diff options
23 files changed, 510 insertions, 238 deletions
diff --git a/Documentation/devicetree/bindings/sound/adi,adau7002.txt b/Documentation/devicetree/bindings/sound/adi,adau7002.txt deleted file mode 100644 index f144ee1abf85..000000000000 --- a/Documentation/devicetree/bindings/sound/adi,adau7002.txt +++ /dev/null @@ -1,19 +0,0 @@ -Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter - -Required properties: - - - compatible: Must be "adi,adau7002" - -Optional properties: - - - IOVDD-supply: Phandle and specifier for the power supply providing the IOVDD - supply as covered in Documentation/devicetree/bindings/regulator/regulator.txt - - If this property is not present it is assumed that the supply pin is - hardwired to always on. - -Example: - adau7002: pdm-to-i2s { - compatible = "adi,adau7002"; - IOVDD-supply = <&supply>; - }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau7002.yaml b/Documentation/devicetree/bindings/sound/adi,adau7002.yaml new file mode 100644 index 000000000000..fcca0fde7d86 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau7002.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau7002.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: adi,adau7002 + + IOVDD-supply: + description: + IOVDD power supply, if skipped then it is assumed that the supply pin is + hardwired to always on. + + wakeup-delay-ms: + description: + Delay after power up needed for device to settle. + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + audio-codec { + compatible = "adi,adau7002"; + IOVDD-supply = <&pp1800_l15a>; + #sound-dai-cells = <0>; + wakeup-delay-ms = <80>; + }; diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index ff1228713f7e..0c398581d52b 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -20,6 +20,9 @@ Optional properties: - realtek,in3-differential Boolean. Indicate MIC1/2/3 input are differential, rather than single-ended. +- realtek,lout-differential + Boolean. Indicate LOUT output is differential, rather than stereo. + - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. - realtek,dmic1-data-pin diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml new file mode 100644 index 000000000000..f0375bbf4c40 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml @@ -0,0 +1,158 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +# Copyright (C) 2022 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320aic3x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320AIC3x Codec + +description: | + TLV320AIC3x are a series of low-power stereo audio codecs with stereo + headphone amplifier, as well as multiple inputs and outputs programmable in + single-ended or fully differential configurations. + + The serial control bus supports SPI or I2C protocols, while the serial audio + data bus is programmable for I2S, left/right-justified, DSP, or TDM modes. + + The following pins can be referred in the sound node's audio routing property: + + CODEC output pins: + LLOUT + RLOUT + MONO_LOUT + HPLOUT + HPROUT + HPLCOM + HPRCOM + + CODEC input pins for TLV320AIC3104: + MIC2L + MIC2R + LINE1L + LINE1R + + CODEC input pins for other compatible codecs: + MIC3L + MIC3R + LINE1L + LINE2L + LINE1R + LINE2R + +maintainers: + - Jai Luthra <j-luthra@ti.com> + +properties: + compatible: + enum: + - ti,tlv320aic3x + - ti,tlv320aic33 + - ti,tlv320aic3007 + - ti,tlv320aic3106 + - ti,tlv320aic3104 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + description: + GPIO specification for the active low RESET input. + + gpio-reset: + maxItems: 1 + description: + Deprecated, please use reset-gpios instead. + deprecated: true + + ai3x-gpio-func: + description: AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + $ref: /schemas/types.yaml#/definitions/uint32-array + maxItems: 2 + + ai3x-micbias-vg: + description: MicBias required voltage. If node is omitted then MicBias is powered down. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 1 + description: MICBIAS output is powered to 2.0V. + - const: 2 + description: MICBIAS output is powered to 2.5V. + - const: 3 + description: MICBIAS output is connected to AVDD. + + ai3x-ocmv: + description: Output Common-Mode Voltage selection. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 0 + description: 1.35V + - const: 1 + description: 1.5V + - const: 2 + description: 1.65V + - const: 3 + description: 1.8V + + AVDD-supply: + description: Analog DAC voltage. + + IOVDD-supply: + description: I/O voltage. + + DRVDD-supply: + description: ADC analog and output driver voltage. + + DVDD-supply: + description: Digital core voltage. + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_i2c: audio-codec@1b { + compatible = "ti,tlv320aic3x"; + reg = <0x1b>; + + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DRVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; + }; + }; + + - | + #include <dt-bindings/gpio/gpio.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_spi: audio-codec@0 { + compatible = "ti,tlv320aic3x"; + reg = <0>; /* CS number */ + #sound-dai-cells = <0>; + + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DRVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; + ai3x-ocmv = <0>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt deleted file mode 100644 index 20931a63fd64..000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ /dev/null @@ -1,97 +0,0 @@ -Texas Instruments - tlv320aic3x Codec module - -The tlv320aic3x serial control bus communicates through both I2C and SPI bus protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tlv320aic3x" - Generic TLV320AIC3x device - "ti,tlv320aic33" - TLV320AIC33 - "ti,tlv320aic3007" - TLV320AIC3007 - "ti,tlv320aic3106" - TLV320AIC3106 - "ti,tlv320aic3104" - TLV320AIC3104 - - -- reg - <int> - I2C slave address - - -Optional properties: - -- reset-gpios - GPIO specification for the active low RESET input. -- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality - - Not supported on tlv320aic3104 -- ai3x-micbias-vg - MicBias Voltage required. - 1 - MICBIAS output is powered to 2.0V, - 2 - MICBIAS output is powered to 2.5V, - 3 - MICBIAS output is connected to AVDD, - If this node is not mentioned or if the value is incorrect, then MicBias - is powered down. -- ai3x-ocmv - Output Common-Mode Voltage selection: - 0 - 1.35V, - 1 - 1.5V, - 2 - 1.65V, - 3 - 1.8V -- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the - device as covered in Documentation/devicetree/bindings/regulator/regulator.txt - -Deprecated properties: - -- gpio-reset - gpio pin number used for codec reset - -CODEC output pins: - * LLOUT - * RLOUT - * MONO_LOUT - * HPLOUT - * HPROUT - * HPLCOM - * HPRCOM - -CODEC input pins for TLV320AIC3104: - * MIC2L - * MIC2R - * LINE1L - * LINE1R - -CODEC input pins for other compatible codecs: - * MIC3L - * MIC3R - * LINE1L - * LINE2L - * LINE1R - * LINE2R - -The pins can be used in referring sound node's audio-routing property. - -I2C example: - -#include <dt-bindings/gpio/gpio.h> - -tlv320aic3x: tlv320aic3x@1b { - compatible = "ti,tlv320aic3x"; - reg = <0x1b>; - - reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; - - AVDD-supply = <®ulator>; - IOVDD-supply = <®ulator>; - DRVDD-supply = <®ulator>; - DVDD-supply = <®ulator>; -}; - -SPI example: - -spi0: spi@f0000000 { - tlv320aic3x: codec@0 { - compatible = "ti,tlv320aic3x"; - reg = <0>; /* CS number */ - #sound-dai-cells = <0>; - spi-max-frequency = <1000000>; - - AVDD-supply = <®ulator>; - IOVDD-supply = <®ulator>; - DRVDD-supply = <®ulator>; - DVDD-supply = <®ulator>; - ai3x-ocmv = <0>; - }; -}; diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index 5e7f9c1c1b0e..8f024cb309c9 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -11,7 +11,6 @@ #define ACP63_REG_START 0x1240000 #define ACP63_REG_END 0x1250200 #define ACP63_DEVS 3 -#define ACP63_PDM_MODE 1 #define ACP_SOFT_RESET_SOFTRESET_AUDDONE_MASK 0x00010001 #define ACP_PGFSM_CNTL_POWER_ON_MASK 1 @@ -54,6 +53,11 @@ /* time in ms for runtime suspend delay */ #define ACP_SUSPEND_DELAY_MS 2000 +#define ACP63_DMIC_ADDR 2 +#define ACP63_PDM_MODE_DEVS 3 +#define ACP63_PDM_DEV_MASK 1 +#define ACP_DMIC_DEV 2 + enum acp_config { ACP_CONFIG_0 = 0, ACP_CONFIG_1, @@ -102,4 +106,7 @@ struct acp63_dev_data { struct resource *res; bool acp63_audio_mode; struct platform_device *pdev[ACP63_DEVS]; + u16 pdev_mask; + u16 pdev_count; + u16 pdm_dev_index; }; diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 489f01a20699..401cfd0036be 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -116,6 +116,7 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) struct acp63_dev_data *adata; struct pdm_dev_data *ps_pdm_data; u32 val; + u16 pdev_index; adata = dev_id; if (!adata) @@ -123,7 +124,8 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) val = acp63_readl(adata->acp63_base + ACP_EXTERNAL_INTR_STAT); if (val & BIT(PDM_DMA_STAT)) { - ps_pdm_data = dev_get_drvdata(&adata->pdev[0]->dev); + pdev_index = adata->pdm_dev_index; + ps_pdm_data = dev_get_drvdata(&adata->pdev[pdev_index]->dev); acp63_writel(BIT(PDM_DMA_STAT), adata->acp63_base + ACP_EXTERNAL_INTR_STAT); if (ps_pdm_data->capture_stream) snd_pcm_period_elapsed(ps_pdm_data->capture_stream); @@ -132,17 +134,119 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) return IRQ_NONE; } +static void get_acp63_device_config(u32 config, struct pci_dev *pci, + struct acp63_dev_data *acp_data) +{ + struct acpi_device *dmic_dev; + const union acpi_object *obj; + bool is_dmic_dev = false; + + dmic_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), ACP63_DMIC_ADDR, 0); + if (dmic_dev) { + if (!acpi_dev_get_property(dmic_dev, "acp-audio-device-type", + ACPI_TYPE_INTEGER, &obj) && + obj->integer.value == ACP_DMIC_DEV) + is_dmic_dev = true; + } + + switch (config) { + case ACP_CONFIG_0: + case ACP_CONFIG_1: + case ACP_CONFIG_2: + case ACP_CONFIG_3: + case ACP_CONFIG_9: + case ACP_CONFIG_15: + dev_dbg(&pci->dev, "Audio Mode %d\n", config); + break; + default: + if (is_dmic_dev) { + acp_data->pdev_mask = ACP63_PDM_DEV_MASK; + acp_data->pdev_count = ACP63_PDM_MODE_DEVS; + } + break; + } +} + +void acp63_fill_platform_dev_info(struct platform_device_info *pdevinfo, struct device *parent, + struct fwnode_handle *fw_node, char *name, unsigned int id, + const struct resource *res, unsigned int num_res, + const void *data, size_t size_data) +{ + pdevinfo->name = name; + pdevinfo->id = id; + pdevinfo->parent = parent; + pdevinfo->num_res = num_res; + pdevinfo->res = res; + pdevinfo->data = data; + pdevinfo->size_data = size_data; + pdevinfo->fwnode = fw_node; +} + +static int create_acp63_platform_devs(struct pci_dev *pci, struct acp63_dev_data *adata, u32 addr) +{ + struct platform_device_info pdevinfo[ACP63_DEVS]; + struct device *parent; + int index; + int ret; + + parent = &pci->dev; + dev_dbg(&pci->dev, + "%s pdev_mask:0x%x pdev_count:0x%x\n", __func__, adata->pdev_mask, + adata->pdev_count); + if (adata->pdev_mask) { + adata->res = devm_kzalloc(&pci->dev, sizeof(struct resource), GFP_KERNEL); + if (!adata->res) { + ret = -ENOMEM; + goto de_init; + } + adata->res->flags = IORESOURCE_MEM; + adata->res->start = addr; + adata->res->end = addr + (ACP63_REG_END - ACP63_REG_START); + memset(&pdevinfo, 0, sizeof(pdevinfo)); + } + + switch (adata->pdev_mask) { + case ACP63_PDM_DEV_MASK: + adata->pdm_dev_index = 0; + acp63_fill_platform_dev_info(&pdevinfo[0], parent, NULL, "acp_ps_pdm_dma", + 0, adata->res, 1, NULL, 0); + acp63_fill_platform_dev_info(&pdevinfo[1], parent, NULL, "dmic-codec", + 0, NULL, 0, NULL, 0); + acp63_fill_platform_dev_info(&pdevinfo[2], parent, NULL, "acp_ps_mach", + 0, NULL, 0, NULL, 0); + break; + default: + dev_dbg(&pci->dev, "No PDM devices found\n"); + goto de_init; + } + + for (index = 0; index < adata->pdev_count; index++) { + adata->pdev[index] = platform_device_register_full(&pdevinfo[index]); + if (IS_ERR(adata->pdev[index])) { + dev_err(&pci->dev, + "cannot register %s device\n", pdevinfo[index].name); + ret = PTR_ERR(adata->pdev[index]); + goto unregister_devs; + } + } + return 0; +unregister_devs: + for (--index; index >= 0; index--) + platform_device_unregister(adata->pdev[index]); +de_init: + if (acp63_deinit(adata->acp63_base, &pci->dev)) + dev_err(&pci->dev, "ACP de-init failed\n"); + return ret; +} + static int snd_acp63_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { struct acp63_dev_data *adata; - struct platform_device_info pdevinfo[ACP63_DEVS]; - int index, ret; - int val = 0x00; - struct acpi_device *adev; - const union acpi_object *obj; u32 addr; - unsigned int irqflags; + u32 irqflags; + int val; + int ret; irqflags = IRQF_SHARED; /* Pink Sardine device check */ @@ -182,83 +286,24 @@ static int snd_acp63_probe(struct pci_dev *pci, ret = acp63_init(adata->acp63_base, &pci->dev); if (ret) goto release_regions; + ret = devm_request_irq(&pci->dev, pci->irq, acp63_irq_handler, + irqflags, "ACP_PCI_IRQ", adata); + if (ret) { + dev_err(&pci->dev, "ACP PCI IRQ request failed\n"); + goto de_init; + } val = acp63_readl(adata->acp63_base + ACP_PIN_CONFIG); - switch (val) { - case ACP_CONFIG_0: - case ACP_CONFIG_1: - case ACP_CONFIG_2: - case ACP_CONFIG_3: - case ACP_CONFIG_9: - case ACP_CONFIG_15: - dev_info(&pci->dev, "Audio Mode %d\n", val); - break; - default: - - /* Checking DMIC hardware*/ - adev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), 0x02, 0); - - if (!adev) - break; - - if (!acpi_dev_get_property(adev, "acp-audio-device-type", - ACPI_TYPE_INTEGER, &obj) && - obj->integer.value == 2) { - adata->res = devm_kzalloc(&pci->dev, sizeof(struct resource), GFP_KERNEL); - if (!adata->res) { - ret = -ENOMEM; - goto de_init; - } - - adata->res->name = "acp_iomem"; - adata->res->flags = IORESOURCE_MEM; - adata->res->start = addr; - adata->res->end = addr + (ACP63_REG_END - ACP63_REG_START); - adata->acp63_audio_mode = ACP63_PDM_MODE; - - memset(&pdevinfo, 0, sizeof(pdevinfo)); - pdevinfo[0].name = "acp_ps_pdm_dma"; - pdevinfo[0].id = 0; - pdevinfo[0].parent = &pci->dev; - pdevinfo[0].num_res = 1; - pdevinfo[0].res = adata->res; - - pdevinfo[1].name = "dmic-codec"; - pdevinfo[1].id = 0; - pdevinfo[1].parent = &pci->dev; - - pdevinfo[2].name = "acp_ps_mach"; - pdevinfo[2].id = 0; - pdevinfo[2].parent = &pci->dev; - - for (index = 0; index < ACP63_DEVS; index++) { - adata->pdev[index] = - platform_device_register_full(&pdevinfo[index]); - - if (IS_ERR(adata->pdev[index])) { - dev_err(&pci->dev, - "cannot register %s device\n", - pdevinfo[index].name); - ret = PTR_ERR(adata->pdev[index]); - goto unregister_devs; - } - ret = devm_request_irq(&pci->dev, pci->irq, acp63_irq_handler, - irqflags, "ACP_PCI_IRQ", adata); - if (ret) { - dev_err(&pci->dev, "ACP PCI IRQ request failed\n"); - goto unregister_devs; - } - } - } - break; + get_acp63_device_config(val, pci, adata); + ret = create_acp63_platform_devs(pci, adata, addr); + if (ret < 0) { + dev_err(&pci->dev, "ACP platform devices creation failed\n"); + goto de_init; } pm_runtime_set_autosuspend_delay(&pci->dev, ACP_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pci->dev); pm_runtime_put_noidle(&pci->dev); pm_runtime_allow(&pci->dev); return 0; -unregister_devs: - for (--index; index >= 0; index--) - platform_device_unregister(adata->pdev[index]); de_init: if (acp63_deinit(adata->acp63_base, &pci->dev)) dev_err(&pci->dev, "ACP de-init failed\n"); @@ -305,10 +350,8 @@ static void snd_acp63_remove(struct pci_dev *pci) int ret, index; adata = pci_get_drvdata(pci); - if (adata->acp63_audio_mode == ACP63_PDM_MODE) { - for (index = 0; index < ACP63_DEVS; index++) - platform_device_unregister(adata->pdev[index]); - } + for (index = 0; index < adata->pdev_count; index++) + platform_device_unregister(adata->pdev[index]); ret = acp63_deinit(adata->acp63_base, &pci->dev); if (ret) dev_err(&pci->dev, "ACP de-init failed\n"); diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 544ccbcfc884..0068780fe0a7 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1157,13 +1157,31 @@ static int da7213_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct da7213_priv *da7213 = snd_soc_component_get_drvdata(component); + u8 dai_clk_mode = DA7213_DAI_BCLKS_PER_WCLK_64; u8 dai_ctrl = 0; u8 fs; + /* Set channels */ + switch (params_channels(params)) { + case 1: + if (da7213->fmt != DA7213_DAI_FORMAT_DSP) { + dev_err(component->dev, "Mono supported only in DSP mode\n"); + return -EINVAL; + } + dai_ctrl |= DA7213_DAI_MONO_MODE_EN; + break; + case 2: + dai_ctrl &= ~(DA7213_DAI_MONO_MODE_EN); + break; + default: + return -EINVAL; + } + /* Set DAI format */ switch (params_width(params)) { case 16: dai_ctrl |= DA7213_DAI_WORD_LENGTH_S16_LE; + dai_clk_mode = DA7213_DAI_BCLKS_PER_WCLK_32; /* 32bit for 1ch and 2ch */ break; case 20: dai_ctrl |= DA7213_DAI_WORD_LENGTH_S20_LE; @@ -1224,8 +1242,11 @@ static int da7213_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_component_update_bits(component, DA7213_DAI_CTRL, DA7213_DAI_WORD_LENGTH_MASK, - dai_ctrl); + snd_soc_component_update_bits(component, DA7213_DAI_CLK_MODE, + DA7213_DAI_BCLKS_PER_WCLK_MASK, dai_clk_mode); + + snd_soc_component_update_bits(component, DA7213_DAI_CTRL, + DA7213_DAI_WORD_LENGTH_MASK | DA7213_DAI_MONO_MODE_MASK, dai_ctrl); snd_soc_component_write(component, DA7213_SR, fs); return 0; @@ -1300,19 +1321,24 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: dai_ctrl |= DA7213_DAI_FORMAT_I2S_MODE; + da7213->fmt = DA7213_DAI_FORMAT_I2S_MODE; break; case SND_SOC_DAIFMT_LEFT_J: dai_ctrl |= DA7213_DAI_FORMAT_LEFT_J; + da7213->fmt = DA7213_DAI_FORMAT_LEFT_J; break; case SND_SOC_DAIFMT_RIGHT_J: dai_ctrl |= DA7213_DAI_FORMAT_RIGHT_J; + da7213->fmt = DA7213_DAI_FORMAT_RIGHT_J; break; case SND_SOC_DAI_FORMAT_DSP_A: /* L data MSB after FRM LRC */ dai_ctrl |= DA7213_DAI_FORMAT_DSP; dai_offset = 1; + da7213->fmt = DA7213_DAI_FORMAT_DSP; break; case SND_SOC_DAI_FORMAT_DSP_B: /* L data MSB during FRM LRC */ dai_ctrl |= DA7213_DAI_FORMAT_DSP; + da7213->fmt = DA7213_DAI_FORMAT_DSP; break; default: return -EINVAL; diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 97ccf0ddd2be..4ca9cfdea06d 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -195,6 +195,8 @@ #define DA7213_DAI_WORD_LENGTH_S24_LE (0x2 << 2) #define DA7213_DAI_WORD_LENGTH_S32_LE (0x3 << 2) #define DA7213_DAI_WORD_LENGTH_MASK (0x3 << 2) +#define DA7213_DAI_MONO_MODE_EN (0x1 << 4) +#define DA7213_DAI_MONO_MODE_MASK (0x1 << 4) #define DA7213_DAI_EN_SHIFT 7 /* DA7213_DIG_ROUTING_DAI = 0x21 */ @@ -542,6 +544,7 @@ struct da7213_priv { bool alc_en; bool fixed_clk_auto_pll; struct da7213_platform_data *pdata; + int fmt; }; #endif /* _DA7213_H */ diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 708e55b7431a..9e6341a68c6b 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2717,6 +2717,10 @@ static int rt5640_probe(struct snd_soc_component *component) snd_soc_component_update_bits(component, RT5640_IN1_IN2, RT5640_IN_DF2, RT5640_IN_DF2); + if (device_property_read_bool(component->dev, "realtek,lout-differential")) + snd_soc_component_update_bits(component, RT5640_DUMMY1, + RT5640_EN_LOUT_DF, RT5640_EN_LOUT_DF); + if (device_property_read_u32(component->dev, "realtek,dmic1-data-pin", &val) == 0 && val) { dmic1_data_pin = val - 1; diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index f58b88e3325b..9847a1ae01f4 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -1978,6 +1978,8 @@ #define RT5640_ZCD_HP_EN (0x1 << 15) /* General Control 1 (0xfa) */ +#define RT5640_EN_LOUT_DF (0x1 << 14) +#define RT5640_EN_LOUT_DF_SFT 14 #define RT5640_M_MONO_ADC_L (0x1 << 13) #define RT5640_M_MONO_ADC_L_SFT 13 #define RT5640_M_MONO_ADC_R (0x1 << 12) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index b78dd5994edb..7cdf184d380b 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -463,6 +463,21 @@ static void rt711_sdca_jack_init(struct rt711_sdca_priv *rt711) RT711_HP_JD_FINAL_RESULT_CTL_JD12, RT711_HP_JD_FINAL_RESULT_CTL_JD12); break; + case RT711_JD2_100K: + rt711_sdca_index_write(rt711, RT711_VENDOR_REG, + RT711_COMBO_JACK_AUTO_CTL3, 0xa47e); + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_REG, + RT711_JD_CTL1, RT711_JD2_DIGITAL_MODE_SEL, + RT711_JD2_DIGITAL_MODE_SEL); + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_REG, + RT711_JD_CTL2, RT711_JD2_2PORT_200K_DECODE_HP | + RT711_JD2_2PORT_100K_DECODE_MASK | RT711_HP_JD_SEL_JD2, + RT711_JD2_2PORT_100K_DECODE_HP | RT711_HP_JD_SEL_JD2); + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_REG, + RT711_CC_DET1, + RT711_HP_JD_FINAL_RESULT_CTL_JD12 | RT711_POW_CC1_AGPI, + RT711_HP_JD_FINAL_RESULT_CTL_JD12 | RT711_POW_CC1_AGPI_OFF); + break; default: dev_warn(rt711->component->dev, "Wrong JD source\n"); break; diff --git a/sound/soc/codecs/rt711-sdca.h b/sound/soc/codecs/rt711-sdca.h index 498ca687c47b..10e3c801b813 100644 --- a/sound/soc/codecs/rt711-sdca.h +++ b/sound/soc/codecs/rt711-sdca.h @@ -127,12 +127,17 @@ struct sdw_stream_data { /* jack detect control 2 (0x09)(NID:20h) */ #define RT711_JD2_2PORT_200K_DECODE_HP (0x1 << 13) +#define RT711_JD2_2PORT_100K_DECODE_MASK (0x1 << 12) +#define RT711_JD2_2PORT_100K_DECODE_HP (0x0 << 12) #define RT711_HP_JD_SEL_JD1 (0x0 << 1) #define RT711_HP_JD_SEL_JD2 (0x1 << 1) /* CC DET1 (0x11)(NID:20h) */ #define RT711_HP_JD_FINAL_RESULT_CTL_JD12 (0x1 << 10) #define RT711_HP_JD_FINAL_RESULT_CTL_CCDET (0x0 << 10) +#define RT711_POW_CC1_AGPI (0x1 << 5) +#define RT711_POW_CC1_AGPI_ON (0x1 << 5) +#define RT711_POW_CC1_AGPI_OFF (0x0 << 5) /* Parameter & Verb control (0x1a)(NID:20h) */ #define RT711_HIDDEN_REG_SW_RESET (0x1 << 14) @@ -226,7 +231,8 @@ enum { enum rt711_sdca_jd_src { RT711_JD_NULL, RT711_JD1, - RT711_JD2 + RT711_JD2, + RT711_JD2_100K }; enum rt711_sdca_ver { diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 8dac9fd88547..8eb4782c9232 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -37,7 +37,9 @@ #include "wm8940.h" struct wm8940_priv { - unsigned int sysclk; + unsigned int mclk; + unsigned int fs; + struct regmap *regmap; }; @@ -387,17 +389,24 @@ static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int wm8940_update_clocks(struct snd_soc_dai *dai); static int wm8940_i2s_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 wm8940_priv *priv = snd_soc_component_get_drvdata(component); u16 iface = snd_soc_component_read(component, WM8940_IFACE) & 0xFD9F; u16 addcntrl = snd_soc_component_read(component, WM8940_ADDCNTRL) & 0xFFF1; u16 companding = snd_soc_component_read(component, WM8940_COMPANDINGCTL) & 0xFFDF; int ret; + priv->fs = params_rate(params); + ret = wm8940_update_clocks(dai); + if (ret) + return ret; + /* LoutR control */ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && params_channels(params) == 2) @@ -611,24 +620,6 @@ static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, return 0; } -static int wm8940_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_component *component = codec_dai->component; - struct wm8940_priv *wm8940 = snd_soc_component_get_drvdata(component); - - switch (freq) { - case 11289600: - case 12000000: - case 12288000: - case 16934400: - case 18432000: - wm8940->sysclk = freq; - return 0; - } - return -EINVAL; -} - static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { @@ -653,6 +644,78 @@ static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai, return ret; } +static unsigned int wm8940_get_mclkdiv(unsigned int f_in, unsigned int f_out, + int *mclkdiv) +{ + unsigned int ratio = 2 * f_in / f_out; + + if (ratio <= 2) { + *mclkdiv = WM8940_MCLKDIV_1; + ratio = 2; + } else if (ratio == 3) { + *mclkdiv = WM8940_MCLKDIV_1_5; + } else if (ratio == 4) { + *mclkdiv = WM8940_MCLKDIV_2; + } else if (ratio <= 6) { + *mclkdiv = WM8940_MCLKDIV_3; + ratio = 6; + } else if (ratio <= 8) { + *mclkdiv = WM8940_MCLKDIV_4; + ratio = 8; + } else if (ratio <= 12) { + *mclkdiv = WM8940_MCLKDIV_6; + ratio = 12; + } else if (ratio <= 16) { + *mclkdiv = WM8940_MCLKDIV_8; + ratio = 16; + } else { + *mclkdiv = WM8940_MCLKDIV_12; + ratio = 24; + } + + return f_out * ratio / 2; +} + +static int wm8940_update_clocks(struct snd_soc_dai *dai) +{ + struct snd_soc_component *codec = dai->component; + struct wm8940_priv *priv = snd_soc_component_get_drvdata(codec); + unsigned int fs256; + unsigned int fpll = 0; + unsigned int f; + int mclkdiv; + + if (!priv->mclk || !priv->fs) + return 0; + + fs256 = 256 * priv->fs; + + f = wm8940_get_mclkdiv(priv->mclk, fs256, &mclkdiv); + if (f != priv->mclk) { + /* The PLL performs best around 90MHz */ + fpll = wm8940_get_mclkdiv(22500000, fs256, &mclkdiv); + } + + wm8940_set_dai_pll(dai, 0, 0, priv->mclk, fpll); + wm8940_set_dai_clkdiv(dai, WM8940_MCLKDIV, mclkdiv); + + return 0; +} + +static int wm8940_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *codec = dai->component; + struct wm8940_priv *priv = snd_soc_component_get_drvdata(codec); + + if (dir != SND_SOC_CLOCK_IN) + return -EINVAL; + + priv->mclk = freq; + + return wm8940_update_clocks(dai); +} + #define WM8940_RATES SNDRV_PCM_RATE_8000_48000 #define WM8940_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ @@ -697,6 +760,17 @@ static int wm8940_probe(struct snd_soc_component *component) int ret; u16 reg; + /* + * Check chip ID for wm8940 - value of 0x00 offset + * SOFTWARE_RESET on write + * CHIP_ID on read + */ + reg = snd_soc_component_read(component, WM8940_SOFTRESET); + if (reg != WM8940_CHIP_ID) { + dev_err(component->dev, "Wrong wm8940 chip ID: 0x%x\n", reg); + return -ENODEV; + } + ret = wm8940_reset(component); if (ret < 0) { dev_err(component->dev, "Failed to issue reset\n"); @@ -709,9 +783,7 @@ static int wm8940_probe(struct snd_soc_component *component) if (ret < 0) return ret; - if (!pdata) - dev_warn(component->dev, "No platform data supplied\n"); - else { + if (pdata) { reg = snd_soc_component_read(component, WM8940_OUTPUTCTL); ret = snd_soc_component_write(component, WM8940_OUTPUTCTL, reg | pdata->vroi); if (ret < 0) diff --git a/sound/soc/codecs/wm8940.h b/sound/soc/codecs/wm8940.h index 0d4f53ada2e6..86bbf902ef5a 100644 --- a/sound/soc/codecs/wm8940.h +++ b/sound/soc/codecs/wm8940.h @@ -95,5 +95,8 @@ struct wm8940_setup_data { #define WM8940_OPCLKDIV_3 2 #define WM8940_OPCLKDIV_4 3 +/* Chip ID */ +#define WM8940_CHIP_ID 0x8940 + #endif /* _WM8940_H */ diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 44303b6eb228..70442315f5c5 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -306,18 +306,12 @@ static int spitz_probe(struct platform_device *pdev) return ret; } -static int spitz_remove(struct platform_device *pdev) -{ - return 0; -} - static struct platform_driver spitz_driver = { .driver = { .name = "spitz-audio", .pm = &snd_soc_pm_ops, }, .probe = spitz_probe, - .remove = spitz_remove, }; module_platform_driver(spitz_driver); diff --git a/sound/soc/sof/amd/acp-dsp-offset.h b/sound/soc/sof/amd/acp-dsp-offset.h index de5726251dc6..920155dee819 100644 --- a/sound/soc/sof/amd/acp-dsp-offset.h +++ b/sound/soc/sof/amd/acp-dsp-offset.h @@ -85,4 +85,8 @@ #define ACP_SCRATCH_REG_0 0x10000 #define ACP6X_DSP_FUSION_RUNSTALL 0x0644 + +/* Cache window registers */ +#define ACP_DSP0_CACHE_OFFSET0 0x0420 +#define ACP_DSP0_CACHE_SIZE0 0x0424 #endif diff --git a/sound/soc/sof/amd/acp-loader.c b/sound/soc/sof/amd/acp-loader.c index 090c8b18c83c..a4bce5a3ae48 100644 --- a/sound/soc/sof/amd/acp-loader.c +++ b/sound/soc/sof/amd/acp-loader.c @@ -151,6 +151,7 @@ static void configure_pte_for_fw_loading(int type, int num_pages, struct acp_dev int acp_dsp_pre_fw_run(struct snd_sof_dev *sdev) { struct pci_dev *pci = to_pci_dev(sdev->dev); + const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); struct acp_dev_data *adata; unsigned int src_addr, size_fw; u32 page_count, dma_size; @@ -183,6 +184,12 @@ int acp_dsp_pre_fw_run(struct snd_sof_dev *sdev) if (ret < 0) dev_err(sdev->dev, "acp dma transfer status: %d\n", ret); + if (desc->rev > 3) { + /* Cache Window enable */ + snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_DSP0_CACHE_OFFSET0, desc->sram_pte_offset); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_DSP0_CACHE_SIZE0, SRAM1_SIZE | BIT(31)); + } + /* Free memory once DMA is complete */ dma_size = (PAGE_ALIGN(sdev->basefw.fw->size) >> PAGE_SHIFT) * ACP_PAGE_SIZE; dma_free_coherent(&pci->dev, dma_size, adata->bin_buf, adata->sha_dma_addr); diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 09e16ef8afa0..4314094a97fd 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -72,6 +72,8 @@ #define EXCEPT_MAX_HDR_SIZE 0x400 #define AMD_STACK_DUMP_SIZE 32 +#define SRAM1_SIZE 0x13A000 + enum clock_source { ACP_CLOCK_96M = 0, ACP_CLOCK_48M, diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 625977a29d8a..26a3f7c8c914 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -362,6 +362,9 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) sdev->first_boot = true; dev_set_drvdata(dev, sdev); + if (sof_core_debug) + dev_info(dev, "sof_debug value: %#x\n", sof_core_debug); + /* check IPC support */ if (!(BIT(plat_data->ipc_type) & plat_data->desc->ipc_supported_mask)) { dev_err(dev, "ipc_type %d is not supported on this platform, mask is %#x\n", diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index b94cc40485ed..989395999d6e 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -1651,6 +1651,7 @@ static int sof_ipc3_route_setup(struct snd_sof_dev *sdev, struct snd_sof_route * static int sof_ipc3_control_load_bytes(struct snd_sof_dev *sdev, struct snd_sof_control *scontrol) { struct sof_ipc_ctrl_data *cdata; + size_t priv_size_check; int ret; if (scontrol->max_size < (sizeof(*cdata) + sizeof(struct sof_abi_hdr))) { @@ -1694,8 +1695,10 @@ static int sof_ipc3_control_load_bytes(struct snd_sof_dev *sdev, struct snd_sof_ goto err; } - if (cdata->data->size + sizeof(struct sof_abi_hdr) != scontrol->priv_size) { - dev_err(sdev->dev, "Conflict in bytes vs. priv size.\n"); + priv_size_check = cdata->data->size + sizeof(struct sof_abi_hdr); + if (priv_size_check != scontrol->priv_size) { + dev_err(sdev->dev, "Conflict in bytes (%zu) vs. priv size (%zu).\n", + priv_size_check, scontrol->priv_size); ret = -EINVAL; goto err; } diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 3c81e84fcecf..7c831e18483c 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -215,11 +215,6 @@ static int platform_parse_resource(struct platform_device *pdev, void *data) adsp->pa_sram = (phys_addr_t)mmio->start; adsp->sramsize = resource_size(mmio); - if (adsp->sramsize < TOTAL_SIZE_SHARED_SRAM_FROM_TAIL) { - dev_err(dev, "adsp SRAM(%#x) is not enough for share\n", - adsp->sramsize); - return -EINVAL; - } dev_dbg(dev, "sram pbase=%pa,%#x\n", &adsp->pa_sram, adsp->sramsize); diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.h b/sound/soc/sof/mediatek/mt8195/mt8195.h index 7ffd523f936c..b4229170049f 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.h +++ b/sound/soc/sof/mediatek/mt8195/mt8195.h @@ -139,8 +139,6 @@ struct snd_sof_dev; #define DSP_MBOX1_BAR 6 #define DSP_MBOX2_BAR 7 -#define TOTAL_SIZE_SHARED_SRAM_FROM_TAIL 0x0 - #define SIZE_SHARED_DRAM_DL 0x40000 /*Shared buffer for Downlink*/ #define SIZE_SHARED_DRAM_UL 0x40000 /*Shared buffer for Uplink*/ |