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

[PATCH v4] dt-bindings: sound: tlv320aic3x: Convert to dtschema
by Jai Luthra 27 Dec '22

27 Dec '22

Convert bindings for TI's TLV320AIC3x audio codecs to dtschema. The following properties are still found in some old dts files, but will be ignored by the driver: - adc-settle-ms - assigned-clock-parents, assigned-clock-rates, assigned-clocks - port Signed-off-by: Jai Luthra <j-luthra(a)ti.com> --- Changes in v4: - Use generic node name in the example - Fix typo in commit message v3: https://lore.kernel.org/linux-devicetree/20221220123951.29959-1-j-luthra@ti… v2: https://lore.kernel.org/linux-devicetree/20220803151726.31628-1-j-luthra@ti… .../bindings/sound/ti,tlv320aic3x.yaml | 158 ++++++++++++++++++ .../devicetree/bindings/sound/tlv320aic3x.txt | 97 ----------- 2 files changed, 158 insertions(+), 97 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.txt 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(a)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 = <&regulator>; + IOVDD-supply = <&regulator>; + DRVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + }; + }; + + - | + #include <dt-bindings/gpio/gpio.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_spi: audio-codec@0 { + compatible = "ti,tlv320aic3x"; + reg = <0>; /* CS number */ + #sound-dai-cells = <0>; + + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DRVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + ai3x-ocmv = <0>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt deleted file mode 100644 index 20931a63fd64..000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ /dev/null @@ -1,97 +0,0 @@ -Texas Instruments - tlv320aic3x Codec module - -The tlv320aic3x serial control bus communicates through both I2C and SPI bus protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tlv320aic3x" - Generic TLV320AIC3x device - "ti,tlv320aic33" - TLV320AIC33 - "ti,tlv320aic3007" - TLV320AIC3007 - "ti,tlv320aic3106" - TLV320AIC3106 - "ti,tlv320aic3104" - TLV320AIC3104 - - -- reg - <int> - I2C slave address - - -Optional properties: - -- reset-gpios - GPIO specification for the active low RESET input. -- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality - - Not supported on tlv320aic3104 -- ai3x-micbias-vg - MicBias Voltage required. - 1 - MICBIAS output is powered to 2.0V, - 2 - MICBIAS output is powered to 2.5V, - 3 - MICBIAS output is connected to AVDD, - If this node is not mentioned or if the value is incorrect, then MicBias - is powered down. -- ai3x-ocmv - Output Common-Mode Voltage selection: - 0 - 1.35V, - 1 - 1.5V, - 2 - 1.65V, - 3 - 1.8V -- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the - device as covered in Documentation/devicetree/bindings/regulator/regulator.txt - -Deprecated properties: - -- gpio-reset - gpio pin number used for codec reset - -CODEC output pins: - * LLOUT - * RLOUT - * MONO_LOUT - * HPLOUT - * HPROUT - * HPLCOM - * HPRCOM - -CODEC input pins for TLV320AIC3104: - * MIC2L - * MIC2R - * LINE1L - * LINE1R - -CODEC input pins for other compatible codecs: - * MIC3L - * MIC3R - * LINE1L - * LINE2L - * LINE1R - * LINE2R - -The pins can be used in referring sound node's audio-routing property. - -I2C example: - -#include <dt-bindings/gpio/gpio.h> - -tlv320aic3x: tlv320aic3x@1b { - compatible = "ti,tlv320aic3x"; - reg = <0x1b>; - - reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; - - AVDD-supply = <&regulator>; - IOVDD-supply = <&regulator>; - DRVDD-supply = <&regulator>; - DVDD-supply = <&regulator>; -}; - -SPI example: - -spi0: spi@f0000000 { - tlv320aic3x: codec@0 { - compatible = "ti,tlv320aic3x"; - reg = <0>; /* CS number */ - #sound-dai-cells = <0>; - spi-max-frequency = <1000000>; - - AVDD-supply = <&regulator>; - IOVDD-supply = <&regulator>; - DRVDD-supply = <&regulator>; - DVDD-supply = <&regulator>; - ai3x-ocmv = <0>; - }; -}; -- 2.17.1

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[PATCH V2 1/4] ASoC: amd: ps: implement api to retrieve acp device config
by Vijendar Mukunda 27 Dec '22

27 Dec '22

Implement API to retrieve acp device config and calculate platform device count and dev mask for platform device node creation. Currently for DMIC configuration, mask and dev count are calculated. Same api will be used to extend support for different ACP device configurations. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- since v1: - fixed below build error sound/soc/amd/ps/pci-ps.c:231:60: error: use of undeclared identifier 'ACP_DMIC_ADDR' - fixed below kernel warning >> sound/soc/amd/ps/pci-ps.c:135:6: warning: no previous prototype for >> function 'get_acp63_device_config' [-Wmissing-prototypes] sound/soc/amd/ps/acp63.h | 7 +++++++ sound/soc/amd/ps/pci-ps.c | 38 ++++++++++++++++++++++++++++++++++++-- 2 files changed, 43 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index 5e7f9c1c1b0e..b015e845e5fa 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -54,6 +54,11 @@ /* 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 +107,6 @@ struct acp63_dev_data { struct resource *res; bool acp63_audio_mode; struct platform_device *pdev[ACP63_DEVS]; + u16 pdev_mask; + u16 pdev_count; }; diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 489f01a20699..86af94f6f06e 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -132,6 +132,39 @@ 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; + } +} + static int snd_acp63_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -183,6 +216,7 @@ static int snd_acp63_probe(struct pci_dev *pci, if (ret) goto release_regions; val = acp63_readl(adata->acp63_base + ACP_PIN_CONFIG); + get_acp63_device_config(val, pci, adata); switch (val) { case ACP_CONFIG_0: case ACP_CONFIG_1: @@ -195,14 +229,14 @@ static int snd_acp63_probe(struct pci_dev *pci, default: /* Checking DMIC hardware*/ - adev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), 0x02, 0); + adev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), ACP63_DMIC_ADDR, 0); if (!adev) break; if (!acpi_dev_get_property(adev, "acp-audio-device-type", ACPI_TYPE_INTEGER, &obj) && - obj->integer.value == 2) { + obj->integer.value == ACP_DMIC_DEV) { adata->res = devm_kzalloc(&pci->dev, sizeof(struct resource), GFP_KERNEL); if (!adata->res) { ret = -ENOMEM; -- 2.34.1

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[PATCH 00/11] ASoC: SOF: Extend the IPC ops optionality
by Peter Ujfalusi 27 Dec '22

27 Dec '22

Hi, This series will extend the IPC ops optionality to cover it up to the existence of the top level ipc pointer itself. There is no functionality change. The reason for the extended optionality is that we have "DSPless" debug/development support coming up (currently it is in SOF's topic/sof-dev stable branch) initially supporting Intel's HDA platforms. As the name suggests, in this mode the DSP is completely ignored by the linux driver stack (no firmware loaded, only using HDA directly). The DSPless mode is aimed to help us to verify our Linux stack on new platforms where the firmware is not yet in the state that we can reliably use it, but the hardware and programming flows can be tested already. There is no plan to make DSPless a production target for SOF Linux stack. While this is preparatory series aimed to unblock the DSPless support, it has been integrated into sof-dev separately and we have lots of new features depending on it (went in between this set and the DSPless support). I still have some minor tasks to complete for the DSPless to make it a bit more versatile, but I don't want to block other, stable features for upstreaming. Regards, Peter --- Peter Ujfalusi (11): ASoC: SOF: sof-audio: Treat tplg_ops->route_setup() as optional ASoC: SOF: sof-audio: Update documentation for sof_ipc_tplg_ops ASoC: SOF: Add helper macro to be used to get an IPC ops ASoC: SOF: pcm: Extend the optionality of IPC ops to IPC as well ASoC: SOF: control: Extend the optionality of IPC ops to IPC as well ASoC: SOF: sof-audio: Extend the optionality of IPC ops to IPC as well ASoC: SOF: topology: Extend the optionality of IPC ops to IPC as well ASoC: SOF: pm: Extend the optionality of IPC ops to IPC as well ASoC: SOF: sof-priv: Mark fw_tracing ops optional in documentation ASoC: SOF: trace: Use sof_ipc_get_ops() in sof_fw_trace_init ASoC: SOF: trace: No need to check for op pointer in sof_fw_trace_free() sound/soc/sof/control.c | 44 +++++++++---------- sound/soc/sof/pcm.c | 18 ++++---- sound/soc/sof/pm.c | 12 +++--- sound/soc/sof/sof-audio.c | 57 +++++++++++++----------- sound/soc/sof/sof-audio.h | 6 ++- sound/soc/sof/sof-priv.h | 6 ++- sound/soc/sof/topology.c | 91 +++++++++++++++++++++++---------------- sound/soc/sof/trace.c | 8 ++-- 8 files changed, 138 insertions(+), 104 deletions(-) -- 2.39.0

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[PATCH] ASoC: SOF: core: Print out the value of sof_debug if it is set
by Peter Ujfalusi 27 Dec '22

27 Dec '22

The sof_debug value is set by the user, developer intentionally. To save time on figuring out what value has been passed to the kernel by the user, developer, print it out if it is not 0. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/sof/core.c | 3 +++ 1 file changed, 3 insertions(+) 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", -- 2.39.0

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[PATCH] ASoC: SOF: core: Print out the value of sof_debug if it is set
by Peter Ujfalusi 27 Dec '22

27 Dec '22

The sof_debug value is set by the user, developer intentionally. To save time on figuring out what value has been passed to the kernel by the user, developer, print it out if it is not 0. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/sof/core.c | 3 +++ 1 file changed, 3 insertions(+) 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", -- 2.39.0

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[PATCH] ASoC: SOF: IPC3 topology: Print the conflicting bytes sizes
by Peter Ujfalusi 27 Dec '22

27 Dec '22

From: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> The error "Conflict in bytes vs. priv size." is too brief. With the printed sizes it's a lot easier to find the size issue in for binary control if such happens. Signed-off-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc3-topology.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) 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; } -- 2.39.0

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[PATCH v3 1/3] ASoC: wm8940: Remove warning when no plat data present
by Lukasz Majewski 27 Dec '22

27 Dec '22

The lack of platform data in the contemporary Linux shall not be the reason to display warnings to the kernel logs. Signed-off-by: Lukasz Majewski <lukma(a)denx.de> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes for v2: - None Changes for v3: - None --- sound/soc/codecs/wm8940.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 8dac9fd88547..0b59020d747f 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -709,9 +709,7 @@ 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) -- 2.20.1

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[PATCH] ASoC: SOF: mediatek: mt8195: remove a redundant comparison of sram
by YC Hung 27 Dec '22

27 Dec '22

DSP SRAM is not used for audio shared buffer between host and DSP so TOTAL_SIZE_SHARED_SRAM_FROM_TAIL is zero. Remove the definition and redundant comparison to fix coverity "unsigned compared against 0". Signed-off-by: YC Hung <yc.hung(a)mediatek.com> --- sound/soc/sof/mediatek/mt8195/mt8195.c | 5 ----- sound/soc/sof/mediatek/mt8195/mt8195.h | 2 -- 2 files changed, 7 deletions(-) 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*/ -- 2.18.0

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[PATCH 0/4] ASoC: Fixes for WM8940 codec
by Lukasz Majewski 27 Dec '22

27 Dec '22

This patch series provides fixes for WM8940 codec. The most notable change is the clock rewrite, so this driver now can either generate proper clock frequency by itself or use one provided from the clock subsystem of the SoC. Lukasz Majewski (4): ASoC: wm8940: Remove warning when no plat data present ASoC: wm8940: Rewrite code to set proper clocks ASoC: wm8940: Mute also the speaker output ASoC: wm8940: Read chip ID when wm8940 codec probing sound/soc/codecs/wm8940.c | 129 +++++++++++++++++++++++++++++++------- sound/soc/codecs/wm8940.h | 6 ++ 2 files changed, 111 insertions(+), 24 deletions(-) -- 2.20.1

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Re: [PATCH 2/2] ASoC: da7213: Add support for mono, set frame width to 32 when possible
by Mark Brown 27 Dec '22

27 Dec '22

On Wed, 14 Dec 2022 04:40:58 +0000, David Rau wrote: > This adds the DAI mono mode support and set the frame width to 32 > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [2/2] ASoC: da7213: Add support for mono, set frame width to 32 when possible commit: c89e652e84f636354213aae42490c7f77a0eadde All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

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