }, /* all powered down intially */ + { SRC4XXX_PORTA_CTL_03, 0x00 }, + { SRC4XXX_PORTA_CTL_04, 0x00 }, + { SRC4XXX_PORTB_CTL_05, 0x00 }, + { SRC4XXX_PORTB_CTL_06, 0x00 }, + { SRC4XXX_TX_CTL_07, 0x00 }, + { SRC4XXX_TX_CTL_08, 0x00 }, + { SRC4XXX_TX_CTL_09, 0x00 }, + { SRC4XXX_SRC_DIT_IRQ_MSK_0B, 0x00 }, + { SRC4XXX_SRC_DIT_IRQ_MODE_0C, 0x00 }, + { SRC4XXX_RCV_CTL_0D, 0x00 }, + { SRC4XXX_RCV_CTL_0E, 0x00 }, + { SRC4XXX_RCV_PLL_0F, 0x00 }, /* not spec. in the datasheet */ + { SRC4XXX_RCV_PLL_10, 0xff }, /* not spec. in the datasheet */ + { SRC4XXX_RCV_PLL_11, 0xff }, /* not spec. in the datasheet */ + { SRC4XXX_RVC_IRQ_MSK_16, 0x00 }, + { SRC4XXX_RVC_IRQ_MSK_17, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_18, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_19, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_1A, 0x00 }, + { SRC4XXX_GPIO_1_1B, 0x00 }, + { SRC4XXX_GPIO_2_1C, 0x00 }, + { SRC4XXX_GPIO_3_1D, 0x00 }, + { SRC4XXX_GPIO_4_1E, 0x00 }, + { SRC4XXX_SCR_CTL_2D, 0x00 }, + { SRC4XXX_SCR_CTL_2E, 0x00 }, + { SRC4XXX_SCR_CTL_2F, 0x00 }, + { SRC4XXX_SCR_CTL_30, 0x00 }, + { SRC4XXX_SCR_CTL_31, 0x00 }, +}; + +int src4xxx_probe(struct device *dev, struct regmap *regmap, + void (*switch_mode)(struct device *dev)) +{ + struct src4xxx *src4xxx; + int ret; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + src4xxx = devm_kzalloc(dev, sizeof(*src4xxx), GFP_KERNEL); + if (!src4xxx) + return -ENOMEM; + + src4xxx->regmap = regmap; + src4xxx->dev = dev; + src4xxx->mclk_hz = 0; /* mclk has not been configured yet */ + dev_set_drvdata(dev, src4xxx); + + ret = regmap_write(regmap, SRC4XXX_PWR_RST_01, SRC4XXX_RESET); + if (ret < 0) + dev_err(dev, "Failed to issue reset: %d\n", ret); + usleep_range(1, 500); /* sleep for more then 500 ns */ + ret = regmap_write(regmap, SRC4XXX_PWR_RST_01, SRC4XXX_POWER_DOWN); + if (ret < 0) + dev_err(dev, "Failed to decommission reset: %d\n", ret); + usleep_range(500, 1000); /* sleep for 500 us or more */ + + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_PWR_RST_01, + SRC4XXX_POWER_ENABLE, SRC4XXX_POWER_ENABLE); + if (ret < 0) + dev_err(dev, "Failed to port A and B : %d\n", ret); + + /* set receiver to use master clock (rcv mclk is most likely jittery) */ + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_RCV_CTL_0D, + SRC4XXX_RXCLK_MCLK, SRC4XXX_RXCLK_MCLK); + if (ret < 0) + dev_err(dev, + "Failed to enable mclk as the PLL1 DIR reference : %d\n", ret); + + /* default to leaving the PLL2 running on loss of lock, divide by 8 */ + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_RCV_CTL_0E, + SRC4XXX_PLL2_DIV_8 | SRC4XXX_REC_MCLK_EN | SRC4XXX_PLL2_LOL, + SRC4XXX_PLL2_DIV_8 | SRC4XXX_REC_MCLK_EN | SRC4XXX_PLL2_LOL); + if (ret < 0) + dev_err(dev, "Failed to enable mclk rec and div : %d\n", ret); + + ret = devm_snd_soc_register_component(dev, &src4xxx_driver, + src4xxx_dai_driver, ARRAY_SIZE(src4xxx_dai_driver)); + if (ret == 0) + dev_info(dev, "src4392 probe ok %d\n", ret); + return ret; +} +EXPORT_SYMBOL_GPL(src4xxx_probe); + +static bool src4xxx_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SRC4XXX_RES_00: + case SRC4XXX_GLOBAL_ITR_STS_02: + case SRC4XXX_SRC_DIT_STS_0A: + case SRC4XXX_NON_AUDIO_D_12: + case SRC4XXX_RVC_STS_13: + case SRC4XXX_RVC_STS_14: + case SRC4XXX_RVC_STS_15: + case SRC4XXX_SUB_CODE_1F: + case SRC4XXX_SUB_CODE_20: + case SRC4XXX_SUB_CODE_21: + case SRC4XXX_SUB_CODE_22: + case SRC4XXX_SUB_CODE_23: + case SRC4XXX_SUB_CODE_24: + case SRC4XXX_SUB_CODE_25: + case SRC4XXX_SUB_CODE_26: + case SRC4XXX_SUB_CODE_27: + case SRC4XXX_SUB_CODE_28: + case SRC4XXX_PC_PREAMBLE_HI_29: + case SRC4XXX_PC_PREAMBLE_LO_2A: + case SRC4XXX_PD_PREAMBLE_HI_2B: + case SRC4XXX_PC_PREAMBLE_LO_2C: + case SRC4XXX_IO_RATIO_32: + case SRC4XXX_IO_RATIO_33: + return true; + } + + if (reg > SRC4XXX_IO_RATIO_33 && reg < SRC4XXX_PAGE_SEL_7F) + return true; + + return false; +} + +const struct regmap_config src4xxx_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + .max_register = SRC4XXX_IO_RATIO_33, + + .reg_defaults = src4xxx_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(src4xxx_reg_defaults), + .volatile_reg = src4xxx_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(src4xxx_regmap_config); + +MODULE_DESCRIPTION("ASoC SRC4XXX CODEC driver"); +MODULE_AUTHOR("Matt Flax <flatmax@flatmax.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/src4xxx.h b/sound/soc/codecs/src4xxx.h new file mode 100644 index 000000000000..5bf778fb9945 --- /dev/null +++ b/sound/soc/codecs/src4xxx.h @@ -0,0 +1,113 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// src4xxx.h -- SRC4XXX ALSA SoC audio driver +// +// Copyright 2021-2022 Deqx Pty Ltd +// Author: Matt R Flax <flatmax@flatmax.com> + +#ifndef __SRC4XXX_H__ +#define __SRC4XXX_H__ + +#define SRC4XXX_RES_00 0x00 +#define SRC4XXX_PWR_RST_01 0x01 +#define SRC4XXX_RESET 0x80 +#define SRC4XXX_POWER_DOWN 0x00 +#define SRC4XXX_POWER_ENABLE 0x20 +#define SRC4XXX_ENABLE_SRC 0x1 +#define SRC4XXX_ENABLE_SRC_SHIFT 0 +#define SRC4XXX_ENABLE_DIR 0x2 +#define SRC4XXX_ENABLE_DIR_SHIFT 1 +#define SRC4XXX_ENABLE_DIT 0x4 +#define SRC4XXX_ENABLE_DIT_SHIFT 2 +#define SRC4XXX_ENABLE_PORT_B 0x8 +#define SRC4XXX_ENABLE_PORT_B_SHIFT 3 +#define SRC4XXX_ENABLE_PORT_A 0x10 +#define SRC4XXX_ENABLE_PORT_A_SHIFT 4 + +#define SRC4XXX_PORTA_CTL_03 0x03 +#define SRC4XXX_BUS_MASTER 0x8 +#define SRC4XXX_BUS_LEFT_J 0x0 +#define SRC4XXX_BUS_I2S 0x1 +#define SRC4XXX_BUS_RIGHT_J_16 0x4 +#define SRC4XXX_BUS_RIGHT_J_18 0x5 +#define SRC4XXX_BUS_RIGHT_J_20 0x6 +#define SRC4XXX_BUS_RIGHT_J_24 0x7 +#define SRC4XXX_BUS_FMT_MS_MASK 0xf + +#define SRC4XXX_PORTA_CTL_04 0x04 +#define SRC4XXX_MCLK_DIV_MASK 0x3 + +#define SRC4XXX_BUS_FMT(id) (SRC4XXX_PORTA_CTL_03+2*id) +#define SRC4XXX_BUS_CLK(id) (SRC4XXX_PORTA_CTL_04+2*id) + +#define SRC4XXX_PORTB_CTL_05 0x05 +#define SRC4XXX_PORTB_CTL_06 0x06 + +#define SRC4XXX_TX_CTL_07 0x07 +#define SRC4XXX_TX_MCLK_DIV_MASK 0x60 +#define SRC4XXX_TX_MCLK_DIV_SHIFT 5 + +#define SRC4XXX_TX_CTL_08 0x08 +#define SRC4XXX_TX_CTL_09 0x09 +#define SRC4XXX_SRC_DIT_IRQ_MSK_0B 0x0B +#define SRC4XXX_SRC_BTI_EN 0x01 +#define SRC4XXX_SRC_TSLIP_EN 0x02 +#define SRC4XXX_SRC_DIT_IRQ_MODE_0C 0x0C +#define SRC4XXX_RCV_CTL_0D 0x0D +#define SRC4XXX_RXCLK_RXCKI 0x0 +#define SRC4XXX_RXCLK_MCLK 0x8 +#define SRC4XXX_RCV_CTL_0E 0x0E +#define SRC4XXX_REC_MCLK_EN 0x1 +#define SRC4XXX_PLL2_DIV_0 (0x0<<1) +#define SRC4XXX_PLL2_DIV_2 (0x1<<1) +#define SRC4XXX_PLL2_DIV_4 (0x2<<1) +#define SRC4XXX_PLL2_DIV_8 (0x3<<1) +#define SRC4XXX_PLL2_LOL 0x8 +#define SRC4XXX_RCV_PLL_0F 0x0F +#define SRC4XXX_RCV_PLL_10 0x10 +#define SRC4XXX_RCV_PLL_11 0x11 +#define SRC4XXX_RVC_IRQ_MSK_16 0x16 +#define SRC4XXX_RVC_IRQ_MSK_17 0x17 +#define SRC4XXX_RVC_IRQ_MODE_18 0x18 +#define SRC4XXX_RVC_IRQ_MODE_19 0x19 +#define SRC4XXX_RVC_IRQ_MODE_1A 0x1A +#define SRC4XXX_GPIO_1_1B 0x1B +#define SRC4XXX_GPIO_2_1C 0x1C +#define SRC4XXX_GPIO_3_1D 0x1D +#define SRC4XXX_GPIO_4_1E 0x1E +#define SRC4XXX_SCR_CTL_2D 0x2D +#define SRC4XXX_SCR_CTL_2E 0x2E +#define SRC4XXX_SCR_CTL_2F 0x2F +#define SRC4XXX_SCR_CTL_30 0x30 +#define SRC4XXX_SCR_CTL_31 0x31 +#define SRC4XXX_PAGE_SEL_7F 0x7F + +// read only registers +#define SRC4XXX_GLOBAL_ITR_STS_02 0x02 +#define SRC4XXX_SRC_DIT_STS_0A 0x0A +#define SRC4XXX_NON_AUDIO_D_12 0x12 +#define SRC4XXX_RVC_STS_13 0x13 +#define SRC4XXX_RVC_STS_14 0x14 +#define SRC4XXX_RVC_STS_15 0x15 +#define SRC4XXX_SUB_CODE_1F 0x1F +#define SRC4XXX_SUB_CODE_20 0x20 +#define SRC4XXX_SUB_CODE_21 0x21 +#define SRC4XXX_SUB_CODE_22 0x22 +#define SRC4XXX_SUB_CODE_23 0x23 +#define SRC4XXX_SUB_CODE_24 0x24 +#define SRC4XXX_SUB_CODE_25 0x25 +#define SRC4XXX_SUB_CODE_26 0x26 +#define SRC4XXX_SUB_CODE_27 0x27 +#define SRC4XXX_SUB_CODE_28 0x28 +#define SRC4XXX_PC_PREAMBLE_HI_29 0x29 +#define SRC4XXX_PC_PREAMBLE_LO_2A 0x2A +#define SRC4XXX_PD_PREAMBLE_HI_2B 0x2B +#define SRC4XXX_PC_PREAMBLE_LO_2C 0x2C +#define SRC4XXX_IO_RATIO_32 0x32 +#define SRC4XXX_IO_RATIO_33 0x33 + +int src4xxx_probe(struct device *dev, struct regmap *regmap, + void (*switch_mode)(struct device *dev)); +extern const struct regmap_config src4xxx_regmap_config; + +#endif /* __SRC4XXX_H__ */ diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index 846d9d3ecc9d..51b87a936179 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -31,11 +31,66 @@ struct tas2764_priv { struct gpio_desc *sdz_gpio; struct regmap *regmap; struct device *dev; + int irq; int v_sense_slot; int i_sense_slot; + + bool dac_powered; + bool unmuted; +}; + +static const char *tas2764_int_ltch0_msgs[8] = { + "fault: over temperature", /* INT_LTCH0 & BIT(0) */ + "fault: over current", + "fault: bad TDM clock", + "limiter active", + "fault: PVDD below limiter inflection point", + "fault: limiter max attenuation", + "fault: BOP infinite hold", + "fault: BOP mute", /* INT_LTCH0 & BIT(7) */ +}; + +static const unsigned int tas2764_int_readout_regs[6] = { + TAS2764_INT_LTCH0, + TAS2764_INT_LTCH1, + TAS2764_INT_LTCH1_0, + TAS2764_INT_LTCH2, + TAS2764_INT_LTCH3, + TAS2764_INT_LTCH4, }; +static irqreturn_t tas2764_irq(int irq, void *data) +{ + struct tas2764_priv *tas2764 = data; + u8 latched[6] = {0, 0, 0, 0, 0, 0}; + int ret = IRQ_NONE; + int i; + + for (i = 0; i < ARRAY_SIZE(latched); i++) + latched[i] = snd_soc_component_read(tas2764->component, + tas2764_int_readout_regs[i]); + + for (i = 0; i < 8; i++) { + if (latched[0] & BIT(i)) { + dev_crit_ratelimited(tas2764->dev, "%s\n", + tas2764_int_ltch0_msgs[i]); + ret = IRQ_HANDLED; + } + } + + if (latched[0]) { + dev_err_ratelimited(tas2764->dev, "other context to the fault: %02x,%02x,%02x,%02x,%02x", + latched[1], latched[2], latched[3], latched[4], latched[5]); + snd_soc_component_update_bits(tas2764->component, + TAS2764_INT_CLK_CFG, + TAS2764_INT_CLK_CFG_IRQZ_CLR, + TAS2764_INT_CLK_CFG_IRQZ_CLR); + } + + return ret; +} + static void tas2764_reset(struct tas2764_priv *tas2764) { if (tas2764->reset_gpio) { @@ -50,34 +105,22 @@ static void tas2764_reset(struct tas2764_priv *tas2764) usleep_range(1000, 2000); } -static int tas2764_set_bias_level(struct snd_soc_component *component, - enum snd_soc_bias_level level) +static int tas2764_update_pwr_ctrl(struct tas2764_priv *tas2764) { - struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component); + struct snd_soc_component *component = tas2764->component; + unsigned int val; + int ret; - switch (level) { - case SND_SOC_BIAS_ON: - snd_soc_component_update_bits(component, TAS2764_PWR_CTRL, - TAS2764_PWR_CTRL_MASK, - TAS2764_PWR_CTRL_ACTIVE); - break; - case SND_SOC_BIAS_STANDBY: - case SND_SOC_BIAS_PREPARE: - snd_soc_component_update_bits(component, TAS2764_PWR_CTRL, - TAS2764_PWR_CTRL_MASK, - TAS2764_PWR_CTRL_MUTE); - break; - case SND_SOC_BIAS_OFF: - snd_soc_component_update_bits(component, TAS2764_PWR_CTRL, - TAS2764_PWR_CTRL_MASK, - TAS2764_PWR_CTRL_SHUTDOWN); - break; + if (tas2764->dac_powered) + val = tas2764->unmuted ? + TAS2764_PWR_CTRL_ACTIVE : TAS2764_PWR_CTRL_MUTE; + else + val = TAS2764_PWR_CTRL_SHUTDOWN; - default: - dev_err(tas2764->dev, - "wrong power level setting %d\n", level); - return -EINVAL; - } + ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL, + TAS2764_PWR_CTRL_MASK, val); + if (ret < 0) + return ret; return 0; } @@ -114,9 +157,7 @@ static int tas2764_codec_resume(struct snd_soc_component *component) usleep_range(1000, 2000); } - ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL, - TAS2764_PWR_CTRL_MASK, - TAS2764_PWR_CTRL_ACTIVE); + ret = tas2764_update_pwr_ctrl(tas2764); if (ret < 0) return ret; @@ -150,14 +191,12 @@ static int tas2764_dac_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: - ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL, - TAS2764_PWR_CTRL_MASK, - TAS2764_PWR_CTRL_MUTE); + tas2764->dac_powered = true; + ret = tas2764_update_pwr_ctrl(tas2764); break; case SND_SOC_DAPM_PRE_PMD: - ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL, - TAS2764_PWR_CTRL_MASK, - TAS2764_PWR_CTRL_SHUTDOWN); + tas2764->dac_powered = false; + ret = tas2764_update_pwr_ctrl(tas2764); break; default: dev_err(tas2764->dev, "Unsupported event\n"); @@ -202,17 +241,11 @@ static const struct snd_soc_dapm_route tas2764_audio_map[] = { static int tas2764_mute(struct snd_soc_dai *dai, int mute, int direction) { - struct snd_soc_component *component = dai->component; - int ret; - - ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL, - TAS2764_PWR_CTRL_MASK, - mute ? TAS2764_PWR_CTRL_MUTE : 0); + struct tas2764_priv *tas2764 = + snd_soc_component_get_drvdata(dai->component); - if (ret < 0) - return ret; - - return 0; + tas2764->unmuted = !mute; + return tas2764_update_pwr_ctrl(tas2764); } static int tas2764_set_bitwidth(struct tas2764_priv *tas2764, int bitwidth) @@ -485,7 +518,7 @@ static struct snd_soc_dai_driver tas2764_dai_driver[] = { .id = 0, .playback = { .stream_name = "ASI1 Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = TAS2764_RATES, .formats = TAS2764_FORMATS, @@ -516,6 +549,34 @@ static int tas2764_codec_probe(struct snd_soc_component *component) tas2764_reset(tas2764); + if (tas2764->irq) { + ret = snd_soc_component_write(tas2764->component, TAS2764_INT_MASK0, 0xff); + if (ret < 0) + return ret; + + ret = snd_soc_component_write(tas2764->component, TAS2764_INT_MASK1, 0xff); + if (ret < 0) + return ret; + + ret = snd_soc_component_write(tas2764->component, TAS2764_INT_MASK2, 0xff); + if (ret < 0) + return ret; + + ret = snd_soc_component_write(tas2764->component, TAS2764_INT_MASK3, 0xff); + if (ret < 0) + return ret; + + ret = snd_soc_component_write(tas2764->component, TAS2764_INT_MASK4, 0xff); + if (ret < 0) + return ret; + + ret = devm_request_threaded_irq(tas2764->dev, tas2764->irq, NULL, tas2764_irq, + IRQF_ONESHOT | IRQF_SHARED | IRQF_TRIGGER_LOW, + "tas2764", tas2764); + if (ret) + dev_warn(tas2764->dev, "failed to request IRQ: %d\n", ret); + } + ret = snd_soc_component_update_bits(tas2764->component, TAS2764_TDM_CFG5, TAS2764_TDM_CFG5_VSNS_ENABLE, 0); if (ret < 0) @@ -526,30 +587,33 @@ static int tas2764_codec_probe(struct snd_soc_component *component) if (ret < 0) return ret; - ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL, - TAS2764_PWR_CTRL_MASK, - TAS2764_PWR_CTRL_MUTE); - if (ret < 0) - return ret; - return 0; } static DECLARE_TLV_DB_SCALE(tas2764_digital_tlv, 1100, 50, 0); static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10050, 50, 1); +static const char * const tas2764_hpf_texts[] = { + "Disabled", "2 Hz", "50 Hz", "100 Hz", "200 Hz", + "400 Hz", "800 Hz" +}; + +static SOC_ENUM_SINGLE_DECL( + tas2764_hpf_enum, TAS2764_DC_BLK0, + TAS2764_DC_BLK0_HPF_FREQ_PB_SHIFT, tas2764_hpf_texts); + static const struct snd_kcontrol_new tas2764_snd_controls[] = { SOC_SINGLE_TLV("Speaker Volume", TAS2764_DVC, 0, TAS2764_DVC_MAX, 1, tas2764_playback_volume), SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 1, 0x14, 0, tas2764_digital_tlv), + SOC_ENUM("HPF Corner Frequency", tas2764_hpf_enum), }; static const struct snd_soc_component_driver soc_component_driver_tas2764 = { .probe = tas2764_codec_probe, .suspend = tas2764_codec_suspend, .resume = tas2764_codec_resume, - .set_bias_level = tas2764_set_bias_level, .controls = tas2764_snd_controls, .num_controls = ARRAY_SIZE(tas2764_snd_controls), .dapm_widgets = tas2764_dapm_widgets, @@ -585,9 +649,21 @@ static const struct regmap_range_cfg tas2764_regmap_ranges[] = { }, }; +static bool tas2764_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS2764_INT_LTCH0 ... TAS2764_INT_LTCH4: + case TAS2764_INT_CLK_CFG: + return true; + default: + return false; + } +} + static const struct regmap_config tas2764_i2c_regmap = { .reg_bits = 8, .val_bits = 8, + .volatile_reg = tas2764_volatile_register, .reg_defaults = tas2764_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas2764_reg_defaults), .cache_type = REGCACHE_RBTREE, @@ -641,6 +717,7 @@ static int tas2764_i2c_probe(struct i2c_client *client) return -ENOMEM; tas2764->dev = &client->dev; + tas2764->irq = client->irq; i2c_set_clientdata(client, tas2764); dev_set_drvdata(&client->dev, tas2764); diff --git a/sound/soc/codecs/tas2764.h b/sound/soc/codecs/tas2764.h index f015f22a083b..168af772a898 100644 --- a/sound/soc/codecs/tas2764.h +++ b/sound/soc/codecs/tas2764.h @@ -33,6 +33,10 @@ #define TAS2764_VSENSE_POWER_EN 3 #define TAS2764_ISENSE_POWER_EN 4 +/* DC Blocker Control */ +#define TAS2764_DC_BLK0 TAS2764_REG(0x0, 0x04) +#define TAS2764_DC_BLK0_HPF_FREQ_PB_SHIFT 0 + /* Digital Volume Control */ #define TAS2764_DVC TAS2764_REG(0X0, 0x1a) #define TAS2764_DVC_MAX 0xc9 @@ -87,4 +91,23 @@ #define TAS2764_TDM_CFG6_ISNS_ENABLE BIT(6) #define TAS2764_TDM_CFG6_50_MASK GENMASK(5, 0) +/* Interrupt Masks */ +#define TAS2764_INT_MASK0 TAS2764_REG(0x0, 0x3b) +#define TAS2764_INT_MASK1 TAS2764_REG(0x0, 0x3c) +#define TAS2764_INT_MASK2 TAS2764_REG(0x0, 0x40) +#define TAS2764_INT_MASK3 TAS2764_REG(0x0, 0x41) +#define TAS2764_INT_MASK4 TAS2764_REG(0x0, 0x3d) + +/* Latched Fault Registers */ +#define TAS2764_INT_LTCH0 TAS2764_REG(0x0, 0x49) +#define TAS2764_INT_LTCH1 TAS2764_REG(0x0, 0x4a) +#define TAS2764_INT_LTCH1_0 TAS2764_REG(0x0, 0x4b) +#define TAS2764_INT_LTCH2 TAS2764_REG(0x0, 0x4f) +#define TAS2764_INT_LTCH3 TAS2764_REG(0x0, 0x50) +#define TAS2764_INT_LTCH4 TAS2764_REG(0x0, 0x51) + +/* Clock/IRQ Settings */ +#define TAS2764_INT_CLK_CFG TAS2764_REG(0x0, 0x5c) +#define TAS2764_INT_CLK_CFG_IRQZ_CLR BIT(2) + #endif /* __TAS2764__ */ diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 3cb634c28261..bb653b664146 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -46,34 +46,22 @@ static void tas2770_reset(struct tas2770_priv *tas2770) usleep_range(1000, 2000); } -static int tas2770_set_bias_level(struct snd_soc_component *component, - enum snd_soc_bias_level level) +static int tas2770_update_pwr_ctrl(struct tas2770_priv *tas2770) { - struct tas2770_priv *tas2770 = - snd_soc_component_get_drvdata(component); + struct snd_soc_component *component = tas2770->component; + unsigned int val; + int ret; - switch (level) { - case SND_SOC_BIAS_ON: - snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, - TAS2770_PWR_CTRL_MASK, - TAS2770_PWR_CTRL_ACTIVE); - break; - case SND_SOC_BIAS_STANDBY: - case SND_SOC_BIAS_PREPARE: - snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, - TAS2770_PWR_CTRL_MASK, - TAS2770_PWR_CTRL_MUTE); - break; - case SND_SOC_BIAS_OFF: - snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, - TAS2770_PWR_CTRL_MASK, - TAS2770_PWR_CTRL_SHUTDOWN); - break; + if (tas2770->dac_powered) + val = tas2770->unmuted ? + TAS2770_PWR_CTRL_ACTIVE : TAS2770_PWR_CTRL_MUTE; + else + val = TAS2770_PWR_CTRL_SHUTDOWN; - default: - dev_err(tas2770->dev, "wrong power level setting %d\n", level); - return -EINVAL; - } + ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, val); + if (ret < 0) + return ret; return 0; } @@ -114,9 +102,7 @@ static int tas2770_codec_resume(struct snd_soc_component *component) gpiod_set_value_cansleep(tas2770->sdz_gpio, 1); usleep_range(1000, 2000); } else { - ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, - TAS2770_PWR_CTRL_MASK, - TAS2770_PWR_CTRL_ACTIVE); + ret = tas2770_update_pwr_ctrl(tas2770); if (ret < 0) return ret; } @@ -152,24 +138,19 @@ static int tas2770_dac_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: - ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, - TAS2770_PWR_CTRL_MASK, - TAS2770_PWR_CTRL_MUTE); + tas2770->dac_powered = 1; + ret = tas2770_update_pwr_ctrl(tas2770); break; case SND_SOC_DAPM_PRE_PMD: - ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, - TAS2770_PWR_CTRL_MASK, - TAS2770_PWR_CTRL_SHUTDOWN); + tas2770->dac_powered = 0; + ret = tas2770_update_pwr_ctrl(tas2770); break; default: dev_err(tas2770->dev, "Not supported evevt\n"); return -EINVAL; } - if (ret < 0) - return ret; - - return 0; + return ret; } static const struct snd_kcontrol_new isense_switch = @@ -203,21 +184,11 @@ static const struct snd_soc_dapm_route tas2770_audio_map[] = { static int tas2770_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; - int ret; - - if (mute) - ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, - TAS2770_PWR_CTRL_MASK, - TAS2770_PWR_CTRL_MUTE); - else - ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, - TAS2770_PWR_CTRL_MASK, - TAS2770_PWR_CTRL_ACTIVE); - - if (ret < 0) - return ret; + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); - return 0; + tas2770->unmuted = !mute; + return tas2770_update_pwr_ctrl(tas2770); } static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) @@ -337,7 +308,7 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct snd_soc_component *component = dai->component; struct tas2770_priv *tas2770 = snd_soc_component_get_drvdata(component); - u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; + u8 tdm_rx_start_slot = 0, invert_fpol = 0, fpol_preinv = 0, asi_cfg_1 = 0; int ret; switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { @@ -349,9 +320,15 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + invert_fpol = 1; + fallthrough; case SND_SOC_DAIFMT_NB_NF: asi_cfg_1 |= TAS2770_TDM_CFG_REG1_RX_RSING; break; + case SND_SOC_DAIFMT_IB_IF: + invert_fpol = 1; + fallthrough; case SND_SOC_DAIFMT_IB_NF: asi_cfg_1 |= TAS2770_TDM_CFG_REG1_RX_FALING; break; @@ -369,15 +346,19 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: tdm_rx_start_slot = 1; + fpol_preinv = 0; break; case SND_SOC_DAIFMT_DSP_A: tdm_rx_start_slot = 0; + fpol_preinv = 1; break; case SND_SOC_DAIFMT_DSP_B: tdm_rx_start_slot = 1; + fpol_preinv = 1; break; case SND_SOC_DAIFMT_LEFT_J: tdm_rx_start_slot = 0; + fpol_preinv = 1; break; default: dev_err(tas2770->dev, @@ -391,6 +372,14 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) if (ret < 0) return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_FPOL_MASK, + (fpol_preinv ^ invert_fpol) + ? TAS2770_TDM_CFG_REG0_FPOL_RSING + : TAS2770_TDM_CFG_REG0_FPOL_FALING); + if (ret < 0) + return ret; + return 0; } @@ -489,7 +478,7 @@ static struct snd_soc_dai_driver tas2770_dai_driver[] = { .id = 0, .playback = { .stream_name = "ASI1 Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = TAS2770_RATES, .formats = TAS2770_FORMATS, @@ -537,7 +526,6 @@ static const struct snd_soc_component_driver soc_component_driver_tas2770 = { .probe = tas2770_codec_probe, .suspend = tas2770_codec_suspend, .resume = tas2770_codec_resume, - .set_bias_level = tas2770_set_bias_level, .controls = tas2770_snd_controls, .num_controls = ARRAY_SIZE(tas2770_snd_controls), .dapm_widgets = tas2770_dapm_widgets, diff --git a/sound/soc/codecs/tas2770.h b/sound/soc/codecs/tas2770.h index d156666bcc55..f75f40781ab1 100644 --- a/sound/soc/codecs/tas2770.h +++ b/sound/soc/codecs/tas2770.h @@ -41,6 +41,9 @@ #define TAS2770_TDM_CFG_REG0_31_44_1_48KHZ 0x6 #define TAS2770_TDM_CFG_REG0_31_88_2_96KHZ 0x8 #define TAS2770_TDM_CFG_REG0_31_176_4_192KHZ 0xa +#define TAS2770_TDM_CFG_REG0_FPOL_MASK BIT(0) +#define TAS2770_TDM_CFG_REG0_FPOL_RSING 0 +#define TAS2770_TDM_CFG_REG0_FPOL_FALING 1 /* TDM Configuration Reg1 */ #define TAS2770_TDM_CFG_REG1 TAS2770_REG(0X0, 0x0B) #define TAS2770_TDM_CFG_REG1_MASK GENMASK(5, 1) @@ -135,6 +138,8 @@ struct tas2770_priv { struct device *dev; int v_sense_slot; int i_sense_slot; + bool dac_powered; + bool unmuted; }; #endif /* __TAS2770__ */ diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 2844a9d2bc4a..91a22d927915 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -31,6 +31,7 @@ struct adcx140_priv { struct device *dev; bool micbias_vg; + bool phase_calib_on; unsigned int dai_fmt; unsigned int slot_width; @@ -592,6 +593,52 @@ static const struct snd_soc_dapm_route adcx140_audio_map[] = { {"MIC4M Input Mux", "Digital", "MIC4M"}, }; +#define ADCX140_PHASE_CALIB_SWITCH(xname) {\ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .info = adcx140_phase_calib_info, \ + .get = adcx140_phase_calib_get, \ + .put = adcx140_phase_calib_put} + +static int adcx140_phase_calib_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int adcx140_phase_calib_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) +{ + struct snd_soc_component *codec = + snd_soc_kcontrol_component(kcontrol); + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(codec); + + value->value.integer.value[0] = adcx140->phase_calib_on ? 1 : 0; + + + return 0; +} + +static int adcx140_phase_calib_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(codec); + + bool v = value->value.integer.value[0] ? true : false; + + if (adcx140->phase_calib_on != v) { + adcx140->phase_calib_on = v; + return 1; + } + return 0; +} + static const struct snd_kcontrol_new adcx140_snd_controls[] = { SOC_SINGLE_TLV("Analog CH1 Mic Gain Volume", ADCX140_CH1_CFG1, 2, 42, 0, adc_tlv), @@ -628,6 +675,7 @@ static const struct snd_kcontrol_new adcx140_snd_controls[] = { 0, 0xff, 0, dig_vol_tlv), SOC_SINGLE_TLV("Digital CH8 Out Volume", ADCX140_CH8_CFG2, 0, 0xff, 0, dig_vol_tlv), + ADCX140_PHASE_CALIB_SWITCH("Phase Calibration Switch"), }; static int adcx140_reset(struct adcx140_priv *adcx140) @@ -653,6 +701,8 @@ static int adcx140_reset(struct adcx140_priv *adcx140) static void adcx140_pwr_ctrl(struct adcx140_priv *adcx140, bool power_state) { int pwr_ctrl = 0; + int ret = 0; + struct snd_soc_component *component = adcx140->component; if (power_state) pwr_ctrl = ADCX140_PWR_CFG_ADC_PDZ | ADCX140_PWR_CFG_PLL_PDZ; @@ -660,6 +710,14 @@ static void adcx140_pwr_ctrl(struct adcx140_priv *adcx140, bool power_state) if (adcx140->micbias_vg && power_state) pwr_ctrl |= ADCX140_PWR_CFG_BIAS_PDZ; + if (pwr_ctrl) { + ret = regmap_write(adcx140->regmap, ADCX140_PHASE_CALIB, + adcx140->phase_calib_on ? 0x00 : 0x40); + if (ret) + dev_err(component->dev, "%s: register write error %d\n", + __func__, ret); + } + regmap_update_bits(adcx140->regmap, ADCX140_PWR_CFG, ADCX140_PWR_CTRL_MSK, pwr_ctrl); } @@ -1095,6 +1153,7 @@ static int adcx140_i2c_probe(struct i2c_client *i2c) if (!adcx140) return -ENOMEM; + adcx140->phase_calib_on = false; adcx140->dev = &i2c->dev; adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev, diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index d7d4e3a88b5c..fd80fac8b327 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -1,5 +1,5 @@ // SPDX-License-Identifier: GPL-2.0 -// TLV320ADCX104 Sound driver +// TLV320ADCX140 Sound driver // Copyright (C) 2020 Texas Instruments Incorporated - https://www.ti.com/ #ifndef _TLV320ADCX140_H @@ -90,6 +90,7 @@ #define ADCX140_PWR_CFG 0x75 #define ADCX140_DEV_STS0 0x76 #define ADCX140_DEV_STS1 0x77 +#define ADCX140_PHASE_CALIB 0X7b #define ADCX140_RESET BIT(0) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 8bae4b475068..e5dfb3d752a3 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -271,7 +271,7 @@ static ssize_t keyclick_show(struct device *dev, freq = (125 << ((val >> 8) & 0x7)) >> 1; len = 2 * (1 + ((val >> 4) & 0xf)); - return sprintf(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len); + return sysfs_emit(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len); } /* Any write to the keyclick attribute will trigger the keyclick event */ diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 4b74805cdd2e..ffe1828a4b7e 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -49,6 +49,8 @@ struct aic32x4_priv { struct aic32x4_setup_data *setup; struct device *dev; enum aic32x4_type type; + + unsigned int fmt; }; static int aic32x4_reset_adc(struct snd_soc_dapm_widget *w, @@ -611,6 +613,7 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_component *component = codec_dai->component; + struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); u8 iface_reg_1 = 0; u8 iface_reg_2 = 0; u8 iface_reg_3 = 0; @@ -653,6 +656,8 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return -EINVAL; } + aic32x4->fmt = fmt; + snd_soc_component_update_bits(component, AIC32X4_IFACE1, AIC32X4_IFACE1_DATATYPE_MASK | AIC32X4_IFACE1_MASTER_MASK, iface_reg_1); @@ -757,6 +762,10 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, return -EINVAL; } + /* PCM over I2S is always 2-channel */ + if ((aic32x4->fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) + channels = 2; + madc = DIV_ROUND_UP((32 * adc_resource_class), aosr); max_dosr = (AIC32X4_MAX_DOSR_FREQ / sample_rate / dosr_increment) * dosr_increment; diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 2db3d8a60c7a..1a62bec94005 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -450,7 +450,7 @@ static int uda134x_soc_probe(struct snd_soc_component *component) struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); struct uda134x_platform_data *pd = uda134x->pd; const struct snd_soc_dapm_widget *widgets; - unsigned num_widgets; + unsigned int num_widgets; int ret; printk(KERN_INFO "UDA134X SoC Audio Codec\n"); diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index beeeb35e8032..2c5aa4df1e66 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -5013,16 +5013,22 @@ static const struct regmap_irq wcd9335_codec_irqs[] = { }, }; +static const unsigned int wcd9335_config_regs[] = { + WCD9335_INTR_LEVEL0, +}; + static const struct regmap_irq_chip wcd9335_regmap_irq1_chip = { .name = "wcd9335_pin1_irq", .status_base = WCD9335_INTR_PIN1_STATUS0, .mask_base = WCD9335_INTR_PIN1_MASK0, .ack_base = WCD9335_INTR_PIN1_CLEAR0, - .type_base = WCD9335_INTR_LEVEL0, - .num_type_reg = 4, .num_regs = 4, .irqs = wcd9335_codec_irqs, .num_irqs = ARRAY_SIZE(wcd9335_codec_irqs), + .config_base = wcd9335_config_regs, + .num_config_bases = ARRAY_SIZE(wcd9335_config_regs), + .num_config_regs = 4, + .set_type_config = regmap_irq_set_type_config_simple, }; static int wcd9335_parse_dt(struct wcd9335_codec *wcd) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 781ae569be29..aca06a4026f3 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -1298,7 +1298,6 @@ static struct regmap_irq_chip wcd938x_regmap_irq_chip = { .num_regs = 3, .status_base = WCD938X_DIGITAL_INTR_STATUS_0, .mask_base = WCD938X_DIGITAL_INTR_MASK_0, - .type_base = WCD938X_DIGITAL_INTR_LEVEL_0, .ack_base = WCD938X_DIGITAL_INTR_CLEAR_0, .use_ack = 1, .runtime_pm = true, diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 7523bb944b21..81f89f6767a2 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -114,11 +114,8 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid) if (flags & FSL_SAI_CSR_SEF) dev_dbg(dev, "isr: Tx Frame sync error detected\n"); - if (flags & FSL_SAI_CSR_FEF) { + if (flags & FSL_SAI_CSR_FEF) dev_dbg(dev, "isr: Transmit underrun detected\n"); - /* FIFO reset for safety */ - xcsr |= FSL_SAI_CSR_FR; - } if (flags & FSL_SAI_CSR_FWF) dev_dbg(dev, "isr: Enabled transmit FIFO is empty\n"); @@ -148,11 +145,8 @@ irq_rx: if (flags & FSL_SAI_CSR_SEF) dev_dbg(dev, "isr: Rx Frame sync error detected\n"); - if (flags & FSL_SAI_CSR_FEF) { + if (flags & FSL_SAI_CSR_FEF) dev_dbg(dev, "isr: Receive overflow detected\n"); - /* FIFO reset for safety */ - xcsr |= FSL_SAI_CSR_FR; - } if (flags & FSL_SAI_CSR_FWF) dev_dbg(dev, "isr: Enabled receive FIFO is full\n"); @@ -533,14 +527,17 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, u32 slot_width = word_width; int adir = tx ? RX : TX; u32 pins, bclk; + u32 watermark; int ret, i; - if (sai->slots) - slots = sai->slots; - if (sai->slot_width) slot_width = sai->slot_width; + if (sai->slots) + slots = sai->slots; + else if (sai->bclk_ratio) + slots = sai->bclk_ratio / slot_width; + pins = DIV_ROUND_UP(channels, slots); /* @@ -625,7 +622,15 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR5_FBT_MASK, val_cr5); } - if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1) + /* + * Combine mode has limation: + * - Can't used for singel dataline/FIFO case except the FIFO0 + * - Can't used for multi dataline/FIFO case except the enabled FIFOs + * are successive and start from FIFO0 + * + * So for common usage, all multi fifo case disable the combine mode. + */ + if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1 || sai->is_multi_fifo_dma) regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_FCOMB_MASK, 0); else @@ -636,6 +641,26 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, dma_params->addr = sai->res->start + FSL_SAI_xDR0(tx) + dl_cfg[dl_cfg_idx].start_off[tx] * 0x4; + if (sai->is_multi_fifo_dma) { + sai->audio_config[tx].words_per_fifo = min(slots, channels); + if (tx) { + sai->audio_config[tx].n_fifos_dst = pins; + sai->audio_config[tx].stride_fifos_dst = dl_cfg[dl_cfg_idx].next_off[tx]; + } else { + sai->audio_config[tx].n_fifos_src = pins; + sai->audio_config[tx].stride_fifos_src = dl_cfg[dl_cfg_idx].next_off[tx]; + } + dma_params->maxburst = sai->audio_config[tx].words_per_fifo * pins; + dma_params->peripheral_config = &sai->audio_config[tx]; + dma_params->peripheral_size = sizeof(sai->audio_config[tx]); + + watermark = tx ? (sai->soc_data->fifo_depth - dma_params->maxburst) : + (dma_params->maxburst - 1); + regmap_update_bits(sai->regmap, FSL_SAI_xCR1(tx, ofs), + FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth), + watermark); + } + /* Find a proper tcre setting */ for (i = 0; i < sai->soc_data->pins; i++) { trce_mask = (1 << (i + 1)) - 1; @@ -1263,6 +1288,7 @@ static int fsl_sai_probe(struct platform_device *pdev) char tmp[8]; int irq, ret, i; int index; + u32 dmas[4]; sai = devm_kzalloc(dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -1306,7 +1332,7 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->mclk_clk[i] = devm_clk_get(dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(dev, "failed to get mclk%d clock: %ld\n", - i + 1, PTR_ERR(sai->mclk_clk[i])); + i, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } } @@ -1319,6 +1345,11 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_asoc_get_pll_clocks(&pdev->dev, &sai->pll8k_clk, &sai->pll11k_clk); + /* Use Multi FIFO mode depending on the support from SDMA script */ + ret = of_property_read_u32_array(np, "dmas", dmas, 4); + if (!sai->soc_data->use_edma && !ret && dmas[2] == IMX_DMATYPE_MULTI_SAI) + sai->is_multi_fifo_dma = true; + /* read dataline mask for rx and tx*/ ret = fsl_sai_read_dlcfg(sai); if (ret < 0) { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 17956b5731dc..697f6690068c 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -6,6 +6,7 @@ #ifndef __FSL_SAI_H #define __FSL_SAI_H +#include <linux/dma/imx-dma.h> #include <sound/dmaengine_pcm.h> #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ @@ -281,6 +282,7 @@ struct fsl_sai { bool is_lsb_first; bool is_dsp_mode; bool is_pdm_mode; + bool is_multi_fifo_dma; bool synchronous[2]; struct fsl_sai_dl_cfg *dl_cfg; unsigned int dl_cfg_cnt; @@ -300,6 +302,7 @@ struct fsl_sai { struct pm_qos_request pm_qos_req; struct pinctrl *pinctrl; struct pinctrl_state *pins_state; + struct sdma_peripheral_config audio_config[2]; }; #define TX 1 diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index 2e117311e582..4d99f4858a14 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -19,6 +19,7 @@ struct imx_rpmsg { struct snd_soc_dai_link dai; struct snd_soc_card card; + unsigned long sysclk; }; static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { @@ -28,6 +29,27 @@ static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { SND_SOC_DAPM_MIC("Main MIC", NULL), }; +static int imx_rpmsg_late_probe(struct snd_soc_card *card) +{ + struct imx_rpmsg *data = snd_soc_card_get_drvdata(card); + struct snd_soc_pcm_runtime *rtd = list_first_entry(&card->rtd_list, + struct snd_soc_pcm_runtime, list); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct device *dev = card->dev; + int ret; + + if (!data->sysclk) + return 0; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, data->sysclk, SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set sysclk in %s\n", __func__); + return ret; + } + + return 0; +} + static int imx_rpmsg_probe(struct platform_device *pdev) { struct snd_soc_dai_link_component *dlc; @@ -72,12 +94,18 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->dai.codecs->dai_name = "snd-soc-dummy-dai"; data->dai.codecs->name = "snd-soc-dummy"; } else { + struct clk *clk; + data->dai.codecs->of_node = args.np; ret = snd_soc_get_dai_name(&args, &data->dai.codecs->dai_name); if (ret) { dev_err(&pdev->dev, "Unable to get codec_dai_name\n"); goto fail; } + + clk = devm_get_clk_from_child(&pdev->dev, args.np, NULL); + if (!IS_ERR(clk)) + data->sysclk = clk_get_rate(clk); } data->dai.cpus->dai_name = dev_name(&rpmsg_pdev->dev); @@ -103,6 +131,7 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->card.owner = THIS_MODULE; data->card.dapm_widgets = imx_rpmsg_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_rpmsg_dapm_widgets); + data->card.late_probe = imx_rpmsg_late_probe; /* * Inoder to use common api to get card name and audio routing. * Use parent of_node for this device, revert it after finishing using diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 4a29e314fa95..1b201dd09259 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -15,6 +15,33 @@ #include <sound/pcm_params.h> #include <sound/simple_card_utils.h> +static void asoc_simple_fixup_sample_fmt(struct asoc_simple_data *data, + struct snd_pcm_hw_params *params) +{ + int i; + struct snd_mask *mask = hw_param_mask(params, + SNDRV_PCM_HW_PARAM_FORMAT); + struct { + char *fmt; + u32 val; + } of_sample_fmt_table[] = { + { "s8", SNDRV_PCM_FORMAT_S8}, + { "s16_le", SNDRV_PCM_FORMAT_S16_LE}, + { "s24_le", SNDRV_PCM_FORMAT_S24_LE}, + { "s24_3le", SNDRV_PCM_FORMAT_S24_3LE}, + { "s32_le", SNDRV_PCM_FORMAT_S32_LE}, + }; + + for (i = 0; i < ARRAY_SIZE(of_sample_fmt_table); i++) { + if (!strcmp(data->convert_sample_format, + of_sample_fmt_table[i].fmt)) { + snd_mask_none(mask); + snd_mask_set(mask, of_sample_fmt_table[i].val); + break; + } + } +} + void asoc_simple_convert_fixup(struct asoc_simple_data *data, struct snd_pcm_hw_params *params) { @@ -30,6 +57,9 @@ void asoc_simple_convert_fixup(struct asoc_simple_data *data, if (data->convert_channels) channels->min = channels->max = data->convert_channels; + + if (data->convert_sample_format) + asoc_simple_fixup_sample_fmt(data, params); } EXPORT_SYMBOL_GPL(asoc_simple_convert_fixup); @@ -49,6 +79,10 @@ void asoc_simple_parse_convert(struct device_node *np, /* channels transfer */ snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-channels"); of_property_read_u32(np, prop, &data->convert_channels); + + /* convert sample format */ + snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-sample-format"); + of_property_read_string(np, prop, &data->convert_sample_format); } EXPORT_SYMBOL_GPL(asoc_simple_parse_convert); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index ded903f95b67..d2ca710ac3fa 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -23,7 +23,7 @@ config SND_SOC_INTEL_CATPT depends on ACPI || COMPILE_TEST depends on DMADEVICES && SND_DMA_SGBUF select DW_DMAC_CORE - select SND_SOC_ACPI_INTEL_MATCH + select SND_SOC_ACPI if ACPI select WANT_DEV_COREDUMP select SND_INTEL_DSP_CONFIG help diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index 160b50f479fb..a0d29510d2bc 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -242,11 +242,11 @@ static ssize_t firmware_version_show(struct device *dev, if (ctx->fw_version.type == 0 && ctx->fw_version.major == 0 && ctx->fw_version.minor == 0 && ctx->fw_version.build == 0) - return sprintf(buf, "FW not yet loaded\n"); + return sysfs_emit(buf, "FW not yet loaded\n"); else - return sprintf(buf, "v%02x.%02x.%02x.%02x\n", - ctx->fw_version.type, ctx->fw_version.major, - ctx->fw_version.minor, ctx->fw_version.build); + return sysfs_emit(buf, "v%02x.%02x.%02x.%02x\n", + ctx->fw_version.type, ctx->fw_version.major, + ctx->fw_version.minor, ctx->fw_version.build); } diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index f21b0cdd3206..8fe5917b1e26 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -636,8 +636,8 @@ static ssize_t topology_name_read(struct file *file, char __user *user_buf, size char buf[64]; size_t len; - len = snprintf(buf, sizeof(buf), "%s/%s\n", component->driver->topology_name_prefix, - mach->tplg_filename); + len = scnprintf(buf, sizeof(buf), "%s/%s\n", component->driver->topology_name_prefix, + mach->tplg_filename); return simple_read_from_buffer(user_buf, count, ppos, buf, len); } diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index eea1e26acfda..53458e748191 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -1,8 +1,8 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-sst-haswell-objs := hsw_rt5640.o +snd-soc-hsw-rt5640-objs := hsw_rt5640.o snd-soc-sst-bdw-rt5650-mach-objs := bdw-rt5650.o snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o -snd-soc-sst-broadwell-objs := bdw_rt286.o +snd-soc-bdw-rt286-objs := bdw_rt286.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o snd-soc-sst-bxt-rt298-objs := bxt_rt298.o snd-soc-sst-sof-pcm512x-objs := sof_pcm512x.o @@ -47,13 +47,13 @@ obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_SOF_CS42L42_MACH) += snd-soc-sof_cs42l42.o obj-$(CONFIG_SND_SOC_INTEL_SOF_ES8336_MACH) += snd-soc-sof_es8336.o obj-$(CONFIG_SND_SOC_INTEL_SOF_NAU8825_MACH) += snd-soc-sof_nau8825.o -obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o +obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-hsw-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o obj-$(CONFIG_SND_SOC_INTEL_SOF_PCM512x_MACH) += snd-soc-sst-sof-pcm512x.o obj-$(CONFIG_SND_SOC_INTEL_SOF_WM8804_MACH) += snd-soc-sst-sof-wm8804.o obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o -obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-bdw-rt286.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index f4192df962d6..6e39eda77385 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -10,6 +10,9 @@ #include "../../codecs/cs35l41.h" #include "sof_cirrus_common.h" +#define CS35L41_HID "CSC3541" +#define CS35L41_MAX_AMPS 4 + /* * Cirrus Logic CS35L41/CS35L53 */ @@ -35,50 +38,12 @@ static const struct snd_soc_dapm_route cs35l41_dapm_routes[] = { {"TR Spk", NULL, "TR SPK"}, }; -static struct snd_soc_dai_link_component cs35l41_components[] = { - { - .name = CS35L41_DEV0_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV1_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV2_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV3_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, -}; +static struct snd_soc_dai_link_component cs35l41_components[CS35L41_MAX_AMPS]; /* * Mapping between ACPI instance id and speaker position. - * - * Four speakers: - * 0: Tweeter left, 1: Woofer left - * 2: Tweeter right, 3: Woofer right */ -static struct snd_soc_codec_conf cs35l41_codec_conf[] = { - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV0_NAME), - .name_prefix = "TL", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV1_NAME), - .name_prefix = "WL", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV2_NAME), - .name_prefix = "TR", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV3_NAME), - .name_prefix = "WR", - }, -}; +static struct snd_soc_codec_conf cs35l41_codec_conf[CS35L41_MAX_AMPS]; static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) { @@ -117,10 +82,10 @@ static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) static const struct { unsigned int rx[2]; } cs35l41_channel_map[] = { - {.rx = {0, 1}}, /* TL */ {.rx = {0, 1}}, /* WL */ - {.rx = {1, 0}}, /* TR */ {.rx = {1, 0}}, /* WR */ + {.rx = {0, 1}}, /* TL */ + {.rx = {1, 0}}, /* TR */ }; static int cs35l41_hw_params(struct snd_pcm_substream *substream, @@ -175,10 +140,51 @@ static const struct snd_soc_ops cs35l41_ops = { .hw_params = cs35l41_hw_params, }; +static const char * const cs35l41_name_prefixes[] = { "WL", "WR", "TL", "TR" }; + +/* + * Expected UIDs are integers (stored as strings). + * UID Mapping is fixed: + * UID 0x0 -> WL + * UID 0x1 -> WR + * UID 0x2 -> TL + * UID 0x3 -> TR + * Note: If there are less than 4 Amps, UIDs still map to WL/WR/TL/TR. Dynamic code will only create + * dai links for UIDs which exist, and ignore non-existant ones. Only 2 or 4 amps are expected. + * Return number of codecs found. + */ +static int cs35l41_compute_codec_conf(void) +{ + const char * const uid_strings[] = { "0", "1", "2", "3" }; + unsigned int uid, sz = 0; + struct acpi_device *adev; + struct device *physdev; + + for (uid = 0; uid < CS35L41_MAX_AMPS; uid++) { + adev = acpi_dev_get_first_match_dev(CS35L41_HID, uid_strings[uid], -1); + if (!adev) { + pr_devel("Cannot find match for HID %s UID %u (%s)\n", CS35L41_HID, uid, + cs35l41_name_prefixes[uid]); + continue; + } + physdev = get_device(acpi_get_first_physical_node(adev)); + cs35l41_components[sz].name = dev_name(physdev); + cs35l41_components[sz].dai_name = CS35L41_CODEC_DAI; + cs35l41_codec_conf[sz].dlc.name = dev_name(physdev); + cs35l41_codec_conf[sz].name_prefix = cs35l41_name_prefixes[uid]; + acpi_dev_put(adev); + sz++; + } + + if (sz != 2 && sz != 4) + pr_warn("Invalid number of cs35l41 amps found: %d, expected 2 or 4\n", sz); + return sz; +} + void cs35l41_set_dai_link(struct snd_soc_dai_link *link) { + link->num_codecs = cs35l41_compute_codec_conf(); link->codecs = cs35l41_components; - link->num_codecs = ARRAY_SIZE(cs35l41_components); link->init = cs35l41_init; link->ops = &cs35l41_ops; } diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index c7f33c89588e..606cc3242a60 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -760,6 +760,9 @@ static int sof_es8336_remove(struct platform_device *pdev) static const struct platform_device_id board_ids[] = { { + .name = "sof-essx8336", /* default quirk == 0 */ + }, + { .name = "adl_es83x6_c1_h02", .driver_data = (kernel_ulong_t)(SOF_ES8336_SSP_CODEC(1) | SOF_NO_OF_HDMI_CAPTURE_SSP(2) | @@ -786,5 +789,4 @@ module_platform_driver(sof_es8336_driver); MODULE_DESCRIPTION("ASoC Intel(R) SOF + ES8336 Machine driver"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:sof-essx8336"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); diff --git a/sound/soc/intel/catpt/device.c b/sound/soc/intel/catpt/device.c index d48a71d2cf1e..d5d08bd766c7 100644 --- a/sound/soc/intel/catpt/device.c +++ b/sound/soc/intel/catpt/device.c @@ -22,7 +22,6 @@ #include <sound/intel-dsp-config.h> #include <sound/soc.h> #include <sound/soc-acpi.h> -#include <sound/soc-acpi-intel-match.h> #include "core.h" #include "registers.h" @@ -310,8 +309,36 @@ static int catpt_acpi_remove(struct platform_device *pdev) return 0; } +static struct snd_soc_acpi_mach lpt_machines[] = { + { + .id = "INT33CA", + .drv_name = "hsw_rt5640", + }, + {} +}; + +static struct snd_soc_acpi_mach wpt_machines[] = { + { + .id = "INT33CA", + .drv_name = "hsw_rt5640", + }, + { + .id = "INT343A", + .drv_name = "bdw_rt286", + }, + { + .id = "10EC5650", + .drv_name = "bdw-rt5650", + }, + { + .id = "RT5677CE", + .drv_name = "bdw-rt5677", + }, + {} +}; + static struct catpt_spec lpt_desc = { - .machines = snd_soc_acpi_intel_haswell_machines, + .machines = lpt_machines, .core_id = 0x01, .host_dram_offset = 0x000000, .host_iram_offset = 0x080000, @@ -326,7 +353,7 @@ static struct catpt_spec lpt_desc = { }; static struct catpt_spec wpt_desc = { - .machines = snd_soc_acpi_intel_broadwell_machines, + .machines = wpt_machines, .core_id = 0x02, .host_dram_offset = 0x000000, .host_iram_offset = 0x0A0000, diff --git a/sound/soc/intel/catpt/sysfs.c b/sound/soc/intel/catpt/sysfs.c index 1bdbcc04dc71..9b6d2d93a2e7 100644 --- a/sound/soc/intel/catpt/sysfs.c +++ b/sound/soc/intel/catpt/sysfs.c @@ -27,8 +27,8 @@ static ssize_t fw_version_show(struct device *dev, if (ret) return CATPT_IPC_ERROR(ret); - return sprintf(buf, "%d.%d.%d.%d\n", version.type, version.major, - version.minor, version.build); + return sysfs_emit(buf, "%d.%d.%d.%d\n", version.type, version.major, + version.minor, version.build); } static DEVICE_ATTR_RO(fw_version); @@ -37,7 +37,7 @@ static ssize_t fw_info_show(struct device *dev, { struct catpt_dev *cdev = dev_get_drvdata(dev); - return sprintf(buf, "%s\n", cdev->ipc.config.fw_info); + return sysfs_emit(buf, "%s\n", cdev->ipc.config.fw_info); } static DEVICE_ATTR_RO(fw_info); diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 8ca8f872ec80..41054cf09ec9 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -9,7 +9,7 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m soc-acpi-intel-cml-match.o soc-acpi-intel-icl-match.o \ soc-acpi-intel-tgl-match.o soc-acpi-intel-ehl-match.o \ soc-acpi-intel-jsl-match.o soc-acpi-intel-adl-match.o \ - soc-acpi-intel-mtl-match.o \ + soc-acpi-intel-rpl-match.o soc-acpi-intel-mtl-match.o \ soc-acpi-intel-hda-match.o \ soc-acpi-intel-sdw-mockup-match.o diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index cbcb649604e5..6daf60b1edf1 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -9,40 +9,25 @@ #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> -struct snd_soc_acpi_mach snd_soc_acpi_intel_haswell_machines[] = { - { - .id = "INT33CA", - .drv_name = "hsw_rt5640", - .fw_filename = "intel/IntcSST1.bin", - .sof_tplg_filename = "sof-hsw.tplg", - }, - {} -}; -EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_haswell_machines); - struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { { .id = "INT343A", .drv_name = "bdw_rt286", - .fw_filename = "intel/IntcSST2.bin", .sof_tplg_filename = "sof-bdw-rt286.tplg", }, { .id = "10EC5650", .drv_name = "bdw-rt5650", - .fw_filename = "intel/IntcSST2.bin", .sof_tplg_filename = "sof-bdw-rt5650.tplg", }, { .id = "RT5677CE", .drv_name = "bdw-rt5677", - .fw_filename = "intel/IntcSST2.bin", .sof_tplg_filename = "sof-bdw-rt5677.tplg", }, { .id = "INT33CA", .drv_name = "hsw_rt5640", - .fw_filename = "intel/IntcSST2.bin", .sof_tplg_filename = "sof-bdw-rt5640.tplg", }, {} diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c new file mode 100644 index 000000000000..0b77401e4e6f --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -0,0 +1,51 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * soc-apci-intel-rpl-match.c - tables and support for RPL ACPI enumeration. + * + * Copyright (c) 2022 Intel Corporation. + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> + +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, +}; + +static const struct snd_soc_acpi_adr_device rt711_0_adr[] = { + { + .adr = 0x000020025D071100ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt711" + } +}; + +static const struct snd_soc_acpi_link_adr rpl_rvp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_0_adr), + .adr_d = rt711_0_adr, + }, + {} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_machines); + +/* this table is used when there is no I2S codec present */ +struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_sdw_machines[] = { + { + .link_mask = 0x1, /* link0 required */ + .links = rpl_rvp, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-rpl-rt711.tplg", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_sdw_machines); diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index deb7b820325e..e617b4c335a4 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -61,7 +61,7 @@ static ssize_t platform_id_show(struct device *dev, nhlt->header.oem_revision); skl_nhlt_trim_space(platform_id); - return sprintf(buf, "%s\n", platform_id); + return sysfs_emit(buf, "%s\n", platform_id); } static DEVICE_ATTR_RO(platform_id); diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c index eb729ab00f5a..d7e94e6a19c7 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c @@ -1359,6 +1359,9 @@ static const struct snd_soc_dapm_widget mt8186_memif_widgets[] = { SND_SOC_DAPM_MUX("UL5_IN_MUX", SND_SOC_NOPM, 0, 0, &ul5_in_mux_control), + SND_SOC_DAPM_MIXER("DSP_DL1_VIRT", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("DSP_DL2_VIRT", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_INPUT("UL1_VIRTUAL_INPUT"), SND_SOC_DAPM_INPUT("UL2_VIRTUAL_INPUT"), SND_SOC_DAPM_INPUT("UL3_VIRTUAL_INPUT"), diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c index ec79e2f2a54d..e553a555d168 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c @@ -658,9 +658,15 @@ static const struct snd_soc_dapm_route mtk_dai_i2s_routes[] = { {"I2S1_CH1", "DL1_CH1 Switch", "DL1"}, {"I2S1_CH2", "DL1_CH2 Switch", "DL1"}, + {"I2S1_CH1", "DL1_CH1 Switch", "DSP_DL1_VIRT"}, + {"I2S1_CH2", "DL1_CH2 Switch", "DSP_DL1_VIRT"}, + {"I2S1_CH1", "DL2_CH1 Switch", "DL2"}, {"I2S1_CH2", "DL2_CH2 Switch", "DL2"}, + {"I2S1_CH1", "DL2_CH1 Switch", "DSP_DL2_VIRT"}, + {"I2S1_CH2", "DL2_CH2 Switch", "DSP_DL2_VIRT"}, + {"I2S1_CH1", "DL3_CH1 Switch", "DL3"}, {"I2S1_CH2", "DL3_CH2 Switch", "DL3"}, @@ -728,9 +734,15 @@ static const struct snd_soc_dapm_route mtk_dai_i2s_routes[] = { {"I2S3_CH1", "DL1_CH1 Switch", "DL1"}, {"I2S3_CH2", "DL1_CH2 Switch", "DL1"}, + {"I2S3_CH1", "DL1_CH1 Switch", "DSP_DL1_VIRT"}, + {"I2S3_CH2", "DL1_CH2 Switch", "DSP_DL1_VIRT"}, + {"I2S3_CH1", "DL2_CH1 Switch", "DL2"}, {"I2S3_CH2", "DL2_CH2 Switch", "DL2"}, + {"I2S3_CH1", "DL2_CH1 Switch", "DSP_DL2_VIRT"}, + {"I2S3_CH2", "DL2_CH2 Switch", "DSP_DL2_VIRT"}, + {"I2S3_CH1", "DL3_CH1 Switch", "DL3"}, {"I2S3_CH2", "DL3_CH2 Switch", "DL3"}, @@ -968,7 +980,7 @@ static int mtk_dai_i2s_config(struct mtk_base_afe *afe, } /* set share i2s */ - if (i2s_priv && i2s_priv->share_i2s_id >= 0) { + if (i2s_priv->share_i2s_id >= 0) { ret = mtk_dai_i2s_config(afe, params, i2s_priv->share_i2s_id); if (ret) return ret; diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c index 387f25cad809..17a15bec41da 100644 --- a/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c +++ b/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c @@ -18,6 +18,8 @@ #include "../../codecs/da7219.h" #include "../../codecs/mt6358.h" #include "../common/mtk-afe-platform-driver.h" +#include "../common/mtk-dsp-sof-common.h" +#include "../common/mtk-soc-card.h" #include "mt8186-afe-common.h" #include "mt8186-afe-clk.h" #include "mt8186-afe-gpio.h" @@ -26,6 +28,11 @@ #define DA7219_CODEC_DAI "da7219-hifi" #define DA7219_DEV_NAME "da7219.5-001a" +#define SOF_DMA_DL1 "SOF_DMA_DL1" +#define SOF_DMA_DL2 "SOF_DMA_DL2" +#define SOF_DMA_UL1 "SOF_DMA_UL1" +#define SOF_DMA_UL2 "SOF_DMA_UL2" + struct mt8186_mt6366_da7219_max98357_priv { struct snd_soc_jack headset_jack, hdmi_jack; }; @@ -47,8 +54,9 @@ static struct snd_soc_codec_conf mt8186_mt6366_da7219_max98357_codec_conf[] = { static int mt8186_da7219_init(struct snd_soc_pcm_runtime *rtd) { - struct mt8186_mt6366_da7219_max98357_priv *priv = + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); + struct mt8186_mt6366_da7219_max98357_priv *priv = soc_card_data->mach_priv; struct snd_soc_jack *jack = &priv->headset_jack; struct snd_soc_component *cmpnt_codec = asoc_rtd_to_codec(rtd, 0)->component; @@ -154,8 +162,9 @@ static int mt8186_mt6366_da7219_max98357_hdmi_init(struct snd_soc_pcm_runtime *r { struct snd_soc_component *cmpnt_codec = asoc_rtd_to_codec(rtd, 0)->component; - struct mt8186_mt6366_da7219_max98357_priv *priv = + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); + struct mt8186_mt6366_da7219_max98357_priv *priv = soc_card_data->mach_priv; int ret; ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, &priv->hdmi_jack); @@ -201,6 +210,24 @@ static int mt8186_anx7625_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return mt8186_hw_params_fixup(rtd, params, SNDRV_PCM_FORMAT_S24_LE); } +/* fixup the BE DAI link to match any values from topology */ +static int mt8186_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + int ret; + + ret = mtk_sof_dai_link_fixup(rtd, params); + + if (!strcmp(rtd->dai_link->name, "I2S0") || + !strcmp(rtd->dai_link->name, "I2S1") || + !strcmp(rtd->dai_link->name, "I2S2")) + mt8186_i2s_hw_params_fixup(rtd, params); + else if (!strcmp(rtd->dai_link->name, "I2S3")) + mt8186_anx7625_i2s_hw_params_fixup(rtd, params); + + return ret; +} + static int mt8186_mt6366_da7219_max98357_playback_startup(struct snd_pcm_substream *substream) { static const unsigned int rates[] = { @@ -474,6 +501,33 @@ SND_SOC_DAILINK_DEFS(hostless_src_aaudio, DAILINK_COMP_ARRAY(COMP_CPU("Hostless SRC AAudio DAI")), DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); +SND_SOC_DAILINK_DEFS(AFE_SOF_DL1, + DAILINK_COMP_ARRAY(COMP_CPU("SOF_DL1")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +SND_SOC_DAILINK_DEFS(AFE_SOF_DL2, + DAILINK_COMP_ARRAY(COMP_CPU("SOF_DL2")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +SND_SOC_DAILINK_DEFS(AFE_SOF_UL1, + DAILINK_COMP_ARRAY(COMP_CPU("SOF_UL1")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +SND_SOC_DAILINK_DEFS(AFE_SOF_UL2, + DAILINK_COMP_ARRAY(COMP_CPU("SOF_UL2")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +static const struct sof_conn_stream g_sof_conn_streams[] = { + { "I2S1", "AFE_SOF_DL1", SOF_DMA_DL1, SNDRV_PCM_STREAM_PLAYBACK}, + { "I2S3", "AFE_SOF_DL2", SOF_DMA_DL2, SNDRV_PCM_STREAM_PLAYBACK}, + { "Primary Codec", "AFE_SOF_UL1", SOF_DMA_UL1, SNDRV_PCM_STREAM_CAPTURE}, + { "I2S0", "AFE_SOF_UL2", SOF_DMA_UL2, SNDRV_PCM_STREAM_CAPTURE}, +}; + static struct snd_soc_dai_link mt8186_mt6366_da7219_max98357_dai_links[] = { /* Front End DAI links */ { @@ -848,12 +902,41 @@ static struct snd_soc_dai_link mt8186_mt6366_da7219_max98357_dai_links[] = { .ignore_suspend = 1, SND_SOC_DAILINK_REG(hostless_ul6), }, + /* SOF BE */ + { + .name = "AFE_SOF_DL1", + .no_pcm = 1, + .dpcm_playback = 1, + SND_SOC_DAILINK_REG(AFE_SOF_DL1), + }, + { + .name = "AFE_SOF_DL2", + .no_pcm = 1, + .dpcm_playback = 1, + SND_SOC_DAILINK_REG(AFE_SOF_DL2), + }, + { + .name = "AFE_SOF_UL1", + .no_pcm = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(AFE_SOF_UL1), + }, + { + .name = "AFE_SOF_UL2", + .no_pcm = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(AFE_SOF_UL2), + }, }; static const struct snd_soc_dapm_widget mt8186_mt6366_da7219_max98357_widgets[] = { SND_SOC_DAPM_SPK("Speakers", NULL), SND_SOC_DAPM_OUTPUT("HDMI1"), + SND_SOC_DAPM_MIXER(SOF_DMA_DL1, SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER(SOF_DMA_DL2, SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER(SOF_DMA_UL1, SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER(SOF_DMA_UL2, SND_SOC_NOPM, 0, 0, NULL, 0), }; static const struct snd_soc_dapm_route @@ -862,6 +945,14 @@ mt8186_mt6366_da7219_max98357_routes[] = { { "Speakers", NULL, "Speaker"}, /* HDMI */ { "HDMI1", NULL, "TX"}, + /* SOF Uplink */ + {SOF_DMA_UL1, NULL, "UL1_CH1"}, + {SOF_DMA_UL1, NULL, "UL1_CH2"}, + {SOF_DMA_UL2, NULL, "UL2_CH1"}, + {SOF_DMA_UL2, NULL, "UL2_CH2"}, + /* SOF Downlink */ + {"DSP_DL1_VIRT", NULL, SOF_DMA_DL1}, + {"DSP_DL2_VIRT", NULL, SOF_DMA_DL2}, }; static const struct snd_kcontrol_new @@ -871,7 +962,7 @@ mt8186_mt6366_da7219_max98357_controls[] = { }; static struct snd_soc_card mt8186_mt6366_da7219_max98357_soc_card = { - .name = "mt8186_mt6366_da7219_max98357", + .name = "mt8186_da7219_max98357", .owner = THIS_MODULE, .dai_link = mt8186_mt6366_da7219_max98357_dai_links, .num_links = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_dai_links), @@ -889,8 +980,10 @@ static int mt8186_mt6366_da7219_max98357_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card; struct snd_soc_dai_link *dai_link; - struct mt8186_mt6366_da7219_max98357_priv *priv; - struct device_node *platform_node, *headset_codec, *playback_codec; + struct mtk_soc_card_data *soc_card_data; + struct mt8186_mt6366_da7219_max98357_priv *mach_priv; + struct device_node *platform_node, *headset_codec, *playback_codec, *adsp_node; + int sof_on = 0; int ret, i; card = (struct snd_soc_card *)device_get_match_data(&pdev->dev); @@ -898,11 +991,60 @@ static int mt8186_mt6366_da7219_max98357_dev_probe(struct platform_device *pdev) return -EINVAL; card->dev = &pdev->dev; + soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*soc_card_data), GFP_KERNEL); + if (!soc_card_data) + return -ENOMEM; + mach_priv = devm_kzalloc(&pdev->dev, sizeof(*mach_priv), GFP_KERNEL); + if (!mach_priv) + return -ENOMEM; + + soc_card_data->mach_priv = mach_priv; + + adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0); + if (adsp_node) { + struct mtk_sof_priv *sof_priv; + + sof_priv = devm_kzalloc(&pdev->dev, sizeof(*sof_priv), GFP_KERNEL); + if (!sof_priv) { + ret = -ENOMEM; + goto err_adsp_node; + } + sof_priv->conn_streams = g_sof_conn_streams; + sof_priv->num_streams = ARRAY_SIZE(g_sof_conn_streams); + sof_priv->sof_dai_link_fixup = mt8186_sof_dai_link_fixup; + soc_card_data->sof_priv = sof_priv; + card->probe = mtk_sof_card_probe; + card->late_probe = mtk_sof_card_late_probe; + if (!card->topology_shortname_created) { + snprintf(card->topology_shortname, 32, "sof-%s", card->name); + card->topology_shortname_created = true; + } + card->name = card->topology_shortname; + sof_on = 1; + } else { + dev_info(&pdev->dev, "Probe without adsp\n"); + } + + if (of_property_read_bool(pdev->dev.of_node, "mediatek,dai-link")) { + ret = mtk_sof_dailink_parse_of(card, pdev->dev.of_node, + "mediatek,dai-link", + mt8186_mt6366_da7219_max98357_dai_links, + ARRAY_SIZE(mt8186_mt6366_da7219_max98357_dai_links)); + if (ret) { + dev_dbg(&pdev->dev, "Parse dai-link fail\n"); + goto err_adsp_node; + } + } else { + if (!sof_on) + card->num_links = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_dai_links) + - ARRAY_SIZE(g_sof_conn_streams); + } + platform_node = of_parse_phandle(pdev->dev.of_node, "mediatek,platform", 0); if (!platform_node) { ret = -EINVAL; dev_err_probe(&pdev->dev, ret, "Property 'platform' missing or invalid\n"); - return ret; + goto err_platform_node; } playback_codec = of_get_child_by_name(pdev->dev.of_node, "playback-codecs"); @@ -941,17 +1083,14 @@ static int mt8186_mt6366_da7219_max98357_dev_probe(struct platform_device *pdev) goto err_probe; } - if (!dai_link->platforms->name) - dai_link->platforms->of_node = platform_node; - } + if (!strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF")) && sof_on) + dai_link->platforms->of_node = adsp_node; - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) { - ret = -ENOMEM; - goto err_probe; + if (!dai_link->platforms->name && !dai_link->platforms->of_node) + dai_link->platforms->of_node = platform_node; } - snd_soc_card_set_drvdata(card, priv); + snd_soc_card_set_drvdata(card, soc_card_data); ret = mt8186_afe_gpio_init(&pdev->dev); if (ret) { @@ -969,6 +1108,9 @@ err_headset_codec: of_node_put(playback_codec); err_playback_codec: of_node_put(platform_node); +err_platform_node: +err_adsp_node: + of_node_put(adsp_node); return ret; } diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c index 891146fd6c2b..393d179d61de 100644 --- a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c +++ b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c @@ -19,6 +19,8 @@ #include "../../codecs/mt6358.h" #include "../../codecs/rt5682.h" #include "../common/mtk-afe-platform-driver.h" +#include "../common/mtk-dsp-sof-common.h" +#include "../common/mtk-soc-card.h" #include "mt8186-afe-common.h" #include "mt8186-afe-clk.h" #include "mt8186-afe-gpio.h" @@ -30,6 +32,11 @@ #define RT5682S_CODEC_DAI "rt5682s-aif1" #define RT5682S_DEV0_NAME "rt5682s.5-001a" +#define SOF_DMA_DL1 "SOF_DMA_DL1" +#define SOF_DMA_DL2 "SOF_DMA_DL2" +#define SOF_DMA_UL1 "SOF_DMA_UL1" +#define SOF_DMA_UL2 "SOF_DMA_UL2" + struct mt8186_mt6366_rt1019_rt5682s_priv { struct snd_soc_jack headset_jack, hdmi_jack; }; @@ -51,8 +58,9 @@ static struct snd_soc_codec_conf mt8186_mt6366_rt1019_rt5682s_codec_conf[] = { static int mt8186_rt5682s_init(struct snd_soc_pcm_runtime *rtd) { - struct mt8186_mt6366_rt1019_rt5682s_priv *priv = + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); + struct mt8186_mt6366_rt1019_rt5682s_priv *priv = soc_card_data->mach_priv; struct snd_soc_jack *jack = &priv->headset_jack; struct snd_soc_component *cmpnt_codec = asoc_rtd_to_codec(rtd, 0)->component; @@ -130,8 +138,9 @@ static int mt8186_mt6366_rt1019_rt5682s_hdmi_init(struct snd_soc_pcm_runtime *rt { struct snd_soc_component *cmpnt_codec = asoc_rtd_to_codec(rtd, 0)->component; - struct mt8186_mt6366_rt1019_rt5682s_priv *priv = + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); + struct mt8186_mt6366_rt1019_rt5682s_priv *priv = soc_card_data->mach_priv; int ret; ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, &priv->hdmi_jack); @@ -177,6 +186,24 @@ static int mt8186_it6505_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return mt8186_hw_params_fixup(rtd, params, SNDRV_PCM_FORMAT_S32_LE); } +/* fixup the BE DAI link to match any values from topology */ +static int mt8186_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + int ret; + + ret = mtk_sof_dai_link_fixup(rtd, params); + + if (!strcmp(rtd->dai_link->name, "I2S0") || + !strcmp(rtd->dai_link->name, "I2S1") || + !strcmp(rtd->dai_link->name, "I2S2")) + mt8186_i2s_hw_params_fixup(rtd, params); + else if (!strcmp(rtd->dai_link->name, "I2S3")) + mt8186_it6505_i2s_hw_params_fixup(rtd, params); + + return ret; +} + static int mt8186_mt6366_rt1019_rt5682s_playback_startup(struct snd_pcm_substream *substream) { static const unsigned int rates[] = { @@ -450,6 +477,33 @@ SND_SOC_DAILINK_DEFS(hostless_src_aaudio, DAILINK_COMP_ARRAY(COMP_CPU("Hostless SRC AAudio DAI")), DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); +SND_SOC_DAILINK_DEFS(AFE_SOF_DL1, + DAILINK_COMP_ARRAY(COMP_CPU("SOF_DL1")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +SND_SOC_DAILINK_DEFS(AFE_SOF_DL2, + DAILINK_COMP_ARRAY(COMP_CPU("SOF_DL2")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +SND_SOC_DAILINK_DEFS(AFE_SOF_UL1, + DAILINK_COMP_ARRAY(COMP_CPU("SOF_UL1")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +SND_SOC_DAILINK_DEFS(AFE_SOF_UL2, + DAILINK_COMP_ARRAY(COMP_CPU("SOF_UL2")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +static const struct sof_conn_stream g_sof_conn_streams[] = { + { "I2S1", "AFE_SOF_DL1", SOF_DMA_DL1, SNDRV_PCM_STREAM_PLAYBACK}, + { "I2S3", "AFE_SOF_DL2", SOF_DMA_DL2, SNDRV_PCM_STREAM_PLAYBACK}, + { "Primary Codec", "AFE_SOF_UL1", SOF_DMA_UL1, SNDRV_PCM_STREAM_CAPTURE}, + { "I2S0", "AFE_SOF_UL2", SOF_DMA_UL2, SNDRV_PCM_STREAM_CAPTURE}, +}; + static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = { /* Front End DAI links */ { @@ -824,12 +878,41 @@ static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = { .ignore_suspend = 1, SND_SOC_DAILINK_REG(hostless_ul6), }, + /* SOF BE */ + { + .name = "AFE_SOF_DL1", + .no_pcm = 1, + .dpcm_playback = 1, + SND_SOC_DAILINK_REG(AFE_SOF_DL1), + }, + { + .name = "AFE_SOF_DL2", + .no_pcm = 1, + .dpcm_playback = 1, + SND_SOC_DAILINK_REG(AFE_SOF_DL2), + }, + { + .name = "AFE_SOF_UL1", + .no_pcm = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(AFE_SOF_UL1), + }, + { + .name = "AFE_SOF_UL2", + .no_pcm = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(AFE_SOF_UL2), + }, }; static const struct snd_soc_dapm_widget mt8186_mt6366_rt1019_rt5682s_widgets[] = { SND_SOC_DAPM_SPK("Speakers", NULL), SND_SOC_DAPM_OUTPUT("HDMI1"), + SND_SOC_DAPM_MIXER(SOF_DMA_DL1, SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER(SOF_DMA_DL2, SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER(SOF_DMA_UL1, SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER(SOF_DMA_UL2, SND_SOC_NOPM, 0, 0, NULL, 0), }; static const struct snd_soc_dapm_route @@ -838,6 +921,14 @@ mt8186_mt6366_rt1019_rt5682s_routes[] = { { "Speakers", NULL, "Speaker" }, /* HDMI */ { "HDMI1", NULL, "TX" }, + /* SOF Uplink */ + {SOF_DMA_UL1, NULL, "UL1_CH1"}, + {SOF_DMA_UL1, NULL, "UL1_CH2"}, + {SOF_DMA_UL2, NULL, "UL2_CH1"}, + {SOF_DMA_UL2, NULL, "UL2_CH2"}, + /* SOF Downlink */ + {"DSP_DL1_VIRT", NULL, SOF_DMA_DL1}, + {"DSP_DL2_VIRT", NULL, SOF_DMA_DL2}, }; static const struct snd_kcontrol_new @@ -847,7 +938,7 @@ mt8186_mt6366_rt1019_rt5682s_controls[] = { }; static struct snd_soc_card mt8186_mt6366_rt1019_rt5682s_soc_card = { - .name = "mt8186_mt6366_rt1019_rt5682s", + .name = "mt8186_rt1019_rt5682s", .owner = THIS_MODULE, .dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links, .num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links), @@ -865,8 +956,10 @@ static int mt8186_mt6366_rt1019_rt5682s_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card; struct snd_soc_dai_link *dai_link; - struct mt8186_mt6366_rt1019_rt5682s_priv *priv; - struct device_node *platform_node, *headset_codec, *playback_codec; + struct mtk_soc_card_data *soc_card_data; + struct mt8186_mt6366_rt1019_rt5682s_priv *mach_priv; + struct device_node *platform_node, *headset_codec, *playback_codec, *adsp_node; + int sof_on = 0; int ret, i; card = (struct snd_soc_card *)device_get_match_data(&pdev->dev); @@ -874,11 +967,60 @@ static int mt8186_mt6366_rt1019_rt5682s_dev_probe(struct platform_device *pdev) return -EINVAL; card->dev = &pdev->dev; + soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*soc_card_data), GFP_KERNEL); + if (!soc_card_data) + return -ENOMEM; + mach_priv = devm_kzalloc(&pdev->dev, sizeof(*mach_priv), GFP_KERNEL); + if (!mach_priv) + return -ENOMEM; + + soc_card_data->mach_priv = mach_priv; + + adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0); + if (adsp_node) { + struct mtk_sof_priv *sof_priv; + + sof_priv = devm_kzalloc(&pdev->dev, sizeof(*sof_priv), GFP_KERNEL); + if (!sof_priv) { + ret = -ENOMEM; + goto err_adsp_node; + } + sof_priv->conn_streams = g_sof_conn_streams; + sof_priv->num_streams = ARRAY_SIZE(g_sof_conn_streams); + sof_priv->sof_dai_link_fixup = mt8186_sof_dai_link_fixup; + soc_card_data->sof_priv = sof_priv; + card->probe = mtk_sof_card_probe; + card->late_probe = mtk_sof_card_late_probe; + if (!card->topology_shortname_created) { + snprintf(card->topology_shortname, 32, "sof-%s", card->name); + card->topology_shortname_created = true; + } + card->name = card->topology_shortname; + sof_on = 1; + } else { + dev_info(&pdev->dev, "Probe without adsp\n"); + } + + if (of_property_read_bool(pdev->dev.of_node, "mediatek,dai-link")) { + ret = mtk_sof_dailink_parse_of(card, pdev->dev.of_node, + "mediatek,dai-link", + mt8186_mt6366_rt1019_rt5682s_dai_links, + ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links)); + if (ret) { + dev_dbg(&pdev->dev, "Parse dai-link fail\n"); + goto err_adsp_node; + } + } else { + if (!sof_on) + card->num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links) + - ARRAY_SIZE(g_sof_conn_streams); + } + platform_node = of_parse_phandle(pdev->dev.of_node, "mediatek,platform", 0); if (!platform_node) { ret = -EINVAL; dev_err_probe(&pdev->dev, ret, "Property 'platform' missing or invalid\n"); - return ret; + goto err_platform_node; } playback_codec = of_get_child_by_name(pdev->dev.of_node, "playback-codecs"); @@ -917,17 +1059,14 @@ static int mt8186_mt6366_rt1019_rt5682s_dev_probe(struct platform_device *pdev) goto err_probe; } - if (!dai_link->platforms->name) - dai_link->platforms->of_node = platform_node; - } + if (!strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF")) && sof_on) + dai_link->platforms->of_node = adsp_node; - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) { - ret = -ENOMEM; - goto err_probe; + if (!dai_link->platforms->name && !dai_link->platforms->of_node) + dai_link->platforms->of_node = platform_node; } - snd_soc_card_set_drvdata(card, priv); + snd_soc_card_set_drvdata(card, soc_card_data); ret = mt8186_afe_gpio_init(&pdev->dev); if (ret) { @@ -945,6 +1084,9 @@ err_headset_codec: of_node_put(playback_codec); err_playback_codec: of_node_put(platform_node); +err_platform_node: +err_adsp_node: + of_node_put(adsp_node); return ret; } diff --git a/sound/soc/qcom/qdsp6/q6prm-clocks.c b/sound/soc/qcom/qdsp6/q6prm-clocks.c index a26cda5140c1..73b0cbac73d4 100644 --- a/sound/soc/qcom/qdsp6/q6prm-clocks.c +++ b/sound/soc/qcom/qdsp6/q6prm-clocks.c @@ -50,6 +50,15 @@ static const struct q6dsp_clk_init q6prm_clks[] = { Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_MCLK), Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_NPL_MCLK), Q6PRM_CLK(LPASS_CLK_ID_VA_CORE_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_TX_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_TX_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA_CORE_TX_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA_CORE_TX_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_TX_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_TX_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK), Q6DSP_VOTE_CLK(LPASS_HW_MACRO_VOTE, Q6PRM_HW_CORE_ID_LPASS, "LPASS_HW_MACRO"), Q6DSP_VOTE_CLK(LPASS_HW_DCODEC_VOTE, Q6PRM_HW_CORE_ID_DCODEC, diff --git a/sound/soc/qcom/qdsp6/q6prm.h b/sound/soc/qcom/qdsp6/q6prm.h index fea4d1954bc1..a988a32086fe 100644 --- a/sound/soc/qcom/qdsp6/q6prm.h +++ b/sound/soc/qcom/qdsp6/q6prm.h @@ -64,6 +64,25 @@ #define Q6PRM_LPASS_CLK_ID_RX_CORE_MCLK 0x30e #define Q6PRM_LPASS_CLK_ID_RX_CORE_NPL_MCLK 0x30f +/* Clock ID for MCLK for WSA2 core */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_MCLK 0x310 +/* Clock ID for NPL MCLK for WSA2 core */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_2X_MCLK 0x311 +/* Clock ID for RX Core TX MCLK */ +#define Q6PRM_LPASS_CLK_ID_RX_CORE_TX_MCLK 0x312 +/* Clock ID for RX CORE TX 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_RX_CORE_TX_2X_MCLK 0x313 +/* Clock ID for WSA core TX MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA_CORE_TX_MCLK 0x314 +/* Clock ID for WSA core TX 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA_CORE_TX_2X_MCLK 0x315 +/* Clock ID for WSA2 core TX MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_TX_MCLK 0x316 +/* Clock ID for WSA2 core TX 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_TX_2X_MCLK 0x317 +/* Clock ID for RX CORE MCLK2 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK 0x318 + #define Q6PRM_LPASS_CLK_SRC_INTERNAL 1 #define Q6PRM_LPASS_CLK_ROOT_DEFAULT 0 #define Q6PRM_HW_CORE_ID_LPASS 1 diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index e7d52d27132e..0fbbf3b02c09 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -1,7 +1,6 @@ // SPDX-License-Identifier: GPL-2.0+ #include <linux/extcon.h> #include <linux/iio/consumer.h> -#include <linux/iio/iio.h> #include <linux/input-event-codes.h> #include <linux/mfd/wm8994/registers.h> #include <linux/module.h> @@ -543,6 +542,7 @@ static int aries_audio_probe(struct platform_device *pdev) struct aries_wm8994_data *priv; struct snd_soc_dai_link *dai_link; const struct of_device_id *match; + enum iio_chan_type channel_type; int ret, i; if (!np) @@ -594,7 +594,11 @@ static int aries_audio_probe(struct platform_device *pdev) return dev_err_probe(dev, PTR_ERR(priv->adc), "Failed to get ADC channel"); - if (priv->adc->channel->type != IIO_VOLTAGE) + ret = iio_get_channel_type(priv->adc, &channel_type); + if (ret) + return dev_err_probe(dev, ret, + "Failed to get ADC channel type"); + if (channel_type != IIO_VOLTAGE) return -EINVAL; priv->gpio_headset_key = devm_gpiod_get(dev, "headset-key", diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 0d0594a0e4f6..7ace0c0db5b1 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1017,32 +1017,36 @@ static int rz_ssi_probe(struct platform_device *pdev) ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL); if (IS_ERR(ssi->rstc)) { - rz_ssi_release_dma_channels(ssi); - return PTR_ERR(ssi->rstc); + ret = PTR_ERR(ssi->rstc); + goto err_reset; } reset_control_deassert(ssi->rstc); pm_runtime_enable(&pdev->dev); ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); - return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n"); + dev_err(&pdev->dev, "pm_runtime_resume_and_get failed\n"); + goto err_pm; } ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component, rz_ssi_soc_dai, ARRAY_SIZE(rz_ssi_soc_dai)); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - - pm_runtime_put(ssi->dev); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); dev_err(&pdev->dev, "failed to register snd component\n"); + goto err_snd_soc; } + return 0; + +err_snd_soc: + pm_runtime_put(ssi->dev); +err_pm: + pm_runtime_disable(ssi->dev); + reset_control_assert(ssi->rstc); +err_reset: + rz_ssi_release_dma_channels(ssi); + return ret; } diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 5f49e3dec3fc..32c5be61e2ec 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -57,7 +57,7 @@ static inline struct snd_soc_component *gpio_to_component(struct gpio_chip *chip return gpio_priv->component; } -static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned offset) +static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned int offset) { if (offset >= AC97_NUM_GPIOS) return -EINVAL; @@ -66,7 +66,7 @@ static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned offset) } static int snd_soc_ac97_gpio_direction_in(struct gpio_chip *chip, - unsigned offset) + unsigned int offset) { struct snd_soc_component *component = gpio_to_component(chip); @@ -75,7 +75,7 @@ static int snd_soc_ac97_gpio_direction_in(struct gpio_chip *chip, 1 << offset, 1 << offset); } -static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) +static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned int offset) { struct snd_soc_component *component = gpio_to_component(chip); int ret; @@ -88,7 +88,7 @@ static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) return !!(ret & (1 << offset)); } -static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset, +static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned int offset, int value) { struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e824ff1a9fc0..e020ab49cfb1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -72,7 +72,7 @@ static ssize_t pmdown_time_show(struct device *dev, { struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); - return sprintf(buf, "%ld\n", rtd->pmdown_time); + return sysfs_emit(buf, "%ld\n", rtd->pmdown_time); } static ssize_t pmdown_time_store(struct device *dev, diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index d530e8c2b77b..49752af0e205 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -124,7 +124,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); */ int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) { - int ret = -EINVAL; + int ret = -ENOTSUPP; if (dai->driver->ops && dai->driver->ops->set_bclk_ratio) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b05231414c1d..73b8bd452ca7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2386,11 +2386,10 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, - char *buf) + char *buf, int count) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cmpnt); struct snd_soc_dapm_widget *w; - int count = 0; char *state = "not set"; /* card won't be set for the dummy component, as a spot fix @@ -2423,7 +2422,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, case snd_soc_dapm_pinctrl: case snd_soc_dapm_clock_supply: if (w->name) - count += sprintf(buf + count, "%s: %s\n", + count += sysfs_emit_at(buf, count, "%s: %s\n", w->name, w->power ? "On":"Off"); break; default: @@ -2445,7 +2444,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, state = "Off"; break; } - count += sprintf(buf + count, "PM State: %s\n", state); + count += sysfs_emit_at(buf, count, "PM State: %s\n", state); return count; } @@ -2463,7 +2462,7 @@ static ssize_t dapm_widget_show(struct device *dev, for_each_rtd_codec_dais(rtd, i, codec_dai) { struct snd_soc_component *cmpnt = codec_dai->component; - count += dapm_widget_show_component(cmpnt, buf + count); + count = dapm_widget_show_component(cmpnt, buf, count); } mutex_unlock(&rtd->card->dapm_mutex); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5b99bf2dbd08..b5720e272c51 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -723,7 +723,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); snd_soc_dpcm_mutex_lock(rtd); - soc_pcm_clean(rtd, substream, 0); + __soc_pcm_close(rtd, substream); snd_soc_dpcm_mutex_unlock(rtd); return 0; } @@ -1317,6 +1317,9 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; + if (!snd_soc_dpcm_get_substream(be, stream)) + continue; + for_each_rtd_dais(be, i, dai) { w = snd_soc_dai_get_widget(dai, stream); @@ -2904,6 +2907,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = pcm; pcm->nonatomic = rtd->dai_link->nonatomic; pcm->private_data = rtd; + pcm->no_device_suspend = true; if (rtd->dai_link->no_pcm || rtd->dai_link->params) { if (playback) @@ -2958,8 +2962,6 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) ret = snd_soc_pcm_component_new(rtd); if (ret < 0) return ret; - - pcm->no_device_suspend = true; out: dev_dbg(rtd->card->dev, "%s <-> %s mapping ok\n", soc_codec_dai_name(rtd), soc_cpu_dai_name(rtd)); diff --git a/sound/soc/soc-utils-test.c b/sound/soc/soc-utils-test.c index 5ad8e23af49a..616d2c926dd1 100644 --- a/sound/soc/soc-utils-test.c +++ b/sound/soc/soc-utils-test.c @@ -170,8 +170,54 @@ static void test_tdm_params_to_bclk(struct kunit *test) } } +static void test_snd_soc_params_to_bclk_one(struct kunit *test, + unsigned int rate, snd_pcm_format_t fmt, + unsigned int channels, + unsigned int expected_bclk) +{ + struct snd_pcm_hw_params params; + int got_bclk; + + _snd_pcm_hw_params_any(¶ms); + snd_mask_none(hw_param_mask(¶ms, SNDRV_PCM_HW_PARAM_FORMAT)); + hw_param_interval(¶ms, SNDRV_PCM_HW_PARAM_RATE)->min = rate; + hw_param_interval(¶ms, SNDRV_PCM_HW_PARAM_RATE)->max = rate; + hw_param_interval(¶ms, SNDRV_PCM_HW_PARAM_CHANNELS)->min = channels; + hw_param_interval(¶ms, SNDRV_PCM_HW_PARAM_CHANNELS)->max = channels; + params_set_format(¶ms, fmt); + + got_bclk = snd_soc_params_to_bclk(¶ms); + pr_debug("%s: r=%u sb=%u ch=%u expected=%u got=%d\n", + __func__, + rate, params_width(¶ms), channels, expected_bclk, got_bclk); + KUNIT_ASSERT_EQ(test, expected_bclk, (unsigned int)got_bclk); +} + +static void test_snd_soc_params_to_bclk(struct kunit *test) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(tdm_params_to_bclk_cases); ++i) { + /* + * snd_soc_params_to_bclk() is all the test cases where + * snd_pcm_hw_params values are not overridden. + */ + if (tdm_params_to_bclk_cases[i].tdm_width | + tdm_params_to_bclk_cases[i].tdm_slots | + tdm_params_to_bclk_cases[i].slot_multiple) + continue; + + test_snd_soc_params_to_bclk_one(test, + tdm_params_to_bclk_cases[i].rate, + tdm_params_to_bclk_cases[i].fmt, + tdm_params_to_bclk_cases[i].channels, + tdm_params_to_bclk_cases[i].bclk); + } +} + static struct kunit_case soc_utils_test_cases[] = { KUNIT_CASE(test_tdm_params_to_bclk), + KUNIT_CASE(test_snd_soc_params_to_bclk), {} }; diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 70c380c0ac7b..a3b6df2378b4 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -56,23 +56,24 @@ EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk); /** * snd_soc_tdm_params_to_bclk - calculate bclk from params and tdm slot info. * - * Calculate the bclk from the params sample rate and the tdm slot count and - * tdm slot width. Either or both of tdm_width and tdm_slots can be 0. + * Calculate the bclk from the params sample rate, the tdm slot count and the + * tdm slot width. Optionally round-up the slot count to a given multiple. + * Either or both of tdm_width and tdm_slots can be 0. * - * If tdm_width == 0 and tdm_slots > 0: the params_width will be used. - * If tdm_width > 0 and tdm_slots == 0: the params_channels will be used - * as the slot count. - * Both tdm_width and tdm_slots are 0: this is equivalent to calling - * snd_soc_params_to_bclk(). + * If tdm_width == 0: use params_width() as the slot width. + * If tdm_slots == 0: use params_channels() as the slot count. * - * If slot_multiple > 1 the slot count (or params_channels if tdm_slots == 0) - * will be rounded up to a multiple of this value. This is mainly useful for + * If slot_multiple > 1 the slot count (or params_channels() if tdm_slots == 0) + * will be rounded up to a multiple of slot_multiple. This is mainly useful for * I2S mode, which has a left and right phase so the number of slots is always * a multiple of 2. * + * If tdm_width == 0 && tdm_slots == 0 && slot_multiple < 2, this is equivalent + * to calling snd_soc_params_to_bclk(). + * * @params: Pointer to struct_pcm_hw_params. - * @tdm_width: Width in bits of the tdm slots. - * @tdm_slots: Number of tdm slots per frame. + * @tdm_width: Width in bits of the tdm slots. Must be >= 0. + * @tdm_slots: Number of tdm slots per frame. Must be >= 0. * @slot_multiple: If >1 roundup slot count to a multiple of this value. * * Return: bclk frequency in Hz, else a negative error code if params format diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index 67139e15f862..c8ae778a50d1 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -11,20 +11,20 @@ #include "sof-priv.h" #include "sof-utils.h" -static void sof_set_transferred_bytes(struct snd_compr_tstamp *tstamp, +static void sof_set_transferred_bytes(struct sof_compr_stream *sstream, u64 host_pos, u64 buffer_size) { u64 prev_pos; unsigned int copied; - div64_u64_rem(tstamp->copied_total, buffer_size, &prev_pos); + div64_u64_rem(sstream->copied_total, buffer_size, &prev_pos); if (host_pos < prev_pos) copied = (buffer_size - prev_pos) + host_pos; else copied = host_pos - prev_pos; - tstamp->copied_total += copied; + sstream->copied_total += copied; } static void snd_sof_compr_fragment_elapsed_work(struct work_struct *work) @@ -49,7 +49,7 @@ void snd_sof_compr_fragment_elapsed(struct snd_compr_stream *cstream) struct snd_soc_pcm_runtime *rtd; struct snd_compr_runtime *crtd; struct snd_soc_component *component; - struct snd_compr_tstamp *tstamp; + struct sof_compr_stream *sstream; struct snd_sof_pcm *spcm; if (!cstream) @@ -57,7 +57,7 @@ void snd_sof_compr_fragment_elapsed(struct snd_compr_stream *cstream) rtd = cstream->private_data; crtd = cstream->runtime; - tstamp = crtd->private_data; + sstream = crtd->private_data; component = snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME); spcm = snd_sof_find_spcm_dai(component, rtd); @@ -67,7 +67,7 @@ void snd_sof_compr_fragment_elapsed(struct snd_compr_stream *cstream) return; } - sof_set_transferred_bytes(tstamp, spcm->stream[cstream->direction].posn.host_posn, + sof_set_transferred_bytes(sstream, spcm->stream[cstream->direction].posn.host_posn, crtd->buffer_size); /* use the same workqueue-based solution as for PCM, cf. snd_sof_pcm_elapsed */ @@ -96,24 +96,24 @@ static int sof_compr_open(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_compr_runtime *crtd = cstream->runtime; - struct snd_compr_tstamp *tstamp; + struct sof_compr_stream *sstream; struct snd_sof_pcm *spcm; int dir; - tstamp = kzalloc(sizeof(*tstamp), GFP_KERNEL); - if (!tstamp) + sstream = kzalloc(sizeof(*sstream), GFP_KERNEL); + if (!sstream) return -ENOMEM; spcm = snd_sof_find_spcm_dai(component, rtd); if (!spcm) { - kfree(tstamp); + kfree(sstream); return -EINVAL; } dir = cstream->direction; if (spcm->stream[dir].cstream) { - kfree(tstamp); + kfree(sstream); return -EBUSY; } @@ -122,7 +122,7 @@ static int sof_compr_open(struct snd_soc_component *component, spcm->stream[dir].posn.dai_posn = 0; spcm->prepared[dir] = false; - crtd->private_data = tstamp; + crtd->private_data = sstream; return 0; } @@ -131,7 +131,7 @@ static int sof_compr_free(struct snd_soc_component *component, struct snd_compr_stream *cstream) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); - struct snd_compr_tstamp *tstamp = cstream->runtime->private_data; + struct sof_compr_stream *sstream = cstream->runtime->private_data; struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct sof_ipc_stream stream; struct sof_ipc_reply reply; @@ -155,7 +155,7 @@ static int sof_compr_free(struct snd_soc_component *component, cancel_work_sync(&spcm->stream[cstream->direction].period_elapsed_work); spcm->stream[cstream->direction].cstream = NULL; - kfree(tstamp); + kfree(sstream); return ret; } @@ -169,7 +169,7 @@ static int sof_compr_set_params(struct snd_soc_component *component, struct sof_ipc_pcm_params_reply ipc_params_reply; struct sof_ipc_fw_ready *ready = &sdev->fw_ready; struct sof_ipc_fw_version *v = &ready->version; - struct snd_compr_tstamp *tstamp; + struct sof_compr_stream *sstream; struct sof_ipc_pcm_params *pcm; struct snd_sof_pcm *spcm; size_t ext_data_size; @@ -184,7 +184,7 @@ static int sof_compr_set_params(struct snd_soc_component *component, return -EINVAL; } - tstamp = crtd->private_data; + sstream = crtd->private_data; spcm = snd_sof_find_spcm_dai(component, rtd); @@ -237,8 +237,9 @@ static int sof_compr_set_params(struct snd_soc_component *component, goto out; } - tstamp->byte_offset = sdev->stream_box.offset + ipc_params_reply.posn_offset; - tstamp->sampling_rate = params->codec.sample_rate; + sstream->sampling_rate = params->codec.sample_rate; + sstream->channels = params->codec.ch_out; + sstream->sample_container_bytes = pcm->params.sample_container_bytes; spcm->prepared[cstream->direction] = true; @@ -326,10 +327,18 @@ static int sof_compr_pointer(struct snd_soc_component *component, struct snd_compr_stream *cstream, struct snd_compr_tstamp *tstamp) { - struct snd_compr_tstamp *pstamp = cstream->runtime->private_data; + struct snd_sof_pcm *spcm; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct sof_compr_stream *sstream = cstream->runtime->private_data; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; - tstamp->sampling_rate = pstamp->sampling_rate; - tstamp->copied_total = pstamp->copied_total; + tstamp->sampling_rate = sstream->sampling_rate; + tstamp->copied_total = sstream->copied_total; + tstamp->pcm_io_frames = div_u64(spcm->stream[cstream->direction].posn.dai_posn, + sstream->channels * sstream->sample_container_bytes); return 0; } diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index c5d797e97c02..d9a3ce7b69e1 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -252,9 +252,9 @@ static int memory_info_update(struct snd_sof_dev *sdev, char *buf, size_t buff_s } for (i = 0, len = 0; i < reply->num_elems; i++) { - ret = snprintf(buf + len, buff_size - len, "zone %d.%d used %#8x free %#8x\n", - reply->elems[i].zone, reply->elems[i].id, - reply->elems[i].used, reply->elems[i].free); + ret = scnprintf(buf + len, buff_size - len, "zone %d.%d used %#8x free %#8x\n", + reply->elems[i].zone, reply->elems[i].id, + reply->elems[i].used, reply->elems[i].free); if (ret < 0) goto error; len += ret; diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index cc6e695f913a..4751b04d5e6f 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -41,4 +41,13 @@ config SND_SOC_SOF_IMX8M Say Y if you have such a device. If unsure select "N". +config SND_SOC_SOF_IMX8ULP + tristate "SOF support for i.MX8ULP" + depends on IMX_DSP + select SND_SOC_SOF_IMX_COMMON + help + This adds support for Sound Open Firmware for NXP i.MX8ULP platforms. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_SOF_IMX_TOPLEVEL diff --git a/sound/soc/sof/imx/Makefile b/sound/soc/sof/imx/Makefile index dba93c3466ec..798b43a415bf 100644 --- a/sound/soc/sof/imx/Makefile +++ b/sound/soc/sof/imx/Makefile @@ -1,9 +1,11 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) snd-sof-imx8-objs := imx8.o snd-sof-imx8m-objs := imx8m.o +snd-sof-imx8ulp-objs := imx8ulp.o snd-sof-imx-common-objs := imx-common.o obj-$(CONFIG_SND_SOC_SOF_IMX8) += snd-sof-imx8.o obj-$(CONFIG_SND_SOC_SOF_IMX8M) += snd-sof-imx8m.o +obj-$(CONFIG_SND_SOC_SOF_IMX8ULP) += snd-sof-imx8ulp.o obj-$(CONFIG_SND_SOC_SOF_IMX_COMMON) += imx-common.o diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c new file mode 100644 index 000000000000..4a562c9856e9 --- /dev/null +++ b/sound/soc/sof/imx/imx8ulp.c @@ -0,0 +1,515 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// Copyright 2021-2022 NXP +// +// Author: Peng Zhang <peng.zhang_8@nxp.com> +// +// Hardware interface for audio DSP on i.MX8ULP + +#include <linux/arm-smccc.h> +#include <linux/clk.h> +#include <linux/firmware.h> +#include <linux/firmware/imx/dsp.h> +#include <linux/firmware/imx/ipc.h> +#include <linux/firmware/imx/svc/misc.h> +#include <linux/mfd/syscon.h> +#include <linux/module.h> +#include <linux/of_address.h> +#include <linux/of_irq.h> +#include <linux/of_platform.h> +#include <linux/of_reserved_mem.h> + +#include <sound/sof.h> +#include <sound/sof/xtensa.h> + +#include "../ops.h" +#include "../sof-of-dev.h" +#include "imx-common.h" + +#define FSL_SIP_HIFI_XRDC 0xc200000e + +/* SIM Domain register */ +#define SYSCTRL0 0x8 +#define EXECUTE_BIT BIT(13) +#define RESET_BIT BIT(16) +#define HIFI4_CLK_BIT BIT(17) +#define PB_CLK_BIT BIT(18) +#define PLAT_CLK_BIT BIT(19) +#define DEBUG_LOGIC_BIT BIT(25) + +#define MBOX_OFFSET 0x800000 +#define MBOX_SIZE 0x1000 + +static struct clk_bulk_data imx8ulp_dsp_clks[] = { + { .id = "core" }, + { .id = "ipg" }, + { .id = "ocram" }, + { .id = "mu" }, +}; + +struct imx8ulp_priv { + struct device *dev; + struct snd_sof_dev *sdev; + + /* DSP IPC handler */ + struct imx_dsp_ipc *dsp_ipc; + struct platform_device *ipc_dev; + + struct regmap *regmap; + struct imx_clocks *clks; +}; + +static void imx8ulp_sim_lpav_start(struct imx8ulp_priv *priv) +{ + /* Controls the HiFi4 DSP Reset: 1 in reset, 0 out of reset */ + regmap_update_bits(priv->regmap, SYSCTRL0, RESET_BIT, 0); + + /* Reset HiFi4 DSP Debug logic: 1 debug reset, 0 out of reset*/ + regmap_update_bits(priv->regmap, SYSCTRL0, DEBUG_LOGIC_BIT, 0); + + /* Stall HIFI4 DSP Execution: 1 stall, 0 run */ + regmap_update_bits(priv->regmap, SYSCTRL0, EXECUTE_BIT, 0); +} + +static int imx8ulp_get_mailbox_offset(struct snd_sof_dev *sdev) +{ + return MBOX_OFFSET; +} + +static int imx8ulp_get_window_offset(struct snd_sof_dev *sdev, u32 id) +{ + return MBOX_OFFSET; +} + +static void imx8ulp_dsp_handle_reply(struct imx_dsp_ipc *ipc) +{ + struct imx8ulp_priv *priv = imx_dsp_get_data(ipc); + unsigned long flags; + + spin_lock_irqsave(&priv->sdev->ipc_lock, flags); + + snd_sof_ipc_process_reply(priv->sdev, 0); + + spin_unlock_irqrestore(&priv->sdev->ipc_lock, flags); +} + +static void imx8ulp_dsp_handle_request(struct imx_dsp_ipc *ipc) +{ + struct imx8ulp_priv *priv = imx_dsp_get_data(ipc); + u32 p; /* panic code */ + + /* Read the message from the debug box. */ + sof_mailbox_read(priv->sdev, priv->sdev->debug_box.offset + 4, &p, sizeof(p)); + + /* Check to see if the message is a panic code (0x0dead***) */ + if ((p & SOF_IPC_PANIC_MAGIC_MASK) == SOF_IPC_PANIC_MAGIC) + snd_sof_dsp_panic(priv->sdev, p, true); + else + snd_sof_ipc_msgs_rx(priv->sdev); +} + +static struct imx_dsp_ops dsp_ops = { + .handle_reply = imx8ulp_dsp_handle_reply, + .handle_request = imx8ulp_dsp_handle_request, +}; + +static int imx8ulp_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) +{ + struct imx8ulp_priv *priv = sdev->pdata->hw_pdata; + + sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data, + msg->msg_size); + imx_dsp_ring_doorbell(priv->dsp_ipc, 0); + + return 0; +} + +static int imx8ulp_run(struct snd_sof_dev *sdev) +{ + struct imx8ulp_priv *priv = sdev->pdata->hw_pdata; + + imx8ulp_sim_lpav_start(priv); + + return 0; +} + +static int imx8ulp_reset(struct snd_sof_dev *sdev) +{ + struct imx8ulp_priv *priv = sdev->pdata->hw_pdata; + struct arm_smccc_res smc_resource; + + /* HiFi4 Platform Clock Enable: 1 enabled, 0 disabled */ + regmap_update_bits(priv->regmap, SYSCTRL0, PLAT_CLK_BIT, PLAT_CLK_BIT); + + /* HiFi4 PBCLK clock enable: 1 enabled, 0 disabled */ + regmap_update_bits(priv->regmap, SYSCTRL0, PB_CLK_BIT, PB_CLK_BIT); + + /* HiFi4 Clock Enable: 1 enabled, 0 disabled */ + regmap_update_bits(priv->regmap, SYSCTRL0, HIFI4_CLK_BIT, HIFI4_CLK_BIT); + + regmap_update_bits(priv->regmap, SYSCTRL0, RESET_BIT, RESET_BIT); + usleep_range(1, 2); + + /* Stall HIFI4 DSP Execution: 1 stall, 0 not stall */ + regmap_update_bits(priv->regmap, SYSCTRL0, EXECUTE_BIT, EXECUTE_BIT); + usleep_range(1, 2); + + arm_smccc_smc(FSL_SIP_HIFI_XRDC, 0, 0, 0, 0, 0, 0, 0, &smc_resource); + + return 0; +} + +static int imx8ulp_probe(struct snd_sof_dev *sdev) +{ + struct platform_device *pdev = + container_of(sdev->dev, struct platform_device, dev); + struct device_node *np = pdev->dev.of_node; + struct device_node *res_node; + struct resource *mmio; + struct imx8ulp_priv *priv; + struct resource res; + u32 base, size; + int ret = 0; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->clks = devm_kzalloc(&pdev->dev, sizeof(*priv->clks), GFP_KERNEL); + if (!priv->clks) + return -ENOMEM; + + sdev->num_cores = 1; + sdev->pdata->hw_pdata = priv; + priv->dev = sdev->dev; + priv->sdev = sdev; + + /* System integration module(SIM) control dsp configuration */ + priv->regmap = syscon_regmap_lookup_by_phandle(np, "fsl,dsp-ctrl"); + if (IS_ERR(priv->regmap)) + return PTR_ERR(priv->regmap); + + priv->ipc_dev = platform_device_register_data(sdev->dev, "imx-dsp", + PLATFORM_DEVID_NONE, + pdev, sizeof(*pdev)); + if (IS_ERR(priv->ipc_dev)) + return PTR_ERR(priv->ipc_dev); + + priv->dsp_ipc = dev_get_drvdata(&priv->ipc_dev->dev); + if (!priv->dsp_ipc) { + /* DSP IPC driver not probed yet, try later */ + ret = -EPROBE_DEFER; + dev_err(sdev->dev, "Failed to get drvdata\n"); + goto exit_pdev_unregister; + } + + imx_dsp_set_data(priv->dsp_ipc, priv); + priv->dsp_ipc->ops = &dsp_ops; + + /* DSP base */ + mmio = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (mmio) { + base = mmio->start; + size = resource_size(mmio); + } else { + dev_err(sdev->dev, "error: failed to get DSP base at idx 0\n"); + ret = -EINVAL; + goto exit_pdev_unregister; + } + + sdev->bar[SOF_FW_BLK_TYPE_IRAM] = devm_ioremap(sdev->dev, base, size); + if (!sdev->bar[SOF_FW_BLK_TYPE_IRAM]) { + dev_err(sdev->dev, "failed to ioremap base 0x%x size 0x%x\n", + base, size); + ret = -ENODEV; + goto exit_pdev_unregister; + } + sdev->mmio_bar = SOF_FW_BLK_TYPE_IRAM; + + res_node = of_parse_phandle(np, "memory-reserved", 0); + if (!res_node) { + dev_err(&pdev->dev, "failed to get memory region node\n"); + ret = -ENODEV; + goto exit_pdev_unregister; + } + + ret = of_address_to_resource(res_node, 0, &res); + of_node_put(res_node); + if (ret) { + dev_err(&pdev->dev, "failed to get reserved region address\n"); + goto exit_pdev_unregister; + } + + sdev->bar[SOF_FW_BLK_TYPE_SRAM] = devm_ioremap_wc(sdev->dev, res.start, + resource_size(&res)); + if (!sdev->bar[SOF_FW_BLK_TYPE_SRAM]) { + dev_err(sdev->dev, "failed to ioremap mem 0x%x size 0x%x\n", + base, size); + ret = -ENOMEM; + goto exit_pdev_unregister; + } + sdev->mailbox_bar = SOF_FW_BLK_TYPE_SRAM; + + /* set default mailbox offset for FW ready message */ + sdev->dsp_box.offset = MBOX_OFFSET; + + ret = of_reserved_mem_device_init(sdev->dev); + if (ret) { + dev_err(&pdev->dev, "failed to init reserved memory region %d\n", ret); + goto exit_pdev_unregister; + } + + priv->clks->dsp_clks = imx8ulp_dsp_clks; + priv->clks->num_dsp_clks = ARRAY_SIZE(imx8ulp_dsp_clks); + + ret = imx8_parse_clocks(sdev, priv->clks); + if (ret < 0) + goto exit_pdev_unregister; + + ret = imx8_enable_clocks(sdev, priv->clks); + if (ret < 0) + goto exit_pdev_unregister; + + return 0; + +exit_pdev_unregister: + platform_device_unregister(priv->ipc_dev); + + return ret; +} + +static int imx8ulp_remove(struct snd_sof_dev *sdev) +{ + struct imx8ulp_priv *priv = sdev->pdata->hw_pdata; + + imx8_disable_clocks(sdev, priv->clks); + platform_device_unregister(priv->ipc_dev); + + return 0; +} + +/* on i.MX8 there is 1 to 1 match between type and BAR idx */ +static int imx8ulp_get_bar_index(struct snd_sof_dev *sdev, u32 type) +{ + return type; +} + +static int imx8ulp_suspend(struct snd_sof_dev *sdev) +{ + int i; + struct imx8ulp_priv *priv = (struct imx8ulp_priv *)sdev->pdata->hw_pdata; + + /*Stall DSP, release in .run() */ + regmap_update_bits(priv->regmap, SYSCTRL0, EXECUTE_BIT, EXECUTE_BIT); + + for (i = 0; i < DSP_MU_CHAN_NUM; i++) + imx_dsp_free_channel(priv->dsp_ipc, i); + + imx8_disable_clocks(sdev, priv->clks); + + return 0; +} + +static int imx8ulp_resume(struct snd_sof_dev *sdev) +{ + struct imx8ulp_priv *priv = (struct imx8ulp_priv *)sdev->pdata->hw_pdata; + int i; + + imx8_enable_clocks(sdev, priv->clks); + + for (i = 0; i < DSP_MU_CHAN_NUM; i++) + imx_dsp_request_channel(priv->dsp_ipc, i); + + return 0; +} + +static int imx8ulp_dsp_runtime_resume(struct snd_sof_dev *sdev) +{ + const struct sof_dsp_power_state target_dsp_state = { + .state = SOF_DSP_PM_D0, + .substate = 0, + }; + + imx8ulp_resume(sdev); + + return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); +} + +static int imx8ulp_dsp_runtime_suspend(struct snd_sof_dev *sdev) +{ + const struct sof_dsp_power_state target_dsp_state = { + .state = SOF_DSP_PM_D3, + .substate = 0, + }; + + imx8ulp_suspend(sdev); + + return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); +} + +static int imx8ulp_dsp_suspend(struct snd_sof_dev *sdev, unsigned int target_state) +{ + const struct sof_dsp_power_state target_dsp_state = { + .state = target_state, + .substate = 0, + }; + + if (!pm_runtime_suspended(sdev->dev)) + imx8ulp_suspend(sdev); + + return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); +} + +static int imx8ulp_dsp_resume(struct snd_sof_dev *sdev) +{ + const struct sof_dsp_power_state target_dsp_state = { + .state = SOF_DSP_PM_D0, + .substate = 0, + }; + + imx8ulp_resume(sdev); + + if (pm_runtime_suspended(sdev->dev)) { + pm_runtime_disable(sdev->dev); + pm_runtime_set_active(sdev->dev); + pm_runtime_mark_last_busy(sdev->dev); + pm_runtime_enable(sdev->dev); + pm_runtime_idle(sdev->dev); + } + + return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); +} + +static struct snd_soc_dai_driver imx8ulp_dai[] = { + { + .name = "sai5", + .playback = { + .channels_min = 1, + .channels_max = 32, + }, + .capture = { + .channels_min = 1, + .channels_max = 32, + }, + }, + { + .name = "sai6", + .playback = { + .channels_min = 1, + .channels_max = 32, + }, + .capture = { + .channels_min = 1, + .channels_max = 32, + }, + }, +}; + +static int imx8ulp_dsp_set_power_state(struct snd_sof_dev *sdev, + const struct sof_dsp_power_state *target_state) +{ + sdev->dsp_power_state = *target_state; + + return 0; +} + +/* i.MX8 ops */ +static struct snd_sof_dsp_ops sof_imx8ulp_ops = { + /* probe and remove */ + .probe = imx8ulp_probe, + .remove = imx8ulp_remove, + /* DSP core boot */ + .run = imx8ulp_run, + .reset = imx8ulp_reset, + + /* Block IO */ + .block_read = sof_block_read, + .block_write = sof_block_write, + + /* Module IO */ + .read64 = sof_io_read64, + + /* Mailbox IO */ + .mailbox_read = sof_mailbox_read, + .mailbox_write = sof_mailbox_write, + + /* ipc */ + .send_msg = imx8ulp_send_msg, + .get_mailbox_offset = imx8ulp_get_mailbox_offset, + .get_window_offset = imx8ulp_get_window_offset, + + .ipc_msg_data = sof_ipc_msg_data, + .set_stream_data_offset = sof_set_stream_data_offset, + + /* stream callbacks */ + .pcm_open = sof_stream_pcm_open, + .pcm_close = sof_stream_pcm_close, + + /* module loading */ + .get_bar_index = imx8ulp_get_bar_index, + /* firmware loading */ + .load_firmware = snd_sof_load_firmware_memcpy, + + /* Debug information */ + .dbg_dump = imx8_dump, + + /* Firmware ops */ + .dsp_arch_ops = &sof_xtensa_arch_ops, + + /* DAI drivers */ + .drv = imx8ulp_dai, + .num_drv = ARRAY_SIZE(imx8ulp_dai), + + /* ALSA HW info flags */ + .hw_info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, + + /* PM */ + .runtime_suspend = imx8ulp_dsp_runtime_suspend, + .runtime_resume = imx8ulp_dsp_runtime_resume, + + .suspend = imx8ulp_dsp_suspend, + .resume = imx8ulp_dsp_resume, + + .set_power_state = imx8ulp_dsp_set_power_state, +}; + +static struct sof_dev_desc sof_of_imx8ulp_desc = { + .ipc_supported_mask = BIT(SOF_IPC), + .ipc_default = SOF_IPC, + .default_fw_path = { + [SOF_IPC] = "imx/sof", + }, + .default_tplg_path = { + [SOF_IPC] = "imx/sof-tplg", + }, + .default_fw_filename = { + [SOF_IPC] = "sof-imx8ulp.ri", + }, + .nocodec_tplg_filename = "sof-imx8ulp-nocodec.tplg", + .ops = &sof_imx8ulp_ops, +}; + +static const struct of_device_id sof_of_imx8ulp_ids[] = { + { .compatible = "fsl,imx8ulp-dsp", .data = &sof_of_imx8ulp_desc}, + { } +}; +MODULE_DEVICE_TABLE(of, sof_of_imx8ulp_ids); + +/* DT driver definition */ +static struct platform_driver snd_sof_of_imx8ulp_driver = { + .probe = sof_of_probe, + .remove = sof_of_remove, + .driver = { + .name = "sof-audio-of-imx8ulp", + .pm = &sof_of_pm, + .of_match_table = sof_of_imx8ulp_ids, + }, +}; +module_platform_driver(snd_sof_of_imx8ulp_driver); + +MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); +MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index eddfd77ad90f..671c3e02d7df 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -620,8 +620,13 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) /* * The memory used for IMR boot loses its content in deeper than S3 state * We must not try IMR boot on next power up (as it will fail). + * + * In case of firmware crash or boot failure set the skip_imr_boot to true + * as well in order to try to re-load the firmware to do a 'cold' boot. */ - if (sdev->system_suspend_target > SOF_SUSPEND_S3) + if (sdev->system_suspend_target > SOF_SUSPEND_S3 || + sdev->fw_state == SOF_FW_CRASHED || + sdev->fw_state == SOF_FW_BOOT_FAILED) hda->skip_imr_boot = true; hda_sdw_int_enable(sdev, false); diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index eb22eb3f6fee..98812d51b31c 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -177,14 +177,13 @@ int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) * - IMR boot: wait for ROM firmware entered (firmware booted up from IMR) */ if (imr_boot) - target_status = HDA_DSP_ROM_FW_ENTERED; + target_status = FSR_STATE_FW_ENTERED; else - target_status = HDA_DSP_ROM_INIT; + target_status = FSR_STATE_INIT_DONE; ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, chip->rom_status_reg, status, - ((status & HDA_DSP_ROM_STS_MASK) - == target_status), + (FSR_TO_STATE_CODE(status) == target_status), HDA_DSP_REG_POLL_INTERVAL_US, chip->rom_init_timeout * USEC_PER_MSEC); @@ -292,8 +291,7 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream status = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, chip->rom_status_reg, reg, - ((reg & HDA_DSP_ROM_STS_MASK) - == HDA_DSP_ROM_FW_ENTERED), + (FSR_TO_STATE_CODE(reg) == FSR_STATE_FW_ENTERED), HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_BASEFW_TIMEOUT_US); diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 8639ea63a10d..6d4ecbe14adf 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -574,7 +574,7 @@ static void hda_dsp_dump_ext_rom_status(struct snd_sof_dev *sdev, const char *le chip = get_chip_info(sdev->pdata); for (i = 0; i < HDA_EXT_ROM_STATUS_SIZE; i++) { value = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg + i * 0x4); - len += snprintf(msg + len, sizeof(msg) - len, " 0x%x", value); + len += scnprintf(msg + len, sizeof(msg) - len, " 0x%x", value); } dev_printk(level, sdev->dev, "extended rom status: %s", msg); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 5ef3e8775e36..ba6feb1b0d3b 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -251,12 +251,6 @@ #define FSR_STATE_BRINGUP_FW_ENTERED FSR_STATE_FW_ENTERED /* ROM status/error values */ -#define HDA_DSP_ROM_STS_MASK GENMASK(23, 0) -#define HDA_DSP_ROM_INIT 0x1 -#define HDA_DSP_ROM_FW_MANIFEST_LOADED 0x3 -#define HDA_DSP_ROM_FW_FW_LOADED 0x4 -#define HDA_DSP_ROM_FW_ENTERED 0x5 -#define HDA_DSP_ROM_RFW_START 0xf #define HDA_DSP_ROM_CSE_ERROR 40 #define HDA_DSP_ROM_CSE_WRONG_RESPONSE 41 #define HDA_DSP_ROM_IMR_TO_SMALL 42 diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index ccc44ba3ad94..aac47cd007e8 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -159,6 +159,62 @@ static const struct sof_dev_desc adl_desc = { .ops_init = sof_tgl_ops_init, }; +static const struct sof_dev_desc rpls_desc = { + .machines = snd_soc_acpi_intel_rpl_machines, + .alt_machines = snd_soc_acpi_intel_rpl_sdw_machines, + .use_acpi_target_states = true, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .chip_info = &adls_chip_info, + .ipc_supported_mask = BIT(SOF_IPC) | BIT(SOF_INTEL_IPC4), + .ipc_default = SOF_IPC, + .default_fw_path = { + [SOF_IPC] = "intel/sof", + [SOF_INTEL_IPC4] = "intel/avs/rpl-s", + }, + .default_tplg_path = { + [SOF_IPC] = "intel/sof-tplg", + [SOF_INTEL_IPC4] = "intel/avs-tplg", + }, + .default_fw_filename = { + [SOF_IPC] = "sof-rpl-s.ri", + [SOF_INTEL_IPC4] = "dsp_basefw.bin", + }, + .nocodec_tplg_filename = "sof-rpl-nocodec.tplg", + .ops = &sof_tgl_ops, + .ops_init = sof_tgl_ops_init, +}; + +static const struct sof_dev_desc rpl_desc = { + .machines = snd_soc_acpi_intel_rpl_machines, + .alt_machines = snd_soc_acpi_intel_rpl_sdw_machines, + .use_acpi_target_states = true, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .chip_info = &tgl_chip_info, + .ipc_supported_mask = BIT(SOF_IPC) | BIT(SOF_INTEL_IPC4), + .ipc_default = SOF_IPC, + .default_fw_path = { + [SOF_IPC] = "intel/sof", + [SOF_INTEL_IPC4] = "intel/avs/rpl", + }, + .default_tplg_path = { + [SOF_IPC] = "intel/sof-tplg", + [SOF_INTEL_IPC4] = "intel/avs-tplg", + }, + .default_fw_filename = { + [SOF_IPC] = "sof-rpl.ri", + [SOF_INTEL_IPC4] = "dsp_basefw.bin", + }, + .nocodec_tplg_filename = "sof-rpl-nocodec.tplg", + .ops = &sof_tgl_ops, + .ops_init = sof_tgl_ops_init, +}; + /* PCI IDs */ static const struct pci_device_id sof_pci_ids[] = { { PCI_DEVICE(0x8086, 0xa0c8), /* TGL-LP */ @@ -172,7 +228,7 @@ static const struct pci_device_id sof_pci_ids[] = { { PCI_DEVICE(0x8086, 0x7ad0), /* ADL-S */ .driver_data = (unsigned long)&adls_desc}, { PCI_DEVICE(0x8086, 0x7a50), /* RPL-S */ - .driver_data = (unsigned long)&adls_desc}, + .driver_data = (unsigned long)&rpls_desc}, { PCI_DEVICE(0x8086, 0x51c8), /* ADL-P */ .driver_data = (unsigned long)&adl_desc}, { PCI_DEVICE(0x8086, 0x51cd), /* ADL-P */ @@ -180,9 +236,9 @@ static const struct pci_device_id sof_pci_ids[] = { { PCI_DEVICE(0x8086, 0x51c9), /* ADL-PS */ .driver_data = (unsigned long)&adl_desc}, { PCI_DEVICE(0x8086, 0x51ca), /* RPL-P */ - .driver_data = (unsigned long)&adl_desc}, + .driver_data = (unsigned long)&rpl_desc}, { PCI_DEVICE(0x8086, 0x51cb), /* RPL-P */ - .driver_data = (unsigned long)&adl_desc}, + .driver_data = (unsigned long)&rpl_desc}, { PCI_DEVICE(0x8086, 0x51cc), /* ADL-M */ .driver_data = (unsigned long)&adl_desc}, { PCI_DEVICE(0x8086, 0x54c8), /* ADL-N */ 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); diff --git a/sound/soc/sof/ipc4-loader.c b/sound/soc/sof/ipc4-loader.c index 9fadae8fd011..8bd2132b4f41 100644 --- a/sound/soc/sof/ipc4-loader.c +++ b/sound/soc/sof/ipc4-loader.c @@ -40,6 +40,17 @@ static size_t sof_ipc4_fw_parse_ext_man(struct snd_sof_dev *sdev) ext_man_hdr = (struct sof_ext_manifest4_hdr *)fw->data; + /* + * At the start of the firmware image we must have an extended manifest. + * Verify that the magic number is correct. + */ + if (ext_man_hdr->id != SOF_EXT_MAN4_MAGIC_NUMBER) { + dev_err(sdev->dev, + "Unexpected extended manifest magic number: %#x\n", + ext_man_hdr->id); + return -EINVAL; + } + fw_hdr_offset = ipc4_data->manifest_fw_hdr_offset; if (!fw_hdr_offset) return -EINVAL; diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index e006532caf2f..a1be5d74f40b 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -460,13 +460,36 @@ static int mt8186_get_bar_index(struct snd_sof_dev *sdev, u32 type) return type; } -static int mt8186_ipc_msg_data(struct snd_sof_dev *sdev, - struct snd_pcm_substream *substream, - void *p, size_t sz) +static struct snd_soc_dai_driver mt8186_dai[] = { { - sof_mailbox_read(sdev, sdev->dsp_box.offset, p, sz); - return 0; -} + .name = "SOF_DL1", + .playback = { + .channels_min = 1, + .channels_max = 2, + }, +}, +{ + .name = "SOF_DL2", + .playback = { + .channels_min = 1, + .channels_max = 2, + }, +}, +{ + .name = "SOF_UL1", + .capture = { + .channels_min = 1, + .channels_max = 2, + }, +}, +{ + .name = "SOF_UL2", + .capture = { + .channels_min = 1, + .channels_max = 2, + }, +}, +}; /* mt8186 ops */ static struct snd_sof_dsp_ops sof_mt8186_ops = { @@ -481,6 +504,10 @@ static struct snd_sof_dsp_ops sof_mt8186_ops = { .block_read = sof_block_read, .block_write = sof_block_write, + /* Mailbox IO */ + .mailbox_read = sof_mailbox_read, + .mailbox_write = sof_mailbox_write, + /* Register IO */ .write = sof_io_write, .read = sof_io_read, @@ -491,18 +518,26 @@ static struct snd_sof_dsp_ops sof_mt8186_ops = { .send_msg = mt8186_send_msg, .get_mailbox_offset = mt8186_get_mailbox_offset, .get_window_offset = mt8186_get_window_offset, - .ipc_msg_data = mt8186_ipc_msg_data, + .ipc_msg_data = sof_ipc_msg_data, .set_stream_data_offset = sof_set_stream_data_offset, /* misc */ .get_bar_index = mt8186_get_bar_index, + /* stream callbacks */ + .pcm_open = sof_stream_pcm_open, + .pcm_close = sof_stream_pcm_close, + /* firmware loading */ .load_firmware = snd_sof_load_firmware_memcpy, /* Firmware ops */ .dsp_arch_ops = &sof_xtensa_arch_ops, + /* DAI drivers */ + .drv = mt8186_dai, + .num_drv = ARRAY_SIZE(mt8186_dai), + /* PM */ .suspend = mt8186_dsp_suspend, .resume = mt8186_dsp_resume, @@ -515,7 +550,16 @@ static struct snd_sof_dsp_ops sof_mt8186_ops = { SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, }; +static struct snd_sof_of_mach sof_mt8186_machs[] = { + { + .compatible = "mediatek,mt8186", + .sof_tplg_filename = "sof-mt8186.tplg", + }, + {} +}; + static const struct sof_dev_desc sof_of_mt8186_desc = { + .of_machines = sof_mt8186_machs, .ipc_supported_mask = BIT(SOF_IPC), .ipc_default = SOF_IPC, .default_fw_path = { diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 6cb6a432be5e..49f7cb049f62 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -13,6 +13,7 @@ #include <linux/pm_runtime.h> #include <sound/pcm_params.h> #include <sound/sof.h> +#include "sof-of-dev.h" #include "sof-priv.h" #include "sof-audio.h" #include "sof-utils.h" @@ -655,7 +656,12 @@ void snd_sof_new_platform_drv(struct snd_sof_dev *sdev) struct snd_sof_pdata *plat_data = sdev->pdata; const char *drv_name; - drv_name = plat_data->machine->drv_name; + if (plat_data->machine) + drv_name = plat_data->machine->drv_name; + else if (plat_data->of_machine) + drv_name = plat_data->of_machine->drv_name; + else + drv_name = NULL; pd->name = "sof-audio-component"; pd->probe = sof_pcm_probe; diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 28976098a89e..c18e723435bd 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -10,6 +10,7 @@ #include <linux/bitfield.h> #include "sof-audio.h" +#include "sof-of-dev.h" #include "ops.h" static void sof_reset_route_setup_status(struct snd_sof_dev *sdev, struct snd_sof_widget *widget) diff --git a/sound/soc/sof/sof-of-dev.h b/sound/soc/sof/sof-of-dev.h index fd950a222ba4..2948b3a0d9fe 100644 --- a/sound/soc/sof/sof-of-dev.h +++ b/sound/soc/sof/sof-of-dev.h @@ -9,6 +9,13 @@ #ifndef __SOUND_SOC_SOF_OF_H #define __SOUND_SOC_SOF_OF_H +struct snd_sof_of_mach { + const char *compatible; + const char *drv_name; + const char *fw_filename; + const char *sof_tplg_filename; +}; + extern const struct dev_pm_ops sof_of_pm; int sof_of_probe(struct platform_device *pdev); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 823583086279..33165299a20f 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -105,6 +105,13 @@ enum sof_debugfs_access_type { SOF_DEBUGFS_ACCESS_D0_ONLY, }; +struct sof_compr_stream { + u64 copied_total; + u32 sampling_rate; + u16 channels; + u16 sample_container_bytes; +}; + struct snd_sof_dev; struct snd_sof_ipc_msg; struct snd_sof_ipc; diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c index 7e8179cae92e..8163f453bf36 100644 --- a/sound/soc/ti/omap-mcbsp-st.c +++ b/sound/soc/ti/omap-mcbsp-st.c @@ -244,10 +244,10 @@ static ssize_t st_taps_show(struct device *dev, spin_lock_irq(&mcbsp->lock); for (i = 0; i < st_data->nr_taps; i++) - status += sprintf(&buf[status], (i ? ", %d" : "%d"), - st_data->taps[i]); + status += sysfs_emit_at(buf, status, (i ? ", %d" : "%d"), + st_data->taps[i]); if (i) - status += sprintf(&buf[status], "\n"); + status += sysfs_emit_at(buf, status, "\n"); spin_unlock_irq(&mcbsp->lock); return status; diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index c4ac1f30b9fe..0b377bb7737f 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -517,7 +517,7 @@ static ssize_t prop##_show(struct device *dev, \ { \ struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ \ - return sprintf(buf, "%u\n", mcbsp->prop); \ + return sysfs_emit(buf, "%u\n", mcbsp->prop); \ } \ \ static ssize_t prop##_store(struct device *dev, \ @@ -560,11 +560,11 @@ static ssize_t dma_op_mode_show(struct device *dev, for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) { if (dma_op_mode == i) - len += sprintf(buf + len, "[%s] ", *s); + len += sysfs_emit_at(buf, len, "[%s] ", *s); else - len += sprintf(buf + len, "%s ", *s); + len += sysfs_emit_at(buf, len, "%s ", *s); } - len += sprintf(buf + len, "\n"); + len += sysfs_emit_at(buf, len, "\n"); return len; } |