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[PATCH] [RFCv3] ASoC: Add Rockchip rk817 audio CODEC support
by Chris Morgan 17 Mar '21

17 Mar '21

Add support for the Rockchip rk817 audio codec integrated into the rk817 PMIC. This is based on the sources provided by Rockchip from their BSP kernel. Would someone advise what they think the best way to get the widgets set up properly would be? I'm still new to this and I don't fully understand all the macros (the alsa-project.org documentation only gets me so far). Changes from v2: Switched to DAPM macros to set audio path. Eliminated unnecessary register value sets. Removed devicetree binding since device is present for every rk817 PMIC. Updated binding documentation to yaml format. Separated mfd changes to separate commit. Notes: headphone detect works now, however speaker path does not disable automatically when headphones are used. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- .../bindings/sound/rockchip,rk817-codec.yaml | 60 ++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 675 ++++++++++++++++++ sound/soc/codecs/rk817_codec.h | 93 +++ 5 files changed, 834 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml create mode 100644 sound/soc/codecs/rk817_codec.c create mode 100644 sound/soc/codecs/rk817_codec.h diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml new file mode 100644 index 000000000000..6bdfb917f76e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-rk817.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk817 audio codec + +description: + The rk817 codec is an I2C codec integrated with every Rockchip + rk817 PMIC MFD. + +properties: + compatible: + enum: + - rockchip,rk817-codec + + clocks: + maxItems: 1 + + clock-names: + const: "mclk" + + mic-in-differential: + description: the microphone is in differential mode. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - clocks + - clock-names + +additionalProperties: false + +examples: + - | +rk817: pmic@20 { + compatible = "rockchip,rk817"; + reg = <0x20>; + interrupt-parent = <&gpio0>; + interrupts = <7 IRQ_TYPE_LEVEL_LOW>; + pinctrl-names = "default"; + pinctrl-0 = <&pmic_int>; + + ........ + + rk817_codec: codec { + #sound-dai-cells = <0>; + compatible = "rockchip,rk817-codec"; + clocks = <&cru SCLK_I2S_8CH_OUT>; + clock-names = "mclk"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s_8ch_mclk>; + mic-in-differential; + status = "okay"; + }; + + ........ + +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e4cf14e66a51..d835145ad65e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -155,6 +155,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_PCM512x_I2C imply SND_SOC_PCM512x_SPI imply SND_SOC_RK3328 + imply SND_SOC_RK817 imply SND_SOC_RT274 imply SND_SOC_RT286 imply SND_SOC_RT298 @@ -1059,6 +1060,11 @@ config SND_SOC_RK3328 tristate "Rockchip RK3328 audio CODEC" select REGMAP_MMIO +config SND_SOC_RK817 + tristate "Rockchip RK817 audio CODEC" + depends on MFD_RK808 + select REGMAP_I2C + config SND_SOC_RL6231 tristate default y if SND_SOC_RT5514=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 81357dc62ea0..4e5e9b28775c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -165,6 +165,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rk3328-objs := rk3328_codec.o +snd-soc-rk817-objs := rk817_codec.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1011-objs := rt1011.o @@ -479,6 +480,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o +obj-$(CONFIG_SND_SOC_RK817) += snd-soc-rk817.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c new file mode 100644 index 000000000000..469b013e1e63 --- /dev/null +++ b/sound/soc/codecs/rk817_codec.c @@ -0,0 +1,673 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rk817 ALSA SoC Audio driver +// +// Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/mfd/rk808.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "rk817_codec.h" + +struct rk817_codec_priv { + struct snd_soc_component *component; + struct regmap *regmap; + struct rk808 *rk817; + struct clk *mclk; + unsigned int stereo_sysclk; + bool mic_in_differential; +}; + +static const struct reg_default rk817_reg_defaults[] = { + { RK817_CODEC_DTOP_VUCTL, 0x03 }, + { RK817_CODEC_DTOP_VUCTIME, 0x00 }, + { RK817_CODEC_DTOP_LPT_SRST, 0x00 }, + { RK817_CODEC_DTOP_DIGEN_CLKE, 0x00 }, + /* from vendor driver, CODEC_AREF_RTCFG0 not defined in data sheet */ + { RK817_CODEC_AREF_RTCFG0, 0x00 }, + { RK817_CODEC_AREF_RTCFG1, 0x06 }, + { RK817_CODEC_AADC_CFG0, 0xc8 }, + /* from vendor driver, CODEC_AADC_CFG1 not defined in data sheet */ + { RK817_CODEC_AADC_CFG1, 0x00 }, + { RK817_CODEC_DADC_VOLL, 0x00 }, + { RK817_CODEC_DADC_VOLR, 0x00 }, + { RK817_CODEC_DADC_SR_ACL0, 0x00 }, + { RK817_CODEC_DADC_ALC1, 0x00 }, + { RK817_CODEC_DADC_ALC2, 0x00 }, + { RK817_CODEC_DADC_NG, 0x00 }, + { RK817_CODEC_DADC_HPF, 0x00 }, + { RK817_CODEC_DADC_RVOLL, 0xff }, + { RK817_CODEC_DADC_RVOLR, 0xff }, + { RK817_CODEC_AMIC_CFG0, 0x70 }, + { RK817_CODEC_AMIC_CFG1, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, 0x66 }, + { RK817_CODEC_DMIC_LMT1, 0x00 }, + { RK817_CODEC_DMIC_LMT2, 0x00 }, + { RK817_CODEC_DMIC_NG1, 0x00 }, + { RK817_CODEC_DMIC_NG2, 0x00 }, + /* from vendor driver, CODEC_ADAC_CFG0 not defined in data sheet */ + { RK817_CODEC_ADAC_CFG0, 0x00 }, + { RK817_CODEC_ADAC_CFG1, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, 0x82 }, + { RK817_CODEC_DDAC_VOLL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, 0x00 }, + { RK817_CODEC_DDAC_LMT1, 0x00 }, + { RK817_CODEC_DDAC_LMT2, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, 0xff }, + { RK817_CODEC_DDAC_RVOLR, 0xff }, + { RK817_CODEC_AHP_ANTI0, 0x00 }, + { RK817_CODEC_AHP_ANTI1, 0x00 }, + { RK817_CODEC_AHP_CFG0, 0xe0 }, + { RK817_CODEC_AHP_CFG1, 0x1f }, + { RK817_CODEC_AHP_CP, 0x09 }, + { RK817_CODEC_ACLASSD_CFG1, 0x69 }, + { RK817_CODEC_ACLASSD_CFG2, 0x44 }, + { RK817_CODEC_APLL_CFG0, 0x04 }, + { RK817_CODEC_APLL_CFG1, 0x00 }, + { RK817_CODEC_APLL_CFG2, 0x30 }, + { RK817_CODEC_APLL_CFG3, 0x19 }, + { RK817_CODEC_APLL_CFG4, 0x65 }, + { RK817_CODEC_APLL_CFG5, 0x01 }, + { RK817_CODEC_DI2S_CKM, 0x01 }, + { RK817_CODEC_DI2S_RSD, 0x00 }, + { RK817_CODEC_DI2S_RXCR1, 0x00 }, + { RK817_CODEC_DI2S_RXCR2, 0x17 }, + { RK817_CODEC_DI2S_RXCMD_TSD, 0x00 }, + { RK817_CODEC_DI2S_TXCR1, 0x00 }, + { RK817_CODEC_DI2S_TXCR2, 0x17 }, + { RK817_CODEC_DI2S_TXCR3_TXCMD, 0x00 }, +}; + +static bool rk817_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RK817_CODEC_DTOP_LPT_SRST: + return true; + default: + return false; + } +} + +static bool rk817_codec_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RK817_CODEC_DTOP_VUCTL: + case RK817_CODEC_DTOP_VUCTIME: + case RK817_CODEC_DTOP_LPT_SRST: + case RK817_CODEC_DTOP_DIGEN_CLKE: + case RK817_CODEC_AREF_RTCFG0: + case RK817_CODEC_AREF_RTCFG1: + case RK817_CODEC_AADC_CFG0: + case RK817_CODEC_AADC_CFG1: + case RK817_CODEC_DADC_VOLL: + case RK817_CODEC_DADC_VOLR: + case RK817_CODEC_DADC_SR_ACL0: + case RK817_CODEC_DADC_ALC1: + case RK817_CODEC_DADC_ALC2: + case RK817_CODEC_DADC_NG: + case RK817_CODEC_DADC_HPF: + case RK817_CODEC_DADC_RVOLL: + case RK817_CODEC_DADC_RVOLR: + case RK817_CODEC_AMIC_CFG0: + case RK817_CODEC_AMIC_CFG1: + case RK817_CODEC_DMIC_PGA_GAIN: + case RK817_CODEC_DMIC_LMT1: + case RK817_CODEC_DMIC_LMT2: + case RK817_CODEC_DMIC_NG1: + case RK817_CODEC_DMIC_NG2: + case RK817_CODEC_ADAC_CFG0: + case RK817_CODEC_ADAC_CFG1: + case RK817_CODEC_DDAC_POPD_DACST: + case RK817_CODEC_DDAC_VOLL: + case RK817_CODEC_DDAC_VOLR: + case RK817_CODEC_DDAC_SR_LMT0: + case RK817_CODEC_DDAC_LMT1: + case RK817_CODEC_DDAC_LMT2: + case RK817_CODEC_DDAC_MUTE_MIXCTL: + case RK817_CODEC_DDAC_RVOLL: + case RK817_CODEC_DDAC_RVOLR: + case RK817_CODEC_AHP_ANTI0: + case RK817_CODEC_AHP_ANTI1: + case RK817_CODEC_AHP_CFG0: + case RK817_CODEC_AHP_CFG1: + case RK817_CODEC_AHP_CP: + case RK817_CODEC_ACLASSD_CFG1: + case RK817_CODEC_ACLASSD_CFG2: + case RK817_CODEC_APLL_CFG0: + case RK817_CODEC_APLL_CFG1: + case RK817_CODEC_APLL_CFG2: + case RK817_CODEC_APLL_CFG3: + case RK817_CODEC_APLL_CFG4: + case RK817_CODEC_APLL_CFG5: + case RK817_CODEC_DI2S_CKM: + case RK817_CODEC_DI2S_RSD: + case RK817_CODEC_DI2S_RXCR1: + case RK817_CODEC_DI2S_RXCR2: + case RK817_CODEC_DI2S_RXCMD_TSD: + case RK817_CODEC_DI2S_TXCR1: + case RK817_CODEC_DI2S_TXCR2: + case RK817_CODEC_DI2S_TXCR3_TXCMD: + return true; + default: + return false; + } +} + +/* + * This sets the codec up with the values defined in the default implementation including the APLL + * from the Rockchip vendor kernel. I do not know if these values are universal despite differing + * from the default values defined above and taken from the datasheet, or implementation specific. + * I don't have another implementation to compare from the Rockchip sources. Hard-coding this for + * now. + */ + +static int rk817_init(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + snd_soc_component_write(component, RK817_CODEC_APLL_CFG1, 0x58); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG2, 0x2d); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG3, 0x0c); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG4, 0xa5); + snd_soc_component_write(component, RK817_CODEC_DDAC_POPD_DACST, 0x02); + snd_soc_component_write(component, RK817_CODEC_DDAC_SR_LMT0, 0x02); + snd_soc_component_write(component, RK817_CODEC_DMIC_PGA_GAIN, 0x99); + snd_soc_component_write(component, RK817_CODEC_DADC_SR_ACL0, 0x02); + snd_soc_component_write(component, RK817_CODEC_DTOP_VUCTIME, 0xf4); + if (rk817->mic_in_differential) { + snd_soc_component_update_bits(component, RK817_CODEC_AMIC_CFG0, MIC_DIFF_MASK, + MIC_DIFF_EN); + }; + return 0; +} + +/* + * DDAC/DADC L/R volume setting + * 0db~-95db,0.375db/step,for example: + * 0: 0dB + * 0x0a: -3.75dB + * 0x7d: -46dB + * 0xff: -95dB + */ + +static const DECLARE_TLV_DB_MINMAX(rk817_vol_tlv, -9500, 0); + +static const struct snd_kcontrol_new rk817_dac_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("Master Playback Volume", RK817_CODEC_DDAC_VOLL, + RK817_CODEC_DDAC_VOLR, 0, 0x00, 0xff, 1, rk817_vol_tlv), + SOC_DOUBLE_R_RANGE_TLV("Master Capture Volume", RK817_CODEC_DADC_VOLL, + RK817_CODEC_DADC_VOLR, 0, 0x00, 0xff, 1, rk817_vol_tlv) +}; + +static const char *dac_mux_text[] = { + "HP", + "SPK", +}; + +static SOC_ENUM_SINGLE_VIRT_DECL(dac_enum, dac_mux_text); + +static const struct snd_kcontrol_new dac_mux = + SOC_DAPM_ENUM("Playback Mux", dac_enum); + +static const struct snd_soc_dapm_widget rk817_dapm_widgets[] = { + + /* capture/playback common */ + SND_SOC_DAPM_SUPPLY("LDO Regulator", RK817_CODEC_AREF_RTCFG1, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBIAS Block", RK817_CODEC_AREF_RTCFG1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("VAvg Buffer", RK817_CODEC_AREF_RTCFG1, 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL Power", RK817_CODEC_APLL_CFG5, 0, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX1 Transfer Start", RK817_CODEC_DI2S_RXCMD_TSD, 5, 0, NULL, 0), + + /* capture path common */ + SND_SOC_DAPM_SUPPLY("ADC Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC Power On", RK817_CODEC_AMIC_CFG0, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX3 Transfer Start", RK817_CODEC_DI2S_TXCR3_TXCMD, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX3 Right Justified", RK817_CODEC_DI2S_TXCR3_TXCMD, 3, 0, NULL, 0), + + /* capture path L */ + SND_SOC_DAPM_ADC("ADC L", "Capture", RK817_CODEC_AADC_CFG0, 7, 1), + SND_SOC_DAPM_SUPPLY("PGA L Power On", RK817_CODEC_AMIC_CFG0, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost L1", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost L2", RK817_CODEC_AMIC_CFG0, 2, 0, NULL, 0), + + /* capture path R */ + SND_SOC_DAPM_ADC("ADC R", "Capture", RK817_CODEC_AADC_CFG0, 6, 1), + SND_SOC_DAPM_SUPPLY("PGA R Power On", RK817_CODEC_AMIC_CFG0, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost R1", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost R2", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + + /* playback path common */ + SND_SOC_DAPM_SUPPLY("DAC Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S RX Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S RX Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Bias", RK817_CODEC_ADAC_CFG1, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Mute Off", RK817_CODEC_DDAC_MUTE_MIXCTL, 0, 1, NULL, 0), + + /* playback path speaker */ + SND_SOC_DAPM_SUPPLY("Class D Mode", RK817_CODEC_DDAC_MUTE_MIXCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("High Pass Filter", RK817_CODEC_DDAC_MUTE_MIXCTL, 7, 0, NULL, 0), + SND_SOC_DAPM_DAC("SPK DAC", "Playback", RK817_CODEC_ADAC_CFG1, 2, 1), + SND_SOC_DAPM_SUPPLY("Enable Class D", RK817_CODEC_ACLASSD_CFG1, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Disable Class D Mute Ramp", RK817_CODEC_ACLASSD_CFG1, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D Mute Rate 1", RK817_CODEC_ACLASSD_CFG1, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D Mute Rate 2", RK817_CODEC_ACLASSD_CFG1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPP 2", RK817_CODEC_ACLASSD_CFG2, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPP 3", RK817_CODEC_ACLASSD_CFG2, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPN 2", RK817_CODEC_ACLASSD_CFG2, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPN 3", RK817_CODEC_ACLASSD_CFG2, 0, 0, NULL, 0), + + /* playback path headphones */ + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", RK817_CODEC_AHP_CP, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone CP Discharge LDO", RK817_CODEC_AHP_CP, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone OStage", RK817_CODEC_AHP_CFG0, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Pre Amp", RK817_CODEC_AHP_CFG0, 5, 1, NULL, 0), + SND_SOC_DAPM_DAC("DAC L", "Playback", RK817_CODEC_ADAC_CFG1, 1, 1), + SND_SOC_DAPM_DAC("DAC R", "Playback", RK817_CODEC_ADAC_CFG1, 0, 1), + + /* Mux for input/output path selection */ + SND_SOC_DAPM_MUX("Playback Mux", SND_SOC_NOPM, 1, 0, &dac_mux), + + /* Pins for Simple Card Bindings */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("SPKO"), +}; + +static const struct snd_soc_dapm_route rk817_dapm_routes[] = { + + /* capture path */ + /* left mic */ + {"ADC L", NULL, "LDO Regulator"}, + {"ADC L", NULL, "IBIAS Block"}, + {"ADC L", NULL, "VAvg Buffer"}, + {"ADC L", NULL, "PLL Power"}, + {"ADC L", NULL, "ADC Clock"}, + {"ADC L", NULL, "I2S TX Clock"}, + {"ADC L", NULL, "ADC Channel Enable"}, + {"ADC L", NULL, "I2S TX Channel Enable"}, + {"ADC L", NULL, "I2S TX1 Transfer Start"}, + {"MICL", NULL, "MIC Power On"}, + {"MICL", NULL, "PGA L Power On"}, + {"MICL", NULL, "Mic Boost L1"}, + {"MICL", NULL, "Mic Boost L2"}, + {"MICL", NULL, "I2S TX3 Transfer Start"}, + {"MICL", NULL, "I2S TX3 Right Justified"}, + {"ADC L", NULL, "MICL"}, + + /* right mic */ + {"ADC R", NULL, "LDO Regulator"}, + {"ADC R", NULL, "IBIAS Block"}, + {"ADC R", NULL, "VAvg Buffer"}, + {"ADC R", NULL, "PLL Power"}, + {"ADC R", NULL, "ADC Clock"}, + {"ADC R", NULL, "I2S TX Clock"}, + {"ADC R", NULL, "ADC Channel Enable"}, + {"ADC R", NULL, "I2S TX Channel Enable"}, + {"ADC R", NULL, "I2S TX1 Transfer Start"}, + {"MICR", NULL, "MIC Power On"}, + {"MICR", NULL, "PGA R Power On"}, + {"MICR", NULL, "Mic Boost R1"}, + {"MICR", NULL, "Mic Boost R2"}, + {"MICR", NULL, "I2S TX3 Transfer Start"}, + {"MICR", NULL, "I2S TX3 Right Justified"}, + {"ADC R", NULL, "MICR"}, + + /* playback path */ + /* speaker path */ + {"SPK DAC", NULL, "LDO Regulator"}, + {"SPK DAC", NULL, "IBIAS Block"}, + {"SPK DAC", NULL, "VAvg Buffer"}, + {"SPK DAC", NULL, "PLL Power"}, + {"SPK DAC", NULL, "I2S TX1 Transfer Start"}, + {"SPK DAC", NULL, "DAC Clock"}, + {"SPK DAC", NULL, "I2S RX Clock"}, + {"SPK DAC", NULL, "DAC Channel Enable"}, + {"SPK DAC", NULL, "I2S RX Channel Enable"}, + {"SPK DAC", NULL, "Class D Mode"}, + {"SPK DAC", NULL, "DAC Bias"}, + {"SPK DAC", NULL, "DAC Mute Off"}, + {"SPK DAC", NULL, "Enable Class D"}, + {"SPK DAC", NULL, "Disable Class D Mute Ramp"}, + {"SPK DAC", NULL, "Class D Mute Rate 1"}, + {"SPK DAC", NULL, "Class D Mute Rate 2"}, + {"SPK DAC", NULL, "Class D OCPP 2"}, + {"SPK DAC", NULL, "Class D OCPP 3"}, + {"SPK DAC", NULL, "Class D OCPN 2"}, + {"SPK DAC", NULL, "Class D OCPN 3"}, + {"SPK DAC", NULL, "High Pass Filter"}, + + /* headphone path L */ + {"DAC L", NULL, "LDO Regulator"}, + {"DAC L", NULL, "IBIAS Block"}, + {"DAC L", NULL, "VAvg Buffer"}, + {"DAC L", NULL, "PLL Power"}, + {"DAC L", NULL, "I2S TX1 Transfer Start"}, + {"DAC L", NULL, "DAC Clock"}, + {"DAC L", NULL, "I2S RX Clock"}, + {"DAC L", NULL, "DAC Channel Enable"}, + {"DAC L", NULL, "I2S RX Channel Enable"}, + {"DAC L", NULL, "DAC Bias"}, + {"DAC L", NULL, "DAC Mute Off"}, + {"DAC L", NULL, "Headphone Charge Pump"}, + {"DAC L", NULL, "Headphone CP Discharge LDO"}, + {"DAC L", NULL, "Headphone OStage"}, + {"DAC L", NULL, "Headphone Pre Amp"}, + + /* headphone path R */ + {"DAC R", NULL, "LDO Regulator"}, + {"DAC R", NULL, "IBIAS Block"}, + {"DAC R", NULL, "VAvg Buffer"}, + {"DAC R", NULL, "PLL Power"}, + {"DAC R", NULL, "I2S TX1 Transfer Start"}, + {"DAC R", NULL, "DAC Clock"}, + {"DAC R", NULL, "I2S RX Clock"}, + {"DAC R", NULL, "DAC Channel Enable"}, + {"DAC R", NULL, "I2S RX Channel Enable"}, + {"DAC R", NULL, "DAC Bias"}, + {"DAC R", NULL, "DAC Mute Off"}, + {"DAC R", NULL, "Headphone Charge Pump"}, + {"DAC R", NULL, "Headphone CP Discharge LDO"}, + {"DAC R", NULL, "Headphone OStage"}, + {"DAC R", NULL, "Headphone Pre Amp"}, + + /* mux path for output selection */ + {"Playback Mux", "HP", "DAC L"}, + {"Playback Mux", "HP", "DAC R"}, + {"Playback Mux", "SPK", "SPK DAC"}, + {"SPKO", NULL, "Playback Mux"}, + {"HPOL", NULL, "Playback Mux"}, + {"HPOR", NULL, "Playback Mux"}, +}; + +static int rk817_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 rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + rk817->stereo_sysclk = freq; + + return 0; +} + +static int rk817_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + unsigned int i2s_mst = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + i2s_mst |= RK817_I2S_MODE_SLV; + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2s_mst |= RK817_I2S_MODE_MST; + break; + default: + dev_err(component->dev, "%s : set master mask failed!\n", __func__); + return -EINVAL; + } + + snd_soc_component_update_bits(component, RK817_CODEC_DI2S_CKM, + RK817_I2S_MODE_MASK, i2s_mst); + + return 0; +} + +static int rk817_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; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_16BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_16BITS); + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_24BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_24BITS); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int rk817_digital_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_component *component = dai->component; + + if (mute) + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_ENABLE); + else + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_DISABLE); + + return 0; +} + +#define RK817_PLAYBACK_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_CAPTURE_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops rk817_dai_ops = { + .hw_params = rk817_hw_params, + .set_fmt = rk817_set_dai_fmt, + .set_sysclk = rk817_set_dai_sysclk, + .mute_stream = rk817_digital_mute, + .no_capture_mute = 1, +}; + +static struct snd_soc_dai_driver rk817_dai[] = { + { + .name = "rk817-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = RK817_PLAYBACK_RATES, + .formats = RK817_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RK817_CAPTURE_RATES, + .formats = RK817_FORMATS, + }, + .ops = &rk817_dai_ops, + }, +}; + +static int rk817_probe(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + snd_soc_component_init_regmap(component, rk817->regmap); + rk817->component = component; + + clk_prepare_enable(rk817->mclk); + + snd_soc_component_write(component, RK817_CODEC_DTOP_LPT_SRST, 0x40); + rk817_init(component); + + return 0; +} + +static void rk817_remove(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + snd_soc_component_exit_regmap(component); + clk_disable_unprepare(rk817->mclk); + mdelay(10); + +} + +static const struct snd_soc_component_driver soc_codec_dev_rk817 = { + .probe = rk817_probe, + .remove = rk817_remove, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, + .controls = rk817_dac_controls, + .num_controls = ARRAY_SIZE(rk817_dac_controls), + .dapm_routes = rk817_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rk817_dapm_routes), + .dapm_widgets = rk817_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rk817_dapm_widgets), +}; + +static int rk817_codec_parse_dt_property(struct device *dev, + struct rk817_codec_priv *rk817) +{ + struct device_node *node = dev->parent->of_node; + + if (!node) { + dev_err(dev, "%s() dev->parent->of_node is NULL\n", + __func__); + return -ENODEV; + } + + node = of_get_child_by_name(dev->parent->of_node, "codec"); + if (!node) { + dev_err(dev, "%s() Can not get child: codec\n", + __func__); + return -ENODEV; + } + + rk817->mic_in_differential = + of_property_read_bool(node, "mic-in-differential"); + + return 0; +} + +static const struct regmap_config rk817_codec_regmap_config = { + .name = "rk817-codec", + .reg_bits = 8, + .val_bits = 8, + .reg_stride = 1, + .max_register = 0x4f, + .cache_type = REGCACHE_NONE, + .volatile_reg = rk817_volatile_register, + .writeable_reg = rk817_codec_register, + .readable_reg = rk817_codec_register, + .reg_defaults = rk817_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rk817_reg_defaults), +}; + +static int rk817_platform_probe(struct platform_device *pdev) +{ + struct rk808 *rk817 = dev_get_drvdata(pdev->dev.parent); + struct rk817_codec_priv *rk817_codec_data; + int ret; + + if (!rk817) { + dev_err(&pdev->dev, "%s : rk817 is NULL\n", __func__); + return -EINVAL; + } + + rk817_codec_data = devm_kzalloc(&pdev->dev, + sizeof(struct rk817_codec_priv), + GFP_KERNEL); + if (!rk817_codec_data) + return -ENOMEM; + + platform_set_drvdata(pdev, rk817_codec_data); + + ret = rk817_codec_parse_dt_property(&pdev->dev, rk817_codec_data); + if (ret < 0) { + dev_err(&pdev->dev, "%s() parse device tree property error %d\n", + __func__, ret); + goto err_; + } + + rk817_codec_data->regmap = devm_regmap_init_i2c(rk817->i2c, + &rk817_codec_regmap_config); + if (IS_ERR(rk817_codec_data->regmap)) { + ret = PTR_ERR(rk817_codec_data->regmap); + dev_err(&pdev->dev, "failed to allocate register map: %d\n", + ret); + goto err_; + } + + rk817_codec_data->mclk = devm_clk_get(&pdev->dev, "mclk"); + if (IS_ERR(rk817_codec_data->mclk)) { + dev_err(&pdev->dev, "Unable to get mclk\n"); + ret = -ENXIO; + goto err_; + } + + clk_prepare_enable(rk817_codec_data->mclk); + + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_dev_rk817, + rk817_dai, ARRAY_SIZE(rk817_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "%s() register codec error %d\n", + __func__, ret); + goto err_; + } + + return 0; +err_: + + return ret; +} + +MODULE_DEVICE_TABLE(of, rk817_codec_dt_ids); + +static struct platform_driver rk817_codec_driver = { + .driver = { + .name = "rk817-codec", + }, + .probe = rk817_platform_probe, +}; + +module_platform_driver(rk817_codec_driver); + +MODULE_DESCRIPTION("ASoC RK817 codec driver"); +MODULE_AUTHOR("binyuan <kevan.lan(a)rock-chips.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rk817_codec.h b/sound/soc/codecs/rk817_codec.h new file mode 100644 index 000000000000..831c80972f92 --- /dev/null +++ b/sound/soc/codecs/rk817_codec.h @@ -0,0 +1,93 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * rk817 ALSA SoC Audio driver + * + */ Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + +#ifndef __RK817_CODEC_H__ +#define __RK817_CODEC_H__ + +/* codec register */ +#define RK817_CODEC_BASE 0x0000 + +#define RK817_CODEC_DTOP_VUCTL (RK817_CODEC_BASE + 0x12) +#define RK817_CODEC_DTOP_VUCTIME (RK817_CODEC_BASE + 0x13) +#define RK817_CODEC_DTOP_LPT_SRST (RK817_CODEC_BASE + 0x14) +#define RK817_CODEC_DTOP_DIGEN_CLKE (RK817_CODEC_BASE + 0x15) +#define RK817_CODEC_AREF_RTCFG0 (RK817_CODEC_BASE + 0x16) +#define RK817_CODEC_AREF_RTCFG1 (RK817_CODEC_BASE + 0x17) +#define RK817_CODEC_AADC_CFG0 (RK817_CODEC_BASE + 0x18) +#define RK817_CODEC_AADC_CFG1 (RK817_CODEC_BASE + 0x19) +#define RK817_CODEC_DADC_VOLL (RK817_CODEC_BASE + 0x1a) +#define RK817_CODEC_DADC_VOLR (RK817_CODEC_BASE + 0x1b) +#define RK817_CODEC_DADC_SR_ACL0 (RK817_CODEC_BASE + 0x1e) +#define RK817_CODEC_DADC_ALC1 (RK817_CODEC_BASE + 0x1f) +#define RK817_CODEC_DADC_ALC2 (RK817_CODEC_BASE + 0x20) +#define RK817_CODEC_DADC_NG (RK817_CODEC_BASE + 0x21) +#define RK817_CODEC_DADC_HPF (RK817_CODEC_BASE + 0x22) +#define RK817_CODEC_DADC_RVOLL (RK817_CODEC_BASE + 0x23) +#define RK817_CODEC_DADC_RVOLR (RK817_CODEC_BASE + 0x24) +#define RK817_CODEC_AMIC_CFG0 (RK817_CODEC_BASE + 0x27) +#define RK817_CODEC_AMIC_CFG1 (RK817_CODEC_BASE + 0x28) +#define RK817_CODEC_DMIC_PGA_GAIN (RK817_CODEC_BASE + 0x29) +#define RK817_CODEC_DMIC_LMT1 (RK817_CODEC_BASE + 0x2a) +#define RK817_CODEC_DMIC_LMT2 (RK817_CODEC_BASE + 0x2b) +#define RK817_CODEC_DMIC_NG1 (RK817_CODEC_BASE + 0x2c) +#define RK817_CODEC_DMIC_NG2 (RK817_CODEC_BASE + 0x2d) +#define RK817_CODEC_ADAC_CFG0 (RK817_CODEC_BASE + 0x2e) +#define RK817_CODEC_ADAC_CFG1 (RK817_CODEC_BASE + 0x2f) +#define RK817_CODEC_DDAC_POPD_DACST (RK817_CODEC_BASE + 0x30) +#define RK817_CODEC_DDAC_VOLL (RK817_CODEC_BASE + 0x31) +#define RK817_CODEC_DDAC_VOLR (RK817_CODEC_BASE + 0x32) +#define RK817_CODEC_DDAC_SR_LMT0 (RK817_CODEC_BASE + 0x35) +#define RK817_CODEC_DDAC_LMT1 (RK817_CODEC_BASE + 0x36) +#define RK817_CODEC_DDAC_LMT2 (RK817_CODEC_BASE + 0x37) +#define RK817_CODEC_DDAC_MUTE_MIXCTL (RK817_CODEC_BASE + 0x38) +#define RK817_CODEC_DDAC_RVOLL (RK817_CODEC_BASE + 0x39) +#define RK817_CODEC_DDAC_RVOLR (RK817_CODEC_BASE + 0x3a) +#define RK817_CODEC_AHP_ANTI0 (RK817_CODEC_BASE + 0x3b) +#define RK817_CODEC_AHP_ANTI1 (RK817_CODEC_BASE + 0x3c) +#define RK817_CODEC_AHP_CFG0 (RK817_CODEC_BASE + 0x3d) +#define RK817_CODEC_AHP_CFG1 (RK817_CODEC_BASE + 0x3e) +#define RK817_CODEC_AHP_CP (RK817_CODEC_BASE + 0x3f) +#define RK817_CODEC_ACLASSD_CFG1 (RK817_CODEC_BASE + 0x40) +#define RK817_CODEC_ACLASSD_CFG2 (RK817_CODEC_BASE + 0x41) +#define RK817_CODEC_APLL_CFG0 (RK817_CODEC_BASE + 0x42) +#define RK817_CODEC_APLL_CFG1 (RK817_CODEC_BASE + 0x43) +#define RK817_CODEC_APLL_CFG2 (RK817_CODEC_BASE + 0x44) +#define RK817_CODEC_APLL_CFG3 (RK817_CODEC_BASE + 0x45) +#define RK817_CODEC_APLL_CFG4 (RK817_CODEC_BASE + 0x46) +#define RK817_CODEC_APLL_CFG5 (RK817_CODEC_BASE + 0x47) +#define RK817_CODEC_DI2S_CKM (RK817_CODEC_BASE + 0x48) +#define RK817_CODEC_DI2S_RSD (RK817_CODEC_BASE + 0x49) +#define RK817_CODEC_DI2S_RXCR1 (RK817_CODEC_BASE + 0x4a) +#define RK817_CODEC_DI2S_RXCR2 (RK817_CODEC_BASE + 0x4b) +#define RK817_CODEC_DI2S_RXCMD_TSD (RK817_CODEC_BASE + 0x4c) +#define RK817_CODEC_DI2S_TXCR1 (RK817_CODEC_BASE + 0x4d) +#define RK817_CODEC_DI2S_TXCR2 (RK817_CODEC_BASE + 0x4e) +#define RK817_CODEC_DI2S_TXCR3_TXCMD (RK817_CODEC_BASE + 0x4f) + +/* RK817_CODEC_DI2S_CKM */ +#define RK817_I2S_MODE_MASK (0x1 << 0) +#define RK817_I2S_MODE_MST (0x1 << 0) +#define RK817_I2S_MODE_SLV (0x0 << 0) + +/* RK817_CODEC_DDAC_MUTE_MIXCTL */ +#define DACMT_MASK (0x1 << 0) +#define DACMT_ENABLE (0x1 << 0) +#define DACMT_DISABLE (0x0 << 0) + +/* RK817_CODEC_DI2S_RXCR2 */ +#define VDW_RX_24BITS (0x17) +#define VDW_RX_16BITS (0x0f) + +/* RK817_CODEC_DI2S_TXCR2 */ +#define VDW_TX_24BITS (0x17) +#define VDW_TX_16BITS (0x0f) + +/* RK817_CODEC_AMIC_CFG0 */ +#define MIC_DIFF_MASK (0x1 << 7) +#define MIC_DIFF_DIS (0x0 << 7) +#define MIC_DIFF_EN (0x1 << 7) + +#endif /* __RK817_CODEC_H__ */ -- 2.25.1

2 1

[PATCH] ASoC: core: Don't set platform name when of_node is set
by Daniel Baluta 17 Mar '21

17 Mar '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> Platform may be specified by either name or OF node but not both. For OF node platforms (e.g i.MX) we end up with both platform name and of_node set and sound card registration will fail with the error: asoc-simple-card sof-sound-wm8960: ASoC: Neither/both platform name/of_node are set for sai1-wm8960-hifi Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/soc-core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16ba54eb8164..76ab42fa9461 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1660,7 +1660,9 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dev_err(card->dev, "init platform error"); continue; } - dai_link->platforms->name = component->name; + + if (!dai_link->platforms->of_node) + dai_link->platforms->name = component->name; /* convert non BE into BE */ if (!dai_link->no_pcm) { -- 2.27.0

3 8

[PATCH v3] ASoC: amd: add support for rt5682 codec in machine driver
by Vijendar Mukunda 17 Mar '21

17 Mar '21

Add support for RT5682 codec in machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v2->v3: updated SPDX license v1->v2: updated kconfig, spdx license, removed unnecessary initialization and updated comment sound/soc/amd/Kconfig | 5 +- sound/soc/amd/acp-da7219-max98357a.c | 380 +++++++++++++++++++++++++++++++---- 2 files changed, 343 insertions(+), 42 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index a6ce000..43f5d29 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -5,14 +5,15 @@ config SND_SOC_AMD_ACP This option enables ACP DMA support on AMD platform. config SND_SOC_AMD_CZ_DA7219MX98357_MACH - tristate "AMD CZ support for DA7219 and MAX9835" + tristate "AMD CZ support for DA7219, RT5682 and MAX9835" select SND_SOC_DA7219 + select SND_SOC_RT5682_I2C select SND_SOC_MAX98357A select SND_SOC_ADAU7002 select REGULATOR depends on SND_SOC_AMD_ACP && I2C && GPIOLIB help - This option enables machine driver for DA7219 and MAX9835. + This option enables machine driver for DA7219, RT5682 and MAX9835. config SND_SOC_AMD_CZ_RT5645_MACH tristate "AMD CZ support for RT5645" diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 849288d..fb7aa74 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -1,27 +1,8 @@ -/* - * Machine driver for AMD ACP Audio engine using DA7219 & MAX98357 codec - * - * Copyright 2017 Advanced Micro Devices, Inc. - * - * Permission is hereby granted, free of charge, to any person obtaining a - * copy of this software and associated documentation files (the "Software"), - * to deal in the Software without restriction, including without limitation - * the rights to use, copy, modify, merge, publish, distribute, sublicense, - * and/or sell copies of the Software, and to permit persons to whom the - * Software is furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE COPYRIGHT HOLDER(S) OR AUTHOR(S) BE LIABLE FOR ANY CLAIM, DAMAGES OR - * OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, - * ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR - * OTHER DEALINGS IN THE SOFTWARE. - * - */ +// SPDX-License-Identifier: MIT +// +// Machine driver for AMD ACP Audio engine using DA7219, RT5682 & MAX98357 codec +// +//Copyright 2017-2021 Advanced Micro Devices, Inc. #include <sound/core.h> #include <sound/soc.h> @@ -41,14 +22,19 @@ #include "acp.h" #include "../codecs/da7219.h" #include "../codecs/da7219-aad.h" +#include "../codecs/rt5682.h" #define CZ_PLAT_CLK 48000000 #define DUAL_CHANNEL 2 +#define RT5682_PLL_FREQ (48000 * 512) static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_wclk; static struct clk *da7219_dai_bclk; -extern bool bt_uart_enable; +static struct clk *rt5682_dai_wclk; +static struct clk *rt5682_dai_bclk; +extern int bt_uart_enable; +void *soc_is_rltk_max(struct device *dev); static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { @@ -128,6 +114,88 @@ static void da7219_clk_disable(void) clk_disable_unprepare(da7219_dai_bclk); } +static int cz_rt5682_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_component *component = codec_dai->component; + + dev_info(codec_dai->dev, "codec dai name = %s\n", codec_dai->name); + + /* Set codec sysclk */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL2, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, + "Failed to set rt5682 SYSCLK: %d\n", ret); + return ret; + } + /* set codec PLL */ + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL2, RT5682_PLL2_S_MCLK, + CZ_PLAT_CLK, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set rt5682 PLL: %d\n", ret); + return ret; + } + + rt5682_dai_wclk = devm_clk_get(component->dev, "rt5682-dai-wclk"); + if (IS_ERR(rt5682_dai_wclk)) + return PTR_ERR(rt5682_dai_wclk); + + rt5682_dai_bclk = devm_clk_get(component->dev, "rt5682-dai-bclk"); + if (IS_ERR(rt5682_dai_bclk)) + return PTR_ERR(rt5682_dai_bclk); + + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &cz_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "HP jack creation failed %d\n", ret); + return ret; + } + + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ret = snd_soc_component_set_jack(component, &cz_jack, NULL); + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + return 0; +} + +static int rt5682_clk_enable(struct snd_pcm_substream *substream) +{ + int ret; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + + /* + * Set wclk to 48000 because the rate constraint of this driver is + * 48000. ADAU7002 spec: "The ADAU7002 requires a BCLK rate that is + * minimum of 64x the LRCLK sample rate." RT5682 is the only clk + * source so for all codecs we have to limit bclk to 64X lrclk. + */ + clk_set_rate(rt5682_dai_wclk, 48000); + clk_set_rate(rt5682_dai_bclk, 48000 * 64); + ret = clk_prepare_enable(rt5682_dai_bclk); + if (ret < 0) { + dev_err(rtd->dev, "can't enable master clock %d\n", ret); + return ret; + } + return ret; +} + +static void rt5682_clk_disable(void) +{ + clk_disable_unprepare(rt5682_dai_bclk); +} + static const unsigned int channels[] = { DUAL_CHANNEL, }; @@ -260,6 +328,118 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_rt5682_play_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->play_i2s_instance = I2S_SP_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_cap_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL1; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_max_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->play_i2s_instance = I2S_BT_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_dmic0_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_BT_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_dmic1_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL0; + return rt5682_clk_enable(substream); +} + +static void cz_rt5682_shutdown(struct snd_pcm_substream *substream) +{ + rt5682_clk_disable(); +} + static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, @@ -285,6 +465,31 @@ static const struct snd_soc_ops cz_dmic1_cap_ops = { .shutdown = cz_da7219_shutdown, }; +static const struct snd_soc_ops cz_rt5682_play_ops = { + .startup = cz_rt5682_play_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_cap_ops = { + .startup = cz_rt5682_cap_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_max_play_ops = { + .startup = cz_rt5682_max_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_dmic0_cap_ops = { + .startup = cz_rt5682_dmic0_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_dmic1_cap_ops = { + .startup = cz_rt5682_dmic1_startup, + .shutdown = cz_rt5682_shutdown, +}; + SND_SOC_DAILINK_DEF(designware1, DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto"))); SND_SOC_DAILINK_DEF(designware2, @@ -294,6 +499,8 @@ SND_SOC_DAILINK_DEF(designware3, SND_SOC_DAILINK_DEF(dlgs, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", "da7219-hifi"))); +SND_SOC_DAILINK_DEF(rt5682, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1"))); SND_SOC_DAILINK_DEF(mx, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi"))); SND_SOC_DAILINK_DEF(adau, @@ -353,6 +560,57 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { }, }; +static struct snd_soc_dai_link cz_dai_5682_98357[] = { + { + .name = "amd-rt5682-play", + .stream_name = "Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .init = cz_rt5682_init, + .dpcm_playback = 1, + .ops = &cz_rt5682_play_ops, + SND_SOC_DAILINK_REG(designware1, rt5682, platform), + }, + { + .name = "amd-rt5682-cap", + .stream_name = "Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_cap_ops, + SND_SOC_DAILINK_REG(designware2, rt5682, platform), + }, + { + .name = "amd-max98357-play", + .stream_name = "HiFi Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_playback = 1, + .ops = &cz_rt5682_max_play_ops, + SND_SOC_DAILINK_REG(designware3, mx, platform), + }, + { + /* C panel DMIC */ + .name = "dmic0", + .stream_name = "DMIC0 Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_dmic0_cap_ops, + SND_SOC_DAILINK_REG(designware3, adau, platform), + }, + { + /* A/B panel DMIC */ + .name = "dmic1", + .stream_name = "DMIC1 Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_dmic1_cap_ops, + SND_SOC_DAILINK_REG(designware2, adau, platform), + }, +}; + static const struct snd_soc_dapm_widget cz_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), @@ -368,6 +626,14 @@ static const struct snd_soc_dapm_route cz_audio_route[] = { {"PDM_DAT", NULL, "Int Mic"}, }; +static const struct snd_soc_dapm_route cz_rt5682_audio_route[] = { + {"Headphones", NULL, "HPOL"}, + {"Headphones", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"Speakers", NULL, "Speaker"}, + {"PDM_DAT", NULL, "Int Mic"}, +}; + static const struct snd_kcontrol_new cz_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), @@ -388,6 +654,28 @@ static struct snd_soc_card cz_card = { .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct snd_soc_card cz_rt5682_card = { + .name = "acpr5682m98357", + .owner = THIS_MODULE, + .dai_link = cz_dai_5682_98357, + .num_links = ARRAY_SIZE(cz_dai_5682_98357), + .dapm_widgets = cz_widgets, + .num_dapm_widgets = ARRAY_SIZE(cz_widgets), + .dapm_routes = cz_rt5682_audio_route, + .controls = cz_mc_controls, + .num_controls = ARRAY_SIZE(cz_mc_controls), +}; + +void *soc_is_rltk_max(struct device *dev) +{ + const struct acpi_device_id *match; + + match = acpi_match_device(dev->driver->acpi_match_table, dev); + if (!match) + return NULL; + return (void *)match->driver_data; +} + static struct regulator_consumer_supply acp_da7219_supplies[] = { REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"), REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"), @@ -425,29 +713,39 @@ static int cz_probe(struct platform_device *pdev) struct snd_soc_card *card; struct acp_platform_info *machine; struct regulator_dev *rdev; - - acp_da7219_cfg.dev = &pdev->dev; - rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, - &acp_da7219_cfg); - if (IS_ERR(rdev)) { - dev_err(&pdev->dev, "Failed to register regulator: %d\n", - (int)PTR_ERR(rdev)); - return -EINVAL; + struct device *dev = &pdev->dev; + + card = (struct snd_soc_card *)soc_is_rltk_max(dev); + if (!card) + return -ENODEV; + if (!strcmp(card->name, "acpd7219m98357")) { + acp_da7219_cfg.dev = &pdev->dev; + rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, + &acp_da7219_cfg); + if (IS_ERR(rdev)) { + dev_err(&pdev->dev, "Failed to register regulator: %d\n", + (int)PTR_ERR(rdev)); + return -EINVAL; + } } machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); if (!machine) return -ENOMEM; - card = &cz_card; - cz_card.dev = &pdev->dev; + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - ret = devm_snd_soc_register_card(&pdev->dev, &cz_card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", - cz_card.name, ret); + card->name, ret); + else + dev_dbg(&pdev->dev, + "devm_snd_soc_register_card(%s) probe deferred: %d\n", + card->name, ret); return ret; } bt_uart_enable = !device_property_read_bool(&pdev->dev, @@ -457,7 +755,8 @@ static int cz_probe(struct platform_device *pdev) #ifdef CONFIG_ACPI static const struct acpi_device_id cz_audio_acpi_match[] = { - { "AMD7219", 0 }, + { "AMD7219", (unsigned long)&cz_card }, + { "AMDI5682", (unsigned long)&cz_rt5682_card}, {}, }; MODULE_DEVICE_TABLE(acpi, cz_audio_acpi_match); @@ -475,5 +774,6 @@ static struct platform_driver cz_pcm_driver = { module_platform_driver(cz_pcm_driver); MODULE_AUTHOR("akshu.agrawal(a)amd.com"); -MODULE_DESCRIPTION("DA7219 & MAX98357A audio support"); +MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); +MODULE_DESCRIPTION("DA7219, RT5682 & MAX98357A audio support"); MODULE_LICENSE("GPL v2"); -- 2.7.4

5 5

[PATCH AUTOSEL 5.4 06/37] ALSA: hda: ignore invalid NHLT table
by Sasha Levin 17 Mar '21

17 Mar '21

From: Mark Pearson <markpearson(a)lenovo.com> [ Upstream commit a14a6219996ee6f6e858d83b11affc7907633687 ] On some Lenovo systems if the microphone is disabled in the BIOS only the NHLT table header is created, with no data. This means the endpoints field is not correctly set to zero - leading to an unintialised variable and hence invalid descriptors are parsed leading to page faults. The Lenovo firmware team is addressing this, but adding a check preventing invalid tables being parsed is worthwhile. Tested on a Lenovo T14. Tested-by: Philipp Leskovitz <philipp.leskovitz(a)secunet.com> Reported-by: Philipp Leskovitz <philipp.leskovitz(a)secunet.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210302141003.7342-1-markpearson@lenovo.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/intel-nhlt.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index baeda6c9716a..6ed80a4cba01 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -72,6 +72,11 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) if (!nhlt) return 0; + if (nhlt->header.length <= sizeof(struct acpi_table_header)) { + dev_warn(dev, "Invalid DMIC description table\n"); + return 0; + } + for (j = 0, epnt = nhlt->desc; j < nhlt->endpoint_count; j++, epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length)) { -- 2.30.1

1 0

[PATCH AUTOSEL 5.10 08/54] ALSA: hda: ignore invalid NHLT table
by Sasha Levin 17 Mar '21

17 Mar '21

From: Mark Pearson <markpearson(a)lenovo.com> [ Upstream commit a14a6219996ee6f6e858d83b11affc7907633687 ] On some Lenovo systems if the microphone is disabled in the BIOS only the NHLT table header is created, with no data. This means the endpoints field is not correctly set to zero - leading to an unintialised variable and hence invalid descriptors are parsed leading to page faults. The Lenovo firmware team is addressing this, but adding a check preventing invalid tables being parsed is worthwhile. Tested on a Lenovo T14. Tested-by: Philipp Leskovitz <philipp.leskovitz(a)secunet.com> Reported-by: Philipp Leskovitz <philipp.leskovitz(a)secunet.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210302141003.7342-1-markpearson@lenovo.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/intel-nhlt.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index d053beccfaec..e2237239d922 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -39,6 +39,11 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) if (!nhlt) return 0; + if (nhlt->header.length <= sizeof(struct acpi_table_header)) { + dev_warn(dev, "Invalid DMIC description table\n"); + return 0; + } + for (j = 0, epnt = nhlt->desc; j < nhlt->endpoint_count; j++, epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length)) { -- 2.30.1

1 0

[PATCH AUTOSEL 5.11 09/61] ALSA: hda: ignore invalid NHLT table
by Sasha Levin 17 Mar '21

17 Mar '21

From: Mark Pearson <markpearson(a)lenovo.com> [ Upstream commit a14a6219996ee6f6e858d83b11affc7907633687 ] On some Lenovo systems if the microphone is disabled in the BIOS only the NHLT table header is created, with no data. This means the endpoints field is not correctly set to zero - leading to an unintialised variable and hence invalid descriptors are parsed leading to page faults. The Lenovo firmware team is addressing this, but adding a check preventing invalid tables being parsed is worthwhile. Tested on a Lenovo T14. Tested-by: Philipp Leskovitz <philipp.leskovitz(a)secunet.com> Reported-by: Philipp Leskovitz <philipp.leskovitz(a)secunet.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210302141003.7342-1-markpearson@lenovo.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/intel-nhlt.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index d053beccfaec..e2237239d922 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -39,6 +39,11 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) if (!nhlt) return 0; + if (nhlt->header.length <= sizeof(struct acpi_table_header)) { + dev_warn(dev, "Invalid DMIC description table\n"); + return 0; + } + for (j = 0, epnt = nhlt->desc; j < nhlt->endpoint_count; j++, epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length)) { -- 2.30.1

1 0

[PATCH] ASoC: rt711: add snd_soc_component remove callback
by Bard Liao 16 Mar '21

16 Mar '21

We do some IO operations in the snd_soc_component_set_jack callback function and snd_soc_component_set_jack() will be called when soc component is removed. However, we should not access SoundWire registers when the bus is suspended. So set regcache_cache_only(regmap, true) to avoid accessing in the soc component removal process. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> --- sound/soc/codecs/rt711.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 202bf6fbc5be..9f5b2dc16c54 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -895,6 +895,13 @@ static int rt711_probe(struct snd_soc_component *component) return 0; } +static void rt711_remove(struct snd_soc_component *component) +{ + struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt711->regmap, true); +} + static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .probe = rt711_probe, .set_bias_level = rt711_set_bias_level, @@ -905,6 +912,7 @@ static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .dapm_routes = rt711_audio_map, .num_dapm_routes = ARRAY_SIZE(rt711_audio_map), .set_jack = rt711_set_jack_detect, + .remove = rt711_remove, }; static int rt711_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, -- 2.17.1

2 1

[PATCH 0/3] ASoC: SOF: debug: cleanups
by Pierre-Louis Bossart 16 Mar '21

16 Mar '21

cleanups of allocation and error handling Guennadi Liakhovetski (3): ASoC: SOF: fix debugfs initialisation error handling ASoC: SOF: only allocate debugfs cache buffers for IPC flood entries ASoC: SOF: remove superfluous NULL check in debugfs read sound/soc/sof/core.c | 5 +++-- sound/soc/sof/debug.c | 21 +++++++++++---------- 2 files changed, 14 insertions(+), 12 deletions(-) -- 2.25.1

2 4

[PATCH 0/2] Do not handle MCLK device clock in simple-card-utils
by Sameer Pujar 16 Mar '21

16 Mar '21

With commit 1e30f642cf29 ("ASoC: simple-card-utils: Fix device module clock") simple-card-utils can control MCLK clock for rate updates or enable/disable. But this is breaking some platforms where it is expected that codec drivers would actually handle the MCLK clock. One such example is following platform. - "arch/arm64/boot/dts/freescale/fsl-ls1028a-kontron-sl28-var3-ads2.dts" In above case codec, wm8904, is using internal PLL and configures sysclk based on fixed MCLK input. In such cases it is expected that, required PLL output or sysclk, is just passed via set_sysclk() callback and card driver need not actually update MCLK rate. Instead, codec can take ownership of this clock and do the necessary configuration. So the original commit is reverted and codec driver for rt5659 is updated to fix my board which has this codec. Sameer Pujar (2): ASoC: simple-card-utils: Do not handle device clock ASoC: rt5659: Update MCLK rate in set_sysclk() sound/soc/codecs/rt5659.c | 5 +++++ sound/soc/generic/simple-card-utils.c | 13 +++++++------ 2 files changed, 12 insertions(+), 6 deletions(-) -- 2.7.4

3 5

[PATCH] ASoC: mx27vis: Remove unused file
by Fabio Estevam 16 Mar '21

16 Mar '21

i.MX has been converted to a devicetree-only platform and asoc-mx27vis.h is no longer used. Get rid of this unused file. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- include/linux/platform_data/asoc-mx27vis.h | 12 ------------ 1 file changed, 12 deletions(-) delete mode 100644 include/linux/platform_data/asoc-mx27vis.h diff --git a/include/linux/platform_data/asoc-mx27vis.h b/include/linux/platform_data/asoc-mx27vis.h deleted file mode 100644 index 2107d0d992dd..000000000000 --- a/include/linux/platform_data/asoc-mx27vis.h +++ /dev/null @@ -1,12 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -#ifndef __PLATFORM_DATA_ASOC_MX27VIS_H -#define __PLATFORM_DATA_ASOC_MX27VIS_H - -struct snd_mx27vis_platform_data { - int amp_gain0_gpio; - int amp_gain1_gpio; - int amp_mutel_gpio; - int amp_muter_gpio; -}; - -#endif /* __PLATFORM_DATA_ASOC_MX27VIS_H */ -- 2.25.1

2 1

[PATCH] ASoC: wm8962: Relax bit clock divider searching
by Shengjiu Wang 16 Mar '21

16 Mar '21

With S20_3LE format case, the sysclk = rate * 384, the bclk = rate * 20 * 2, there is no proper bclk divider for 384 / 40, because current condition needs exact match. So driver fails to configure the clocking: wm8962 3-001a: Unsupported BCLK ratio 9 Fix this by relaxing bitclk divider searching, so that when no exact value can be derived from sysclk pick the closest value greater than expected bitclk. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8962.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index ce4666a74793..f5cd22450190 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2403,6 +2403,7 @@ static const int sysclk_rates[] = { static void wm8962_configure_bclk(struct snd_soc_component *component) { struct wm8962_priv *wm8962 = snd_soc_component_get_drvdata(component); + int best, min_diff, diff; int dspclk, i; int clocking2 = 0; int clocking4 = 0; @@ -2473,23 +2474,23 @@ static void wm8962_configure_bclk(struct snd_soc_component *component) dev_dbg(component->dev, "DSPCLK is %dHz, BCLK %d\n", dspclk, wm8962->bclk); - /* We're expecting an exact match */ + /* Search a proper bclk, not exact match. */ + best = 0; + min_diff = INT_MAX; for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { if (bclk_divs[i] < 0) continue; - if (dspclk / bclk_divs[i] == wm8962->bclk) { - dev_dbg(component->dev, "Selected BCLK_DIV %d for %dHz\n", - bclk_divs[i], wm8962->bclk); - clocking2 |= i; + diff = (dspclk / bclk_divs[i]) - wm8962->bclk; + if (diff < 0) /* Table is sorted */ break; + if (diff < min_diff) { + best = i; + min_diff = diff; } } - if (i == ARRAY_SIZE(bclk_divs)) { - dev_err(component->dev, "Unsupported BCLK ratio %d\n", - dspclk / wm8962->bclk); - return; - } + wm8962->bclk = dspclk / bclk_divs[best]; + clocking2 |= best; aif2 |= wm8962->bclk / wm8962->lrclk; dev_dbg(component->dev, "Selected LRCLK divisor %d for %dHz\n", -- 2.27.0

4 3

[PATCH v0] ASoC: rsnd: core: Check convert rate in rsnd_hw_params
by mdurnev＠gmail.com 16 Mar '21

16 Mar '21

From: Mikhail Durnev <mikhail_durnev(a)mentor.com> snd_pcm_hw_params_set_rate_near can return incorrect sample rate in some cases, e.g. when the backend output rate is set to some value higher than 48000 Hz and the input rate is 8000 Hz. So passing the value returned by snd_pcm_hw_params_set_rate_near to snd_pcm_hw_params will result in "FSO/FSI ratio error" and playing no audio at all while the userland is not properly notified about the issue. If SRC is unable to convert the requested sample rate to the sample rate the backend is using, then the requested sample rate should be adjusted in rsnd_hw_params. The userland will be notified about that change in the returned hw_params structure. Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> --- sound/soc/sh/rcar/core.c | 69 +++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index d1612d3..8696a99 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1424,8 +1424,75 @@ static int rsnd_hw_params(struct snd_soc_component *component, } if (io->converted_chan) dev_dbg(dev, "convert channels = %d\n", io->converted_chan); - if (io->converted_rate) + if (io->converted_rate) { + /* + * SRC supports convert rates from params_rate(hw_params)/k_down + * to params_rate(hw_params)*k_up, where k_up is always 6, and + * k_down depends on number of channels and SRC unit. + * So all SRC units can upsample audio up to 6 times regardless + * its number of channels. And all SRC units can downsample + * 2 channel audio up to 6 times too. + */ + int k_up = 6; + int k_down = 6; + int channel; + struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); + dev_dbg(dev, "convert rate = %d\n", io->converted_rate); + + channel = io->converted_chan ? io->converted_chan : + params_channels(hw_params); + + switch (rsnd_mod_id(src_mod)) { + /* + * SRC0 can downsample 4, 6 and 8 channel audio up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 4 channel audio + * up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 6 and 8 channel audio + * no more than twice. + */ + case 1: + case 3: + case 4: + if (channel > 4) { + k_down = 2; + break; + } + fallthrough; + case 0: + if (channel > 2) + k_down = 4; + break; + + /* Other SRC units do not support more than 2 channels */ + default: + if (channel > 2) + return -EINVAL; + } + + if (params_rate(hw_params) > io->converted_rate * k_down) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + io->converted_rate * k_down; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + io->converted_rate * k_down; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } else if (params_rate(hw_params) * k_up < io->converted_rate) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + (io->converted_rate + k_up - 1) / k_up; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + (io->converted_rate + k_up - 1) / k_up; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } + + /* + * TBD: Max SRC input and output rates also depend on number + * of channels and SRC unit: + * SRC1, SRC3 and SRC4 do not support more than 128kHz + * for 6 channel and 96kHz for 8 channel audio. + * Perhaps this function should return EINVAL if the input or + * the output rate exceeds the limitation. + */ + } } return rsnd_dai_call(hw_params, io, substream, hw_params); -- 2.7.4

3 2

[PATCH v2] ASoC: imx-hdmi: fix platform_no_drv_owner.cocci warnings
by Yang Li 16 Mar '21

16 Mar '21

./sound/soc/fsl/imx-hdmi.c:226:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- Change in v2: -use imx-hdmi instead of hdmi-codec for Subject sound/soc/fsl/imx-hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index dbbb761..cd0235a 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -223,7 +223,6 @@ static int imx_hdmi_probe(struct platform_device *pdev) static struct platform_driver imx_hdmi_driver = { .driver = { .name = "imx-hdmi", - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = imx_hdmi_dt_ids, }, -- 1.8.3.1

3 2

[PATCH] ASoC: fsl_spdif: use snd_ctl_boolean_mono_info
by Shengjiu Wang 16 Mar '21

16 Mar '21

From: Viorel Suman <viorel.suman(a)nxp.com> Remove redundant code and use snd_ctl_boolean_mono_info instead. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 30 +++--------------------------- 1 file changed, 3 insertions(+), 27 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 174e558224d8..1cd3441d1c03 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -801,18 +801,6 @@ static int fsl_spdif_qget(struct snd_kcontrol *kcontrol, return ret; } -/* Valid bit information */ -static int fsl_spdif_vbit_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; -} - /* Get valid good bit from interrupt status register */ static int fsl_spdif_rx_vbit_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -925,18 +913,6 @@ static int fsl_spdif_rxrate_get(struct snd_kcontrol *kcontrol, return 0; } -/* User bit sync mode info */ -static int fsl_spdif_usync_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; -} - /* * User bit sync mode: * 1 CD User channel subcode @@ -1018,7 +994,7 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = { .name = "IEC958 RX V-Bit Errors", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, - .info = fsl_spdif_vbit_info, + .info = snd_ctl_boolean_mono_info, .get = fsl_spdif_rx_vbit_get, }, { @@ -1027,7 +1003,7 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = { .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE | SNDRV_CTL_ELEM_ACCESS_VOLATILE, - .info = fsl_spdif_vbit_info, + .info = snd_ctl_boolean_mono_info, .get = fsl_spdif_tx_vbit_get, .put = fsl_spdif_tx_vbit_put, }, @@ -1047,7 +1023,7 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = { .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE | SNDRV_CTL_ELEM_ACCESS_VOLATILE, - .info = fsl_spdif_usync_info, + .info = snd_ctl_boolean_mono_info, .get = fsl_spdif_usync_get, .put = fsl_spdif_usync_put, }, -- 2.27.0

2 1

[PATCH] ASoC: dt-bindings: fsl_spdif: Add compatible string for new platforms
by Shengjiu Wang 16 Mar '21

16 Mar '21

Add compatible string for new added platforms which support spdif module. They are i.MX8QXP, i.MX8MM, i.MX8MN, i.MX8MQ. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml index 50449b6d1048..4454aca34d56 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -21,6 +21,10 @@ properties: - fsl,vf610-spdif - fsl,imx6sx-spdif - fsl,imx8qm-spdif + - fsl,imx8qxp-spdif + - fsl,imx8mq-spdif + - fsl,imx8mm-spdif + - fsl,imx8mn-spdif reg: maxItems: 1 -- 2.27.0

2 1

[PATCH] ASoC: hdmi-codec: fix platform_no_drv_owner.cocci warnings
by Yang Li 16 Mar '21

16 Mar '21

./sound/soc/fsl/imx-hdmi.c:226:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/fsl/imx-hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index dbbb761..cd0235a 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -223,7 +223,6 @@ static int imx_hdmi_probe(struct platform_device *pdev) static struct platform_driver imx_hdmi_driver = { .driver = { .name = "imx-hdmi", - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = imx_hdmi_dt_ids, }, -- 1.8.3.1

3 2

[PATCH v2 0/7] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 16 Mar '21

16 Mar '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Even though this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Am planning to send support for MBHC once this driver gets accepted! Thanks, srini Many thanks for reviewing v1. Changes since v1: - updated PA gain to a proper TLV control - move various if else checks to switch case. - return error on check failure rather than fallback to default - cleaned up micbias pull up control - fixed all the gain tlvs correctly - update dt-bindings to use microvolt for button thresholds - removed unnecessary debug at various places. Srinivas Kandagatla (7): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing .../bindings/sound/qcom,wcd938x.yaml | 127 + sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh-v2.c | 350 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 291 ++ sound/soc/codecs/wcd938x.c | 3623 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 676 +++ 8 files changed, 5084 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

2 8

conf: No SPDIF on Plantronics 3220 headset
by GitHub pull_request - opened 16 Mar '21

16 Mar '21

alsa-project/alsa-lib pull request #126 was opened from drjhe: The Plantronics Blackwire 3220 Series headset (USB ID 047f:c056) shows up in GNOME Settings with a non-existent digital output. Add it to the list to suppress this. Request URL : https://github.com/alsa-project/alsa-lib/pull/126 Patch URL : https://github.com/alsa-project/alsa-lib/pull/126.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: Intel: sof_rt5682: Add ALC1015Q-VB speaker amp support
by Brent Lu 16 Mar '21

16 Mar '21

This patch adds jsl_rt5682_rt1015p which supports the RT5682 headset codec and ALC1015Q-VB speaker amplifier combination on JasperLake platform. This driver also applies for ALC1015Q-CG running in auto-mode. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_realtek_common.c | 99 +++++++++++++++++++ sound/soc/intel/boards/sof_realtek_common.h | 7 ++ sound/soc/intel/boards/sof_rt5682.c | 19 +++- .../intel/common/soc-acpi-intel-jsl-match.c | 13 +++ 5 files changed, 137 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index d1d28129a32b..58379393b8e4 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -457,6 +457,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_MAX98373_I2C select SND_SOC_RT1011 select SND_SOC_RT1015 + select SND_SOC_RT1015P select SND_SOC_RT5682_I2C select SND_SOC_DMIC select SND_SOC_HDAC_HDMI diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index f3cf73c620ba..5dd0eb438aa3 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -7,6 +7,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/soc-dai.h> #include <sound/soc-dapm.h> #include <uapi/sound/asound.h> @@ -136,3 +137,101 @@ void sof_rt1011_codec_conf(struct snd_soc_card *card) card->codec_conf = rt1011_codec_confs; card->num_configs = ARRAY_SIZE(rt1011_codec_confs); } + +/* + * rt1015: i2c mode driver for ALC1015 and ALC1015Q + * rt1015p: auto-mode driver for ALC1015, ALC1015Q, and ALC1015Q-VB + */ +static const struct snd_soc_dapm_route rt1015p_1dev_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Speaker" }, + { "Right Spk", NULL, "Speaker" }, +}; + +static const struct snd_soc_dapm_route rt1015p_2dev_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left Speaker" }, + { "Right Spk", NULL, "Right Speaker" }, +}; + +static struct snd_soc_codec_conf rt1015p_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(RT1015P_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(RT1015P_DEV1_NAME), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component rt1015p_dai_link_components[] = { + { + .name = RT1015P_DEV0_NAME, + .dai_name = RT1015P_CODEC_DAI, + }, + { + .name = RT1015P_DEV1_NAME, + .dai_name = RT1015P_CODEC_DAI, + }, +}; + +static int rt1015p_get_num_codecs(void) +{ + static int dev_num; + + if (dev_num) + return dev_num; + + if (!acpi_dev_present("RTL1015", "1", -1)) + dev_num = 1; + else + dev_num = 2; + + return dev_num; +} + +static int rt1015p_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + /* reserved for debugging purpose */ + + return 0; +} + +static const struct snd_soc_ops rt1015p_ops = { + .hw_params = rt1015p_hw_params, +}; + +static int rt1015p_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + if (rt1015p_get_num_codecs() == 1) + ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_1dev_dapm_routes, + ARRAY_SIZE(rt1015p_1dev_dapm_routes)); + else + ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_2dev_dapm_routes, + ARRAY_SIZE(rt1015p_2dev_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +void sof_rt1015p_dai_link(struct snd_soc_dai_link *link) +{ + link->codecs = rt1015p_dai_link_components; + link->num_codecs = rt1015p_get_num_codecs(); + link->init = rt1015p_init; + link->ops = &rt1015p_ops; +} + +void sof_rt1015p_codec_conf(struct snd_soc_card *card) +{ + if (rt1015p_get_num_codecs() == 1) + return; + + card->codec_conf = rt1015p_codec_confs; + card->num_configs = ARRAY_SIZE(rt1015p_codec_confs); +} diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h index 87cb3812b926..cb0b49b2855c 100644 --- a/sound/soc/intel/boards/sof_realtek_common.h +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -21,4 +21,11 @@ void sof_rt1011_dai_link(struct snd_soc_dai_link *link); void sof_rt1011_codec_conf(struct snd_soc_card *card); +#define RT1015P_CODEC_DAI "HiFi" +#define RT1015P_DEV0_NAME "RTL1015:00" +#define RT1015P_DEV1_NAME "RTL1015:01" + +void sof_rt1015p_dai_link(struct snd_soc_dai_link *link); +void sof_rt1015p_codec_conf(struct snd_soc_card *card); + #endif /* __SOF_REALTEK_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 55505e207bc0..f4b898c1719f 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -45,8 +45,9 @@ #define SOF_RT1011_SPEAKER_AMP_PRESENT BIT(13) #define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(14) #define SOF_RT1015_SPEAKER_AMP_100FS BIT(15) -#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(16) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(17) +#define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(17) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(18) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -723,6 +724,8 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].num_codecs = ARRAY_SIZE(rt1015_components); links[id].init = speaker_codec_init_lr; links[id].ops = &sof_rt1015_ops; + } else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) { + sof_rt1015p_dai_link(&links[id]); } else if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { links[id].codecs = max_98373_components; @@ -851,6 +854,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_max98373_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) sof_rt1011_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) + sof_rt1015p_codec_conf(&sof_audio_card_rt5682); dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, dmic_be_num, hdmi_num); @@ -940,6 +945,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "jsl_rt5682_rt1015p", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1015P_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; @@ -966,3 +980,4 @@ MODULE_ALIAS("platform:tgl_max98373_rt5682"); MODULE_ALIAS("platform:jsl_rt5682_max98360a"); MODULE_ALIAS("platform:cml_rt1015_rt5682"); MODULE_ALIAS("platform:tgl_rt1011_rt5682"); +MODULE_ALIAS("platform:jsl_rt5682_rt1015p"); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index 52238db0bcb5..73fe4f89a82d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -19,6 +19,11 @@ static struct snd_soc_acpi_codecs rt1015_spk = { .codecs = {"10EC1015"} }; +static struct snd_soc_acpi_codecs rt1015p_spk = { + .num_codecs = 1, + .codecs = {"RTL1015"} +}; + static struct snd_soc_acpi_codecs mx98360a_spk = { .num_codecs = 1, .codecs = {"MX98360A"} @@ -52,6 +57,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .quirk_data = &rt1015_spk, .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", }, + { + .id = "10EC5682", + .drv_name = "jsl_rt5682_rt1015p", + .sof_fw_filename = "sof-jsl.ri", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1015p_spk, + .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", + }, { .id = "10EC5682", .drv_name = "jsl_rt5682_max98360a", -- 2.17.1

3 2

aaf plugin xrun:1711 error
by GitHub issues - opened 16 Mar '21

16 Mar '21

alsa-project/alsa-plugins issue #19 was opened from elabbing: Hello, I set 2 host to use aaf plugin for TSN test. I followed TSN-doc guidelines and `aaf.txt` doc. Simply, after hosts is ready and syncing though `ptp4l, phc3sys, qdisc,` then I try to run a basic test, no switch between, back to back linked hosts, obtaining the following error. ``` #Talker sudo speaker-test -p 12500 -F S16_BE -c 2 -r 48000 -D aaf0 #Listener sudo arecord -F 25000 -t raw -f S16_BE -c 2 -r 48000 -D aaf0 | \ > aplay -F 25000 -t raw -f S16_BE -c 2 -r 48000 -D default Playing raw data 'stdin' : Signed 16 bit Big Endian, Rate 48000 Hz, Stereo Recording raw data 'stdin' : Signed 16 bit Big Endian, Rate 48000 Hz, Stereo underrun!!! (at least 24,460 ms long) arecord: xrun:1711: read/write error, state = RUNNING ``` This error happens after a random period, tipically less than 60sec. I followed TSN doc settings, but I noticed that value in aaf plugin configuration and cbs qdisc are suggested differrently between aaf.txt doc and TSN doc. Any prompt to get out of this? Many thanks Issue URL : https://github.com/alsa-project/alsa-plugins/issues/19 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP 850 G8
by Jeremy Szu 16 Mar '21

16 Mar '21

The HP EliteBook 850 G8 Notebook PC is using ALC285 codec which is using 0x04 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e34ba64dd0ae..8239e5efc12d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8067,6 +8067,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), + SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 2.30.1

2 1

[PATCH] ASoC: Intel: sof_rt5682: Add rt1015p speaker amp support
by Brent Lu 16 Mar '21

16 Mar '21

This patch adds jsl_rt5682_rt1015p which supports the RT5682 headset codec and RT1015P speaker amplifier combination on JasperLake platform. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/codecs/rt1015p.c | 10 ++ sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_realtek_common.c | 94 +++++++++++++++++++ sound/soc/intel/boards/sof_realtek_common.h | 8 ++ sound/soc/intel/boards/sof_rt5682.c | 31 +++++- .../intel/common/soc-acpi-intel-jsl-match.c | 13 +++ 6 files changed, 155 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt1015p.c b/sound/soc/codecs/rt1015p.c index 671f2a2130fe..3a564272156b 100644 --- a/sound/soc/codecs/rt1015p.c +++ b/sound/soc/codecs/rt1015p.c @@ -4,6 +4,7 @@ // // Copyright 2020 The Linux Foundation. All rights reserved. +#include <linux/acpi.h> #include <linux/delay.h> #include <linux/device.h> #include <linux/err.h> @@ -130,10 +131,19 @@ static const struct of_device_id rt1015p_device_id[] = { MODULE_DEVICE_TABLE(of, rt1015p_device_id); #endif +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt1015p_acpi_match[] = { + { "RTL1015", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt1015p_acpi_match); +#endif + static struct platform_driver rt1015p_platform_driver = { .driver = { .name = "rt1015p", .of_match_table = of_match_ptr(rt1015p_device_id), + .acpi_match_table = ACPI_PTR(rt1015p_acpi_match), }, .probe = rt1015p_platform_probe, }; diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index d1d28129a32b..58379393b8e4 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -457,6 +457,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_MAX98373_I2C select SND_SOC_RT1011 select SND_SOC_RT1015 + select SND_SOC_RT1015P select SND_SOC_RT5682_I2C select SND_SOC_DMIC select SND_SOC_HDAC_HDMI diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index f3cf73c620ba..4b95aa2cbff8 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -136,3 +136,97 @@ void sof_rt1011_codec_conf(struct snd_soc_card *card) card->codec_conf = rt1011_codec_confs; card->num_configs = ARRAY_SIZE(rt1011_codec_confs); } + +#define RT1015P_SHARE_EN_SPK BIT(0) + +static unsigned long rt1015p_quirk; + +static const struct snd_soc_dapm_route rt1015p_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left Speaker" }, + { "Right Spk", NULL, "Right Speaker" }, +}; + +static const struct snd_soc_dapm_route rt1015p_share_en_spk_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Speaker" }, + { "Right Spk", NULL, "Speaker" }, +}; + +static struct snd_soc_codec_conf rt1015p_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(RT1015P_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(RT1015P_DEV1_NAME), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component rt1015p_dai_link_components[] = { + { + .name = RT1015P_DEV0_NAME, + .dai_name = RT1015P_CODEC_DAI, + }, + { + .name = RT1015P_DEV1_NAME, + .dai_name = RT1015P_CODEC_DAI, + }, +}; + +void sof_rt1015p_set_share_en_spk(void) +{ + /* Two amps share one en pin so there is only one device in acpi + * table + */ + rt1015p_quirk |= RT1015P_SHARE_EN_SPK; +} + +static int rt1015p_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + /* reserved for debugging purpose */ + + return 0; +} + +static const struct snd_soc_ops rt1015p_ops = { + .hw_params = rt1015p_hw_params, +}; + +static int rt1015p_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + if (rt1015p_quirk & RT1015P_SHARE_EN_SPK) + ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_share_en_spk_dapm_routes, + ARRAY_SIZE(rt1015p_share_en_spk_dapm_routes)); + else + ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_dapm_routes, + ARRAY_SIZE(rt1015p_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +void sof_rt1015p_dai_link(struct snd_soc_dai_link *link) +{ + link->codecs = rt1015p_dai_link_components; + if (rt1015p_quirk & RT1015P_SHARE_EN_SPK) + link->num_codecs = 1; + else + link->num_codecs = ARRAY_SIZE(rt1015p_dai_link_components); + link->init = rt1015p_init; + link->ops = &rt1015p_ops; +} + +void sof_rt1015p_codec_conf(struct snd_soc_card *card) +{ + if (rt1015p_quirk & RT1015P_SHARE_EN_SPK) + return; + + card->codec_conf = rt1015p_codec_confs; + card->num_configs = ARRAY_SIZE(rt1015p_codec_confs); +} diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h index 87cb3812b926..d7865681f2bd 100644 --- a/sound/soc/intel/boards/sof_realtek_common.h +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -21,4 +21,12 @@ void sof_rt1011_dai_link(struct snd_soc_dai_link *link); void sof_rt1011_codec_conf(struct snd_soc_card *card); +#define RT1015P_CODEC_DAI "HiFi" +#define RT1015P_DEV0_NAME "RTL1015:00" +#define RT1015P_DEV1_NAME "RTL1015:01" + +void sof_rt1015p_set_share_en_spk(void); +void sof_rt1015p_dai_link(struct snd_soc_dai_link *link); +void sof_rt1015p_codec_conf(struct snd_soc_card *card); + #endif /* __SOF_REALTEK_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 55505e207bc0..154f2ee9f406 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -45,8 +45,9 @@ #define SOF_RT1011_SPEAKER_AMP_PRESENT BIT(13) #define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(14) #define SOF_RT1015_SPEAKER_AMP_100FS BIT(15) -#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(16) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(17) +#define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(17) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(18) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -723,6 +724,8 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].num_codecs = ARRAY_SIZE(rt1015_components); links[id].init = speaker_codec_init_lr; links[id].ops = &sof_rt1015_ops; + } else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) { + sof_rt1015p_dai_link(&links[id]); } else if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { links[id].codecs = max_98373_components; @@ -847,10 +850,24 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_rt5682.num_links++; + /* setup amp quirk if any */ + if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) { + /* There may be only one device instance if two amps + * sharing one en pin + */ + if (!acpi_dev_present("RTL1015", "1", -1)) { + dev_dbg(&pdev->dev, "Device %s not exist\n", + RT1015P_DEV1_NAME); + sof_rt1015p_set_share_en_spk(); + } + } + if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) sof_max98373_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) sof_rt1011_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) + sof_rt1015p_codec_conf(&sof_audio_card_rt5682); dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, dmic_be_num, hdmi_num); @@ -940,6 +957,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "jsl_rt5682_rt1015p", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1015P_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; @@ -966,3 +992,4 @@ MODULE_ALIAS("platform:tgl_max98373_rt5682"); MODULE_ALIAS("platform:jsl_rt5682_max98360a"); MODULE_ALIAS("platform:cml_rt1015_rt5682"); MODULE_ALIAS("platform:tgl_rt1011_rt5682"); +MODULE_ALIAS("platform:jsl_rt5682_rt1015p"); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index 52238db0bcb5..73fe4f89a82d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -19,6 +19,11 @@ static struct snd_soc_acpi_codecs rt1015_spk = { .codecs = {"10EC1015"} }; +static struct snd_soc_acpi_codecs rt1015p_spk = { + .num_codecs = 1, + .codecs = {"RTL1015"} +}; + static struct snd_soc_acpi_codecs mx98360a_spk = { .num_codecs = 1, .codecs = {"MX98360A"} @@ -52,6 +57,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .quirk_data = &rt1015_spk, .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", }, + { + .id = "10EC5682", + .drv_name = "jsl_rt5682_rt1015p", + .sof_fw_filename = "sof-jsl.ri", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1015p_spk, + .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", + }, { .id = "10EC5682", .drv_name = "jsl_rt5682_max98360a", -- 2.17.1

3 3

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP 440 G8
by Jeremy Szu 16 Mar '21

16 Mar '21

The HP EliteBook 840 G8 Notebook PC is using ALC236 codec which is using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b47504fa8dfd..05dae87735bb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4225,6 +4225,12 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, } } +static void alc236_fixup_hp_gpio_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_hp_gpio_led(codec, action, 0x02, 0x01); +} + static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6381,6 +6387,7 @@ enum { ALC294_FIXUP_ASUS_GX502_VERBS, ALC285_FIXUP_HP_GPIO_LED, ALC285_FIXUP_HP_MUTE_LED, + ALC236_FIXUP_HP_GPIO_LED, ALC236_FIXUP_HP_MUTE_LED, ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -7616,6 +7623,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_mute_led, }, + [ALC236_FIXUP_HP_GPIO_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc236_fixup_hp_gpio_led, + }, [ALC236_FIXUP_HP_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc236_fixup_hp_mute_led, @@ -8045,6 +8056,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), -- 2.30.1

2 1

[PATCH v3 0/4] ALSA: hda/cirrus: Make CS8409 driver more generic by using fixups
by Vitaly Rodionov 16 Mar '21

16 Mar '21

This series of patches will address comments by Pierre-Louis Bossart, cleans up patch_cirrus.c source, reducing checkpatch.pl warnings from 19 to 0, fixing an issue reported by Canonical: BugLink: https://bugs.launchpad.net/bugs/1918378, and makes the CS8409 patch more generic by using fixups. Stefan Binding (4): ALSA: hda/cirrus: Add error handling into CS8409 I2C functions ALSA: hda/cirrus: Cleanup patch_cirrus.c code. ALSA: hda/cirrus: Fix CS42L42 Headset Mic volume control name ALSA: hda/cirrus: Make CS8409 driver more generic by using fixups. sound/pci/hda/patch_cirrus.c | 525 ++++++++++++++++------------------- 1 file changed, 242 insertions(+), 283 deletions(-) -- 2.25.1

2 5

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP 840 G8
by Jeremy Szu 16 Mar '21

16 Mar '21

* The HP EliteBook 840 G8 Notebook PC is using ALC285 codec which is using 0x04 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b47504fa8dfd..e568bf460bae 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8048,6 +8048,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), + SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.30.1

2 1

[PATCH v3] ALSA: hda/realtek: apply pin quirk for XiaomiNotebook Pro
by Xiaoliang Yu 15 Mar '21

15 Mar '21

Built-in microphone and combojack on Xiaomi Notebook Pro (1d72:1701) needs to be fixed, the existing quirk for Dell works well on that machine. Signed-off-by: Xiaoliang Yu <yxl_22(a)outlook.com> --- Changes in V3: - Sort the added entry in its PCI SSID order. --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5fea2b999afc..18646694ff51 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8249,6 +8249,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), + SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), -- 2.30.2

2 1

[PATCH v2 0/4] ALSA: hda/cirrus: Make CS8409 driver more generic by using fixups
by Vitaly Rodionov 15 Mar '21

15 Mar '21

This series of patches will address comments by Pierre-Louis Bossart, cleans up patch_cirrus.c source, reducing checkpatch.pl warnings from 19 to 0, fixing an issue reported by Canonical: BugLink: https://bugs.launchpad.net/bugs/1918378, and makes the CS8409 patch more generic by using fixups. Stefan Binding (4): ALSA: hda/cirrus: Add error handling into CS8409 I2C functions ALSA: hda/cirrus: Cleanup patch_cirrus.c code. ALSA: hda/cirrus: Fix CS42L42 Headset Mic volume control name ALSA: hda/cirrus: Make CS8409 driver more generic by using fixups. sound/pci/hda/patch_cirrus.c | 525 ++++++++++++++++------------------- 1 file changed, 242 insertions(+), 283 deletions(-) -- 2.25.1

2 5

[PATCH 0/3] Use clocks property in a device node
by Sameer Pujar 15 Mar '21

15 Mar '21

It is recommended to not specifiy clocks property in an endpoint subnode. This series moves clocks to device node. However after moving the clocks to device node, the audio playback or capture fails. The specified clock is not actually getting enabled and hence the failure is seen. There seems to be a bug in simple-card-utils.c where clock handle is not assigned when parsing clocks from device node. Fix the same and revert original change which actually added clocks property in endpoint subnode. Also update Jetson AGX Xavier DT where the usage is found. Sameer Pujar (3): ASoC: simple-card-utils: Fix device module clock Revert "ASoC: audio-graph-card: Add clocks property to endpoint node" arm64: tegra: Move clocks from RT5658 endpoint to device node .../devicetree/bindings/sound/audio-graph-port.yaml | 3 --- arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 2 +- sound/soc/generic/simple-card-utils.c | 13 ++++++------- 3 files changed, 7 insertions(+), 11 deletions(-) -- 2.7.4

4 36

[PATCH] sound: soc: codecs: Remove unnecessary THIS_MODULE
by Wang Qing 15 Mar '21

15 Mar '21

As THIS_MODULE has been set in module_platform_driver(), so remove it. Signed-off-by: Wang Qing <wangqing(a)vivo.com> --- sound/soc/codecs/lpass-rx-macro.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index c9c21d22..5b9d4e9 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -3585,7 +3585,6 @@ static const struct of_device_id rx_macro_dt_match[] = { static struct platform_driver rx_macro_driver = { .driver = { .name = "rx_macro", - .owner = THIS_MODULE, .of_match_table = rx_macro_dt_match, .suppress_bind_attrs = true, }, -- 2.7.4

2 2

[PATCH v1 0/4] ALSA: hda/cirrus: Make CS8409 driver more generic by using fixups
by Vitaly Rodionov 15 Mar '21

15 Mar '21

This series of patches will address comments by Pierre-Louis Bossart, cleans up patch_cirrus.c source, reducing checkpatch.pl warnings from 19 to 0, fixing an issue reported by Canonical: BugLink: https://bugs.launchpad.net/bugs/1918378, and makes the CS8409 patch more generic by using fixups. Stefan Binding (4): ALSA: hda/cirrus: Add error handling into CS8409 I2C functions ALSA: hda/cirrus: Cleanup patch_cirrus.c code. ALSA: hda/cirrus: Fix CS42L42 Headset Mic volume control name ALSA: hda/cirrus: Make CS8409 driver more generic by using fixups. sound/pci/hda/patch_cirrus.c | 506 ++++++++++++++++------------------- 1 file changed, 228 insertions(+), 278 deletions(-) -- 2.25.1

2 11

[PATCH v2] ASoC: amd: add support for rt5682 codec in machine driver
by Vijendar Mukunda 15 Mar '21

15 Mar '21

Add support for RT5682 codec in machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v1->v2 : updated kconfig, spdx license, removed unnecessary initialization and updated comment sound/soc/amd/Kconfig | 5 +- sound/soc/amd/acp-da7219-max98357a.c | 380 +++++++++++++++++++++++++++++++---- 2 files changed, 343 insertions(+), 42 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index a6ce000..43f5d29 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -5,14 +5,15 @@ config SND_SOC_AMD_ACP This option enables ACP DMA support on AMD platform. config SND_SOC_AMD_CZ_DA7219MX98357_MACH - tristate "AMD CZ support for DA7219 and MAX9835" + tristate "AMD CZ support for DA7219, RT5682 and MAX9835" select SND_SOC_DA7219 + select SND_SOC_RT5682_I2C select SND_SOC_MAX98357A select SND_SOC_ADAU7002 select REGULATOR depends on SND_SOC_AMD_ACP && I2C && GPIOLIB help - This option enables machine driver for DA7219 and MAX9835. + This option enables machine driver for DA7219, RT5682 and MAX9835. config SND_SOC_AMD_CZ_RT5645_MACH tristate "AMD CZ support for RT5645" diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 849288d..9b3520e 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -1,27 +1,8 @@ -/* - * Machine driver for AMD ACP Audio engine using DA7219 & MAX98357 codec - * - * Copyright 2017 Advanced Micro Devices, Inc. - * - * Permission is hereby granted, free of charge, to any person obtaining a - * copy of this software and associated documentation files (the "Software"), - * to deal in the Software without restriction, including without limitation - * the rights to use, copy, modify, merge, publish, distribute, sublicense, - * and/or sell copies of the Software, and to permit persons to whom the - * Software is furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE COPYRIGHT HOLDER(S) OR AUTHOR(S) BE LIABLE FOR ANY CLAIM, DAMAGES OR - * OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, - * ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR - * OTHER DEALINGS IN THE SOFTWARE. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Machine driver for AMD ACP Audio engine using DA7219, RT5682 & MAX98357 codec +// +//Copyright 2017-2021 Advanced Micro Devices, Inc. #include <sound/core.h> #include <sound/soc.h> @@ -41,14 +22,19 @@ #include "acp.h" #include "../codecs/da7219.h" #include "../codecs/da7219-aad.h" +#include "../codecs/rt5682.h" #define CZ_PLAT_CLK 48000000 #define DUAL_CHANNEL 2 +#define RT5682_PLL_FREQ (48000 * 512) static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_wclk; static struct clk *da7219_dai_bclk; -extern bool bt_uart_enable; +static struct clk *rt5682_dai_wclk; +static struct clk *rt5682_dai_bclk; +extern int bt_uart_enable; +void *soc_is_rltk_max(struct device *dev); static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { @@ -128,6 +114,88 @@ static void da7219_clk_disable(void) clk_disable_unprepare(da7219_dai_bclk); } +static int cz_rt5682_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_component *component = codec_dai->component; + + dev_info(codec_dai->dev, "codec dai name = %s\n", codec_dai->name); + + /* Set codec sysclk */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL2, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, + "Failed to set rt5682 SYSCLK: %d\n", ret); + return ret; + } + /* set codec PLL */ + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL2, RT5682_PLL2_S_MCLK, + CZ_PLAT_CLK, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set rt5682 PLL: %d\n", ret); + return ret; + } + + rt5682_dai_wclk = devm_clk_get(component->dev, "rt5682-dai-wclk"); + if (IS_ERR(rt5682_dai_wclk)) + return PTR_ERR(rt5682_dai_wclk); + + rt5682_dai_bclk = devm_clk_get(component->dev, "rt5682-dai-bclk"); + if (IS_ERR(rt5682_dai_bclk)) + return PTR_ERR(rt5682_dai_bclk); + + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &cz_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "HP jack creation failed %d\n", ret); + return ret; + } + + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ret = snd_soc_component_set_jack(component, &cz_jack, NULL); + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + return 0; +} + +static int rt5682_clk_enable(struct snd_pcm_substream *substream) +{ + int ret; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + + /* + * Set wclk to 48000 because the rate constraint of this driver is + * 48000. ADAU7002 spec: "The ADAU7002 requires a BCLK rate that is + * minimum of 64x the LRCLK sample rate." RT5682 is the only clk + * source so for all codecs we have to limit bclk to 64X lrclk. + */ + clk_set_rate(rt5682_dai_wclk, 48000); + clk_set_rate(rt5682_dai_bclk, 48000 * 64); + ret = clk_prepare_enable(rt5682_dai_bclk); + if (ret < 0) { + dev_err(rtd->dev, "can't enable master clock %d\n", ret); + return ret; + } + return ret; +} + +static void rt5682_clk_disable(void) +{ + clk_disable_unprepare(rt5682_dai_bclk); +} + static const unsigned int channels[] = { DUAL_CHANNEL, }; @@ -260,6 +328,118 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_rt5682_play_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->play_i2s_instance = I2S_SP_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_cap_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL1; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_max_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->play_i2s_instance = I2S_BT_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_dmic0_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_BT_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_dmic1_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL0; + return rt5682_clk_enable(substream); +} + +static void cz_rt5682_shutdown(struct snd_pcm_substream *substream) +{ + rt5682_clk_disable(); +} + static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, @@ -285,6 +465,31 @@ static const struct snd_soc_ops cz_dmic1_cap_ops = { .shutdown = cz_da7219_shutdown, }; +static const struct snd_soc_ops cz_rt5682_play_ops = { + .startup = cz_rt5682_play_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_cap_ops = { + .startup = cz_rt5682_cap_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_max_play_ops = { + .startup = cz_rt5682_max_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_dmic0_cap_ops = { + .startup = cz_rt5682_dmic0_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_dmic1_cap_ops = { + .startup = cz_rt5682_dmic1_startup, + .shutdown = cz_rt5682_shutdown, +}; + SND_SOC_DAILINK_DEF(designware1, DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto"))); SND_SOC_DAILINK_DEF(designware2, @@ -294,6 +499,8 @@ SND_SOC_DAILINK_DEF(designware3, SND_SOC_DAILINK_DEF(dlgs, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", "da7219-hifi"))); +SND_SOC_DAILINK_DEF(rt5682, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1"))); SND_SOC_DAILINK_DEF(mx, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi"))); SND_SOC_DAILINK_DEF(adau, @@ -353,6 +560,57 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { }, }; +static struct snd_soc_dai_link cz_dai_5682_98357[] = { + { + .name = "amd-rt5682-play", + .stream_name = "Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .init = cz_rt5682_init, + .dpcm_playback = 1, + .ops = &cz_rt5682_play_ops, + SND_SOC_DAILINK_REG(designware1, rt5682, platform), + }, + { + .name = "amd-rt5682-cap", + .stream_name = "Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_cap_ops, + SND_SOC_DAILINK_REG(designware2, rt5682, platform), + }, + { + .name = "amd-max98357-play", + .stream_name = "HiFi Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_playback = 1, + .ops = &cz_rt5682_max_play_ops, + SND_SOC_DAILINK_REG(designware3, mx, platform), + }, + { + /* C panel DMIC */ + .name = "dmic0", + .stream_name = "DMIC0 Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_dmic0_cap_ops, + SND_SOC_DAILINK_REG(designware3, adau, platform), + }, + { + /* A/B panel DMIC */ + .name = "dmic1", + .stream_name = "DMIC1 Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_dmic1_cap_ops, + SND_SOC_DAILINK_REG(designware2, adau, platform), + }, +}; + static const struct snd_soc_dapm_widget cz_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), @@ -368,6 +626,14 @@ static const struct snd_soc_dapm_route cz_audio_route[] = { {"PDM_DAT", NULL, "Int Mic"}, }; +static const struct snd_soc_dapm_route cz_rt5682_audio_route[] = { + {"Headphones", NULL, "HPOL"}, + {"Headphones", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"Speakers", NULL, "Speaker"}, + {"PDM_DAT", NULL, "Int Mic"}, +}; + static const struct snd_kcontrol_new cz_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), @@ -388,6 +654,28 @@ static struct snd_soc_card cz_card = { .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct snd_soc_card cz_rt5682_card = { + .name = "acpr5682m98357", + .owner = THIS_MODULE, + .dai_link = cz_dai_5682_98357, + .num_links = ARRAY_SIZE(cz_dai_5682_98357), + .dapm_widgets = cz_widgets, + .num_dapm_widgets = ARRAY_SIZE(cz_widgets), + .dapm_routes = cz_rt5682_audio_route, + .controls = cz_mc_controls, + .num_controls = ARRAY_SIZE(cz_mc_controls), +}; + +void *soc_is_rltk_max(struct device *dev) +{ + const struct acpi_device_id *match; + + match = acpi_match_device(dev->driver->acpi_match_table, dev); + if (!match) + return NULL; + return (void *)match->driver_data; +} + static struct regulator_consumer_supply acp_da7219_supplies[] = { REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"), REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"), @@ -425,29 +713,39 @@ static int cz_probe(struct platform_device *pdev) struct snd_soc_card *card; struct acp_platform_info *machine; struct regulator_dev *rdev; - - acp_da7219_cfg.dev = &pdev->dev; - rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, - &acp_da7219_cfg); - if (IS_ERR(rdev)) { - dev_err(&pdev->dev, "Failed to register regulator: %d\n", - (int)PTR_ERR(rdev)); - return -EINVAL; + struct device *dev = &pdev->dev; + + card = (struct snd_soc_card *)soc_is_rltk_max(dev); + if (!card) + return -ENODEV; + if (!strcmp(card->name, "acpd7219m98357")) { + acp_da7219_cfg.dev = &pdev->dev; + rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, + &acp_da7219_cfg); + if (IS_ERR(rdev)) { + dev_err(&pdev->dev, "Failed to register regulator: %d\n", + (int)PTR_ERR(rdev)); + return -EINVAL; + } } machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); if (!machine) return -ENOMEM; - card = &cz_card; - cz_card.dev = &pdev->dev; + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - ret = devm_snd_soc_register_card(&pdev->dev, &cz_card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", - cz_card.name, ret); + card->name, ret); + else + dev_dbg(&pdev->dev, + "devm_snd_soc_register_card(%s) probe deferred: %d\n", + card->name, ret); return ret; } bt_uart_enable = !device_property_read_bool(&pdev->dev, @@ -457,7 +755,8 @@ static int cz_probe(struct platform_device *pdev) #ifdef CONFIG_ACPI static const struct acpi_device_id cz_audio_acpi_match[] = { - { "AMD7219", 0 }, + { "AMD7219", (unsigned long)&cz_card }, + { "AMDI5682", (unsigned long)&cz_rt5682_card}, {}, }; MODULE_DEVICE_TABLE(acpi, cz_audio_acpi_match); @@ -475,5 +774,6 @@ static struct platform_driver cz_pcm_driver = { module_platform_driver(cz_pcm_driver); MODULE_AUTHOR("akshu.agrawal(a)amd.com"); -MODULE_DESCRIPTION("DA7219 & MAX98357A audio support"); +MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); +MODULE_DESCRIPTION("DA7219, RT5682 & MAX98357A audio support"); MODULE_LICENSE("GPL v2"); -- 2.7.4

2 1

[PATCH] sound:fix typo in rt5663.c
by mateng 15 Mar '21

15 Mar '21

From: mateng <mateng(a)yulong.com> interrput -> interrupt Signed-off-by: mateng <mateng(a)yulong.com> --- sound/soc/codecs/rt5663.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 619fb9a..411ff30 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1967,7 +1967,7 @@ static void rt5663_jack_detect_work(struct work_struct *work) btn_type); break; } - /* button release or spurious interrput*/ + /* button release or spurious interrupt*/ if (btn_type == 0) { report = rt5663->jack_type; cancel_delayed_work_sync( -- 1.9.1

1 0

[PATCH 0/7] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 15 Mar '21

15 Mar '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Even though this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Am planning to send support for MBHC once this driver gets accepted! Thanks, srini Srinivas Kandagatla (7): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing .../bindings/sound/qcom,wcd938x.yaml | 127 + sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh-v2.c | 350 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 294 ++ sound/soc/codecs/wcd938x.c | 3688 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 675 +++ 8 files changed, 5151 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

2 13

kernel branch for submitting patches
by Mikhail Durnev 15 Mar '21

15 Mar '21

Dear All, I am new to ALSA development in Linux kernel. Is my understanding correct that new patches should be created against the for-next branch in https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git ? When I tried to create a patch on top of the master branch of https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git and send it to linux-kernel(a)vger.kernel.org, my patch was ignored. Reading docs from www.alsa-project.org did not help much. In this mailing list I found that patches should probably go to broonie/sound.git for-next first. Looking forward to receiving your help. Thanks, Mikhail

2 1

Wrong description in emu0404_netlist.h
by GitHub issues - reopened 15 Mar '21

15 Mar '21

alsa-project/alsa-firmware issue #4 was reopened from Usr666x: It says: > This is the FPGA firmware for EMU Systems's Audio Dock, > Present in the EMU1820 and EMU1820m products. > It is nicknamed "audio_dock". This is a mistake because it's a description taken from audio_dock_netlist.h. emu0404 should have its own description. Issue URL : https://github.com/alsa-project/alsa-firmware/issues/4 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

Wrong description in emu0404_netlist.h
by GitHub issues - edited 15 Mar '21

15 Mar '21

alsa-project/alsa-firmware issue #4 was edited from Usr666x: It says: > This is the FPGA firmware for EMU Systems's Audio Dock, > Present in the EMU1820 and EMU1820m products. > It is nicknamed "audio_dock". This is a mistake because it's a description taken from audio_dock_netlist.h. emu0404 should have its own description. Issue URL : https://github.com/alsa-project/alsa-firmware/issues/4 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

Wrong description in emu0404_netlist.h
by GitHub issues - opened 15 Mar '21

15 Mar '21

alsa-project/alsa-firmware issue #4 was opened from Usr666x: It says: /* This is the FPGA firmware for EMU Systems's Audio Dock, * Present in the EMU1820 and EMU1820m products. * It is nicknamed "audio_dock". */ This is a mistake because it's a description taken from audio_dock_netlist.h. emu0404 should have its own description. Issue URL : https://github.com/alsa-project/alsa-firmware/issues/4 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

Midi timestamps
by David Henningsson 14 Mar '21

14 Mar '21

Hi, When writing an application that records midi (and e g saves it to disk), ultra-low latency isn't really needed, all we need to know is exactly when the midi message came in. The application can then wake up once a second or so, to write the incoming data, including an accurate timestamp for every event, to disk. As far as I can see, the rawmidi interface does not support such a feature at all. There is a snd_rawmidi_status syscall, but its timestamp field is not even filled by the kernel (!). But even if that was fixed, it would not fix the problem as there could be several midi events in the buffer with different timestamps. You could use the seq interface, it does support timestamps, but I can see at least two potential problems with this: First, the seq code runs in a work queue, not in the actual IRQ. This means that midi event is timestamped too late, especially so if the work is delayed for some reason. Second, seq hard-codes the timestamp type to monotonic - there is no monotonic_raw, so the timestamp would be affected by NTP changes. Also, the timespec uses 32-bit for sec and nsec, but I suspect this is less of a problem (unless people constantly record midi for sixty years or so...). It seems like it could be useful to have a rawmidi mode which automatically prefixes every incoming message with a timestamp. This would replace the current "raw" midi buffer with some type of header "length, timestamp, data" or so. Arguably this would make the rawmidi slightly less raw, but anyway. Does this make sense? Should I attempt to write something? Or would it be better to try to use (and potentially improve) the seq code? // David

1 0

[PATCH] speaker-test: add support for S24_LE and S24_BE
by Pierre-Louis Bossart 14 Mar '21

14 Mar '21

These formats are sometimes advertised by drivers, e.g. SOF. The format is 3 bytes packed in 32-bit container, with the MSB zeroed out. sample: 0x00123456 S24_LE format: b0 56 b1 34 b2 12 b3 00 S24_BE format: b0 00 b1 12 b2 34 b3 56 I only tested the S24_LE format with the SOF driver, S24_BE was added for symmetry only. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- speaker-test/speaker-test.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index 773af0a..fd13d88 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -285,6 +285,8 @@ static const int supported_formats[] = { SND_PCM_FORMAT_FLOAT_LE, SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_S24_3BE, + SND_PCM_FORMAT_S24_LE, + SND_PCM_FORMAT_S24_BE, SND_PCM_FORMAT_S32_LE, SND_PCM_FORMAT_S32_BE, -1 @@ -338,6 +340,20 @@ static void do_generate(uint8_t *frames, int channel, int count, *samp8++ = BE_INT(res.i) >> 8; *samp8++ = BE_INT(res.i) >> 16; break; + case SND_PCM_FORMAT_S24_LE: + res.i >>= 8; + *samp8++ = LE_INT(res.i); + *samp8++ = LE_INT(res.i) >> 8; + *samp8++ = LE_INT(res.i) >> 16; + *samp8++ = 0; + break; + case SND_PCM_FORMAT_S24_BE: + res.i >>= 8; + *samp8++ = 0; + *samp8++ = BE_INT(res.i); + *samp8++ = BE_INT(res.i) >> 8; + *samp8++ = BE_INT(res.i) >> 16; + break; case SND_PCM_FORMAT_S32_LE: *samp32++ = LE_INT(res.i); break; -- 2.25.1

2 1

alsa-lib's new API issue (snd_ctl_elem_id_compare)
by Takashi Sakamoto 14 Mar '21

14 Mar '21

Hi Jaroslav, I have some concern about your commit 2cfe6addaef6[1] for alsa-lib, It adds new library API, 'snd_ctl_elem_id_compare()', to compare a pair of IDs for control element. In the implementation, the API call returns 0 if they are the same, else negative or positive numeric value according to contents of the IDs. My concerns are: 1. evaluation of numid field is not covered. This is not preferable since ALSA control core implementation covers two types of comparison; numid only, and the combination iface, device, subdevice, name, and index fields. If the API is produced for general use cases, it should correctly handle the numid-only case, in my opinion. 2. tri-state return value is semantically inconsistent The ID structure includes three types of values; integer, enumeration, and string. In my opinion, tri-state return value from them has different meaning each other. It's better just to return comparison result by boolean value, in my opinion. The reason I post this message instead of posting any fix is that the fix to the API affects to alsa-utils, in which the API is used by a patch you applied a few days ago[2]. The patch also includes change to 'AM_PATH_ALSA' declaration in configure.ac with bumped-up version to '1.2.5', and it disables to rebuild alsa-utils on the latest toolchain. (alsa-lib 1.2.5 is not released yet.) I'd like to drop the usage of API from alsa-utils with equivalent alternative codes in alsa-utils side at first, apart from the new API in alsa-lib so that alsa-utils can be still build on the latest toolchains to avoid any confusion in user side. Then I'd fix the issued API carefully so that it can be used so long as exposed API without any issue. [1] https://github.com/alsa-project/alsa-lib/commit/2cfe6addaef6a0afa72699ec07a… [2] https://github.com/alsa-project/alsa-utils/commit/17b4129e6c89d1a96d4d86dab… Regards Takashi Sakamoto

2 8

[PATCH] sound: soc: fsl: Remove unnecessary THIS_MODULE
by Wang Qing 13 Mar '21

13 Mar '21

As THIS_MODULE has been set in module_platform_driver(), so remove it. Signed-off-by: Wang Qing <wangqing(a)vivo.com> --- sound/soc/fsl/imx-hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index dbbb761..cd0235a --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -223,7 +223,6 @@ MODULE_DEVICE_TABLE(of, imx_hdmi_dt_ids); static struct platform_driver imx_hdmi_driver = { .driver = { .name = "imx-hdmi", - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = imx_hdmi_dt_ids, }, -- 2.7.4

2 1

broadwell-rt286, bdw-rt5677: Fix inconsistency in card detection
by GitHub pull_request - edited 13 Mar '21

13 Mar '21

alsa-project/alsa-ucm-conf pull request #85 was edited from dpward: This conveniently provides a workaround for a bug in kernel 5.10: https://bugzilla.kernel.org/show_bug.cgi?id=211985 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/85 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/85.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH alsa-ucm-conf v2 0/7] codecs/rt5640: Cleanup + HW volume control support
by Hans de Goede 13 Mar '21

13 Mar '21

Hi All, Here is v2 of my patch series consisting of some codecs/rt5640 cleanups and addition of HW volume control support to the rt5640 .conf snippets. This new version has been updated to deal with the "[PATCH 3/5] ASoC: rt5640: Add emulated 'DAC1 Playback Switch' control" kernel-patch being dropped. Note patch 7 depends on a new "aif:%d" part being added to the components string, the kernel patches for this are pending upstream in the "[PATCH resend 0/2] AsoC: rt5640/rt5651: Volume control fixes" patch-series. As such patch 7 should probably not be merged until the kernel patches have landed, because theoretically the new component string part could change as part of the review of the kernel patches. Patches 1-6 are ready to be merged now. Regards, Hans Hans de Goede (7): codecs/rt5640: Cleanup: unify Stereo / Mono ADC handling codecs/rt5640: Cleanup: Initially disable all inputs and outputs codecs/rt5640: Cleanup: Move 'SPK MIX' setup to main EnableSequene codecs/rt5640: Drop bogus 'DIG MIXL DAC L2 Switch' setting codecs/rt5640: Unify capture volume for AIF1 and AIF2 recording paths codecs/rt5640: Add hardware volume-control support codecs/rt5640: Specify Playback/CaptureMasterElem for HW volume-control ucm2/bytcr-rt5640/HiFi.conf | 21 +++++++++++ ucm2/codecs/rt5640/DigitalMics.conf | 18 ++++++++- ucm2/codecs/rt5640/EnableSeq.conf | 50 ++++++++++++++++++++----- ucm2/codecs/rt5640/HeadPhones.conf | 15 +++++++- ucm2/codecs/rt5640/HeadsetMic.conf | 24 ++++++++---- ucm2/codecs/rt5640/IN1-InternalMic.conf | 27 +++++++------ ucm2/codecs/rt5640/IN3-InternalMic.conf | 27 +++++++------ ucm2/codecs/rt5640/MonoSpeaker.conf | 17 +++++++-- ucm2/codecs/rt5640/Speaker.conf | 17 +++++++-- 9 files changed, 167 insertions(+), 49 deletions(-) -- 2.30.1

2 10

[PATCH alsa-ucm-conf 1/2] codecs/rt5672: Move +12dB input boost from 'STO1 ADC Boost Gain' to 'ADC Capture Volume'
by Hans de Goede 13 Mar '21

13 Mar '21

Both the 'STO1 ADC Boost Gain' and the 'ADC Capture Volume' controls are digital and the 'ADC Capture Volume' control goes up to +30dB. When adding hw-volume-control support adding the +12dB boost gain on to of the max +30dB of 'ADC Capture Volume' is a bit too much. So move the +12dB from the 'STO1 ADC Boost Gain' control to the 'ADC Capture Volume' control. This way we keep the same setting for userspace which does not support hw-volume-control, while getting a better range when we start using 'ADC Capture Volume' as hw-volume-control. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/codecs/rt5672/EnableSeq.conf | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/ucm2/codecs/rt5672/EnableSeq.conf b/ucm2/codecs/rt5672/EnableSeq.conf index ddfb1c7..e373275 100644 --- a/ucm2/codecs/rt5672/EnableSeq.conf +++ b/ucm2/codecs/rt5672/EnableSeq.conf @@ -26,14 +26,13 @@ EnableSequence [ # cset "name='LOUT MIX OUTMIX L Switch' on" # cset "name='LOUT MIX OUTMIX R Switch' on" - # 47=0dB, 0.375 dB/step - cset "name='ADC Capture Volume' 47" + # Both the DMICs and the headset mic (even with the 20dB boost below) + # are quite soft, set the default capture volume to +12dB + # 47=0dB, 0.375dB/step, 79=+12dB + cset "name='ADC Capture Volume' 79" cset "name='ADC Capture Switch' on" - - # Both the DMICs and the headset mic (even with the 20 dB boost below) - # are quite soft, set the ADC Boost gain (range 0-3) to 1 = 12 dB the - # next step is 24 dB which is a bit too much - cset "name='STO1 ADC Boost Gain Volume' 1" + # 0=0dB + cset "name='STO1 ADC Boost Gain Volume' 0" # Headset mic is quite soft, boost it a bit, 1 = 20dB which is the first # available boost step -- 2.30.1

2 3

alsa_snd_pcm_hw_params() on Ubuntu 20.04 started returning an unexpected -ENOMEM
by GitHub issues - opened 13 Mar '21

13 Mar '21

alsa-project/alsa-lib issue #125 was opened from philburk: This seems to be a regression in a recent release. It was reported on the PortAudio GitHub repo. See here for more details: https://github.com/PortAudio/portaudio/issues/526 Issue URL : https://github.com/alsa-project/alsa-lib/issues/125 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH resend 0/2] AsoC: rt5640/rt5651: Volume control fixes
by Hans de Goede 13 Mar '21

13 Mar '21

Hi All, Here is a resent of the remaining patches from my "[PATCH 0/5] AsoC: rt5640/rt5651: Volume control fixes" series, with the controversial "[PATCH 3/5] ASoC: rt5640: Add emulated 'DAC1 Playback Switch' control" patch dropped, and these remaining 2 patches rebased to still apply with that patch dropped. Regards, Hans Hans de Goede (2): ASoC: rt5640: Rename 'Mono DAC Playback Volume' to 'DAC2 Playback Volume' ASoC: Intel: bytcr_rt5640: Add used AIF to the components string sound/soc/codecs/rt5640.c | 6 +++--- sound/soc/intel/boards/bytcr_rt5640.c | 11 ++++++++--- 2 files changed, 11 insertions(+), 6 deletions(-) -- 2.30.1

2 3

[PATCH v3 0/2] ASoC: samsung: remove cppcheck warnings
by Pierre-Louis Bossart 13 Mar '21

13 Mar '21

v3: Added Sylwester tag Rebased and squashed fix with initial patch which was merged at some point but can't be found in broonie/for-next (not sure what happened?) Corrected patch subjects to tm2_wm5110 Reverted second patch to initial v1, after agreement between Krzysztof and Sylwester v2: added Krzysztof Kozlowski's tags added fix for first patch already merged as suggested by Krzysztof Kozlowski moved variable to lower scope in patch6 Pierre-Louis Bossart (2): ASoC: samsung: tm2_wm5110: check of of_parse return value ASoC: samsung: tm2_wm5110: remove shadowing variable sound/soc/samsung/tm2_wm5110.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) -- 2.25.1

3 10

[PATCH] ASoC: fsl_asrc_dma: request dma channel from specific controller
by Shengjiu Wang 13 Mar '21

13 Mar '21

From: Robin Gong <yibin.gong(a)nxp.com> Request dma channel from specific dma controller instead of generic dma controller list, otherwise, may get the wrong dma controller if there are multi dma controllers such as i.MX8MP. Signed-off-by: Robin Gong <yibin.gong(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc_dma.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 29f91cdecbc3..c313a26c8f95 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -141,6 +141,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct dma_slave_config config_fe, config_be; enum asrc_pair_index index = pair->index; struct device *dev = component->dev; + struct device_node *of_dma_node; int stream = substream->stream; struct imx_dma_data *tmp_data; struct snd_soc_dpcm *dpcm; @@ -231,8 +232,10 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, pair->dma_data.priority = tmp_data->priority; dma_release_channel(tmp_chan); + of_dma_node = pair->dma_chan[!dir]->device->dev->of_node; pair->dma_chan[dir] = - dma_request_channel(mask, filter, &pair->dma_data); + __dma_request_channel(&mask, filter, &pair->dma_data, + of_dma_node); pair->req_dma_chan = true; } else { pair->dma_chan[dir] = tmp_chan; -- 2.27.0

2 1

[PATCH 0/4] ALSA: hda/cirrus: Make CS8409 driver more generic by using fixups
by Vitaly Rodionov 12 Mar '21

12 Mar '21

This series of patches will address comments by Pierre-Louis Bossart, cleans up patch_cirrus.c source, reducing checkpatch.pl warnings from 19 to 0, fixing an issue reported by Canonical: BugLink: https://bugs.launchpad.net/bugs/1918378, and makes the CS8409 patch more generic by using fixups. Stefan Binding (4): ALSA: hda/cirrus: Add error handling into CS8409 I2C functions ALSA: hda/cirrus: Cleanup patch_cirrus.c code. ALSA: hda/cirrus: Fix CS42L42 Headset Mic volume control name ALSA: hda/cirrus: Make CS8409 driver more generic by using fixups. sound/pci/hda/patch_cirrus.c | 506 ++++++++++++++++------------------- 1 file changed, 228 insertions(+), 278 deletions(-) -- 2.25.1

2 5

[PATCH v3 0/8] ASoC: soc-pcm: cleanup each functions
by Kuninori Morimoto 12 Mar '21

12 Mar '21

Hi Mark These are v2 of soc-pcm cleanup patches. These has no relationship to each other. My 1 concern is [3/8] patch. I think it is no problem, but I'm not 100% sure why current code was such code. Pierre-Louis / Liam might about something. v1 -> v2 - soc_cpu/codec_dai_name() is now inline function - rename soc_pcm_care_symmetry() to soc_pcm_update_symmetry() v2 -> v3 - log fix at [6/8] - Thus, it will be minus users at (2). + Thus, users will be a negative number at (2) Link: https://lore.kernel.org/r/87tupuqqc8.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87tupqpg9x.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (8): ASoC: soc-pcm: check DAI activity under soc_pcm_apply_symmetry() ASoC: soc-pcm: add soc_cpu/codec_dai_name() macro ASoC: soc-pcm: direct copy at snd_soc_set_runtime_hwparams() ASoC: soc-pcm: add soc_pcm_update_symmetry() ASoC: soc-pcm: add soc_hw_sanity_check() ASoC: soc-pcm: fixup dpcm_be_dai_startup() user count ASoC: soc-pcm: remove unneeded !rtd->dai_link check ASoC: soc-pcm: share DPCM BE DAI stop operation include/sound/soc-dpcm.h | 8 +- sound/soc/soc-compress.c | 2 +- sound/soc/soc-pcm.c | 243 ++++++++++++++++----------------------- 3 files changed, 105 insertions(+), 148 deletions(-) -- 2.25.1

2 9

[PATCH 0/4] ASoC: mediatek: remove cppcheck warnings
by Pierre-Louis Bossart 12 Mar '21

12 Mar '21

There should be no functionality change, this patchset only fixes shadowed variables, prototypes and removes a useless assignment. Pierre-Louis Bossart (4): ASoC: mediatek: mtk-btcvsd: remove useless assignment ASoC: mediatek: mt2701: align function prototype ASoC: mediatek: mt2701: rename shadowed array ASoC: mediatek: mt8173: rename local irq variable sound/soc/mediatek/common/mtk-btcvsd.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h | 4 ++-- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 4 ++-- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 6 +++--- 4 files changed, 8 insertions(+), 8 deletions(-) -- 2.25.1

2 5

[PATCH] ASoC: mediatek: mt8192: fix tdm out data is valid on rising edge
by Jiaxin Yu 12 Mar '21

12 Mar '21

This patch correct tdm out bck inverse register to AUDIO_TOP_CON3[3]. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 4 +++- sound/soc/mediatek/mt8192/mt8192-reg.h | 8 +++++--- 2 files changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c index f5de1d769679..f3bebed2428a 100644 --- a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c +++ b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c @@ -555,7 +555,9 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream, /* set tdm */ if (tdm_priv->bck_invert) - tdm_con |= 1 << BCK_INVERSE_SFT; + regmap_update_bits(afe->regmap, AUDIO_TOP_CON3, + BCK_INVERSE_MASK_SFT, + 0x1 << BCK_INVERSE_SFT); if (tdm_priv->lck_invert) tdm_con |= 1 << LRCK_INVERSE_SFT; diff --git a/sound/soc/mediatek/mt8192/mt8192-reg.h b/sound/soc/mediatek/mt8192/mt8192-reg.h index 562f25c79c34..b9fb80d4afec 100644 --- a/sound/soc/mediatek/mt8192/mt8192-reg.h +++ b/sound/soc/mediatek/mt8192/mt8192-reg.h @@ -21,6 +21,11 @@ enum { /***************************************************************************** * R E G I S T E R D E F I N I T I O N *****************************************************************************/ +/* AUDIO_TOP_CON3 */ +#define BCK_INVERSE_SFT 3 +#define BCK_INVERSE_MASK 0x1 +#define BCK_INVERSE_MASK_SFT (0x1 << 3) + /* AFE_DAC_CON0 */ #define VUL12_ON_SFT 31 #define VUL12_ON_MASK 0x1 @@ -2079,9 +2084,6 @@ enum { #define TDM_EN_SFT 0 #define TDM_EN_MASK 0x1 #define TDM_EN_MASK_SFT (0x1 << 0) -#define BCK_INVERSE_SFT 1 -#define BCK_INVERSE_MASK 0x1 -#define BCK_INVERSE_MASK_SFT (0x1 << 1) #define LRCK_INVERSE_SFT 2 #define LRCK_INVERSE_MASK 0x1 #define LRCK_INVERSE_MASK_SFT (0x1 << 2) -- 2.18.0

2 1

[PATCH v2] ASoC: rt1015p: add acpi device id for rt1015p
by Jack Yu 12 Mar '21

12 Mar '21

Add acpi device id for rt1015p. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015p.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/rt1015p.c b/sound/soc/codecs/rt1015p.c index 671f2a2130fe..39f4c4ed2ab8 100644 --- a/sound/soc/codecs/rt1015p.c +++ b/sound/soc/codecs/rt1015p.c @@ -4,6 +4,7 @@ // // Copyright 2020 The Linux Foundation. All rights reserved. +#include <linux/acpi.h> #include <linux/delay.h> #include <linux/device.h> #include <linux/err.h> @@ -130,10 +131,19 @@ static const struct of_device_id rt1015p_device_id[] = { MODULE_DEVICE_TABLE(of, rt1015p_device_id); #endif +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt1015p_acpi_match[] = { + { "RTL1015", 0}, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt1015p_acpi_match); +#endif + static struct platform_driver rt1015p_platform_driver = { .driver = { .name = "rt1015p", .of_match_table = of_match_ptr(rt1015p_device_id), + .acpi_match_table = ACPI_PTR(rt1015p_acpi_match), }, .probe = rt1015p_platform_probe, }; -- 2.29.0

2 1

[PATCH] ASoC: Intel: bytcr_rt5640: Enable jack-detect support on Asus T100TAF
by Hans de Goede 12 Mar '21

12 Mar '21

The Asus T100TAF uses the same jack-detect settings as the T100TA, this has been confirmed on actual hardware. Add these settings to the T100TAF quirks to enable jack-detect support on the T100TAF. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 1f6a636571c2..59d6d47c8d82 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -482,6 +482,9 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MONO_SPEAKER | BYT_RT5640_DIFF_MIC | BYT_RT5640_SSP0_AIF2 | -- 2.30.1

3 2

[PATCH] ASoC: rt1015p: add acpi device id for rt1015p
by jack.yu＠realtek.com 12 Mar '21

12 Mar '21

From: Jack Yu <jack.yu(a)realtek.com> Add acpi device id for rt1015p. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015p.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/rt1015p.c b/sound/soc/codecs/rt1015p.c index 671f2a2130fe..61694de080e6 100644 --- a/sound/soc/codecs/rt1015p.c +++ b/sound/soc/codecs/rt1015p.c @@ -4,6 +4,7 @@ // // Copyright 2020 The Linux Foundation. All rights reserved. +#include <linux/acpi.h> #include <linux/delay.h> #include <linux/device.h> #include <linux/err.h> @@ -130,10 +131,19 @@ static const struct of_device_id rt1015p_device_id[] = { MODULE_DEVICE_TABLE(of, rt1015p_device_id); #endif +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt1015p_acpi_match[] = { + { "ALCP1015", 0}, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt1015p_acpi_match); +#endif + static struct platform_driver rt1015p_platform_driver = { .driver = { .name = "rt1015p", .of_match_table = of_match_ptr(rt1015p_device_id), + .acpi_match_table = ACPI_PTR(rt1015p_acpi_match), }, .probe = rt1015p_platform_probe, }; -- 2.29.0

4 6

[PATCH v4 0/2] ASoC: samsung: remove cppcheck warnings
by Pierre-Louis Bossart 12 Mar '21

12 Mar '21

v4: corrected Fixes tag Added added Krzysztof Kozlowski's r-v-b tag reverted patch2 to v2 since this is the agreement. v3: Added Sylwester tag Rebased and squashed fix with initial patch which was merged at some point but can't be found in broonie/for-next (not sure what happened?) Corrected patch subjects to tm2_wm5110 Reverted second patch to initial v1, after agreement between Krzysztof and Sylwester v2: added Krzysztof Kozlowski's tags added fix for first patch already merged as suggested by Krzysztof Kozlowski moved variable to lower scope in patch6 Pierre-Louis Bossart (2): ASoC: samsung: tm2_wm5110: check of of_parse return value ASoC: samsung: tm2_wm5110: remove shadowed variable sound/soc/samsung/tm2_wm5110.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) -- 2.25.1

3 4

[PATCH] ASoC: soc-core: fix DMI handling
by Pierre-Louis Bossart 12 Mar '21

12 Mar '21

When DMI information is not present, trying to assign the card long name results in the following warning. WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name! The initial solution suggested was to test if the card device is an ACPI one. This causes a regression visible to userspace on all Intel platforms, with UCM unable to load card profiles based on DMI information: the card devices are not necessarily ACPI ones, e.g. when the parent creates platform devices on Intel devices. To fix this problem, this patch exports the existing dmi_available variable and tests it in the ASoC core. Fixes: c014170408bc ("ASoC: soc-core: Prevent warning if no DMI table is present") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/firmware/dmi_scan.c | 1 + sound/soc/soc-core.c | 2 +- 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/firmware/dmi_scan.c b/drivers/firmware/dmi_scan.c index d51ca0428bb8..f191a1f901ac 100644 --- a/drivers/firmware/dmi_scan.c +++ b/drivers/firmware/dmi_scan.c @@ -166,6 +166,7 @@ static int __init dmi_checksum(const u8 *buf, u8 len) static const char *dmi_ident[DMI_STRING_MAX]; static LIST_HEAD(dmi_devices); int dmi_available; +EXPORT_SYMBOL_GPL(dmi_available); /* * Save a DMI string diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16ba54eb8164..c7e4600b2dd4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1574,7 +1574,7 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) if (card->long_name) return 0; /* long name already set by driver or from DMI */ - if (!is_acpi_device_node(card->dev->fwnode)) + if (!dmi_available) return 0; /* make up dmi long name as: vendor-product-version-board */ -- 2.25.1

5 4

[PATCH] ALSA: dice: fix null pointer dereference when node is disconnected
by Takashi Sakamoto 12 Mar '21

12 Mar '21

When node is removed from IEEE 1394 bus, any transaction fails to the node. In the case, ALSA dice driver doesn't stop isochronous contexts even if they are running. As a result, null pointer dereference occurs in callback from the running context. This commit fixes the bug to release isochronous contexts always. Cc: <stable(a)vger.kernel.org> # v5.4 or later Fixes: e9f21129b8d8 ("ALSA: dice: support AMDTP domain") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-stream.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 8e0c0380b4c4..1a14c083e8ce 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -493,11 +493,10 @@ void snd_dice_stream_stop_duplex(struct snd_dice *dice) struct reg_params tx_params, rx_params; if (dice->substreams_counter == 0) { - if (get_register_params(dice, &tx_params, &rx_params) >= 0) { - amdtp_domain_stop(&dice->domain); + if (get_register_params(dice, &tx_params, &rx_params) >= 0) finish_session(dice, &tx_params, &rx_params); - } + amdtp_domain_stop(&dice->domain); release_resources(dice); } } -- 2.27.0

2 1

[PATCH] ALSA: hda: generic: Fix the micmute led init state
by Hui Wang 12 Mar '21

12 Mar '21

Recently we found the micmute led init state is not correct after freshly installing the ubuntu linux on a Lenovo AIO machine. The internal mic is not muted, but the micmute led is on and led mode is 'follow mute'. If we mute internal mic, the led is keeping on, then unmute the internal mic, the led is off. And from then on, the micmute led will work correctly. So the micmute led init state is not correct. The led is controlled by codec gpio (ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), in the patch_realtek, the gpio data is set to 0x4 initially and the led is on with this data. In the hda_generic, the led_value is set to 0 initially, suppose users set the 'capture switch' to on from user space and the micmute led should change to be off with this operation, but the check "if (val == spec->micmute_led.led_value)" in the call_micmute_led_update() will skip the led setting. To guarantee the led state will be set by the 1st time of changing "Capture Switch", set -1 to the init led_value. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8b7c5508f368..f5cba7afd1c6 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4065,7 +4065,7 @@ static int add_micmute_led_hook(struct hda_codec *codec) spec->micmute_led.led_mode = MICMUTE_LED_FOLLOW_MUTE; spec->micmute_led.capture = 0; - spec->micmute_led.led_value = 0; + spec->micmute_led.led_value = -1; spec->micmute_led.old_hook = spec->cap_sync_hook; spec->cap_sync_hook = update_micmute_led; if (!snd_hda_gen_add_kctl(spec, NULL, &micmute_led_mode_ctl)) -- 2.25.1

2 1

[RESEND PATCH v3 0/5] soundwire: add static port map support
by Srinivas Kandagatla 12 Mar '21

12 Mar '21

Resending this as I realized that alsa-devel list was missing in cc! In some cases, SoundWire device ports are statically mapped to Controller ports during design, however there is no way to expose this information to the controller. Controllers like Qualcomm ones use this info to setup static bandwidth parameters for those ports. A generic port allocation is not possible in this cases! This patch adds a new member m_port_map to SoundWire device so that it can populate the static master port map and share it with controller to be able to setup correct bandwidth parameters. As a user of this feature this patchset also adds new bindings for wsa881x smart speaker which has 4 ports which are statically mapped to the 3 output and 1 input port of the controller. Tested it on DB845c and SM8250 MTP. thanks, srini Changes since v2: - fixed dt_binding_check error and kernel test robot reported errors - updated comments as suggested by Pierre - updated wsa881x to populate m_ports from valid index. Srinivas Kandagatla (5): soundwire: add static port mapping support soundwire: qcom: update port map allocation bit mask soundwire: qcom: add static port map support ASoC: dt-bindings: wsa881x: add bindings for port mapping ASoC: codecs: wsa881x: add static port map support .../bindings/sound/qcom,wsa881x.yaml | 9 ++++++ drivers/soundwire/qcom.c | 31 +++++++++++++++---- include/linux/soundwire/sdw.h | 2 ++ sound/soc/codecs/wsa881x.c | 7 +++++ 4 files changed, 43 insertions(+), 6 deletions(-) -- 2.21.0

1 5

[PATCH v4 0/5] Add missing reset controls to NVIDIA Tegra ASoC drivers
by Dmitry Osipenko 11 Mar '21

11 Mar '21

Hi, This series adds missing hardware reset controls to I2S and AC97 drivers. Currently drivers happen to work properly because reset is implicitly deasserted by tegra-clk driver, but clk driver shouldn't touch the resets and we need to fix it because this breaks other Tegra drivers. Previously we fixed the resets of the AHUB and HDMI codec drivers, but turned out that we missed the I2C and AC97 drivers. Thanks to Paul Fertser for testing the pending clk patches and finding that audio got broken on Tegra20 AC100 netbook because of the missing I2S reset. Changelog: v4: - Added missing prototype for reset_control_bulk_put(). v3: - Fixed reset stubs for !CONFIG_RESET_CONTROLLER. v2: - After some more testing I found that I2S control logic doesn't require I2S clock to be enabled for resetting. Hence it's fine to have I2S to be reset by parent AHUB driver, so I dropped "tegra30: i2s: Add reset control" patch. - While I was double-checking resets on Tegra30, I found that that Tegra30 I2S driver has a broken runtime PM which doesn't restore hardware state on resume and it's lost after AHUB RPM-resume. Thus, added this new patch "tegra30: i2s: Restore hardware state on runtime PM resume". - Added new patches which switch AHUB driver to use reset-bulk API. I took the RFC patch from Philipp Zabel, fixed it and added devm_reset_control_bulk_optional_get_exclusive_released() that will be useful for further Tegra GPU patches. This is a minor improvement which makes code cleaner. Dmitry Osipenko (4): ASoC: tegra20: ac97: Add reset control ASoC: tegra20: i2s: Add reset control ASoC: tegra30: i2s: Restore hardware state on runtime PM resume ASoC: tegra: ahub: Switch to use reset-bulk API Philipp Zabel (1): reset: Add reset_control_bulk API drivers/reset/core.c | 215 ++++++++++++++++++++++ include/linux/reset.h | 315 +++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra20_ac97.c | 21 +++ sound/soc/tegra/tegra20_ac97.h | 1 + sound/soc/tegra/tegra20_i2s.c | 31 ++++ sound/soc/tegra/tegra20_i2s.h | 1 + sound/soc/tegra/tegra30_ahub.c | 104 ++++------- sound/soc/tegra/tegra30_ahub.h | 5 +- sound/soc/tegra/tegra30_i2s.c | 41 ++--- 9 files changed, 636 insertions(+), 98 deletions(-) -- 2.29.2

1 5

[PATCH v3 0/5] Add missing reset controls to NVIDIA Tegra ASoC drivers
by Dmitry Osipenko 11 Mar '21

11 Mar '21

Hi, This series adds missing hardware reset controls to I2S and AC97 drivers. Currently drivers happen to work properly because reset is implicitly deasserted by tegra-clk driver, but clk driver shouldn't touch the resets and we need to fix it because this breaks other Tegra drivers. Previously we fixed the resets of the AHUB and HDMI codec drivers, but turned out that we missed the I2C and AC97 drivers. Thanks to Paul Fertser for testing the pending clk patches and finding that audio got broken on Tegra20 AC100 netbook because of the missing I2S reset. Changelog: v3: - Fixed reset stubs for !CONFIG_RESET_CONTROLLER. v2: - After some more testing I found that I2S control logic doesn't require I2S clock to be enabled for resetting. Hence it's fine to have I2S to be reset by parent AHUB driver, so I dropped "tegra30: i2s: Add reset control" patch. - While I was double-checking resets on Tegra30, I found that that Tegra30 I2S driver has a broken runtime PM which doesn't restore hardware state on resume and it's lost after AHUB RPM-resume. Thus, added this new patch "tegra30: i2s: Restore hardware state on runtime PM resume". - Added new patches which switch AHUB driver to use reset-bulk API. I took the RFC patch from Philipp Zabel, fixed it and added devm_reset_control_bulk_optional_get_exclusive_released() that will be useful for further Tegra GPU patches. This is a minor improvement which makes code cleaner. Dmitry Osipenko (4): ASoC: tegra20: ac97: Add reset control ASoC: tegra20: i2s: Add reset control ASoC: tegra30: i2s: Restore hardware state on runtime PM resume ASoC: tegra: ahub: Switch to use reset-bulk API Philipp Zabel (1): reset: Add reset_control_bulk API drivers/reset/core.c | 215 +++++++++++++++++++++++ include/linux/reset.h | 308 +++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra20_ac97.c | 21 +++ sound/soc/tegra/tegra20_ac97.h | 1 + sound/soc/tegra/tegra20_i2s.c | 31 ++++ sound/soc/tegra/tegra20_i2s.h | 1 + sound/soc/tegra/tegra30_ahub.c | 104 ++++------- sound/soc/tegra/tegra30_ahub.h | 5 +- sound/soc/tegra/tegra30_i2s.c | 41 ++--- 9 files changed, 629 insertions(+), 98 deletions(-) -- 2.29.2

1 5

[PATCH v2 0/5] Add missing reset controls to NVIDIA Tegra ASoC drivers
by Dmitry Osipenko 11 Mar '21

11 Mar '21

Hi, This series adds missing hardware reset controls to I2S and AC97 drivers. Currently drivers happen to work properly because reset is implicitly deasserted by tegra-clk driver, but clk driver shouldn't touch the resets and we need to fix it because this breaks other Tegra drivers. Previously we fixed the resets of the AHUB and HDMI codec drivers, but turned out that we missed the I2C and AC97 drivers. Thanks to Paul Fertser for testing the pending clk patches and finding that audio got broken on Tegra20 AC100 netbook because of the missing I2S reset. Changelog: v2: - After some more testing I found that I2S control logic doesn't require I2S clock to be enabled for resetting. Hence it's fine to have I2S to be reset by parent AHUB driver, so I dropped "tegra30: i2s: Add reset control" patch. - While I was double-checking resets on Tegra30, I found that that Tegra30 I2S driver has a broken runtime PM which doesn't restore hardware state on resume and it's lost after AHUB RPM-resume. Thus, added this new patch "tegra30: i2s: Restore hardware state on runtime PM resume". - Added new patches which switch AHUB driver to use reset-bulk API. I took the RFC patch from Philipp Zabel, fixed it and added devm_reset_control_bulk_optional_get_exclusive_released() that will be useful for further Tegra GPU patches. This is a minor improvement which makes code cleaner. Dmitry Osipenko (4): ASoC: tegra20: ac97: Add reset control ASoC: tegra20: i2s: Add reset control ASoC: tegra30: i2s: Restore hardware state on runtime PM resume ASoC: tegra: ahub: Switch to use reset-bulk API Philipp Zabel (1): reset: Add reset_control_bulk API drivers/reset/core.c | 215 +++++++++++++++++++++++++ include/linux/reset.h | 279 +++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra20_ac97.c | 21 +++ sound/soc/tegra/tegra20_ac97.h | 1 + sound/soc/tegra/tegra20_i2s.c | 31 ++++ sound/soc/tegra/tegra20_i2s.h | 1 + sound/soc/tegra/tegra30_ahub.c | 104 +++++------- sound/soc/tegra/tegra30_ahub.h | 5 +- sound/soc/tegra/tegra30_i2s.c | 41 ++--- 9 files changed, 600 insertions(+), 98 deletions(-) -- 2.29.2

2 8

[PATCH] ASoC: qcom: lpass-cpu: Fix lpass dai ids parse
by Srinivasa Rao Mandadapu 11 Mar '21

11 Mar '21

The max boundary check while parsing dai ids makes sound card registration fail after common up dai ids. Fixes: cd3484f7f1386 (ASoC: dt-bindings: lpass: Fix and common up lpass dai ids) Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 4762286b33fe..c62d2612e8f5 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -739,7 +739,7 @@ static void of_lpass_cpu_parse_dai_data(struct device *dev, for_each_child_of_node(dev->of_node, node) { ret = of_property_read_u32(node, "reg", &id); - if (ret || id < 0 || id >= data->variant->num_dai) { + if (ret || id < 0) { dev_err(dev, "valid dai id not found: %d\n", ret); continue; } -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 3

[PATCH v2] ASoC: qcom: lpass-cpu: Fix lpass dai ids parse
by Srinivasa Rao Mandadapu 11 Mar '21

11 Mar '21

The max boundary check while parsing dai ids makes sound card registration fail after common up dai ids. Fixes: cd3484f7f138 ("ASoC: qcom: Fix broken support to MI2S TERTIARY and QUATERNARY") Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes since v1: -- Fixes commit message chnaged. sound/soc/qcom/lpass-cpu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 4762286b33fe..c62d2612e8f5 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -739,7 +739,7 @@ static void of_lpass_cpu_parse_dai_data(struct device *dev, for_each_child_of_node(dev->of_node, node) { ret = of_property_read_u32(node, "reg", &id); - if (ret || id < 0 || id >= data->variant->num_dai) { + if (ret || id < 0) { dev_err(dev, "valid dai id not found: %d\n", ret); continue; } -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 1

[PATCH 0/2] ASoC: mediatek: mt8183-mt6358: support machine driver for rt1015p
by Tzung-Bi Shih 11 Mar '21

11 Mar '21

The series reuses mt8183-mt6358-ts3a227-max98357.c for supporting machine driver with rt1015p speaker amplifier. The 1st patch adds document for the new proposed compatible string. The 2nd patch changes the machine driver to support "RT1015P" codec. Tzung-Bi Shih (2): ASoC: dt-bindings: mt8183: add compatible string for using rt1015p ASoC: mediatek: mt8183: support machine driver with rt1015p .../sound/mt8183-mt6358-ts3a227-max98357.txt | 1 + sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 29 +++++++++++++++++++ 3 files changed, 31 insertions(+) -- 2.31.0.rc2.261.g7f71774620-goog

2 3

[PATCH 00/10] ASoC: codecs: wolfson: remove cppcheck warnings
by Pierre-Louis Bossart 11 Mar '21

11 Mar '21

There should be no functionality change, just minor fixes to make warnings go away. Pierre-Louis Bossart (10): ASoC: arizona: fix function argument ASoC: madera: align function prototype ASoC: wm2200: remove unused structure ASoC: wm8903: remove useless assignments ASoC: wm8958-dsp2: rename local 'control' arrays ASoC: wm8978: clarify expression ASoC: wm8994: align function prototype ASoC: wm8996: clarify expression ASoC: wm_adsp: simplify return value ASoC: wm_hubs: align function prototype sound/soc/codecs/arizona.h | 2 +- sound/soc/codecs/madera.h | 2 +- sound/soc/codecs/wm2200.c | 7 ------- sound/soc/codecs/wm8903.c | 2 -- sound/soc/codecs/wm8958-dsp2.c | 16 ++++++++-------- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8994.h | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm_adsp.c | 2 +- sound/soc/codecs/wm_hubs.h | 2 +- 10 files changed, 15 insertions(+), 24 deletions(-) -- 2.25.1

3 21

[PATCH v2 0/5] soundwire: add static port map support
by Srinivas Kandagatla 11 Mar '21

11 Mar '21

In some cases, SoundWire device ports are statically mapped to Controller ports during design, however there is no way to expose this information to the controller. Controllers like Qualcomm ones use this info to setup static bandwith parameters for those ports. A generic port allocation is not possible in this cases! This patch adds a new member m_port_map to SoundWire device so that it can populate the static master port map and share it with controller to be able to setup correct bandwidth parameters. As a user of this feature this patchset also adds new bindings for wsa881x smart speaker which has 4 ports which are statically mapped to the 3 output and 1 input port of the controller. Tested it on DB845c and SM8250 MTP. thanks, srini Srinivas Kandagatla (5): soundwire: add static port mapping support soundwire: qcom: update port map allocation bit mask soundwire: qcom: add static port map support ASoC: dt-bindings: wsa881x: add bindings for port mapping ASoC: codecs: wsa881x: add static port map support .../bindings/sound/qcom,wsa881x.yaml | 9 ++++++ drivers/soundwire/qcom.c | 30 +++++++++++++++---- include/linux/soundwire/sdw.h | 2 ++ sound/soc/codecs/wsa881x.c | 5 ++++ 4 files changed, 40 insertions(+), 6 deletions(-) -- 2.21.0

4 12

[PATCH v1 1/1] ALSA: hda/realtek: Sort alphanumerically the SND_PCI_QUIRK() entries.
by Andy Shevchenko 11 Mar '21

11 Mar '21

Sort alphanumerically the SND_PCI_QUIRK() entries for better maintenance. No functional change implied. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/pci/hda/patch_realtek.c | 73 +++++++++++++++++------------------ 1 file changed, 36 insertions(+), 37 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c2ae077bd010..d23c8c2e6833 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1084,10 +1084,10 @@ static const struct snd_pci_quirk beep_allow_list[] = { SND_PCI_QUIRK(0x1043, 0x103c, "ASUS", 1), SND_PCI_QUIRK(0x1043, 0x115d, "ASUS", 1), SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1), - SND_PCI_QUIRK(0x1043, 0x8376, "EeePC", 1), - SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1), SND_PCI_QUIRK(0x1043, 0x834a, "EeePC", 1), + SND_PCI_QUIRK(0x1043, 0x8376, "EeePC", 1), + SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), SND_PCI_QUIRK(0x1458, 0xa002, "GA-MA790X", 1), SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1), /* denylist -- no beep available */ @@ -2489,11 +2489,11 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS), SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3), + SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), + SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9060, "Sony Vaio VPCL14M1R", ALC882_FIXUP_NO_PRIMARY_HP), - SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), - SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP), /* All Apple entries are in codec SSIDs */ SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), @@ -2536,6 +2536,14 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), + SND_PCI_QUIRK(0x1558, 0x50d3, "Clevo PC50[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x65d1, "Clevo PB51[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x65d2, "Clevo PB51R[CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x65e1, "Clevo PB51[ED][DF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x70d1, "Clevo PC70[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x9501, "Clevo P950HR", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x9506, "Clevo P955HQ", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x950A, "Clevo P955H[PR]", ALC1220_FIXUP_CLEVO_P950), @@ -2548,14 +2556,6 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x96e1, "Clevo P960[ER][CDFN]-K", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x97e1, "Clevo P970[ER][CDFN]", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x97e2, "Clevo P970RC-M", ALC1220_FIXUP_CLEVO_P950), - SND_PCI_QUIRK(0x1558, 0x50d3, "Clevo PC50[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), - SND_PCI_QUIRK(0x1558, 0x65d1, "Clevo PB51[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), - SND_PCI_QUIRK(0x1558, 0x65d2, "Clevo PB51R[CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), - SND_PCI_QUIRK(0x1558, 0x65e1, "Clevo PB51[ED][DF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), - SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), - SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), - SND_PCI_QUIRK(0x1558, 0x70d1, "Clevo PC70[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), - SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), @@ -7890,12 +7890,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), SND_PCI_QUIRK(0x1025, 0x072d, "Acer Aspire V5-571G", ALC269_FIXUP_ASPIRE_HEADSET_MIC), - SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), + SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x101c, "Acer Veriton N2510G", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC), @@ -7950,8 +7950,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), - SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), + SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), @@ -7961,8 +7961,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x08ad, "Dell WYSE AIO", ALC225_FIXUP_DELL_WYSE_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x08ae, "Dell WYSE NB", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0935, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), - SND_PCI_QUIRK(0x1028, 0x097e, "Dell Precision", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x097d, "Dell Precision", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x097e, "Dell Precision", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x098d, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x09bf, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a2e, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), @@ -8000,6 +8000,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x22fb, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), /* ALC290 */ SND_PCI_QUIRK(0x103c, 0x221b, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2225, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2253, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), @@ -8020,15 +8021,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2282, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x228b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x228e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c4, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22c5, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22c7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22c8, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), - SND_PCI_QUIRK(0x103c, 0x22c4, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2334, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), - SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), SND_PCI_QUIRK(0x103c, 0x802e, "HP Z240 SFF", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), @@ -8063,16 +8063,18 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1043, 0x125e, "ASUS Q524UQK", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), + SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE), @@ -8085,31 +8087,30 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x1043, 0x125e, "ASUS Q524UQK", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), - SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), - SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9099, "Sony VAIO S13", ALC275_FIXUP_SONY_DISABLE_AAMIX), + SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x10cf, 0x159f, "Lifebook E780", ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT), SND_PCI_QUIRK(0x10cf, 0x15dc, "Lifebook T731", ALC269_FIXUP_LIFEBOOK_HP_PIN), - SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC), + SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), + SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1252, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), @@ -8119,9 +8120,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), - SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), @@ -8177,9 +8178,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST), SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), @@ -8223,6 +8224,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), + SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), @@ -8241,18 +8243,16 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), - SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI), SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ - SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), - SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), - SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), + SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), + SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), @@ -8262,11 +8262,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { * If BIOS doesn't give a proper info, enable the corresponding * fixup entry. */ - SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", - ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC), - SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x11b3, "ASUS K52DR", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269_FIXUP_AMIC), @@ -8274,16 +8272,16 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82JV", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_FIXUP_AMIC), - SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_FIXUP_AMIC), - SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1593, "ASUS N51Vn", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_FIXUP_AMIC), @@ -8298,6 +8296,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x17aa, 0x3be9, "Quanta Wistron", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_AMIC), @@ -9197,8 +9196,8 @@ static const struct snd_pci_quirk alc861_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F), SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT), - SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", ALC861_FIXUP_AMP_VREF_0F), SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", ALC861_FIXUP_AMP_VREF_0F), + SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", ALC861_FIXUP_AMP_VREF_0F), SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", ALC861_FIXUP_FSC_AMILO_PI1505), {} }; @@ -9993,6 +9992,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x0566, "Acer Aspire Ethos 8951G", ALC669_FIXUP_ACER_ASPIRE_ETHOS), SND_PCI_QUIRK(0x1025, 0x123c, "Acer Nitro N50-600", ALC662_FIXUP_ACER_NITRO_HEADSET_MODE), SND_PCI_QUIRK(0x1025, 0x124e, "Acer 2660G", ALC662_FIXUP_ACER_X2660G_HEADSET_MODE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), @@ -10009,9 +10009,9 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50), - SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A), SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x12ff, "ASUS G751", ALC668_FIXUP_ASUS_G751), + SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), @@ -10031,7 +10031,6 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1b0a, 0x01b8, "ACER Veriton", ALC662_FIXUP_ACER_VERITON), SND_PCI_QUIRK(0x1b35, 0x1234, "CZC ET26", ALC662_FIXUP_CZC_ET26), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), - SND_PCI_QUIRK(0x1025, 0x0566, "Acer Aspire Ethos 8951G", ALC669_FIXUP_ACER_ASPIRE_ETHOS), #if 0 /* Below is a quirk table taken from the old code. -- 2.30.1

3 5

No sound cards detected on Kabylake laptops after upgrade to kernel 5.8
by Chris Chiu 11 Mar '21

11 Mar '21

Hi Guys, We have received reports that on some Kabylake laptops (Acer Swift SF314-54/55 and Lenovo Yoga C930...etc), all sound cards no longer be detected after upgrade to kernel later than 5.8. These laptops have one thing in common, all of them have Realtek audio codec and connect the internal microphone to DMIC of the Intel SST controller either [8086:9d71] or [8086:9dc8]. Please refer to https://bugzilla.kernel.org/show_bug.cgi?id=201251#c246 and https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1915117. From the dmesg from kernel 5.8, the sound related parts only show as follows but the expected snd_hda_codec_realtek and the snd_soc_skl are not even loaded then. [ 13.357495] snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 [ 13.357500] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SST driver Building the kernel with the CONFIG_SND_SOC_INTEL_KBL removed can load the snd_hda_codec_realtek successfully and the pulseaudio and alsa-utils can detect the sound cards again. The result of bisecting between kernel 5.4 and 5.8 also get similar result, reverting the commit "ALSA: hda: Allow SST driver on SKL and KBL platforms with DMIC" can fix the issue. I tried to generate the required firmware for snd_soc_skl but it did not help. Please refer to what I did in https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1915117/comments/14 and https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1915117/comments/18. Since the skl_hda_dsp_generic-tplg.bin and dfw_sst.bin are not in the linux-firmware. The Intel SST support for Skylake family is not yet complete. Can we simply revert the "ALSA: hda: Allow SST driver on SKL and KBL platforms with DMIC" in the current stage and wait for SOF support for Skylake family? Or please suggest a better solution for this. Thanks Chris

4 6

[PATCH v1 1/1] ALSA: hda/realtek: Sort alphanumerically the HDA_CODEC_ENTRY() entries.
by Andy Shevchenko 11 Mar '21

11 Mar '21

Sort alphanumerically the HDA_CODEC_ENTRY() entries for better maintenance. No functional change implied. Note, that HDA_CODEC_REV_ENTRY() goes after HDA_CODEC_ENTRY() as per default value in the latter macro. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/pci/hda/patch_realtek.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d23c8c2e6833..c9e62693db6d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10341,10 +10341,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0299, "ALC299", patch_alc269), HDA_CODEC_ENTRY(0x10ec0300, "ALC300", patch_alc269), HDA_CODEC_ENTRY(0x10ec0623, "ALC623", patch_alc269), - HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861), HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd), - HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861), - HDA_CODEC_ENTRY(0x10ec0862, "ALC861-VD", patch_alc861vd), HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100002, "ALC662 rev2", patch_alc882), HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100101, "ALC662 rev1", patch_alc662), HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100300, "ALC662 rev3", patch_alc662), @@ -10359,16 +10356,19 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0701, "ALC701", patch_alc269), HDA_CODEC_ENTRY(0x10ec0703, "ALC703", patch_alc269), HDA_CODEC_ENTRY(0x10ec0711, "ALC711", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861), + HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861), + HDA_CODEC_ENTRY(0x10ec0862, "ALC861-VD", patch_alc861vd), HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc662), HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880), HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882), HDA_CODEC_ENTRY(0x10ec0883, "ALC883", patch_alc882), + HDA_CODEC_ENTRY(0x10ec0885, "ALC885", patch_alc882), HDA_CODEC_REV_ENTRY(0x10ec0885, 0x100101, "ALC889A", patch_alc882), HDA_CODEC_REV_ENTRY(0x10ec0885, 0x100103, "ALC889A", patch_alc882), - HDA_CODEC_ENTRY(0x10ec0885, "ALC885", patch_alc882), HDA_CODEC_ENTRY(0x10ec0887, "ALC887", patch_alc882), - HDA_CODEC_REV_ENTRY(0x10ec0888, 0x100101, "ALC1200", patch_alc882), HDA_CODEC_ENTRY(0x10ec0888, "ALC888", patch_alc882), + HDA_CODEC_REV_ENTRY(0x10ec0888, 0x100101, "ALC1200", patch_alc882), HDA_CODEC_ENTRY(0x10ec0889, "ALC889", patch_alc882), HDA_CODEC_ENTRY(0x10ec0892, "ALC892", patch_alc662), HDA_CODEC_ENTRY(0x10ec0897, "ALC897", patch_alc662), -- 2.30.1

3 2

Re: [PATCH v2 3/3] ASoC: codecs: wcd934x: add a sanity check in set channel map
by Sufyan Sheikh 11 Mar '21

11 Mar '21

1 0

[RFCv2] ASoC: Add Rockchip rk817 audio CODEC support
by Chris Morgan 11 Mar '21

11 Mar '21

Add support for the Rockchip rk817 audio codec integrated into the rk817 PMIC. This is based on the sources provided by Rockchip from their BSP kernel. Would someone advise what they think the best way to get the widgets set up properly would be? I'm still new to this and I don't fully understand all the macros (the alsa-project.org documentation only gets me so far). TODO: Get HP detect working, see if I can improve the audio path widgets. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- .../bindings/sound/rockchip,rk817-codec.txt | 39 + drivers/mfd/rk808.c | 6 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 946 ++++++++++++++++++ sound/soc/codecs/rk817_codec.h | 142 +++ 6 files changed, 1141 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk817-codec.txt create mode 100644 sound/soc/codecs/rk817_codec.c create mode 100644 sound/soc/codecs/rk817_codec.h diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.txt new file mode 100644 index 000000000000..6d4e05440dae --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.txt @@ -0,0 +1,39 @@ +* Rockchip rk817 codec + +Required properties: + +- compatible: "rockchip,rk817-codec" +- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. +- clock-names: should be "mclk". + +Optional properties: + +- mic-in-differential: + Boolean. Indicate MIC input are differential, rather than single-ended. + +Example for rk817 codec: + +rk817: pmic@20 { + compatible = "rockchip,rk817"; + reg = <0x20>; + interrupt-parent = <&gpio0>; + interrupts = <7 IRQ_TYPE_LEVEL_LOW>; + pinctrl-names = "default"; + pinctrl-0 = <&pmic_int>; + + ........ + + rk817_codec: codec { + #sound-dai-cells = <0>; + compatible = "rockchip,rk817-codec"; + clocks = <&cru SCLK_I2S_8CH_OUT>; + clock-names = "mclk"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s_8ch_mclk>; + mic-in-differential; + status = "okay"; + }; + + ........ + +}; diff --git a/drivers/mfd/rk808.c b/drivers/mfd/rk808.c index ad923dd4e007..adb8a7da29db 100644 --- a/drivers/mfd/rk808.c +++ b/drivers/mfd/rk808.c @@ -163,6 +163,12 @@ static const struct mfd_cell rk817s[] = { .num_resources = ARRAY_SIZE(rk817_rtc_resources), .resources = &rk817_rtc_resources[0], }, +#ifdef CONFIG_SND_SOC_RK817 + { + .name = "rk817-codec", + .of_compatible = "rockchip,rk817-codec", + }, +#endif }; static const struct mfd_cell rk818s[] = { diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e4cf14e66a51..d835145ad65e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -155,6 +155,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_PCM512x_I2C imply SND_SOC_PCM512x_SPI imply SND_SOC_RK3328 + imply SND_SOC_RK817 imply SND_SOC_RT274 imply SND_SOC_RT286 imply SND_SOC_RT298 @@ -1059,6 +1060,11 @@ config SND_SOC_RK3328 tristate "Rockchip RK3328 audio CODEC" select REGMAP_MMIO +config SND_SOC_RK817 + tristate "Rockchip RK817 audio CODEC" + depends on MFD_RK808 + select REGMAP_I2C + config SND_SOC_RL6231 tristate default y if SND_SOC_RT5514=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 81357dc62ea0..4e5e9b28775c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -165,6 +165,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rk3328-objs := rk3328_codec.o +snd-soc-rk817-objs := rk817_codec.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1011-objs := rt1011.o @@ -479,6 +480,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o +obj-$(CONFIG_SND_SOC_RK817) += snd-soc-rk817.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c new file mode 100644 index 000000000000..285c3bc9ce5d --- /dev/null +++ b/sound/soc/codecs/rk817_codec.c @@ -0,0 +1,946 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rk817 ALSA SoC Audio driver +// +// Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/mfd/rk808.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "rk817_codec.h" + +/* For route */ +#define RK817_CODEC_PLAYBACK 1 +#define RK817_CODEC_CAPTURE 2 +#define RK817_CODEC_ALL (RK817_CODEC_PLAYBACK |\ + RK817_CODEC_CAPTURE) + +struct rk817_codec_priv { + struct snd_soc_component *component; + struct regmap *regmap; + struct rk808 *rk817; + struct clk *mclk; + + unsigned int stereo_sysclk; + + bool mic_in_differential; + + int playback_path; + int capture_path; +}; + +static const struct reg_default rk817_reg_defaults[] = { + { RK817_CODEC_DTOP_VUCTL, 0x03 }, + { RK817_CODEC_DTOP_VUCTIME, 0x00 }, + { RK817_CODEC_DTOP_LPT_SRST, 0x00 }, + { RK817_CODEC_DTOP_DIGEN_CLKE, 0x00 }, + /* CODEC_AREF_RTCFG0 not defined in data sheet */ + { RK817_CODEC_AREF_RTCFG0, 0x00 }, + { RK817_CODEC_AREF_RTCFG1, 0x06 }, + { RK817_CODEC_AADC_CFG0, 0xc8 }, + /* CODEC_AADC_CFG1 not defined in data sheet */ + { RK817_CODEC_AADC_CFG1, 0x00 }, + { RK817_CODEC_DADC_VOLL, 0x00 }, + { RK817_CODEC_DADC_VOLR, 0x00 }, + { RK817_CODEC_DADC_SR_ACL0, 0x00 }, + { RK817_CODEC_DADC_ALC1, 0x00 }, + { RK817_CODEC_DADC_ALC2, 0x00 }, + { RK817_CODEC_DADC_NG, 0x00 }, + { RK817_CODEC_DADC_HPF, 0x00 }, + { RK817_CODEC_DADC_RVOLL, 0xff }, + { RK817_CODEC_DADC_RVOLR, 0xff }, + { RK817_CODEC_AMIC_CFG0, 0x70 }, + { RK817_CODEC_AMIC_CFG1, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, 0x66 }, + { RK817_CODEC_DMIC_LMT1, 0x00 }, + { RK817_CODEC_DMIC_LMT2, 0x00 }, + { RK817_CODEC_DMIC_NG1, 0x00 }, + { RK817_CODEC_DMIC_NG2, 0x00 }, + /* CODEC_ADAC_CFG0 not defined in data sheet */ + { RK817_CODEC_ADAC_CFG0, 0x00 }, + { RK817_CODEC_ADAC_CFG1, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, 0x82 }, + { RK817_CODEC_DDAC_VOLL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, 0x00 }, + { RK817_CODEC_DDAC_LMT1, 0x00 }, + { RK817_CODEC_DDAC_LMT2, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, 0xff }, + { RK817_CODEC_DDAC_RVOLR, 0xff }, + { RK817_CODEC_AHP_ANTI0, 0x00 }, + { RK817_CODEC_AHP_ANTI1, 0x00 }, + { RK817_CODEC_AHP_CFG0, 0xe0 }, + { RK817_CODEC_AHP_CFG1, 0x1f }, + { RK817_CODEC_AHP_CP, 0x09 }, + { RK817_CODEC_ACLASSD_CFG1, 0x69 }, + { RK817_CODEC_ACLASSD_CFG2, 0x44 }, + { RK817_CODEC_APLL_CFG0, 0x04 }, + { RK817_CODEC_APLL_CFG1, 0x00 }, + { RK817_CODEC_APLL_CFG2, 0x30 }, + { RK817_CODEC_APLL_CFG3, 0x19 }, + { RK817_CODEC_APLL_CFG4, 0x65 }, + { RK817_CODEC_APLL_CFG5, 0x01 }, + { RK817_CODEC_DI2S_CKM, 0x01 }, + { RK817_CODEC_DI2S_RSD, 0x00 }, + { RK817_CODEC_DI2S_RXCR1, 0x00 }, + { RK817_CODEC_DI2S_RXCR2, 0x17 }, + { RK817_CODEC_DI2S_RXCMD_TSD, 0x00 }, + { RK817_CODEC_DI2S_TXCR1, 0x00 }, + { RK817_CODEC_DI2S_TXCR2, 0x17 }, + { RK817_CODEC_DI2S_TXCR3_TXCMD, 0x00 }, +}; + +/* + * DDAC L/R volume setting + * 0db~-95db,0.375db/step,for example: + * 0: 0dB + * 0x0a: -3.75dB + * 0x7d: -46dB + * 0xff: -95dB + * + * DADC L/R volume setting + * 0db~-95db,0.375db/step,for example: + * 0: 0dB + * 0x0a: -3.75dB + * 0x7d: -46dB + * 0xff: -95dB + */ + +static const DECLARE_TLV_DB_MINMAX(rk817_dacvol_tlv, -9500, 0); + +static const DECLARE_TLV_DB_MINMAX(rk817_adcvol_tlv, -9500, 0); + +static const struct snd_kcontrol_new rk817_dac_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("Playback Volume", RK817_CODEC_DDAC_VOLL, + RK817_CODEC_DDAC_VOLR, 0, 0x00, 0xff, 1, rk817_dacvol_tlv), + SOC_DOUBLE_R_RANGE_TLV("Capture Volume", RK817_CODEC_DADC_VOLL, RK817_CODEC_DADC_VOLR, + 0, 0x00, 0xff, 1, rk817_adcvol_tlv) +}; + +static bool rk817_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RK817_CODEC_DTOP_LPT_SRST: + return true; + default: + return false; + } +} + +static bool rk817_codec_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RK817_CODEC_DTOP_VUCTL: + case RK817_CODEC_DTOP_VUCTIME: + case RK817_CODEC_DTOP_LPT_SRST: + case RK817_CODEC_DTOP_DIGEN_CLKE: + case RK817_CODEC_AREF_RTCFG0: + case RK817_CODEC_AREF_RTCFG1: + case RK817_CODEC_AADC_CFG0: + case RK817_CODEC_AADC_CFG1: + case RK817_CODEC_DADC_VOLL: + case RK817_CODEC_DADC_VOLR: + case RK817_CODEC_DADC_SR_ACL0: + case RK817_CODEC_DADC_ALC1: + case RK817_CODEC_DADC_ALC2: + case RK817_CODEC_DADC_NG: + case RK817_CODEC_DADC_HPF: + case RK817_CODEC_DADC_RVOLL: + case RK817_CODEC_DADC_RVOLR: + case RK817_CODEC_AMIC_CFG0: + case RK817_CODEC_AMIC_CFG1: + case RK817_CODEC_DMIC_PGA_GAIN: + case RK817_CODEC_DMIC_LMT1: + case RK817_CODEC_DMIC_LMT2: + case RK817_CODEC_DMIC_NG1: + case RK817_CODEC_DMIC_NG2: + case RK817_CODEC_ADAC_CFG0: + case RK817_CODEC_ADAC_CFG1: + case RK817_CODEC_DDAC_POPD_DACST: + case RK817_CODEC_DDAC_VOLL: + case RK817_CODEC_DDAC_VOLR: + case RK817_CODEC_DDAC_SR_LMT0: + case RK817_CODEC_DDAC_LMT1: + case RK817_CODEC_DDAC_LMT2: + case RK817_CODEC_DDAC_MUTE_MIXCTL: + case RK817_CODEC_DDAC_RVOLL: + case RK817_CODEC_DDAC_RVOLR: + case RK817_CODEC_AHP_ANTI0: + case RK817_CODEC_AHP_ANTI1: + case RK817_CODEC_AHP_CFG0: + case RK817_CODEC_AHP_CFG1: + case RK817_CODEC_AHP_CP: + case RK817_CODEC_ACLASSD_CFG1: + case RK817_CODEC_ACLASSD_CFG2: + case RK817_CODEC_APLL_CFG0: + case RK817_CODEC_APLL_CFG1: + case RK817_CODEC_APLL_CFG2: + case RK817_CODEC_APLL_CFG3: + case RK817_CODEC_APLL_CFG4: + case RK817_CODEC_APLL_CFG5: + case RK817_CODEC_DI2S_CKM: + case RK817_CODEC_DI2S_RSD: + case RK817_CODEC_DI2S_RXCR1: + case RK817_CODEC_DI2S_RXCR2: + case RK817_CODEC_DI2S_RXCMD_TSD: + case RK817_CODEC_DI2S_TXCR1: + case RK817_CODEC_DI2S_TXCR2: + case RK817_CODEC_DI2S_TXCR3_TXCMD: + return true; + default: + return false; + } +} + +static int rk817_reset(struct snd_soc_component *component) +{ + snd_soc_component_write(component, RK817_CODEC_DTOP_LPT_SRST, 0x40); + snd_soc_component_write(component, RK817_CODEC_DDAC_POPD_DACST, 0x02); + snd_soc_component_write(component, RK817_CODEC_DTOP_DIGEN_CLKE, 0x0f); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG0, 0x04); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG1, 0x58); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG2, 0x2d); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG3, 0x0c); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG4, 0xa5); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG5, 0x00); + snd_soc_component_write(component, RK817_CODEC_DTOP_DIGEN_CLKE, 0x00); + + return 0; +} + +static struct rk817_reg_val_typ playback_power_up_list[] = { + {RK817_CODEC_AREF_RTCFG1, 0x40}, + {RK817_CODEC_DDAC_POPD_DACST, 0x02}, + {RK817_CODEC_DDAC_SR_LMT0, 0x02}, + /* APLL */ + {RK817_CODEC_APLL_CFG0, 0x04}, + {RK817_CODEC_APLL_CFG1, 0x58}, + {RK817_CODEC_APLL_CFG2, 0x2d}, + {RK817_CODEC_APLL_CFG3, 0x0c}, + {RK817_CODEC_APLL_CFG4, 0xa5}, + {RK817_CODEC_APLL_CFG5, 0x00}, + + {RK817_CODEC_DI2S_RXCMD_TSD, 0x00}, + {RK817_CODEC_DI2S_RSD, 0x00}, + {RK817_CODEC_DI2S_RXCR1, 0x00}, + {RK817_CODEC_DI2S_RXCMD_TSD, 0x20}, + {RK817_CODEC_DTOP_VUCTIME, 0xf4}, + {RK817_CODEC_DDAC_MUTE_MIXCTL, 0x00}, + + {RK817_CODEC_DDAC_VOLL, 0x0a}, + {RK817_CODEC_DDAC_VOLR, 0x0a}, +}; + +#define RK817_CODEC_PLAYBACK_POWER_UP_LIST_LEN \ + ARRAY_SIZE(playback_power_up_list) + +static struct rk817_reg_val_typ playback_power_down_list[] = { + {RK817_CODEC_DDAC_MUTE_MIXCTL, 0x01}, + {RK817_CODEC_ADAC_CFG1, 0x0f}, + /* HP */ + {RK817_CODEC_AHP_CFG0, 0xe0}, + {RK817_CODEC_AHP_CP, 0x09}, + /* SPK */ + {RK817_CODEC_ACLASSD_CFG1, 0x69}, +}; + +#define RK817_CODEC_PLAYBACK_POWER_DOWN_LIST_LEN \ + ARRAY_SIZE(playback_power_down_list) + +static struct rk817_reg_val_typ capture_power_up_list[] = { + {RK817_CODEC_AREF_RTCFG1, 0x40}, + {RK817_CODEC_DDAC_SR_LMT0, 0x02}, + {RK817_CODEC_DADC_SR_ACL0, 0x02}, + + {RK817_CODEC_APLL_CFG0, 0x04}, + {RK817_CODEC_APLL_CFG1, 0x58}, + {RK817_CODEC_APLL_CFG2, 0x2d}, + {RK817_CODEC_APLL_CFG3, 0x0c}, + {RK817_CODEC_APLL_CFG4, 0xa5}, + {RK817_CODEC_APLL_CFG5, 0x00}, + + {RK817_CODEC_DI2S_RSD, 0x00}, + {RK817_CODEC_DI2S_RXCR1, 0x00}, + {RK817_CODEC_DI2S_RXCMD_TSD, 0x20}, + {RK817_CODEC_DTOP_VUCTIME, 0xf4}, + + {RK817_CODEC_DDAC_MUTE_MIXCTL, 0x00}, + {RK817_CODEC_AADC_CFG0, 0x08}, + {RK817_CODEC_AMIC_CFG0, 0x0f}, + {RK817_CODEC_DI2S_TXCR3_TXCMD, 0x88}, + {RK817_CODEC_DDAC_POPD_DACST, 0x02}, + /* 0x29: -18db to 27db */ + {RK817_CODEC_DMIC_PGA_GAIN, 0x99}, +}; + +#define RK817_CODEC_CAPTURE_POWER_UP_LIST_LEN \ + ARRAY_SIZE(capture_power_up_list) + +static struct rk817_reg_val_typ capture_power_down_list[] = { + {RK817_CODEC_AADC_CFG0, 0xc8}, + {RK817_CODEC_AMIC_CFG0, 0x70}, +}; + +#define RK817_CODEC_CAPTURE_POWER_DOWN_LIST_LEN \ + ARRAY_SIZE(capture_power_down_list) + +static int rk817_codec_power_up(struct snd_soc_component *component, int type) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + int i; + + if (type & RK817_CODEC_PLAYBACK) { + snd_soc_component_update_bits(component, + RK817_CODEC_DTOP_DIGEN_CLKE, + DAC_DIG_CLK_MASK, DAC_DIG_CLK_EN); + for (i = 0; i < RK817_CODEC_PLAYBACK_POWER_UP_LIST_LEN; i++) { + snd_soc_component_write(component, + playback_power_up_list[i].reg, + playback_power_up_list[i].value); + } + } + + if (type & RK817_CODEC_CAPTURE) { + snd_soc_component_update_bits(component, + RK817_CODEC_DTOP_DIGEN_CLKE, + ADC_DIG_CLK_MASK, + ADC_DIG_CLK_EN); + for (i = 0; i < RK817_CODEC_CAPTURE_POWER_UP_LIST_LEN; i++) { + snd_soc_component_write(component, + capture_power_up_list[i].reg, + capture_power_up_list[i].value); + } + + if (rk817->mic_in_differential) + snd_soc_component_update_bits(component, + RK817_CODEC_AMIC_CFG0, + MIC_DIFF_MASK, MIC_DIFF_EN); + else + snd_soc_component_update_bits(component, + RK817_CODEC_AMIC_CFG0, + MIC_DIFF_MASK, + MIC_DIFF_DIS); + + } + + return 0; +} + +static int rk817_codec_power_down(struct snd_soc_component *component, int type) +{ + int i; + + /* mute output for pop noise */ + if (type & RK817_CODEC_PLAYBACK) { + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_ENABLE); + } + + if (type & RK817_CODEC_CAPTURE) { + for (i = 0; i < RK817_CODEC_CAPTURE_POWER_DOWN_LIST_LEN; i++) { + snd_soc_component_write(component, + capture_power_down_list[i].reg, + capture_power_down_list[i].value); + } + snd_soc_component_update_bits(component, RK817_CODEC_DTOP_DIGEN_CLKE, + ADC_DIG_CLK_MASK, ADC_DIG_CLK_DIS); + } + + if (type & RK817_CODEC_PLAYBACK) { + for (i = 0; i < RK817_CODEC_PLAYBACK_POWER_DOWN_LIST_LEN; i++) { + snd_soc_component_write(component, + playback_power_down_list[i].reg, + playback_power_down_list[i].value); + } + snd_soc_component_update_bits(component, + RK817_CODEC_DTOP_DIGEN_CLKE, + DAC_DIG_CLK_MASK, DAC_DIG_CLK_DIS); + } + + if (type == RK817_CODEC_ALL) { + for (i = 0; i < RK817_CODEC_PLAYBACK_POWER_DOWN_LIST_LEN; i++) { + snd_soc_component_write(component, + playback_power_down_list[i].reg, + playback_power_down_list[i].value); + } + for (i = 0; i < RK817_CODEC_CAPTURE_POWER_DOWN_LIST_LEN; i++) { + snd_soc_component_write(component, + capture_power_down_list[i].reg, + capture_power_down_list[i].value); + } + snd_soc_component_write(component, RK817_CODEC_DTOP_DIGEN_CLKE, 0x00); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG5, 0x01); + snd_soc_component_write(component, RK817_CODEC_AREF_RTCFG1, 0x06); + } + + return 0; +} + +/* For tiny alsa playback/capture path*/ +static const char * const rk817_playback_path_mode[] = { + "OFF", "SPK", "HP", "SPK_HP"}; + +static const char * const rk817_capture_path_mode[] = { + "MIC OFF", "MIC"}; + +static SOC_ENUM_SINGLE_DECL(rk817_playback_path_type, + 0, 0, rk817_playback_path_mode); + +static SOC_ENUM_SINGLE_DECL(rk817_capture_path_type, + 0, 0, rk817_capture_path_mode); + +static int rk817_playback_path_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = rk817->playback_path; + + return 0; +} + +static int rk817_playback_path_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + int pre_path; + + if (rk817->playback_path == ucontrol->value.integer.value[0]) { + return 0; + } + + pre_path = rk817->playback_path; + rk817->playback_path = ucontrol->value.integer.value[0]; + + if (rk817->playback_path != OFF) + clk_prepare_enable(rk817->mclk); + else + clk_disable_unprepare(rk817->mclk); + + switch (rk817->playback_path) { + case OFF: + if (pre_path != OFF && pre_path != HP_PATH) { + rk817_codec_power_down(component, RK817_CODEC_PLAYBACK); + if (rk817->capture_path == 0) + rk817_codec_power_down(component, RK817_CODEC_ALL); + } + break; + case SPK_PATH: + if (pre_path == OFF) + rk817_codec_power_up(component, RK817_CODEC_PLAYBACK); + + /* power on dac ibias/l/r */ + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, + PWD_DACBIAS_ON | PWD_DACD_ON | + PWD_DACL_DOWN | PWD_DACR_DOWN); + /* CLASS D mode, combine LR channels */ + snd_soc_component_write(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + 0x10); + /* CLASS D enable */ + snd_soc_component_write(component, + RK817_CODEC_ACLASSD_CFG1, + 0xa5); + /* restart CLASS D, OCPP/N */ + snd_soc_component_write(component, + RK817_CODEC_ACLASSD_CFG2, + 0xf7); + break; + case HP_PATH: + if (pre_path == OFF) + rk817_codec_power_up(component, RK817_CODEC_PLAYBACK); + /* HP_CP_EN , CP 2.3V */ + snd_soc_component_write(component, RK817_CODEC_AHP_CP, 0x11); + /* power on HP two stage opamp ,HP amplitude 0db */ + snd_soc_component_write(component, RK817_CODEC_AHP_CFG0, 0x80); + /* power on dac ibias/l/r */ + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, + PWD_DACBIAS_ON | PWD_DACD_DOWN | + PWD_DACL_ON | PWD_DACR_ON); + /* CLASS D mode disable, split LR channels */ + snd_soc_component_write(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + 0x00); + + break; + case SPK_HP: + if (pre_path == OFF) + rk817_codec_power_up(component, RK817_CODEC_PLAYBACK); + + /* HP_CP_EN , CP 2.3V */ + snd_soc_component_write(component, RK817_CODEC_AHP_CP, 0x11); + /* power on HP two stage opamp ,HP amplitude 0db */ + snd_soc_component_write(component, RK817_CODEC_AHP_CFG0, 0x80); + + /* power on dac ibias/l/r */ + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, + PWD_DACBIAS_ON | PWD_DACD_ON | + PWD_DACL_ON | PWD_DACR_ON); + + /* CLASS D mode, combine LR channels */ + snd_soc_component_write(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + 0x10); + /* CLASS D enable */ + snd_soc_component_write(component, + RK817_CODEC_ACLASSD_CFG1, + 0xa5); + /* restart CLASS D, OCPP/N */ + snd_soc_component_write(component, + RK817_CODEC_ACLASSD_CFG2, + 0xf7); + + break; + default: + return -EINVAL; + } + + return 0; +} + +static int rk817_capture_path_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "%s:capture_path %d\n", __func__, rk817->capture_path); + ucontrol->value.integer.value[0] = rk817->capture_path; + return 0; +} + +static int rk817_capture_path_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + int pre_path; + + if (rk817->capture_path == ucontrol->value.integer.value[0]) { + dev_dbg(component->dev, "%s:capture_path is not changed!\n", + __func__); + return 0; + } + + pre_path = rk817->capture_path; + rk817->capture_path = ucontrol->value.integer.value[0]; + + if (rk817->capture_path != MIC_OFF) + clk_prepare_enable(rk817->mclk); + else + clk_disable_unprepare(rk817->mclk); + + switch (rk817->capture_path) { + case MIC_OFF: + if (pre_path != MIC_OFF) + rk817_codec_power_down(component, RK817_CODEC_CAPTURE); + break; + case MIC: + if (pre_path == MIC_OFF) + rk817_codec_power_up(component, RK817_CODEC_CAPTURE); + + if (!rk817->mic_in_differential) { + snd_soc_component_write(component, + RK817_CODEC_DADC_VOLR, + 0xff); + snd_soc_component_update_bits(component, + RK817_CODEC_AADC_CFG0, + ADC_R_PWD_MASK, + ADC_R_PWD_EN); + snd_soc_component_update_bits(component, + RK817_CODEC_AMIC_CFG0, + PWD_PGA_R_MASK, + PWD_PGA_R_EN); + } + break; + default: + return -EINVAL; + } + + return 0; +} + +static struct snd_kcontrol_new rk817_snd_path_controls[] = { + SOC_ENUM_EXT("Playback Path", rk817_playback_path_type, + rk817_playback_path_get, rk817_playback_path_put), + + SOC_ENUM_EXT("Capture Path", rk817_capture_path_type, + rk817_capture_path_get, rk817_capture_path_put), +}; + +static const struct snd_soc_dapm_widget rk817_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static int rk817_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 rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + rk817->stereo_sysclk = freq; + + return 0; +} + +static int rk817_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + unsigned int i2s_mst = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + i2s_mst |= RK817_I2S_MODE_SLV; + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2s_mst |= RK817_I2S_MODE_MST; + break; + default: + dev_err(component->dev, "%s : set master mask failed!\n", __func__); + return -EINVAL; + } + + snd_soc_component_update_bits(component, RK817_CODEC_DI2S_CKM, + RK817_I2S_MODE_MASK, i2s_mst); + + return 0; +} + +static int rk817_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; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_16BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_16BITS); + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_24BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_24BITS); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int rk817_digital_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_component *component = dai->component; + + if (mute) + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_ENABLE); + else + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_DISABLE); + + return 0; +} + +#define RK817_PLAYBACK_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_CAPTURE_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops rk817_dai_ops = { + .hw_params = rk817_hw_params, + .set_fmt = rk817_set_dai_fmt, + .set_sysclk = rk817_set_dai_sysclk, + .mute_stream = rk817_digital_mute, + .no_capture_mute = 1, +}; + +static struct snd_soc_dai_driver rk817_dai[] = { + { + .name = "rk817-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = RK817_PLAYBACK_RATES, + .formats = RK817_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RK817_CAPTURE_RATES, + .formats = RK817_FORMATS, + }, + .ops = &rk817_dai_ops, + }, +}; + +static int rk817_suspend(struct snd_soc_component *component) +{ + rk817_codec_power_down(component, RK817_CODEC_ALL); + return 0; +} + +static int rk817_resume(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + rk817_codec_power_up(component, RK817_CODEC_ALL); + + switch (rk817->playback_path) { + case OFF: + break; + case SPK_PATH: + /* power on dac ibias/l/r */ + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, + PWD_DACBIAS_ON | PWD_DACD_ON | + PWD_DACL_ON | PWD_DACR_ON); + /* CLASS D mode */ + snd_soc_component_write(component, RK817_CODEC_DDAC_MUTE_MIXCTL, 0x10); + /* CLASS D enable */ + snd_soc_component_write(component, RK817_CODEC_ACLASSD_CFG1, 0xa5); + /* restart CLASS D, OCPP/N */ + snd_soc_component_write(component, RK817_CODEC_ACLASSD_CFG2, 0xc4); + break; + case HP_PATH: + /* HP_CP_EN , CP 2.3V */ + snd_soc_component_write(component, RK817_CODEC_AHP_CP, 0x11); + /* power on HP two stage opamp ,HP amplitude 0db */ + snd_soc_component_write(component, RK817_CODEC_AHP_CFG0, 0x80); + /* power on dac ibias/l/r */ + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, + PWD_DACBIAS_ON | PWD_DACD_DOWN | + PWD_DACL_ON | PWD_DACR_ON); + snd_soc_component_update_bits(component, RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_DISABLE); + break; + case SPK_HP: + /* HP_CP_EN , CP 2.3V */ + snd_soc_component_write(component, RK817_CODEC_AHP_CP, 0x11); + /* power on HP two stage opamp ,HP amplitude 0db */ + snd_soc_component_write(component, RK817_CODEC_AHP_CFG0, 0x80); + + /* power on dac ibias/l/r */ + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, + PWD_DACBIAS_ON | PWD_DACD_ON | + PWD_DACL_ON | PWD_DACR_ON); + + /* CLASS D mode */ + snd_soc_component_write(component, RK817_CODEC_DDAC_MUTE_MIXCTL, 0x10); + /* CLASS D enable */ + snd_soc_component_write(component, RK817_CODEC_ACLASSD_CFG1, 0xa5); + /* restart CLASS D, OCPP/N */ + snd_soc_component_write(component, RK817_CODEC_ACLASSD_CFG2, 0xc4); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int rk817_probe(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + if (!rk817) { + dev_err(component->dev, "%s : rk817 priv is NULL!\n", + __func__); + return -EINVAL; + } + snd_soc_component_init_regmap(component, rk817->regmap); + rk817->component = component; + rk817->playback_path = OFF; + rk817->capture_path = MIC_OFF; + + rk817_reset(component); + snd_soc_add_component_controls(component, rk817_snd_path_controls, + ARRAY_SIZE(rk817_snd_path_controls)); + + snd_soc_dapm_new_controls(dapm, rk817_dapm_widgets, + ARRAY_SIZE(rk817_dapm_widgets)); + return 0; +} + +/* power down chip */ +static void rk817_remove(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + if (!rk817) { + dev_err(component->dev, "%s : rk817 is NULL\n", __func__); + return; + } + + rk817_codec_power_down(component, RK817_CODEC_ALL); + snd_soc_component_exit_regmap(component); + mdelay(10); + +} + +static const struct snd_soc_component_driver soc_codec_dev_rk817 = { + .probe = rk817_probe, + .remove = rk817_remove, + .suspend = rk817_suspend, + .resume = rk817_resume, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, + .controls = rk817_dac_controls, + .num_controls = ARRAY_SIZE(rk817_dac_controls), +}; + +static int rk817_codec_parse_dt_property(struct device *dev, + struct rk817_codec_priv *rk817) +{ + struct device_node *node = dev->parent->of_node; + + if (!node) { + dev_err(dev, "%s() dev->parent->of_node is NULL\n", + __func__); + return -ENODEV; + } + + node = of_get_child_by_name(dev->parent->of_node, "codec"); + if (!node) { + dev_err(dev, "%s() Can not get child: codec\n", + __func__); + return -ENODEV; + } + + rk817->mic_in_differential = + of_property_read_bool(node, "mic-in-differential"); + + return 0; +} + +static const struct regmap_config rk817_codec_regmap_config = { + .name = "rk817-codec", + .reg_bits = 8, + .val_bits = 8, + .reg_stride = 1, + .max_register = 0x4f, + .cache_type = REGCACHE_NONE, + .volatile_reg = rk817_volatile_register, + .writeable_reg = rk817_codec_register, + .readable_reg = rk817_codec_register, + .reg_defaults = rk817_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rk817_reg_defaults), +}; + +static int rk817_platform_probe(struct platform_device *pdev) +{ + struct rk808 *rk817 = dev_get_drvdata(pdev->dev.parent); + struct rk817_codec_priv *rk817_codec_data; + int ret; + + if (!rk817) { + dev_err(&pdev->dev, "%s : rk817 is NULL\n", __func__); + return -EINVAL; + } + + rk817_codec_data = devm_kzalloc(&pdev->dev, + sizeof(struct rk817_codec_priv), + GFP_KERNEL); + if (!rk817_codec_data) + return -ENOMEM; + + platform_set_drvdata(pdev, rk817_codec_data); + + ret = rk817_codec_parse_dt_property(&pdev->dev, rk817_codec_data); + if (ret < 0) { + dev_err(&pdev->dev, "%s() parse device tree property error %d\n", + __func__, ret); + goto err_; + } + + rk817_codec_data->regmap = devm_regmap_init_i2c(rk817->i2c, + &rk817_codec_regmap_config); + if (IS_ERR(rk817_codec_data->regmap)) { + ret = PTR_ERR(rk817_codec_data->regmap); + dev_err(&pdev->dev, "failed to allocate register map: %d\n", + ret); + goto err_; + } + + rk817_codec_data->mclk = devm_clk_get(&pdev->dev, "mclk"); + if (IS_ERR(rk817_codec_data->mclk)) { + dev_err(&pdev->dev, "Unable to get mclk\n"); + ret = -ENXIO; + goto err_; + } + + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_dev_rk817, + rk817_dai, ARRAY_SIZE(rk817_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "%s() register codec error %d\n", + __func__, ret); + goto err_; + } + + return 0; +err_: + + return ret; +} + +static const struct of_device_id rk817_codec_dt_ids[] = { + { .compatible = "rockchip,rk817-codec" }, + {}, +}; +MODULE_DEVICE_TABLE(of, rk817_codec_dt_ids); + +static struct platform_driver rk817_codec_driver = { + .driver = { + .name = "rk817-codec", + .of_match_table = rk817_codec_dt_ids, + }, + .probe = rk817_platform_probe, +}; + +module_platform_driver(rk817_codec_driver); + +MODULE_DESCRIPTION("ASoC RK817 codec driver"); +MODULE_AUTHOR("binyuan <kevan.lan(a)rock-chips.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rk817_codec.h b/sound/soc/codecs/rk817_codec.h new file mode 100644 index 000000000000..5350f7100d67 --- /dev/null +++ b/sound/soc/codecs/rk817_codec.h @@ -0,0 +1,142 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rk817 ALSA SoC Audio driver +// +// Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + +#ifndef __RK817_CODEC_H__ +#define __RK817_CODEC_H__ + +/* codec register */ +#define RK817_CODEC_BASE 0x0000 + +#define RK817_CODEC_DTOP_VUCTL (RK817_CODEC_BASE + 0x12) +#define RK817_CODEC_DTOP_VUCTIME (RK817_CODEC_BASE + 0x13) +#define RK817_CODEC_DTOP_LPT_SRST (RK817_CODEC_BASE + 0x14) +#define RK817_CODEC_DTOP_DIGEN_CLKE (RK817_CODEC_BASE + 0x15) +#define RK817_CODEC_AREF_RTCFG0 (RK817_CODEC_BASE + 0x16) +#define RK817_CODEC_AREF_RTCFG1 (RK817_CODEC_BASE + 0x17) +#define RK817_CODEC_AADC_CFG0 (RK817_CODEC_BASE + 0x18) +#define RK817_CODEC_AADC_CFG1 (RK817_CODEC_BASE + 0x19) +#define RK817_CODEC_DADC_VOLL (RK817_CODEC_BASE + 0x1a) +#define RK817_CODEC_DADC_VOLR (RK817_CODEC_BASE + 0x1b) +#define RK817_CODEC_DADC_SR_ACL0 (RK817_CODEC_BASE + 0x1e) +#define RK817_CODEC_DADC_ALC1 (RK817_CODEC_BASE + 0x1f) +#define RK817_CODEC_DADC_ALC2 (RK817_CODEC_BASE + 0x20) +#define RK817_CODEC_DADC_NG (RK817_CODEC_BASE + 0x21) +#define RK817_CODEC_DADC_HPF (RK817_CODEC_BASE + 0x22) +#define RK817_CODEC_DADC_RVOLL (RK817_CODEC_BASE + 0x23) +#define RK817_CODEC_DADC_RVOLR (RK817_CODEC_BASE + 0x24) +#define RK817_CODEC_AMIC_CFG0 (RK817_CODEC_BASE + 0x27) +#define RK817_CODEC_AMIC_CFG1 (RK817_CODEC_BASE + 0x28) +#define RK817_CODEC_DMIC_PGA_GAIN (RK817_CODEC_BASE + 0x29) +#define RK817_CODEC_DMIC_LMT1 (RK817_CODEC_BASE + 0x2a) +#define RK817_CODEC_DMIC_LMT2 (RK817_CODEC_BASE + 0x2b) +#define RK817_CODEC_DMIC_NG1 (RK817_CODEC_BASE + 0x2c) +#define RK817_CODEC_DMIC_NG2 (RK817_CODEC_BASE + 0x2d) +#define RK817_CODEC_ADAC_CFG0 (RK817_CODEC_BASE + 0x2e) +#define RK817_CODEC_ADAC_CFG1 (RK817_CODEC_BASE + 0x2f) +#define RK817_CODEC_DDAC_POPD_DACST (RK817_CODEC_BASE + 0x30) +#define RK817_CODEC_DDAC_VOLL (RK817_CODEC_BASE + 0x31) +#define RK817_CODEC_DDAC_VOLR (RK817_CODEC_BASE + 0x32) +#define RK817_CODEC_DDAC_SR_LMT0 (RK817_CODEC_BASE + 0x35) +#define RK817_CODEC_DDAC_LMT1 (RK817_CODEC_BASE + 0x36) +#define RK817_CODEC_DDAC_LMT2 (RK817_CODEC_BASE + 0x37) +#define RK817_CODEC_DDAC_MUTE_MIXCTL (RK817_CODEC_BASE + 0x38) +#define RK817_CODEC_DDAC_RVOLL (RK817_CODEC_BASE + 0x39) +#define RK817_CODEC_DDAC_RVOLR (RK817_CODEC_BASE + 0x3a) +#define RK817_CODEC_AHP_ANTI0 (RK817_CODEC_BASE + 0x3b) +#define RK817_CODEC_AHP_ANTI1 (RK817_CODEC_BASE + 0x3c) +#define RK817_CODEC_AHP_CFG0 (RK817_CODEC_BASE + 0x3d) +#define RK817_CODEC_AHP_CFG1 (RK817_CODEC_BASE + 0x3e) +#define RK817_CODEC_AHP_CP (RK817_CODEC_BASE + 0x3f) +#define RK817_CODEC_ACLASSD_CFG1 (RK817_CODEC_BASE + 0x40) +#define RK817_CODEC_ACLASSD_CFG2 (RK817_CODEC_BASE + 0x41) +#define RK817_CODEC_APLL_CFG0 (RK817_CODEC_BASE + 0x42) +#define RK817_CODEC_APLL_CFG1 (RK817_CODEC_BASE + 0x43) +#define RK817_CODEC_APLL_CFG2 (RK817_CODEC_BASE + 0x44) +#define RK817_CODEC_APLL_CFG3 (RK817_CODEC_BASE + 0x45) +#define RK817_CODEC_APLL_CFG4 (RK817_CODEC_BASE + 0x46) +#define RK817_CODEC_APLL_CFG5 (RK817_CODEC_BASE + 0x47) +#define RK817_CODEC_DI2S_CKM (RK817_CODEC_BASE + 0x48) +#define RK817_CODEC_DI2S_RSD (RK817_CODEC_BASE + 0x49) +#define RK817_CODEC_DI2S_RXCR1 (RK817_CODEC_BASE + 0x4a) +#define RK817_CODEC_DI2S_RXCR2 (RK817_CODEC_BASE + 0x4b) +#define RK817_CODEC_DI2S_RXCMD_TSD (RK817_CODEC_BASE + 0x4c) +#define RK817_CODEC_DI2S_TXCR1 (RK817_CODEC_BASE + 0x4d) +#define RK817_CODEC_DI2S_TXCR2 (RK817_CODEC_BASE + 0x4e) +#define RK817_CODEC_DI2S_TXCR3_TXCMD (RK817_CODEC_BASE + 0x4f) + +/* RK817_CODEC_DTOP_DIGEN_CLKE */ +#define ADC_DIG_CLK_MASK (0xf << 4) +#define ADC_DIG_CLK_DIS (0x0 << 4) +#define ADC_DIG_CLK_EN (0xf << 4) + +#define DAC_DIG_CLK_MASK (0xf << 0) +#define DAC_DIG_CLK_DIS (0x0 << 0) +#define DAC_DIG_CLK_EN (0xf << 0) + +#define RK817_I2S_MODE_MASK (0x1 << 0) +#define RK817_I2S_MODE_MST (0x1 << 0) +#define RK817_I2S_MODE_SLV (0x0 << 0) + +/* RK817_CODEC_DDAC_MUTE_MIXCTL */ +#define DACMT_MASK (0x1 << 0) +#define DACMT_ENABLE (0x1 << 0) +#define DACMT_DISABLE (0x0 << 0) + +/* RK817_CODEC_DI2S_RXCR2 */ +#define VDW_RX_24BITS (0x17) +#define VDW_RX_16BITS (0x0f) + +/* RK817_CODEC_DI2S_TXCR2 */ +#define VDW_TX_24BITS (0x17) +#define VDW_TX_16BITS (0x0f) + +/* RK817_CODEC_ADAC_CFG1 */ +#define PWD_DACBIAS_MASK (0x1 << 3) +#define PWD_DACBIAS_DOWN (0x1 << 3) +#define PWD_DACBIAS_ON (0x0 << 3) +#define PWD_DACD_MASK (0x1 << 2) +#define PWD_DACD_DOWN (0x1 << 2) +#define PWD_DACD_ON (0x0 << 2) +#define PWD_DACL_MASK (0x1 << 1) +#define PWD_DACL_DOWN (0x1 << 1) +#define PWD_DACL_ON (0x0 << 1) +#define PWD_DACR_MASK (0x1 << 0) +#define PWD_DACR_DOWN (0x1 << 0) +#define PWD_DACR_ON (0x0 << 0) + +/* RK817_CODEC_AADC_CFG0 */ +#define ADC_R_PWD_MASK (0x1 << 6) +#define ADC_R_PWD_DIS (0x0 << 6) +#define ADC_R_PWD_EN (0x1 << 6) + +/* RK817_CODEC_AMIC_CFG0 */ +#define MIC_DIFF_MASK (0x1 << 7) +#define MIC_DIFF_DIS (0x0 << 7) +#define MIC_DIFF_EN (0x1 << 7) + +/* RK817_CODEC_AMIC_CFG0 */ +#define PWD_PGA_R_MASK (0x1 << 4) +#define PWD_PGA_R_DIS (0x0 << 4) +#define PWD_PGA_R_EN (0x1 << 4) + +enum { + OFF, + SPK_PATH, + HP_PATH, + SPK_HP, +}; + +enum { + MIC_OFF, + MIC, +}; + +struct rk817_reg_val_typ { + unsigned int reg; + unsigned int value; +}; + +#endif /* __RK817_CODEC_H__ */ -- 2.25.1

2 1

[PATCH v4 0/6] Add audio driver base on rpmsg on i.MX platform
by Shengjiu Wang 11 Mar '21

11 Mar '21

On Asymmetric multiprocessor, there is Cortex-A core and Cortex-M core, Linux is running on A core, RTOS is running on M core. The audio hardware device can be controlled by Cortex-M device, So audio playback/capture can be handled by M core. Rpmsg is the interface for sending and receiving msg to and from M core, that we can create a virtual sound on Cortex-A core side. A core will tell the Cortex-M core sound format/rate/channel, where is the data buffer, what is the period size, when to start, when to stop and when suspend or resume happen, each of this behavior there is defined rpmsg command. Especially we designed the low power audio case, that is to allocate a large buffer and fill the data, then Cortex-A core can go to sleep mode, Cortex-M core continue to play the sound, when the buffer is consumed, Cortex-M core will trigger the Cortex-A core to wakeup to fill data. changes in v4: - remove the sound card node, merge the property to cpu dai node according to Rob's comments. - sound card device will be registered by cpu dai driver. - Fix do_div issue reported by kernel test robot changes in v3: - add local refcount for clk enablement in hw_params() - update the document according Rob's comments changes in v2: - update codes and comments according to Mark's comments Shengjiu Wang (6): ASoC: soc-component: Add snd_soc_pcm_component_ack ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg ASoC: dt-bindings: fsl_rpmsg: Add binding doc for rpmsg cpu dai driver ASoC: imx-audio-rpmsg: Add rpmsg_driver for audio channel ASoC: imx-pcm-rpmsg: Add platform driver for audio base on rpmsg ASoC: imx-rpmsg: Add machine driver for audio base on rpmsg .../devicetree/bindings/sound/fsl,rpmsg.yaml | 118 +++ include/sound/soc-component.h | 3 + sound/soc/fsl/Kconfig | 28 + sound/soc/fsl/Makefile | 6 + sound/soc/fsl/fsl_rpmsg.c | 283 ++++++ sound/soc/fsl/fsl_rpmsg.h | 42 + sound/soc/fsl/imx-audio-rpmsg.c | 151 +++ sound/soc/fsl/imx-pcm-rpmsg.c | 919 ++++++++++++++++++ sound/soc/fsl/imx-pcm-rpmsg.h | 512 ++++++++++ sound/soc/fsl/imx-rpmsg.c | 127 +++ sound/soc/soc-component.c | 14 + sound/soc/soc-pcm.c | 2 + 12 files changed, 2205 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml create mode 100644 sound/soc/fsl/fsl_rpmsg.c create mode 100644 sound/soc/fsl/fsl_rpmsg.h create mode 100644 sound/soc/fsl/imx-audio-rpmsg.c create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.c create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.h create mode 100644 sound/soc/fsl/imx-rpmsg.c -- 2.27.0

4 11

[PATCH] ALSA: ppc: keywest: remove outdated comment
by Wolfram Sang 11 Mar '21

11 Mar '21

The I2C attach_adapter callback is gone. Remove this reference. Signed-off-by: Wolfram Sang <wsa(a)kernel.org> --- sound/ppc/keywest.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index a6c1905039de..a8915100d6bb 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -13,12 +13,7 @@ #include <sound/core.h> #include "pmac.h" -/* - * we have to keep a static variable here since i2c attach_adapter - * callback cannot pass a private data. - */ static struct pmac_keywest *keywest_ctx; - static bool keywest_probed; static int keywest_probe(struct i2c_client *client, -- 2.30.0

2 1

[alsa-utils][PATCH 00/14] axfer: reduce test time
by Takashi Sakamoto 11 Mar '21

11 Mar '21

Hi, The axfer in alsa-utils has test programs for internal modules and overall time to execute them takes much. The issue is filing in github repository[1]. It comes from three causes: 1. file I/O operations are done in file system for actual storage 2. Some cases operate much audio data frames (over 4MB, approx.) 3. much test iteration count (so simple...) This patchset uses shm by memfd_create(2) for the cause 1. In addition, the number of maximum audio data frame to test is reduced as well as the number of samples per frame. I got benchmark with berow machine environment. * AMD Ryzen 5 2400G * memory total 30823852 KB * linux-image-5.8.0-44-generic on Ubuntu 20.04 amd64 * ext4 on SATA SSD As a result: |container| mapper | | test | test | ================== | ======= | ======= | Current | 112 min | 5 min | +shm | 58 min | 50 sec | +maximum reduction | 38 min | - | +iter reduction | 4 min | - | In my opinion, the issue comes from package build server in each distribution. 5 min for test execution is not so worse time. Finally, test programs run on shm and commit c3f2344b7209 is reverted[1] since it's useless now. (I have free time at last from reverse-engineering work to analyze protocols of RME Fireface series[1], and start to solve stacked issues.) [1] https://github.com/alsa-project/snd-firewire-ctl-services/pull/37 [2] https://github.com/alsa-project/alsa-utils/issues/19 [3] https://github.com/alsa-project/alsa-utils/commit/c3f2344b7209 Regards Takashi Sakamoto (14): axfer: minor code arrangement for container module in a point of nonblocking flag axfer: minor code arrangement in a point of stdio detection axfer: minor code arrangement in a point of opened file descriptor axfer: minor code arrangement to allocate containers axfer: open file descriptor outside of container module axfer: maintain lifetime of file descriptor outside of container module autotools: preparation to use memfd_create(2) axfer: test: minor code arrangement to use the same file descriptor for container-test axfer: test: use memfd_create() for container-test axfer: test: minor code arrangement to use the same file descriptor for mappter-test axfer: test: use memfd_create() for mapper-test axfer: test: reduce test case for maximum number of frame count axfer: test: reduce test case for maximum number of samples per frame Revert "axfer: test - add run-test-in-tmpdir.sh script" axfer/container.c | 57 +++++++++++----------------- axfer/container.h | 10 ++--- axfer/subcmd-transfer.c | 64 ++++++++++++++++++++++++++------ axfer/test/container-test.c | 53 +++++++++++++++++++------- axfer/test/mapper-test.c | 55 ++++++++++++++++++++++----- axfer/test/run-test-in-tmpdir.sh | 19 ---------- configure.ac | 5 +++ 7 files changed, 167 insertions(+), 96 deletions(-) delete mode 100755 axfer/test/run-test-in-tmpdir.sh -- 2.27.0

2 17

Support for Tascam US-366 card
by GitHub issues - opened 11 Mar '21

11 Mar '21

alsa-project/alsa-tools issue #5 was opened from sxiii: Greetings! It would be really nice to have US-366 card support. I bet this card can be reversed just as the others was. Here's the link: https://www.tascam.eu/en/downloads/us-366 I am available to test stuff if somebody can give me a hand with this task. Thanks Den Issue URL : https://github.com/alsa-project/alsa-tools/issues/5 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH V2] ASoC: soc-core: Prevent warning if no DMI table is present
by Jon Hunter 10 Mar '21

10 Mar '21

Many systems do not use ACPI and hence do not provide a DMI table. On non-ACPI systems a warning, such as the following, is printed on boot. WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name! The variable 'dmi_available' is not exported and so currently cannot be used by kernel modules without adding an accessor. However, it is possible to use the function is_acpi_device_node() to determine if the sound card is an ACPI device and hence indicate if we expect a DMI table to be present. Therefore, call is_acpi_device_node() to see if we are using ACPI and only parse the DMI table if we are booting with ACPI. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- Changes since V1: - Use is_acpi_device_node() to determine if we expect the DMI table to be present. sound/soc/soc-core.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f6d4e99b590c..0cffc9527e28 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -31,6 +31,7 @@ #include <linux/of.h> #include <linux/of_graph.h> #include <linux/dmi.h> +#include <linux/acpi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -1573,6 +1574,9 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) if (card->long_name) return 0; /* long name already set by driver or from DMI */ + if (!is_acpi_device_node(card->dev->fwnode)) + return 0; + /* make up dmi long name as: vendor-product-version-board */ vendor = dmi_get_system_info(DMI_BOARD_VENDOR); if (!vendor || !is_dmi_valid(vendor)) { -- 2.25.1

4 14

[PATCH] sound: Remove redundant steps
by zuoqilin1＠163.com 10 Mar '21

10 Mar '21

From: zuoqilin <zuoqilin(a)yulong.com> If kzalloc fail,not need free it,so just return -ENOMEM when kzalloc fail. Signed-off-by: zuoqilin <zuoqilin(a)yulong.com> --- sound/core/oss/mixer_oss.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index bec9283..eec61dc 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -800,8 +800,7 @@ static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (uinfo == NULL || uctl == NULL) { - err = -ENOMEM; - goto __free_only; + return -ENOMEM; } down_read(&card->controls_rwsem); kctl = snd_mixer_oss_test_id(mixer, "Capture Source", 0); @@ -851,8 +850,7 @@ static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (uinfo == NULL || uctl == NULL) { - err = -ENOMEM; - goto __free_only; + return -ENOMEM; } down_read(&card->controls_rwsem); kctl = snd_mixer_oss_test_id(mixer, "Capture Source", 0); -- 1.9.1

4 3

[PATCH] ASoC: rt1019: add rt1019 amplifier driver
by jack.yu＠realtek.com 10 Mar '21

10 Mar '21

From: Jack Yu <jack.yu(a)realtek.com> This is initial amplifier driver for rt1019. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- .../devicetree/bindings/sound/rt1019.txt | 16 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt1019.c | 940 ++++++++++++++++++ sound/soc/codecs/rt1019.h | 320 ++++++ 5 files changed, 1285 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt1019.txt create mode 100644 sound/soc/codecs/rt1019.c create mode 100644 sound/soc/codecs/rt1019.h diff --git a/Documentation/devicetree/bindings/sound/rt1019.txt b/Documentation/devicetree/bindings/sound/rt1019.txt new file mode 100644 index 000000000000..fdefd4ff1d3d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt1019.txt @@ -0,0 +1,16 @@ +RT1019 Mono Class-D Audio Amplifier + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt1019". + +- reg : The I2C address of the device. + +Example: + +rt1019: codec@28 { + compatible = "realtek,rt1019"; + reg = <0x28>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6ce74c99a305..4ab34bca71aa 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -161,6 +161,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT1011 imply SND_SOC_RT1015 imply SND_SOC_RT1015P + imply SND_SOC_RT1019 imply SND_SOC_RT1305 imply SND_SOC_RT1308 imply SND_SOC_RT5514 @@ -1079,6 +1080,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT1011=y default y if SND_SOC_RT1015=y default y if SND_SOC_RT1015P=y + default y if SND_SOC_RT1019=y default y if SND_SOC_RT1305=y default y if SND_SOC_RT1308=y default m if SND_SOC_RT5514=m @@ -1097,6 +1099,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT1011=m default m if SND_SOC_RT1015=m default m if SND_SOC_RT1015P=m + default m if SND_SOC_RT1019=m default m if SND_SOC_RT1305=m default m if SND_SOC_RT1308=m @@ -1133,6 +1136,10 @@ config SND_SOC_RT1015P tristate depends on GPIOLIB +config SND_SOC_RT1019 + tristate + depends on I2C + config SND_SOC_RT1305 tristate depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dcc2f757bb82..edff5c5b92d3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -170,6 +170,7 @@ snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1011-objs := rt1011.o snd-soc-rt1015-objs := rt1015.o snd-soc-rt1015p-objs := rt1015p.o +snd-soc-rt1019-objs := rt1019.o snd-soc-rt1305-objs := rt1305.o snd-soc-rt1308-objs := rt1308.o snd-soc-rt1308-sdw-objs := rt1308-sdw.o @@ -487,6 +488,7 @@ obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o obj-$(CONFIG_SND_SOC_RT1015) += snd-soc-rt1015.o obj-$(CONFIG_SND_SOC_RT1015P) += snd-soc-rt1015p.o +obj-$(CONFIG_SND_SOC_RT1019) += snd-soc-rt1019.o obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o obj-$(CONFIG_SND_SOC_RT1308) += snd-soc-rt1308.o obj-$(CONFIG_SND_SOC_RT1308_SDW) += snd-soc-rt1308-sdw.o diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c new file mode 100644 index 000000000000..fb275686a00f --- /dev/null +++ b/sound/soc/codecs/rt1019.c @@ -0,0 +1,940 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt1019.c -- RT1019 ALSA SoC audio amplifier driver +// Author: Jack Yu <jack.yu(a)realtek.com> +// +// Copyright(c) 2021 Realtek Semiconductor Corp. +// +// + +#include <linux/acpi.h> +#include <linux/fs.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rl6231.h" +#include "rt1019.h" + +static const struct reg_default rt1019_reg[] = { + { 0x0000, 0x00 }, + { 0x0002, 0x55 }, + { 0x0003, 0x55 }, + { 0x0005, 0x54 }, + { 0x0006, 0x05 }, + { 0x0007, 0x01 }, + { 0x0008, 0x70 }, + { 0x0009, 0x28 }, + { 0x000a, 0x7f }, + { 0x0011, 0x04 }, + { 0x0013, 0x00 }, + { 0x0015, 0x00 }, + { 0x0017, 0x00 }, + { 0x0019, 0x30 }, + { 0x001b, 0x01 }, + { 0x001d, 0x18 }, + { 0x001f, 0x29 }, + { 0x0021, 0x09 }, + { 0x0023, 0x02 }, + { 0x0025, 0x00 }, + { 0x0026, 0x00 }, + { 0x0028, 0x03 }, + { 0x0053, 0x00 }, + { 0x0055, 0x00 }, + { 0x0056, 0x00 }, + { 0x0057, 0x00 }, + { 0x0058, 0x00 }, + { 0x005a, 0x00 }, + { 0x005c, 0x00 }, + { 0x005d, 0x00 }, + { 0x005e, 0x10 }, + { 0x005f, 0xec }, + { 0x0061, 0x10 }, + { 0x0062, 0x19 }, + { 0x0064, 0x00 }, + { 0x0066, 0x08 }, + { 0x0068, 0x00 }, + { 0x006a, 0x00 }, + { 0x006c, 0x00 }, + { 0x006e, 0x00 }, + { 0x0100, 0x80 }, + { 0x0100, 0x51 }, + { 0x0102, 0x23 }, + { 0x0102, 0x0f }, + { 0x0104, 0x6c }, + { 0x0105, 0xec }, + { 0x0106, 0x00 }, + { 0x0107, 0x00 }, + { 0x0108, 0x00 }, + { 0x0200, 0x40 }, + { 0x0201, 0x00 }, + { 0x0202, 0x00 }, + { 0x0203, 0x00 }, + { 0x0301, 0x02 }, + { 0x0302, 0xaa }, + { 0x0303, 0x2a }, + { 0x0304, 0x6a }, + { 0x0306, 0xb0 }, + { 0x0308, 0x48 }, + { 0x030a, 0x0a }, + { 0x030b, 0x4b }, + { 0x030d, 0x7d }, + { 0x030e, 0xef }, + { 0x030f, 0x5a }, + { 0x0311, 0x00 }, + { 0x0312, 0x3e }, + { 0x0313, 0x86 }, + { 0x0315, 0xa8 }, + { 0x0318, 0x1b }, + { 0x031a, 0x3d }, + { 0x031c, 0x40 }, + { 0x031d, 0x40 }, + { 0x031e, 0x30 }, + { 0x031f, 0xbb }, + { 0x0320, 0xa5 }, + { 0x0321, 0xa5 }, + { 0x0323, 0x5a }, + { 0x0324, 0xaa }, + { 0x0325, 0x80 }, + { 0x0326, 0xaa }, + { 0x0327, 0x66 }, + { 0x0328, 0x94 }, + { 0x0329, 0x00 }, + { 0x0330, 0x00 }, + { 0x0331, 0x30 }, + { 0x0332, 0x05 }, + { 0x0400, 0x03 }, + { 0x0401, 0x02 }, + { 0x0402, 0x01 }, + { 0x0403, 0x23 }, + { 0x0404, 0x45 }, + { 0x0405, 0x67 }, + { 0x0500, 0x80 }, + { 0x0501, 0x00 }, + { 0x0502, 0x00 }, + { 0x0503, 0x00 }, + { 0x0504, 0xff }, + { 0x0505, 0x24 }, + { 0x0600, 0x75 }, + { 0x0601, 0x41 }, + { 0x0602, 0x60 }, + { 0x0603, 0x7f }, + { 0x0604, 0x65 }, + { 0x0605, 0x43 }, + { 0x0606, 0x00 }, + { 0x0607, 0x00 }, + { 0x0608, 0x00 }, + { 0x0609, 0x00 }, + { 0x060a, 0x00 }, + { 0x060b, 0x00 }, + { 0x060c, 0x00 }, + { 0x060d, 0x00 }, + { 0x060e, 0x00 }, + { 0x060f, 0x00 }, + { 0x0700, 0x15 }, + { 0x0701, 0xc8 }, + { 0x0704, 0x02 }, + { 0x0705, 0x00 }, + { 0x0706, 0x00 }, + { 0x0707, 0x80 }, + { 0x0708, 0x08 }, + { 0x0709, 0x00 }, + { 0x0800, 0x00 }, + { 0x0801, 0x00 }, + { 0x0802, 0x09 }, + { 0x0803, 0x00 }, + { 0x0900, 0x87 }, + { 0x0a01, 0x99 }, + { 0x0a02, 0x40 }, + { 0x0a03, 0x10 }, + { 0x0b00, 0x50 }, + { 0x0b01, 0xc3 }, + { 0x0c00, 0x84 }, + { 0x0c01, 0x00 }, + { 0x0c02, 0xbb }, + { 0x0c03, 0x80 }, + { 0x0c04, 0x10 }, + { 0x0c05, 0x30 }, + { 0x0c06, 0x00 }, + { 0x0d00, 0x80 }, + { 0x0d01, 0xbb }, + { 0x0d02, 0x80 }, + { 0x0d03, 0x00 }, + { 0x0d04, 0x00 }, + { 0x0d05, 0x00 }, + { 0x0e00, 0x80 }, + { 0x0e01, 0xbb }, + { 0x0e02, 0x80 }, + { 0x0e03, 0x00 }, + { 0x0e04, 0x10 }, + { 0x0e05, 0x30 }, + { 0x0f00, 0x80 }, + { 0x0f01, 0xbb }, + { 0x0f02, 0x80 }, + { 0x0f03, 0x00 }, + { 0x0f04, 0x10 }, + { 0x0f05, 0x30 }, + { 0x0f06, 0x88 }, + { 0x0f07, 0x88 }, + { 0x0f08, 0x00 }, + { 0x0f09, 0x00 }, +}; + +static bool rt1019_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT1019_PWR_STRP_1: + case RT1019_PWR_STRP_2: + case RT1019_SIL_DET_GAT: + case RT1019_PHASE_SYNC: + case RT1019_STAT_MACH_2: + case RT1019_FS_DET_1: + case RT1019_FS_DET_2: + case RT1019_FS_DET_3: + case RT1019_FS_DET_4: + case RT1019_FS_DET_5: + case RT1019_FS_DET_6: + case RT1019_FS_DET_7: + case RT1019_ANA_READ: + case RT1019_VER_ID: + case RT1019_CUSTOM_ID: + case RT1019_VEND_ID_1: + case RT1019_VEND_ID_2: + case RT1019_DEV_ID_1: + case RT1019_DEV_ID_2: + case RT1019_CAL_TOP_3: + case RT1019_CAL_TOP_7: + case RT1019_CAL_TOP_17: + case RT1019_CAL_TOP_18: + case RT1019_CAL_TOP_19: + case RT1019_CAL_TOP_20: + case RT1019_CAL_TOP_21: + case RT1019_CAL_TOP_22: + case RT1019_MDRE_CTRL_2: + case RT1019_MDRE_CTRL_3: + case RT1019_MDRE_CTRL_4: + case RT1019_SIL_DET_2: + case RT1019_PWM_DC_DET_1: + case RT1019_PMC_8: + case RT1019_PMC_9: + case RT1019_SPKDRC_7: + case RT1019_HALF_FREQ_7: + case RT1019_CUR_CTRL_11: + case RT1019_CUR_CTRL_12: + case RT1019_CUR_CTRL_13: + return true; + + default: + return false; + } +} + +static bool rt1019_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT1019_RESET: + case RT1019_PAD_DRV_1: + case RT1019_PAD_DRV_2: + case RT1019_PAD_PULL_1: + case RT1019_PAD_PULL_2: + case RT1019_PAD_PULL_3: + case RT1019_I2C_CTRL_1: + case RT1019_I2C_CTRL_2: + case RT1019_I2C_CTRL_3: + case RT1019_IDS_CTRL: + case RT1019_ASEL_CTRL: + case RT1019_PLL_RESET: + case RT1019_PWR_STRP_1: + case RT1019_PWR_STRP_2: + case RT1019_BEEP_TONE: + case RT1019_SIL_DET_GAT: + case RT1019_CLASSD_TIME: + case RT1019_CLASSD_OCP: + case RT1019_PHASE_SYNC: + case RT1019_STAT_MACH_1: + case RT1019_STAT_MACH_2: + case RT1019_EFF_CTRL: + case RT1019_FS_DET_1: + case RT1019_FS_DET_2: + case RT1019_FS_DET_3: + case RT1019_FS_DET_4: + case RT1019_FS_DET_5: + case RT1019_FS_DET_6: + case RT1019_FS_DET_7: + case RT1019_ANA_CTRL: + case RT1019_DUMMY_A: + case RT1019_DUMMY_B: + case RT1019_DUMMY_C: + case RT1019_DUMMY_D: + case RT1019_ANA_READ: + case RT1019_VER_ID: + case RT1019_CUSTOM_ID: + case RT1019_VEND_ID_1: + case RT1019_VEND_ID_2: + case RT1019_DEV_ID_1: + case RT1019_DEV_ID_2: + case RT1019_TEST_PAD: + case RT1019_SDB_CTRL: + case RT1019_TEST_CTRL_1: + case RT1019_TEST_CTRL_2: + case RT1019_TEST_CTRL_3: + case RT1019_SCAN_MODE: + case RT1019_CLK_TREE_1: + case RT1019_CLK_TREE_2: + case RT1019_CLK_TREE_3: + case RT1019_CLK_TREE_4: + case RT1019_CLK_TREE_5: + case RT1019_CLK_TREE_6: + case RT1019_CLK_TREE_7: + case RT1019_CLK_TREE_8: + case RT1019_CLK_TREE_9: + case RT1019_ASRC_1: + case RT1019_ASRC_2: + case RT1019_ASRC_3: + case RT1019_ASRC_4: + case RT1019_SYS_CLK: + case RT1019_BIAS_CUR_1: + case RT1019_BIAS_CUR_2: + case RT1019_BIAS_CUR_3: + case RT1019_BIAS_CUR_4: + case RT1019_CHOP_CLK_DAC: + case RT1019_CHOP_CLK_ADC: + case RT1019_LDO_CTRL_1: + case RT1019_LDO_CTRL_2: + case RT1019_PM_ANA_1: + case RT1019_PM_ANA_2: + case RT1019_PM_ANA_3: + case RT1019_PLL_1: + case RT1019_PLL_2: + case RT1019_PLL_3: + case RT1019_PLL_INT_1: + case RT1019_PLL_INT_3: + case RT1019_MIXER: + case RT1019_CLD_OUT_1: + case RT1019_CLD_OUT_2: + case RT1019_CLD_OUT_3: + case RT1019_CLD_OUT_4: + case RT1019_CLD_OUT_5: + case RT1019_CLD_OUT_6: + case RT1019_CLS_INT_REG_1: + case RT1019_CLS_INT_REG_2: + case RT1019_CLS_INT_REG_3: + case RT1019_CLS_INT_REG_4: + case RT1019_CLS_INT_REG_5: + case RT1019_CLS_INT_REG_6: + case RT1019_CLS_INT_REG_7: + case RT1019_CLS_INT_REG_8: + case RT1019_CLS_INT_REG_9: + case RT1019_CLS_INT_REG_10: + case RT1019_TDM_1: + case RT1019_TDM_2: + case RT1019_TDM_3: + case RT1019_TDM_4: + case RT1019_TDM_5: + case RT1019_TDM_6: + case RT1019_DVOL_1: + case RT1019_DVOL_2: + case RT1019_DVOL_3: + case RT1019_DVOL_4: + case RT1019_DMIX_MONO_1: + case RT1019_DMIX_MONO_2: + case RT1019_CAL_TOP_1: + case RT1019_CAL_TOP_2: + case RT1019_CAL_TOP_3: + case RT1019_CAL_TOP_4: + case RT1019_CAL_TOP_5: + case RT1019_CAL_TOP_6: + case RT1019_CAL_TOP_7: + case RT1019_CAL_TOP_8: + case RT1019_CAL_TOP_9: + case RT1019_CAL_TOP_10: + case RT1019_CAL_TOP_11: + case RT1019_CAL_TOP_12: + case RT1019_CAL_TOP_13: + case RT1019_CAL_TOP_14: + case RT1019_CAL_TOP_15: + case RT1019_CAL_TOP_16: + case RT1019_CAL_TOP_17: + case RT1019_CAL_TOP_18: + case RT1019_CAL_TOP_19: + case RT1019_CAL_TOP_20: + case RT1019_CAL_TOP_21: + case RT1019_CAL_TOP_22: + case RT1019_MDRE_CTRL_1: + case RT1019_MDRE_CTRL_2: + case RT1019_MDRE_CTRL_3: + case RT1019_MDRE_CTRL_4: + case RT1019_MDRE_CTRL_5: + case RT1019_MDRE_CTRL_6: + case RT1019_MDRE_CTRL_7: + case RT1019_MDRE_CTRL_8: + case RT1019_MDRE_CTRL_9: + case RT1019_MDRE_CTRL_10: + case RT1019_SCC_CTRL_1: + case RT1019_SCC_CTRL_2: + case RT1019_SCC_CTRL_3: + case RT1019_SCC_DUMMY: + case RT1019_SIL_DET_1: + case RT1019_SIL_DET_2: + case RT1019_PWM_DC_DET_1: + case RT1019_PWM_DC_DET_2: + case RT1019_PWM_DC_DET_3: + case RT1019_PWM_DC_DET_4: + case RT1019_BEEP_1: + case RT1019_BEEP_2: + case RT1019_PMC_1: + case RT1019_PMC_2: + case RT1019_PMC_3: + case RT1019_PMC_4: + case RT1019_PMC_5: + case RT1019_PMC_6: + case RT1019_PMC_7: + case RT1019_PMC_8: + case RT1019_PMC_9: + case RT1019_SPKDRC_1: + case RT1019_SPKDRC_2: + case RT1019_SPKDRC_3: + case RT1019_SPKDRC_4: + case RT1019_SPKDRC_5: + case RT1019_SPKDRC_6: + case RT1019_SPKDRC_7: + case RT1019_HALF_FREQ_1: + case RT1019_HALF_FREQ_2: + case RT1019_HALF_FREQ_3: + case RT1019_HALF_FREQ_4: + case RT1019_HALF_FREQ_5: + case RT1019_HALF_FREQ_6: + case RT1019_HALF_FREQ_7: + case RT1019_CUR_CTRL_1: + case RT1019_CUR_CTRL_2: + case RT1019_CUR_CTRL_3: + case RT1019_CUR_CTRL_4: + case RT1019_CUR_CTRL_5: + case RT1019_CUR_CTRL_6: + case RT1019_CUR_CTRL_7: + case RT1019_CUR_CTRL_8: + case RT1019_CUR_CTRL_9: + case RT1019_CUR_CTRL_10: + case RT1019_CUR_CTRL_11: + case RT1019_CUR_CTRL_12: + case RT1019_CUR_CTRL_13: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9525, 75, 0); + +static const char * const rt1019_din_source_select[] = { + "Left", + "Right", + "Left + Right average", +}; + +static SOC_ENUM_SINGLE_DECL(rt1019_mono_lr_sel, RT1019_IDS_CTRL, 0, + rt1019_din_source_select); + +static const struct snd_kcontrol_new rt1019_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", RT1019_DMIX_MONO_1, 0, + 127, 0, dac_vol_tlv), + SOC_ENUM("Mono LR Select", rt1019_mono_lr_sel), +}; + +static int r1019_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write(component, RT1019_SDB_CTRL, 0xb); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_write(component, RT1019_SDB_CTRL, 0xa); + break; + default: + break; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt1019_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("AIFRX", "AIF Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, + r1019_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("SPO"), +}; + +static const struct snd_soc_dapm_route rt1019_dapm_routes[] = { + { "DAC", NULL, "AIFRX" }, + { "SPO", NULL, "DAC" }, +}; + +static int rt1019_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 rt1019_priv *rt1019 = snd_soc_component_get_drvdata(component); + int pre_div, bclk_ms, frame_size; + unsigned int val_len = 0, sys_div_da_filter = 0; + unsigned int sys_dac_osr = 0, sys_fifo_clk = 0; + unsigned int sys_clk_cal = 0, sys_asrc_in = 0; + + rt1019->lrck = params_rate(params); + pre_div = rl6231_get_clk_info(rt1019->sysclk, rt1019->lrck); + if (pre_div < 0) { + dev_err(component->dev, "Unsupported clock setting\n"); + return -EINVAL; + } + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(component->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + + bclk_ms = frame_size > 32; + rt1019->bclk = rt1019->lrck * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt1019->bclk, rt1019->lrck); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (pre_div) { + case 0: + sys_div_da_filter = RT1019_SYS_DIV_DA_FIL_DIV1; + sys_dac_osr = RT1019_SYS_DA_OSR_DIV1; + sys_asrc_in = RT1019_ASRC_256FS_DIV1; + sys_fifo_clk = RT1019_SEL_FIFO_DIV1; + sys_clk_cal = RT1019_SEL_CLK_CAL_DIV1; + break; + case 1: + sys_div_da_filter = RT1019_SYS_DIV_DA_FIL_DIV2; + sys_dac_osr = RT1019_SYS_DA_OSR_DIV2; + sys_asrc_in = RT1019_ASRC_256FS_DIV2; + sys_fifo_clk = RT1019_SEL_FIFO_DIV2; + sys_clk_cal = RT1019_SEL_CLK_CAL_DIV2; + break; + case 3: + sys_div_da_filter = RT1019_SYS_DIV_DA_FIL_DIV4; + sys_dac_osr = RT1019_SYS_DA_OSR_DIV4; + sys_asrc_in = RT1019_ASRC_256FS_DIV4; + sys_fifo_clk = RT1019_SEL_FIFO_DIV4; + sys_clk_cal = RT1019_SEL_CLK_CAL_DIV4; + break; + default: + return -EINVAL; + } + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len = RT1019_I2S_DL_20; + break; + case 24: + val_len = RT1019_I2S_DL_24; + break; + case 32: + val_len = RT1019_I2S_DL_32; + break; + case 8: + val_len = RT1019_I2S_DL_8; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT1019_TDM_2, RT1019_I2S_DL_MASK, + val_len); + snd_soc_component_update_bits(component, RT1019_CLK_TREE_1, + RT1019_SEL_FIFO_MASK, sys_fifo_clk); + snd_soc_component_update_bits(component, RT1019_CLK_TREE_2, + RT1019_SYS_DIV_DA_FIL_MASK | RT1019_SYS_DA_OSR_MASK | + RT1019_ASRC_256FS_MASK, sys_div_da_filter | sys_dac_osr | + sys_asrc_in); + snd_soc_component_update_bits(component, RT1019_CLK_TREE_3, + RT1019_SEL_CLK_CAL_MASK, sys_clk_cal); + + return 0; +} + +static int rt1019_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + unsigned int reg_val = 0, reg_val2 = 0; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val2 |= RT1019_TDM_BCLK_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT1019_I2S_DF_LEFT; + break; + + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT1019_I2S_DF_PCM_A_R; + break; + + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT1019_I2S_DF_PCM_B_R; + break; + + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT1019_TDM_2, + RT1019_I2S_DF_MASK, reg_val); + snd_soc_component_update_bits(component, RT1019_TDM_1, + RT1019_TDM_BCLK_MASK, reg_val2); + + return 0; +} + +static int rt1019_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct rt1019_priv *rt1019 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0; + + if (freq == rt1019->sysclk && clk_id == rt1019->sysclk_src) + return 0; + + switch (clk_id) { + case RT1019_SCLK_S_BCLK: + reg_val |= RT1019_CLK_SYS_PRE_SEL_BCLK; + break; + + case RT1019_SCLK_S_PLL: + reg_val |= RT1019_CLK_SYS_PRE_SEL_PLL; + break; + + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + rt1019->sysclk = freq; + rt1019->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + snd_soc_component_update_bits(component, RT1019_CLK_TREE_1, + RT1019_CLK_SYS_PRE_SEL_MASK, reg_val); + + return 0; +} + +static int rt1019_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_component *component = dai->component; + struct rt1019_priv *rt1019 = snd_soc_component_get_drvdata(component); + struct rl6231_pll_code pll_code; + int ret; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + rt1019->pll_in = 0; + rt1019->pll_out = 0; + return 0; + } + + if (source == rt1019->pll_src && freq_in == rt1019->pll_in && + freq_out == rt1019->pll_out) + return 0; + + switch (source) { + case RT1019_PLL_S_BCLK: + snd_soc_component_update_bits(component, RT1019_CLK_TREE_1, + RT1019_PLL_SRC_MASK, RT1019_PLL_SRC_SEL_BCLK); + break; + + case RT1019_PLL_S_RC25M: + snd_soc_component_update_bits(component, RT1019_CLK_TREE_1, + RT1019_PLL_SRC_MASK, RT1019_PLL_SRC_SEL_RC); + break; + + default: + dev_err(component->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_component_update_bits(component, RT1019_PWR_STRP_2, + RT1019_AUTO_BITS_SEL_MASK | RT1019_AUTO_CLK_SEL_MASK, + RT1019_AUTO_BITS_SEL_MANU | RT1019_AUTO_CLK_SEL_MANU); + snd_soc_component_update_bits(component, RT1019_PLL_1, + RT1019_PLL_M_MASK | RT1019_PLL_M_BP_MASK | RT1019_PLL_Q_8_8_MASK, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT1019_PLL_M_SFT | + pll_code.m_bp << RT1019_PLL_M_BP_SFT | + ((pll_code.n_code >> 8) & RT1019_PLL_Q_8_8_MASK)); + snd_soc_component_update_bits(component, RT1019_PLL_2, + RT1019_PLL_Q_7_0_MASK, pll_code.n_code & RT1019_PLL_Q_7_0_MASK); + snd_soc_component_update_bits(component, RT1019_PLL_3, + RT1019_PLL_K_MASK, pll_code.k_code); + + rt1019->pll_in = freq_in; + rt1019->pll_out = freq_out; + rt1019->pll_src = source; + + return 0; +} + +static int rt1019_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int val = 0, rx_slotnum; + int ret = 0, first_bit; + + switch (slots) { + case 4: + val |= RT1019_I2S_TX_4CH; + break; + case 6: + val |= RT1019_I2S_TX_6CH; + break; + case 8: + val |= RT1019_I2S_TX_8CH; + break; + case 2: + break; + default: + return -EINVAL; + } + + switch (slot_width) { + case 20: + val |= RT1019_I2S_DL_20; + break; + case 24: + val |= RT1019_I2S_DL_24; + break; + case 32: + val |= RT1019_I2S_DL_32; + break; + case 8: + val |= RT1019_I2S_DL_8; + break; + case 16: + break; + default: + return -EINVAL; + } + + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + if (rx_slotnum != 1) { + ret = -EINVAL; + dev_err(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + /* This is an assumption that the system sends stereo audio to the + * amplifier typically. And the stereo audio is placed in slot 0/2/4/6 + * as the starting slot. The users could select the channel from + * L/R/L+R by "Mono LR Select" control. + */ + first_bit = __ffs(rx_mask); + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + snd_soc_component_update_bits(component, + RT1019_TDM_3, + RT1019_TDM_I2S_TX_L_DAC1_1_MASK | + RT1019_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1019_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit + 1) << RT1019_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + case 1: + case 3: + case 5: + case 7: + snd_soc_component_update_bits(component, + RT1019_TDM_3, + RT1019_TDM_I2S_TX_L_DAC1_1_MASK | + RT1019_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit - 1) << RT1019_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1019_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + snd_soc_component_update_bits(component, RT1019_TDM_2, + RT1019_I2S_CH_TX_MASK | RT1019_I2S_DF_MASK, val); + +_set_tdm_err_: + return ret; +} + +static int rt1019_probe(struct snd_soc_component *component) +{ + struct rt1019_priv *rt1019 = snd_soc_component_get_drvdata(component); + + rt1019->component = component; + snd_soc_component_write(component, RT1019_SDB_CTRL, 0xa); + + return 0; +} + +#define RT1019_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT1019_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static struct snd_soc_dai_ops rt1019_aif_dai_ops = { + .hw_params = rt1019_hw_params, + .set_fmt = rt1019_set_dai_fmt, + .set_sysclk = rt1019_set_dai_sysclk, + .set_pll = rt1019_set_dai_pll, + .set_tdm_slot = rt1019_set_tdm_slot, +}; + +static struct snd_soc_dai_driver rt1019_dai[] = { + { + .name = "rt1019-aif", + .id = 0, + .playback = { + .stream_name = "AIF Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT1019_STEREO_RATES, + .formats = RT1019_FORMATS, + }, + .ops = &rt1019_aif_dai_ops, + } +}; + +static const struct snd_soc_component_driver soc_component_dev_rt1019 = { + .probe = rt1019_probe, + .controls = rt1019_snd_controls, + .num_controls = ARRAY_SIZE(rt1019_snd_controls), + .dapm_widgets = rt1019_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt1019_dapm_widgets), + .dapm_routes = rt1019_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt1019_dapm_routes), +}; + +static const struct regmap_config rt1019_regmap = { + .reg_bits = 16, + .val_bits = 8, + .use_single_read = true, + .use_single_write = true, + .max_register = RT1019_CUR_CTRL_13, + .volatile_reg = rt1019_volatile_register, + .readable_reg = rt1019_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt1019_reg, + .num_reg_defaults = ARRAY_SIZE(rt1019_reg), +}; + +static const struct i2c_device_id rt1019_i2c_id[] = { + { "rt1019", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt1019_i2c_id); + +static const struct of_device_id rt1019_of_match[] = { + { .compatible = "realtek,rt1019", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt1019_of_match); + +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt1019_acpi_match[] = { + { "10EC1019", 0}, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt1019_acpi_match); +#endif + +static int rt1019_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt1019_priv *rt1019; + int ret; + unsigned int val, val2, dev_id; + + rt1019 = devm_kzalloc(&i2c->dev, sizeof(struct rt1019_priv), + GFP_KERNEL); + if (!rt1019) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt1019); + + rt1019->regmap = devm_regmap_init_i2c(i2c, &rt1019_regmap); + if (IS_ERR(rt1019->regmap)) { + ret = PTR_ERR(rt1019->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt1019->regmap, RT1019_DEV_ID_1, &val); + regmap_read(rt1019->regmap, RT1019_DEV_ID_2, &val2); + dev_id = val << 8 | val2; + if (dev_id != RT1019_DEVICE_ID_VAL && dev_id != RT1019_DEVICE_ID_VAL2) { + dev_err(&i2c->dev, + "Device with ID register 0x%x is not rt1019\n", dev_id); + return -ENODEV; + } + + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_rt1019, rt1019_dai, ARRAY_SIZE(rt1019_dai)); +} + +struct i2c_driver rt1019_i2c_driver = { + .driver = { + .name = "rt1019", + .of_match_table = of_match_ptr(rt1019_of_match), + .acpi_match_table = ACPI_PTR(rt1019_acpi_match), + }, + .probe = rt1019_i2c_probe, + .id_table = rt1019_i2c_id, +}; +module_i2c_driver(rt1019_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT1019 driver"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt1019.h b/sound/soc/codecs/rt1019.h new file mode 100644 index 000000000000..46973e71c963 --- /dev/null +++ b/sound/soc/codecs/rt1019.h @@ -0,0 +1,320 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt1019.h -- RT1019 ALSA SoC audio amplifier driver + * + * Copyright(c) 2021 Realtek Semiconductor Corp. + */ + +#ifndef __RT1019_H__ +#define __RT1019_H__ + +#define RT1019_DEVICE_ID_VAL 0x1019 +#define RT1019_DEVICE_ID_VAL2 0x6731 + +#define RT1019_RESET 0x0000 +#define RT1019_PAD_DRV_1 0x0002 +#define RT1019_PAD_DRV_2 0x0003 +#define RT1019_PAD_PULL_1 0x0005 +#define RT1019_PAD_PULL_2 0x0006 +#define RT1019_PAD_PULL_3 0x0007 +#define RT1019_I2C_CTRL_1 0x0008 +#define RT1019_I2C_CTRL_2 0x0009 +#define RT1019_I2C_CTRL_3 0x000a +#define RT1019_IDS_CTRL 0x0011 +#define RT1019_ASEL_CTRL 0x0013 +#define RT1019_PLL_RESET 0x0015 +#define RT1019_PWR_STRP_1 0x0017 +#define RT1019_PWR_STRP_2 0x0019 +#define RT1019_BEEP_TONE 0x001b +#define RT1019_SIL_DET_GAT 0x001d +#define RT1019_CLASSD_TIME 0x001f +#define RT1019_CLASSD_OCP 0x0021 +#define RT1019_PHASE_SYNC 0x0023 +#define RT1019_STAT_MACH_1 0x0025 +#define RT1019_STAT_MACH_2 0x0026 +#define RT1019_EFF_CTRL 0x0028 +#define RT1019_FS_DET_1 0x002a +#define RT1019_FS_DET_2 0x002b +#define RT1019_FS_DET_3 0x002c +#define RT1019_FS_DET_4 0x002d +#define RT1019_FS_DET_5 0x002e +#define RT1019_FS_DET_6 0x002f +#define RT1019_FS_DET_7 0x0030 +#define RT1019_ANA_CTRL 0x0053 +#define RT1019_DUMMY_A 0x0055 +#define RT1019_DUMMY_B 0x0056 +#define RT1019_DUMMY_C 0x0057 +#define RT1019_DUMMY_D 0x0058 +#define RT1019_ANA_READ 0x005a +#define RT1019_VER_ID 0x005c +#define RT1019_CUSTOM_ID 0x005d +#define RT1019_VEND_ID_1 0x005e +#define RT1019_VEND_ID_2 0x005f +#define RT1019_DEV_ID_1 0x0061 +#define RT1019_DEV_ID_2 0x0062 +#define RT1019_TEST_PAD 0x0064 +#define RT1019_SDB_CTRL 0x0066 +#define RT1019_TEST_CTRL_1 0x0068 +#define RT1019_TEST_CTRL_2 0x006a +#define RT1019_TEST_CTRL_3 0x006c +#define RT1019_SCAN_MODE 0x006e +#define RT1019_CLK_TREE_1 0x0100 +#define RT1019_CLK_TREE_2 0x0101 +#define RT1019_CLK_TREE_3 0x0102 +#define RT1019_CLK_TREE_4 0x0103 +#define RT1019_CLK_TREE_5 0x0104 +#define RT1019_CLK_TREE_6 0x0105 +#define RT1019_CLK_TREE_7 0x0106 +#define RT1019_CLK_TREE_8 0x0107 +#define RT1019_CLK_TREE_9 0x0108 +#define RT1019_ASRC_1 0x0200 +#define RT1019_ASRC_2 0x0201 +#define RT1019_ASRC_3 0x0202 +#define RT1019_ASRC_4 0x0203 +#define RT1019_SYS_CLK 0x0300 +#define RT1019_BIAS_CUR_1 0x0301 +#define RT1019_BIAS_CUR_2 0x0302 +#define RT1019_BIAS_CUR_3 0x0303 +#define RT1019_BIAS_CUR_4 0x0304 +#define RT1019_CHOP_CLK_DAC 0x0306 +#define RT1019_CHOP_CLK_ADC 0x0308 +#define RT1019_LDO_CTRL_1 0x030a +#define RT1019_LDO_CTRL_2 0x030b +#define RT1019_PM_ANA_1 0x030d +#define RT1019_PM_ANA_2 0x030e +#define RT1019_PM_ANA_3 0x030f +#define RT1019_PLL_1 0x0311 +#define RT1019_PLL_2 0x0312 +#define RT1019_PLL_3 0x0313 +#define RT1019_PLL_INT_1 0x0315 +#define RT1019_PLL_INT_3 0x0318 +#define RT1019_MIXER 0x031a +#define RT1019_CLD_OUT_1 0x031c +#define RT1019_CLD_OUT_2 0x031d +#define RT1019_CLD_OUT_3 0x031e +#define RT1019_CLD_OUT_4 0x031f +#define RT1019_CLD_OUT_5 0x0320 +#define RT1019_CLD_OUT_6 0x0321 +#define RT1019_CLS_INT_REG_1 0x0323 +#define RT1019_CLS_INT_REG_2 0x0324 +#define RT1019_CLS_INT_REG_3 0x0325 +#define RT1019_CLS_INT_REG_4 0x0326 +#define RT1019_CLS_INT_REG_5 0x0327 +#define RT1019_CLS_INT_REG_6 0x0328 +#define RT1019_CLS_INT_REG_7 0x0329 +#define RT1019_CLS_INT_REG_8 0x0330 +#define RT1019_CLS_INT_REG_9 0x0331 +#define RT1019_CLS_INT_REG_10 0x0332 +#define RT1019_TDM_1 0x0400 +#define RT1019_TDM_2 0x0401 +#define RT1019_TDM_3 0x0402 +#define RT1019_TDM_4 0x0403 +#define RT1019_TDM_5 0x0404 +#define RT1019_TDM_6 0x0405 +#define RT1019_DVOL_1 0x0500 +#define RT1019_DVOL_2 0x0501 +#define RT1019_DVOL_3 0x0502 +#define RT1019_DVOL_4 0x0503 +#define RT1019_DMIX_MONO_1 0x0504 +#define RT1019_DMIX_MONO_2 0x0505 +#define RT1019_CAL_TOP_1 0x0600 +#define RT1019_CAL_TOP_2 0x0601 +#define RT1019_CAL_TOP_3 0x0602 +#define RT1019_CAL_TOP_4 0x0603 +#define RT1019_CAL_TOP_5 0x0604 +#define RT1019_CAL_TOP_6 0x0605 +#define RT1019_CAL_TOP_7 0x0606 +#define RT1019_CAL_TOP_8 0x0607 +#define RT1019_CAL_TOP_9 0x0608 +#define RT1019_CAL_TOP_10 0x0609 +#define RT1019_CAL_TOP_11 0x060a +#define RT1019_CAL_TOP_12 0x060b +#define RT1019_CAL_TOP_13 0x060c +#define RT1019_CAL_TOP_14 0x060d +#define RT1019_CAL_TOP_15 0x060e +#define RT1019_CAL_TOP_16 0x060f +#define RT1019_CAL_TOP_17 0x0610 +#define RT1019_CAL_TOP_18 0x0611 +#define RT1019_CAL_TOP_19 0x0612 +#define RT1019_CAL_TOP_20 0x0613 +#define RT1019_CAL_TOP_21 0x0614 +#define RT1019_CAL_TOP_22 0x0615 +#define RT1019_MDRE_CTRL_1 0x0700 +#define RT1019_MDRE_CTRL_2 0x0701 +#define RT1019_MDRE_CTRL_3 0x0702 +#define RT1019_MDRE_CTRL_4 0x0703 +#define RT1019_MDRE_CTRL_5 0x0704 +#define RT1019_MDRE_CTRL_6 0x0705 +#define RT1019_MDRE_CTRL_7 0x0706 +#define RT1019_MDRE_CTRL_8 0x0707 +#define RT1019_MDRE_CTRL_9 0x0708 +#define RT1019_MDRE_CTRL_10 0x0709 +#define RT1019_SCC_CTRL_1 0x0800 +#define RT1019_SCC_CTRL_2 0x0801 +#define RT1019_SCC_CTRL_3 0x0802 +#define RT1019_SCC_DUMMY 0x0803 +#define RT1019_SIL_DET_1 0x0900 +#define RT1019_SIL_DET_2 0x0901 +#define RT1019_PWM_DC_DET_1 0x0a00 +#define RT1019_PWM_DC_DET_2 0x0a01 +#define RT1019_PWM_DC_DET_3 0x0a02 +#define RT1019_PWM_DC_DET_4 0x0a03 +#define RT1019_BEEP_1 0x0b00 +#define RT1019_BEEP_2 0x0b01 +#define RT1019_PMC_1 0x0c00 +#define RT1019_PMC_2 0x0c01 +#define RT1019_PMC_3 0x0c02 +#define RT1019_PMC_4 0x0c03 +#define RT1019_PMC_5 0x0c04 +#define RT1019_PMC_6 0x0c05 +#define RT1019_PMC_7 0x0c06 +#define RT1019_PMC_8 0x0c07 +#define RT1019_PMC_9 0x0c08 +#define RT1019_SPKDRC_1 0x0d00 +#define RT1019_SPKDRC_2 0x0d01 +#define RT1019_SPKDRC_3 0x0d02 +#define RT1019_SPKDRC_4 0x0d03 +#define RT1019_SPKDRC_5 0x0d04 +#define RT1019_SPKDRC_6 0x0d05 +#define RT1019_SPKDRC_7 0x0d06 +#define RT1019_HALF_FREQ_1 0x0e00 +#define RT1019_HALF_FREQ_2 0x0e01 +#define RT1019_HALF_FREQ_3 0x0e02 +#define RT1019_HALF_FREQ_4 0x0e03 +#define RT1019_HALF_FREQ_5 0x0e04 +#define RT1019_HALF_FREQ_6 0x0e05 +#define RT1019_HALF_FREQ_7 0x0e06 +#define RT1019_CUR_CTRL_1 0x0f00 +#define RT1019_CUR_CTRL_2 0x0f01 +#define RT1019_CUR_CTRL_3 0x0f02 +#define RT1019_CUR_CTRL_4 0x0f03 +#define RT1019_CUR_CTRL_5 0x0f04 +#define RT1019_CUR_CTRL_6 0x0f05 +#define RT1019_CUR_CTRL_7 0x0f06 +#define RT1019_CUR_CTRL_8 0x0f07 +#define RT1019_CUR_CTRL_9 0x0f08 +#define RT1019_CUR_CTRL_10 0x0f09 +#define RT1019_CUR_CTRL_11 0x0f0a +#define RT1019_CUR_CTRL_12 0x0f0b +#define RT1019_CUR_CTRL_13 0x0f0c + +/* 0x0019 Power On Strap Control-2 */ +#define RT1019_AUTO_BITS_SEL_MASK (0x1 << 5) +#define RT1019_AUTO_BITS_SEL_AUTO (0x1 << 5) +#define RT1019_AUTO_BITS_SEL_MANU (0x0 << 5) +#define RT1019_AUTO_CLK_SEL_MASK (0x1 << 4) +#define RT1019_AUTO_CLK_SEL_AUTO (0x1 << 4) +#define RT1019_AUTO_CLK_SEL_MANU (0x0 << 4) + +/* 0x0100 Clock Tree Control-1 */ +#define RT1019_CLK_SYS_PRE_SEL_MASK (0x1 << 7) +#define RT1019_CLK_SYS_PRE_SEL_SFT 7 +#define RT1019_CLK_SYS_PRE_SEL_BCLK (0x0 << 7) +#define RT1019_CLK_SYS_PRE_SEL_PLL (0x1 << 7) +#define RT1019_PLL_SRC_MASK (0x1 << 4) +#define RT1019_PLL_SRC_SFT 4 +#define RT1019_PLL_SRC_SEL_BCLK (0x0 << 4) +#define RT1019_PLL_SRC_SEL_RC (0x1 << 4) +#define RT1019_SEL_FIFO_MASK (0x3 << 2) +#define RT1019_SEL_FIFO_DIV1 (0x0 << 2) +#define RT1019_SEL_FIFO_DIV2 (0x1 << 2) +#define RT1019_SEL_FIFO_DIV4 (0x2 << 2) + +/* 0x0101 clock tree control-2 */ +#define RT1019_SYS_DIV_DA_FIL_MASK (0x7 << 5) +#define RT1019_SYS_DIV_DA_FIL_DIV1 (0x2 << 5) +#define RT1019_SYS_DIV_DA_FIL_DIV2 (0x3 << 5) +#define RT1019_SYS_DIV_DA_FIL_DIV4 (0x4 << 5) +#define RT1019_SYS_DA_OSR_MASK (0x3 << 2) +#define RT1019_SYS_DA_OSR_DIV1 (0x0 << 2) +#define RT1019_SYS_DA_OSR_DIV2 (0x1 << 2) +#define RT1019_SYS_DA_OSR_DIV4 (0x2 << 2) +#define RT1019_ASRC_256FS_MASK 0x3 +#define RT1019_ASRC_256FS_DIV1 0x0 +#define RT1019_ASRC_256FS_DIV2 0x1 +#define RT1019_ASRC_256FS_DIV4 0x2 + +/* 0x0102 clock tree control-3 */ +#define RT1019_SEL_CLK_CAL_MASK (0x3 << 6) +#define RT1019_SEL_CLK_CAL_DIV1 (0x0 << 6) +#define RT1019_SEL_CLK_CAL_DIV2 (0x1 << 6) +#define RT1019_SEL_CLK_CAL_DIV4 (0x2 << 6) + +/* 0x0311 PLL-1 */ +#define RT1019_PLL_M_MASK (0xf << 4) +#define RT1019_PLL_M_SFT 4 +#define RT1019_PLL_M_BP_MASK (0x1 << 1) +#define RT1019_PLL_M_BP_SFT 1 +#define RT1019_PLL_Q_8_8_MASK (0x1) + +/* 0x0312 PLL-2 */ +#define RT1019_PLL_Q_7_0_MASK 0xff + +/* 0x0313 PLL-3 */ +#define RT1019_PLL_K_MASK 0x1f + +/* 0x0400 TDM Control-1 */ +#define RT1019_TDM_BCLK_MASK (0x1 << 6) +#define RT1019_TDM_BCLK_NORM (0x0 << 6) +#define RT1019_TDM_BCLK_INV (0x1 << 6) + +/* 0x0401 TDM Control-2 */ +#define RT1019_I2S_CH_TX_MASK (0x3 << 6) +#define RT1019_I2S_CH_TX_SFT 6 +#define RT1019_I2S_TX_2CH (0x0 << 6) +#define RT1019_I2S_TX_4CH (0x1 << 6) +#define RT1019_I2S_TX_6CH (0x2 << 6) +#define RT1019_I2S_TX_8CH (0x3 << 6) +#define RT1019_I2S_DF_MASK (0x7 << 3) +#define RT1019_I2S_DF_SFT 3 +#define RT1019_I2S_DF_I2S (0x0 << 3) +#define RT1019_I2S_DF_LEFT (0x1 << 3) +#define RT1019_I2S_DF_PCM_A_R (0x2 << 3) +#define RT1019_I2S_DF_PCM_B_R (0x3 << 3) +#define RT1019_I2S_DF_PCM_A_F (0x6 << 3) +#define RT1019_I2S_DF_PCM_B_F (0x7 << 3) +#define RT1019_I2S_DL_MASK 0x7 +#define RT1019_I2S_DL_SFT 0 +#define RT1019_I2S_DL_16 0x0 +#define RT1019_I2S_DL_20 0x1 +#define RT1019_I2S_DL_24 0x2 +#define RT1019_I2S_DL_32 0x3 +#define RT1019_I2S_DL_8 0x4 + +/* TDM1 Control-3 (0x0402) */ +#define RT1019_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 4) +#define RT1019_TDM_I2S_TX_R_DAC1_1_MASK 0x7 +#define RT1019_TDM_I2S_TX_L_DAC1_1_SFT 4 +#define RT1019_TDM_I2S_TX_R_DAC1_1_SFT 0 + +/* System Clock Source */ +enum { + RT1019_SCLK_S_BCLK, + RT1019_SCLK_S_PLL, +}; + +/* PLL1 Source */ +enum { + RT1019_PLL_S_BCLK, + RT1019_PLL_S_RC25M, +}; + +enum { + RT1019_AIF1, + RT1019_AIFS +}; + +struct rt1019_priv { + struct snd_soc_component *component; + struct regmap *regmap; + int sysclk; + int sysclk_src; + int lrck; + int bclk; + int pll_src; + int pll_in; + int pll_out; + unsigned int bclk_ratio; +}; + +#endif /* __RT1019_H__ */ -- 2.29.0

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[PATCH 5.10 02/49] [PATCH v2] ASoC: SOF: Intel: broadwell: fix mutual exclusion with catpt driver
by gregkh＠linuxfoundation.org 10 Mar '21

10 Mar '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> In v5.10, the "haswell" driver was replaced by the "catpt" driver, but the mutual exclusion with the SOF driver was not updated. This leads to errors with card names and UCM profiles not being loaded by PulseAudio. This fix should only be applied on v5.10-stable, the mutual exclusion was removed in 5.11. Reported-by: David Ward <david.ward(a)ll.mit.edu> Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=211985 Fixes: 6cbfa11d2694 ("ASoC: Intel: Select catpt and deprecate haswell") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Mark Brown <broonie(a)kernel.org> Cc: <stable(a)vger.kernel.org> Cc: Sasha Levin <sashal(a)kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- sound/soc/sof/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -84,7 +84,7 @@ config SND_SOC_SOF_BAYTRAIL config SND_SOC_SOF_BROADWELL_SUPPORT bool "SOF support for Broadwell" - depends on SND_SOC_INTEL_HASWELL=n + depends on SND_SOC_INTEL_CATPT=n help This adds support for Sound Open Firmware for Intel(R) platforms using the Broadwell processors.

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[PATCH v2] ASoC: SOF: Intel: broadwell: fix mutual exclusion with catpt driver
by Pierre-Louis Bossart 10 Mar '21

10 Mar '21

In v5.10, the "haswell" driver was replaced by the "catpt" driver, but the mutual exclusion with the SOF driver was not updated. This leads to errors with card names and UCM profiles not being loaded by PulseAudio. This fix should only be applied on v5.10-stable, the mutual exclusion was removed in 5.11. Reported-by: David Ward <david.ward(a)ll.mit.edu> Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=211985 Fixes: 6cbfa11d2694 ("ASoC: Intel: Select catpt and deprecate haswell") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Mark Brown <broonie(a)kernel.org> Cc: <stable(a)vger.kernel.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Sasha Levin <sashal(a)kernel.org> --- v2: added Mark and Cezary tags, fixed stable address, added maintainers sound/soc/sof/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index de7ff2d097ab..6708a2c5a838 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -84,7 +84,7 @@ config SND_SOC_SOF_BAYTRAIL config SND_SOC_SOF_BROADWELL_SUPPORT bool "SOF support for Broadwell" - depends on SND_SOC_INTEL_HASWELL=n + depends on SND_SOC_INTEL_CATPT=n help This adds support for Sound Open Firmware for Intel(R) platforms using the Broadwell processors. -- 2.25.1

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[PATCH 0/3] ALSA: hda: Fix potential bad accesses at suspend/resume
by Takashi Iwai 10 Mar '21

10 Mar '21

Hi, this is a small patch series to address the bugs that triggers the unexpected accesses during the system suspend/resume. It happens often with Nvidia driver and HDMI codec driver, and it may lead to the serious CORB/RIRB errors. Although the issues are seen mostly on DP/HDMI, a part of the problems is rather generic to all platforms. Takashi === Takashi Iwai (3): ALSA: hda: Flush pending unsolicited events before suspend ALSA: hda: Avoid spurious unsol event handling during S3/S4 ALSA: hda/hdmi: Cancel pending works before suspend sound/pci/hda/hda_bind.c | 4 ++++ sound/pci/hda/hda_intel.c | 2 ++ sound/pci/hda/patch_hdmi.c | 13 +++++++++++++ 3 files changed, 19 insertions(+) -- 2.26.2

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[PATCH alsa-ucm-conf 1/3] codecs/es8316: Fix capture settings
by Hans de Goede 10 Mar '21

10 Mar '21

Fix the following issues with the capture settings: 1. Disable ALC / Auto Level Control, it tries to always get an input signal even when the user is not talking into the mic leading to it cranking up the volume till there is noise at the same level as the user talking. 2. The 'ADC PGA Gain Volume' volume control is not the main 'ADC vol' control, it is the mic amplifier control and setting it to 10 sets it to 24dB not 0dB. Adjust the comment and set it to 7 which is 16dB which gives a good microphone signal strength without introducing too much noise (with 'ADC Capture Volume' set to 0dB). 3. There actually is a main 'ADC vol' control which goes from -96 to 0dB which is simply called the 'ADC Capture Volume' and when disabling ALC it defaults to -96 dB. Add a line setting it to 0dB. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/codecs/es8316/EnableSeq.conf | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/ucm2/codecs/es8316/EnableSeq.conf b/ucm2/codecs/es8316/EnableSeq.conf index 97ccfc9..272ef1a 100644 --- a/ucm2/codecs/es8316/EnableSeq.conf +++ b/ucm2/codecs/es8316/EnableSeq.conf @@ -11,11 +11,15 @@ EnableSequence [ cset "name='Headphone Mixer Volume' 4" # Set DAC vol to 0 dB (192/192) cset "name='DAC Playback Volume' 192" - # Set ADC vold to 0 dB (10/10) - cset "name='ADC PGA Gain Volume' 10" + + # Disable Auto Level Control + cset "name='ALC Capture Switch' off" + # Set ADC vol to 0 dB (192/192) + cset "name='ADC Capture Volume' 192" + # Set Mic amplifier to +16 dB + cset "name='ADC PGA Gain Volume' 7" # Setup muxes / switches cset "name='Left Headphone Mixer Left DAC Switch' on" cset "name='Right Headphone Mixer Right DAC Switch' on" - cset "name='ALC Capture Switch' on" ] -- 2.30.1

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RE: [PATCH] Add audio driver for rk817 upstream
by Christopher Morgan 10 Mar '21

10 Mar '21

Thank you. Turns out my mail client crapped out on me (I'm trying to switch to using mutt for replies and accidentally deleted the thread) so I'm afraid my nice and neat reply didn't go quite as expected. I'll work on the things you mention below and resubmit this as an RFC. I'll include the devicetree bindings that I did so you can see how things are mapped. To answer your questions: Acknowledged on the maintainers, the signed-off tag, the pointer, and the commit message. I don't know for sure the best way to accomplish things, I read in the documentation that the simple-audio-card has an option of setting a GPIO which should automatically register with the jack for headphone or microphone sources, that's what I'm trying to do. The specific chip I'm working with has 2 headphone ports (R and L), a single speaker port (let's say LR combined to L), and a single microphone port (L). The codec itself supports 2 channels for the DAC which either power the headphones or the speaker (or both at the same time, if you don't mind either only sending L to the speaker with a proper RL split on the headphones or RL combined to the L headphone with a proper RL combined to the speaker). There are also 2 channels on the ADC, but again only one microphone port which seems odd. Register documentation here for the codec: https://rockchip.fr/RK817%20datasheet%20V1.01.pdf As for the specific implementation I'm working with its here: https://dn.odroid.com/ODROID_GO_ADVANCE/ODROID_GO_ADVANCE_rev1.1.pdf Here's my simple-audio-card DT node: rk817-sound { compatible = "simple-audio-card"; simple-audio-card,format = "i2s"; simple-audio-card,name = "rockchip,rk817-codec"; simple-audio-card,mclk-fs = <256>; simple-audio-card,widgets = "Microphone", "Mic Jack", "Headphone", "Headphones", simple-audio-card,routing = "MIC", "Mic Jack", "Headphones", "HPOL", "Headphones", "HPOR", simple-audio-card,hp-det-gpio = <&gpio2 RK_PC6 GPIO_ACTIVE_LOW>; simple-audio-card,cpu { sound-dai = <&i2s1_2ch>; }; simple-audio-card,codec { sound-dai = <&rk817_codec>; }; }; I copied the rockchip code from the BSP kernel and it had some of the variables, such as the pre_path, as a long. Since it looks like it's just checking the path enums to see if there was a change, I imagine int will do just fine on the pre_path. Speaking of which, that is pre as opposed to post. In this specific step it's looking to see if we're going from off to on or on to off and setting the register values accordingly. As for whether or not the steps are trying to bypass or reimplement something that should be done in DAPM, I'm not sure honestly; this is my first audio driver (or any driver for that matter). I tried to read the DAPM documentation and didn't understand much of it. What I can say is that the code in question is setting the register values via I2C depending upon which audio path is chosen. For example on the SPK_PATH it sets the register at address 0x38 (RK817_CODEC_DDAC_MUTE_MIXCTL) to 0x10, which tells it to switch the amplifier on and combine the channels into a single channel. The HP path likewise does the opposite, setting this same register to 0x00 which turns off the amplifier/splits it back into L+R channels. Curiously enough, this is actually a bug I found in Rockchip's BSP kernel where the channels don't get split apart again when jumping from external speaker back to headphones and your headphones start to output R to R and RL to L... In regards to the DT bindings, I will document all of them, however that poses a dilemma. Most of the bindings I'm unable to test, as aside from mic-in-differential I'm not really using any of them on the single hardware implementation I have available to me. Should I document them based on Rockchip's BSP kernel and note that they are untested, or should I simply delete them and their associated functionality (and focus on that which I can test)? I'll test removing the unregister_component code and see if that does anything unexpected. Thank you again for all of your help. On 3/9/21 5:41 PM, Chris Morgan wrote: > I'm wondering if you all can help me. I'm trying to get the rk817 > codec driver working from Rockchip's BSP kernel sources (GPL per > the license) and I'm struggling with a few parts. The first part is to Cc: audio maintainers (Takashi Iwai and Mark Brown). Then you need to make sure there is a Signed-off-by tag from the original contribution this code is based on, along with yours. It doesn't hurt to have a pointer to that code either in the commit message. > is I'm not sure if I have my audio paths set up correctly. For example > the sinks I have set up are for HPOL and HPOR, and the source is for > MIC. While this does work (audio output seems fine) I'm having issues > with the GPIO to detect headphone insertion. When I insert headphones > I expect the audio to output to the headphones, and when I remove > headphones I expect the audio to output to a speaker. Right now I > have to manually change the output between the different paths. You don't necessarily have to do everything at the kernel level, it's not uncommon to have the driver set a kcontrol for jack detection, and let userspace change settings on a jack detection event. PulseAudio relies on UCM JackControl to switch to Headphones and Headset devices. > Additionally, while the codec "technically" has dual channel inputs > for the microphone, I'm only using the L channel. Should I have a > stereo mixer? Note that I'm using the simple-audio-card to set my > paths, widgets, and the gpio pin. They all work, just not seamlessly > together. > > Basically, I'm wanting to know if and how I should set up my audio > paths for the speaker and headphones, they use mostly the same pins > and only really differ in setting the external amp settings. > Additionally, once I set up my paths how do I ensure that the GPIO > events will result in switching between the speaker/headphone path? > > Any help you can provide is appreciated. This is my first attempt at > something of this magnitude (for me it's a big step), even if it's > just trying to facilitate in getting a vendor's code ready for upstream. > --- /dev/null > +++ b/sound/soc/codecs/rk817_codec.c > @@ -0,0 +1,1148 @@ > +/* > + * Copyright (c) 2018 Rockchip Electronics Co. Ltd. > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation; either version 2 of the License, or > + * (at your option) any later version. > + * > + * This program is distributed in the hope that it will be useful, > + * but WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the > + * GNU General Public License for more details. remove and replace by an SPDX line. > +static int rk817_playback_path_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); > + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); > + long int pre_path; it's uncommon to see this, use either int or u32/u64 if your require a specific size. > + > + if (rk817->playback_path == ucontrol->value.integer.value[0]) { > + return 0; > + } > + > + pre_path = rk817->playback_path; 'pre' as in 'previous' or 'preamp' or as opposed to 'post'? > + rk817->playback_path = ucontrol->value.integer.value[0]; > + > + if (rk817->playback_path != OFF) > + clk_prepare_enable(rk817->mclk); > + else > + clk_disable_unprepare(rk817->mclk); > + > + switch (rk817->playback_path) { > + case OFF: > + if (pre_path != OFF && pre_path != HP_PATH) { > + rk817_codec_power_down(component, RK817_CODEC_PLAYBACK); > + if (rk817->capture_path == 0) > + rk817_codec_power_down(component, RK817_CODEC_ALL); > + } > + break; > + case SPK_PATH: > + if (pre_path == OFF) > + rk817_codec_power_up(component, RK817_CODEC_PLAYBACK); > + if (!rk817->use_ext_amplifier) { > + /* power on dac ibias/l/r */ > + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, > + PWD_DACBIAS_ON | PWD_DACD_ON | > + PWD_DACL_DOWN | PWD_DACR_DOWN); > + /* CLASS D mode, combine LR channels */ > + snd_soc_component_write(component, > + RK817_CODEC_DDAC_MUTE_MIXCTL, > + 0x10); > + /* CLASS D enable */ > + snd_soc_component_write(component, > + RK817_CODEC_ACLASSD_CFG1, > + 0xa5); > + /* restart CLASS D, OCPP/N */ > + snd_soc_component_write(component, > + RK817_CODEC_ACLASSD_CFG2, > + 0xf7); > + } else { > + /* HP_CP_EN , CP 2.3V */ > + snd_soc_component_write(component, RK817_CODEC_AHP_CP, > + 0x11); > + /* power on HP two stage opamp ,HP amplitude 0db */ > + snd_soc_component_write(component, RK817_CODEC_AHP_CFG0, > + 0x80); > + /* power on dac ibias/l/r */ > + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, > + PWD_DACBIAS_ON | PWD_DACD_DOWN | > + PWD_DACL_ON | PWD_DACR_ON); > + snd_soc_component_update_bits(component, > + RK817_CODEC_DDAC_MUTE_MIXCTL, > + DACMT_ENABLE, DACMT_DISABLE); > + } > + snd_soc_component_write(component, RK817_CODEC_DDAC_VOLL, > + rk817->spk_volume); > + snd_soc_component_write(component, RK817_CODEC_DDAC_VOLR, > + rk817->spk_volume); > + break; > + case HP_PATH: > + if (pre_path == OFF) > + rk817_codec_power_up(component, RK817_CODEC_PLAYBACK); > + /* HP_CP_EN , CP 2.3V */ > + snd_soc_component_write(component, RK817_CODEC_AHP_CP, 0x11); > + /* power on HP two stage opamp ,HP amplitude 0db */ > + snd_soc_component_write(component, RK817_CODEC_AHP_CFG0, 0x80); > + /* power on dac ibias/l/r */ > + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, > + PWD_DACBIAS_ON | PWD_DACD_DOWN | > + PWD_DACL_ON | PWD_DACR_ON); > + /* CLASS D mode disable, split LR channels */ > + snd_soc_component_write(component, > + RK817_CODEC_DDAC_MUTE_MIXCTL, > + 0x00); > + > + snd_soc_component_write(component, RK817_CODEC_DDAC_VOLL, > + rk817->hp_volume); > + snd_soc_component_write(component, RK817_CODEC_DDAC_VOLR, > + rk817->hp_volume); > + break; > + case SPK_HP: > + if (pre_path == OFF) > + rk817_codec_power_up(component, RK817_CODEC_PLAYBACK); > + > + /* HP_CP_EN , CP 2.3V */ > + snd_soc_component_write(component, RK817_CODEC_AHP_CP, 0x11); > + /* power on HP two stage opamp ,HP amplitude 0db */ > + snd_soc_component_write(component, RK817_CODEC_AHP_CFG0, 0x80); > + > + /* power on dac ibias/l/r */ > + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, > + PWD_DACBIAS_ON | PWD_DACD_ON | > + PWD_DACL_ON | PWD_DACR_ON); > + > + if (!rk817->use_ext_amplifier) { > + /* CLASS D mode, combine LR channels */ > + snd_soc_component_write(component, > + RK817_CODEC_DDAC_MUTE_MIXCTL, > + 0x10); > + /* CLASS D enable */ > + snd_soc_component_write(component, > + RK817_CODEC_ACLASSD_CFG1, > + 0xa5); > + /* restart CLASS D, OCPP/N */ > + snd_soc_component_write(component, > + RK817_CODEC_ACLASSD_CFG2, > + 0xf7); > + } > + > + snd_soc_component_write(component, RK817_CODEC_DDAC_VOLL, > + rk817->hp_volume); > + snd_soc_component_write(component, RK817_CODEC_DDAC_VOLR, > + rk817->hp_volume); > + break; > + default: > + return -EINVAL; > + } > + > + return 0; > +} > + > +static int rk817_capture_path_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); > + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); > + > + dev_dbg(component->dev, "%s:capture_path %ld\n", __func__, rk817->capture_path); > + ucontrol->value.integer.value[0] = rk817->capture_path; > + return 0; > +} > + > +static int rk817_capture_path_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); > + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); > + long int pre_path; > + > + if (rk817->capture_path == ucontrol->value.integer.value[0]) { > + dev_dbg(component->dev, "%s:capture_path is not changed!\n", > + __func__); > + return 0; > + } > + > + pre_path = rk817->capture_path; > + rk817->capture_path = ucontrol->value.integer.value[0]; > + > + if (rk817->capture_path != MIC_OFF) > + clk_prepare_enable(rk817->mclk); > + else > + clk_disable_unprepare(rk817->mclk); > + > + switch (rk817->capture_path) { > + case MIC_OFF: > + if (pre_path != MIC_OFF) > + rk817_codec_power_down(component, RK817_CODEC_CAPTURE); > + break; > + case MIC: > + if (pre_path == MIC_OFF) > + rk817_codec_power_up(component, RK817_CODEC_CAPTURE); these sequences look like trying to bypass DAPM or re-invent it with custom state machines. > + > + if (rk817->adc_for_loopback) { > + /* don't need to gain when adc use for loopback */ > + snd_soc_component_update_bits(component, > + RK817_CODEC_AMIC_CFG0, > + 0xf, > + 0x0); > + snd_soc_component_write(component, > + RK817_CODEC_DMIC_PGA_GAIN, > + 0x66); > + snd_soc_component_write(component, > + RK817_CODEC_DADC_VOLL, > + 0x00); > + snd_soc_component_write(component, > + RK817_CODEC_DADC_VOLR, > + 0x00); > + break; > + } > + if (!rk817->mic_in_differential) { > + snd_soc_component_write(component, > + RK817_CODEC_DADC_VOLR, > + 0xff); > + snd_soc_component_update_bits(component, > + RK817_CODEC_AADC_CFG0, > + ADC_R_PWD_MASK, > + ADC_R_PWD_EN); > + snd_soc_component_update_bits(component, > + RK817_CODEC_AMIC_CFG0, > + PWD_PGA_R_MASK, > + PWD_PGA_R_EN); > + } > + break; > + default: > + return -EINVAL; > + } > + > + return 0; > +} > + [...] > +static int rk817_codec_parse_dt_property(struct device *dev, > + struct rk817_codec_priv *rk817) > +{ > + struct device_node *node = dev->parent->of_node; > + int ret; > + > + if (!node) { > + dev_err(dev, "%s() dev->parent->of_node is NULL\n", > + __func__); > + return -ENODEV; > + } > + > + node = of_get_child_by_name(dev->parent->of_node, "codec"); > + if (!node) { > + dev_err(dev, "%s() Can not get child: codec\n", > + __func__); > + return -ENODEV; > + } > + > + rk817->hp_ctl_gpio = devm_gpiod_get_optional(dev, "hp-ctl", > + GPIOD_OUT_LOW); > + > + rk817->spk_ctl_gpio = devm_gpiod_get_optional(dev, "spk-ctl", > + GPIOD_OUT_LOW); > + > + ret = of_property_read_u32(node, "spk-mute-delay-ms", > + &rk817->spk_mute_delay); > + if (ret < 0) { > + rk817->spk_mute_delay = 0; > + } > + > + ret = of_property_read_u32(node, "hp-mute-delay-ms", > + &rk817->hp_mute_delay); > + if (ret < 0) { > + rk817->hp_mute_delay = 0; > + } > + > + ret = of_property_read_u32(node, "spk-volume", &rk817->spk_volume); > + if (ret < 0) { > + rk817->spk_volume = OUT_VOLUME; > + } > + if (rk817->spk_volume < 3) > + rk817->spk_volume = 3; > + > + ret = of_property_read_u32(node, "hp-volume", > + &rk817->hp_volume); > + if (ret < 0) { > + rk817->hp_volume = OUT_VOLUME; > + } > + if (rk817->hp_volume < 3) > + rk817->hp_volume = 3; > + > + ret = of_property_read_u32(node, "capture-volume", > + &rk817->capture_volume); > + if (ret < 0) { > + rk817->capture_volume = CAPTURE_VOLUME; > + } > + > + rk817->mic_in_differential = > + of_property_read_bool(node, "mic-in-differential"); > + > + rk817->pdmdata_out_enable = > + of_property_read_bool(node, "pdmdata-out-enable"); > + > + rk817->use_ext_amplifier = > + of_property_read_bool(node, "use-ext-amplifier"); > + > + rk817->adc_for_loopback = > + of_property_read_bool(node, "adc-for-loopback"); you will need DT bindings for all these properties. > + > + return 0; > +} > + > +static const struct regmap_config rk817_codec_regmap_config = { > + .name = "rk817-codec", > + .reg_bits = 8, > + .val_bits = 8, > + .reg_stride = 1, > + .max_register = 0x4f, > + .cache_type = REGCACHE_NONE, > + .volatile_reg = rk817_volatile_register, > + .writeable_reg = rk817_codec_register, > + .readable_reg = rk817_codec_register, > + .reg_defaults = rk817_reg_defaults, > + .num_reg_defaults = ARRAY_SIZE(rk817_reg_defaults), > +}; > + > +static int rk817_platform_probe(struct platform_device *pdev) > +{ > + struct rk808 *rk817 = dev_get_drvdata(pdev->dev.parent); > + struct rk817_codec_priv *rk817_codec_data; > + int ret; > + > + if (!rk817) { > + dev_err(&pdev->dev, "%s : rk817 is NULL\n", __func__); > + return -EINVAL; > + } > + > + rk817_codec_data = devm_kzalloc(&pdev->dev, > + sizeof(struct rk817_codec_priv), > + GFP_KERNEL); > + if (!rk817_codec_data) > + return -ENOMEM; > + > + platform_set_drvdata(pdev, rk817_codec_data); > + > + ret = rk817_codec_parse_dt_property(&pdev->dev, rk817_codec_data); > + if (ret < 0) { > + dev_err(&pdev->dev, "%s() parse device tree property error %d\n", > + __func__, ret); > + goto err_; > + } > + > + rk817_codec_data->regmap = devm_regmap_init_i2c(rk817->i2c, > + &rk817_codec_regmap_config); > + if (IS_ERR(rk817_codec_data->regmap)) { > + ret = PTR_ERR(rk817_codec_data->regmap); > + dev_err(&pdev->dev, "failed to allocate register map: %d\n", > + ret); > + goto err_; > + } > + > + rk817_codec_data->mclk = devm_clk_get(&pdev->dev, "mclk"); > + if (IS_ERR(rk817_codec_data->mclk)) { > + dev_err(&pdev->dev, "Unable to get mclk\n"); > + ret = -ENXIO; > + goto err_; > + } > + > + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_dev_rk817, > + rk817_dai, ARRAY_SIZE(rk817_dai)); > + if (ret < 0) { > + dev_err(&pdev->dev, "%s() register codec error %d\n", > + __func__, ret); > + goto err_; > + } > + > + return 0; > +err_: > + > + return ret; > +} > + > +static int rk817_platform_remove(struct platform_device *pdev) > +{ > + snd_soc_unregister_component(&pdev->dev); humm, that looks like a bug. If you used devm_soc_register_component() in the probe, you don't need to release it here manually? > +++ b/sound/soc/codecs/rk817_codec.h > @@ -0,0 +1,197 @@ > +/* > + * Copyright (c) 2018 Rockchip Electronics Co. Ltd. > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation; either version 2 of the License, or > + * (at your option) any later version. > + * > + * This program is distributed in the hope that it will be useful, > + * but WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the > + * GNU General Public License for more details. remove and use SPDX

1 0

[PATCH] Add audio driver for rk817 upstream
by Chris Morgan 10 Mar '21

10 Mar '21

I'm wondering if you all can help me. I'm trying to get the rk817 codec driver working from Rockchip's BSP kernel sources (GPL per the license) and I'm struggling with a few parts. The first part is I'm not sure if I have my audio paths set up correctly. For example the sinks I have set up are for HPOL and HPOR, and the source is for MIC. While this does work (audio output seems fine) I'm having issues with the GPIO to detect headphone insertion. When I insert headphones I expect the audio to output to the headphones, and when I remove headphones I expect the audio to output to a speaker. Right now I have to manually change the output between the different paths. Additionally, while the codec "technically" has dual channel inputs for the microphone, I'm only using the L channel. Should I have a stereo mixer? Note that I'm using the simple-audio-card to set my paths, widgets, and the gpio pin. They all work, just not seamlessly together. Basically, I'm wanting to know if and how I should set up my audio paths for the speaker and headphones, they use mostly the same pins and only really differ in setting the external amp settings. Additionally, once I set up my paths how do I ensure that the GPIO events will result in switching between the speaker/headphone path? Any help you can provide is appreciated. This is my first attempt at something of this magnitude (for me it's a big step), even if it's just trying to facilitate in getting a vendor's code ready for upstream. Thank you. --- drivers/mfd/rk808.c | 6 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 1148 ++++++++++++++++++++++++++++++++ sound/soc/codecs/rk817_codec.h | 197 ++++++ 5 files changed, 1359 insertions(+) create mode 100644 sound/soc/codecs/rk817_codec.c create mode 100644 sound/soc/codecs/rk817_codec.h diff --git a/drivers/mfd/rk808.c b/drivers/mfd/rk808.c index ad923dd4e007..adb8a7da29db 100644 --- a/drivers/mfd/rk808.c +++ b/drivers/mfd/rk808.c @@ -163,6 +163,12 @@ static const struct mfd_cell rk817s[] = { .num_resources = ARRAY_SIZE(rk817_rtc_resources), .resources = &rk817_rtc_resources[0], }, +#ifdef CONFIG_SND_SOC_RK817 + { + .name = "rk817-codec", + .of_compatible = "rockchip,rk817-codec", + }, +#endif }; static const struct mfd_cell rk818s[] = { diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e4cf14e66a51..d835145ad65e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -155,6 +155,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_PCM512x_I2C imply SND_SOC_PCM512x_SPI imply SND_SOC_RK3328 + imply SND_SOC_RK817 imply SND_SOC_RT274 imply SND_SOC_RT286 imply SND_SOC_RT298 @@ -1059,6 +1060,11 @@ config SND_SOC_RK3328 tristate "Rockchip RK3328 audio CODEC" select REGMAP_MMIO +config SND_SOC_RK817 + tristate "Rockchip RK817 audio CODEC" + depends on MFD_RK808 + select REGMAP_I2C + config SND_SOC_RL6231 tristate default y if SND_SOC_RT5514=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 81357dc62ea0..4e5e9b28775c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -165,6 +165,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rk3328-objs := rk3328_codec.o +snd-soc-rk817-objs := rk817_codec.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1011-objs := rt1011.o @@ -479,6 +480,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o +obj-$(CONFIG_SND_SOC_RK817) += snd-soc-rk817.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c new file mode 100644 index 000000000000..973e2f7562f8 --- /dev/null +++ b/sound/soc/codecs/rk817_codec.c @@ -0,0 +1,1148 @@ +/* + * Copyright (c) 2018 Rockchip Electronics Co. Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/mfd/rk808.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "rk817_codec.h" + +/* For route */ +#define RK817_CODEC_PLAYBACK 1 +#define RK817_CODEC_CAPTURE 2 +#define RK817_CODEC_ALL (RK817_CODEC_PLAYBACK |\ + RK817_CODEC_CAPTURE) + +/* + * DDAC L/R volume setting + * 0db~-95db,0.375db/step,for example: + * 0: 0dB + * 0x0a: -3.75dB + * 0x7d: -46dB + * 0xff: -95dB + */ +#define OUT_VOLUME (0x03) + +/* + * DADC L/R volume setting + * 0db~-95db,0.375db/step,for example: + * 0: 0dB + * 0x0a: -3.75dB + * 0x7d: -46dB + * 0xff: -95dB + */ +#define CAPTURE_VOLUME (0x03) + +#define CODEC_SET_SPK 1 +#define CODEC_SET_HP 2 + +struct rk817_codec_priv { + struct snd_soc_component *component; + struct regmap *regmap; + struct rk808 *rk817; + struct clk *mclk; + + unsigned int stereo_sysclk; + unsigned int rate; + + unsigned int spk_volume; + unsigned int hp_volume; + unsigned int capture_volume; + + bool mic_in_differential; + bool pdmdata_out_enable; + bool use_ext_amplifier; + bool adc_for_loopback; + + long int playback_path; + long int capture_path; + + struct gpio_desc *spk_ctl_gpio; + struct gpio_desc *hp_ctl_gpio; + int spk_mute_delay; + int hp_mute_delay; +}; + +static const struct reg_default rk817_reg_defaults[] = { + { RK817_CODEC_DTOP_VUCTL, 0x003 }, + { RK817_CODEC_DTOP_VUCTIME, 0x00 }, + { RK817_CODEC_DTOP_LPT_SRST, 0x00 }, + { RK817_CODEC_DTOP_DIGEN_CLKE, 0x00 }, + { RK817_CODEC_AREF_RTCFG0, 0x00 }, + { RK817_CODEC_AREF_RTCFG1, 0x06 }, + { RK817_CODEC_AADC_CFG0, 0xc8 }, + { RK817_CODEC_AADC_CFG1, 0x00 }, + { RK817_CODEC_DADC_VOLL, 0x00 }, + { RK817_CODEC_DADC_VOLR, 0x00 }, + { RK817_CODEC_DADC_SR_ACL0, 0x00 }, + { RK817_CODEC_DADC_ALC1, 0x00 }, + { RK817_CODEC_DADC_ALC2, 0x00 }, + { RK817_CODEC_DADC_NG, 0x00 }, + { RK817_CODEC_DADC_HPF, 0x00 }, + { RK817_CODEC_DADC_RVOLL, 0xff }, + { RK817_CODEC_DADC_RVOLR, 0xff }, + { RK817_CODEC_AMIC_CFG0, 0x70 }, + { RK817_CODEC_AMIC_CFG1, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, 0x66 }, + { RK817_CODEC_DMIC_LMT1, 0x00 }, + { RK817_CODEC_DMIC_LMT2, 0x00 }, + { RK817_CODEC_DMIC_NG1, 0x00 }, + { RK817_CODEC_DMIC_NG2, 0x00 }, + { RK817_CODEC_ADAC_CFG0, 0x00 }, + { RK817_CODEC_ADAC_CFG1, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, 0x82 }, + { RK817_CODEC_DDAC_VOLL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, 0x00 }, + { RK817_CODEC_DDAC_LMT1, 0x00 }, + { RK817_CODEC_DDAC_LMT2, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, 0xff }, + { RK817_CODEC_DDAC_RVOLR, 0xff }, + { RK817_CODEC_AHP_ANTI0, 0x00 }, + { RK817_CODEC_AHP_ANTI1, 0x00 }, + { RK817_CODEC_AHP_CFG0, 0xe0 }, + { RK817_CODEC_AHP_CFG1, 0x1f }, + { RK817_CODEC_AHP_CP, 0x09 }, + { RK817_CODEC_ACLASSD_CFG1, 0x69 }, + { RK817_CODEC_ACLASSD_CFG2, 0x44 }, + { RK817_CODEC_APLL_CFG0, 0x04 }, + { RK817_CODEC_APLL_CFG1, 0x00 }, + { RK817_CODEC_APLL_CFG2, 0x30 }, + { RK817_CODEC_APLL_CFG3, 0x19 }, + { RK817_CODEC_APLL_CFG4, 0x65 }, + { RK817_CODEC_APLL_CFG5, 0x01 }, + { RK817_CODEC_DI2S_CKM, 0x01 }, + { RK817_CODEC_DI2S_RSD, 0x00 }, + { RK817_CODEC_DI2S_RXCR1, 0x00 }, + { RK817_CODEC_DI2S_RXCR2, 0x17 }, + { RK817_CODEC_DI2S_RXCMD_TSD, 0x00 }, + { RK817_CODEC_DI2S_TXCR1, 0x00 }, + { RK817_CODEC_DI2S_TXCR2, 0x17 }, + { RK817_CODEC_DI2S_TXCR3_TXCMD, 0x00 }, +}; + +static const DECLARE_TLV_DB_MINMAX(rk817_vol_tlv, -9500, -675); + +static const struct snd_kcontrol_new rk817_dac_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("Playback Volume", RK817_CODEC_DDAC_VOLL, + RK817_CODEC_DDAC_VOLR, 0, 0x12, 0xff, 1, rk817_vol_tlv), + SOC_DOUBLE_R("Record Volume", RK817_CODEC_DADC_VOLL, RK817_CODEC_DADC_VOLR, + 0, 0xFF, 1) +}; + +static bool rk817_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RK817_CODEC_DTOP_LPT_SRST: + return true; + default: + return false; + } +} + +static bool rk817_codec_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RK817_CODEC_DTOP_VUCTL: + case RK817_CODEC_DTOP_VUCTIME: + case RK817_CODEC_DTOP_LPT_SRST: + case RK817_CODEC_DTOP_DIGEN_CLKE: + case RK817_CODEC_AREF_RTCFG0: + case RK817_CODEC_AREF_RTCFG1: + case RK817_CODEC_AADC_CFG0: + case RK817_CODEC_AADC_CFG1: + case RK817_CODEC_DADC_VOLL: + case RK817_CODEC_DADC_VOLR: + case RK817_CODEC_DADC_SR_ACL0: + case RK817_CODEC_DADC_ALC1: + case RK817_CODEC_DADC_ALC2: + case RK817_CODEC_DADC_NG: + case RK817_CODEC_DADC_HPF: + case RK817_CODEC_DADC_RVOLL: + case RK817_CODEC_DADC_RVOLR: + case RK817_CODEC_AMIC_CFG0: + case RK817_CODEC_AMIC_CFG1: + case RK817_CODEC_DMIC_PGA_GAIN: + case RK817_CODEC_DMIC_LMT1: + case RK817_CODEC_DMIC_LMT2: + case RK817_CODEC_DMIC_NG1: + case RK817_CODEC_DMIC_NG2: + case RK817_CODEC_ADAC_CFG0: + case RK817_CODEC_ADAC_CFG1: + case RK817_CODEC_DDAC_POPD_DACST: + case RK817_CODEC_DDAC_VOLL: + case RK817_CODEC_DDAC_VOLR: + case RK817_CODEC_DDAC_SR_LMT0: + case RK817_CODEC_DDAC_LMT1: + case RK817_CODEC_DDAC_LMT2: + case RK817_CODEC_DDAC_MUTE_MIXCTL: + case RK817_CODEC_DDAC_RVOLL: + case RK817_CODEC_DDAC_RVOLR: + case RK817_CODEC_AHP_ANTI0: + case RK817_CODEC_AHP_ANTI1: + case RK817_CODEC_AHP_CFG0: + case RK817_CODEC_AHP_CFG1: + case RK817_CODEC_AHP_CP: + case RK817_CODEC_ACLASSD_CFG1: + case RK817_CODEC_ACLASSD_CFG2: + case RK817_CODEC_APLL_CFG0: + case RK817_CODEC_APLL_CFG1: + case RK817_CODEC_APLL_CFG2: + case RK817_CODEC_APLL_CFG3: + case RK817_CODEC_APLL_CFG4: + case RK817_CODEC_APLL_CFG5: + case RK817_CODEC_DI2S_CKM: + case RK817_CODEC_DI2S_RSD: + case RK817_CODEC_DI2S_RXCR1: + case RK817_CODEC_DI2S_RXCR2: + case RK817_CODEC_DI2S_RXCMD_TSD: + case RK817_CODEC_DI2S_TXCR1: + case RK817_CODEC_DI2S_TXCR2: + case RK817_CODEC_DI2S_TXCR3_TXCMD: + return true; + default: + return false; + } +} + +static int rk817_codec_ctl_gpio(struct rk817_codec_priv *rk817, + int gpio, int level) +{ + if ((gpio & CODEC_SET_SPK) && + rk817->spk_ctl_gpio) { + gpiod_set_value(rk817->spk_ctl_gpio, level); + msleep(rk817->spk_mute_delay); + } + + if ((gpio & CODEC_SET_HP) && + rk817->hp_ctl_gpio) { + gpiod_set_value(rk817->hp_ctl_gpio, level); + msleep(rk817->hp_mute_delay); + } + + return 0; +} + +static int rk817_reset(struct snd_soc_component *component) +{ + snd_soc_component_write(component, RK817_CODEC_DTOP_LPT_SRST, 0x40); + snd_soc_component_write(component, RK817_CODEC_DDAC_POPD_DACST, 0x02); + snd_soc_component_write(component, RK817_CODEC_DTOP_DIGEN_CLKE, 0x0f); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG0, 0x04); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG1, 0x58); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG2, 0x2d); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG3, 0x0c); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG4, 0xa5); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG5, 0x00); + snd_soc_component_write(component, RK817_CODEC_DTOP_DIGEN_CLKE, 0x00); + + return 0; +} + +static struct rk817_reg_val_typ playback_power_up_list[] = { + {RK817_CODEC_AREF_RTCFG1, 0x40}, + {RK817_CODEC_DDAC_POPD_DACST, 0x02}, + {RK817_CODEC_DDAC_SR_LMT0, 0x02}, + /* APLL */ + {RK817_CODEC_APLL_CFG0, 0x04}, + {RK817_CODEC_APLL_CFG1, 0x58}, + {RK817_CODEC_APLL_CFG2, 0x2d}, + {RK817_CODEC_APLL_CFG3, 0x0c}, + {RK817_CODEC_APLL_CFG4, 0xa5}, + {RK817_CODEC_APLL_CFG5, 0x00}, + + {RK817_CODEC_DI2S_RXCMD_TSD, 0x00}, + {RK817_CODEC_DI2S_RSD, 0x00}, + {RK817_CODEC_DI2S_RXCR1, 0x00}, + {RK817_CODEC_DI2S_RXCMD_TSD, 0x20}, + {RK817_CODEC_DTOP_VUCTIME, 0xf4}, + {RK817_CODEC_DDAC_MUTE_MIXCTL, 0x00}, + + {RK817_CODEC_DDAC_VOLL, 0x0a}, + {RK817_CODEC_DDAC_VOLR, 0x0a}, +}; + +#define RK817_CODEC_PLAYBACK_POWER_UP_LIST_LEN \ + ARRAY_SIZE(playback_power_up_list) + +static struct rk817_reg_val_typ playback_power_down_list[] = { + {RK817_CODEC_DDAC_MUTE_MIXCTL, 0x01}, + {RK817_CODEC_ADAC_CFG1, 0x0f}, + /* HP */ + {RK817_CODEC_AHP_CFG0, 0xe0}, + {RK817_CODEC_AHP_CP, 0x09}, + /* SPK */ + {RK817_CODEC_ACLASSD_CFG1, 0x69}, +}; + +#define RK817_CODEC_PLAYBACK_POWER_DOWN_LIST_LEN \ + ARRAY_SIZE(playback_power_down_list) + +static struct rk817_reg_val_typ capture_power_up_list[] = { + {RK817_CODEC_AREF_RTCFG1, 0x40}, + {RK817_CODEC_DDAC_SR_LMT0, 0x02}, + {RK817_CODEC_DADC_SR_ACL0, 0x02}, + + {RK817_CODEC_APLL_CFG0, 0x04}, + {RK817_CODEC_APLL_CFG1, 0x58}, + {RK817_CODEC_APLL_CFG2, 0x2d}, + {RK817_CODEC_APLL_CFG3, 0x0c}, + {RK817_CODEC_APLL_CFG4, 0xa5}, + {RK817_CODEC_APLL_CFG5, 0x00}, + + {RK817_CODEC_DI2S_RSD, 0x00}, + {RK817_CODEC_DI2S_RXCR1, 0x00}, + {RK817_CODEC_DI2S_RXCMD_TSD, 0x20}, + {RK817_CODEC_DTOP_VUCTIME, 0xf4}, + + {RK817_CODEC_DDAC_MUTE_MIXCTL, 0x00}, + {RK817_CODEC_AADC_CFG0, 0x08}, + {RK817_CODEC_AMIC_CFG0, 0x0f}, + {RK817_CODEC_DI2S_TXCR3_TXCMD, 0x88}, + {RK817_CODEC_DDAC_POPD_DACST, 0x02}, + /* 0x29: -18db to 27db */ + {RK817_CODEC_DMIC_PGA_GAIN, 0x99}, +}; + +#define RK817_CODEC_CAPTURE_POWER_UP_LIST_LEN \ + ARRAY_SIZE(capture_power_up_list) + +static struct rk817_reg_val_typ capture_power_down_list[] = { + {RK817_CODEC_AADC_CFG0, 0xc8}, + {RK817_CODEC_AMIC_CFG0, 0x70}, +}; + +#define RK817_CODEC_CAPTURE_POWER_DOWN_LIST_LEN \ + ARRAY_SIZE(capture_power_down_list) + +static int rk817_codec_power_up(struct snd_soc_component *component, int type) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + int i; + + if (type & RK817_CODEC_PLAYBACK) { + snd_soc_component_update_bits(component, + RK817_CODEC_DTOP_DIGEN_CLKE, + DAC_DIG_CLK_MASK, DAC_DIG_CLK_EN); + for (i = 0; i < RK817_CODEC_PLAYBACK_POWER_UP_LIST_LEN; i++) { + snd_soc_component_write(component, + playback_power_up_list[i].reg, + playback_power_up_list[i].value); + } + } + + if (type & RK817_CODEC_CAPTURE) { + snd_soc_component_update_bits(component, + RK817_CODEC_DTOP_DIGEN_CLKE, + ADC_DIG_CLK_MASK, + ADC_DIG_CLK_EN); + for (i = 0; i < RK817_CODEC_CAPTURE_POWER_UP_LIST_LEN; i++) { + snd_soc_component_write(component, + capture_power_up_list[i].reg, + capture_power_up_list[i].value); + } + + if (rk817->mic_in_differential) + snd_soc_component_update_bits(component, + RK817_CODEC_AMIC_CFG0, + MIC_DIFF_MASK, MIC_DIFF_EN); + else + snd_soc_component_update_bits(component, + RK817_CODEC_AMIC_CFG0, + MIC_DIFF_MASK, + MIC_DIFF_DIS); + + if (rk817->pdmdata_out_enable) + snd_soc_component_update_bits(component, + RK817_CODEC_DI2S_CKM, + PDM_EN_MASK, + PDM_EN_ENABLE); + + snd_soc_component_write(component, RK817_CODEC_DADC_VOLL, + rk817->capture_volume); + snd_soc_component_write(component, RK817_CODEC_DADC_VOLR, + rk817->capture_volume); + } + + return 0; +} + +static int rk817_codec_power_down(struct snd_soc_component *component, int type) +{ + int i; + + /* mute output for pop noise */ + if (type & RK817_CODEC_PLAYBACK) { + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_ENABLE, DACMT_ENABLE); + } + + if (type & RK817_CODEC_CAPTURE) { + for (i = 0; i < RK817_CODEC_CAPTURE_POWER_DOWN_LIST_LEN; i++) { + snd_soc_component_write(component, + capture_power_down_list[i].reg, + capture_power_down_list[i].value); + } + snd_soc_component_update_bits(component, RK817_CODEC_DTOP_DIGEN_CLKE, + ADC_DIG_CLK_MASK, ADC_DIG_CLK_DIS); + } + + if (type & RK817_CODEC_PLAYBACK) { + for (i = 0; i < RK817_CODEC_PLAYBACK_POWER_DOWN_LIST_LEN; i++) { + snd_soc_component_write(component, + playback_power_down_list[i].reg, + playback_power_down_list[i].value); + } + snd_soc_component_update_bits(component, + RK817_CODEC_DTOP_DIGEN_CLKE, + DAC_DIG_CLK_MASK, DAC_DIG_CLK_DIS); + } + + if (type == RK817_CODEC_ALL) { + for (i = 0; i < RK817_CODEC_PLAYBACK_POWER_DOWN_LIST_LEN; i++) { + snd_soc_component_write(component, + playback_power_down_list[i].reg, + playback_power_down_list[i].value); + } + for (i = 0; i < RK817_CODEC_CAPTURE_POWER_DOWN_LIST_LEN; i++) { + snd_soc_component_write(component, + capture_power_down_list[i].reg, + capture_power_down_list[i].value); + } + snd_soc_component_write(component, RK817_CODEC_DTOP_DIGEN_CLKE, 0x00); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG5, 0x01); + snd_soc_component_write(component, RK817_CODEC_AREF_RTCFG1, 0x06); + } + + return 0; +} + +/* For tiny alsa playback/capture path*/ +static const char * const rk817_playback_path_mode[] = { + "OFF", "SPK", "HP", "SPK_HP"}; + +static const char * const rk817_capture_path_mode[] = { + "MIC OFF", "MIC"}; + +static SOC_ENUM_SINGLE_DECL(rk817_playback_path_type, + 0, 0, rk817_playback_path_mode); + +static SOC_ENUM_SINGLE_DECL(rk817_capture_path_type, + 0, 0, rk817_capture_path_mode); + +static int rk817_playback_path_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = rk817->playback_path; + + return 0; +} + +static int rk817_playback_path_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + long int pre_path; + + if (rk817->playback_path == ucontrol->value.integer.value[0]) { + return 0; + } + + pre_path = rk817->playback_path; + rk817->playback_path = ucontrol->value.integer.value[0]; + + if (rk817->playback_path != OFF) + clk_prepare_enable(rk817->mclk); + else + clk_disable_unprepare(rk817->mclk); + + switch (rk817->playback_path) { + case OFF: + if (pre_path != OFF && pre_path != HP_PATH) { + rk817_codec_power_down(component, RK817_CODEC_PLAYBACK); + if (rk817->capture_path == 0) + rk817_codec_power_down(component, RK817_CODEC_ALL); + } + break; + case SPK_PATH: + if (pre_path == OFF) + rk817_codec_power_up(component, RK817_CODEC_PLAYBACK); + if (!rk817->use_ext_amplifier) { + /* power on dac ibias/l/r */ + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, + PWD_DACBIAS_ON | PWD_DACD_ON | + PWD_DACL_DOWN | PWD_DACR_DOWN); + /* CLASS D mode, combine LR channels */ + snd_soc_component_write(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + 0x10); + /* CLASS D enable */ + snd_soc_component_write(component, + RK817_CODEC_ACLASSD_CFG1, + 0xa5); + /* restart CLASS D, OCPP/N */ + snd_soc_component_write(component, + RK817_CODEC_ACLASSD_CFG2, + 0xf7); + } else { + /* HP_CP_EN , CP 2.3V */ + snd_soc_component_write(component, RK817_CODEC_AHP_CP, + 0x11); + /* power on HP two stage opamp ,HP amplitude 0db */ + snd_soc_component_write(component, RK817_CODEC_AHP_CFG0, + 0x80); + /* power on dac ibias/l/r */ + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, + PWD_DACBIAS_ON | PWD_DACD_DOWN | + PWD_DACL_ON | PWD_DACR_ON); + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_ENABLE, DACMT_DISABLE); + } + snd_soc_component_write(component, RK817_CODEC_DDAC_VOLL, + rk817->spk_volume); + snd_soc_component_write(component, RK817_CODEC_DDAC_VOLR, + rk817->spk_volume); + break; + case HP_PATH: + if (pre_path == OFF) + rk817_codec_power_up(component, RK817_CODEC_PLAYBACK); + /* HP_CP_EN , CP 2.3V */ + snd_soc_component_write(component, RK817_CODEC_AHP_CP, 0x11); + /* power on HP two stage opamp ,HP amplitude 0db */ + snd_soc_component_write(component, RK817_CODEC_AHP_CFG0, 0x80); + /* power on dac ibias/l/r */ + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, + PWD_DACBIAS_ON | PWD_DACD_DOWN | + PWD_DACL_ON | PWD_DACR_ON); + /* CLASS D mode disable, split LR channels */ + snd_soc_component_write(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + 0x00); + + snd_soc_component_write(component, RK817_CODEC_DDAC_VOLL, + rk817->hp_volume); + snd_soc_component_write(component, RK817_CODEC_DDAC_VOLR, + rk817->hp_volume); + break; + case SPK_HP: + if (pre_path == OFF) + rk817_codec_power_up(component, RK817_CODEC_PLAYBACK); + + /* HP_CP_EN , CP 2.3V */ + snd_soc_component_write(component, RK817_CODEC_AHP_CP, 0x11); + /* power on HP two stage opamp ,HP amplitude 0db */ + snd_soc_component_write(component, RK817_CODEC_AHP_CFG0, 0x80); + + /* power on dac ibias/l/r */ + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, + PWD_DACBIAS_ON | PWD_DACD_ON | + PWD_DACL_ON | PWD_DACR_ON); + + if (!rk817->use_ext_amplifier) { + /* CLASS D mode, combine LR channels */ + snd_soc_component_write(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + 0x10); + /* CLASS D enable */ + snd_soc_component_write(component, + RK817_CODEC_ACLASSD_CFG1, + 0xa5); + /* restart CLASS D, OCPP/N */ + snd_soc_component_write(component, + RK817_CODEC_ACLASSD_CFG2, + 0xf7); + } + + snd_soc_component_write(component, RK817_CODEC_DDAC_VOLL, + rk817->hp_volume); + snd_soc_component_write(component, RK817_CODEC_DDAC_VOLR, + rk817->hp_volume); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int rk817_capture_path_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "%s:capture_path %ld\n", __func__, rk817->capture_path); + ucontrol->value.integer.value[0] = rk817->capture_path; + return 0; +} + +static int rk817_capture_path_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + long int pre_path; + + if (rk817->capture_path == ucontrol->value.integer.value[0]) { + dev_dbg(component->dev, "%s:capture_path is not changed!\n", + __func__); + return 0; + } + + pre_path = rk817->capture_path; + rk817->capture_path = ucontrol->value.integer.value[0]; + + if (rk817->capture_path != MIC_OFF) + clk_prepare_enable(rk817->mclk); + else + clk_disable_unprepare(rk817->mclk); + + switch (rk817->capture_path) { + case MIC_OFF: + if (pre_path != MIC_OFF) + rk817_codec_power_down(component, RK817_CODEC_CAPTURE); + break; + case MIC: + if (pre_path == MIC_OFF) + rk817_codec_power_up(component, RK817_CODEC_CAPTURE); + + if (rk817->adc_for_loopback) { + /* don't need to gain when adc use for loopback */ + snd_soc_component_update_bits(component, + RK817_CODEC_AMIC_CFG0, + 0xf, + 0x0); + snd_soc_component_write(component, + RK817_CODEC_DMIC_PGA_GAIN, + 0x66); + snd_soc_component_write(component, + RK817_CODEC_DADC_VOLL, + 0x00); + snd_soc_component_write(component, + RK817_CODEC_DADC_VOLR, + 0x00); + break; + } + if (!rk817->mic_in_differential) { + snd_soc_component_write(component, + RK817_CODEC_DADC_VOLR, + 0xff); + snd_soc_component_update_bits(component, + RK817_CODEC_AADC_CFG0, + ADC_R_PWD_MASK, + ADC_R_PWD_EN); + snd_soc_component_update_bits(component, + RK817_CODEC_AMIC_CFG0, + PWD_PGA_R_MASK, + PWD_PGA_R_EN); + } + break; + default: + return -EINVAL; + } + + return 0; +} + +static struct snd_kcontrol_new rk817_snd_path_controls[] = { + SOC_ENUM_EXT("Playback Path", rk817_playback_path_type, + rk817_playback_path_get, rk817_playback_path_put), + + SOC_ENUM_EXT("Capture Path", rk817_capture_path_type, + rk817_capture_path_get, rk817_capture_path_put), +}; + +static const struct snd_soc_dapm_widget rk817_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static int rk817_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 rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + rk817->stereo_sysclk = freq; + + return 0; +} + +static int rk817_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + unsigned int i2s_mst = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + i2s_mst |= RK817_I2S_MODE_SLV; + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2s_mst |= RK817_I2S_MODE_MST; + break; + default: + dev_err(component->dev, "%s : set master mask failed!\n", __func__); + return -EINVAL; + } + + snd_soc_component_update_bits(component, RK817_CODEC_DI2S_CKM, + RK817_I2S_MODE_MASK, i2s_mst); + + return 0; +} + +static int rk817_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; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_16BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_16BITS); + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_24BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_24BITS); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int rk817_digital_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_component *component = dai->component; + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + if (mute) + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_ENABLE, DACMT_ENABLE); + else + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_ENABLE, DACMT_DISABLE); + + if (mute) { + rk817_codec_ctl_gpio(rk817, CODEC_SET_SPK, 0); + rk817_codec_ctl_gpio(rk817, CODEC_SET_HP, 0); + } else { + switch (rk817->playback_path) { + case SPK_PATH: + rk817_codec_ctl_gpio(rk817, CODEC_SET_SPK, 1); + rk817_codec_ctl_gpio(rk817, CODEC_SET_HP, 0); + break; + case HP_PATH: + rk817_codec_ctl_gpio(rk817, CODEC_SET_SPK, 0); + rk817_codec_ctl_gpio(rk817, CODEC_SET_HP, 1); + break; + case SPK_HP: + rk817_codec_ctl_gpio(rk817, CODEC_SET_SPK, 1); + rk817_codec_ctl_gpio(rk817, CODEC_SET_HP, 1); + break; + default: + break; + } + } + + return 0; +} + +#define RK817_PLAYBACK_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_CAPTURE_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops rk817_dai_ops = { + .hw_params = rk817_hw_params, + .set_fmt = rk817_set_dai_fmt, + .set_sysclk = rk817_set_dai_sysclk, + .mute_stream = rk817_digital_mute, + .no_capture_mute = 1, +}; + +static struct snd_soc_dai_driver rk817_dai[] = { + { + .name = "rk817-hifi", + .id = RK817_HIFI, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = RK817_PLAYBACK_RATES, + .formats = RK817_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RK817_CAPTURE_RATES, + .formats = RK817_FORMATS, + }, + .ops = &rk817_dai_ops, + }, +}; + +static int rk817_suspend(struct snd_soc_component *component) +{ + rk817_codec_power_down(component, RK817_CODEC_ALL); + return 0; +} + +static int rk817_resume(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + rk817_codec_power_up(component, RK817_CODEC_ALL); + + switch (rk817->playback_path) { + case OFF: + break; + case SPK_PATH: + if (!rk817->use_ext_amplifier) { + /* power on dac ibias/l/r */ + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, + PWD_DACBIAS_ON | PWD_DACD_ON | + PWD_DACL_ON | PWD_DACR_ON); + /* CLASS D mode */ + snd_soc_component_write(component, RK817_CODEC_DDAC_MUTE_MIXCTL, 0x10); + /* CLASS D enable */ + snd_soc_component_write(component, RK817_CODEC_ACLASSD_CFG1, 0xa5); + /* restart CLASS D, OCPP/N */ + snd_soc_component_write(component, RK817_CODEC_ACLASSD_CFG2, 0xc4); + } else { + /* HP_CP_EN , CP 2.3V */ + snd_soc_component_write(component, RK817_CODEC_AHP_CP, 0x11); + /* power on HP two stage opamp ,HP amplitude 0db */ + snd_soc_component_write(component, RK817_CODEC_AHP_CFG0, 0x80); + /* power on dac ibias/l/r */ + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, + PWD_DACBIAS_ON | PWD_DACD_DOWN | + PWD_DACL_ON | PWD_DACR_ON); + snd_soc_component_update_bits(component, RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_ENABLE, DACMT_DISABLE); + } + break; + case HP_PATH: + /* HP_CP_EN , CP 2.3V */ + snd_soc_component_write(component, RK817_CODEC_AHP_CP, 0x11); + /* power on HP two stage opamp ,HP amplitude 0db */ + snd_soc_component_write(component, RK817_CODEC_AHP_CFG0, 0x80); + /* power on dac ibias/l/r */ + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, + PWD_DACBIAS_ON | PWD_DACD_DOWN | + PWD_DACL_ON | PWD_DACR_ON); + snd_soc_component_update_bits(component, RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_ENABLE, DACMT_DISABLE); + break; + case SPK_HP: + /* HP_CP_EN , CP 2.3V */ + snd_soc_component_write(component, RK817_CODEC_AHP_CP, 0x11); + /* power on HP two stage opamp ,HP amplitude 0db */ + snd_soc_component_write(component, RK817_CODEC_AHP_CFG0, 0x80); + + /* power on dac ibias/l/r */ + snd_soc_component_write(component, RK817_CODEC_ADAC_CFG1, + PWD_DACBIAS_ON | PWD_DACD_ON | + PWD_DACL_ON | PWD_DACR_ON); + + if (!rk817->use_ext_amplifier) { + /* CLASS D mode */ + snd_soc_component_write(component, RK817_CODEC_DDAC_MUTE_MIXCTL, 0x10); + /* CLASS D enable */ + snd_soc_component_write(component, RK817_CODEC_ACLASSD_CFG1, 0xa5); + /* restart CLASS D, OCPP/N */ + snd_soc_component_write(component, RK817_CODEC_ACLASSD_CFG2, 0xc4); + } + break; + default: + return -EINVAL; + } + + return 0; +} + +static int rk817_probe(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + if (!rk817) { + dev_err(component->dev, "%s : rk817 priv is NULL!\n", + __func__); + return -EINVAL; + } + snd_soc_component_init_regmap(component, rk817->regmap); + rk817->component = component; + rk817->playback_path = OFF; + rk817->capture_path = MIC_OFF; + + rk817_reset(component); + snd_soc_add_component_controls(component, rk817_snd_path_controls, + ARRAY_SIZE(rk817_snd_path_controls)); + + snd_soc_dapm_new_controls(dapm, rk817_dapm_widgets, + ARRAY_SIZE(rk817_dapm_widgets)); + return 0; +} + +/* power down chip */ +static void rk817_remove(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + if (!rk817) { + dev_err(component->dev, "%s : rk817 is NULL\n", __func__); + return; + } + + rk817_codec_power_down(component, RK817_CODEC_ALL); + snd_soc_component_exit_regmap(component); + mdelay(10); + +} + +static const struct snd_soc_component_driver soc_codec_dev_rk817 = { + .probe = rk817_probe, + .remove = rk817_remove, + .suspend = rk817_suspend, + .resume = rk817_resume, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, + .controls = rk817_dac_controls, + .num_controls = ARRAY_SIZE(rk817_dac_controls), +}; + +static int rk817_codec_parse_dt_property(struct device *dev, + struct rk817_codec_priv *rk817) +{ + struct device_node *node = dev->parent->of_node; + int ret; + + if (!node) { + dev_err(dev, "%s() dev->parent->of_node is NULL\n", + __func__); + return -ENODEV; + } + + node = of_get_child_by_name(dev->parent->of_node, "codec"); + if (!node) { + dev_err(dev, "%s() Can not get child: codec\n", + __func__); + return -ENODEV; + } + + rk817->hp_ctl_gpio = devm_gpiod_get_optional(dev, "hp-ctl", + GPIOD_OUT_LOW); + + rk817->spk_ctl_gpio = devm_gpiod_get_optional(dev, "spk-ctl", + GPIOD_OUT_LOW); + + ret = of_property_read_u32(node, "spk-mute-delay-ms", + &rk817->spk_mute_delay); + if (ret < 0) { + rk817->spk_mute_delay = 0; + } + + ret = of_property_read_u32(node, "hp-mute-delay-ms", + &rk817->hp_mute_delay); + if (ret < 0) { + rk817->hp_mute_delay = 0; + } + + ret = of_property_read_u32(node, "spk-volume", &rk817->spk_volume); + if (ret < 0) { + rk817->spk_volume = OUT_VOLUME; + } + if (rk817->spk_volume < 3) + rk817->spk_volume = 3; + + ret = of_property_read_u32(node, "hp-volume", + &rk817->hp_volume); + if (ret < 0) { + rk817->hp_volume = OUT_VOLUME; + } + if (rk817->hp_volume < 3) + rk817->hp_volume = 3; + + ret = of_property_read_u32(node, "capture-volume", + &rk817->capture_volume); + if (ret < 0) { + rk817->capture_volume = CAPTURE_VOLUME; + } + + rk817->mic_in_differential = + of_property_read_bool(node, "mic-in-differential"); + + rk817->pdmdata_out_enable = + of_property_read_bool(node, "pdmdata-out-enable"); + + rk817->use_ext_amplifier = + of_property_read_bool(node, "use-ext-amplifier"); + + rk817->adc_for_loopback = + of_property_read_bool(node, "adc-for-loopback"); + + return 0; +} + +static const struct regmap_config rk817_codec_regmap_config = { + .name = "rk817-codec", + .reg_bits = 8, + .val_bits = 8, + .reg_stride = 1, + .max_register = 0x4f, + .cache_type = REGCACHE_NONE, + .volatile_reg = rk817_volatile_register, + .writeable_reg = rk817_codec_register, + .readable_reg = rk817_codec_register, + .reg_defaults = rk817_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rk817_reg_defaults), +}; + +static int rk817_platform_probe(struct platform_device *pdev) +{ + struct rk808 *rk817 = dev_get_drvdata(pdev->dev.parent); + struct rk817_codec_priv *rk817_codec_data; + int ret; + + if (!rk817) { + dev_err(&pdev->dev, "%s : rk817 is NULL\n", __func__); + return -EINVAL; + } + + rk817_codec_data = devm_kzalloc(&pdev->dev, + sizeof(struct rk817_codec_priv), + GFP_KERNEL); + if (!rk817_codec_data) + return -ENOMEM; + + platform_set_drvdata(pdev, rk817_codec_data); + + ret = rk817_codec_parse_dt_property(&pdev->dev, rk817_codec_data); + if (ret < 0) { + dev_err(&pdev->dev, "%s() parse device tree property error %d\n", + __func__, ret); + goto err_; + } + + rk817_codec_data->regmap = devm_regmap_init_i2c(rk817->i2c, + &rk817_codec_regmap_config); + if (IS_ERR(rk817_codec_data->regmap)) { + ret = PTR_ERR(rk817_codec_data->regmap); + dev_err(&pdev->dev, "failed to allocate register map: %d\n", + ret); + goto err_; + } + + rk817_codec_data->mclk = devm_clk_get(&pdev->dev, "mclk"); + if (IS_ERR(rk817_codec_data->mclk)) { + dev_err(&pdev->dev, "Unable to get mclk\n"); + ret = -ENXIO; + goto err_; + } + + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_dev_rk817, + rk817_dai, ARRAY_SIZE(rk817_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "%s() register codec error %d\n", + __func__, ret); + goto err_; + } + + return 0; +err_: + + return ret; +} + +static int rk817_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_component(&pdev->dev); + + return 0; +} + +static void rk817_platform_shutdown(struct platform_device *pdev) +{ + struct rk817_codec_priv *rk817 = dev_get_drvdata(&pdev->dev); + + if (rk817 && rk817->component) + rk817_codec_power_down(rk817->component, RK817_CODEC_ALL); +} + +static const struct of_device_id rk817_codec_dt_ids[] = { + { .compatible = "rockchip,rk817-codec" }, + {}, +}; +MODULE_DEVICE_TABLE(of, rk817_codec_dt_ids); + +static struct platform_driver rk817_codec_driver = { + .driver = { + .name = "rk817-codec", + .of_match_table = rk817_codec_dt_ids, + }, + .probe = rk817_platform_probe, + .remove = rk817_platform_remove, + .shutdown = rk817_platform_shutdown, +}; + +module_platform_driver(rk817_codec_driver); + +MODULE_DESCRIPTION("ASoC RK817 codec driver"); +MODULE_AUTHOR("binyuan <kevan.lan(a)rock-chips.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rk817_codec.h b/sound/soc/codecs/rk817_codec.h new file mode 100644 index 000000000000..3f13eade80b3 --- /dev/null +++ b/sound/soc/codecs/rk817_codec.h @@ -0,0 +1,197 @@ +/* + * Copyright (c) 2018 Rockchip Electronics Co. Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef __RK817_CODEC_H__ +#define __RK817_CODEC_H__ + +/* codec register */ +#define RK817_CODEC_BASE 0x0000 + +#define RK817_CODEC_DTOP_VUCTL (RK817_CODEC_BASE + 0x12) +#define RK817_CODEC_DTOP_VUCTIME (RK817_CODEC_BASE + 0x13) +#define RK817_CODEC_DTOP_LPT_SRST (RK817_CODEC_BASE + 0x14) +#define RK817_CODEC_DTOP_DIGEN_CLKE (RK817_CODEC_BASE + 0x15) +#define RK817_CODEC_AREF_RTCFG0 (RK817_CODEC_BASE + 0x16) +#define RK817_CODEC_AREF_RTCFG1 (RK817_CODEC_BASE + 0x17) +#define RK817_CODEC_AADC_CFG0 (RK817_CODEC_BASE + 0x18) +#define RK817_CODEC_AADC_CFG1 (RK817_CODEC_BASE + 0x19) +#define RK817_CODEC_DADC_VOLL (RK817_CODEC_BASE + 0x1a) +#define RK817_CODEC_DADC_VOLR (RK817_CODEC_BASE + 0x1b) +#define RK817_CODEC_DADC_SR_ACL0 (RK817_CODEC_BASE + 0x1e) +#define RK817_CODEC_DADC_ALC1 (RK817_CODEC_BASE + 0x1f) +#define RK817_CODEC_DADC_ALC2 (RK817_CODEC_BASE + 0x20) +#define RK817_CODEC_DADC_NG (RK817_CODEC_BASE + 0x21) +#define RK817_CODEC_DADC_HPF (RK817_CODEC_BASE + 0x22) +#define RK817_CODEC_DADC_RVOLL (RK817_CODEC_BASE + 0x23) +#define RK817_CODEC_DADC_RVOLR (RK817_CODEC_BASE + 0x24) +#define RK817_CODEC_AMIC_CFG0 (RK817_CODEC_BASE + 0x27) +#define RK817_CODEC_AMIC_CFG1 (RK817_CODEC_BASE + 0x28) +#define RK817_CODEC_DMIC_PGA_GAIN (RK817_CODEC_BASE + 0x29) +#define RK817_CODEC_DMIC_LMT1 (RK817_CODEC_BASE + 0x2a) +#define RK817_CODEC_DMIC_LMT2 (RK817_CODEC_BASE + 0x2b) +#define RK817_CODEC_DMIC_NG1 (RK817_CODEC_BASE + 0x2c) +#define RK817_CODEC_DMIC_NG2 (RK817_CODEC_BASE + 0x2d) +#define RK817_CODEC_ADAC_CFG0 (RK817_CODEC_BASE + 0x2e) +#define RK817_CODEC_ADAC_CFG1 (RK817_CODEC_BASE + 0x2f) +#define RK817_CODEC_DDAC_POPD_DACST (RK817_CODEC_BASE + 0x30) +#define RK817_CODEC_DDAC_VOLL (RK817_CODEC_BASE + 0x31) +#define RK817_CODEC_DDAC_VOLR (RK817_CODEC_BASE + 0x32) +#define RK817_CODEC_DDAC_SR_LMT0 (RK817_CODEC_BASE + 0x35) +#define RK817_CODEC_DDAC_LMT1 (RK817_CODEC_BASE + 0x36) +#define RK817_CODEC_DDAC_LMT2 (RK817_CODEC_BASE + 0x37) +#define RK817_CODEC_DDAC_MUTE_MIXCTL (RK817_CODEC_BASE + 0x38) +#define RK817_CODEC_DDAC_RVOLL (RK817_CODEC_BASE + 0x39) +#define RK817_CODEC_DDAC_RVOLR (RK817_CODEC_BASE + 0x3a) +#define RK817_CODEC_AHP_ANTI0 (RK817_CODEC_BASE + 0x3b) +#define RK817_CODEC_AHP_ANTI1 (RK817_CODEC_BASE + 0x3c) +#define RK817_CODEC_AHP_CFG0 (RK817_CODEC_BASE + 0x3d) +#define RK817_CODEC_AHP_CFG1 (RK817_CODEC_BASE + 0x3e) +#define RK817_CODEC_AHP_CP (RK817_CODEC_BASE + 0x3f) +#define RK817_CODEC_ACLASSD_CFG1 (RK817_CODEC_BASE + 0x40) +#define RK817_CODEC_ACLASSD_CFG2 (RK817_CODEC_BASE + 0x41) +#define RK817_CODEC_APLL_CFG0 (RK817_CODEC_BASE + 0x42) +#define RK817_CODEC_APLL_CFG1 (RK817_CODEC_BASE + 0x43) +#define RK817_CODEC_APLL_CFG2 (RK817_CODEC_BASE + 0x44) +#define RK817_CODEC_APLL_CFG3 (RK817_CODEC_BASE + 0x45) +#define RK817_CODEC_APLL_CFG4 (RK817_CODEC_BASE + 0x46) +#define RK817_CODEC_APLL_CFG5 (RK817_CODEC_BASE + 0x47) +#define RK817_CODEC_DI2S_CKM (RK817_CODEC_BASE + 0x48) +#define RK817_CODEC_DI2S_RSD (RK817_CODEC_BASE + 0x49) +#define RK817_CODEC_DI2S_RXCR1 (RK817_CODEC_BASE + 0x4a) +#define RK817_CODEC_DI2S_RXCR2 (RK817_CODEC_BASE + 0x4b) +#define RK817_CODEC_DI2S_RXCMD_TSD (RK817_CODEC_BASE + 0x4c) +#define RK817_CODEC_DI2S_TXCR1 (RK817_CODEC_BASE + 0x4d) +#define RK817_CODEC_DI2S_TXCR2 (RK817_CODEC_BASE + 0x4e) +#define RK817_CODEC_DI2S_TXCR3_TXCMD (RK817_CODEC_BASE + 0x4f) + +/* RK817_CODEC_DTOP_DIGEN_CLKE */ +#define ADC_DIG_CLK_MASK (0xf << 4) +#define ADC_DIG_CLK_SFT 4 +#define ADC_DIG_CLK_DIS (0x0 << 4) +#define ADC_DIG_CLK_EN (0xf << 4) + +#define DAC_DIG_CLK_MASK (0xf << 0) +#define DAC_DIG_CLK_SFT 0 +#define DAC_DIG_CLK_DIS (0x0 << 0) +#define DAC_DIG_CLK_EN (0xf << 0) + +/* RK817_CODEC_APLL_CFG5 */ +#define PLL_PW_DOWN (0x01 << 0) +#define PLL_PW_UP (0x00 << 0) + +/* RK817_CODEC_DI2S_CKM */ +#define PDM_EN_MASK (0x1 << 3) +#define PDM_EN_SFT 3 +#define PDM_EN_DISABLE (0x0 << 3) +#define PDM_EN_ENABLE (0x1 << 3) + +#define SCK_EN_ENABLE (0x1 << 2) +#define SCK_EN_DISABLE (0x0 << 2) + +#define RK817_I2S_MODE_MASK (0x1 << 0) +#define RK817_I2S_MODE_SFT 0 +#define RK817_I2S_MODE_MST (0x1 << 0) +#define RK817_I2S_MODE_SLV (0x0 << 0) + +/* RK817_CODEC_DDAC_MUTE_MIXCTL */ +#define DACMT_ENABLE (0x1 << 0) +#define DACMT_DISABLE (0x0 << 0) + +/* RK817_CODEC_DI2S_RXCR2 */ +#define VDW_RX_24BITS (0x17) +#define VDW_RX_16BITS (0x0f) +/* RK817_CODEC_DI2S_TXCR2 */ +#define VDW_TX_24BITS (0x17) +#define VDW_TX_16BITS (0x0f) + +/* RK817_CODEC_AHP_CFG1 */ +#define HP_ANTIPOP_ENABLE (0x1 << 4) +#define HP_ANTIPOP_DISABLE (0x0 << 4) + +/* RK817_CODEC_ADAC_CFG1 */ +#define PWD_DACBIAS_MASK (0x1 << 3) +#define PWD_DACBIAS_SFT 3 +#define PWD_DACBIAS_DOWN (0x1 << 3) +#define PWD_DACBIAS_ON (0x0 << 3) + +#define PWD_DACD_MASK (0x1 << 2) +#define PWD_DACD_SFT 2 +#define PWD_DACD_DOWN (0x1 << 2) +#define PWD_DACD_ON (0x0 << 2) + +#define PWD_DACL_MASK (0x1 << 1) +#define PWD_DACL_SFT 1 +#define PWD_DACL_DOWN (0x1 << 1) +#define PWD_DACL_ON (0x0 << 1) + +#define PWD_DACR_MASK (0x1 << 0) +#define PWD_DACR_SFT 0 +#define PWD_DACR_DOWN (0x1 << 0) +#define PWD_DACR_ON (0x0 << 0) + +/* RK817_CODEC_AADC_CFG0 */ +#define ADC_L_PWD_MASK (0x1 << 7) +#define ADC_L_PWD_SFT 7 +#define ADC_L_PWD_DIS (0x0 << 7) +#define ADC_L_PWD_EN (0x1 << 7) + +#define ADC_R_PWD_MASK (0x1 << 6) +#define ADC_R_PWD_SFT 6 +#define ADC_R_PWD_DIS (0x0 << 6) +#define ADC_R_PWD_EN (0x1 << 6) + +/* RK817_CODEC_AMIC_CFG0 */ +#define MIC_DIFF_MASK (0x1 << 7) +#define MIC_DIFF_SFT 7 +#define MIC_DIFF_DIS (0x0 << 7) +#define MIC_DIFF_EN (0x1 << 7) + +#define PWD_PGA_L_MASK (0x1 << 5) +#define PWD_PGA_L_SFT 5 +#define PWD_PGA_L_DIS (0x0 << 5) +#define PWD_PGA_L_EN (0x1 << 5) + +#define PWD_PGA_R_MASK (0x1 << 4) +#define PWD_PGA_R_SFT 4 +#define PWD_PGA_R_DIS (0x0 << 4) +#define PWD_PGA_R_EN (0x1 << 4) + +enum { + RK817_HIFI, +}; + +enum { + OFF, + SPK_PATH, + HP_PATH, + SPK_HP, +}; + +enum { + MIC_OFF, + MIC, +}; + +struct rk817_reg_val_typ { + unsigned int reg; + unsigned int value; +}; + +struct rk817_init_bit_typ { + unsigned int reg; + unsigned int power_bit; + unsigned int init_bit; +}; + +#endif /* __RK817_CODEC_H__ */ -- 2.25.1

2 1

[PATCH v2 0/3] ASoC: sdm845: array out of bound issues
by Srinivas Kandagatla 09 Mar '21

09 Mar '21

During testing John Stultz and Amit reported few array our bound issues after enabling bound sanitizer This patch series attempts to fix those! changes since v1: - make sure the wcd is not de-referenced without intialization Srinivas Kandagatla (3): ASoC: qcom: sdm845: Fix array out of bounds access ASoC: qcom: sdm845: Fix array out of range on rx slim channels ASoC: codecs: wcd934x: add a sanity check in set channel map sound/soc/codecs/wcd934x.c | 6 ++++++ sound/soc/qcom/sdm845.c | 6 +++--- 2 files changed, 9 insertions(+), 3 deletions(-) -- 2.21.0

3 5

[PATCH v3 00/15] Report jack and button detection + Capture Support
by Lucas Tanure 09 Mar '21

09 Mar '21

Hi All, Here is a patch series for reporting to user space jack and button events and add the support for Capture. With some cleanups and fixes along the way. Regards, Lucas Tanure Lucas Tanure (12): ASoC: cs42l42: Fix Bitclock polarity inversion ASoC: cs42l42: Fix channel width support ASoC: cs42l42: Fix mixer volume control ASoC: cs42l42: Don't enable/disable regulator at Bias Level ASoC: cs42l42: Always wait at least 3ms after reset ASoC: cs42l42: Remove power if the driver is being removed ASoC: cs42l42: Disable regulators if probe fails ASoC: cs42l42: Provide finer control on playback path ASoC: cs42l42: Set clock source for both ways of stream ASoC: cs42l42: Add Capture Support ASoC: cs42l42: Report jack and button detection ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called Richard Fitzgerald (3): ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT ASoC: cs42l42: Only start PLL if it is needed ASoC: cs42l42: Wait for PLL to lock before switching to it sound/soc/codecs/cs42l42.c | 435 +++++++++++++++++++++---------------- sound/soc/codecs/cs42l42.h | 41 +++- 2 files changed, 282 insertions(+), 194 deletions(-) -- 2.30.1

2 16

[PATCH] ASoC: SOF: intel: fix wrong poll bits in dsp power down
by Pierre-Louis Bossart 09 Mar '21

09 Mar '21

From: Pan Xiuli <xiuli.pan(a)linux.intel.com> The ADSPCS_SPA is Set Power Active bit. To check if DSP is powered down, we need to check ADSPCS_CPA, the Current Power Active bit. Fixes: 747503b1813a3 ("ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations") Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-dsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 9900a058ba6e..6e24e1cb13f9 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -207,7 +207,7 @@ int hda_dsp_core_power_down(struct snd_sof_dev *sdev, unsigned int core_mask) ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPCS, adspcs, - !(adspcs & HDA_DSP_ADSPCS_SPA_MASK(core_mask)), + !(adspcs & HDA_DSP_ADSPCS_CPA_MASK(core_mask)), HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_PD_TIMEOUT * USEC_PER_MSEC); if (ret < 0) -- 2.25.1

2 1

[PATCH] ASoC: rt5682: add delay time of workqueue to control next IRQ event
by shumingf＠realtek.com 09 Mar '21

09 Mar '21

From: Shuming Fan <shumingf(a)realtek.com> This patch keeps the delay time (50 ms) for jack detection and zero delay time for the button press. This patch improves the reaction of the button press. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682-i2c.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt5682.c | 2 ++ sound/soc/codecs/rt5682.h | 1 + 4 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 93c1603b42f1..8ea9f1d9fec0 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -78,7 +78,7 @@ static irqreturn_t rt5682_irq(int irq, void *data) struct rt5682_priv *rt5682 = data; mod_delayed_work(system_power_efficient_wq, - &rt5682->jack_detect_work, msecs_to_jiffies(250)); + &rt5682->jack_detect_work, msecs_to_jiffies(rt5682->irq_work_delay_time)); return IRQ_HANDLED; } diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 5e097f776561..fed80c8f994f 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -677,7 +677,7 @@ static int rt5682_interrupt_callback(struct sdw_slave *slave, if (status->control_port & 0x4) { mod_delayed_work(system_power_efficient_wq, - &rt5682->jack_detect_work, msecs_to_jiffies(250)); + &rt5682->jack_detect_work, msecs_to_jiffies(rt5682->irq_work_delay_time)); } return 0; diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 559dc6db1f7c..0e2a10ed11da 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1094,6 +1094,7 @@ void rt5682_jack_detect_handler(struct work_struct *work) /* jack was out, report jack type */ rt5682->jack_type = rt5682_headset_detect(rt5682->component, 1); + rt5682->irq_work_delay_time = 0; } else if ((rt5682->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) { /* jack is already in, report button event */ @@ -1139,6 +1140,7 @@ void rt5682_jack_detect_handler(struct work_struct *work) } else { /* jack out */ rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0); + rt5682->irq_work_delay_time = 50; } snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 1f9c51a5b9bf..74ff66767016 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1439,6 +1439,7 @@ struct rt5682_priv { int pll_out[RT5682_PLLS]; int jack_type; + int irq_work_delay_time; }; extern const char *rt5682_supply_names[RT5682_NUM_SUPPLIES]; -- 2.29.0

3 2

[PATCH -next] ASoC: rt715-sdca: Remove unused including <linux/version.h>
by 'Wei Yongjun 09 Mar '21

09 Mar '21

From: Wei Yongjun <weiyongjun1(a)huawei.com> Remove including <linux/version.h> that don't need it. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/rt715-sdca.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 92ad6fa408ec..20528afbdc57 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -9,7 +9,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/pm_runtime.h>

2 1

[PATCH -next] ASoC: rt715-sdca: Fix return value check in rt715_sdca_sdw_probe()
by 'Wei Yongjun 09 Mar '21

09 Mar '21

From: Wei Yongjun <weiyongjun1(a)huawei.com> In case of error, the function devm_regmap_init_sdw_mbq() and devm_regmap_init_sdw() returns ERR_PTR() not NULL. The NULL test in the return value check should be replaced with IS_ERR(). Fixes: 393c52d2d109 ("ASoC: rt715-sdca: Add RT715 sdca vendor-specific driver") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/rt715-sdca-sdw.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index bcced85876b0..1350798406f0 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -184,12 +184,12 @@ static int rt715_sdca_sdw_probe(struct sdw_slave *slave, /* Regmap Initialization */ mbq_regmap = devm_regmap_init_sdw_mbq(slave, &rt715_sdca_mbq_regmap); - if (!mbq_regmap) - return -EINVAL; + if (IS_ERR(mbq_regmap)) + return PTR_ERR(mbq_regmap); regmap = devm_regmap_init_sdw(slave, &rt715_sdca_regmap); - if (!regmap) - return -EINVAL; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); return rt715_sdca_init(&slave->dev, mbq_regmap, regmap, slave); }

2 1

[PATCH v2 0/7] Add I2S-MCC support for Microchip's SAMA7G5
by Codrin Ciubotariu 09 Mar '21

09 Mar '21

SAMA7G5 includes an updated version of I2S-MCC, found previously on SAM9X60. This controller includes 8 data pins, 4 for playback and 4 for capture. For I2S and LEFT_J formats, these pins can be used to send/receive up to 8 audio channels. For DSP_A, with TDM, any pins pair (DIN/DOUT) from these 4 can be selected to send/receive data. This version also includes 2 FIFOs (send and receive). This patch set starts by moving the driver's bindings to yaml and continues with adding a new compatible for the SAMA7G5 variant, followed by the changes needed for I2S/LEFT_J support, TDM pin pair selection and FIFO support, exclusively for SAMA7G5. Changes in v2: - moved DT binding conversion patch from the beginning to the end of the patch serieses - patches that update the DT binding are modified to change .txt file instead of .yaml Codrin Ciubotariu (7): dt-bindings: mchp,i2s-mcc: Add SAMA7G5 to binding ASoC: mchp-i2s-mcc: Add compatible for SAMA7G5 ASoC: mchp-i2s-mcc: Add multi-channel support for I2S and LEFT_J formats dt-bindings: mchp,i2s-mcc: Add property to specify pin pair for TDM ASoC: mchp-i2s-mcc: Add support to select TDM pins ASoC: mchp-i2s-mcc: Add FIFOs support ASoC: convert Microchip I2SMCC binding to yaml .../bindings/sound/mchp,i2s-mcc.yaml | 108 ++++++++++++ .../bindings/sound/mchp-i2s-mcc.txt | 43 ----- sound/soc/atmel/Kconfig | 3 + sound/soc/atmel/mchp-i2s-mcc.c | 161 +++++++++++++++--- 4 files changed, 252 insertions(+), 63 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mchp,i2s-mcc.yaml delete mode 100644 Documentation/devicetree/bindings/sound/mchp-i2s-mcc.txt -- 2.27.0

3 10

[PATCH] ASoC: amd: add support for rt5682 codec in machine driver
by Vijendar Mukunda 09 Mar '21

09 Mar '21

Add support for RT5682 codec in machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/Kconfig | 5 +- sound/soc/amd/acp-da7219-max98357a.c | 353 +++++++++++++++++++++++++++++++++-- 2 files changed, 339 insertions(+), 19 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index a6ce000..13f5b1b2 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -5,14 +5,15 @@ config SND_SOC_AMD_ACP This option enables ACP DMA support on AMD platform. config SND_SOC_AMD_CZ_DA7219MX98357_MACH - tristate "AMD CZ support for DA7219 and MAX9835" + tristate "AMD CZ support for DA7219, RT5682 and MAX9835" select SND_SOC_DA7219 + select SND_SOC_RT5682 select SND_SOC_MAX98357A select SND_SOC_ADAU7002 select REGULATOR depends on SND_SOC_AMD_ACP && I2C && GPIOLIB help - This option enables machine driver for DA7219 and MAX9835. + This option enables machine driver for DA7219, RT5682 and MAX9835. config SND_SOC_AMD_CZ_RT5645_MACH tristate "AMD CZ support for RT5645" diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 849288d..ff2b639 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -1,7 +1,7 @@ /* * Machine driver for AMD ACP Audio engine using DA7219 & MAX98357 codec * - * Copyright 2017 Advanced Micro Devices, Inc. + * Copyright 2017-2021 Advanced Micro Devices, Inc. * * Permission is hereby granted, free of charge, to any person obtaining a * copy of this software and associated documentation files (the "Software"), @@ -41,14 +41,19 @@ #include "acp.h" #include "../codecs/da7219.h" #include "../codecs/da7219-aad.h" +#include "../codecs/rt5682.h" #define CZ_PLAT_CLK 48000000 #define DUAL_CHANNEL 2 +#define RT5682_PLL_FREQ (48000 * 512) static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_wclk; static struct clk *da7219_dai_bclk; -extern bool bt_uart_enable; +static struct clk *rt5682_dai_wclk; +static struct clk *rt5682_dai_bclk; +extern int bt_uart_enable; +void *soc_is_rltk_max(struct device *dev); static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { @@ -128,6 +133,88 @@ static void da7219_clk_disable(void) clk_disable_unprepare(da7219_dai_bclk); } +static int cz_rt5682_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_component *component = codec_dai->component; + + dev_info(codec_dai->dev, "codec dai name = %s\n", codec_dai->name); + + /* Set codec sysclk */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL2, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, + "Failed to set rt5682 SYSCLK: %d\n", ret); + return ret; + } + /* set codec PLL */ + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL2, RT5682_PLL2_S_MCLK, + CZ_PLAT_CLK, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set rt5682 PLL: %d\n", ret); + return ret; + } + + rt5682_dai_wclk = devm_clk_get(component->dev, "rt5682-dai-wclk"); + if (IS_ERR(rt5682_dai_wclk)) + return PTR_ERR(rt5682_dai_wclk); + + rt5682_dai_bclk = devm_clk_get(component->dev, "rt5682-dai-bclk"); + if (IS_ERR(rt5682_dai_bclk)) + return PTR_ERR(rt5682_dai_bclk); + + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &cz_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "HP jack creation failed %d\n", ret); + return ret; + } + + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ret = snd_soc_component_set_jack(component, &cz_jack, NULL); + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + return 0; +} + +static int rt5682_clk_enable(struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + + /* + * Set wclk to 48000 because the rate constraint of this driver is + * 48000. ADAU7002 spec: "The ADAU7002 requires a BCLK rate that is + * minimum of 64x the LRCLK sample rate." DA7219 is the only clk + * source so for all codecs we have to limit bclk to 64X lrclk. + */ + clk_set_rate(rt5682_dai_wclk, 48000); + clk_set_rate(rt5682_dai_bclk, 48000 * 64); + ret = clk_prepare_enable(rt5682_dai_bclk); + if (ret < 0) { + dev_err(rtd->dev, "can't enable master clock %d\n", ret); + return ret; + } + return ret; +} + +static void rt5682_clk_disable(void) +{ + clk_disable_unprepare(rt5682_dai_bclk); +} + static const unsigned int channels[] = { DUAL_CHANNEL, }; @@ -260,6 +347,118 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_rt5682_play_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->play_i2s_instance = I2S_SP_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_cap_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL1; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_max_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->play_i2s_instance = I2S_BT_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_dmic0_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_BT_INSTANCE; + return rt5682_clk_enable(substream); +} + +static int cz_rt5682_dmic1_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL0; + return rt5682_clk_enable(substream); +} + +static void cz_rt5682_shutdown(struct snd_pcm_substream *substream) +{ + rt5682_clk_disable(); +} + static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, @@ -285,6 +484,31 @@ static const struct snd_soc_ops cz_dmic1_cap_ops = { .shutdown = cz_da7219_shutdown, }; +static const struct snd_soc_ops cz_rt5682_play_ops = { + .startup = cz_rt5682_play_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_cap_ops = { + .startup = cz_rt5682_cap_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_max_play_ops = { + .startup = cz_rt5682_max_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_dmic0_cap_ops = { + .startup = cz_rt5682_dmic0_startup, + .shutdown = cz_rt5682_shutdown, +}; + +static const struct snd_soc_ops cz_rt5682_dmic1_cap_ops = { + .startup = cz_rt5682_dmic1_startup, + .shutdown = cz_rt5682_shutdown, +}; + SND_SOC_DAILINK_DEF(designware1, DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto"))); SND_SOC_DAILINK_DEF(designware2, @@ -294,6 +518,8 @@ SND_SOC_DAILINK_DEF(designware3, SND_SOC_DAILINK_DEF(dlgs, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", "da7219-hifi"))); +SND_SOC_DAILINK_DEF(rt5682, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1"))); SND_SOC_DAILINK_DEF(mx, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi"))); SND_SOC_DAILINK_DEF(adau, @@ -353,6 +579,57 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { }, }; +static struct snd_soc_dai_link cz_dai_5682_98357[] = { + { + .name = "amd-rt5682-play", + .stream_name = "Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .init = cz_rt5682_init, + .dpcm_playback = 1, + .ops = &cz_rt5682_play_ops, + SND_SOC_DAILINK_REG(designware1, rt5682, platform), + }, + { + .name = "amd-rt5682-cap", + .stream_name = "Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_cap_ops, + SND_SOC_DAILINK_REG(designware2, rt5682, platform), + }, + { + .name = "amd-max98357-play", + .stream_name = "HiFi Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_playback = 1, + .ops = &cz_rt5682_max_play_ops, + SND_SOC_DAILINK_REG(designware3, mx, platform), + }, + { + /* C panel DMIC */ + .name = "dmic0", + .stream_name = "DMIC0 Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_dmic0_cap_ops, + SND_SOC_DAILINK_REG(designware3, adau, platform), + }, + { + /* A/B panel DMIC */ + .name = "dmic1", + .stream_name = "DMIC1 Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_rt5682_dmic1_cap_ops, + SND_SOC_DAILINK_REG(designware2, adau, platform), + }, +}; + static const struct snd_soc_dapm_widget cz_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), @@ -368,6 +645,14 @@ static const struct snd_soc_dapm_route cz_audio_route[] = { {"PDM_DAT", NULL, "Int Mic"}, }; +static const struct snd_soc_dapm_route cz_rt5682_audio_route[] = { + {"Headphones", NULL, "HPOL"}, + {"Headphones", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"Speakers", NULL, "Speaker"}, + {"PDM_DAT", NULL, "Int Mic"}, +}; + static const struct snd_kcontrol_new cz_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), @@ -388,6 +673,28 @@ static struct snd_soc_card cz_card = { .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct snd_soc_card cz_rt5682_card = { + .name = "acpr5682m98357", + .owner = THIS_MODULE, + .dai_link = cz_dai_5682_98357, + .num_links = ARRAY_SIZE(cz_dai_5682_98357), + .dapm_widgets = cz_widgets, + .num_dapm_widgets = ARRAY_SIZE(cz_widgets), + .dapm_routes = cz_rt5682_audio_route, + .controls = cz_mc_controls, + .num_controls = ARRAY_SIZE(cz_mc_controls), +}; + +void *soc_is_rltk_max(struct device *dev) +{ + const struct acpi_device_id *match; + + match = acpi_match_device(dev->driver->acpi_match_table, dev); + if (!match) + return NULL; + return (void *)match->driver_data; +} + static struct regulator_consumer_supply acp_da7219_supplies[] = { REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"), REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"), @@ -425,29 +732,39 @@ static int cz_probe(struct platform_device *pdev) struct snd_soc_card *card; struct acp_platform_info *machine; struct regulator_dev *rdev; - - acp_da7219_cfg.dev = &pdev->dev; - rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, - &acp_da7219_cfg); - if (IS_ERR(rdev)) { - dev_err(&pdev->dev, "Failed to register regulator: %d\n", - (int)PTR_ERR(rdev)); - return -EINVAL; + struct device *dev = &pdev->dev; + + card = (struct snd_soc_card *)soc_is_rltk_max(dev); + if (!card) + return -ENODEV; + if (!strcmp(card->name, "acpd7219m98357")) { + acp_da7219_cfg.dev = &pdev->dev; + rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, + &acp_da7219_cfg); + if (IS_ERR(rdev)) { + dev_err(&pdev->dev, "Failed to register regulator: %d\n", + (int)PTR_ERR(rdev)); + return -EINVAL; + } } machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); if (!machine) return -ENOMEM; - card = &cz_card; - cz_card.dev = &pdev->dev; + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - ret = devm_snd_soc_register_card(&pdev->dev, &cz_card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", - cz_card.name, ret); + card->name, ret); + else + dev_dbg(&pdev->dev, + "devm_snd_soc_register_card(%s) probe deferred: %d\n", + card->name, ret); return ret; } bt_uart_enable = !device_property_read_bool(&pdev->dev, @@ -457,7 +774,8 @@ static int cz_probe(struct platform_device *pdev) #ifdef CONFIG_ACPI static const struct acpi_device_id cz_audio_acpi_match[] = { - { "AMD7219", 0 }, + { "AMD7219", (unsigned long)&cz_card }, + { "AMDI5682", (unsigned long)&cz_rt5682_card}, {}, }; MODULE_DEVICE_TABLE(acpi, cz_audio_acpi_match); @@ -475,5 +793,6 @@ static struct platform_driver cz_pcm_driver = { module_platform_driver(cz_pcm_driver); MODULE_AUTHOR("akshu.agrawal(a)amd.com"); -MODULE_DESCRIPTION("DA7219 & MAX98357A audio support"); +MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); +MODULE_DESCRIPTION("DA7219, RT5682 & MAX98357A audio support"); MODULE_LICENSE("GPL v2"); -- 2.7.4

2 1

[PATCH] SOUND: pci: hda: Add Sound BlasterX AE-5 Plus support
by Simeon Simeonoff 09 Mar '21

09 Mar '21

The new AE-5 Plus model has a different Subsystem ID compared to the non-plus model. Adding the new id to the list of quirks. Signed-off-by: Simeon Simeonoff <sim.simeonoff(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index c966f49fa942..b2b620f6c832 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1309,6 +1309,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D), SND_PCI_QUIRK(0x1102, 0x0018, "Recon3D", QUIRK_R3D), SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5), + SND_PCI_QUIRK(0x1102, 0x0191, "Sound Blaster AE-5 Plus", QUIRK_AE5), SND_PCI_QUIRK(0x1102, 0x0081, "Sound Blaster AE-7", QUIRK_AE7), {} }; -- 2.27.0

2 2

[PATCH v3 0/4] ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec
by Vitaly Rodionov 09 Mar '21

09 Mar '21

Dell's laptops Inspiron 3500, Inspiron 3501, Inspiron 3505 are using Cirrus Logic CS8409 HDA bridge with CS42L42 companion codec. The CS8409 is a multichannel HD audio routing controller. CS8409 includes support for four channels of digital microphone data and two bidirectional ASPs for up to 32 channels of TDM data or 4 channels of I2S data. The CS8409 is intended to be used with a remote companion codec that implements high performance analog functions in close physical proximity to the end-equipment audio port or speaker driver. The CS42L42 is a low-power audio codec with integrated MIPI SoundWire interface or I2C/I2S/TDM interfaces designed for portable applications. It provides a high-dynamic range, stereo DAC for audio playback and a mono high-dynamic-range ADC for audio capture Changes since version 1: ALSA: hda/cirrus: Increase AUTO_CFG_MAX_INS from 8 to 18 * No change ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec. * Removed redundant fields in fixup table * Handle gpio via spec->gpio_dir, spec->gpio_data and spec->gpio_mask * Moved cs8409_cs42l42_init() from patch 2, to handle resume correctly ALSA: hda/cirrus: Add jack detect interrupt support from CS42L42 companion codec. * Run scripts/checkpatch.pl, fixed new warnings ALSA: hda/cirrus: Add Headphone and Headset MIC Volume Control * Moved control values to cache to avoid i2c read at each time. Stefan Binding (1): ALSA: hda/cirrus: Add Headphone and Headset MIC Volume Control Vitaly Rodionov (3): ALSA: hda/cirrus: Increase AUTO_CFG_MAX_INS from 8 to 18 ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec. ALSA: hda/cirrus: Add jack detect interrupt support from CS42L42 companion codec. sound/pci/hda/hda_auto_parser.h | 2 +- sound/pci/hda/hda_local.h | 2 +- sound/pci/hda/patch_cirrus.c | 1081 +++++++++++++++++++++++++++++++ 3 files changed, 1083 insertions(+), 2 deletions(-) -- 2.25.1

4 11

[syzbot] KASAN: use-after-free Read in usb_audio_disconnect
by syzbot 09 Mar '21

09 Mar '21

Hello, syzbot found the following issue on: HEAD commit: 4641b323 Add linux-next specific files for 20210305 git tree: linux-next console output: https://syzkaller.appspot.com/x/log.txt?x=103252ead00000 kernel config: https://syzkaller.appspot.com/x/.config?x=b87b6a083d302a90 dashboard link: https://syzkaller.appspot.com/bug?extid=ffad4c74b3b3ea3aa9c3 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=1122d35cd00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=17f6476cd00000 The issue was bisected to: commit 9799110825dba087c2bdce886977cf84dada2005 Author: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Date: Thu Mar 4 04:34:16 2021 +0000 ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend() bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=166d91ead00000 final oops: https://syzkaller.appspot.com/x/report.txt?x=156d91ead00000 console output: https://syzkaller.appspot.com/x/log.txt?x=116d91ead00000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+ffad4c74b3b3ea3aa9c3(a)syzkaller.appspotmail.com Fixes: 9799110825db ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()") usb 1-1: USB disconnect, device number 2 ================================================================== BUG: KASAN: use-after-free in usb_audio_disconnect+0x750/0x800 sound/usb/card.c:918 Read of size 2 at addr ffff888027a08f24 by task kworker/0:2/2966 CPU: 0 PID: 2966 Comm: kworker/0:2 Not tainted 5.12.0-rc1-next-20210305-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:79 [inline] dump_stack+0xfa/0x151 lib/dump_stack.c:120 print_address_description.constprop.0.cold+0x5b/0x2f8 mm/kasan/report.c:232 __kasan_report mm/kasan/report.c:399 [inline] kasan_report.cold+0x7c/0xd8 mm/kasan/report.c:416 usb_audio_disconnect+0x750/0x800 sound/usb/card.c:918 usb_unbind_interface+0x1d8/0x8d0 drivers/usb/core/driver.c:458 __device_release_driver+0x3bd/0x6f0 drivers/base/dd.c:1156 device_release_driver_internal drivers/base/dd.c:1187 [inline] device_release_driver+0x26/0x40 drivers/base/dd.c:1210 bus_remove_device+0x2eb/0x5a0 drivers/base/bus.c:533 device_del+0x502/0xd40 drivers/base/core.c:3421 usb_disable_device+0x35b/0x7b0 drivers/usb/core/message.c:1413 usb_disconnect.cold+0x27d/0x791 drivers/usb/core/hub.c:2218 hub_port_connect drivers/usb/core/hub.c:5074 [inline] hub_port_connect_change drivers/usb/core/hub.c:5363 [inline] port_event drivers/usb/core/hub.c:5509 [inline] hub_event+0x1c9c/0x4320 drivers/usb/core/hub.c:5591 process_one_work+0x98d/0x1600 kernel/workqueue.c:2275 worker_thread+0x64c/0x1120 kernel/workqueue.c:2421 kthread+0x3b1/0x4a0 kernel/kthread.c:292 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:294 Allocated by task 5: kasan_save_stack+0x1b/0x40 mm/kasan/common.c:38 kasan_set_track mm/kasan/common.c:46 [inline] set_alloc_info mm/kasan/common.c:427 [inline] ____kasan_kmalloc mm/kasan/common.c:506 [inline] ____kasan_kmalloc mm/kasan/common.c:465 [inline] __kasan_kmalloc+0x99/0xc0 mm/kasan/common.c:515 kmalloc include/linux/slab.h:561 [inline] kzalloc include/linux/slab.h:686 [inline] snd_card_new+0xc2/0xcb0 sound/core/init.c:176 snd_usb_audio_create sound/usb/card.c:588 [inline] usb_audio_probe+0x1547/0x2c70 sound/usb/card.c:755 usb_probe_interface+0x315/0x7f0 drivers/usb/core/driver.c:396 really_probe+0x291/0xe60 drivers/base/dd.c:554 driver_probe_device+0x26b/0x3d0 drivers/base/dd.c:740 __device_attach_driver+0x1d1/0x290 drivers/base/dd.c:846 bus_for_each_drv+0x15f/0x1e0 drivers/base/bus.c:431 __device_attach+0x228/0x4a0 drivers/base/dd.c:914 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:491 device_add+0xbdb/0x1db0 drivers/base/core.c:3242 usb_set_configuration+0x113f/0x1910 drivers/usb/core/message.c:2164 usb_generic_driver_probe+0xba/0x100 drivers/usb/core/generic.c:238 usb_probe_device+0xd9/0x2c0 drivers/usb/core/driver.c:293 really_probe+0x291/0xe60 drivers/base/dd.c:554 driver_probe_device+0x26b/0x3d0 drivers/base/dd.c:740 __device_attach_driver+0x1d1/0x290 drivers/base/dd.c:846 bus_for_each_drv+0x15f/0x1e0 drivers/base/bus.c:431 __device_attach+0x228/0x4a0 drivers/base/dd.c:914 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:491 device_add+0xbdb/0x1db0 drivers/base/core.c:3242 usb_new_device.cold+0x721/0x1058 drivers/usb/core/hub.c:2555 hub_port_connect drivers/usb/core/hub.c:5223 [inline] hub_port_connect_change drivers/usb/core/hub.c:5363 [inline] port_event drivers/usb/core/hub.c:5509 [inline] hub_event+0x2357/0x4320 drivers/usb/core/hub.c:5591 process_one_work+0x98d/0x1600 kernel/workqueue.c:2275 worker_thread+0x64c/0x1120 kernel/workqueue.c:2421 kthread+0x3b1/0x4a0 kernel/kthread.c:292 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:294 Freed by task 2966: kasan_save_stack+0x1b/0x40 mm/kasan/common.c:38 kasan_set_track+0x1c/0x30 mm/kasan/common.c:46 kasan_set_free_info+0x20/0x30 mm/kasan/generic.c:357 ____kasan_slab_free mm/kasan/common.c:360 [inline] ____kasan_slab_free mm/kasan/common.c:325 [inline] __kasan_slab_free+0xf5/0x130 mm/kasan/common.c:367 kasan_slab_free include/linux/kasan.h:199 [inline] slab_free_hook mm/slub.c:1562 [inline] slab_free_freelist_hook+0x72/0x1b0 mm/slub.c:1600 slab_free mm/slub.c:3161 [inline] kfree+0xe5/0x7b0 mm/slub.c:4215 device_release+0x9f/0x240 drivers/base/core.c:2108 kobject_cleanup lib/kobject.c:705 [inline] kobject_release lib/kobject.c:736 [inline] kref_put include/linux/kref.h:65 [inline] kobject_put+0x1c8/0x540 lib/kobject.c:753 put_device+0x1b/0x30 drivers/base/core.c:3341 snd_card_free_when_closed+0x35/0x50 sound/core/init.c:518 usb_audio_disconnect+0x2ba/0x800 sound/usb/card.c:913 usb_unbind_interface+0x1d8/0x8d0 drivers/usb/core/driver.c:458 __device_release_driver+0x3bd/0x6f0 drivers/base/dd.c:1156 device_release_driver_internal drivers/base/dd.c:1187 [inline] device_release_driver+0x26/0x40 drivers/base/dd.c:1210 bus_remove_device+0x2eb/0x5a0 drivers/base/bus.c:533 device_del+0x502/0xd40 drivers/base/core.c:3421 usb_disable_device+0x35b/0x7b0 drivers/usb/core/message.c:1413 usb_disconnect.cold+0x27d/0x791 drivers/usb/core/hub.c:2218 hub_port_connect drivers/usb/core/hub.c:5074 [inline] hub_port_connect_change drivers/usb/core/hub.c:5363 [inline] port_event drivers/usb/core/hub.c:5509 [inline] hub_event+0x1c9c/0x4320 drivers/usb/core/hub.c:5591 process_one_work+0x98d/0x1600 kernel/workqueue.c:2275 worker_thread+0x64c/0x1120 kernel/workqueue.c:2421 kthread+0x3b1/0x4a0 kernel/kthread.c:292 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:294 The buggy address belongs to the object at ffff888027a08000 which belongs to the cache kmalloc-8k of size 8192 The buggy address is located 3876 bytes inside of 8192-byte region [ffff888027a08000, ffff888027a0a000) The buggy address belongs to the page: page:0000000024fc4526 refcount:1 mapcount:0 mapping:0000000000000000 index:0x0 pfn:0x27a08 head:0000000024fc4526 order:3 compound_mapcount:0 compound_pincount:0 flags: 0xfff00000010200(slab|head) raw: 00fff00000010200 dead000000000100 dead000000000122 ffff888010842280 raw: 0000000000000000 0000000000020002 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff888027a08e00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff888027a08e80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb >ffff888027a08f00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff888027a08f80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff888027a09000: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. For information about bisection process see: https://goo.gl/tpsmEJ#bisection syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

4 3

[PATCH v2] soundwire: bus: Fix device found flag correctly
by Srinivas Kandagatla 09 Mar '21

09 Mar '21

found flag is used to indicate SoundWire devices that are both enumerated on the bus and available in the device list. However this flag is not reset correctly after one iteration, This could miss some of the devices that are enumerated on the bus but not in device list. So reset this correctly to fix this issue! Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/soundwire/bus.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index a9ab4a6264e0..188e391c5a83 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -712,7 +712,7 @@ static int sdw_program_device_num(struct sdw_bus *bus) struct sdw_slave *slave, *_s; struct sdw_slave_id id; struct sdw_msg msg; - bool found = false; + bool found; int count = 0, ret; u64 addr; @@ -744,6 +744,7 @@ static int sdw_program_device_num(struct sdw_bus *bus) sdw_extract_slave_id(bus, addr, &id); + found = false; /* Now compare with entries */ list_for_each_entry_safe(slave, _s, &bus->slaves, node) { if (sdw_compare_devid(slave, id) == 0) { -- 2.21.0

1 0

[PATCH 0/3] ASoC: sdm845: array out of bound issues
by Srinivas Kandagatla 09 Mar '21

09 Mar '21

During testing John Stultz and Amit reported few array our bound issues after enabling bound sanitizer This patch series attempts to fix those! Srinivas Kandagatla (3): ASoC: qcom: sdm845: Fix array out of bounds access ASoC: qcom: sdm845: Fix array out of range on rx slim channels ASoC: codecs: wcd934x: add a sanity check in set channel map sound/soc/codecs/wcd934x.c | 6 ++++++ sound/soc/qcom/sdm845.c | 6 +++--- 2 files changed, 9 insertions(+), 3 deletions(-) -- 2.21.0

1 3

[PATCH next 0/2] fixes for sound: usb:
by Pavel Skripkin 09 Mar '21

09 Mar '21

This small patch series fixes 2 errors from commit 9799110825db ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()"). One of them was reported by syzbot, but second one appeared while testing fixes for the first one. Pavel Skripkin (2): sound: usb: fix NULL ptr dereference in usb_audio_probe sound: usb: fix use after free in usb_audio_disconnect sound/usb/card.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) -- 2.25.1

2 4

[PATCH 0/3] ASoc: rt286: Fix combojack detection for ALC3263 (Dell XPS 13 9343)
by David Ward 09 Mar '21

09 Mar '21

This fixes a longstanding issue that prevented jack detection from working on systems with the ALC3263 codec, such as the Dell XPS 13 9343, when using I2S mode for audio (instead of HDA mode). There is currently an ACPI quirk for the Dell XPS 13 9343 that forces it to use HDA mode instead of I2S mode, because jack detection did not work. See: https://lore.kernel.org/alsa-devel/CAPeXnHv07HkvcHrYFmZMr8OTp7U7F=k_k=LPYnU… There are very likely codec fixups for the HDA driver which should be added to the I2S codec driver, to attempt to achieve parity. Commit 394c97f824fa ("ALSA: hda/realtek - Change EAPD to verb control") describes an issue that still seems to be present in the I2S codec driver, for example. In any case this change should move closer to allowing the Dell XPS 13 9343 to default to I2S mode; and it resolves this issue for systems which use I2S mode now, including the Dell Latitude 7350 and the Dell Venue 11 Pro 7140. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=114171 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961 Signed-off-by: David Ward <david.ward(a)gatech.edu> David Ward (3): ASoC: rt286: Make RT286_SET_GPIO_* readable and writable ASoC: rt286: Handle all devices with ALC3263 codec ASoc: rt286: Handle headphones in combo jack correctly sound/soc/codecs/rt286.c | 29 +++++++++++++++++++---------- 1 file changed, 19 insertions(+), 10 deletions(-) -- 2.30.1

1 3

[PATCH] ASoC: rt286: Fix upper byte in DMIC2 configuration
by David Ward 09 Mar '21

09 Mar '21

This HDA verb sets the upper byte of the Configuration Default register, so it expects an 8-bit value here. For the rt298, the same fix was applied in commit f8f2dc4a7127 ("ASoC: rt298: fix wrong setting of gpio2_en"). Signed-off-by: David Ward <david.ward(a)gatech.edu> --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 8abe232ca4a4..f9b29782b62a 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1204,7 +1204,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, mdelay(10); if (!rt286->pdata.gpio2_en) - regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0x4000); + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0x40); else regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0); -- 2.30.1

1 0

[PATCH] ASoC: bcm: add missing call to of_node_put()
by Yang Li 08 Mar '21

08 Mar '21

In one of the error paths of the for_each_child_of_node() loop, add missing call to of_node_put(). Fix the following coccicheck warning: ./sound/soc/bcm/cygnus-ssp.c:1346:1-33: WARNING: Function "for_each_available_child_of_node" should have of_node_put() before return around line 1352. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/bcm/cygnus-ssp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index 6e634b4..aa16a23 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1348,8 +1348,10 @@ static int cygnus_ssp_probe(struct platform_device *pdev) &cygnus_ssp_dai[active_port_count]); /* negative is err, 0 is active and good, 1 is disabled */ - if (err < 0) + if (err < 0) { + of_node_put(child_node); return err; + } else if (!err) { dev_dbg(dev, "Activating DAI: %s\n", cygnus_ssp_dai[active_port_count].name); -- 1.8.3.1

2 1

[syzbot] general protection fault in usb_audio_probe
by syzbot 08 Mar '21

08 Mar '21

Hello, syzbot found the following issue on: HEAD commit: 4641b323 Add linux-next specific files for 20210305 git tree: linux-next console output: https://syzkaller.appspot.com/x/log.txt?x=12584542d00000 kernel config: https://syzkaller.appspot.com/x/.config?x=b87b6a083d302a90 dashboard link: https://syzkaller.appspot.com/bug?extid=719da9b149a931f5143f syz repro: https://syzkaller.appspot.com/x/repro.syz?x=125b52aed00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=123df2aed00000 The issue was bisected to: commit 9799110825dba087c2bdce886977cf84dada2005 Author: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Date: Thu Mar 4 04:34:16 2021 +0000 ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend() bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=178418a2d00000 final oops: https://syzkaller.appspot.com/x/report.txt?x=144418a2d00000 console output: https://syzkaller.appspot.com/x/log.txt?x=104418a2d00000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+719da9b149a931f5143f(a)syzkaller.appspotmail.com Fixes: 9799110825db ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()") usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: syz usb 1-1: Manufacturer: syz usb 1-1: SerialNumber: syz usb 1-1: 0:2 : does not exist general protection fault, probably for non-canonical address 0xdffffc0000000002: 0000 [#1] PREEMPT SMP KASAN KASAN: null-ptr-deref in range [0x0000000000000010-0x0000000000000017] CPU: 0 PID: 2965 Comm: kworker/0:2 Not tainted 5.12.0-rc1-next-20210305-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event RIP: 0010:usb_audio_probe+0xa80/0x2c70 sound/usb/card.c:833 Code: 8b 74 24 08 48 89 df e8 0e 31 06 00 e8 59 64 8d fa 48 8b 04 24 48 8d 78 12 48 b8 00 00 00 00 00 fc ff df 48 89 fa 48 c1 ea 03 <0f> b6 14 02 48 89 f8 83 e0 07 83 c0 01 38 d0 7c 08 84 d2 0f 85 af RSP: 0018:ffffc900016cf0c8 EFLAGS: 00010202 RAX: dffffc0000000000 RBX: ffff888023e48e88 RCX: 0000000000000000 RDX: 0000000000000002 RSI: ffffffff86e61767 RDI: 0000000000000012 RBP: 0000000000000020 R08: 0000000000000000 R09: ffffffff8d580a07 R10: ffffffff86e61bfe R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: dffffc0000000000 R15: 000000000000001f FS: 0000000000000000(0000) GS:ffff8880b9c00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 000056098ddcb140 CR3: 00000000147bd000 CR4: 00000000001506f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: usb_probe_interface+0x315/0x7f0 drivers/usb/core/driver.c:396 really_probe+0x291/0xe60 drivers/base/dd.c:554 driver_probe_device+0x26b/0x3d0 drivers/base/dd.c:740 __device_attach_driver+0x1d1/0x290 drivers/base/dd.c:846 bus_for_each_drv+0x15f/0x1e0 drivers/base/bus.c:431 __device_attach+0x228/0x4a0 drivers/base/dd.c:914 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:491 device_add+0xbdb/0x1db0 drivers/base/core.c:3242 usb_set_configuration+0x113f/0x1910 drivers/usb/core/message.c:2164 usb_generic_driver_probe+0xba/0x100 drivers/usb/core/generic.c:238 usb_probe_device+0xd9/0x2c0 drivers/usb/core/driver.c:293 really_probe+0x291/0xe60 drivers/base/dd.c:554 driver_probe_device+0x26b/0x3d0 drivers/base/dd.c:740 __device_attach_driver+0x1d1/0x290 drivers/base/dd.c:846 bus_for_each_drv+0x15f/0x1e0 drivers/base/bus.c:431 __device_attach+0x228/0x4a0 drivers/base/dd.c:914 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:491 device_add+0xbdb/0x1db0 drivers/base/core.c:3242 usb_new_device.cold+0x721/0x1058 drivers/usb/core/hub.c:2555 hub_port_connect drivers/usb/core/hub.c:5223 [inline] hub_port_connect_change drivers/usb/core/hub.c:5363 [inline] port_event drivers/usb/core/hub.c:5509 [inline] hub_event+0x2357/0x4320 drivers/usb/core/hub.c:5591 process_one_work+0x98d/0x1600 kernel/workqueue.c:2275 worker_thread+0x64c/0x1120 kernel/workqueue.c:2421 kthread+0x3b1/0x4a0 kernel/kthread.c:292 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:294 Modules linked in: ---[ end trace 4d5988887d9521a9 ]--- RIP: 0010:usb_audio_probe+0xa80/0x2c70 sound/usb/card.c:833 Code: 8b 74 24 08 48 89 df e8 0e 31 06 00 e8 59 64 8d fa 48 8b 04 24 48 8d 78 12 48 b8 00 00 00 00 00 fc ff df 48 89 fa 48 c1 ea 03 <0f> b6 14 02 48 89 f8 83 e0 07 83 c0 01 38 d0 7c 08 84 d2 0f 85 af RSP: 0018:ffffc900016cf0c8 EFLAGS: 00010202 RAX: dffffc0000000000 RBX: ffff888023e48e88 RCX: 0000000000000000 RDX: 0000000000000002 RSI: ffffffff86e61767 RDI: 0000000000000012 RBP: 0000000000000020 R08: 0000000000000000 R09: ffffffff8d580a07 R10: ffffffff86e61bfe R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: dffffc0000000000 R15: 000000000000001f FS: 0000000000000000(0000) GS:ffff8880b9c00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00005572425f2908 CR3: 0000000020f01000 CR4: 00000000001506f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. For information about bisection process see: https://goo.gl/tpsmEJ#bisection syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

1 0

[PATCH v3 0/9] soundwire: qcom: various improvements
by Srinivas Kandagatla 08 Mar '21

08 Mar '21

Thanks for reviewing v2 of this patchset! During testing SoundWire controller on SM8250 MTP, we found few issues like all the interrupts are not handled, all transport parameters are not read from device tree. Patch to add Auto Enumeration supported by the controller is also part of this series. Other major issue was register read/writes which was interrupt based was an overhead and puts lot of limitation on context it can be used from. With previous approach number of interrupts generated after enumeration are around 130: $ cat /proc/interrupts | grep soundwire 21: 130 0 0 0 0 0 0 0 GICv3 234 Edge soundwire after this patch they are just 3 interrupts $ cat /proc/interrupts | grep soundwire 21: 3 0 0 0 0 0 0 0 GICv3 234 Edge soundwire So this patchset add various improvements to the existing driver to address above issues. Tested it on SM8250 MTP with 2x WSA881x speakers, HeadPhones on WCD938x via lpass-rx-macro and Analog MICs via lpass-tx-macro. Also tested on DragonBoard DB845c with 2xWSA881x speakers. Changes since v1: - added auto enumeration patches to this patchset - added comments to qcom bindings to clarify invalid values - cleanup trasport parameters to use internal port config Srinivas Kandagatla (9): dt-bindings: soundwire: qcom: clarify data port bus parameters soundwire: qcom: add support to missing transport params soundwire: qcom: set continue execution flag for ignored commands soundwire: qcom: start the clock during initialization soundwire: qcom: update register read/write routine soundwire: qcom: add support to new interrupts soundwire: export sdw_compare_devid() and sdw_extract_slave_id() soundwire: qcom: add auto enumeration support soundwire: qcom: wait for enumeration to be complete in probe .../bindings/soundwire/qcom,sdw.txt | 20 + drivers/soundwire/bus.c | 4 +- drivers/soundwire/qcom.c | 529 ++++++++++++++---- include/linux/soundwire/sdw.h | 2 + 4 files changed, 441 insertions(+), 114 deletions(-) -- 2.21.0

2 11

[PATCH] ALSA: hda: Drop the BATCH workaround for AMD controllers
by Takashi Iwai 08 Mar '21

08 Mar '21

The commit c02f77d32d2c ("ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)") introduced a few workarounds for the recent AMD HD-audio controller, and one of them is the forced BATCH PCM mode so that PulseAudio avoids the timer-based scheduling. This was thought to cover for some badly working applications, but this actually worsens for more others. In total, this wasn't a good idea to enforce it. This is a partial revert of the commit above for dropping the PCM BATCH enforcement part to recover from the regression again. Fixes: c02f77d32d2c ("ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 9087981cd1f7..ca2f2ecd1488 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -609,13 +609,6 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) 20, 178000000); - /* by some reason, the playback stream stalls on PulseAudio with - * tsched=1 when a capture stream triggers. Until we figure out the - * real cause, disable tsched mode by telling the PCM info flag. - */ - if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) - runtime->hw.info |= SNDRV_PCM_INFO_BATCH; - if (chip->align_buffer_size) /* constrain buffer sizes to be multiple of 128 bytes. This is more efficient in terms of memory -- 2.26.2

1 0

[PATCH] ASoC: codecs: lpass-va-macro: mute/unmute all active decimators
by Jonathan Marek 08 Mar '21

08 Mar '21

An interface can have multiple decimators enabled, so loop over all active decimators. Otherwise only one channel will be unmuted, and other channels will be zero. This fixes recording from dual DMIC as a single two channel stream. Also remove the now unused "active_decimator" field. Fixes: 908e6b1df26e ("ASoC: codecs: lpass-va-macro: Add support to VA Macro") Signed-off-by: Jonathan Marek <jonathan(a)marek.ca> --- sound/soc/codecs/lpass-va-macro.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 91e6890d6efcb..3d6976a3d9e42 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -189,7 +189,6 @@ struct va_macro { struct device *dev; unsigned long active_ch_mask[VA_MACRO_MAX_DAIS]; unsigned long active_ch_cnt[VA_MACRO_MAX_DAIS]; - unsigned long active_decimator[VA_MACRO_MAX_DAIS]; u16 dmic_clk_div; int dec_mode[VA_MACRO_NUM_DECIMATORS]; @@ -549,11 +548,9 @@ static int va_macro_tx_mixer_put(struct snd_kcontrol *kcontrol, if (enable) { set_bit(dec_id, &va->active_ch_mask[dai_id]); va->active_ch_cnt[dai_id]++; - va->active_decimator[dai_id] = dec_id; } else { clear_bit(dec_id, &va->active_ch_mask[dai_id]); va->active_ch_cnt[dai_id]--; - va->active_decimator[dai_id] = -1; } snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, enable, update); @@ -880,18 +877,19 @@ static int va_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) struct va_macro *va = snd_soc_component_get_drvdata(component); u16 tx_vol_ctl_reg, decimator; - decimator = va->active_decimator[dai->id]; - - tx_vol_ctl_reg = CDC_VA_TX0_TX_PATH_CTL + - VA_MACRO_TX_PATH_OFFSET * decimator; - if (mute) - snd_soc_component_update_bits(component, tx_vol_ctl_reg, - CDC_VA_TX_PATH_PGA_MUTE_EN_MASK, - CDC_VA_TX_PATH_PGA_MUTE_EN); - else - snd_soc_component_update_bits(component, tx_vol_ctl_reg, - CDC_VA_TX_PATH_PGA_MUTE_EN_MASK, - CDC_VA_TX_PATH_PGA_MUTE_DISABLE); + for_each_set_bit(decimator, &va->active_ch_mask[dai->id], + VA_MACRO_DEC_MAX) { + tx_vol_ctl_reg = CDC_VA_TX0_TX_PATH_CTL + + VA_MACRO_TX_PATH_OFFSET * decimator; + if (mute) + snd_soc_component_update_bits(component, tx_vol_ctl_reg, + CDC_VA_TX_PATH_PGA_MUTE_EN_MASK, + CDC_VA_TX_PATH_PGA_MUTE_EN); + else + snd_soc_component_update_bits(component, tx_vol_ctl_reg, + CDC_VA_TX_PATH_PGA_MUTE_EN_MASK, + CDC_VA_TX_PATH_PGA_MUTE_DISABLE); + } return 0; } -- 2.26.1

3 2

[PATCH] ASoC: codecs: lpass-wsa-macro: fix RX MIX input controls
by Jonathan Marek 08 Mar '21

08 Mar '21

Attempting to use the RX MIX path at 48kHz plays at 96kHz, because these controls are incorrectly toggling the first bit of the register, which is part of the FS_RATE field. Fix the problem by using the same method used by the "WSA RX_MIX EC0_MUX" control, which is to use SND_SOC_NOPM as the register and use an enum in the shift field instead. Fixes: 2c4066e5d428 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route") Signed-off-by: Jonathan Marek <jonathan(a)marek.ca> --- sound/soc/codecs/lpass-wsa-macro.c | 20 +++++++++++--------- 1 file changed, 11 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index f399f4dff5511..bd2561f9fb9fa 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -1211,14 +1211,16 @@ static int wsa_macro_enable_mix_path(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - u16 gain_reg; + u16 path_reg, gain_reg; int val; - switch (w->reg) { - case CDC_WSA_RX0_RX_PATH_MIX_CTL: + switch (w->shift) { + case WSA_MACRO_RX_MIX0: + path_reg = CDC_WSA_RX0_RX_PATH_MIX_CTL; gain_reg = CDC_WSA_RX0_RX_VOL_MIX_CTL; break; - case CDC_WSA_RX1_RX_PATH_MIX_CTL: + case WSA_MACRO_RX_MIX1: + path_reg = CDC_WSA_RX1_RX_PATH_MIX_CTL; gain_reg = CDC_WSA_RX1_RX_VOL_MIX_CTL; break; default: @@ -1231,7 +1233,7 @@ static int wsa_macro_enable_mix_path(struct snd_soc_dapm_widget *w, snd_soc_component_write(component, gain_reg, val); break; case SND_SOC_DAPM_POST_PMD: - snd_soc_component_update_bits(component, w->reg, + snd_soc_component_update_bits(component, path_reg, CDC_WSA_RX_PATH_MIX_CLK_EN_MASK, CDC_WSA_RX_PATH_MIX_CLK_DISABLE); break; @@ -2068,14 +2070,14 @@ static const struct snd_soc_dapm_widget wsa_macro_dapm_widgets[] = { SND_SOC_DAPM_MUX("WSA_RX0 INP0", SND_SOC_NOPM, 0, 0, &rx0_prim_inp0_mux), SND_SOC_DAPM_MUX("WSA_RX0 INP1", SND_SOC_NOPM, 0, 0, &rx0_prim_inp1_mux), SND_SOC_DAPM_MUX("WSA_RX0 INP2", SND_SOC_NOPM, 0, 0, &rx0_prim_inp2_mux), - SND_SOC_DAPM_MUX_E("WSA_RX0 MIX INP", CDC_WSA_RX0_RX_PATH_MIX_CTL, - 0, 0, &rx0_mix_mux, wsa_macro_enable_mix_path, + SND_SOC_DAPM_MUX_E("WSA_RX0 MIX INP", SND_SOC_NOPM, WSA_MACRO_RX_MIX0, + 0, &rx0_mix_mux, wsa_macro_enable_mix_path, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("WSA_RX1 INP0", SND_SOC_NOPM, 0, 0, &rx1_prim_inp0_mux), SND_SOC_DAPM_MUX("WSA_RX1 INP1", SND_SOC_NOPM, 0, 0, &rx1_prim_inp1_mux), SND_SOC_DAPM_MUX("WSA_RX1 INP2", SND_SOC_NOPM, 0, 0, &rx1_prim_inp2_mux), - SND_SOC_DAPM_MUX_E("WSA_RX1 MIX INP", CDC_WSA_RX1_RX_PATH_MIX_CTL, - 0, 0, &rx1_mix_mux, wsa_macro_enable_mix_path, + SND_SOC_DAPM_MUX_E("WSA_RX1 MIX INP", SND_SOC_NOPM, WSA_MACRO_RX_MIX1, + 0, &rx1_mix_mux, wsa_macro_enable_mix_path, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER_E("WSA_RX INT0 MIX", SND_SOC_NOPM, 0, 0, NULL, 0, -- 2.26.1

3 2

[PATCH v2 00/15] Report jack and button detection + Capture Support
by Lucas Tanure 08 Mar '21

08 Mar '21

Hi All, Here is a patch series for reporting to user space jack and button events and add the support for Capture. With some cleanups and fixes along the way. Regards, Lucas Tanure Lucas Tanure (12): ASoC: cs42l42: Fix Bitclock polarity inversion ASoC: cs42l42: Fix channel width support ASoC: cs42l42: Fix mixer volume control ASoC: cs42l42: Don't enable/disable regulator at Bias Level ASoC: cs42l42: Always wait at least 3ms after reset ASoC: cs42l42: Remove power if the driver is being removed ASoC: cs42l42: Disable regulators if probe fails ASoC: cs42l42: Provide finer control on playback path ASoC: cs42l42: Set clock source for both ways of stream ASoC: cs42l42: Add Capture Support ASoC: cs42l42: Report jack and button detection ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called Richard Fitzgerald (3): ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT ASoC: cs42l42: Only start PLL if it is needed ASoC: cs42l42: Wait for PLL to lock before switching to it sound/soc/codecs/cs42l42.c | 437 +++++++++++++++++++++---------------- sound/soc/codecs/cs42l42.h | 41 +++- 2 files changed, 282 insertions(+), 196 deletions(-) -- 2.30.1

2 17

[PATCH 1/2] ASoC: rt5645: Move rt5645_platform_data to sound/soc/codecs/rt5645.c
by Hans de Goede 08 Mar '21

08 Mar '21

sound/soc/codecs/rt5645.c is the only user of the rt5645_platform_data, move its definition to sound/soc/codecs/rt5645.c and remove the now empty include/sound/rt5645.h file. Note since the DMI quirk mechanism uses pointers to the rt5645_platform_data struct we can NOT simply add its members to the rt5645_priv struct and completely remove the struct. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- include/sound/rt5645.h | 32 -------------------------------- sound/soc/codecs/rt5645.c | 23 ++++++++++++++++++++++- sound/soc/codecs/rt5645.h | 2 -- 3 files changed, 22 insertions(+), 35 deletions(-) delete mode 100644 include/sound/rt5645.h diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h deleted file mode 100644 index 710c95be5509..000000000000 --- a/include/sound/rt5645.h +++ /dev/null @@ -1,32 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * linux/sound/rt5645.h -- Platform data for RT5645 - * - * Copyright 2013 Realtek Microelectronics - */ - -#ifndef __LINUX_SND_RT5645_H -#define __LINUX_SND_RT5645_H - -struct rt5645_platform_data { - /* IN2 can optionally be differential */ - bool in2_diff; - - unsigned int dmic1_data_pin; - /* 0 = IN2N; 1 = GPIO5; 2 = GPIO11 */ - unsigned int dmic2_data_pin; - /* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */ - - unsigned int jd_mode; - /* Use level triggered irq */ - bool level_trigger_irq; - /* Invert JD1_1 status polarity */ - bool inv_jd1_1; - /* Invert HP detect status polarity */ - bool inv_hp_pol; - - /* Value to asign to snd_soc_card.long_name */ - const char *long_name; -}; - -#endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 63a7e052eaa0..06ea6fce427c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -411,6 +411,27 @@ static const char *const rt5645_supply_names[] = { "cpvdd", }; +struct rt5645_platform_data { + /* IN2 can optionally be differential */ + bool in2_diff; + + unsigned int dmic1_data_pin; + /* 0 = IN2N; 1 = GPIO5; 2 = GPIO11 */ + unsigned int dmic2_data_pin; + /* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */ + + unsigned int jd_mode; + /* Use level triggered irq */ + bool level_trigger_irq; + /* Invert JD1_1 status polarity */ + bool inv_jd1_1; + /* Invert HP detect status polarity */ + bool inv_hp_pol; + + /* Value to assign to snd_soc_card.long_name */ + const char *long_name; +}; + struct rt5645_priv { struct snd_soc_component *component; struct rt5645_platform_data pdata; @@ -3834,7 +3855,7 @@ static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) static int rt5645_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct rt5645_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5645_platform_data *pdata = NULL; const struct dmi_system_id *dmi_data; struct rt5645_priv *rt5645; int ret, i; diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index e2d72ae17484..ac3de6f3bc2f 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -9,8 +9,6 @@ #ifndef __RT5645_H__ #define __RT5645_H__ -#include <sound/rt5645.h> - /* Info */ #define RT5645_RESET 0x00 #define RT5645_VENDOR_ID 0xfd -- 2.30.1

2 2

[PATCH] ASoC: codecs/jz4770: Remove superfluous error message
by Tang Bin 08 Mar '21

08 Mar '21

The function devm_platform_ioremap_resource has already contained error message if failed, so remove superfluous dev_err here. Signed-off-by: Zhang Shengju <zhangshengju(a)cmss.chinamobile.com> Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/codecs/jz4770.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/jz4770.c b/sound/soc/codecs/jz4770.c index 298689a07..5a24471a5 100644 --- a/sound/soc/codecs/jz4770.c +++ b/sound/soc/codecs/jz4770.c @@ -900,11 +900,8 @@ static int jz4770_codec_probe(struct platform_device *pdev) codec->dev = dev; codec->base = devm_platform_ioremap_resource(pdev, 0); - if (IS_ERR(codec->base)) { - ret = PTR_ERR(codec->base); - dev_err(dev, "Failed to ioremap mmio memory: %d\n", ret); - return ret; - } + if (IS_ERR(codec->base)) + return PTR_ERR(codec->base); codec->regmap = devm_regmap_init(dev, NULL, codec, &jz4770_codec_regmap_config); -- 2.20.1.windows.1

3 2

[PATCH 1/2] ASoC: remove remnants of sirf prima/atlas audio codec
by Peter Robinson 08 Mar '21

08 Mar '21

In 61fbeb5 the sirf prima/atlas drivers were removed. This cleans up a stray header and some Kconfig entries for the codec that were missed in the process. Fixes: 61fbeb5dcb3d (ASoC: remove sirf prima/atlas drivers) Signed-off-by: Peter Robinson <pbrobinson(a)gmail.com> Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 5 -- sound/soc/codecs/sirf-audio-codec.h | 124 ---------------------------- 2 files changed, 129 deletions(-) delete mode 100644 sound/soc/codecs/sirf-audio-codec.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e4cf14e66a51..1c87b42606c9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -186,7 +186,6 @@ config SND_SOC_ALL_CODECS imply SND_SOC_SI476X imply SND_SOC_SIMPLE_AMPLIFIER imply SND_SOC_SIMPLE_MUX - imply SND_SOC_SIRF_AUDIO_CODEC imply SND_SOC_SPDIF imply SND_SOC_SSM2305 imply SND_SOC_SSM2518 @@ -1279,10 +1278,6 @@ config SND_SOC_SIMPLE_MUX tristate "Simple Audio Mux" select GPIOLIB -config SND_SOC_SIRF_AUDIO_CODEC - tristate "SiRF SoC internal audio codec" - select REGMAP_MMIO - config SND_SOC_SPDIF tristate "S/PDIF CODEC" diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h deleted file mode 100644 index a7fe2680f4c7..000000000000 --- a/sound/soc/codecs/sirf-audio-codec.h +++ /dev/null @@ -1,124 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-or-later */ -/* - * SiRF inner codec controllers define - * - * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. - */ - -#ifndef _SIRF_AUDIO_CODEC_H -#define _SIRF_AUDIO_CODEC_H - - -#define AUDIO_IC_CODEC_PWR (0x00E0) -#define AUDIO_IC_CODEC_CTRL0 (0x00E4) -#define AUDIO_IC_CODEC_CTRL1 (0x00E8) -#define AUDIO_IC_CODEC_CTRL2 (0x00EC) -#define AUDIO_IC_CODEC_CTRL3 (0x00F0) - -#define MICBIASEN (1 << 3) - -#define IC_RDACEN (1 << 0) -#define IC_LDACEN (1 << 1) -#define IC_HSREN (1 << 2) -#define IC_HSLEN (1 << 3) -#define IC_SPEN (1 << 4) -#define IC_CPEN (1 << 5) - -#define IC_HPRSELR (1 << 6) -#define IC_HPLSELR (1 << 7) -#define IC_HPRSELL (1 << 8) -#define IC_HPLSELL (1 << 9) -#define IC_SPSELR (1 << 10) -#define IC_SPSELL (1 << 11) - -#define IC_MONOR (1 << 12) -#define IC_MONOL (1 << 13) - -#define IC_RXOSRSEL (1 << 28) -#define IC_CPFREQ (1 << 29) -#define IC_HSINVEN (1 << 30) - -#define IC_MICINREN (1 << 0) -#define IC_MICINLEN (1 << 1) -#define IC_MICIN1SEL (1 << 2) -#define IC_MICIN2SEL (1 << 3) -#define IC_MICDIFSEL (1 << 4) -#define IC_LINEIN1SEL (1 << 5) -#define IC_LINEIN2SEL (1 << 6) -#define IC_RADCEN (1 << 7) -#define IC_LADCEN (1 << 8) -#define IC_ALM (1 << 9) - -#define IC_DIGMICEN (1 << 22) -#define IC_DIGMICFREQ (1 << 23) -#define IC_ADC14B_12 (1 << 24) -#define IC_FIRDAC_HSL_EN (1 << 25) -#define IC_FIRDAC_HSR_EN (1 << 26) -#define IC_FIRDAC_LOUT_EN (1 << 27) -#define IC_POR (1 << 28) -#define IC_CODEC_CLK_EN (1 << 29) -#define IC_HP_3DB_BOOST (1 << 30) - -#define IC_ADC_LEFT_GAIN_SHIFT 16 -#define IC_ADC_RIGHT_GAIN_SHIFT 10 -#define IC_ADC_GAIN_MASK 0x3F -#define IC_MIC_MAX_GAIN 0x39 - -#define IC_RXPGAR_MASK 0x3F -#define IC_RXPGAR_SHIFT 14 -#define IC_RXPGAL_MASK 0x3F -#define IC_RXPGAL_SHIFT 21 -#define IC_RXPGAR 0x7B -#define IC_RXPGAL 0x7B - -#define AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK 0x3F -#define AUDIO_PORT_TX_FIFO_SC_OFFSET 0 -#define AUDIO_PORT_TX_FIFO_LC_OFFSET 10 -#define AUDIO_PORT_TX_FIFO_HC_OFFSET 20 - -#define TX_FIFO_SC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_SC_OFFSET) -#define TX_FIFO_LC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_LC_OFFSET) -#define TX_FIFO_HC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_HC_OFFSET) - -#define AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK 0x0F -#define AUDIO_PORT_RX_FIFO_SC_OFFSET 0 -#define AUDIO_PORT_RX_FIFO_LC_OFFSET 10 -#define AUDIO_PORT_RX_FIFO_HC_OFFSET 20 - -#define RX_FIFO_SC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_SC_OFFSET) -#define RX_FIFO_LC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_LC_OFFSET) -#define RX_FIFO_HC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_HC_OFFSET) -#define AUDIO_PORT_IC_CODEC_TX_CTRL (0x00F4) -#define AUDIO_PORT_IC_CODEC_RX_CTRL (0x00F8) - -#define AUDIO_PORT_IC_TXFIFO_OP (0x00FC) -#define AUDIO_PORT_IC_TXFIFO_LEV_CHK (0x0100) -#define AUDIO_PORT_IC_TXFIFO_STS (0x0104) -#define AUDIO_PORT_IC_TXFIFO_INT (0x0108) -#define AUDIO_PORT_IC_TXFIFO_INT_MSK (0x010C) - -#define AUDIO_PORT_IC_RXFIFO_OP (0x0110) -#define AUDIO_PORT_IC_RXFIFO_LEV_CHK (0x0114) -#define AUDIO_PORT_IC_RXFIFO_STS (0x0118) -#define AUDIO_PORT_IC_RXFIFO_INT (0x011C) -#define AUDIO_PORT_IC_RXFIFO_INT_MSK (0x0120) - -#define AUDIO_FIFO_START (1 << 0) -#define AUDIO_FIFO_RESET (1 << 1) - -#define AUDIO_FIFO_FULL (1 << 0) -#define AUDIO_FIFO_EMPTY (1 << 1) -#define AUDIO_FIFO_OFLOW (1 << 2) -#define AUDIO_FIFO_UFLOW (1 << 3) - -#define IC_TX_ENABLE (0x03) -#define IC_RX_ENABLE_MONO (0x01) -#define IC_RX_ENABLE_STEREO (0x03) - -#endif /*__SIRF_AUDIO_CODEC_H*/ -- 2.29.2

3 4

[PATCH v2] ASoC: wm8962: Relax bit clock divider searching
by Shengjiu Wang 08 Mar '21

08 Mar '21

With S20_3LE format case, the sysclk = rate * 384, the bclk = rate * 20 * 2, there is no proper bclk divider for 384 / 40, because current condition needs exact match. So driver fails to configure the clocking: wm8962 3-001a: Unsupported BCLK ratio 9 Fix this by relaxing bitclk divider searching, so that when no exact value can be derived from sysclk pick the closest value greater than expected bitclk. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- changes in v2: - Add debug message according to Charles's comments sound/soc/codecs/wm8962.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index ce4666a74793..34080f497584 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2403,6 +2403,7 @@ static const int sysclk_rates[] = { static void wm8962_configure_bclk(struct snd_soc_component *component) { struct wm8962_priv *wm8962 = snd_soc_component_get_drvdata(component); + int best, min_diff, diff; int dspclk, i; int clocking2 = 0; int clocking4 = 0; @@ -2473,23 +2474,25 @@ static void wm8962_configure_bclk(struct snd_soc_component *component) dev_dbg(component->dev, "DSPCLK is %dHz, BCLK %d\n", dspclk, wm8962->bclk); - /* We're expecting an exact match */ + /* Search a proper bclk, not exact match. */ + best = 0; + min_diff = INT_MAX; for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { if (bclk_divs[i] < 0) continue; - if (dspclk / bclk_divs[i] == wm8962->bclk) { - dev_dbg(component->dev, "Selected BCLK_DIV %d for %dHz\n", - bclk_divs[i], wm8962->bclk); - clocking2 |= i; + diff = (dspclk / bclk_divs[i]) - wm8962->bclk; + if (diff < 0) /* Table is sorted */ break; + if (diff < min_diff) { + best = i; + min_diff = diff; } } - if (i == ARRAY_SIZE(bclk_divs)) { - dev_err(component->dev, "Unsupported BCLK ratio %d\n", - dspclk / wm8962->bclk); - return; - } + wm8962->bclk = dspclk / bclk_divs[best]; + clocking2 |= best; + dev_dbg(component->dev, "Selected BCLK_DIV %d for %dHz\n", + bclk_divs[best], wm8962->bclk); aif2 |= wm8962->bclk / wm8962->lrclk; dev_dbg(component->dev, "Selected LRCLK divisor %d for %dHz\n", -- 2.27.0

3 2

[PATCH] ASoC: rt711-sdca: Add RT711 SDCA vendor-specific driver
by shumingf＠realtek.com 08 Mar '21

08 Mar '21

From: Shuming Fan <shumingf(a)realtek.com> This is the initial codec driver for rt711 SDCA version. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt711-sdca-sdw.c | 425 ++++++++ sound/soc/codecs/rt711-sdca-sdw.h | 99 ++ sound/soc/codecs/rt711-sdca.c | 1492 +++++++++++++++++++++++++++++ sound/soc/codecs/rt711-sdca.h | 238 +++++ 6 files changed, 2263 insertions(+) create mode 100644 sound/soc/codecs/rt711-sdca-sdw.c create mode 100644 sound/soc/codecs/rt711-sdca-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca.c create mode 100644 sound/soc/codecs/rt711-sdca.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ae833a54bcb3..8de56531ffb4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -180,6 +180,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT5682_SDW imply SND_SOC_RT700_SDW imply SND_SOC_RT711_SDW + imply SND_SOC_RT711_SDCA_SDW imply SND_SOC_RT715_SDW imply SND_SOC_RT1308_SDW imply SND_SOC_RT1316_SDW @@ -1248,6 +1249,12 @@ config SND_SOC_RT711_SDW select SND_SOC_RT711 select REGMAP_SOUNDWIRE +config SND_SOC_RT711_SDCA_SDW + tristate "Realtek RT711 SDCA Codec - SDW" + depends on SOUNDWIRE + select REGMAP_SOUNDWIRE + select REGMAP_SOUNDWIRE_MBQ + config SND_SOC_RT715 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1585bc1d9a25..3ad169fb22c4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -197,6 +197,7 @@ snd-soc-rt5682-sdw-objs := rt5682-sdw.o snd-soc-rt5682-i2c-objs := rt5682-i2c.o snd-soc-rt700-objs := rt700.o rt700-sdw.o snd-soc-rt711-objs := rt711.o rt711-sdw.o +snd-soc-rt711-sdca-objs := rt711-sdca.o rt711-sdca-sdw.o snd-soc-rt715-objs := rt715.o rt715-sdw.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -513,6 +514,7 @@ obj-$(CONFIG_SND_SOC_RT5682_I2C) += snd-soc-rt5682-i2c.o obj-$(CONFIG_SND_SOC_RT5682_SDW) += snd-soc-rt5682-sdw.o obj-$(CONFIG_SND_SOC_RT700) += snd-soc-rt700.o obj-$(CONFIG_SND_SOC_RT711) += snd-soc-rt711.o +obj-$(CONFIG_SND_SOC_RT711_SDCA_SDW) += snd-soc-rt711-sdca.o obj-$(CONFIG_SND_SOC_RT715) += snd-soc-rt715.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c new file mode 100644 index 000000000000..9685c8905468 --- /dev/null +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -0,0 +1,425 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt711-sdw-sdca.c -- rt711 SDCA ALSA SoC audio driver +// +// Copyright(c) 2021 Realtek Semiconductor Corp. +// +// + +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/mod_devicetable.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/module.h> + +#include "rt711-sdca.h" +#include "rt711-sdca-sdw.h" + +static bool rt711_sdca_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x201a ... 0x2027: + case 0x2029 ... 0x202a: + case 0x202d ... 0x2034: + case 0x2200 ... 0x2204: + case 0x2206 ... 0x2212: + case 0x2220 ... 0x2223: + case 0x2230 ... 0x2239: + case 0x2f01 ... 0x2f0f: + case 0x2f30 ... 0x2f36: + case 0x2f50 ... 0x2f5a: + case 0x2f60: + case 0x3200 ... 0x3212: + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_GE49, RT711_SDCA_CTL_SELECTED_MODE, 0): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_GE49, RT711_SDCA_CTL_DETECTED_MODE, 0): + case SDW_SDCA_CTL(FUNC_NUM_HID, RT711_SDCA_ENT_HID01, RT711_SDCA_CTL_HIDTX_CURRENT_OWNER, 0) ... + SDW_SDCA_CTL(FUNC_NUM_HID, RT711_SDCA_ENT_HID01, RT711_SDCA_CTL_HIDTX_MESSAGE_LENGTH, 0): + case RT711_BUF_ADDR_HID1 ... RT711_BUF_ADDR_HID2: + return true; + default: + return false; + } +} + +static bool rt711_sdca_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x201b: + case 0x201c: + case 0x201d: + case 0x201f: + case 0x2021: + case 0x2023: + case 0x2230: + case 0x202d ... 0x202f: /* BRA */ + case 0x2200 ... 0x2212: /* i2c debug */ + case RT711_RC_CAL_STATUS: + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_GE49, RT711_SDCA_CTL_DETECTED_MODE, 0): + case SDW_SDCA_CTL(FUNC_NUM_HID, RT711_SDCA_ENT_HID01, RT711_SDCA_CTL_HIDTX_CURRENT_OWNER, 0) ... + SDW_SDCA_CTL(FUNC_NUM_HID, RT711_SDCA_ENT_HID01, RT711_SDCA_CTL_HIDTX_MESSAGE_LENGTH, 0): + case RT711_BUF_ADDR_HID1 ... RT711_BUF_ADDR_HID2: + return true; + default: + return false; + } +} + +static bool rt711_sdca_mbq_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2000000 ... 0x20000ff: + case 0x5600000 ... 0x56000ff: + case 0x5700000 ... 0x57000ff: + case 0x5800000 ... 0x58000ff: + case 0x5900000 ... 0x59000ff: + case 0x5b00000 ... 0x5b000ff: + case 0x5f00000 ... 0x5f000ff: + case 0x6100000 ... 0x61000ff: + return true; + default: + return false; + } +} + +static bool rt711_sdca_mbq_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2000000: + case 0x200001a: + case 0x2000046: + case 0x2000080: + case 0x2000081: + case 0x2000083: + case 0x5800000: + case 0x5800001: + case 0x5f00001: + case 0x6100008: + return true; + default: + return false; + } +} + +static const struct regmap_config rt711_sdca_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = rt711_sdca_readable_register, + .volatile_reg = rt711_sdca_volatile_register, + .max_register = 0x44ffffff, + .reg_defaults = rt711_sdca_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rt711_sdca_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct regmap_config rt711_sdca_mbq_regmap = { + .name = "sdw-mbq", + .reg_bits = 32, + .val_bits = 16, + .readable_reg = rt711_sdca_mbq_readable_register, + .volatile_reg = rt711_sdca_mbq_volatile_register, + .max_register = 0x40800f12, + .reg_defaults = rt711_sdca_mbq_defaults, + .num_reg_defaults = ARRAY_SIZE(rt711_sdca_mbq_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt711_sdca_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt711_sdca_priv *rt711 = dev_get_drvdata(&slave->dev); + + /* Update the status */ + rt711->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + rt711->hw_init = false; + + if (status == SDW_SLAVE_ATTACHED) { + if (rt711->hs_jack) { + /* + * Due to the SCP_SDCA_INTMASK will be cleared by any reset, and then + * if the device attached again, we will need to set the setting back. + * It could avoid losing the jack detection interrupt. + * This also could sync with the cache value as the rt711_sdca_jack_init set. + */ + sdw_write_no_pm(rt711->slave, SDW_SCP_SDCA_INTMASK1, + SDW_SCP_SDCA_INTMASK_SDCA_0); + sdw_write_no_pm(rt711->slave, SDW_SCP_SDCA_INTMASK2, + SDW_SCP_SDCA_INTMASK_SDCA_8); + } + } + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt711->hw_init || rt711->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt711_sdca_io_init(&slave->dev, slave); +} + +static int rt711_sdca_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + prop->is_sdca = true; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x14; /* BITMAP: 00010100 */ + prop->sink_ports = 0x8; /* BITMAP: 00001000 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + /* wake-up event */ + prop->wake_capable = 1; + + return 0; +} + +static int rt711_sdca_interrupt_callback(struct sdw_slave *slave, + struct sdw_slave_intr_status *status) +{ + struct rt711_sdca_priv *rt711 = dev_get_drvdata(&slave->dev); + int ret, stat; + int count = 0, retry = 3; + unsigned int sdca_cascade, scp_sdca_stat1, scp_sdca_stat2 = 0; + + dev_dbg(&slave->dev, + "%s control_port_stat=%x, sdca_cascade=%x", __func__, + status->control_port, status->sdca_cascade); + + if (cancel_delayed_work_sync(&rt711->jack_detect_work)) { + dev_warn(&slave->dev, "%s the pending delayed_work was cancelled", __func__); + /* avoid the HID owner doesn't change to device */ + if (rt711->scp_sdca_stat2) + scp_sdca_stat2 = rt711->scp_sdca_stat2; + } + + ret = sdw_read_no_pm(rt711->slave, SDW_SCP_SDCA_INT1); + if (ret < 0) + goto io_error; + rt711->scp_sdca_stat1 = ret; + ret = sdw_read_no_pm(rt711->slave, SDW_SCP_SDCA_INT2); + if (ret < 0) + goto io_error; + rt711->scp_sdca_stat2 = ret; + if (scp_sdca_stat2) + rt711->scp_sdca_stat2 |= scp_sdca_stat2; + + do { + /* clear flag */ + ret = sdw_read_no_pm(rt711->slave, SDW_SCP_SDCA_INT1); + if (ret < 0) + goto io_error; + if (ret & SDW_SCP_SDCA_INTMASK_SDCA_0) { + ret = sdw_write_no_pm(rt711->slave, SDW_SCP_SDCA_INT1, + SDW_SCP_SDCA_INTMASK_SDCA_0); + if (ret < 0) + goto io_error; + } + ret = sdw_read_no_pm(rt711->slave, SDW_SCP_SDCA_INT2); + if (ret < 0) + goto io_error; + if (ret & SDW_SCP_SDCA_INTMASK_SDCA_8) { + ret = sdw_write_no_pm(rt711->slave, SDW_SCP_SDCA_INT2, + SDW_SCP_SDCA_INTMASK_SDCA_8); + if (ret < 0) + goto io_error; + } + + /* check if flag clear or not */ + ret = sdw_read_no_pm(rt711->slave, SDW_DP0_INT); + if (ret < 0) + goto io_error; + sdca_cascade = ret & SDW_DP0_SDCA_CASCADE; + + ret = sdw_read_no_pm(rt711->slave, SDW_SCP_SDCA_INT1); + if (ret < 0) + goto io_error; + scp_sdca_stat1 = ret & SDW_SCP_SDCA_INTMASK_SDCA_0; + + ret = sdw_read_no_pm(rt711->slave, SDW_SCP_SDCA_INT2); + if (ret < 0) + goto io_error; + scp_sdca_stat2 = ret & SDW_SCP_SDCA_INTMASK_SDCA_8; + + stat = scp_sdca_stat1 || scp_sdca_stat2 || sdca_cascade; + + count++; + } while (stat != 0 && count < retry); + + if (stat) + dev_warn(&slave->dev, + "%s scp_sdca_stat1=0x%x, scp_sdca_stat2=0x%x\n", __func__, + rt711->scp_sdca_stat1, rt711->scp_sdca_stat2); + + if (status->sdca_cascade) + mod_delayed_work(system_power_efficient_wq, + &rt711->jack_detect_work, msecs_to_jiffies(30)); + + return 0; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); + return ret; +} + +static struct sdw_slave_ops rt711_sdca_slave_ops = { + .read_prop = rt711_sdca_read_prop, + .interrupt_callback = rt711_sdca_interrupt_callback, + .update_status = rt711_sdca_update_status, +}; + +static int rt711_sdca_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *regmap, *mbq_regmap; + + /* Regmap Initialization */ + mbq_regmap = devm_regmap_init_sdw_mbq(slave, &rt711_sdca_mbq_regmap); + if (IS_ERR(mbq_regmap)) + return PTR_ERR(mbq_regmap); + + regmap = devm_regmap_init_sdw(slave, &rt711_sdca_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return rt711_sdca_init(&slave->dev, regmap, mbq_regmap, slave); +} + +static int rt711_sdca_sdw_remove(struct sdw_slave *slave) +{ + struct rt711_sdca_priv *rt711 = dev_get_drvdata(&slave->dev); + + if (rt711 && rt711->hw_init) { + cancel_delayed_work_sync(&rt711->jack_detect_work); + cancel_delayed_work_sync(&rt711->jack_btn_check_work); + } + + return 0; +} + +static const struct sdw_device_id rt711_sdca_id[] = { + SDW_SLAVE_ENTRY_EXT(0x025d, 0x711, 0x3, 0x1, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt711_sdca_id); + +static int __maybe_unused rt711_sdca_dev_suspend(struct device *dev) +{ + struct rt711_sdca_priv *rt711 = dev_get_drvdata(dev); + + if (!rt711->hw_init) + return 0; + + cancel_delayed_work_sync(&rt711->jack_detect_work); + cancel_delayed_work_sync(&rt711->jack_btn_check_work); + + regcache_cache_only(rt711->regmap, true); + regcache_cache_only(rt711->mbq_regmap, true); + + return 0; +} + +#define RT711_PROBE_TIMEOUT 5000 + +static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt711_sdca_priv *rt711 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt711->hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT711_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt711->regmap, false); + regcache_sync(rt711->regmap); + regcache_cache_only(rt711->mbq_regmap, false); + regcache_sync(rt711->mbq_regmap); + return 0; +} + +static const struct dev_pm_ops rt711_sdca_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt711_sdca_dev_suspend, rt711_sdca_dev_resume) + SET_RUNTIME_PM_OPS(rt711_sdca_dev_suspend, rt711_sdca_dev_resume, NULL) +}; + +static struct sdw_driver rt711_sdca_sdw_driver = { + .driver = { + .name = "rt711-sdca", + .owner = THIS_MODULE, + .pm = &rt711_sdca_pm, + }, + .probe = rt711_sdca_sdw_probe, + .remove = rt711_sdca_sdw_remove, + .ops = &rt711_sdca_slave_ops, + .id_table = rt711_sdca_id, +}; +module_sdw_driver(rt711_sdca_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT711 SDCA SDW driver"); +MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt711-sdca-sdw.h b/sound/soc/codecs/rt711-sdca-sdw.h new file mode 100644 index 000000000000..0d774e473ab9 --- /dev/null +++ b/sound/soc/codecs/rt711-sdca-sdw.h @@ -0,0 +1,99 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt711-sdw-sdca.h -- RT711 SDCA ALSA SoC audio driver header + * + * Copyright(c) 2021 Realtek Semiconductor Corp. + */ + +#ifndef __RT711_SDW_SDCA_H__ +#define __RT711_SDW_SDCA_H__ + +#include <linux/regmap.h> +#include <linux/soundwire/sdw_registers.h> + +static const struct reg_default rt711_sdca_reg_defaults[] = { + { 0x201a, 0x00 }, + { 0x201e, 0x00 }, + { 0x201f, 0x00 }, + { 0x2020, 0x00 }, + { 0x2021, 0x00 }, + { 0x2022, 0x00 }, + { 0x2023, 0x00 }, + { 0x2024, 0x00 }, + { 0x2025, 0x01 }, + { 0x2026, 0x00 }, + { 0x2027, 0x00 }, + { 0x2029, 0x00 }, + { 0x202a, 0x00 }, + { 0x202d, 0x00 }, + { 0x202e, 0x00 }, + { 0x202f, 0x00 }, + { 0x2030, 0x00 }, + { 0x2031, 0x00 }, + { 0x2032, 0x00 }, + { 0x2033, 0x00 }, + { 0x2230, 0x00 }, + { 0x2231, 0x2f }, + { 0x2232, 0x80 }, + { 0x2233, 0x00 }, + { 0x2234, 0x00 }, + { 0x2235, 0x00 }, + { 0x2236, 0x00 }, + { 0x2237, 0x00 }, + { 0x2238, 0x00 }, + { 0x2239, 0x00 }, + { 0x2f01, 0x00 }, + { 0x2f02, 0x09 }, + { 0x2f03, 0x00 }, + { 0x2f04, 0x00 }, + { 0x2f05, 0x0b }, + { 0x2f06, 0x01 }, + { 0x2f08, 0x00 }, + { 0x2f09, 0x00 }, + { 0x2f0a, 0x00 }, + { 0x2f0b, 0x00 }, + { 0x2f0c, 0x00 }, + { 0x2f0d, 0x00 }, + { 0x2f0e, 0x14 }, + { 0x2f0f, 0x00 }, + { 0x2f50, 0x03 }, + { 0x2f5a, 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_CS01, RT711_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, RT711_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, RT711_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_PDE28, RT711_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, +}; + +static const struct reg_default rt711_sdca_mbq_defaults[] = { + { 0x2000009, 0x1029 }, + { 0x2000011, 0x007a }, + { 0x200001a, 0x8003 }, + { 0x2000045, 0x5289 }, + { 0x2000048, 0x8049 }, + { 0x200004a, 0xa83b }, + { 0x200006b, 0x5064 }, + { 0x200006f, 0x058b }, + { 0x5800000, 0x0008 }, + { 0x5800001, 0x0000 }, + { 0x5f00001, 0x000a }, + { 0x6100000, 0x6100 }, + { 0x6100035, 0x0060 }, + { 0x6100036, 0x0029 }, + { 0x610003f, 0xff12 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, RT711_SDCA_CTL_FU_VOLUME, CH_L), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, RT711_SDCA_CTL_FU_VOLUME, CH_R), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_L), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_R), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_VOLUME, CH_L), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_VOLUME, CH_R), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_PLATFORM_FU44, RT711_SDCA_CTL_FU_CH_GAIN, CH_L), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_PLATFORM_FU44, RT711_SDCA_CTL_FU_CH_GAIN, CH_R), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_PLATFORM_FU15, RT711_SDCA_CTL_FU_CH_GAIN, CH_L), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_PLATFORM_FU15, RT711_SDCA_CTL_FU_CH_GAIN, CH_R), 0x00 }, +}; + +#endif /* __RT711_SDW_SDCA_H__ */ diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c new file mode 100644 index 000000000000..381893b640c6 --- /dev/null +++ b/sound/soc/codecs/rt711-sdca.c @@ -0,0 +1,1492 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt711-sdca.c -- rt711 SDCA ALSA SoC audio driver +// +// Copyright(c) 2021 Realtek Semiconductor Corp. +// +// + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm_runtime.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/slab.h> +#include <linux/bitops.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/jack.h> + +#include "rt711-sdca.h" + +static int rt711_sdca_index_write(struct rt711_sdca_priv *rt711, + unsigned int nid, unsigned int reg, unsigned int value) +{ + int ret; + struct regmap *regmap = rt711->mbq_regmap; + unsigned int addr = (nid << 20) | reg; + + ret = regmap_write(regmap, addr, value); + if (ret < 0) + dev_err(rt711->component->dev, + "Failed to set private value: %06x <= %04x ret=%d\n", + addr, value, ret); + + return ret; +} + +static int rt711_sdca_index_read(struct rt711_sdca_priv *rt711, + unsigned int nid, unsigned int reg, unsigned int *value) +{ + int ret; + struct regmap *regmap = rt711->mbq_regmap; + unsigned int addr = (nid << 20) | reg; + + ret = regmap_read(regmap, addr, value); + if (ret < 0) + dev_err(rt711->component->dev, + "Failed to get private value: %06x => %04x ret=%d\n", + addr, *value, ret); + + return ret; +} + +static int rt711_sdca_index_update_bits(struct rt711_sdca_priv *rt711, + unsigned int nid, unsigned int reg, unsigned int mask, unsigned int val) +{ + unsigned int tmp; + int ret; + + ret = rt711_sdca_index_read(rt711, nid, reg, &tmp); + if (ret < 0) + return ret; + + set_mask_bits(&tmp, mask, val); + return rt711_sdca_index_write(rt711, nid, reg, tmp); +} + +static void rt711_sdca_reset(struct rt711_sdca_priv *rt711) +{ + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_REG, + RT711_PARA_VERB_CTL, RT711_HIDDEN_REG_SW_RESET, + RT711_HIDDEN_REG_SW_RESET); + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, + RT711_HDA_LEGACY_RESET_CTL, 0x1, 0x1); +} + +static int rt711_sdca_calibration(struct rt711_sdca_priv *rt711) +{ + unsigned int val, loop_rc = 0, loop_dc = 0; + struct device *dev; + struct regmap *regmap = rt711->regmap; + int chk_cnt = 100; + int ret = 0; + + mutex_lock(&rt711->calibrate_mutex); + dev = regmap_get_device(regmap); + + regmap_read(rt711->regmap, RT711_RC_CAL_STATUS, &val); + /* RC calibration */ + if (!(val & 0x40)) + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_ANALOG_CTL, + RT711_MISC_POWER_CTL0, 0x0010, 0x0010); + + for (loop_rc = 0; loop_rc < chk_cnt && !(val & 0x40); loop_rc++) { + usleep_range(10000, 11000); + ret = regmap_read(rt711->regmap, RT711_RC_CAL_STATUS, &val); + if (ret < 0) + goto _cali_fail_; + } + if (loop_rc == chk_cnt) + dev_err(dev, "%s, RC calibration time-out!\n", __func__); + + /* HP calibration by manual mode setting */ + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_REG, + RT711_FSM_CTL, 0x2000, 0x2000); + + /* Calibration manual mode */ + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_REG, + RT711_FSM_CTL, 0xf, RT711_CALI_CTL); + + /* reset HP calibration */ + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_CALI, + RT711_DAC_DC_CALI_CTL1, RT711_DAC_DC_FORCE_CALI_RST, 0x00); + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_CALI, + RT711_DAC_DC_CALI_CTL1, RT711_DAC_DC_FORCE_CALI_RST, + RT711_DAC_DC_FORCE_CALI_RST); + + /* cal_clk_en_reg */ + if (rt711->hw_ver == RT711_VER_VD0) + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_CALI, + RT711_DAC_DC_CALI_CTL1, RT711_DAC_DC_CALI_CLK_EN, + RT711_DAC_DC_CALI_CLK_EN); + + /* trigger */ + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_CALI, + RT711_DAC_DC_CALI_CTL1, RT711_DAC_DC_CALI_TRIGGER, + RT711_DAC_DC_CALI_TRIGGER); + + /* wait for calibration process */ + rt711_sdca_index_read(rt711, RT711_VENDOR_CALI, + RT711_DAC_DC_CALI_CTL1, &val); + + for (loop_dc = 0; loop_dc < chk_cnt && + (val & RT711_DAC_DC_CALI_TRIGGER); loop_dc++) { + usleep_range(10000, 11000); + ret = rt711_sdca_index_read(rt711, RT711_VENDOR_CALI, + RT711_DAC_DC_CALI_CTL1, &val); + if (ret < 0) + goto _cali_fail_; + } + if (loop_dc == chk_cnt) + dev_err(dev, "%s, calibration time-out!\n", __func__); + + if (loop_dc == chk_cnt || loop_rc == chk_cnt) + ret = -ETIMEDOUT; + +_cali_fail_: + /* enable impedance sense */ + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_REG, + RT711_FSM_CTL, RT711_FSM_IMP_EN, RT711_FSM_IMP_EN); + + /* release HP-JD and trigger FSM */ + rt711_sdca_index_write(rt711, RT711_VENDOR_REG, + RT711_DIGITAL_MISC_CTRL4, 0x201b); + + mutex_unlock(&rt711->calibrate_mutex); + dev_dbg(dev, "%s calibration complete, ret=%d\n", __func__, ret); + return ret; +} + +static unsigned int rt711_sdca_button_detect(struct rt711_sdca_priv *rt711) +{ + unsigned int btn_type = 0, offset, idx, val, owner; + int ret; + unsigned char buf[3]; + + /* get current UMP message owner */ + ret = regmap_read(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_HID, RT711_SDCA_ENT_HID01, RT711_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), + &owner); + if (ret < 0) + return 0; + + /* if owner is device then there is no button event from device */ + if (owner == 1) + return 0; + + /* read UMP message offset */ + ret = regmap_read(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_HID, RT711_SDCA_ENT_HID01, RT711_SDCA_CTL_HIDTX_MESSAGE_OFFSET, 0), + &offset); + if (ret < 0) + goto _end_btn_det_; + + for (idx = 0; idx < sizeof(buf); idx++) { + ret = regmap_read(rt711->regmap, + RT711_BUF_ADDR_HID1 + offset + idx, &val); + if (ret < 0) + goto _end_btn_det_; + buf[idx] = val & 0xff; + } + + if (buf[0] == 0x11) { + switch (buf[1] & 0xf0) { + case 0x10: + btn_type |= SND_JACK_BTN_2; + break; + case 0x20: + btn_type |= SND_JACK_BTN_3; + break; + case 0x40: + btn_type |= SND_JACK_BTN_0; + break; + case 0x80: + btn_type |= SND_JACK_BTN_1; + break; + } + switch (buf[2]) { + case 0x01: + case 0x10: + btn_type |= SND_JACK_BTN_2; + break; + case 0x02: + case 0x20: + btn_type |= SND_JACK_BTN_3; + break; + case 0x04: + case 0x40: + btn_type |= SND_JACK_BTN_0; + break; + case 0x08: + case 0x80: + btn_type |= SND_JACK_BTN_1; + break; + } + } + +_end_btn_det_: + /* Host is owner, so set back to device */ + if (owner == 0) + /* set owner to device */ + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_HID, RT711_SDCA_ENT_HID01, + RT711_SDCA_CTL_HIDTX_SET_OWNER_TO_DEVICE, 0), 0x01); + + return btn_type; +} + +static int rt711_sdca_headset_detect(struct rt711_sdca_priv *rt711) +{ + unsigned int det_mode; + int ret; + + /* get detected_mode */ + ret = regmap_read(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_GE49, RT711_SDCA_CTL_DETECTED_MODE, 0), + &det_mode); + if (ret < 0) + goto io_error; + + switch (det_mode) { + case 0x00: + rt711->jack_type = 0; + break; + case 0x03: + rt711->jack_type = SND_JACK_HEADPHONE; + break; + case 0x05: + rt711->jack_type = SND_JACK_HEADSET; + break; + } + + /* write selected_mode */ + if (det_mode) { + ret = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_GE49, RT711_SDCA_CTL_SELECTED_MODE, 0), + det_mode); + if (ret < 0) + goto io_error; + } + + dev_dbg(&rt711->slave->dev, + "%s, detected_mode=0x%x\n", __func__, det_mode); + + return 0; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); + return ret; +} + +static void rt711_sdca_jack_detect_handler(struct work_struct *work) +{ + struct rt711_sdca_priv *rt711 = + container_of(work, struct rt711_sdca_priv, jack_detect_work.work); + int btn_type = 0, ret; + + if (!rt711->hs_jack) + return; + + if (!rt711->component->card->instantiated) + return; + + /* SDW_SCP_SDCA_INT_SDCA_0 is used for jack detection */ + if (rt711->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_0) { + ret = rt711_sdca_headset_detect(rt711); + if (ret < 0) + return; + } + + /* SDW_SCP_SDCA_INT_SDCA_8 is used for button detection */ + if (rt711->scp_sdca_stat2 & SDW_SCP_SDCA_INT_SDCA_8) + btn_type = rt711_sdca_button_detect(rt711); + + if (rt711->jack_type == 0) + btn_type = 0; + + dev_dbg(&rt711->slave->dev, + "in %s, jack_type=0x%x\n", __func__, rt711->jack_type); + dev_dbg(&rt711->slave->dev, + "in %s, btn_type=0x%x\n", __func__, btn_type); + dev_dbg(&rt711->slave->dev, + "in %s, scp_sdca_stat1=0x%x, scp_sdca_stat2=0x%x\n", __func__, + rt711->scp_sdca_stat1, rt711->scp_sdca_stat2); + + snd_soc_jack_report(rt711->hs_jack, rt711->jack_type | btn_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (btn_type) { + /* button released */ + snd_soc_jack_report(rt711->hs_jack, rt711->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + mod_delayed_work(system_power_efficient_wq, + &rt711->jack_btn_check_work, msecs_to_jiffies(200)); + } +} + +static void rt711_sdca_btn_check_handler(struct work_struct *work) +{ + struct rt711_sdca_priv *rt711 = + container_of(work, struct rt711_sdca_priv, jack_btn_check_work.work); + int btn_type = 0, ret, idx; + unsigned int det_mode, offset, val; + unsigned char buf[3]; + + ret = regmap_read(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_GE49, RT711_SDCA_CTL_DETECTED_MODE, 0), + &det_mode); + if (ret < 0) + goto io_error; + + /* pin attached */ + if (det_mode) { + /* read UMP message offset */ + ret = regmap_read(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_HID, RT711_SDCA_ENT_HID01, RT711_SDCA_CTL_HIDTX_MESSAGE_OFFSET, 0), + &offset); + if (ret < 0) + goto io_error; + + for (idx = 0; idx < sizeof(buf); idx++) { + ret = regmap_read(rt711->regmap, + RT711_BUF_ADDR_HID1 + offset + idx, &val); + if (ret < 0) + goto io_error; + buf[idx] = val & 0xff; + } + + if (buf[0] == 0x11) { + switch (buf[1] & 0xf0) { + case 0x10: + btn_type |= SND_JACK_BTN_2; + break; + case 0x20: + btn_type |= SND_JACK_BTN_3; + break; + case 0x40: + btn_type |= SND_JACK_BTN_0; + break; + case 0x80: + btn_type |= SND_JACK_BTN_1; + break; + } + switch (buf[2]) { + case 0x01: + case 0x10: + btn_type |= SND_JACK_BTN_2; + break; + case 0x02: + case 0x20: + btn_type |= SND_JACK_BTN_3; + break; + case 0x04: + case 0x40: + btn_type |= SND_JACK_BTN_0; + break; + case 0x08: + case 0x80: + btn_type |= SND_JACK_BTN_1; + break; + } + } + } else + rt711->jack_type = 0; + + dev_dbg(&rt711->slave->dev, "%s, btn_type=0x%x\n", __func__, btn_type); + snd_soc_jack_report(rt711->hs_jack, rt711->jack_type | btn_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (btn_type) { + /* button released */ + snd_soc_jack_report(rt711->hs_jack, rt711->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + mod_delayed_work(system_power_efficient_wq, + &rt711->jack_btn_check_work, msecs_to_jiffies(200)); + } + + return; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); +} + +static void rt711_sdca_jack_init(struct rt711_sdca_priv *rt711) +{ + mutex_lock(&rt711->calibrate_mutex); + + if (rt711->hs_jack) { + /* Enable HID1 event & set button RTC mode */ + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, + RT711_PUSH_BTN_INT_CTL6, 0x80f0, 0x8000); + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, + RT711_PUSH_BTN_INT_CTL2, 0x11dd, 0x11dd); + rt711_sdca_index_write(rt711, RT711_VENDOR_HDA_CTL, + RT711_PUSH_BTN_INT_CTL7, 0xffff); + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, + RT711_PUSH_BTN_INT_CTL9, 0xf000, 0x0000); + + /* GE_mode_change_event_en & Hid1_push_button_event_en */ + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, + RT711_GE_MODE_RELATED_CTL, 0x0c00, 0x0c00); + + switch (rt711->jd_src) { + case RT711_JD1: + /* default settings was already for JD1 */ + break; + case RT711_JD2: + 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_HP_JD_SEL_JD2, + RT711_JD2_2PORT_200K_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_HP_JD_FINAL_RESULT_CTL_JD12); + break; + default: + dev_warn(rt711->component->dev, "Wrong JD source\n"); + break; + } + + /* set SCP_SDCA_IntMask1[0]=1 */ + sdw_write_no_pm(rt711->slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); + /* set SCP_SDCA_IntMask2[0]=1 */ + sdw_write_no_pm(rt711->slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + dev_dbg(&rt711->slave->dev, "in %s enable\n", __func__); + } else { + /* disable HID 1/2 event */ + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, + RT711_GE_MODE_RELATED_CTL, 0x0c00, 0x0000); + + dev_dbg(&rt711->slave->dev, "in %s disable\n", __func__); + } + + mutex_unlock(&rt711->calibrate_mutex); +} + +static int rt711_sdca_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + + rt711->hs_jack = hs_jack; + + if (!rt711->hw_init) { + dev_dbg(&rt711->slave->dev, + "%s hw_init not ready yet\n", __func__); + return 0; + } + + rt711_sdca_jack_init(rt711); + return 0; +} + +/* For SDCA control DAC/ADC Gain */ +static int rt711_sdca_set_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + unsigned int read_l, read_r, gain_l_val, gain_r_val; + unsigned int i, adc_vol_flag = 0; + + if (strstr(ucontrol->id.name, "FU1E Capture Volume") || + strstr(ucontrol->id.name, "FU0F Capture Volume")) + adc_vol_flag = 1; + + /* control value to 2's complement value */ + /* L Channel */ + gain_l_val = ucontrol->value.integer.value[0]; + if (gain_l_val > mc->max) + gain_l_val = mc->max; + read_l = gain_l_val; + + if (mc->shift == 8) /* boost gain */ + gain_l_val = (gain_l_val * 10) << mc->shift; + else { /* ADC/DAC gain */ + if (adc_vol_flag && gain_l_val > mc->shift) + gain_l_val = (gain_l_val - mc->shift) * 75; + else + gain_l_val = (mc->shift - gain_l_val) * 75; + gain_l_val <<= 8; + gain_l_val /= 100; + if (!(adc_vol_flag && read_l > mc->shift)) { + gain_l_val = ~gain_l_val; + gain_l_val += 1; + } + gain_l_val &= 0xffff; + } + + /* R Channel */ + gain_r_val = ucontrol->value.integer.value[1]; + if (gain_r_val > mc->max) + gain_r_val = mc->max; + read_r = gain_r_val; + + if (mc->shift == 8) /* boost gain */ + gain_r_val = (gain_r_val * 10) << mc->shift; + else { /* ADC/DAC gain */ + if (adc_vol_flag && gain_r_val > mc->shift) + gain_r_val = (gain_r_val - mc->shift) * 75; + else + gain_r_val = (mc->shift - gain_r_val) * 75; + gain_r_val <<= 8; + gain_r_val /= 100; + if (!(adc_vol_flag && read_r > mc->shift)) { + gain_r_val = ~gain_r_val; + gain_r_val += 1; + } + gain_r_val &= 0xffff; + } + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + /* Lch*/ + regmap_write(rt711->mbq_regmap, mc->reg, gain_l_val); + + /* Rch */ + regmap_write(rt711->mbq_regmap, mc->rreg, gain_r_val); + + regmap_read(rt711->mbq_regmap, mc->reg, &read_l); + regmap_read(rt711->mbq_regmap, mc->rreg, &read_r); + if (read_r == gain_r_val && read_l == gain_l_val) + break; + } + + return i == 3 ? -EIO : 0; +} + +static int rt711_sdca_set_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int read_l, read_r, ctl_l = 0, ctl_r = 0; + unsigned int adc_vol_flag = 0, neg_flag = 0; + + if (strstr(ucontrol->id.name, "FU1E Capture Volume") || + strstr(ucontrol->id.name, "FU0F Capture Volume")) + adc_vol_flag = 1; + + regmap_read(rt711->mbq_regmap, mc->reg, &read_l); + regmap_read(rt711->mbq_regmap, mc->rreg, &read_r); + + /* 2's complement value to control value */ + if (mc->shift == 8) /* boost gain */ + ctl_l = (read_l >> mc->shift) / 10; + else { /* ADC/DAC gain */ + ctl_l = read_l; + if (read_l & BIT(15)) { + ctl_l = 0xffff & ~(read_l - 1); + neg_flag = 1; + } + ctl_l *= 100; + ctl_l >>= 8; + if (adc_vol_flag) { + if (neg_flag) + ctl_l = mc->shift - (ctl_l / 75); + else + ctl_l = mc->shift + (ctl_l / 75); + } else + ctl_l = mc->max - (ctl_l / 75); + } + + neg_flag = 0; + if (read_l != read_r) { + if (mc->shift == 8) /* boost gain */ + ctl_r = (read_r >> mc->shift) / 10; + else { /* ADC/DAC gain */ + ctl_r = read_r; + if (read_r & BIT(15)) { + ctl_r = 0xffff & ~(read_r - 1); + neg_flag = 1; + } + ctl_r *= 100; + ctl_r >>= 8; + if (adc_vol_flag) { + if (neg_flag) + ctl_r = mc->shift - (ctl_r / 75); + else + ctl_r = mc->shift + (ctl_r / 75); + } else + ctl_r = mc->max - (ctl_r / 75); + } + } else + ctl_r = ctl_l; + + ucontrol->value.integer.value[0] = ctl_l; + ucontrol->value.integer.value[1] = ctl_r; + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static const struct snd_kcontrol_new rt711_sdca_snd_controls[] = { + SOC_DOUBLE_R_EXT_TLV("FU05 Playback Volume", + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, RT711_SDCA_CTL_FU_VOLUME, CH_L), + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, RT711_SDCA_CTL_FU_VOLUME, CH_R), + 0x57, 0x57, 0, + rt711_sdca_set_gain_get, rt711_sdca_set_gain_put, out_vol_tlv), + SOC_DOUBLE_R("FU1E Capture Switch", + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_L), + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_R), + 0, 1, 1), + SOC_DOUBLE_R("FU0F Capture Switch", + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_L), + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_R), + 0, 1, 1), + SOC_DOUBLE_R_EXT_TLV("FU1E Capture Volume", + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_L), + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_R), + 0x17, 0x3f, 0, + rt711_sdca_set_gain_get, rt711_sdca_set_gain_put, in_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("FU0F Capture Volume", + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_VOLUME, CH_L), + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_VOLUME, CH_R), + 0x17, 0x3f, 0, + rt711_sdca_set_gain_get, rt711_sdca_set_gain_put, in_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("FU44 Gain Volume", + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_PLATFORM_FU44, RT711_SDCA_CTL_FU_CH_GAIN, CH_L), + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_PLATFORM_FU44, RT711_SDCA_CTL_FU_CH_GAIN, CH_R), + 8, 3, 0, + rt711_sdca_set_gain_get, rt711_sdca_set_gain_put, mic_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("FU15 Gain Volume", + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_PLATFORM_FU15, RT711_SDCA_CTL_FU_CH_GAIN, CH_L), + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_PLATFORM_FU15, RT711_SDCA_CTL_FU_CH_GAIN, CH_R), + 8, 3, 0, + rt711_sdca_set_gain_get, rt711_sdca_set_gain_put, mic_vol_tlv), +}; + +static int rt711_sdca_mux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + unsigned int val = 0, mask_sft; + + if (strstr(ucontrol->id.name, "ADC 22 Mux")) + mask_sft = 10; + else if (strstr(ucontrol->id.name, "ADC 23 Mux")) + mask_sft = 13; + else + return -EINVAL; + + rt711_sdca_index_read(rt711, RT711_VENDOR_HDA_CTL, + RT711_HDA_LEGACY_MUX_CTL1, &val); + + ucontrol->value.enumerated.item[0] = (val >> mask_sft) & 0x7; + + return 0; +} + +static int rt711_sdca_mux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int val, val2 = 0, change, mask_sft; + + if (item[0] >= e->items) + return -EINVAL; + + if (strstr(ucontrol->id.name, "ADC 22 Mux")) + mask_sft = 10; + else if (strstr(ucontrol->id.name, "ADC 23 Mux")) + mask_sft = 13; + else + return -EINVAL; + + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + + rt711_sdca_index_read(rt711, RT711_VENDOR_HDA_CTL, + RT711_HDA_LEGACY_MUX_CTL1, &val2); + val2 = (val2 >> mask_sft) & 0x7; + + if (val == val2) + change = 0; + else + change = 1; + + if (change) + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, + RT711_HDA_LEGACY_MUX_CTL1, 0x7 << mask_sft, + val << mask_sft); + + snd_soc_dapm_mux_update_power(dapm, kcontrol, + item[0], e, NULL); + + return change; +} + +static const char * const adc_mux_text[] = { + "MIC2", + "LINE1", + "LINE2", + "DMIC", +}; + +static SOC_ENUM_SINGLE_DECL( + rt711_adc22_enum, SND_SOC_NOPM, 0, adc_mux_text); + +static SOC_ENUM_SINGLE_DECL( + rt711_adc23_enum, SND_SOC_NOPM, 0, adc_mux_text); + +static const struct snd_kcontrol_new rt711_sdca_adc22_mux = + SOC_DAPM_ENUM_EXT("ADC 22 Mux", rt711_adc22_enum, + rt711_sdca_mux_get, rt711_sdca_mux_put); + +static const struct snd_kcontrol_new rt711_sdca_adc23_mux = + SOC_DAPM_ENUM_EXT("ADC 23 Mux", rt711_adc23_enum, + rt711_sdca_mux_get, rt711_sdca_mux_put); + +static int rt711_sdca_fu05_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + unsigned char unmute = 0x0, mute = 0x1; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, + RT711_SDCA_CTL_FU_MUTE, CH_L), + unmute); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, + RT711_SDCA_CTL_FU_MUTE, CH_R), + unmute); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, + RT711_SDCA_CTL_FU_MUTE, CH_L), + mute); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, + RT711_SDCA_CTL_FU_MUTE, CH_R), + mute); + break; + } + return 0; +} + +static int rt711_sdca_fu0f_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + unsigned char unmute = 0x0, mute = 0x1; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_L), + unmute); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_R), + unmute); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_L), + mute); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_R), + mute); + break; + } + return 0; +} + +static int rt711_sdca_fu1e_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + unsigned char unmute = 0x0, mute = 0x1; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_L), + unmute); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_R), + unmute); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_L), + mute); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_R), + mute); + break; + } + return 0; +} + +static int rt711_sdca_pde28_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_PDE28, + RT711_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_PDE28, + RT711_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + break; + } + return 0; +} + +static int rt711_sdca_pde29_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_PDE29, + RT711_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_PDE29, + RT711_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + break; + } + return 0; +} + +static int rt711_sdca_pde2a_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_PDE2A, + RT711_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_PDE2A, + RT711_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + break; + } + return 0; +} + +static int rt711_sdca_line1_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + static unsigned int sel_mode = 0xffff; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_read(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_GE49, + RT711_SDCA_CTL_SELECTED_MODE, 0), + &sel_mode); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_LINE1, + RT711_SDCA_CTL_VENDOR_DEF, 0), + 0x1); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_GE49, + RT711_SDCA_CTL_SELECTED_MODE, 0), + 0x7); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_LINE1, + RT711_SDCA_CTL_VENDOR_DEF, 0), + 0x0); + if (sel_mode != 0xffff) + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_GE49, + RT711_SDCA_CTL_SELECTED_MODE, 0), + sel_mode); + break; + } + + return 0; +} + +static int rt711_sdca_line2_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_PDELINE2, + RT711_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_LINE2, + RT711_SDCA_CTL_VENDOR_DEF, 0), + 0x1); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_LINE2, + RT711_SDCA_CTL_VENDOR_DEF, 0), + 0x0); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_PDELINE2, + RT711_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + break; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt711_sdca_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("HP"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("DMIC2"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("LINE2"), + + SND_SOC_DAPM_PGA_E("LINE1 Power", SND_SOC_NOPM, + 0, 0, NULL, 0, rt711_sdca_line1_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("LINE2 Power", SND_SOC_NOPM, + 0, 0, NULL, 0, rt711_sdca_line2_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_SUPPLY("PDE 28", SND_SOC_NOPM, 0, 0, + rt711_sdca_pde28_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("PDE 29", SND_SOC_NOPM, 0, 0, + rt711_sdca_pde29_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("PDE 2A", SND_SOC_NOPM, 0, 0, + rt711_sdca_pde2a_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_DAC_E("FU 05", NULL, SND_SOC_NOPM, 0, 0, + rt711_sdca_fu05_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("FU 0F", NULL, SND_SOC_NOPM, 0, 0, + rt711_sdca_fu0f_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("FU 1E", NULL, SND_SOC_NOPM, 0, 0, + rt711_sdca_fu1e_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MUX("ADC 22 Mux", SND_SOC_NOPM, 0, 0, + &rt711_sdca_adc22_mux), + SND_SOC_DAPM_MUX("ADC 23 Mux", SND_SOC_NOPM, 0, 0, + &rt711_sdca_adc23_mux), + + SND_SOC_DAPM_AIF_IN("DP3RX", "DP3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP2TX", "DP2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP4TX", "DP4 Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route rt711_sdca_audio_map[] = { + {"FU 05", NULL, "DP3RX"}, + {"DP2TX", NULL, "FU 0F"}, + {"DP4TX", NULL, "FU 1E"}, + + {"LINE1 Power", NULL, "LINE1"}, + {"LINE2 Power", NULL, "LINE2"}, + {"HP", NULL, "PDE 28"}, + {"FU 0F", NULL, "PDE 29"}, + {"FU 1E", NULL, "PDE 2A"}, + + {"FU 0F", NULL, "ADC 22 Mux"}, + {"FU 1E", NULL, "ADC 23 Mux"}, + {"ADC 22 Mux", "DMIC", "DMIC1"}, + {"ADC 22 Mux", "LINE1", "LINE1 Power"}, + {"ADC 22 Mux", "LINE2", "LINE2 Power"}, + {"ADC 22 Mux", "MIC2", "MIC2"}, + {"ADC 23 Mux", "DMIC", "DMIC2"}, + {"ADC 23 Mux", "LINE1", "LINE1 Power"}, + {"ADC 23 Mux", "LINE2", "LINE2 Power"}, + {"ADC 23 Mux", "MIC2", "MIC2"}, + + {"HP", NULL, "FU 05"}, +}; + +static int rt711_sdca_parse_dt(struct rt711_sdca_priv *rt711, struct device *dev) +{ + device_property_read_u32(dev, "realtek,jd-src", &rt711->jd_src); + + return 0; +} + +static int rt711_sdca_probe(struct snd_soc_component *component) +{ + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + + rt711_sdca_parse_dt(rt711, &rt711->slave->dev); + rt711->component = component; + + return 0; +} + +static void rt711_sdca_remove(struct snd_soc_component *component) +{ + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt711->regmap, true); + regcache_cache_only(rt711->mbq_regmap, true); +} + +static const struct snd_soc_component_driver soc_sdca_dev_rt711 = { + .probe = rt711_sdca_probe, + .controls = rt711_sdca_snd_controls, + .num_controls = ARRAY_SIZE(rt711_sdca_snd_controls), + .dapm_widgets = rt711_sdca_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt711_sdca_dapm_widgets), + .dapm_routes = rt711_sdca_audio_map, + .num_dapm_routes = ARRAY_SIZE(rt711_sdca_audio_map), + .set_jack = rt711_sdca_set_jack_detect, + .remove = rt711_sdca_remove, +}; + +static int rt711_sdca_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + struct sdw_stream_data *stream; + + if (!sdw_stream) + return 0; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void rt711_sdca_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int rt711_sdca_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 rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_data *stream; + int retval, port, num_channels; + unsigned int sampling_rate; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!rt711->slave) + return -EINVAL; + + /* SoundWire specific configuration */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + port = 3; + } else { + direction = SDW_DATA_DIR_TX; + if (dai->id == RT711_AIF1) + port = 2; + else if (dai->id == RT711_AIF2) + port = 4; + else + return -EINVAL; + } + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = params_channels(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + num_channels = params_channels(params); + port_config.ch_mask = GENMASK(num_channels - 1, 0); + port_config.num = port; + + retval = sdw_stream_add_slave(rt711->slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + if (params_channels(params) > 16) { + dev_err(component->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + /* sampling rate configuration */ + switch (params_rate(params)) { + case 44100: + sampling_rate = RT711_SDCA_RATE_44100HZ; + break; + case 48000: + sampling_rate = RT711_SDCA_RATE_48000HZ; + break; + case 96000: + sampling_rate = RT711_SDCA_RATE_96000HZ; + break; + case 192000: + sampling_rate = RT711_SDCA_RATE_192000HZ; + break; + default: + dev_err(component->dev, "Rate %d is not supported\n", + params_rate(params)); + return -EINVAL; + } + + /* set sampling frequency */ + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_CS01, RT711_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), + sampling_rate); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_CS11, RT711_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), + sampling_rate); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_CS1F, RT711_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), + sampling_rate); + + return 0; +} + +static int rt711_sdca_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt711->slave) + return -EINVAL; + + sdw_stream_remove_slave(rt711->slave, stream->sdw_stream); + return 0; +} + +#define RT711_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) +#define RT711_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops rt711_sdca_ops = { + .hw_params = rt711_sdca_pcm_hw_params, + .hw_free = rt711_sdca_pcm_hw_free, + .set_sdw_stream = rt711_sdca_set_sdw_stream, + .shutdown = rt711_sdca_shutdown, +}; + +static struct snd_soc_dai_driver rt711_sdca_dai[] = { + { + .name = "rt711-sdca-aif1", + .id = RT711_AIF1, + .playback = { + .stream_name = "DP3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT711_STEREO_RATES, + .formats = RT711_FORMATS, + }, + .capture = { + .stream_name = "DP2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT711_STEREO_RATES, + .formats = RT711_FORMATS, + }, + .ops = &rt711_sdca_ops, + }, + { + .name = "rt711-sdca-aif2", + .id = RT711_AIF2, + .capture = { + .stream_name = "DP4 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT711_STEREO_RATES, + .formats = RT711_FORMATS, + }, + .ops = &rt711_sdca_ops, + } +}; + +int rt711_sdca_init(struct device *dev, struct regmap *regmap, + struct regmap *mbq_regmap, struct sdw_slave *slave) +{ + struct rt711_sdca_priv *rt711; + int ret; + + rt711 = devm_kzalloc(dev, sizeof(*rt711), GFP_KERNEL); + if (!rt711) + return -ENOMEM; + + dev_set_drvdata(dev, rt711); + rt711->slave = slave; + rt711->regmap = regmap; + rt711->mbq_regmap = mbq_regmap; + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt711->hw_init = false; + rt711->first_hw_init = false; + + /* JD source uses JD2 in default */ + rt711->jd_src = RT711_JD2; + + ret = devm_snd_soc_register_component(dev, + &soc_sdca_dev_rt711, + rt711_sdca_dai, + ARRAY_SIZE(rt711_sdca_dai)); + + dev_dbg(&slave->dev, "%s\n", __func__); + + return ret; +} + +static void rt711_sdca_vd0_io_init(struct rt711_sdca_priv *rt711) +{ + rt711_sdca_index_write(rt711, RT711_VENDOR_REG, + RT711_GPIO_TEST_MODE_CTL2, 0x0e00); + rt711_sdca_index_write(rt711, RT711_VENDOR_HDA_CTL, + RT711_HDA_LEGACY_GPIO_CTL, 0x0008); + + regmap_write(rt711->regmap, 0x2f5a, 0x01); + + rt711_sdca_index_write(rt711, RT711_VENDOR_REG, + RT711_ADC27_VOL_SET, 0x8728); + + rt711_sdca_index_write(rt711, RT711_VENDOR_REG, + RT711_COMBO_JACK_AUTO_CTL3, 0xa472); + + regmap_write(rt711->regmap, 0x2f50, 0x02); + + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_ANALOG_CTL, + RT711_MISC_POWER_CTL4, 0x6000, 0x6000); + + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_REG, + RT711_COMBO_JACK_AUTO_CTL3, 0x000c, 0x000c); + + rt711_sdca_index_write(rt711, RT711_VENDOR_HDA_CTL, + RT711_HDA_LEGACY_CONFIG_CTL, 0x0000); + + rt711_sdca_index_write(rt711, RT711_VENDOR_VAD, + RT711_VAD_SRAM_CTL1, 0x0050); +} + +static void rt711_sdca_vd1_io_init(struct rt711_sdca_priv *rt711) +{ + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, + RT711_HDA_LEGACY_UNSOLICITED_CTL, 0x0300, 0x0000); + + rt711_sdca_index_write(rt711, RT711_VENDOR_REG, + RT711_COMBO_JACK_AUTO_CTL3, 0xa43e); + + regmap_write(rt711->regmap, 0x2f5a, 0x05); + + rt711_sdca_index_write(rt711, RT711_VENDOR_REG, + RT711_JD_CTRL6, 0x0500); + + rt711_sdca_index_write(rt711, RT711_VENDOR_REG, + RT711_DMIC_CTL1, 0x6173); + + rt711_sdca_index_write(rt711, RT711_VENDOR_HDA_CTL, + RT711_HDA_LEGACY_CONFIG_CTL, 0x0000); + + rt711_sdca_index_write(rt711, RT711_VENDOR_VAD, + RT711_VAD_SRAM_CTL1, 0x0050); +} + +int rt711_sdca_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt711_sdca_priv *rt711 = dev_get_drvdata(dev); + int ret = 0; + unsigned int val; + + if (rt711->hw_init) + return 0; + + if (rt711->first_hw_init) { + regcache_cache_only(rt711->regmap, false); + regcache_cache_bypass(rt711->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + rt711_sdca_reset(rt711); + + rt711_sdca_index_read(rt711, RT711_VENDOR_REG, RT711_JD_PRODUCT_NUM, &val); + rt711->hw_ver = val & 0xf; + + if (rt711->hw_ver == RT711_VER_VD0) + rt711_sdca_vd0_io_init(rt711); + else + rt711_sdca_vd1_io_init(rt711); + + /* DP4 mux select from 08_filter_Out_pri */ + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_REG, + RT711_FILTER_SRC_SEL, 0x1800, 0x0800); + + /* ge_exclusive_inbox_en disable */ + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, + RT711_PUSH_BTN_INT_CTL0, 0x20, 0x00); + + if (!rt711->first_hw_init) { + INIT_DELAYED_WORK(&rt711->jack_detect_work, + rt711_sdca_jack_detect_handler); + INIT_DELAYED_WORK(&rt711->jack_btn_check_work, + rt711_sdca_btn_check_handler); + mutex_init(&rt711->calibrate_mutex); + } + + /* calibration */ + ret = rt711_sdca_calibration(rt711); + if (ret < 0) + dev_err(dev, "%s, calibration failed!\n", __func__); + + /* HP output enable */ + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_OT1, RT711_SDCA_CTL_VENDOR_DEF, 0), 0x4); + + /* + * if set_jack callback occurred early than io_init, + * we set up the jack detection function now + */ + if (rt711->hs_jack) + rt711_sdca_jack_init(rt711); + + if (rt711->first_hw_init) { + regcache_cache_bypass(rt711->regmap, false); + regcache_mark_dirty(rt711->regmap); + } else + rt711->first_hw_init = true; + + /* Mark Slave initialization complete */ + rt711->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + return 0; +} + +MODULE_DESCRIPTION("ASoC RT711 SDCA SDW driver"); +MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt711-sdca.h b/sound/soc/codecs/rt711-sdca.h new file mode 100644 index 000000000000..98a022cec0bd --- /dev/null +++ b/sound/soc/codecs/rt711-sdca.h @@ -0,0 +1,238 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt711-sdca.h -- RT711 SDCA ALSA SoC audio driver header + * + * Copyright(c) 2021 Realtek Semiconductor Corp. + */ + +#ifndef __RT711_SDCA_H__ +#define __RT711_SDCA_H__ + +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/soc.h> +#include <linux/workqueue.h> + +struct rt711_sdca_priv { + struct regmap *regmap, *mbq_regmap; + struct snd_soc_component *component; + struct sdw_slave *slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; + struct snd_soc_jack *hs_jack; + struct delayed_work jack_detect_work; + struct delayed_work jack_btn_check_work; + struct mutex calibrate_mutex; /* for headset calibration */ + int jack_type, jd_src; + unsigned int scp_sdca_stat1, scp_sdca_stat2; + int hw_ver; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +/* NID */ +#define RT711_AUDIO_FUNCTION_GROUP 0x01 +#define RT711_DAC_OUT2 0x03 +#define RT711_ADC_IN1 0x09 +#define RT711_ADC_IN2 0x08 +#define RT711_DMIC1 0x12 +#define RT711_DMIC2 0x13 +#define RT711_MIC2 0x19 +#define RT711_LINE1 0x1a +#define RT711_LINE2 0x1b +#define RT711_BEEP 0x1d +#define RT711_VENDOR_REG 0x20 +#define RT711_HP_OUT 0x21 +#define RT711_MIXER_IN1 0x22 +#define RT711_MIXER_IN2 0x23 +#define RT711_INLINE_CMD 0x55 +#define RT711_VENDOR_CALI 0x58 +#define RT711_VENDOR_IMS_DRE 0x5b +#define RT711_VENDOR_VAD 0x5e +#define RT711_VENDOR_ANALOG_CTL 0x5f +#define RT711_VENDOR_HDA_CTL 0x61 + +/* Index (NID:20h) */ +#define RT711_JD_PRODUCT_NUM 0x00 +#define RT711_DMIC_CTL1 0x06 +#define RT711_JD_CTL1 0x08 +#define RT711_JD_CTL2 0x09 +#define RT711_CC_DET1 0x11 +#define RT711_PARA_VERB_CTL 0x1a +#define RT711_COMBO_JACK_AUTO_CTL1 0x45 +#define RT711_COMBO_JACK_AUTO_CTL2 0x46 +#define RT711_COMBO_JACK_AUTO_CTL3 0x47 +#define RT711_INLINE_CMD_CTL 0x48 +#define RT711_DIGITAL_MISC_CTRL4 0x4a +#define RT711_JD_CTRL6 0x6a +#define RT711_VREFOUT_CTL 0x6b +#define RT711_GPIO_TEST_MODE_CTL2 0x6d +#define RT711_FSM_CTL 0x6f +#define RT711_IRQ_FLAG_TABLE1 0x80 +#define RT711_IRQ_FLAG_TABLE2 0x81 +#define RT711_IRQ_FLAG_TABLE3 0x82 +#define RT711_HP_FSM_CTL 0x83 +#define RT711_TX_RX_MUX_CTL 0x91 +#define RT711_FILTER_SRC_SEL 0xb0 +#define RT711_ADC27_VOL_SET 0xb7 + +/* Index (NID:58h) */ +#define RT711_DAC_DC_CALI_CTL1 0x00 +#define RT711_DAC_DC_CALI_CTL2 0x01 + +/* Index (NID:5bh) */ +#define RT711_IMS_DIGITAL_CTL1 0x00 +#define RT711_HP_IMS_RESULT_L 0x20 +#define RT711_HP_IMS_RESULT_R 0x21 + +/* Index (NID:5eh) */ +#define RT711_VAD_SRAM_CTL1 0x10 + +/* Index (NID:5fh) */ +#define RT711_MISC_POWER_CTL0 0x01 +#define RT711_MISC_POWER_CTL4 0x05 + +/* Index (NID:61h) */ +#define RT711_HDA_LEGACY_MUX_CTL1 0x00 +#define RT711_HDA_LEGACY_UNSOLICITED_CTL 0x03 +#define RT711_HDA_LEGACY_CONFIG_CTL 0x06 +#define RT711_HDA_LEGACY_RESET_CTL 0x08 +#define RT711_HDA_LEGACY_GPIO_CTL 0x0a +#define RT711_ADC08_09_PDE_CTL 0x24 +#define RT711_GE_MODE_RELATED_CTL 0x35 +#define RT711_PUSH_BTN_INT_CTL0 0x36 +#define RT711_PUSH_BTN_INT_CTL1 0x37 +#define RT711_PUSH_BTN_INT_CTL2 0x38 +#define RT711_PUSH_BTN_INT_CTL6 0x3c +#define RT711_PUSH_BTN_INT_CTL7 0x3d +#define RT711_PUSH_BTN_INT_CTL9 0x3f + +/* DAC DC offset calibration control-1 (0x00)(NID:20h) */ +#define RT711_DAC_DC_CALI_TRIGGER (0x1 << 15) +#define RT711_DAC_DC_CALI_CLK_EN (0x1 << 14) +#define RT711_DAC_DC_FORCE_CALI_RST (0x1 << 3) + +/* jack detect control 1 (0x08)(NID:20h) */ +#define RT711_JD2_DIGITAL_MODE_SEL (0x1 << 1) + +/* jack detect control 2 (0x09)(NID:20h) */ +#define RT711_JD2_2PORT_200K_DECODE_HP (0x1 << 13) +#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) + +/* Parameter & Verb control (0x1a)(NID:20h) */ +#define RT711_HIDDEN_REG_SW_RESET (0x1 << 14) + +/* combo jack auto switch control 2 (0x46)(NID:20h) */ +#define RT711_COMBOJACK_AUTO_DET_STATUS (0x1 << 11) +#define RT711_COMBOJACK_AUTO_DET_TRS (0x1 << 10) +#define RT711_COMBOJACK_AUTO_DET_CTIA (0x1 << 9) +#define RT711_COMBOJACK_AUTO_DET_OMTP (0x1 << 8) + +/* FSM control (0x6f)(NID:20h) */ +#define RT711_CALI_CTL (0x0 << 0) +#define RT711_COMBOJACK_CTL (0x1 << 0) +#define RT711_IMS_CTL (0x2 << 0) +#define RT711_DEPOP_CTL (0x3 << 0) +#define RT711_FSM_IMP_EN (0x1 << 6) + +/* Impedance Sense Digital Control 1 (0x00)(NID:5bh) */ +#define RT711_TRIGGER_IMS (0x1 << 15) +#define RT711_IMS_EN (0x1 << 6) + +#define RT711_EAPD_HIGH 0x2 +#define RT711_EAPD_LOW 0x0 +#define RT711_MUTE_SFT 7 +/* set input/output mapping to payload[14][15] separately */ +#define RT711_DIR_IN_SFT 6 +#define RT711_DIR_OUT_SFT 7 + +/* RC Calibration register */ +#define RT711_RC_CAL_STATUS 0x320c + +/* Buffer address for HID */ +#define RT711_BUF_ADDR_HID1 0x44030000 +#define RT711_BUF_ADDR_HID2 0x44030020 + +/* RT711 SDCA Control - function number */ +#define FUNC_NUM_JACK_CODEC 0x01 +#define FUNC_NUM_MIC_ARRAY 0x02 +#define FUNC_NUM_HID 0x03 + +/* RT711 SDCA entity */ +#define RT711_SDCA_ENT_HID01 0x01 +#define RT711_SDCA_ENT_GE49 0x49 +#define RT711_SDCA_ENT_USER_FU05 0x05 +#define RT711_SDCA_ENT_USER_FU0F 0x0f +#define RT711_SDCA_ENT_USER_FU1E 0x1e +#define RT711_SDCA_ENT_PLATFORM_FU15 0x15 +#define RT711_SDCA_ENT_PLATFORM_FU44 0x44 +#define RT711_SDCA_ENT_PDE28 0x28 +#define RT711_SDCA_ENT_PDE29 0x29 +#define RT711_SDCA_ENT_PDE2A 0x2a +#define RT711_SDCA_ENT_CS01 0x01 +#define RT711_SDCA_ENT_CS11 0x11 +#define RT711_SDCA_ENT_CS1F 0x1f +#define RT711_SDCA_ENT_OT1 0x06 +#define RT711_SDCA_ENT_LINE1 0x09 +#define RT711_SDCA_ENT_LINE2 0x31 +#define RT711_SDCA_ENT_PDELINE2 0x36 +#define RT711_SDCA_ENT_USER_FU9 0x41 + +/* RT711 SDCA control */ +#define RT711_SDCA_CTL_SAMPLE_FREQ_INDEX 0x10 +#define RT711_SDCA_CTL_FU_CH_GAIN 0x0b +#define RT711_SDCA_CTL_FU_MUTE 0x01 +#define RT711_SDCA_CTL_FU_VOLUME 0x02 +#define RT711_SDCA_CTL_HIDTX_CURRENT_OWNER 0x10 +#define RT711_SDCA_CTL_HIDTX_SET_OWNER_TO_DEVICE 0x11 +#define RT711_SDCA_CTL_HIDTX_MESSAGE_OFFSET 0x12 +#define RT711_SDCA_CTL_HIDTX_MESSAGE_LENGTH 0x13 +#define RT711_SDCA_CTL_SELECTED_MODE 0x01 +#define RT711_SDCA_CTL_DETECTED_MODE 0x02 +#define RT711_SDCA_CTL_REQ_POWER_STATE 0x01 +#define RT711_SDCA_CTL_VENDOR_DEF 0x30 + +/* RT711 SDCA channel */ +#define CH_L 0x01 +#define CH_R 0x02 + +/* sample frequency index */ +#define RT711_SDCA_RATE_44100HZ 0x08 +#define RT711_SDCA_RATE_48000HZ 0x09 +#define RT711_SDCA_RATE_96000HZ 0x0b +#define RT711_SDCA_RATE_192000HZ 0x0d + +enum { + RT711_AIF1, + RT711_AIF2, + RT711_AIFS, +}; + +enum rt711_sdca_jd_src { + RT711_JD_NULL, + RT711_JD1, + RT711_JD2 +}; + +enum rt711_sdca_ver { + RT711_VER_VD0, + RT711_VER_VD1 +}; + +int rt711_sdca_io_init(struct device *dev, struct sdw_slave *slave); +int rt711_sdca_init(struct device *dev, struct regmap *regmap, + struct regmap *mbq_regmap, struct sdw_slave *slave); + +int rt711_sdca_jack_detect(struct rt711_sdca_priv *rt711, bool *hp, bool *mic); +#endif /* __RT711_SDCA_H__ */ -- 2.29.0

2 1

[PATCH] ASoC: rt715-sdca: Add RT715 sdca vendor-specific driver
by jack.yu＠realtek.com 08 Mar '21

08 Mar '21

From: Jack Yu <jack.yu(a)realtek.com> This is initial sdca version of codec driver for rt715. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt715-sdca-sdw.c | 278 ++++++++ sound/soc/codecs/rt715-sdca-sdw.h | 170 +++++ sound/soc/codecs/rt715-sdca.c | 1076 +++++++++++++++++++++++++++++ sound/soc/codecs/rt715-sdca.h | 136 ++++ 6 files changed, 1669 insertions(+) create mode 100644 sound/soc/codecs/rt715-sdca-sdw.c create mode 100644 sound/soc/codecs/rt715-sdca-sdw.h create mode 100644 sound/soc/codecs/rt715-sdca.c create mode 100644 sound/soc/codecs/rt715-sdca.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ae833a54bcb3..d4d3b4e87207 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -181,6 +181,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT700_SDW imply SND_SOC_RT711_SDW imply SND_SOC_RT715_SDW + imply SND_SOC_RT715_SDCA_SDW imply SND_SOC_RT1308_SDW imply SND_SOC_RT1316_SDW imply SND_SOC_SGTL5000 @@ -1257,6 +1258,12 @@ config SND_SOC_RT715_SDW select SND_SOC_RT715 select REGMAP_SOUNDWIRE +config SND_SOC_RT715_SDCA_SDW + tristate "Realtek RT715 SDCA Codec - SDW" + depends on SOUNDWIRE + select REGMAP_SOUNDWIRE + select REGMAP_SOUNDWIRE_MBQ + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate "Freescale SGTL5000 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1585bc1d9a25..09dd9c168aac 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -198,6 +198,7 @@ snd-soc-rt5682-i2c-objs := rt5682-i2c.o snd-soc-rt700-objs := rt700.o rt700-sdw.o snd-soc-rt711-objs := rt711.o rt711-sdw.o snd-soc-rt715-objs := rt715.o rt715-sdw.o +snd-soc-rt715-sdca-objs := rt715-sdca.o rt715-sdca-sdw.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -514,6 +515,7 @@ obj-$(CONFIG_SND_SOC_RT5682_SDW) += snd-soc-rt5682-sdw.o obj-$(CONFIG_SND_SOC_RT700) += snd-soc-rt700.o obj-$(CONFIG_SND_SOC_RT711) += snd-soc-rt711.o obj-$(CONFIG_SND_SOC_RT715) += snd-soc-rt715.o +obj-$(CONFIG_SND_SOC_RT715_SDCA_SDW) += snd-soc-rt715-sdca.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c new file mode 100644 index 000000000000..bcced85876b0 --- /dev/null +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -0,0 +1,278 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt715-sdca-sdw.c -- rt715 ALSA SoC audio driver +// +// Copyright(c) 2020 Realtek Semiconductor Corp. +// +// + +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/mod_devicetable.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include "rt715-sdca.h" +#include "rt715-sdca-sdw.h" + +static bool rt715_sdca_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x201a ... 0x2027: + case 0x2029 ... 0x202a: + case 0x202d ... 0x2034: + case 0x2200 ... 0x2204: + case 0x2206 ... 0x2212: + case 0x2230 ... 0x2239: + case 0x2f5b: + case SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_SMPU_TRIG_ST_EN, + RT715_SDCA_SMPU_TRIG_ST_CTRL, CH_00): + return true; + default: + return false; + } +} + +static bool rt715_sdca_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x201b: + case 0x201c: + case 0x201d: + case 0x201f: + case 0x2021: + case 0x2023: + case 0x2230: + case 0x202d ... 0x202f: /* BRA */ + case 0x2200 ... 0x2212: /* i2c debug */ + case 0x2f07: + case 0x2f1b ... 0x2f1e: + case 0x2f30 ... 0x2f34: + case 0x2f50 ... 0x2f51: + case 0x2f53 ... 0x2f59: + case 0x2f5c ... 0x2f5f: + case SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_SMPU_TRIG_ST_EN, + RT715_SDCA_SMPU_TRIG_ST_CTRL, CH_00): /* VAD Searching status */ + return true; + default: + return false; + } +} + +static bool rt715_sdca_mbq_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2000000: + case 0x200002b: + case 0x2000036: + case 0x2000037: + case 0x2000039: + case 0x6100000: + return true; + default: + return false; + } +} + +static bool rt715_sdca_mbq_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2000000: + return true; + default: + return false; + } +} + +static const struct regmap_config rt715_sdca_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = rt715_sdca_readable_register, + .volatile_reg = rt715_sdca_volatile_register, + .max_register = 0x43ffffff, + .reg_defaults = rt715_reg_defaults_sdca, + .num_reg_defaults = ARRAY_SIZE(rt715_reg_defaults_sdca), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct regmap_config rt715_sdca_mbq_regmap = { + .name = "sdw-mbq", + .reg_bits = 32, + .val_bits = 16, + .readable_reg = rt715_sdca_mbq_readable_register, + .volatile_reg = rt715_sdca_mbq_volatile_register, + .max_register = 0x43ffffff, + .reg_defaults = rt715_mbq_reg_defaults_sdca, + .num_reg_defaults = ARRAY_SIZE(rt715_mbq_reg_defaults_sdca), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt715_sdca_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt715_sdca_priv *rt715 = dev_get_drvdata(&slave->dev); + + /* Update the status */ + rt715->status = status; + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt715->hw_init || rt715->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt715_sdca_io_init(&slave->dev, slave); +} + +static int rt715_sdca_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval, i; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x50;/* BITMAP: 01010000 */ + prop->sink_ports = 0x0; /* BITMAP: 00000000 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), + GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + dpn = prop->src_dpn_prop; + i = 0; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static struct sdw_slave_ops rt715_sdca_slave_ops = { + .read_prop = rt715_sdca_read_prop, + .update_status = rt715_sdca_update_status, +}; + +static int rt715_sdca_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *mbq_regmap, *regmap; + + slave->ops = &rt715_sdca_slave_ops; + + /* Regmap Initialization */ + mbq_regmap = devm_regmap_init_sdw_mbq(slave, &rt715_sdca_mbq_regmap); + if (!mbq_regmap) + return -EINVAL; + + regmap = devm_regmap_init_sdw(slave, &rt715_sdca_regmap); + if (!regmap) + return -EINVAL; + + return rt715_sdca_init(&slave->dev, mbq_regmap, regmap, slave); +} + +static const struct sdw_device_id rt715_sdca_id[] = { + SDW_SLAVE_ENTRY_EXT(0x025d, 0x715, 0x3, 0x1, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x714, 0x3, 0x1, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt715_sdca_id); + +static int __maybe_unused rt715_dev_suspend(struct device *dev) +{ + struct rt715_sdca_priv *rt715 = dev_get_drvdata(dev); + + if (!rt715->hw_init) + return 0; + + regcache_cache_only(rt715->regmap, true); + regcache_mark_dirty(rt715->regmap); + regcache_cache_only(rt715->mbq_regmap, true); + regcache_mark_dirty(rt715->mbq_regmap); + + return 0; +} + +#define RT715_PROBE_TIMEOUT 5000 + +static int __maybe_unused rt715_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt715_sdca_priv *rt715 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt715->hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->enumeration_complete, + msecs_to_jiffies(RT715_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Enumeration not complete, timed out\n"); + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt715->regmap, false); + regcache_sync_region(rt715->regmap, + SDW_SDCA_CTL(FUN_JACK_CODEC, RT715_SDCA_ST_EN, RT715_SDCA_ST_CTRL, + CH_00), + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_SMPU_TRIG_ST_EN, + RT715_SDCA_SMPU_TRIG_ST_CTRL, CH_00)); + regcache_cache_only(rt715->mbq_regmap, false); + regcache_sync_region(rt715->mbq_regmap, 0x2000000, 0x61020ff); + regcache_sync_region(rt715->mbq_regmap, + SDW_SDCA_CTL(FUN_JACK_CODEC, RT715_SDCA_ST_EN, RT715_SDCA_ST_CTRL, + CH_00), + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_SMPU_TRIG_ST_EN, + RT715_SDCA_SMPU_TRIG_ST_CTRL, CH_00)); + + return 0; +} + +static const struct dev_pm_ops rt715_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt715_dev_suspend, rt715_dev_resume) + SET_RUNTIME_PM_OPS(rt715_dev_suspend, rt715_dev_resume, NULL) +}; + +static struct sdw_driver rt715_sdw_driver = { + .driver = { + .name = "rt715-sdca", + .owner = THIS_MODULE, + .pm = &rt715_pm, + }, + .probe = rt715_sdca_sdw_probe, + .ops = &rt715_sdca_slave_ops, + .id_table = rt715_sdca_id, +}; +module_sdw_driver(rt715_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT715 driver SDW SDCA"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt715-sdca-sdw.h b/sound/soc/codecs/rt715-sdca-sdw.h new file mode 100644 index 000000000000..cd365bb60747 --- /dev/null +++ b/sound/soc/codecs/rt715-sdca-sdw.h @@ -0,0 +1,170 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt715-sdca-sdw.h -- RT715 ALSA SoC audio driver header + * + * Copyright(c) 2020 Realtek Semiconductor Corp. + */ + +#ifndef __RT715_SDW_SDCA_H__ +#define __RT715_SDW_SDCA_H__ + +#include <linux/soundwire/sdw_registers.h> + +static const struct reg_default rt715_reg_defaults_sdca[] = { + { 0x201a, 0x00 }, + { 0x201e, 0x00 }, + { 0x2020, 0x00 }, + { 0x2021, 0x00 }, + { 0x2022, 0x00 }, + { 0x2023, 0x00 }, + { 0x2024, 0x00 }, + { 0x2025, 0x01 }, + { 0x2026, 0x00 }, + { 0x2027, 0x00 }, + { 0x2029, 0x00 }, + { 0x202a, 0x00 }, + { 0x202d, 0x00 }, + { 0x202e, 0x00 }, + { 0x202f, 0x00 }, + { 0x2030, 0x00 }, + { 0x2031, 0x00 }, + { 0x2032, 0x00 }, + { 0x2033, 0x00 }, + { 0x2034, 0x00 }, + { 0x2230, 0x00 }, + { 0x2231, 0x2f }, + { 0x2232, 0x80 }, + { 0x2233, 0x00 }, + { 0x2234, 0x00 }, + { 0x2235, 0x00 }, + { 0x2236, 0x00 }, + { 0x2237, 0x00 }, + { 0x2238, 0x00 }, + { 0x2239, 0x00 }, + { 0x2f01, 0x00 }, + { 0x2f02, 0x09 }, + { 0x2f03, 0x0b }, + { 0x2f04, 0x00 }, + { 0x2f05, 0x0e }, + { 0x2f06, 0x01 }, + { 0x2f08, 0x00 }, + { 0x2f09, 0x00 }, + { 0x2f0a, 0x00 }, + { 0x2f0b, 0x00 }, + { 0x2f0c, 0x00 }, + { 0x2f0d, 0x00 }, + { 0x2f0e, 0x12 }, + { 0x2f0f, 0x00 }, + { 0x2f10, 0x00 }, + { 0x2f11, 0x00 }, + { 0x2f12, 0x00 }, + { 0x2f13, 0x00 }, + { 0x2f14, 0x00 }, + { 0x2f15, 0x00 }, + { 0x2f16, 0x00 }, + { 0x2f17, 0x00 }, + { 0x2f18, 0x00 }, + { 0x2f19, 0x03 }, + { 0x2f1a, 0x00 }, + { 0x2f1f, 0x10 }, + { 0x2f20, 0x00 }, + { 0x2f21, 0x00 }, + { 0x2f22, 0x00 }, + { 0x2f23, 0x00 }, + { 0x2f24, 0x00 }, + { 0x2f25, 0x00 }, + { 0x2f52, 0x01 }, + { 0x2f5a, 0x02 }, + { 0x2f5b, 0x05 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_CX_CLK_SEL_EN, + RT715_SDCA_CX_CLK_SEL_CTRL, CH_00), 0x1 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC8_9_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_01), 0x01 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC8_9_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_02), 0x01 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC8_9_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_03), 0x01 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC8_9_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_04), 0x01 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC10_11_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_01), 0x01 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC10_11_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_02), 0x01 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC10_11_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_03), 0x01 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC10_11_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_04), 0x01 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC7_27_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_01), 0x01 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC7_27_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_02), 0x01 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_SMPU_TRIG_ST_EN, + RT715_SDCA_SMPU_TRIG_EN_CTRL, CH_00), 0x02 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_SMPU_TRIG_ST_EN, + RT715_SDCA_SMPU_TRIG_ST_CTRL, CH_00), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC7_27_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_01), 0x01 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC7_27_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_02), 0x01 }, +}; + +static const struct reg_default rt715_mbq_reg_defaults_sdca[] = { + { 0x200002b, 0x0420 }, + { 0x2000036, 0x0000 }, + { 0x2000037, 0x0000 }, + { 0x2000039, 0xaa81 }, + { 0x6100000, 0x0100 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC8_9_VOL, + RT715_SDCA_FU_VOL_CTRL, CH_01), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC8_9_VOL, + RT715_SDCA_FU_VOL_CTRL, CH_02), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC8_9_VOL, + RT715_SDCA_FU_VOL_CTRL, CH_03), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC8_9_VOL, + RT715_SDCA_FU_VOL_CTRL, CH_04), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC10_11_VOL, + RT715_SDCA_FU_VOL_CTRL, CH_01), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC10_11_VOL, + RT715_SDCA_FU_VOL_CTRL, CH_02), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC10_11_VOL, + RT715_SDCA_FU_VOL_CTRL, CH_03), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC10_11_VOL, + RT715_SDCA_FU_VOL_CTRL, CH_04), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC7_27_VOL, + RT715_SDCA_FU_VOL_CTRL, CH_01), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC7_27_VOL, + RT715_SDCA_FU_VOL_CTRL, CH_02), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_AMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_01), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_AMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_02), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_AMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_03), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_AMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_04), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_AMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_05), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_AMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_06), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_AMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_07), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_AMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_08), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_DMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_01), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_DMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_02), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_DMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_03), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_DMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_04), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_DMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_05), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_DMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_06), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_DMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_07), 0x00 }, + { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_DMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_08), 0x00 }, +}; +#endif /* __RT715_SDW_SDCA_H__ */ diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c new file mode 100644 index 000000000000..92ad6fa408ec --- /dev/null +++ b/sound/soc/codecs/rt715-sdca.c @@ -0,0 +1,1076 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt715-sdca.c -- rt715 ALSA SoC audio driver +// +// Copyright(c) 2020 Realtek Semiconductor Corp. +// +// +// + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/pm_runtime.h> +#include <linux/pm.h> +#include <linux/soundwire/sdw.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/soundwire/sdw_registers.h> + +#include "rt715-sdca.h" + +static int rt715_sdca_index_write(struct rt715_sdca_priv *rt715, + unsigned int nid, unsigned int reg, unsigned int value) +{ + struct regmap *regmap = rt715->mbq_regmap; + unsigned int addr; + int ret; + + addr = (nid << 20) | reg; + + ret = regmap_write(regmap, addr, value); + if (ret < 0) + dev_err(&rt715->slave->dev, + "Failed to set private value: %08x <= %04x %d\n", ret, addr, + value); + + return ret; +} + +static int rt715_sdca_index_read(struct rt715_sdca_priv *rt715, + unsigned int nid, unsigned int reg, unsigned int *value) +{ + struct regmap *regmap = rt715->mbq_regmap; + unsigned int addr; + int ret; + + addr = (nid << 20) | reg; + + ret = regmap_read(regmap, addr, value); + if (ret < 0) + dev_err(&rt715->slave->dev, + "Failed to get private value: %06x => %04x ret=%d\n", + addr, *value, ret); + + return ret; +} + +static int rt715_sdca_index_update_bits(struct rt715_sdca_priv *rt715, + unsigned int nid, unsigned int reg, unsigned int mask, unsigned int val) +{ + unsigned int tmp; + int ret; + + ret = rt715_sdca_index_read(rt715, nid, reg, &tmp); + if (ret < 0) + return ret; + + set_mask_bits(&tmp, mask, val); + + return rt715_sdca_index_write(rt715, nid, reg, tmp); +} + +static inline unsigned int rt715_sdca_vol_gain(unsigned int u_ctrl_val, + unsigned int vol_max, unsigned int vol_gain_sft) +{ + unsigned int val; + + if (u_ctrl_val > vol_max) + u_ctrl_val = vol_max; + val = u_ctrl_val; + u_ctrl_val = + ((abs(u_ctrl_val - vol_gain_sft) * RT715_SDCA_DB_STEP) << 8) / 1000; + if (val <= vol_gain_sft) { + u_ctrl_val = ~u_ctrl_val; + u_ctrl_val += 1; + } + u_ctrl_val &= 0xffff; + + return u_ctrl_val; +} + +static inline unsigned int rt715_sdca_boost_gain(unsigned int u_ctrl_val, + unsigned int b_max, unsigned int b_gain_sft) +{ + if (u_ctrl_val > b_max) + u_ctrl_val = b_max; + + return (u_ctrl_val * 10) << b_gain_sft; +} + +static inline unsigned int rt715_sdca_get_gain(unsigned int reg_val, + unsigned int gain_sft) +{ + unsigned int neg_flag = 0; + + if (reg_val & BIT(15)) { + reg_val = ~(reg_val - 1) & 0xffff; + neg_flag = 1; + } + reg_val *= 1000; + reg_val >>= 8; + if (neg_flag) + reg_val = gain_sft - reg_val / RT715_SDCA_DB_STEP; + else + reg_val = gain_sft + reg_val / RT715_SDCA_DB_STEP; + + return reg_val; +} + +/* SDCA Volume/Boost control */ +static int rt715_sdca_set_amp_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct rt715_sdca_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int gain_val, i, k_changed = 0; + int ret; + + for (i = 0; i < 2; i++) { + if (ucontrol->value.integer.value[i] != rt715->kctl_2ch_orig[i]) { + k_changed = 1; + break; + } + } + + for (i = 0; i < 2; i++) { + rt715->kctl_2ch_orig[i] = ucontrol->value.integer.value[i]; + gain_val = + rt715_sdca_vol_gain(ucontrol->value.integer.value[i], mc->max, + mc->shift); + ret = regmap_write(rt715->mbq_regmap, mc->reg + i, gain_val); + if (ret != 0) { + dev_err(component->dev, "Failed to write 0x%x=0x%x\n", + mc->reg + i, gain_val); + return ret; + } + } + + return k_changed; +} + +static int rt715_sdca_set_amp_gain_4ch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_sdca_priv *rt715 = snd_soc_component_get_drvdata(component); + struct rt715_sdca_kcontrol_private *p = + (struct rt715_sdca_kcontrol_private *)kcontrol->private_value; + unsigned int reg_base = p->reg_base, k_changed = 0; + const unsigned int gain_sft = 0x2f; + unsigned int gain_val, i; + int ret; + + for (i = 0; i < 4; i++) { + if (ucontrol->value.integer.value[i] != rt715->kctl_4ch_orig[i]) { + k_changed = 1; + break; + } + } + + for (i = 0; i < 4; i++) { + rt715->kctl_4ch_orig[i] = ucontrol->value.integer.value[i]; + gain_val = + rt715_sdca_vol_gain(ucontrol->value.integer.value[i], p->max, + gain_sft); + ret = regmap_write(rt715->mbq_regmap, reg_base + i, + gain_val); + if (ret != 0) { + dev_err(component->dev, "Failed to write 0x%x=0x%x\n", + reg_base + i, gain_val); + return ret; + } + } + + return k_changed; +} + +static int rt715_sdca_set_amp_gain_8ch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_sdca_priv *rt715 = snd_soc_component_get_drvdata(component); + struct rt715_sdca_kcontrol_private *p = + (struct rt715_sdca_kcontrol_private *)kcontrol->private_value; + unsigned int reg_base = p->reg_base, i, k_changed = 0; + const unsigned int gain_sft = 8; + unsigned int gain_val, reg; + int ret; + + for (i = 0; i < 8; i++) { + if (ucontrol->value.integer.value[i] != rt715->kctl_8ch_orig[i]) { + k_changed = 1; + break; + } + } + + for (i = 0; i < 8; i++) { + rt715->kctl_8ch_orig[i] = ucontrol->value.integer.value[i]; + gain_val = + rt715_sdca_boost_gain(ucontrol->value.integer.value[i], p->max, + gain_sft); + reg = i < 7 ? reg_base + i : (reg_base - 1) | BIT(15); + ret = regmap_write(rt715->mbq_regmap, reg, gain_val); + if (ret != 0) { + dev_err(component->dev, "Failed to write 0x%x=0x%x\n", + reg, gain_val); + return ret; + } + } + + return k_changed; +} + +static int rt715_sdca_set_amp_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct rt715_sdca_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int val, i; + int ret; + + for (i = 0; i < 2; i++) { + ret = regmap_read(rt715->mbq_regmap, mc->reg + i, &val); + if (ret < 0) { + dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", + mc->reg + i, ret); + return ret; + } + ucontrol->value.integer.value[i] = rt715_sdca_get_gain(val, mc->shift); + } + + return 0; +} + +static int rt715_sdca_set_amp_gain_4ch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_sdca_priv *rt715 = snd_soc_component_get_drvdata(component); + struct rt715_sdca_kcontrol_private *p = + (struct rt715_sdca_kcontrol_private *)kcontrol->private_value; + unsigned int reg_base = p->reg_base, i; + const unsigned int gain_sft = 0x2f; + unsigned int val; + int ret; + + for (i = 0; i < 4; i++) { + ret = regmap_read(rt715->mbq_regmap, reg_base + i, &val); + if (ret < 0) { + dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", + reg_base + i, ret); + return ret; + } + ucontrol->value.integer.value[i] = rt715_sdca_get_gain(val, gain_sft); + } + + return 0; +} + +static int rt715_sdca_set_amp_gain_8ch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_sdca_priv *rt715 = snd_soc_component_get_drvdata(component); + struct rt715_sdca_kcontrol_private *p = + (struct rt715_sdca_kcontrol_private *)kcontrol->private_value; + unsigned int reg_base = p->reg_base; + const unsigned int gain_sft = 8; + unsigned int val_l, val_r; + unsigned int i, reg; + int ret; + + for (i = 0; i < 8; i += 2) { + ret = regmap_read(rt715->mbq_regmap, reg_base + i, &val_l); + if (ret < 0) { + dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", + reg_base + i, ret); + return ret; + } + ucontrol->value.integer.value[i] = (val_l >> gain_sft) / 10; + + reg = (i == 6) ? (reg_base - 1) | BIT(15) : reg_base + 1 + i; + ret = regmap_read(rt715->mbq_regmap, reg, &val_r); + if (ret < 0) { + dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", + reg, ret); + return ret; + } + ucontrol->value.integer.value[i + 1] = (val_r >> gain_sft) / 10; + } + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static int rt715_sdca_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_sdca_kcontrol_private *p = + (struct rt715_sdca_kcontrol_private *)kcontrol->private_value; + unsigned int reg_base = p->reg_base; + unsigned int invert = p->invert, i; + int val; + + for (i = 0; i < p->count; i += 2) { + val = snd_soc_component_read(component, reg_base + i); + if (val < 0) + return -EINVAL; + ucontrol->value.integer.value[i] = invert ? p->max - val : val; + + val = snd_soc_component_read(component, reg_base + 1 + i); + if (val < 0) + return -EINVAL; + ucontrol->value.integer.value[i + 1] = + invert ? p->max - val : val; + } + + return 0; +} + +static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_sdca_priv *rt715 = snd_soc_component_get_drvdata(component); + struct rt715_sdca_kcontrol_private *p = + (struct rt715_sdca_kcontrol_private *)kcontrol->private_value; + unsigned int val[4] = {0}, val_mask, i, k_changed = 0; + unsigned int reg = p->reg_base; + unsigned int shift = p->shift; + unsigned int max = p->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = p->invert; + int err; + + for (i = 0; i < 4; i++) { + if (ucontrol->value.integer.value[i] != rt715->kctl_switch_orig[i]) { + k_changed = 1; + break; + } + } + + for (i = 0; i < 2; i++) { + rt715->kctl_switch_orig[i * 2] = ucontrol->value.integer.value[i * 2]; + val[i * 2] = ucontrol->value.integer.value[i * 2] & mask; + if (invert) + val[i * 2] = max - val[i * 2]; + val_mask = mask << shift; + val[i * 2] <<= shift; + + rt715->kctl_switch_orig[i * 2 + 1] = + ucontrol->value.integer.value[i * 2 + 1]; + val[i * 2 + 1] = + ucontrol->value.integer.value[i * 2 + 1] & mask; + if (invert) + val[i * 2 + 1] = max - val[i * 2 + 1]; + + val[i * 2 + 1] <<= shift; + + err = snd_soc_component_update_bits(component, reg + i * 2, val_mask, + val[i * 2]); + if (err < 0) + return err; + + err = snd_soc_component_update_bits(component, reg + 1 + i * 2, + val_mask, val[i * 2 + 1]); + if (err < 0) + return err; + } + + return k_changed; +} + +static int rt715_sdca_fu_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct rt715_sdca_kcontrol_private *p = + (struct rt715_sdca_kcontrol_private *)kcontrol->private_value; + + if (p->max == 1) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = p->count; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = p->max; + return 0; +} + +#define RT715_SDCA_PR_VALUE(xreg_base, xcount, xmax, xshift, xinvert) \ + ((unsigned long)&(struct rt715_sdca_kcontrol_private) \ + {.reg_base = xreg_base, .count = xcount, .max = xmax, \ + .shift = xshift, .invert = xinvert}) + +#define RT715_SDCA_FU_CTRL(xname, reg_base, xshift, xmax, xinvert, xcount) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = rt715_sdca_fu_info, \ + .get = rt715_sdca_get_volsw, \ + .put = rt715_sdca_put_volsw, \ + .private_value = RT715_SDCA_PR_VALUE(reg_base, xcount, xmax, \ + xshift, xinvert)} + +#define SOC_DOUBLE_R_EXT(xname, reg_left, reg_right, xshift, xmax, xinvert,\ + xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ + xmax, xinvert) } + +#define RT715_SDCA_EXT_TLV(xname, reg_base, xhandler_get,\ + xhandler_put, tlv_array, xcount, xmax) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = rt715_sdca_fu_info, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = RT715_SDCA_PR_VALUE(reg_base, xcount, xmax, 0, 0) } + +#define RT715_SDCA_BOOST_EXT_TLV(xname, reg_base, xhandler_get,\ + xhandler_put, tlv_array, xcount, xmax) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = rt715_sdca_fu_info, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = RT715_SDCA_PR_VALUE(reg_base, xcount, xmax, 0, 0) } + +static const struct snd_kcontrol_new rt715_sdca_snd_controls[] = { + /* Capture switch */ + SOC_DOUBLE_R("FU0A Capture Switch", + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC7_27_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_01), + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC7_27_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_02), + 0, 1, 1), + RT715_SDCA_FU_CTRL("FU02 Capture Switch", + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC8_9_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_01), + 0, 1, 1, 4), + RT715_SDCA_FU_CTRL("FU06 Capture Switch", + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC10_11_VOL, + RT715_SDCA_FU_MUTE_CTRL, CH_01), + 0, 1, 1, 4), + /* Volume Control */ + SOC_DOUBLE_R_EXT_TLV("FU0A Capture Volume", + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC7_27_VOL, + RT715_SDCA_FU_VOL_CTRL, CH_01), + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC7_27_VOL, + RT715_SDCA_FU_VOL_CTRL, CH_02), + 0x2f, 0x7f, 0, + rt715_sdca_set_amp_gain_get, rt715_sdca_set_amp_gain_put, + in_vol_tlv), + RT715_SDCA_EXT_TLV("FU02 Capture Volume", + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC8_9_VOL, + RT715_SDCA_FU_VOL_CTRL, CH_01), + rt715_sdca_set_amp_gain_4ch_get, + rt715_sdca_set_amp_gain_4ch_put, + in_vol_tlv, 4, 0x7f), + RT715_SDCA_EXT_TLV("FU06 Capture Volume", + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC10_11_VOL, + RT715_SDCA_FU_VOL_CTRL, CH_01), + rt715_sdca_set_amp_gain_4ch_get, + rt715_sdca_set_amp_gain_4ch_put, + in_vol_tlv, 4, 0x7f), + /* MIC Boost Control */ + RT715_SDCA_BOOST_EXT_TLV("FU0E Boost", + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_DMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_01), + rt715_sdca_set_amp_gain_8ch_get, + rt715_sdca_set_amp_gain_8ch_put, + mic_vol_tlv, 8, 3), + RT715_SDCA_BOOST_EXT_TLV("FU0C Boost", + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_AMIC_GAIN_EN, + RT715_SDCA_FU_DMIC_GAIN_CTRL, CH_01), + rt715_sdca_set_amp_gain_8ch_get, + rt715_sdca_set_amp_gain_8ch_put, + mic_vol_tlv, 8, 3), +}; + +static int rt715_sdca_mux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct rt715_sdca_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int val, mask_sft; + + if (strstr(ucontrol->id.name, "ADC 22 Mux")) + mask_sft = 12; + else if (strstr(ucontrol->id.name, "ADC 23 Mux")) + mask_sft = 8; + else if (strstr(ucontrol->id.name, "ADC 24 Mux")) + mask_sft = 4; + else if (strstr(ucontrol->id.name, "ADC 25 Mux")) + mask_sft = 0; + else + return -EINVAL; + + rt715_sdca_index_read(rt715, RT715_VENDOR_HDA_CTL, + RT715_HDA_LEGACY_MUX_CTL1, &val); + val = (val >> mask_sft) & 0xf; + + /* + * The first two indices of ADC Mux 24/25 are routed to the same + * hardware source. ie, ADC Mux 24 0/1 will both connect to MIC2. + * To have a unique set of inputs, we skip the index1 of the muxes. + */ + if ((strstr(ucontrol->id.name, "ADC 24 Mux") || + strstr(ucontrol->id.name, "ADC 25 Mux")) && val > 0) + val -= 1; + ucontrol->value.enumerated.item[0] = val; + + return 0; +} + +static int rt715_sdca_mux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct rt715_sdca_priv *rt715 = snd_soc_component_get_drvdata(component); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int val, val2 = 0, change, mask_sft; + + if (item[0] >= e->items) + return -EINVAL; + + if (strstr(ucontrol->id.name, "ADC 22 Mux")) + mask_sft = 12; + else if (strstr(ucontrol->id.name, "ADC 23 Mux")) + mask_sft = 8; + else if (strstr(ucontrol->id.name, "ADC 24 Mux")) + mask_sft = 4; + else if (strstr(ucontrol->id.name, "ADC 25 Mux")) + mask_sft = 0; + else + return -EINVAL; + + /* Verb ID = 0x701h, nid = e->reg */ + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + + rt715_sdca_index_read(rt715, RT715_VENDOR_HDA_CTL, + RT715_HDA_LEGACY_MUX_CTL1, &val2); + val2 = (val2 >> mask_sft) & 0xf; + + change = val != val2; + + if (change) + rt715_sdca_index_update_bits(rt715, RT715_VENDOR_HDA_CTL, + RT715_HDA_LEGACY_MUX_CTL1, 0xf << mask_sft, val << mask_sft); + + snd_soc_dapm_mux_update_power(dapm, kcontrol, item[0], e, NULL); + + return change; +} + +static const char * const adc_22_23_mux_text[] = { + "MIC1", + "MIC2", + "LINE1", + "LINE2", + "DMIC1", + "DMIC2", + "DMIC3", + "DMIC4", +}; + +/* + * Due to mux design for nid 24 (MUX_IN3)/25 (MUX_IN4), connection index 0 and + * 1 will be connected to the same dmic source, therefore we skip index 1 to + * avoid misunderstanding on usage of dapm routing. + */ +static int rt715_adc_24_25_values[] = { + 0, + 2, + 3, + 4, + 5, +}; + +static const char * const adc_24_mux_text[] = { + "MIC2", + "DMIC1", + "DMIC2", + "DMIC3", + "DMIC4", +}; + +static const char * const adc_25_mux_text[] = { + "MIC1", + "DMIC1", + "DMIC2", + "DMIC3", + "DMIC4", +}; + +static SOC_ENUM_SINGLE_DECL(rt715_adc22_enum, SND_SOC_NOPM, 0, + adc_22_23_mux_text); + +static SOC_ENUM_SINGLE_DECL(rt715_adc23_enum, SND_SOC_NOPM, 0, + adc_22_23_mux_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(rt715_adc24_enum, + SND_SOC_NOPM, 0, 0xf, + adc_24_mux_text, rt715_adc_24_25_values); +static SOC_VALUE_ENUM_SINGLE_DECL(rt715_adc25_enum, + SND_SOC_NOPM, 0, 0xf, + adc_25_mux_text, rt715_adc_24_25_values); + +static const struct snd_kcontrol_new rt715_adc22_mux = + SOC_DAPM_ENUM_EXT("ADC 22 Mux", rt715_adc22_enum, + rt715_sdca_mux_get, rt715_sdca_mux_put); + +static const struct snd_kcontrol_new rt715_adc23_mux = + SOC_DAPM_ENUM_EXT("ADC 23 Mux", rt715_adc23_enum, + rt715_sdca_mux_get, rt715_sdca_mux_put); + +static const struct snd_kcontrol_new rt715_adc24_mux = + SOC_DAPM_ENUM_EXT("ADC 24 Mux", rt715_adc24_enum, + rt715_sdca_mux_get, rt715_sdca_mux_put); + +static const struct snd_kcontrol_new rt715_adc25_mux = + SOC_DAPM_ENUM_EXT("ADC 25 Mux", rt715_adc25_enum, + rt715_sdca_mux_get, rt715_sdca_mux_put); + +static int rt715_sdca_pde23_24_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt715_sdca_priv *rt715 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt715->regmap, + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_CREQ_POW_EN, + RT715_SDCA_REQ_POW_CTRL, + CH_00), 0x00); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt715->regmap, + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_CREQ_POW_EN, + RT715_SDCA_REQ_POW_CTRL, + CH_00), 0x03); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget rt715_sdca_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("DMIC2"), + SND_SOC_DAPM_INPUT("DMIC3"), + SND_SOC_DAPM_INPUT("DMIC4"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("LINE2"), + + SND_SOC_DAPM_SUPPLY("PDE23_24", SND_SOC_NOPM, 0, 0, + rt715_sdca_pde23_24_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_ADC("ADC 07", NULL, SND_SOC_NOPM, 4, 0), + SND_SOC_DAPM_ADC("ADC 08", NULL, SND_SOC_NOPM, 4, 0), + SND_SOC_DAPM_ADC("ADC 09", NULL, SND_SOC_NOPM, 4, 0), + SND_SOC_DAPM_ADC("ADC 27", NULL, SND_SOC_NOPM, 4, 0), + SND_SOC_DAPM_MUX("ADC 22 Mux", SND_SOC_NOPM, 0, 0, + &rt715_adc22_mux), + SND_SOC_DAPM_MUX("ADC 23 Mux", SND_SOC_NOPM, 0, 0, + &rt715_adc23_mux), + SND_SOC_DAPM_MUX("ADC 24 Mux", SND_SOC_NOPM, 0, 0, + &rt715_adc24_mux), + SND_SOC_DAPM_MUX("ADC 25 Mux", SND_SOC_NOPM, 0, 0, + &rt715_adc25_mux), + SND_SOC_DAPM_AIF_OUT("DP4TX", "DP4 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP6TX", "DP6 Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route rt715_sdca_audio_map[] = { + {"DP6TX", NULL, "ADC 09"}, + {"DP6TX", NULL, "ADC 08"}, + {"DP4TX", NULL, "ADC 07"}, + {"DP4TX", NULL, "ADC 27"}, + {"DP4TX", NULL, "ADC 09"}, + {"DP4TX", NULL, "ADC 08"}, + + {"LINE1", NULL, "PDE23_24"}, + {"LINE2", NULL, "PDE23_24"}, + {"MIC1", NULL, "PDE23_24"}, + {"MIC2", NULL, "PDE23_24"}, + {"DMIC1", NULL, "PDE23_24"}, + {"DMIC2", NULL, "PDE23_24"}, + {"DMIC3", NULL, "PDE23_24"}, + {"DMIC4", NULL, "PDE23_24"}, + + {"ADC 09", NULL, "ADC 22 Mux"}, + {"ADC 08", NULL, "ADC 23 Mux"}, + {"ADC 07", NULL, "ADC 24 Mux"}, + {"ADC 27", NULL, "ADC 25 Mux"}, + {"ADC 22 Mux", "MIC1", "MIC1"}, + {"ADC 22 Mux", "MIC2", "MIC2"}, + {"ADC 22 Mux", "LINE1", "LINE1"}, + {"ADC 22 Mux", "LINE2", "LINE2"}, + {"ADC 22 Mux", "DMIC1", "DMIC1"}, + {"ADC 22 Mux", "DMIC2", "DMIC2"}, + {"ADC 22 Mux", "DMIC3", "DMIC3"}, + {"ADC 22 Mux", "DMIC4", "DMIC4"}, + {"ADC 23 Mux", "MIC1", "MIC1"}, + {"ADC 23 Mux", "MIC2", "MIC2"}, + {"ADC 23 Mux", "LINE1", "LINE1"}, + {"ADC 23 Mux", "LINE2", "LINE2"}, + {"ADC 23 Mux", "DMIC1", "DMIC1"}, + {"ADC 23 Mux", "DMIC2", "DMIC2"}, + {"ADC 23 Mux", "DMIC3", "DMIC3"}, + {"ADC 23 Mux", "DMIC4", "DMIC4"}, + {"ADC 24 Mux", "MIC2", "MIC2"}, + {"ADC 24 Mux", "DMIC1", "DMIC1"}, + {"ADC 24 Mux", "DMIC2", "DMIC2"}, + {"ADC 24 Mux", "DMIC3", "DMIC3"}, + {"ADC 24 Mux", "DMIC4", "DMIC4"}, + {"ADC 25 Mux", "MIC1", "MIC1"}, + {"ADC 25 Mux", "DMIC1", "DMIC1"}, + {"ADC 25 Mux", "DMIC2", "DMIC2"}, + {"ADC 25 Mux", "DMIC3", "DMIC3"}, + {"ADC 25 Mux", "DMIC4", "DMIC4"}, +}; + +static const struct snd_soc_component_driver soc_codec_dev_rt715_sdca = { + .controls = rt715_sdca_snd_controls, + .num_controls = ARRAY_SIZE(rt715_sdca_snd_controls), + .dapm_widgets = rt715_sdca_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt715_sdca_dapm_widgets), + .dapm_routes = rt715_sdca_audio_map, + .num_dapm_routes = ARRAY_SIZE(rt715_sdca_audio_map), +}; + +static int rt715_sdca_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + struct rt715_sdw_stream_data *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void rt715_sdca_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) + +{ + struct rt715_sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + if (!stream) + return; + + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int rt715_sdca_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 rt715_sdca_priv *rt715 = snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct rt715_sdw_stream_data *stream; + int retval, port, num_channels; + unsigned int val; + + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!rt715->slave) + return -EINVAL; + + switch (dai->id) { + case RT715_AIF1: + direction = SDW_DATA_DIR_TX; + port = 6; + rt715_sdca_index_write(rt715, RT715_VENDOR_REG, RT715_SDW_INPUT_SEL, + 0xa500); + break; + case RT715_AIF2: + direction = SDW_DATA_DIR_TX; + port = 4; + rt715_sdca_index_write(rt715, RT715_VENDOR_REG, RT715_SDW_INPUT_SEL, + 0xaf00); + break; + default: + dev_err(component->dev, "Invalid DAI id %d\n", dai->id); + return -EINVAL; + } + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = params_channels(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + num_channels = params_channels(params); + port_config.ch_mask = GENMASK(num_channels - 1, 0); + port_config.num = port; + + retval = sdw_stream_add_slave(rt715->slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(component->dev, "Unable to configure port, retval:%d\n", + retval); + return retval; + } + + switch (params_rate(params)) { + case 8000: + val = 0x1; + break; + case 11025: + val = 0x2; + break; + case 12000: + val = 0x3; + break; + case 16000: + val = 0x4; + break; + case 22050: + val = 0x5; + break; + case 24000: + val = 0x6; + break; + case 32000: + val = 0x7; + break; + case 44100: + val = 0x8; + break; + case 48000: + val = 0x9; + break; + case 88200: + val = 0xa; + break; + case 96000: + val = 0xb; + break; + case 176400: + val = 0xc; + break; + case 192000: + val = 0xd; + break; + case 384000: + val = 0xe; + break; + case 768000: + val = 0xf; + break; + default: + dev_err(component->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + regmap_write(rt715->regmap, + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_CS_FREQ_IND_EN, + RT715_SDCA_FREQ_IND_CTRL, CH_00), val); + + return 0; +} + +static int rt715_sdca_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt715_sdca_priv *rt715 = snd_soc_component_get_drvdata(component); + struct rt715_sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt715->slave) + return -EINVAL; + + sdw_stream_remove_slave(rt715->slave, stream->sdw_stream); + return 0; +} + +#define RT715_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT715_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static struct snd_soc_dai_ops rt715_sdca_ops = { + .hw_params = rt715_sdca_pcm_hw_params, + .hw_free = rt715_sdca_pcm_hw_free, + .set_sdw_stream = rt715_sdca_set_sdw_stream, + .shutdown = rt715_sdca_shutdown, +}; + +static struct snd_soc_dai_driver rt715_sdca_dai[] = { + { + .name = "rt715-aif1", + .id = RT715_AIF1, + .capture = { + .stream_name = "DP6 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT715_STEREO_RATES, + .formats = RT715_FORMATS, + }, + .ops = &rt715_sdca_ops, + }, + { + .name = "rt715-aif2", + .id = RT715_AIF2, + .capture = { + .stream_name = "DP4 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT715_STEREO_RATES, + .formats = RT715_FORMATS, + }, + .ops = &rt715_sdca_ops, + }, +}; + +/* Bus clock frequency */ +#define RT715_CLK_FREQ_9600000HZ 9600000 +#define RT715_CLK_FREQ_12000000HZ 12000000 +#define RT715_CLK_FREQ_6000000HZ 6000000 +#define RT715_CLK_FREQ_4800000HZ 4800000 +#define RT715_CLK_FREQ_2400000HZ 2400000 +#define RT715_CLK_FREQ_12288000HZ 12288000 + +int rt715_sdca_init(struct device *dev, struct regmap *mbq_regmap, + struct regmap *regmap, struct sdw_slave *slave) +{ + struct rt715_sdca_priv *rt715; + int ret; + + rt715 = devm_kzalloc(dev, sizeof(*rt715), GFP_KERNEL); + if (!rt715) + return -ENOMEM; + + dev_set_drvdata(dev, rt715); + rt715->slave = slave; + rt715->regmap = regmap; + rt715->mbq_regmap = mbq_regmap; + rt715->hw_sdw_ver = slave->id.sdw_version; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt715->hw_init = false; + rt715->first_init = false; + + ret = devm_snd_soc_register_component(dev, + &soc_codec_dev_rt715_sdca, + rt715_sdca_dai, + ARRAY_SIZE(rt715_sdca_dai)); + + return ret; +} + +int rt715_sdca_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt715_sdca_priv *rt715 = dev_get_drvdata(dev); + unsigned int hw_ver; + + if (rt715->hw_init) + return 0; + + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + if (!rt715->first_init) { + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + + rt715->first_init = true; + } + + pm_runtime_get_noresume(&slave->dev); + + rt715_sdca_index_read(rt715, RT715_VENDOR_REG, + RT715_PRODUCT_NUM, &hw_ver); + hw_ver = hw_ver & 0x000f; + + /* set clock selector = external */ + regmap_write(rt715->regmap, + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_CX_CLK_SEL_EN, + RT715_SDCA_CX_CLK_SEL_CTRL, CH_00), 0x1); + /* set GPIO_4/5/6 to be 3rd/4th DMIC usage */ + if (hw_ver == 0x0) + rt715_sdca_index_update_bits(rt715, RT715_VENDOR_REG, + RT715_AD_FUNC_EN, 0x54, 0x54); + else if (hw_ver == 0x1) { + rt715_sdca_index_update_bits(rt715, RT715_VENDOR_REG, + RT715_AD_FUNC_EN, 0x55, 0x55); + rt715_sdca_index_update_bits(rt715, RT715_VENDOR_REG, + RT715_REV_1, 0x40, 0x40); + } + /* trigger mode = VAD enable */ + regmap_write(rt715->regmap, + SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_SMPU_TRIG_ST_EN, + RT715_SDCA_SMPU_TRIG_EN_CTRL, CH_00), 0x2); + /* SMPU-1 interrupt enable mask */ + regmap_update_bits(rt715->regmap, RT715_INT_MASK, 0x1, 0x1); + + /* Mark Slave initialization complete */ + rt715->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + return 0; +} + +MODULE_DESCRIPTION("ASoC rt715 driver SDW SDCA"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt715-sdca.h b/sound/soc/codecs/rt715-sdca.h new file mode 100644 index 000000000000..85ce4d95e5eb --- /dev/null +++ b/sound/soc/codecs/rt715-sdca.h @@ -0,0 +1,136 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt715-sdca.h -- RT715 ALSA SoC audio driver header + * + * Copyright(c) 2020 Realtek Semiconductor Corp. + */ + +#ifndef __RT715_SDCA_H__ +#define __RT715_SDCA_H__ + +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/soc.h> +#include <linux/workqueue.h> +#include <linux/device.h> + +struct rt715_sdca_priv { + struct regmap *regmap; + struct regmap *mbq_regmap; + struct snd_soc_codec *codec; + struct sdw_slave *slave; + struct delayed_work adc_mute_work; + int dbg_nid; + int dbg_vid; + int dbg_payload; + enum sdw_slave_status status; + struct sdw_bus_params params; + bool hw_init; + bool first_init; + int l_is_unmute; + int r_is_unmute; + int hw_sdw_ver; + int kctl_switch_orig[4]; + int kctl_2ch_orig[2]; + int kctl_4ch_orig[4]; + int kctl_8ch_orig[8]; +}; + +struct rt715_sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +struct rt715_sdca_kcontrol_private { + unsigned int reg_base; + unsigned int count; + unsigned int max; + unsigned int shift; + unsigned int invert; +}; + +/* MIPI Register */ +#define RT715_INT_CTRL 0x005a +#define RT715_INT_MASK 0x005e + +/* NID */ +#define RT715_AUDIO_FUNCTION_GROUP 0x01 +#define RT715_MIC_ADC 0x07 +#define RT715_LINE_ADC 0x08 +#define RT715_MIX_ADC 0x09 +#define RT715_DMIC1 0x12 +#define RT715_DMIC2 0x13 +#define RT715_MIC1 0x18 +#define RT715_MIC2 0x19 +#define RT715_LINE1 0x1a +#define RT715_LINE2 0x1b +#define RT715_DMIC3 0x1d +#define RT715_DMIC4 0x29 +#define RT715_VENDOR_REG 0x20 +#define RT715_MUX_IN1 0x22 +#define RT715_MUX_IN2 0x23 +#define RT715_MUX_IN3 0x24 +#define RT715_MUX_IN4 0x25 +#define RT715_MIX_ADC2 0x27 +#define RT715_INLINE_CMD 0x55 +#define RT715_VENDOR_HDA_CTL 0x61 + +/* Index (NID:20h) */ +#define RT715_PRODUCT_NUM 0x0 +#define RT715_IRQ_CTRL 0x2b +#define RT715_AD_FUNC_EN 0x36 +#define RT715_REV_1 0x37 +#define RT715_SDW_INPUT_SEL 0x39 +#define RT715_EXT_DMIC_CLK_CTRL2 0x54 + +/* Index (NID:61h) */ +#define RT715_HDA_LEGACY_MUX_CTL1 0x00 + +/* SDCA (Function) */ +#define FUN_JACK_CODEC 0x01 +#define FUN_MIC_ARRAY 0x02 +#define FUN_HID 0x03 +/* SDCA (Entity) */ +#define RT715_SDCA_ST_EN 0x00 +#define RT715_SDCA_CS_FREQ_IND_EN 0x01 +#define RT715_SDCA_FU_ADC8_9_VOL 0x02 +#define RT715_SDCA_SMPU_TRIG_ST_EN 0x05 +#define RT715_SDCA_FU_ADC10_11_VOL 0x06 +#define RT715_SDCA_FU_ADC7_27_VOL 0x0a +#define RT715_SDCA_FU_AMIC_GAIN_EN 0x0c +#define RT715_SDCA_FU_DMIC_GAIN_EN 0x0e +#define RT715_SDCA_CX_CLK_SEL_EN 0x10 +#define RT715_SDCA_CREQ_POW_EN 0x18 +/* SDCA (Control) */ +#define RT715_SDCA_ST_CTRL 0x00 +#define RT715_SDCA_CX_CLK_SEL_CTRL 0x01 +#define RT715_SDCA_REQ_POW_CTRL 0x01 +#define RT715_SDCA_FU_MUTE_CTRL 0x01 +#define RT715_SDCA_FU_VOL_CTRL 0x02 +#define RT715_SDCA_FU_DMIC_GAIN_CTRL 0x0b +#define RT715_SDCA_FREQ_IND_CTRL 0x10 +#define RT715_SDCA_SMPU_TRIG_EN_CTRL 0x10 +#define RT715_SDCA_SMPU_TRIG_ST_CTRL 0x11 +/* SDCA (Channel) */ +#define CH_00 0x00 +#define CH_01 0x01 +#define CH_02 0x02 +#define CH_03 0x03 +#define CH_04 0x04 +#define CH_05 0x05 +#define CH_06 0x06 +#define CH_07 0x07 +#define CH_08 0x08 + +#define RT715_SDCA_DB_STEP 375 + +enum { + RT715_AIF1, + RT715_AIF2, +}; + +int rt715_sdca_io_init(struct device *dev, struct sdw_slave *slave); +int rt715_sdca_init(struct device *dev, struct regmap *mbq_regmap, + struct regmap *regmap, struct sdw_slave *slave); + +#endif /* __RT715_SDCA_H__ */ -- 2.29.0

2 1

[PATCH] ASoC: fsl_xcvr: Use devm_platform_ioremap_resource_byname() to simplify code
by Tang Bin 08 Mar '21

08 Mar '21

In this function, devm_platform_ioremap_resource_byname() should be suitable to simplify code. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/fsl/fsl_xcvr.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 6dd0a5fcd455..5e8284db857b 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1131,7 +1131,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct fsl_xcvr *xcvr; - struct resource *ram_res, *regs_res, *rx_res, *tx_res; + struct resource *rx_res, *tx_res; void __iomem *regs; int ret, irq; @@ -1166,13 +1166,11 @@ static int fsl_xcvr_probe(struct platform_device *pdev) return PTR_ERR(xcvr->pll_ipg_clk); } - ram_res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "ram"); - xcvr->ram_addr = devm_ioremap_resource(dev, ram_res); + xcvr->ram_addr = devm_platform_ioremap_resource_byname(pdev, "ram"); if (IS_ERR(xcvr->ram_addr)) return PTR_ERR(xcvr->ram_addr); - regs_res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "regs"); - regs = devm_ioremap_resource(dev, regs_res); + regs = devm_platform_ioremap_resource_byname(pdev, "regs"); if (IS_ERR(regs)) return PTR_ERR(regs); -- 2.18.2

2 1

[PATCH] soundwire: bus: Fix device found flag correctly
by Srinivas Kandagatla 08 Mar '21

08 Mar '21

found flag that is used to indicate SoundWire device on the bus is not reset correctly after one iteration, this could miss adding devices that are enumerated on the bus but on in device list. Fixes: d52d7a1be02c ("soundwire: Add Slave status handling helpers") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/bus.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index a9ab4a6264e0..188e391c5a83 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -712,7 +712,7 @@ static int sdw_program_device_num(struct sdw_bus *bus) struct sdw_slave *slave, *_s; struct sdw_slave_id id; struct sdw_msg msg; - bool found = false; + bool found; int count = 0, ret; u64 addr; @@ -744,6 +744,7 @@ static int sdw_program_device_num(struct sdw_bus *bus) sdw_extract_slave_id(bus, addr, &id); + found = false; /* Now compare with entries */ list_for_each_entry_safe(slave, _s, &bus->slaves, node) { if (sdw_compare_devid(slave, id) == 0) { -- 2.21.0

2 1

[PATCH v2 0/8] ASoC: soc-pcm: cleanup each functions
by Kuninori Morimoto 08 Mar '21

08 Mar '21

Hi Mark These are v2 of soc-pcm cleanup patches. These has no relationship to each other. My 1 concern is [3/8] patch. I think it is no problem, but I'm not 100% sure why current code was such code. v1 -> v2 - soc_cpu/codec_dai_name() is now inline function - rename soc_pcm_care_symmetry() to soc_pcm_update_symmetry() Link: https://lore.kernel.org/r/87tupuqqc8.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (8): ASoC: soc-pcm: check DAI activity under soc_pcm_apply_symmetry() ASoC: soc-pcm: add soc_cpu/codec_dai_name() macro ASoC: soc-pcm: direct copy at snd_soc_set_runtime_hwparams() ASoC: soc-pcm: add soc_pcm_update_symmetry() ASoC: soc-pcm: add soc_hw_sanity_check() ASoC: soc-pcm: fixup dpcm_be_dai_startup() user count ASoC: soc-pcm: remove unneeded !rtd->dai_link check ASoC: soc-pcm: share DPCM BE DAI stop operation include/sound/soc-dpcm.h | 8 +- sound/soc/soc-compress.c | 2 +- sound/soc/soc-pcm.c | 243 ++++++++++++++++----------------------- 3 files changed, 105 insertions(+), 148 deletions(-) -- 2.25.1

3 12

[PATCH v3 0/7] Add audio driver base on rpmsg on i.MX platform
by Shengjiu Wang 08 Mar '21

08 Mar '21

On Asymmetric multiprocessor, there is Cortex-A core and Cortex-M core, Linux is running on A core, RTOS is running on M core. The audio hardware device can be controlled by Cortex-M device, So audio playback/capture can be handled by M core. Rpmsg is the interface for sending and receiving msg to and from M core, that we can create a virtual sound on Cortex-A core side. A core will tell the Cortex-M core sound format/rate/channel, where is the data buffer, what is the period size, when to start, when to stop and when suspend or resume happen, each of this behavior there is defined rpmsg command. Especially we designed the low power audio case, that is to allocate a large buffer and fill the data, then Cortex-A core can go to sleep mode, Cortex-M core continue to play the sound, when the buffer is consumed, Cortex-M core will trigger the Cortex-A core to wakeup to fill data. changes in v3: - add local refcount for clk enablement in hw_params() - update the document according Rob's comments changes in v2: - update codes and comments according to Mark's comments Shengjiu Wang (7): ASoC: soc-component: Add snd_soc_pcm_component_ack ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg ASoC: dt-bindings: fsl_rpmsg: Add binding doc for rpmsg cpu dai driver ASoC: imx-audio-rpmsg: Add rpmsg_driver for audio channel ASoC: imx-pcm-rpmsg: Add platform driver for audio base on rpmsg ASoC: imx-rpmsg: Add machine driver for audio base on rpmsg ASoC: dt-bindings: imx-rpmsg: Add binding doc for rpmsg machine driver .../devicetree/bindings/sound/fsl,rpmsg.yaml | 94 ++ .../bindings/sound/imx-audio-rpmsg.yaml | 55 ++ include/sound/soc-component.h | 3 + sound/soc/fsl/Kconfig | 28 + sound/soc/fsl/Makefile | 6 + sound/soc/fsl/fsl_rpmsg.c | 261 +++++ sound/soc/fsl/fsl_rpmsg.h | 40 + sound/soc/fsl/imx-audio-rpmsg.c | 151 +++ sound/soc/fsl/imx-pcm-rpmsg.c | 919 ++++++++++++++++++ sound/soc/fsl/imx-pcm-rpmsg.h | 512 ++++++++++ sound/soc/fsl/imx-rpmsg.c | 148 +++ sound/soc/soc-component.c | 14 + sound/soc/soc-pcm.c | 2 + 13 files changed, 2233 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml create mode 100644 sound/soc/fsl/fsl_rpmsg.c create mode 100644 sound/soc/fsl/fsl_rpmsg.h create mode 100644 sound/soc/fsl/imx-audio-rpmsg.c create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.c create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.h create mode 100644 sound/soc/fsl/imx-rpmsg.c -- 2.27.0

4 10

[PATCH] sound: soc: codecs: Fix a spello in the file wm8955.c
by Bhaskar Chowdhury 08 Mar '21

08 Mar '21

s/sortd/sorted/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- sound/soc/codecs/wm8955.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 513df47bd87d..538bb8b0db39 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -151,7 +151,7 @@ static int wm8955_pll_factors(struct device *dev, /* The oscilator should run at should be 90-100MHz, and * there's a divide by 4 plus an optional divide by 2 in the * output path to generate the system clock. The clock table - * is sortd so we should always generate a suitable target. */ + * is sorted so we should always generate a suitable target. */ target = Fout * 4; if (target < 90000000) { pll->outdiv = 1; -- 2.26.2

3 2

[PATCH] ASoC: Intel: Skylake: Compile when any configuration is selected
by Cezary Rojewski 08 Mar '21

08 Mar '21

Skylake is dependent on SND_SOC_INTEL_SKYLAKE (aka "all SST platforms") whereas selecting specific configuration such as KBL-only will not cause driver code to compile. Switch to SND_SOC_INTEL_SKYLAKE_COMMON dependency so selecting any configuration causes the driver to be built. Reported-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Suggested-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Fixes: 35bc99aaa1a3 ("ASoC: Intel: Skylake: Add more platform granularity") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/Makefile | 2 +- sound/soc/intel/skylake/Makefile | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 4e0248d2accc..7c5038803be7 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -5,7 +5,7 @@ obj-$(CONFIG_SND_SOC) += common/ # Platform Support obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += atom/ obj-$(CONFIG_SND_SOC_INTEL_CATPT) += catpt/ -obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += skylake/ +obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE_COMMON) += skylake/ obj-$(CONFIG_SND_SOC_INTEL_KEEMBAY) += keembay/ # Machine support diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile index dd39149b89b1..1c4649bccec5 100644 --- a/sound/soc/intel/skylake/Makefile +++ b/sound/soc/intel/skylake/Makefile @@ -7,7 +7,7 @@ ifdef CONFIG_DEBUG_FS snd-soc-skl-objs += skl-debug.o endif -obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o +obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE_COMMON) += snd-soc-skl.o #Skylake Clock device support snd-soc-skl-ssp-clk-objs := skl-ssp-clk.o -- 2.17.1

3 12

[RFC 0/2] ASoC: rt5670: Add LED trigger support
by Hans de Goede 07 Mar '21

07 Mar '21

Hi All, This series applies on top of the: "[PATCH 0/4] ASoC: rt5670: Various kcontrol fixes" series which I just send out; And it also requires the: "[PATCH 0/5] ALSA: control - add generic LED trigger code" from Jaroslav. This second requirement is still being discussed / reviewed, which is why I split out these 2 patches into a separate RFC patch series. Regards, Hans Hans de Goede (2): ASoC: Add new SOC_DOUBLE*_ACCESS() macros ASoC: rt5670: Add LED trigger support include/sound/soc.h | 20 ++++++++++++++++---- sound/soc/codecs/rt5670.c | 10 +++++++--- 2 files changed, 23 insertions(+), 7 deletions(-) -- 2.30.1

4 45

[PATCH alsa-lib v3 0/5] mixer: simple - Volume control fixes
by Hans de Goede 07 Mar '21

07 Mar '21

Hi All, Here is v3 of my series of alsa-lib volume-control fixes which I wrote while working on adding hw-volume control to UCM profiles for various x86 ASoC setups. New in version i3 s first adding a new generic exceptions table to deal with controls with names which don't use the expected suffixes to indicate their type. Also new this version is a patch to deal with some output mute switches which are called "Foo Channel Switch" instead of "Foo Playback Switch" (patch 5/5). Note patch 1/5 also fixes a regression introduced by the changes which Jaroslav made to commit 86b9c67774bc ("mixer: simple - Unify simple_none: base_len() exception handling") before merging it. Regards, Hans Hans de Goede (5): mixer: simple - Add generic exception mechanism for non-standard control-names mixer: simple - Move handling of 3D Control - Depth controls to the exceptions list mixer: simple - Add exceptions for non " Volume" suffixed capture vol-ctls used in ASoC realtek codec drivers mixer: simple - Add exceptions for some capture-vol-ctls which have a " Volume" suffix mixer: simple - Add exceptions for some Playback Switches with a " Channel Switch" suffix src/mixer/simple_none.c | 74 +++++++++++++++++++++++++---------------- 1 file changed, 46 insertions(+), 28 deletions(-) -- 2.30.1

1 5

[PATCH v7 0/9] ALSA: add virtio sound driver
by Anton Yakovlev 07 Mar '21

07 Mar '21

This series implements a driver part of the virtio sound device specification v8 [1]. The driver supports PCM playback and capture substreams, jack and channel map controls. A message-based transport is used to write/read PCM frames to/from a device. As a device part was used OpenSynergy proprietary implementation. v7 changes: - Moved the snd_pcm_period_elapsed() call from the interrupt handler to the kernel worker for being consistent with the non-atomic mode of the PCM device. - Removed SNDRV_PCM_INFO_RESUME flag. Now ops->prepare() sets the parameters for the substream if it was previously suspended. - Some additional code readability improvements/comments. [1] https://lists.oasis-open.org/archives/virtio-dev/202003/msg00185.html Anton Yakovlev (9): uapi: virtio_ids: add a sound device type ID from OASIS spec ALSA: virtio: add virtio sound driver ALSA: virtio: handling control messages ALSA: virtio: build PCM devices and substream hardware descriptors ALSA: virtio: handling control and I/O messages for the PCM device ALSA: virtio: PCM substream operators ALSA: virtio: introduce jack support ALSA: virtio: introduce PCM channel map support ALSA: virtio: introduce device suspend/resume support MAINTAINERS | 9 + include/uapi/linux/virtio_ids.h | 1 + include/uapi/linux/virtio_snd.h | 334 +++++++++++++++++++++ sound/Kconfig | 2 + sound/Makefile | 3 +- sound/virtio/Kconfig | 10 + sound/virtio/Makefile | 13 + sound/virtio/virtio_card.c | 449 ++++++++++++++++++++++++++++ sound/virtio/virtio_card.h | 111 +++++++ sound/virtio/virtio_chmap.c | 219 ++++++++++++++ sound/virtio/virtio_ctl_msg.c | 310 +++++++++++++++++++ sound/virtio/virtio_ctl_msg.h | 78 +++++ sound/virtio/virtio_jack.c | 233 +++++++++++++++ sound/virtio/virtio_pcm.c | 513 ++++++++++++++++++++++++++++++++ sound/virtio/virtio_pcm.h | 124 ++++++++ sound/virtio/virtio_pcm_msg.c | 414 ++++++++++++++++++++++++++ sound/virtio/virtio_pcm_ops.c | 464 +++++++++++++++++++++++++++++ 17 files changed, 3286 insertions(+), 1 deletion(-) create mode 100644 include/uapi/linux/virtio_snd.h create mode 100644 sound/virtio/Kconfig create mode 100644 sound/virtio/Makefile create mode 100644 sound/virtio/virtio_card.c create mode 100644 sound/virtio/virtio_card.h create mode 100644 sound/virtio/virtio_chmap.c create mode 100644 sound/virtio/virtio_ctl_msg.c create mode 100644 sound/virtio/virtio_ctl_msg.h create mode 100644 sound/virtio/virtio_jack.c create mode 100644 sound/virtio/virtio_pcm.c create mode 100644 sound/virtio/virtio_pcm.h create mode 100644 sound/virtio/virtio_pcm_msg.c create mode 100644 sound/virtio/virtio_pcm_ops.c -- 2.30.1

2 10

alsactl: init command now honors -g flag as well
by GitHub pull_request - opened 06 Mar '21

06 Mar '21

alsa-project/alsa-utils pull request #80 was opened from MichaIng: As of #75 this commit aligns the manpage with the new behaviour. Request URL : https://github.com/alsa-project/alsa-utils/pull/80 Patch URL : https://github.com/alsa-project/alsa-utils/pull/80.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: hda/conexant: Add quirk for mute LED control on HP ZBook G5
by Takashi Iwai 06 Mar '21

06 Mar '21

The mute and mic-mute LEDs on HP ZBook Studio G5 are controlled via GPIO bits 0x10 and 0x20, respectively, and we need the extra setup for those. As the similar code is already present for other HP models but with different GPIO pins, this patch factors out the common helper code and applies those GPIO values for each model. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211893 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 62 ++++++++++++++++++++++++---------- 1 file changed, 45 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f2aa226d1373..c20dad46a7c9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -149,6 +149,21 @@ static int cx_auto_vmaster_mute_led(struct led_classdev *led_cdev, return 0; } +static void cxt_init_gpio_led(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + unsigned int mask = spec->gpio_mute_led_mask | spec->gpio_mic_led_mask; + + if (mask) { + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, + mask); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, + mask); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_led); + } +} + static int cx_auto_init(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -156,6 +171,7 @@ static int cx_auto_init(struct hda_codec *codec) if (!spec->dynamic_eapd) cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true); + cxt_init_gpio_led(codec); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); return 0; @@ -215,6 +231,7 @@ enum { CXT_FIXUP_HP_SPECTRE, CXT_FIXUP_HP_GATE_MIC, CXT_FIXUP_MUTE_LED_GPIO, + CXT_FIXUP_HP_ZBOOK_MUTE_LED, CXT_FIXUP_HEADSET_MIC, CXT_FIXUP_HP_MIC_NO_PRESENCE, }; @@ -654,31 +671,36 @@ static int cxt_gpio_micmute_update(struct led_classdev *led_cdev, return 0; } - -static void cxt_fixup_mute_led_gpio(struct hda_codec *codec, - const struct hda_fixup *fix, int action) +static void cxt_setup_mute_led(struct hda_codec *codec, + unsigned int mute, unsigned int mic_mute) { struct conexant_spec *spec = codec->spec; - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x03 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03 }, - {} - }; - if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gpio_led = 0; + spec->mute_led_polarity = 0; + if (mute) { snd_hda_gen_add_mute_led_cdev(codec, cxt_gpio_mute_update); - spec->gpio_led = 0; - spec->mute_led_polarity = 0; - spec->gpio_mute_led_mask = 0x01; - spec->gpio_mic_led_mask = 0x02; + spec->gpio_mute_led_mask = mute; + } + if (mic_mute) { snd_hda_gen_add_micmute_led_cdev(codec, cxt_gpio_micmute_update); + spec->gpio_mic_led_mask = mic_mute; } - snd_hda_add_verbs(codec, gpio_init); - if (spec->gpio_led) - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - spec->gpio_led); } +static void cxt_fixup_mute_led_gpio(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + cxt_setup_mute_led(codec, 0x01, 0x02); +} + +static void cxt_fixup_hp_zbook_mute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + cxt_setup_mute_led(codec, 0x10, 0x20); +} /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { @@ -839,6 +861,10 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_mute_led_gpio, }, + [CXT_FIXUP_HP_ZBOOK_MUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_hp_zbook_mute_led, + }, [CXT_FIXUP_HEADSET_MIC] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_headset_mic, @@ -917,6 +943,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8402, "HP ProBook 645 G4", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x8427, "HP ZBook Studio G5", CXT_FIXUP_HP_ZBOOK_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8456, "HP Z2 G4 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8457, "HP Z2 G4 mini", CXT_FIXUP_HP_MIC_NO_PRESENCE), @@ -956,6 +983,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" }, { .id = CXT_FIXUP_HP_DOCK, .name = "hp-dock" }, { .id = CXT_FIXUP_MUTE_LED_GPIO, .name = "mute-led-gpio" }, + { .id = CXT_FIXUP_HP_ZBOOK_MUTE_LED, .name = "hp-zbook-mute-led" }, { .id = CXT_FIXUP_HP_MIC_NO_PRESENCE, .name = "hp-mic-fix" }, {} }; -- 2.26.2

1 0

Use HTTPS instead of HTTP
by GitHub pull_request - opened 05 Mar '21

05 Mar '21

alsa-project/alsa-utils pull request #79 was opened from BrunoVernay: Request URL : https://github.com/alsa-project/alsa-utils/pull/79 Patch URL : https://github.com/alsa-project/alsa-utils/pull/79.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 0/3] soundwire: qcom: add Clock Stop and Auto Enumeration support
by Srinivas Kandagatla 05 Mar '21

05 Mar '21

Qualcomm Soundwire controller have IP built-in feature to enumerate SoundWire slave devices aka Auto Enumeration. This patchset adds support to this feature along with Clock stop feature using runtime PM. Tested it on DB845c, RB5 and SM8250-MTP. thanks, srini Srinivas Kandagatla (3): soundwire: export sdw_compare_devid() and sdw_extract_slave_id() soundwire: qcom: add auto enumeration support soundwire: qcom: add clock stop via runtime pm support drivers/soundwire/bus.c | 4 +- drivers/soundwire/qcom.c | 187 +++++++++++++++++++++++++++++++--- include/linux/soundwire/sdw.h | 2 + 3 files changed, 179 insertions(+), 14 deletions(-) -- 2.21.0

2 13

[PATCH V1] Add debug print in soc-dapm
by Laxminath Kasam 05 Mar '21

05 Mar '21

Add debug print useful for issue analysis in soc-dapm. This print for dapm widgets useful for debugging to find widgets powering up and down. Signed-off-by: Laxminath Kasam <lkasam(a)codeaurora.org> --- sound/soc/soc-dapm.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b005f9e..09d35df 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1717,6 +1717,13 @@ static void dapm_seq_run(struct snd_soc_card *card, break; } + /* + * Add this debug log to keep track of widgets being + * powered-up and powered-down. + */ + dev_dbg(w->dapm->dev, "dapm: powering %s widget %s\n", + power_up ? "up" : "down", w->name); + if (ret < 0) dev_err(w->dapm->dev, "ASoC: Failed to apply widget power: %d\n", ret); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 3

[PATCH v2] ALSA: hda - bind headset buttons to the headphone jack
by Hui Wang 05 Mar '21

05 Mar '21

With the HDA driver, if the headset buttons are supported, an audio Jack will be created for them. This audio Jack is a bit confusing to users since it can't report headphone/mic insertion events but it claims to support these events. And in addition, the driver already builds a headphone Jack and a mic Jack, and most of those buttons are used for headphone playback, so do some change to bind those buttons to the headphone Jack. After this change, the key events are generated from NID 0x55, and are reported to the input layer via headphone jack (NID 0x21). If there is no headphone Jack, then build an audio jack to support those buttons same as previously. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- In the v2, add a function snd_hda_jack_set_button_state(), the codec driver calls it to report the button state. sound/pci/hda/hda_jack.c | 73 ++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_jack.h | 8 ++++ sound/pci/hda/patch_realtek.c | 15 +++++-- 3 files changed, 91 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index ac00866d8032..f29975e3e98d 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -388,6 +388,69 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, } EXPORT_SYMBOL_GPL(snd_hda_jack_set_gating_jack); +/** + * snd_hda_jack_bind_keymap - bind keys generated from one NID to another jack. + * @codec: the HDA codec + * @key_nid: key event is generated by this pin NID + * @keymap: map of key type and key code + * @jack_nid: key reports to the jack of this pin NID + * + * This function is used in the case of key is generated from one NID while is + * reported to the jack of another NID. + */ +int snd_hda_jack_bind_keymap(struct hda_codec *codec, hda_nid_t key_nid, + const struct hda_jack_keymap *keymap, + hda_nid_t jack_nid) +{ + const struct hda_jack_keymap *map; + struct hda_jack_tbl *key_gen = snd_hda_jack_tbl_get(codec, key_nid); + struct hda_jack_tbl *report_to = snd_hda_jack_tbl_get(codec, jack_nid); + + WARN_ON(codec->dp_mst); + + if (!key_gen || !report_to || !report_to->jack) + return -EINVAL; + + key_gen->key_report_jack = jack_nid; + + if (keymap) + for (map = keymap; map->type; map++) + snd_jack_set_key(report_to->jack, map->type, map->key); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hda_jack_bind_keymap); + +/** + * snd_hda_jack_set_button_state - report button event to the hda_jack_tbl button_state. + * @codec: the HDA codec + * @jack_nid: the button event reports to the jack_tbl of this NID + * @button_state: the button event captured by codec + * + * Codec driver calls this function to report the button event. + */ +void snd_hda_jack_set_button_state(struct hda_codec *codec, hda_nid_t jack_nid, + int button_state) +{ + struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, jack_nid); + + if (!jack) + return; + + if (jack->key_report_jack) { + struct hda_jack_tbl *report_to = + snd_hda_jack_tbl_get(codec, jack->key_report_jack); + + if (report_to) { + report_to->button_state = button_state; + return; + } + } + + jack->button_state = button_state; +} +EXPORT_SYMBOL_GPL(snd_hda_jack_set_button_state); + /** * snd_hda_jack_report_sync - sync the states of all jacks and report if changed * @codec: the HDA codec @@ -651,7 +714,15 @@ void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res) } if (!event) return; - event->jack_dirty = 1; + + if (event->key_report_jack) { + struct hda_jack_tbl *report_to = + snd_hda_jack_tbl_get_mst(codec, event->key_report_jack, + event->dev_id); + if (report_to) + report_to->jack_dirty = 1; + } else + event->jack_dirty = 1; call_jack_callback(codec, res, event); snd_hda_jack_report_sync(codec); diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 8ceaf0ef5df1..2abf7aac243a 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -40,6 +40,7 @@ struct hda_jack_tbl { unsigned int block_report:1; /* in a transitional state - do not report to userspace */ hda_nid_t gating_jack; /* valid when gating jack plugged */ hda_nid_t gated_jack; /* gated is dependent on this jack */ + hda_nid_t key_report_jack; /* key reports to this jack */ int type; int button_state; struct snd_jack *jack; @@ -99,6 +100,13 @@ snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid); +int snd_hda_jack_bind_keymap(struct hda_codec *codec, hda_nid_t key_nid, + const struct hda_jack_keymap *keymap, + hda_nid_t jack_nid); + +void snd_hda_jack_set_button_state(struct hda_codec *codec, hda_nid_t jack_nid, + int button_state); + u32 snd_hda_jack_pin_sense(struct hda_codec *codec, hda_nid_t nid, int dev_id); /* the jack state returned from snd_hda_jack_detect_state() */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b47504fa8dfd..c2ae077bd010 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3103,7 +3103,7 @@ static void alc_headset_btn_callback(struct hda_codec *codec, if (jack->unsol_res & (7 << 10)) report |= SND_JACK_BTN_3; - jack->jack->button_state = report; + snd_hda_jack_set_button_state(codec, jack->nid, report); } static void alc_disable_headset_jack_key(struct hda_codec *codec) @@ -3164,16 +3164,23 @@ static void alc_fixup_headset_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: spec->has_hs_key = 1; snd_hda_jack_detect_enable_callback(codec, 0x55, alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, - SND_JACK_HEADSET, alc_headset_btn_keymap); break; - case HDA_FIXUP_ACT_INIT: + case HDA_FIXUP_ACT_BUILD: + hp_pin = alc_get_hp_pin(spec); + if (!hp_pin || snd_hda_jack_bind_keymap(codec, 0x55, + alc_headset_btn_keymap, + hp_pin)) + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", + false, SND_JACK_HEADSET, + alc_headset_btn_keymap); + alc_enable_headset_jack_key(codec); break; } -- 2.25.1

2 1

[PATCH alsa-lib v2 0/3] mixer: Volume control fixes
by Hans de Goede 05 Mar '21

05 Mar '21

Hi All, Here is v2 of my series of alsa-lib volume-control fixes which I wrote while working on adding hw-volume control to UCM profiles for various x86 ASoC setups. New in v2 is adding an extra entry to the capture_volume_names array for the "ADC PGA Gain Volume" control used in the es8316 codec driver in patch 3/3. Note that patch 3/3 is not strictly necessary to make hw-volume control work. At least the PA hw-vol-control code does not care if the controls are wronly marked as global-volume-controls instead of capture-volume-controls. I mostly wrote patch 3/3 because for correctness reasons, so if it is deemed controversial we can drop it. Regards, Hans Hans de Goede (3): mixer: Unify simple_none: base_len() exception handling mixer: Add exception for non " Volume" suffixed capture vol-ctls used in ASoC realtek codec drivers mixer: Add exception for some capture-vol-ctls which have a " Volume" suffix src/mixer/simple_none.c | 83 +++++++++++++++++++++++++++++++---------- 1 file changed, 64 insertions(+), 19 deletions(-) -- 2.30.1

2 6

[PATCH v2 0/4] ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec
by Vitaly Rodionov 05 Mar '21

05 Mar '21

Dell's laptops Inspiron 3500, Inspiron 3501, Inspiron 3505 are using Cirrus Logic CS8409 HDA bridge with CS42L42 companion codec. The CS8409 is a multichannel HD audio routing controller. CS8409 includes support for four channels of digital microphone data and two bidirectional ASPs for up to 32 channels of TDM data or 4 channels of I2S data. The CS8409 is intended to be used with a remote companion codec that implements high performance analog functions in close physical proximity to the end-equipment audio port or speaker driver. The CS42L42 is a low-power audio codec with integrated MIPI SoundWire interface or I2C/I2S/TDM interfaces designed for portable applications. It provides a high-dynamic range, stereo DAC for audio playback and a mono high-dynamic-range ADC for audio capture Changes since version 1: ALSA: hda/cirrus: Increase AUTO_CFG_MAX_INS from 8 to 18 * No change ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec. * Removed redundant fields in fixup table * Handle gpio via spec->gpio_dir, spec->gpio_data and spec->gpio_mask * Moved cs8409_cs42l42_init() from patch 2, to handle resume correctly ALSA: hda/cirrus: Add jack detect interrupt support from CS42L42 companion codec. * Run scripts/checkpatch.pl, fixed new warnings ALSA: hda/cirrus: Add Headphone and Headset MIC Volume Control * Moved control values to cache to avoid i2c read at each time. Stefan Binding (1): ALSA: hda/cirrus: Add Headphone and Headset MIC Volume Control Vitaly Rodionov (3): ALSA: hda/cirrus: Increase AUTO_CFG_MAX_INS from 8 to 18 ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec. ALSA: hda/cirrus: Add jack detect interrupt support from CS42L42 companion codec. sound/pci/hda/hda_auto_parser.h | 2 +- sound/pci/hda/hda_local.h | 2 +- sound/pci/hda/patch_cirrus.c | 1068 +++++++++++++++++++++++++++++++ 3 files changed, 1070 insertions(+), 2 deletions(-) -- 2.25.1

3 8

[PATCH v3 1/3] platform/x86: dell-privacy: Add support for Dell hardware privacy
by Perry Yuan 05 Mar '21

05 Mar '21

From: Perry Yuan <perry_yuan(a)dell.com> add support for dell privacy driver for the dell units equipped hardware privacy design, which protect users privacy of audio and camera from hardware level. once the audio or camera privacy mode enabled, any applications will not get any audio or video stream when user pressed ctrl+F4 hotkey, audio privacy mode will be enabled,Micmute led will be also changed accordingly The micmute led is fully controlled by hardware & EC(embedded controller) and camera mute hotkey is ctrl+f9. currently design only emmit SW_CAMERA_LENS_COVER event while the camera lens shutter will be changed by EC & hw(hadware) control *The flow is like this: 1) User presses key. HW does stuff with this key (timeout is started) 2) WMI event is emitted from BIOS 3) WMI event is received by dell-privacy 4) KEY_MICMUTE emitted from dell-privacy 5) Userland picks up key and modifies kcontrol for SW mute 6) Codec kernel driver catches and calls ledtrig_audio_set, like this: ledtrig_audio_set(LED_AUDIO_MICMUTE, rt715->micmute_led ? LED_ON :LED_OFF); 7) If "LED" is set to on dell-privacy notifies EC, and timeout is cancelled, HW mic mute activated. Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> --- v3 -> v4: * changed as Barnabás`s review comments v2 -> v3: * add sysfs attributes doc v1 -> v2: * query EC handle from EC driver directly * fix some code style * add KEY_END to keymap array * clean platform device when cleanup called * use hexadecimal format for log print in dev_dbg * remove __set_bit for the report keys from probe. * fix keymap leak * add err_free_keymap in dell_privacy_wmi_probe * wmi driver will be unregistered if privacy_acpi_init() fails * add sysfs attribute files for user space query. * add leds micmute driver to privacy acpi * add more design info the commit info --- --- drivers/platform/x86/Kconfig | 17 ++ drivers/platform/x86/Makefile | 4 +- drivers/platform/x86/dell-laptop.c | 25 +- drivers/platform/x86/dell-privacy-acpi.c | 167 ++++++++++++ drivers/platform/x86/dell-privacy-wmi.c | 320 +++++++++++++++++++++++ drivers/platform/x86/dell-privacy-wmi.h | 33 +++ drivers/platform/x86/dell-wmi.c | 37 ++- 7 files changed, 587 insertions(+), 16 deletions(-) create mode 100644 drivers/platform/x86/dell-privacy-acpi.c create mode 100644 drivers/platform/x86/dell-privacy-wmi.c create mode 100644 drivers/platform/x86/dell-privacy-wmi.h diff --git a/drivers/platform/x86/Kconfig b/drivers/platform/x86/Kconfig index 91e6176cdfbd..9d5cc2454b3e 100644 --- a/drivers/platform/x86/Kconfig +++ b/drivers/platform/x86/Kconfig @@ -491,6 +491,23 @@ config DELL_WMI_LED This adds support for the Latitude 2100 and similar notebooks that have an external LED. +config DELL_PRIVACY + tristate "Dell Hardware Privacy Support" + depends on ACPI + depends on ACPI_WMI + depends on INPUT + depends on DELL_LAPTOP + depends on LEDS_TRIGGER_AUDIO + select DELL_WMI + help + This driver provides support for the "Dell Hardware Privacy" feature + of Dell laptops. + Support for a micmute and camera mute privacy will be provided as + hardware button Ctrl+F4 and Ctrl+F9 hotkey + + To compile this driver as a module, choose M here: the module will + be called dell_privacy. + config AMILO_RFKILL tristate "Fujitsu-Siemens Amilo rfkill support" depends on RFKILL diff --git a/drivers/platform/x86/Makefile b/drivers/platform/x86/Makefile index 581475f59819..18c430456de7 100644 --- a/drivers/platform/x86/Makefile +++ b/drivers/platform/x86/Makefile @@ -51,7 +51,9 @@ obj-$(CONFIG_DELL_WMI_DESCRIPTOR) += dell-wmi-descriptor.o obj-$(CONFIG_DELL_WMI_AIO) += dell-wmi-aio.o obj-$(CONFIG_DELL_WMI_LED) += dell-wmi-led.o obj-$(CONFIG_DELL_WMI_SYSMAN) += dell-wmi-sysman/ - +obj-$(CONFIG_DELL_PRIVACY) += dell-privacy.o +dell-privacy-objs := dell-privacy-wmi.o \ + dell-privacy-acpi.o # Fujitsu obj-$(CONFIG_AMILO_RFKILL) += amilo-rfkill.o obj-$(CONFIG_FUJITSU_LAPTOP) += fujitsu-laptop.o diff --git a/drivers/platform/x86/dell-laptop.c b/drivers/platform/x86/dell-laptop.c index 70edc5bb3a14..2fea1f34fcf9 100644 --- a/drivers/platform/x86/dell-laptop.c +++ b/drivers/platform/x86/dell-laptop.c @@ -30,6 +30,7 @@ #include <acpi/video.h> #include "dell-rbtn.h" #include "dell-smbios.h" +#include "dell-privacy-wmi.h" struct quirk_entry { bool touchpad_led; @@ -90,6 +91,7 @@ static struct rfkill *wifi_rfkill; static struct rfkill *bluetooth_rfkill; static struct rfkill *wwan_rfkill; static bool force_rfkill; +static bool privacy_valid; module_param(force_rfkill, bool, 0444); MODULE_PARM_DESC(force_rfkill, "enable rfkill on non whitelisted models"); @@ -2205,11 +2207,18 @@ static int __init dell_init(void) dell_laptop_register_notifier(&dell_laptop_notifier); if (dell_smbios_find_token(GLOBAL_MIC_MUTE_DISABLE) && - dell_smbios_find_token(GLOBAL_MIC_MUTE_ENABLE)) { - micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); - ret = led_classdev_register(&platform_device->dev, &micmute_led_cdev); - if (ret < 0) - goto fail_led; + dell_smbios_find_token(GLOBAL_MIC_MUTE_ENABLE)) { +#if IS_ENABLED(CONFIG_DELL_PRIVACY) + ret = dell_privacy_valid(); + if (!ret) + privacy_valid = true; +#endif + if (!privacy_valid) { + micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + ret = led_classdev_register(&platform_device->dev, &micmute_led_cdev); + if (ret < 0) + goto fail_led; + } } if (acpi_video_get_backlight_type() != acpi_backlight_vendor) @@ -2257,7 +2266,8 @@ static int __init dell_init(void) fail_get_brightness: backlight_device_unregister(dell_backlight_device); fail_backlight: - led_classdev_unregister(&micmute_led_cdev); + if (!privacy_valid) + led_classdev_unregister(&micmute_led_cdev); fail_led: dell_cleanup_rfkill(); fail_rfkill: @@ -2278,7 +2288,8 @@ static void __exit dell_exit(void) touchpad_led_exit(); kbd_led_exit(); backlight_device_unregister(dell_backlight_device); - led_classdev_unregister(&micmute_led_cdev); + if (!privacy_valid) + led_classdev_unregister(&micmute_led_cdev); dell_cleanup_rfkill(); if (platform_device) { platform_device_unregister(platform_device); diff --git a/drivers/platform/x86/dell-privacy-acpi.c b/drivers/platform/x86/dell-privacy-acpi.c new file mode 100644 index 000000000000..df6a86e1345c --- /dev/null +++ b/drivers/platform/x86/dell-privacy-acpi.c @@ -0,0 +1,167 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/acpi.h> +#include <linux/bits.h> +#include <linux/device.h> +#include <linux/fs.h> +#include <linux/kernel.h> +#include <linux/leds.h> +#include <linux/module.h> +#include <linux/mutex.h> +#include <linux/platform_device.h> +#include <linux/string.h> +#include <linux/sysfs.h> +#include <linux/slab.h> +#include <linux/types.h> +#include <linux/wmi.h> + +#include "dell-privacy-wmi.h" + +#define PRIVACY_PLATFORM_NAME "dell-privacy-acpi" +#define DELL_PRIVACY_GUID "6932965F-1671-4CEB-B988-D3AB0A901919" + +struct privacy_acpi_priv { + struct device *dev; + struct platform_device *platform_device; + struct led_classdev cdev; +}; +static struct privacy_acpi_priv *privacy_acpi; + +static int dell_privacy_micmute_led_set(struct led_classdev *led_cdev, + enum led_brightness brightness) +{ + struct privacy_acpi_priv *priv = privacy_acpi; + acpi_status status; + acpi_handle handle; + char *acpi_method; + + handle = ec_get_handle(); + if (!handle) + return -EIO; + acpi_method = "ECAK"; + status = acpi_evaluate_object(handle, acpi_method, NULL, NULL); + if (ACPI_FAILURE(status)) { + dev_err(priv->dev, "Error setting privacy EC ack value: %s\n", + acpi_format_exception(status)); + return -EIO; + } + return 0; +} + +static int dell_privacy_acpi_remove(struct platform_device *pdev) +{ + struct privacy_acpi_priv *priv = dev_get_drvdata(privacy_acpi->dev); + + led_classdev_unregister(&priv->cdev); + + return 0; +} +/* + * Pressing the mute key activates a time delayed circuit to physically cut + * off the mute. The LED is in the same circuit, so it reflects the true + * state of the HW mute. The reason for the EC "ack" is so that software + * can first invoke a SW mute before the HW circuit is cut off. Without SW + * cutting this off first does not affect the time delayed muting or status + * of the LED but there is a possibility of a "popping" noise. + * + * If the EC receives the SW ack, the circuit will be activated before the + * delay completed. + * + * Exposing as an LED device allows the codec drivers notification path to + * EC ACK to work + */ +static int dell_privacy_leds_setup(struct device *dev) +{ + struct privacy_acpi_priv *priv = dev_get_drvdata(dev); + int ret = 0; + + priv->cdev.name = "dell-privacy::micmute"; + priv->cdev.max_brightness = 1; + priv->cdev.brightness_set_blocking = dell_privacy_micmute_led_set; + priv->cdev.default_trigger = "audio-micmute"; + priv->cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + ret = devm_led_classdev_register(dev, &priv->cdev); + if (ret < 0) + return ret; + return 0; +} + +static int dell_privacy_acpi_probe(struct platform_device *pdev) +{ + platform_set_drvdata(pdev, privacy_acpi); + privacy_acpi->dev = &pdev->dev; + privacy_acpi->platform_device = pdev; + return 0; +} + +static const struct acpi_device_id privacy_acpi_device_ids[] = { + {"PNP0C09", 0}, + { }, +}; +MODULE_DEVICE_TABLE(acpi, privacy_acpi_device_ids); + +static struct platform_driver dell_privacy_platform_drv = { + .driver = { + .name = PRIVACY_PLATFORM_NAME, + .acpi_match_table = ACPI_PTR(privacy_acpi_device_ids), + }, + .remove = dell_privacy_acpi_remove, +}; + +int __init dell_privacy_acpi_init(void) +{ + int err; + struct platform_device *pdev; + int privacy_capable = wmi_has_guid(DELL_PRIVACY_GUID); + + if (!wmi_has_guid(DELL_PRIVACY_GUID)) + return -ENODEV; + + privacy_acpi = kzalloc(sizeof(*privacy_acpi), GFP_KERNEL); + if (!privacy_acpi) + return -ENOMEM; + + pdev = platform_device_register_simple( + PRIVACY_PLATFORM_NAME, PLATFORM_DEVID_NONE, NULL, 0); + if (IS_ERR(pdev)) { + err = PTR_ERR(pdev); + goto pdev_err; + } + err = platform_driver_probe(&dell_privacy_platform_drv, + dell_privacy_acpi_probe); + if (err) + goto pdrv_err; + + err = dell_privacy_leds_setup(&pdev->dev); + if (err < 0) + goto pdrv_err; + + return 0; + +pdrv_err: + platform_device_unregister(pdev); +pdev_err: + kfree(privacy_acpi); + return err; +} + +void __exit dell_privacy_acpi_exit(void) +{ + struct platform_device *pdev = to_platform_device(privacy_acpi->dev); + + platform_device_unregister(pdev); + platform_driver_unregister(&dell_privacy_platform_drv); + kfree(privacy_acpi); +} + +MODULE_AUTHOR("Perry Yuan <perry_yuan(a)dell.com>"); +MODULE_DESCRIPTION("DELL Privacy ACPI Driver"); +MODULE_LICENSE("GPL"); diff --git a/drivers/platform/x86/dell-privacy-wmi.c b/drivers/platform/x86/dell-privacy-wmi.c new file mode 100644 index 000000000000..432a3f4ed226 --- /dev/null +++ b/drivers/platform/x86/dell-privacy-wmi.c @@ -0,0 +1,320 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/acpi.h> +#include <linux/input.h> +#include <linux/input/sparse-keymap.h> +#include <linux/list.h> +#include <linux/module.h> +#include <linux/wmi.h> +#include "dell-privacy-wmi.h" + +#define DELL_PRIVACY_GUID "6932965F-1671-4CEB-B988-D3AB0A901919" +#define MICROPHONE_STATUS BIT(0) +#define CAMERA_STATUS BIT(1) +#define PRIVACY_SCREEN_STATUS BIT(2) + +static int privacy_valid = -EPROBE_DEFER; +static LIST_HEAD(wmi_list); +static DEFINE_MUTEX(list_mutex); + +struct privacy_wmi_data { + struct input_dev *input_dev; + struct wmi_device *wdev; + struct list_head list; + u32 features_present; + u32 last_status; +}; + +/* + * Keymap for WMI privacy events of type 0x0012 + */ +static const struct key_entry dell_wmi_keymap_type_0012[] = { + /* privacy mic mute */ + { KE_KEY, 0x0001, { KEY_MICMUTE } }, + /* privacy camera mute */ + { KE_SW, 0x0002, { SW_CAMERA_LENS_COVER } }, + { KE_END, 0}, +}; + +int dell_privacy_valid(void) +{ + int ret; + + ret = wmi_has_guid(DELL_PRIVACY_GUID); + if (!ret) + return -ENODEV; + ret = privacy_valid; + return ret; +} +EXPORT_SYMBOL_GPL(dell_privacy_valid); + +void dell_privacy_process_event(int type, int code, int status) +{ + struct privacy_wmi_data *priv; + const struct key_entry *key; + + mutex_lock(&list_mutex); + priv = list_first_entry_or_null(&wmi_list, + struct privacy_wmi_data, + list); + if (!priv) { + pr_err("dell privacy priv is NULL\n"); + goto error; + } + key = sparse_keymap_entry_from_scancode(priv->input_dev, (type << 16)|code); + if (!key) { + dev_dbg(&priv->wdev->dev, "Unknown key with type 0x%04x and code 0x%04x pressed\n", + type, code); + goto error; + } + switch (code) { + case DELL_PRIVACY_TYPE_AUDIO: /* Mic mute */ + priv->last_status = status; + sparse_keymap_report_entry(priv->input_dev, key, 1, true); + break; + case DELL_PRIVACY_TYPE_CAMERA: /* Camera mute */ + priv->last_status = status; + sparse_keymap_report_entry(priv->input_dev, key, 1, true); + break; + default: + dev_dbg(&priv->wdev->dev, "unknown event type 0x%04x 0x%04x", + type, code); + } +error: + mutex_unlock(&list_mutex); +} +EXPORT_SYMBOL_GPL(dell_privacy_process_event); + +static ssize_t devices_supported_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + + return sysfs_emit(buf, "%x\n", priv->features_present); +} + +static ssize_t current_state_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + + return sysfs_emit(buf, "%x\n", priv->last_status); +} + +static DEVICE_ATTR_RO(devices_supported); +static DEVICE_ATTR_RO(current_state); + +static struct attribute *privacy_attributes[] = { + &dev_attr_devices_supported.attr, + &dev_attr_current_state.attr, + NULL, +}; + +static const struct attribute_group privacy_attribute_group = { + .attrs = privacy_attributes +}; + +/* + * Describes the Device State class exposed by BIOS which can be consumed by + * various applications interested in knowing the Privacy feature capabilities. + * class DeviceState + * { + * [key, read] string InstanceName; + * [read] boolean ReadOnly; + * [WmiDataId(1), read] uint32 DevicesSupported; + * 0 – None, 0x1 – Microphone, 0x2 – Camera, 0x4 -ePrivacy Screen + * [WmiDataId(2), read] uint32 CurrentState; + * 0:Off; 1:On. Bit0 – Microphone, Bit1 – Camera, Bit2 - ePrivacyScreen + * }; + */ + +static int get_current_status(struct wmi_device *wdev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(&wdev->dev); + union acpi_object *obj_present; + u32 *buffer; + int ret = 0; + + if (!priv) { + pr_err("dell privacy priv is NULL\n"); + return -EINVAL; + } + /* check privacy support features and device states */ + obj_present = wmidev_block_query(wdev, 0); + if (!obj_present) { + dev_err(&wdev->dev, "failed to read Binary MOF\n"); + ret = -EIO; + privacy_valid = ret; + return ret; + } + + if (obj_present->type != ACPI_TYPE_BUFFER) { + dev_err(&wdev->dev, "Binary MOF is not a buffer!\n"); + ret = -EIO; + privacy_valid = ret; + goto obj_free; + } + /* Although it's not technically a failure, this would lead to + * unexpected behavior + */ + if (obj_present->buffer.length != 8) { + dev_err(&wdev->dev, "Dell privacy buffer has unexpected length (%d)!\n", + obj_present->buffer.length); + ret = -EINVAL; + privacy_valid = ret; + goto obj_free; + } + buffer = (u32 *)obj_present->buffer.pointer; + priv->features_present = buffer[0]; + priv->last_status = buffer[1]; + privacy_valid = 0; + +obj_free: + kfree(obj_present); + return ret; +} + +static int dell_privacy_wmi_probe(struct wmi_device *wdev, const void *context) +{ + struct privacy_wmi_data *priv; + struct key_entry *keymap; + int ret, i, pos = 0; + + priv = devm_kzalloc(&wdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(&wdev->dev, priv); + priv->wdev = wdev; + /* create evdev passing interface */ + priv->input_dev = devm_input_allocate_device(&wdev->dev); + if (!priv->input_dev) + return -ENOMEM; + /* remap the wmi keymap event to new keymap */ + keymap = kcalloc(ARRAY_SIZE(dell_wmi_keymap_type_0012) + + 1, + sizeof(struct key_entry), GFP_KERNEL); + if (!keymap) { + ret = -ENOMEM; + goto err_free_dev; + } + /* remap the keymap code with Dell privacy key type 0x12 as prefix + * KEY_MICMUTE scancode will be reported as 0x120001 + */ + for (i = 0; i < ARRAY_SIZE(dell_wmi_keymap_type_0012); i++) { + keymap[pos] = dell_wmi_keymap_type_0012[i]; + keymap[pos].code |= (0x0012 << 16); + pos++; + } + ret = sparse_keymap_setup(priv->input_dev, keymap, NULL); + if (ret) + return ret; + priv->input_dev->dev.parent = &wdev->dev; + priv->input_dev->name = "Dell Privacy Driver"; + priv->input_dev->id.bustype = BUS_HOST; + if (input_register_device(priv->input_dev)) { + pr_debug("input_register_device failed to register!\n"); + goto err_free_keymap; + } + mutex_lock(&list_mutex); + list_add_tail(&priv->list, &wmi_list); + mutex_unlock(&list_mutex); + if (get_current_status(priv->wdev)) + goto err_free_input; + ret = devm_device_add_group(&wdev->dev, &privacy_attribute_group); + if (ret) + goto err_free_input; + kfree(keymap); + return 0; + +err_free_input: + input_unregister_device(priv->input_dev); +err_free_keymap: + privacy_valid = -ENODEV; + kfree(keymap); +err_free_dev: + input_free_device(priv->input_dev); + return ret; +} + +static int dell_privacy_wmi_remove(struct wmi_device *wdev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(&wdev->dev); + + mutex_lock(&list_mutex); + list_del(&priv->list); + mutex_unlock(&list_mutex); + privacy_valid = -ENODEV; + input_unregister_device(priv->input_dev); + + return 0; +} + +static const struct wmi_device_id dell_wmi_privacy_wmi_id_table[] = { + { .guid_string = DELL_PRIVACY_GUID }, + { }, +}; + +static struct wmi_driver dell_privacy_wmi_driver = { + .driver = { + .name = "dell-privacy", + }, + .probe = dell_privacy_wmi_probe, + .remove = dell_privacy_wmi_remove, + .id_table = dell_wmi_privacy_wmi_id_table, +}; + +static int __init init_dell_privacy(void) +{ + int ret; + + ret = wmi_has_guid(DELL_PRIVACY_GUID); + if (!ret) + return -ENODEV; + + ret = wmi_driver_register(&dell_privacy_wmi_driver); + if (ret) { + pr_err("failed to initialize privacy wmi driver: %d\n", ret); + return ret; + } + + ret = dell_privacy_acpi_init(); + if (ret) { + pr_err("failed to initialize privacy acpi driver: %d\n", ret); + goto err_init; + } + + return 0; + +err_init: + wmi_driver_unregister(&dell_privacy_wmi_driver); + return ret; +} + +static void dell_privacy_wmi_exit(void) +{ + wmi_driver_unregister(&dell_privacy_wmi_driver); +} + +static void __exit exit_dell_privacy(void) +{ + dell_privacy_wmi_exit(); + dell_privacy_acpi_exit(); +} + +module_init(init_dell_privacy); +module_exit(exit_dell_privacy); + +MODULE_DEVICE_TABLE(wmi, dell_wmi_privacy_wmi_id_table); +MODULE_AUTHOR("Perry Yuan <perry_yuan(a)dell.com>"); +MODULE_DESCRIPTION("Dell Privacy WMI Driver"); +MODULE_LICENSE("GPL"); diff --git a/drivers/platform/x86/dell-privacy-wmi.h b/drivers/platform/x86/dell-privacy-wmi.h new file mode 100644 index 000000000000..9fa01d084f7d --- /dev/null +++ b/drivers/platform/x86/dell-privacy-wmi.h @@ -0,0 +1,33 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#ifndef _DELL_PRIVACY_WMI_H_ +#define _DELL_PRIVACY_WMI_H_ + +#if IS_ENABLED(CONFIG_DELL_PRIVACY) +int dell_privacy_valid(void); +void dell_privacy_process_event(int type, int code, int status); +#else /* CONFIG_DELL_PRIVACY */ +static inline int dell_privacy_valid(void) +{ + return -ENODEV; +} + +static inline void dell_privacy_process_event(int type, int code, int status) +{} +#endif /* CONFIG_DELL_PRIVACY */ + +int dell_privacy_acpi_init(void); +void dell_privacy_acpi_exit(void); + +/* DELL Privacy Type */ +enum { + DELL_PRIVACY_TYPE_UNKNOWN = 0x0, + DELL_PRIVACY_TYPE_AUDIO, + DELL_PRIVACY_TYPE_CAMERA, +}; +#endif diff --git a/drivers/platform/x86/dell-wmi.c b/drivers/platform/x86/dell-wmi.c index bbdb3e860892..93f478edf5d1 100644 --- a/drivers/platform/x86/dell-wmi.c +++ b/drivers/platform/x86/dell-wmi.c @@ -13,20 +13,22 @@ #define pr_fmt(fmt) KBUILD_MODNAME ": " fmt -#include <linux/kernel.h> -#include <linux/module.h> +#include <linux/acpi.h> +#include <linux/dmi.h> #include <linux/init.h> -#include <linux/slab.h> -#include <linux/types.h> #include <linux/input.h> #include <linux/input/sparse-keymap.h> -#include <linux/acpi.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/slab.h> #include <linux/string.h> -#include <linux/dmi.h> +#include <linux/types.h> #include <linux/wmi.h> #include <acpi/video.h> + #include "dell-smbios.h" #include "dell-wmi-descriptor.h" +#include "dell-privacy-wmi.h" MODULE_AUTHOR("Matthew Garrett <mjg(a)redhat.com>"); MODULE_AUTHOR("Pali Rohár <pali(a)kernel.org>"); @@ -381,6 +383,7 @@ static void dell_wmi_notify(struct wmi_device *wdev, u16 *buffer_entry, *buffer_end; acpi_size buffer_size; int len, i; + int err; if (obj->type != ACPI_TYPE_BUFFER) { pr_warn("bad response type %x\n", obj->type); @@ -427,10 +430,9 @@ static void dell_wmi_notify(struct wmi_device *wdev, switch (buffer_entry[1]) { case 0x0000: /* One key pressed or event occurred */ - case 0x0012: /* Event with extended data occurred */ if (len > 2) dell_wmi_process_key(wdev, buffer_entry[1], - buffer_entry[2]); + buffer_entry[2]); /* Extended data is currently ignored */ break; case 0x0010: /* Sequence of keys pressed */ @@ -439,6 +441,25 @@ static void dell_wmi_notify(struct wmi_device *wdev, dell_wmi_process_key(wdev, buffer_entry[1], buffer_entry[i]); break; + case 0x0012: +#if IS_ENABLED(CONFIG_DELL_PRIVACY) + err = dell_privacy_valid(); + if (err == 0) { + dell_privacy_process_event(buffer_entry[1], + buffer_entry[3], buffer_entry[4]); + } else { + if (len > 2) + dell_wmi_process_key(wdev, buffer_entry[1], + buffer_entry[2]); + /* Extended data is currently ignored */ + } +#else + if (len > 2) + dell_wmi_process_key(wdev, buffer_entry[1], + buffer_entry[2]); + /* Extended data is currently ignored */ +#endif + break; default: /* Unknown event */ pr_info("Unknown WMI event type 0x%x\n", (int)buffer_entry[1]); -- 2.25.1

5 9

[PATCH v2 0/7] Add audio driver base on rpmsg on i.MX platform
by Shengjiu Wang 05 Mar '21

05 Mar '21

On Asymmetric multiprocessor, there is Cortex-A core and Cortex-M core, Linux is running on A core, RTOS is running on M core. The audio hardware device can be controlled by Cortex-M device, So audio playback/capture can be handled by M core. Rpmsg is the interface for sending and receiving msg to and from M core, that we can create a virtual sound on Cortex-A core side. A core will tell the Cortex-M core sound format/rate/channel, where is the data buffer, what is the period size, when to start, when to stop and when suspend or resume happen, each of this behavior there is defined rpmsg command. Especially we designed the low power audio case, that is to allocate a large buffer and fill the data, then Cortex-A core can go to sleep mode, Cortex-M core continue to play the sound, when the buffer is consumed, Cortex-M core will trigger the Cortex-A core to wakeup to fill data. changes in v2: - update codes and comments according to Mark's comments Shengjiu Wang (7): ASoC: soc-component: Add snd_soc_pcm_component_ack ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg ASoC: dt-bindings: fsl_rpmsg: Add binding doc for rpmsg cpu dai driver ASoC: imx-audio-rpmsg: Add rpmsg_driver for audio channel ASoC: imx-pcm-rpmsg: Add platform driver for audio base on rpmsg ASoC: imx-rpmsg: Add machine driver for audio base on rpmsg ASoC: dt-bindings: imx-rpmsg: Add binding doc for rpmsg machine driver .../devicetree/bindings/sound/fsl,rpmsg.yaml | 80 ++ .../bindings/sound/imx-audio-rpmsg.yaml | 48 + include/sound/soc-component.h | 3 + sound/soc/fsl/Kconfig | 28 + sound/soc/fsl/Makefile | 6 + sound/soc/fsl/fsl_rpmsg.c | 252 +++++ sound/soc/fsl/fsl_rpmsg.h | 38 + sound/soc/fsl/imx-audio-rpmsg.c | 151 +++ sound/soc/fsl/imx-pcm-rpmsg.c | 919 ++++++++++++++++++ sound/soc/fsl/imx-pcm-rpmsg.h | 512 ++++++++++ sound/soc/fsl/imx-rpmsg.c | 148 +++ sound/soc/soc-component.c | 14 + sound/soc/soc-pcm.c | 2 + 13 files changed, 2201 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml create mode 100644 sound/soc/fsl/fsl_rpmsg.c create mode 100644 sound/soc/fsl/fsl_rpmsg.h create mode 100644 sound/soc/fsl/imx-audio-rpmsg.c create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.c create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.h create mode 100644 sound/soc/fsl/imx-rpmsg.c -- 2.27.0

5 21

[PATCH 0/8] ASoC: soc-pcm: cleanup each functions
by Kuninori Morimoto 05 Mar '21

05 Mar '21

Hi Mark These are soc-pcm cleanup patches, but has no relationship to each other. My 1 concern is [3/8] patch. I think it is no problem, but I'm not 100% sure why current code was such code. Kuninori Morimoto (8): 1) ASoC: soc-pcm: check DAI activity under soc_pcm_apply_symmetry() 2) ASoC: soc-pcm: add soc_cpu/codec_dai_name() macro 3) ASoC: soc-pcm: direct copy at snd_soc_set_runtime_hwparams() 4) ASoC: soc-pcm: add soc_pcm_care_symmetry() 5) ASoC: soc-pcm: add soc_hw_sanity_check() 6) ASoC: soc-pcm: fixup dpcm_be_dai_startup() user count 7) ASoC: soc-pcm: remove unneeded !rtd->dai_link check 8) ASoC: soc-pcm: share DPCM BE DAI stop operation include/sound/soc-dpcm.h | 8 +- sound/soc/soc-compress.c | 2 +- sound/soc/soc-pcm.c | 239 +++++++++++++++------------------------ 3 files changed, 101 insertions(+), 148 deletions(-) -- 2.25.1

2 12

Re: [PATCH 4/4] ALSA: hda/cirrus: Add Headphone and Headset MIC Volume Control
by Takashi Iwai 04 Mar '21

04 Mar '21

On Wed, 03 Mar 2021 19:29:59 +0100, Vitaly Rodionov wrote: > > +static int cs8409_cs42l42_volume_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); > + hda_nid_t nid = get_amp_nid(kcontrol); > + int chs = get_amp_channels(kcontrol); > + long *valp = ucontrol->value.integer.value; > + char vol = 0; > + > + codec_dbg(codec, "%s() nid: %d\n", __func__, nid); > + snd_hda_power_up(codec); > + switch (nid) { > + case CS8409_CS42L42_HP_PIN_NID: > + mutex_lock(&cs8409_i2c_mux); > + if (chs & 1) { > + vol = -(cs8409_i2c_read(codec, CS42L42_I2C_ADDR, > + CS8409_CS42L42_REG_HS_VOLUME_CHA, 1)); Better to cache the values instead of i2c read at each time? Then the unnecessary power up/down sequence can be avoided, too. thanks, Takashi

2 1

Re: [PATCH 3/4] ALSA: hda/cirrus: Add jack detect interrupt support from CS42L42 companion codec.
by Takashi Iwai 04 Mar '21

04 Mar '21

On Wed, 03 Mar 2021 19:29:58 +0100, Vitaly Rodionov wrote: > @@ -1243,6 +1258,8 @@ static int patch_cs4213(struct hda_codec *codec) > #define CIR_I2C_QWRITE 0x005D > #define CIR_I2C_QREAD 0x005E > > +static struct mutex cs8409_i2c_mux; Any reason that this must be the global mutex? I suppose it can fit in own spec->i2c_mutex instead? > +static void cs8409_cs42l42_cap_sync_hook(struct hda_codec *codec, > + struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct cs_spec *spec = codec->spec; > + unsigned int curval, expval; > + /* CS8409 DMIC Pin only allows the setting of the Stream Parameters in > + * Power State D0. When a headset is unplugged, and the path is switched to > + * the DMIC, the Stream is restarted with the new ADC, but this is done in > + * Power State D3. Restart the Stream now DMIC is in D0. > + */ > + if (spec->gen.cur_adc == CS8409_CS42L42_DMIC_ADC_PIN_NID) { > + curval = snd_hda_codec_read(codec, spec->gen.cur_adc, > + 0, AC_VERB_GET_CONV, 0); > + expval = (spec->gen.cur_adc_stream_tag << 4) | 0; > + if (curval != expval) { > + codec_dbg(codec, "%s Restarting Stream after DMIC switch\n", __func__); > + __snd_hda_codec_cleanup_stream(codec, spec->gen.cur_adc, 1); > + snd_hda_codec_setup_stream(codec, spec->gen.cur_adc, > + spec->gen.cur_adc_stream_tag, 0, > + spec->gen.cur_adc_format); Hrm, this looks a big scary. We may need to reconsider how to handle this better later, but it's OK as long as you've tested with this code... > +static int cs8409_cs42l42_init(struct hda_codec *codec) > +{ > + int ret = 0; > + > + ret = snd_hda_gen_init(codec); > + > + if (!ret) { > + /* On Dell platforms with suspend D3 mode support we > + * have to re-initialise cs8409 bridge and companion > + * cs42l42 codec > + */ > + snd_hda_sequence_write(codec, cs8409_cs42l42_init_verbs); > + snd_hda_sequence_write(codec, cs8409_cs42l42_add_verbs); > + > + cs8409_cs42l42_hw_init(codec); Ah... the init stuff at resume appears finally here. This part should be in the second patch instead. > +static int cs8409_cs42l42_exec_verb(struct hdac_device *dev, > + unsigned int cmd, unsigned int flags, unsigned int *res) > +{ > + struct hda_codec *codec = container_of(dev, struct hda_codec, core); > + struct cs_spec *spec = codec->spec; > + > + unsigned int nid = 0; > + unsigned int verb = 0; The blank line above should be removed. > + case CS8409_CS42L42_HP_PIN_NID: > + if (verb == AC_VERB_GET_PIN_SENSE) { > + *res = (spec->cs42l42_hp_jack_in)?AC_PINSENSE_PRESENCE:0; The spaces are needed around operators. Similar coding-style issues are seen other places. Please try to run scripts/checkpatch.pl. thanks, Takashi

2 1

Re: [PATCH 2/4] ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec.
by Takashi Iwai 04 Mar '21

04 Mar '21

On Wed, 03 Mar 2021 19:29:57 +0100, Vitaly Rodionov wrote: > > +static const struct hda_verb cs8409_cs42l42_init_verbs[] = { > + { 0x01, AC_VERB_SET_POWER_STATE, 0x0000 }, /* AFG: D0 */ I guess this power state change is superfluous. The AFG node is already powered up when the codec probe or init is called. > + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0020 }, /* GPIO 5 out, 3,4 in */ > + { 0x01, AC_VERB_SET_GPIO_DATA, 0x0000 }, /* GPIO data 0 */ > + { 0x01, AC_VERB_SET_GPIO_MASK, 0x003f }, /* Enable GPIO */ Those are handled in spec->gpio_dir, gpio->data and gpio->mask fields. > + { 0x01, AC_VERB_SET_GPIO_WAKE_MASK, 0x0018 }, /* WAKE from GPIO 3,4 */ > + { 0x47, AC_VERB_SET_PROC_STATE, 0x0001 }, /* Enable VPW processing */ > + { 0x47, AC_VERB_SET_COEF_INDEX, 0x0002 }, /* Configure GPIO 6,7 */ > + { 0x47, AC_VERB_SET_PROC_COEF, 0x0080 }, /* I2C mode */ > + { 0x47, AC_VERB_SET_COEF_INDEX, 0x005b }, /* Set I2C bus speed */ > + { 0x47, AC_VERB_SET_PROC_COEF, 0x0200 }, /* 100kHz I2C_STO = 2 */ Those remaining verbs are good in the init verbs. But I suppose they have to be applied at the resume as well? But... > +static int cs8409_cs42l42_fixup(struct hda_codec *codec) > +{ > + int err = 0; > + struct cs_spec *spec = codec->spec; > + unsigned int pincap = 0; > + > + /* Basic initial sequence for specific hw configuration */ > + snd_hda_sequence_write(codec, cs8409_cs42l42_init_verbs); ... it seems applied only at the fixup call at parsing? Ditto about cs8409_cs42l42_hw_init(codec). thanks, Takashi

2 1

[PATCH] ALSA: hda: ignore invalid NHLT table
by Mark Pearson 04 Mar '21

04 Mar '21

On some Lenovo systems if the microphone is disabled in the BIOS only the NHLT table header is created, with no data. This means the endpoints field is not correctly set to zero - leading to an unintialised variable and hence invalid descriptors are parsed leading to page faults. The Lenovo firmware team is addressing this, but adding a check preventing invalid tables being parsed is worthwhile. Tested on a Lenovo T14. Tested-by: Philipp Leskovitz <philipp.leskovitz(a)secunet.com> Reported-by: Philipp Leskovitz <philipp.leskovitz(a)secunet.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> --- sound/hda/intel-nhlt.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 059aaf04f..0889f2cc5 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -37,6 +37,11 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) if (!nhlt) return 0; + if (nhlt->header.length <= sizeof(struct acpi_table_header)) { + dev_warn(dev, "Invalid DMIC description table\n"); + return 0; + } + epnt = (struct nhlt_endpoint *)nhlt->desc; for (j = 0; j < nhlt->endpoint_count; j++) { -- 2.29.2

3 4

[PATCH] ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()
by Kai-Heng Feng 04 Mar '21

04 Mar '21

Rear audio on Lenovo ThinkStation P620 stops working after commit 1965c4364bdd ("ALSA: usb-audio: Disable autosuspend for Lenovo ThinkStation P620"): [ 6.013526] usbcore: registered new interface driver snd-usb-audio [ 6.023064] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1 [ 6.023083] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4 [ 6.023090] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1 [ 6.023098] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4 [ 6.023103] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1 [ 6.023110] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4 [ 6.045846] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1 [ 6.045866] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4 [ 6.045877] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1 [ 6.045886] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4 [ 6.045894] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1 [ 6.045908] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4 I overlooked the issue because when I was working on the said commit, only the front audio is tested. Apology for that. Changing supports_autosuspend in driver is too late for disabling autosuspend, because it was already used by USB probe routine, so it can break the balance on the following code that depends on supports_autosuspend. Fix it by using usb_disable_autosuspend() helper, and balance the suspend count in disconnect callback. Fixes: 1965c4364bdd ("ALSA: usb-audio: Disable autosuspend for Lenovo ThinkStation P620") Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/usb/card.c | 5 +++++ sound/usb/quirks.c | 2 +- sound/usb/usbaudio.h | 1 + 3 files changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 85ed8507e41a..08c794883299 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -830,6 +830,8 @@ static int usb_audio_probe(struct usb_interface *intf, snd_media_device_create(chip, intf); } + chip->quirk_type = quirk->type; + usb_chip[chip->index] = chip; chip->intf[chip->num_interfaces] = intf; chip->num_interfaces++; @@ -912,6 +914,9 @@ static void usb_audio_disconnect(struct usb_interface *intf) } else { mutex_unlock(&register_mutex); } + + if (chip->quirk_type & QUIRK_SETUP_DISABLE_AUTOSUSPEND) + usb_enable_autosuspend(interface_to_usbdev(intf)); } /* lock the shutdown (disconnect) task and autoresume */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 9ba4682ebc48..ef5ee899db26 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -547,7 +547,7 @@ static int setup_disable_autosuspend(struct snd_usb_audio *chip, struct usb_driver *driver, const struct snd_usb_audio_quirk *quirk) { - driver->supports_autosuspend = 0; + usb_disable_autosuspend(interface_to_usbdev(iface)); return 1; /* Continue with creating streams and mixer */ } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 215c1771dd57..60b9dd7df6bb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -27,6 +27,7 @@ struct snd_usb_audio { struct snd_card *card; struct usb_interface *intf[MAX_CARD_INTERFACES]; u32 usb_id; + uint16_t quirk_type; struct mutex mutex; unsigned int system_suspend; atomic_t active; -- 2.30.0

3 3

[PATCH v1 0/5] Add missing reset controls to NVIDIA Tegra ASoC drivers
by Dmitry Osipenko 04 Mar '21

04 Mar '21

Hi, This series adds missing hardware reset controls to I2S and AC97 drivers. Currently drivers happen to work properly because reset is implicitly deasserted by tegra-clk driver, but clk driver shouldn't touch the resets and we need to fix it because this breaks other Tegra drivers. Previously we fixed the resets of the AHUB and HDMI codec drivers, but turned out that we missed the I2C and AC97 drivers. Thanks to Paul Fertser for testing the pending clk patches and finding that audio got broken on Tegra20 AC100 netbook because of the missing I2S reset. Dmitry Osipenko (5): reset: Allow devm_reset_control_array_get() to get resets in a released state reset: Add devm_reset_control_array_get_exclusive_released() ASoC: tegra20: ac97: Add reset control ASoC: tegra20: i2s: Add reset control ASoC: tegra30: i2s: Add reset control drivers/reset/core.c | 8 ++++++-- include/linux/reset.h | 20 +++++++++++++------ sound/soc/tegra/tegra20_ac97.c | 21 ++++++++++++++++++++ sound/soc/tegra/tegra20_ac97.h | 1 + sound/soc/tegra/tegra20_i2s.c | 31 +++++++++++++++++++++++++++++ sound/soc/tegra/tegra20_i2s.h | 1 + sound/soc/tegra/tegra30_ahub.c | 14 ++++++++++--- sound/soc/tegra/tegra30_i2s.c | 36 +++++++++++++++++++++++++++++++++- sound/soc/tegra/tegra30_i2s.h | 1 + 9 files changed, 121 insertions(+), 12 deletions(-) -- 2.29.2

2 8

[PATCH] ALSA: usb-audio: Apply the control quirk to Plantronics headsets
by Takashi Iwai 04 Mar '21

04 Mar '21

Other Plantronics headset models seem requiring the same workaround as C320-M to add the 20ms delay for the control messages, too. Apply the workaround generically for devices with the vendor ID 0x047f. Note that the problem didn't surface before 5.11 just with luck. Since 5.11 got a big code rewrite about the stream handling, the parameter setup procedure has changed, and this seemed triggering the problem more often. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182552 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0864692d8a7b..84e5dd66a56b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1671,10 +1671,10 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, msleep(20); /* - * Plantronics C320-M needs a delay to avoid random - * microhpone failures. + * Plantronics headsets (C320, C320-M, etc) need a delay to avoid + * random microhpone failures. */ - if (chip->usb_id == USB_ID(0x047f, 0xc025) && + if (USB_ID_VENDOR(chip->usb_id) == 0x047f && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) msleep(20); -- 2.26.2

1 0

[PATCH] ALSA: usb-audio: Fix "cannot get freq eq" errors on Dell AE515 sound bar
by Takashi Iwai 04 Mar '21

04 Mar '21

Dell AE515 sound bar (413c:a506) spews the error messages when the driver tries to read the current sample frequency, hence it needs to be on the list in snd_usb_get_sample_rate_quirk(). BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211551 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d02dac5fcd40..58f1dff79e52 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1523,6 +1523,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x1901, 0x0191): /* GE B850V3 CP2114 audio interface */ case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */ + case USB_ID(0x413c, 0xa506): /* Dell AE515 sound bar */ return true; } -- 2.26.2

1 0

[PATCH] ALSA: hda: fix kernel-doc warnings
by Pierre-Louis Bossart 04 Mar '21

04 Mar '21

v5.12-rc1 flags new warnings with make W=1, fix missing or broken function descriptors. sound/pci/hda/hda_codec.c:3492: warning: expecting prototype for snd_hda_input_mux_info_info(). Prototype was for snd_hda_input_mux_info() instead sound/pci/hda/hda_codec.c:3521: warning: expecting prototype for snd_hda_input_mux_info_put(). Prototype was for snd_hda_input_mux_put() instead sound/pci/hda/hda_codec.c:3958: warning: expecting prototype for _snd_hda_pin_ctl(). Prototype was for _snd_hda_set_pin_ctl() instead sound/pci/hda/hda_jack.c:223: warning: expecting prototype for snd_hda_set_dirty_all(). Prototype was for snd_hda_jack_set_dirty_all() instead sound/pci/hda/hda_jack.c:309: warning: expecting prototype for snd_hda_jack_detect_enable_mst(). Prototype was for snd_hda_jack_detect_enable_callback_mst() instead sound/pci/hda/hda_generic.c:3933: warning: expecting prototype for snd_dha_gen_add_mute_led_cdev(). Prototype was for snd_hda_gen_add_mute_led_cdev() instead sound/pci/hda/hda_generic.c:4093: warning: expecting prototype for snd_dha_gen_add_micmute_led_cdev(). Prototype was for snd_hda_gen_add_micmute_led_cdev() instead sound/pci/hda/patch_ca0132.c:2357: warning: expecting prototype for Prepare and send the SCP message to DSP(). Prototype was for dspio_scp() instead sound/pci/hda/patch_ca0132.c:2883: warning: expecting prototype for Allocate router ports(). Prototype was for dsp_allocate_router_ports() instead sound/pci/hda/patch_ca0132.c:3202: warning: expecting prototype for Write a block of data into DSP code or data RAM using pre(). Prototype was for dspxfr_one_seg() instead sound/pci/hda/patch_ca0132.c:3397: warning: expecting prototype for data overlay to DSP memories(). Prototype was for dspxfr_image() instead sound/hda/hdac_regmap.c:393: warning: expecting prototype for snd_hdac_regmap_init(). Prototype was for snd_hdac_regmap_exit() instead sound/hda/ext/hdac_ext_controller.c:142: warning: expecting prototype for snd_hdac_ext_bus_get_link_index(). Prototype was for snd_hdac_ext_bus_get_link() instead sound/hda/ext/hdac_ext_stream.c:140: warning: expecting prototype for snd_hdac_ext_linkstream_start(). Prototype was for snd_hdac_ext_link_stream_start() instead Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/ext/hdac_ext_controller.c | 2 +- sound/hda/ext/hdac_ext_stream.c | 2 +- sound/hda/hdac_regmap.c | 2 +- sound/pci/hda/hda_codec.c | 6 +++--- sound/pci/hda/hda_generic.c | 4 ++-- sound/pci/hda/hda_jack.c | 4 ++-- sound/pci/hda/patch_ca0132.c | 9 ++++----- 7 files changed, 14 insertions(+), 15 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index a9bd39b93697..b2df7b4f9227 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -133,7 +133,7 @@ void snd_hdac_link_free_all(struct hdac_bus *bus) EXPORT_SYMBOL_GPL(snd_hdac_link_free_all); /** - * snd_hdac_ext_bus_get_link_index - get link based on codec name + * snd_hdac_ext_bus_get_link - get link based on codec name * @bus: the pointer to HDAC bus object * @codec_name: codec name */ diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index c4d54a838773..0c005d67fa89 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -133,7 +133,7 @@ void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_decouple); /** - * snd_hdac_ext_linkstream_start - start a stream + * snd_hdac_ext_link_stream_start - start a stream * @stream: HD-audio ext core stream to start */ void snd_hdac_ext_link_stream_start(struct hdac_ext_stream *stream) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index d75f31eb9d78..fe3587547cfe 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -386,7 +386,7 @@ int snd_hdac_regmap_init(struct hdac_device *codec) EXPORT_SYMBOL_GPL(snd_hdac_regmap_init); /** - * snd_hdac_regmap_init - Release the regmap from HDA codec + * snd_hdac_regmap_exit - Release the regmap from HDA codec * @codec: the codec object */ void snd_hdac_regmap_exit(struct hdac_device *codec) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9b755062d841..2026f1ccaf5a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3483,7 +3483,7 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power); */ /** - * snd_hda_input_mux_info_info - Info callback helper for the input-mux enum + * snd_hda_input_mux_info - Info callback helper for the input-mux enum * @imux: imux helper object * @uinfo: pointer to get/store the data */ @@ -3506,7 +3506,7 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux, EXPORT_SYMBOL_GPL(snd_hda_input_mux_info); /** - * snd_hda_input_mux_info_put - Put callback helper for the input-mux enum + * snd_hda_input_mux_put - Put callback helper for the input-mux enum * @codec: the HDA codec * @imux: imux helper object * @ucontrol: pointer to get/store the data @@ -3941,7 +3941,7 @@ unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_correct_pin_ctl); /** - * _snd_hda_pin_ctl - Helper to set pin ctl value + * _snd_hda_set_pin_ctl - Helper to set pin ctl value * @codec: the HDA codec * @pin: referred pin NID * @val: pin control value to set diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 5e40944e7342..8b7c5508f368 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3923,7 +3923,7 @@ static void vmaster_update_mute_led(void *private_data, int enabled) } /** - * snd_dha_gen_add_mute_led_cdev - Create a LED classdev and enable as vmaster mute LED + * snd_hda_gen_add_mute_led_cdev - Create a LED classdev and enable as vmaster mute LED * @codec: the HDA codec * @callback: the callback for LED classdev brightness_set_blocking */ @@ -4074,7 +4074,7 @@ static int add_micmute_led_hook(struct hda_codec *codec) } /** - * snd_dha_gen_add_micmute_led_cdev - Create a LED classdev and enable as mic-mute LED + * snd_hda_gen_add_micmute_led_cdev - Create a LED classdev and enable as mic-mute LED * @codec: the HDA codec * @callback: the callback for LED classdev brightness_set_blocking * diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index b8b568046592..ac00866d8032 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -213,7 +213,7 @@ static void jack_detect_update(struct hda_codec *codec, } /** - * snd_hda_set_dirty_all - Mark all the cached as dirty + * snd_hda_jack_set_dirty_all - Mark all the cached as dirty * @codec: the HDA codec * * This function sets the dirty flag to all entries of jack table. @@ -293,7 +293,7 @@ find_callback_from_list(struct hda_jack_tbl *jack, } /** - * snd_hda_jack_detect_enable_mst - enable the jack-detection + * snd_hda_jack_detect_enable_callback_mst - enable the jack-detection * @codec: the HDA codec * @nid: pin NID to enable * @func: callback function to register diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 7e62aed172a9..c966f49fa942 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2338,7 +2338,7 @@ static int dspio_send_scp_message(struct hda_codec *codec, } /** - * Prepare and send the SCP message to DSP + * dspio_scp - Prepare and send the SCP message to DSP * @codec: the HDA codec * @mod_id: ID of the DSP module to send the command * @src_id: ID of the source @@ -2865,7 +2865,7 @@ static int dsp_dma_stop(struct hda_codec *codec, } /** - * Allocate router ports + * dsp_allocate_router_ports - Allocate router ports * * @codec: the HDA codec * @num_chans: number of channels in the stream @@ -3178,8 +3178,7 @@ static int dspxfr_hci_write(struct hda_codec *codec, } /** - * Write a block of data into DSP code or data RAM using pre-allocated - * DMA engine. + * dspxfr_one_seg - Write a block of data into DSP code or data RAM using pre-allocated DMA engine. * * @codec: the HDA codec * @fls: pointer to a fast load image @@ -3376,7 +3375,7 @@ static int dspxfr_one_seg(struct hda_codec *codec, } /** - * Write the entire DSP image of a DSP code/data overlay to DSP memories + * dspxfr_image - Write the entire DSP image of a DSP code/data overlay to DSP memories * * @codec: the HDA codec * @fls_data: pointer to a fast load image -- 2.25.1

2 7

[PATCH] ASoC: SOF: Intel: hda: enable async suspend
by Ranjani Sridharan 04 Mar '21

04 Mar '21

The skylake driver disabled async suspend to prevent disabling the DSP before the card was suspended during system suspend. This code was carried over to the SOF driver. But, there is no risk of the DSP getting disabled before the card is suspended with the SOF driver. Therefore, it is safe to enable async suspend and thereby optimize the system resume time. Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 0dc3a8c0f5e3..25b799f0accc 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -614,8 +614,6 @@ static int hda_init_caps(struct snd_sof_dev *sdev) u32 link_mask; int ret = 0; - device_disable_async_suspend(bus->dev); - /* check if dsp is there */ if (bus->ppcap) dev_dbg(sdev->dev, "PP capability, will probe DSP later.\n"); -- 2.25.1

2 1

[PATCH v2] ASoC: codecs: nau8825: fix kernel-doc
by Pierre-Louis Bossart 04 Mar '21

04 Mar '21

v5.12-rc1 flags new warnings with make W=1, fix missing or broken function descriptors. sound/soc/codecs/nau8825.c:298: warning: wrong kernel-doc identifier on line: * Ramp up the headphone volume change gradually to target level. sound/soc/codecs/nau8825.c:358: warning: expecting prototype for This func(). Prototype was for nau8825_intlog10_dec3() instead sound/soc/codecs/nau8825.c:411: warning: wrong kernel-doc identifier on line: * computes cross talk suppression sidetone gain. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/nau8825.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index f0cba7b5758b..e19db30c457b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -295,7 +295,7 @@ static inline void nau8825_sema_reset(struct nau8825 *nau8825) } /** - * Ramp up the headphone volume change gradually to target level. + * nau8825_hpvol_ramp - Ramp up the headphone volume change gradually to target level. * * @nau8825: component to register the codec private data with * @vol_from: the volume to start up @@ -347,8 +347,9 @@ static void nau8825_hpvol_ramp(struct nau8825 *nau8825, } /** - * Computes log10 of a value; the result is round off to 3 decimal. This func- - * tion takes reference to dvb-math. The source code locates as the following. + * nau8825_intlog10_dec3 - Computes log10 of a value + * the result is round off to 3 decimal. This function takes reference to + * dvb-math. The source code locates as the following. * Linux/drivers/media/dvb-core/dvb_math.c * @value: input for log10 * @@ -408,7 +409,7 @@ static u32 nau8825_intlog10_dec3(u32 value) } /** - * computes cross talk suppression sidetone gain. + * nau8825_xtalk_sidetone - computes cross talk suppression sidetone gain. * * @sig_org: orignal signal level * @sig_cros: cross talk signal level -- 2.25.1

2 1

[PATCH v2] ASoC: Intel: atom: fix kernel-doc
by Pierre-Louis Bossart 04 Mar '21

04 Mar '21

v5.12-rc1 flags new warnings with make W=1, fix missing or broken function descriptors. sound/soc/intel/atom/sst/sst_loader.c:85: warning: expecting prototype for sst_start_merrifield(). Prototype was for sst_start_mrfld() instead sound/soc/intel/atom/sst/sst_acpi.c:339: warning: expecting prototype for intel_sst_remove(). Prototype was for sst_acpi_remove() instead Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/atom/sst/sst_acpi.c | 2 +- sound/soc/intel/atom/sst/sst_loader.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 2c1b8a2e3506..3be64430c256 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -328,7 +328,7 @@ static int sst_acpi_probe(struct platform_device *pdev) } /** -* intel_sst_remove - remove function +* sst_acpi_remove - remove function * * @pdev: platform device structure * diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index 1c9b0c9ec483..eea889001c24 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -76,7 +76,7 @@ int intel_sst_reset_dsp_mrfld(struct intel_sst_drv *sst_drv_ctx) } /** - * sst_start_merrifield - Start the SST DSP processor + * sst_start_mrfld - Start the SST DSP processor * @sst_drv_ctx: intel_sst_drv context pointer * * This starts the DSP in MERRIFIELD platfroms -- 2.25.1

2 1

[PATCH v2] ASoC: soc-dai: fix kernel-doc
by Pierre-Louis Bossart 04 Mar '21

04 Mar '21

v5.12-rc1 flags new warnings with make W=1, fix missing or broken function descriptors. sound/soc/soc-dai.c:167: warning: expecting prototype for snd_soc_xlate_tdm_slot(). Prototype was for snd_soc_xlate_tdm_slot_mask() instead Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/soc-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index cd3bb9a7983f..080fbe053fc5 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -154,7 +154,7 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** - * snd_soc_xlate_tdm_slot - generate tx/rx slot mask. + * snd_soc_xlate_tdm_slot_mask - generate tx/rx slot mask. * @slots: Number of slots in use. * @tx_mask: bitmask representing active TX slots. * @rx_mask: bitmask representing active RX slots. -- 2.25.1

2 1

[PATCH] ASoC: wm8524: Do not print probe defer error
by Fabio Estevam 04 Mar '21

04 Mar '21

On an imx8mq-evk the following error is seen: [ 1.375809] wm8524-codec audio-codec: Failed to get mute line: -517 It happens because the codec driver may probe prior to the imx gpio driver, which causes a probe defer. Change to dev_err_probe() to avoid printing this error. Reported-by: Chris Healy <cphealy(a)gmail.com> Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/codecs/wm8524.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index 4e9ab542f648..81f858f6bd67 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -227,7 +227,7 @@ static int wm8524_codec_probe(struct platform_device *pdev) wm8524->mute = devm_gpiod_get(&pdev->dev, "wlf,mute", GPIOD_OUT_LOW); if (IS_ERR(wm8524->mute)) { ret = PTR_ERR(wm8524->mute); - dev_err(&pdev->dev, "Failed to get mute line: %d\n", ret); + dev_err_probe(&pdev->dev, ret, "Failed to get mute line\n"); return ret; } -- 2.25.1

3 2

[PATCH 0/2] ASoC: Intel: soc-acpi: remove unused TigerLake configurations
by Pierre-Louis Bossart 04 Mar '21

04 Mar '21

Remove two machine descriptors which were added for development on Intel RVPs but are no longer used or productized. This will not have any impact on end-users. The corresponding cleanup was applied in the SOF topologies. Pierre-Louis Bossart (2): ASoC: Intel: soc-acpi: remove unused TGL table with rt5682 only ASoC: Intel: soc-acpi: remove TGL RVP mixed SoundWire/TDM config .../intel/common/soc-acpi-intel-tgl-match.c | 23 ------------------- 1 file changed, 23 deletions(-) -- 2.25.1

2 3

[PATCH] ASoC: SOF: Intel: unregister DMIC device on probe error
by Pierre-Louis Bossart 04 Mar '21

04 Mar '21

We only unregister the platform device during the .remove operation, but if the probe fails we will never reach this sequence. Suggested-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Fixes: dd96daca6c83e ("ASoC: SOF: Intel: Add APL/CNL HW DSP support") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 0e092877f550..5317dfa4a4bf 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -895,6 +895,7 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) /* dsp_unmap: not currently used */ iounmap(sdev->bar[HDA_DSP_BAR]); hdac_bus_unmap: + platform_device_unregister(hdev->dmic_dev); iounmap(bus->remap_addr); hda_codec_i915_exit(sdev); err: -- 2.25.1

2 1

[PATCH v2] ASoC: codec: Omit superfluous error message in jz4760_codec_probe()
by Tang Bin 04 Mar '21

04 Mar '21

The function devm_platform_ioremap_resource has already contained error message, so remove the redundant dev_err here. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- Changes from v1 - to streamline the code. --- sound/soc/codecs/jz4760.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/jz4760.c b/sound/soc/codecs/jz4760.c index e8f28ccc145a..5ae0e312bcfc 100644 --- a/sound/soc/codecs/jz4760.c +++ b/sound/soc/codecs/jz4760.c @@ -841,11 +841,8 @@ static int jz4760_codec_probe(struct platform_device *pdev) codec->dev = dev; codec->base = devm_platform_ioremap_resource(pdev, 0); - if (IS_ERR(codec->base)) { - ret = PTR_ERR(codec->base); - dev_err(dev, "Failed to ioremap mmio memory: %d\n", ret); - return ret; - } + if (IS_ERR(codec->base)) + return PTR_ERR(codec->base); codec->regmap = devm_regmap_init(dev, NULL, codec, &jz4760_codec_regmap_config); -- 2.18.2

3 2

[PATCH] ASoC: sgtl5000: Fix identation of .driver elements
by Fabio Estevam 04 Mar '21

04 Mar '21

The .driver elements are not correctly idented. Fix the identation of the .driver elements. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/codecs/sgtl5000.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 73551e36695e..34ac2e7fd3b5 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1819,9 +1819,9 @@ MODULE_DEVICE_TABLE(of, sgtl5000_dt_ids); static struct i2c_driver sgtl5000_i2c_driver = { .driver = { - .name = "sgtl5000", - .of_match_table = sgtl5000_dt_ids, - }, + .name = "sgtl5000", + .of_match_table = sgtl5000_dt_ids, + }, .probe = sgtl5000_i2c_probe, .remove = sgtl5000_i2c_remove, .id_table = sgtl5000_id, -- 2.25.1

2 1

[PATCH 0/9] ASoC: remove more make W=1 warnings
by Pierre-Louis Bossart 04 Mar '21

04 Mar '21

These warnings get in the way of automation/CI, let's remove them. Pierre-Louis Bossart (9): ASoC: cs4270: fix kernel-doc ASoC: jz4760: fix set but not used warning ASoC: rt5631: fix kernel-doc warning ASoC: sigmadsp-regmap: fix kernel-doc warning ASoC: amd: renoir: remove invalid kernel-doc comment ASoC: fsl: fsl_ssi: fix kernel-doc warning ASoC: fsl: fsl_easrc: fix kernel-doc warning ASoC: Intel: bytcr_wm5102: remove unused static variable ASoC: qcom: q6dsp: fix kernel-doc warning sound/soc/amd/renoir/rn-pci-acp3x.c | 2 +- sound/soc/codecs/cs4270.c | 1 + sound/soc/codecs/jz4760.c | 4 ++-- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/sigmadsp-regmap.c | 2 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 8 -------- sound/soc/qcom/qdsp6/q6afe.c | 2 +- 9 files changed, 9 insertions(+), 16 deletions(-) -- 2.25.1

2 10

[PATCH 00/24] ASoC: realtek: fix cppcheck warnings
by Pierre-Louis Bossart 04 Mar '21

04 Mar '21

Lots of trivial issues that make cppcheck too verbose. There shouldn't be any functionality change with this patchset. Pierre-Louis Bossart (24): ASoC: rt1011: use logical OR ASoC: rt1011: remove redundant test ASoC: rt1011: clarify expression ASoC: rt1015: clarify expression ASoC: rt1016: clarify expression ASoC: rt1305: clarify expression ASoC: rt1308: clarify expression ASoC: rt5640: clarify expression ASoC: rt5645: use logical OR ASoC: rt5645: clarify expression ASoC: rt5651: clarify expression ASoC: rt5651: remove useless assignment ASoC: rt5659: clarify expression ASoC: rt5660: clarify expression ASoC: rt5663: clarify expression ASoC: rt5665: clarify expression ASoC: rt5668: clarify expression ASoC: rt5668: remove useless assignments ASoC: rt5670: clarify expression ASoC: rt5677: clarify expression ASoC: rt5677: remove useless assignment ASoC: rt5682: clarify expression ASoC: rt5682: remove useless assignments ASoC: rt5682: remove useless initialization sound/soc/codecs/rt1011.c | 47 ++++++++++++++++++--------------------- sound/soc/codecs/rt1015.c | 5 +++-- sound/soc/codecs/rt1016.c | 7 +++--- sound/soc/codecs/rt1305.c | 4 ++-- sound/soc/codecs/rt1308.c | 8 +++---- sound/soc/codecs/rt5640.c | 6 ++--- sound/soc/codecs/rt5645.c | 6 ++--- sound/soc/codecs/rt5651.c | 6 ++--- sound/soc/codecs/rt5659.c | 4 ++-- sound/soc/codecs/rt5660.c | 4 ++-- sound/soc/codecs/rt5663.c | 4 ++-- sound/soc/codecs/rt5665.c | 4 ++-- sound/soc/codecs/rt5668.c | 8 +++---- sound/soc/codecs/rt5670.c | 4 ++-- sound/soc/codecs/rt5677.c | 6 ++--- sound/soc/codecs/rt5682.c | 12 +++++----- 16 files changed, 67 insertions(+), 68 deletions(-) -- 2.25.1

2 25

[PATCH -next] ASoC: rt1316: Fix return value check in rt1316_sdw_probe()
by Wei Yongjun 04 Mar '21

04 Mar '21

In case of error, the function devm_regmap_init_sdw() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Fixes: a262057df513 ("ASoC: rt1316: Add RT1316 SDCA vendor-specific driver") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/rt1316-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index a6fb34a48f33..3b029c56467d 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -669,8 +669,8 @@ static int rt1316_sdw_probe(struct sdw_slave *slave, /* Regmap Initialization */ regmap = devm_regmap_init_sdw(slave, &rt1316_sdw_regmap); - if (!regmap) - return -EINVAL; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); return rt1316_sdw_init(&slave->dev, regmap, slave); }

2 1

[PATCH] SoC: codecs: fix platform_no_drv_owner.cocci warnings
by Yang Li 04 Mar '21

04 Mar '21

./sound/soc/codecs/lpass-rx-macro.c:3588:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/codecs/lpass-rx-macro.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index c9c21d22..5b9d4e9 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -3585,7 +3585,6 @@ static int rx_macro_remove(struct platform_device *pdev) static struct platform_driver rx_macro_driver = { .driver = { .name = "rx_macro", - .owner = THIS_MODULE, .of_match_table = rx_macro_dt_match, .suppress_bind_attrs = true, }, -- 1.8.3.1

2 2

[PATCH][next] ASoC: fsl: fsl_easrc: Fix uninitialized variable st2_mem_alloc
by Colin King 04 Mar '21

04 Mar '21

From: Colin Ian King <colin.king(a)canonical.com> A previous cleanup commit removed the ininitialization of st2_mem_alloc. Fix this by restoring the original behaviour by initializing it to zero. Addresses-Coverity: ("Uninitialized scalar variable") Fixes: e80382fe721f ("ASoC: fsl: fsl_easrc: remove useless assignments") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/fsl/fsl_easrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 725a5d3aaa02..e823c9c13764 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -710,7 +710,7 @@ static int fsl_easrc_max_ch_for_slot(struct fsl_asrc_pair *ctx, struct fsl_easrc_slot *slot) { struct fsl_easrc_ctx_priv *ctx_priv = ctx->private; - int st1_mem_alloc = 0, st2_mem_alloc; + int st1_mem_alloc = 0, st2_mem_alloc = 0; int pf_mem_alloc = 0; int max_channels = 8 - slot->num_channel; int channels = 0; -- 2.30.0

2 1

conf: Remove non-existent SPDIF output on Dell AE515
by GitHub pull_request - opened 03 Mar '21

03 Mar '21

alsa-project/alsa-lib pull request #124 was opened from drjhe: The Dell Professional Sound Bar AE515 shows up in GNOME Settings with a digital output that doesn't physically exist. Add it to the list to suppress this. Request URL : https://github.com/alsa-project/alsa-lib/pull/124 Patch URL : https://github.com/alsa-project/alsa-lib/pull/124.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: usb: Add Plantronics C320-M USB ctrl msg delay quirk
by John Ernberg 03 Mar '21

03 Mar '21

The microphone in the Plantronics C320-M headset will randomly fail to initialize properly, at least when using Microsoft Teams. Introducing a 20ms delay on the control messages appears to resolve the issue. Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1065 Tested-by: Andreas Kempe <kempe(a)lysator.liu.se> Signed-off-by: John Ernberg <john.ernberg(a)actia.se> --- Note: Microsoft Teams is the only application tests have been performed with and no other kinds of investigations have been performed, like USB packet logging. Originally reported as a possible PulseAudio issue, where it was suggested to try a delay quirk in the kernel. If other tests should be performed, please let me know which tests and I will try to do them. All testing was done on 5.10, patch was then applied on sound.git master for submission I did not register to the mailing lists, please keep me CC on any comments. Thanks! sound/usb/quirks.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d02dac5fcd40..a31b7db92b1e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1673,6 +1673,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) msleep(20); + /* + * Plantronics C320-M needs a delay to avoid random + * microhpone failures. + */ + if (chip->usb_id == USB_ID(0x047f, 0xc025) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + msleep(20); + /* Zoom R16/24, many Logitech(at least H650e/H570e/BCC950), * Jabra 550a, Kingston HyperX needs a tiny delay here, * otherwise requests like get/set frequency return -- 2.30.0

2 1

[PATCH 0/4] ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec.
by Vitaly Rodionov 03 Mar '21

03 Mar '21

Dell's laptops Inspiron 3500, Inspiron 3501, Inspiron 3505 are using Cirrus Logic CS8409 HDA bridge with CS42L42 companion codec. The CS8409 is a multichannel HD audio routing controller. CS8409 includes support for four channels of digital microphone data and two bidirectional ASPs for up to 32 channels of TDM data or 4 channels of I2S data. The CS8409 is intended to be used with a remote companion codec that implements high performance analog functions in close physical proximity to the end-equipment audio port or speaker driver. The CS42L42 is a low-power audio codec with integrated MIPI SoundWire interface or I2C/I2S/TDM interfaces designed for portable applications. It provides a high-dynamic range, stereo DAC for audio playback and a mono high-dynamic-range ADC for audio capture Stefan Binding (1): ALSA: hda/cirrus: Add Headphone and Headset MIC Volume Control Vitaly Rodionov (3): ALSA: hda/cirrus: Increase AUTO_CFG_MAX_INS from 8 to 18 ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec. ALSA: hda/cirrus: Add jack detect interrupt support from CS42L42 companion codec. sound/pci/hda/hda_auto_parser.h | 2 +- sound/pci/hda/hda_local.h | 2 +- sound/pci/hda/patch_cirrus.c | 1033 +++++++++++++++++++++++++++++++ 3 files changed, 1035 insertions(+), 2 deletions(-) -- 2.25.1

1 4

[PATCH] ASoC: SOF: Intel: broadwell: fix mutual exclusion with catpt driver
by Pierre-Louis Bossart 03 Mar '21

03 Mar '21

In v5.10, the "haswell" driver was replaced by the "catpt" driver, but the mutual exclusion with the SOF driver was not updated. This leads to errors with card names and UCM profiles not being loaded by PulseAudio. This fix should only be applied on v5.10-stable, the mutual exclusion was removed in 5.11. Reported-by: David Ward <david.ward(a)ll.mit.edu> Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=211985 Fixes: 6cbfa11d2694 ("ASoC: Intel: Select catpt and deprecate haswell") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: <stable(a)kernel.org> --- sound/soc/sof/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index de7ff2d097ab..6708a2c5a838 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -84,7 +84,7 @@ config SND_SOC_SOF_BAYTRAIL config SND_SOC_SOF_BROADWELL_SUPPORT bool "SOF support for Broadwell" - depends on SND_SOC_INTEL_HASWELL=n + depends on SND_SOC_INTEL_CATPT=n help This adds support for Sound Open Firmware for Intel(R) platforms using the Broadwell processors. -- 2.25.1

3 2

[PATCH v2 0/5] soundwire: qcom: various improvements
by Srinivas Kandagatla 03 Mar '21

03 Mar '21

Thanks for reviewing v1 of this patchset! During testing SoundWire controller on SM8250 MTP, we found few issues like all the interrupts are not handled, all transport parameters are not read from device tree. Other major issue was register read/writes which was interrupt based was an overhead and puts lot of limitation on context it can be used from. With previous approach number of interrupts generated after enumeration are around 130: $ cat /proc/interrupts | grep soundwire 21: 130 0 0 0 0 0 0 0 GICv3 234 Edge soundwire after this patch they are just 3 interrupts $ cat /proc/interrupts | grep soundwire 21: 3 0 0 0 0 0 0 0 GICv3 234 Edge soundwire So this patchset add various improvements to the existing driver to address above issues. Tested it on SM8250 MTP with 2x WSA881x speakers, HeadPhones on WCD938x via lpass-rx-macro and Analog MICs via lpass-tx-macro. Also tested on DragonBoard DB845c with 2xWSA881x speakers. Changes since v1: - changed bgp_count to blk_group_count as suggested by Pierre - used version raw value as suggested by VKoul - updated read fifo logic and interrupt handling to do while() - renamed some of the variable to make reading easy! - removed patch that parses version Srinivas Kandagatla (5): soundwire: qcom: add support to missing transport params soundwire: qcom: set continue execution flag for ignored commands soundwire: qcom: start the clock during initialization soundwire: qcom: update register read/write routine soundwire: qcom: add support to new interrupts drivers/soundwire/qcom.c | 438 ++++++++++++++++++++++++++++++--------- 1 file changed, 335 insertions(+), 103 deletions(-) -- 2.21.0

2 10

[PATCH] ALSA: hda/realtek: Apply dual codec quirks for MSI Godlike X570 board
by Takashi Iwai 03 Mar '21

03 Mar '21

There is another MSI board (1462:cc34) that has dual Realtek codecs, and we need to apply the existing quirk for fixing the conflicts of Master control. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211743 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7c24d13181e1..b47504fa8dfd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2532,6 +2532,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), + SND_PCI_QUIRK(0x1462, 0xcc34, "MSI Godlike X570", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), -- 2.26.2

1 0

[PATCH] ALSA: hda - bind headset buttons to the headphone jack
by Hui Wang 03 Mar '21

03 Mar '21

With the HDA driver, if the headset buttons are supported, an audio Jack will be created for them. This audio Jack is a bit confusing to users since it can't report headphone/mic insertion events but it claims to support these events. And in addition, the driver already builds a headphone Jack and a mic Jack, and most of those buttons are used for headphone playback, so do some change to bind those buttons to the headphone Jack. After this change, the key events are generated from NID 0x55, and are reported to the input layer via headphone jack (NID 0x21). If there is no headphone Jack, then build an audio jack to support those buttons same as previously. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_jack.c | 43 ++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_jack.h | 5 ++++ sound/pci/hda/patch_realtek.c | 23 ++++++++++++++++--- 3 files changed, 67 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index b8b568046592..640b72b598e8 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -388,6 +388,39 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, } EXPORT_SYMBOL_GPL(snd_hda_jack_set_gating_jack); +/** + * snd_hda_jack_bind_keymap - bind keys generated from one NID to another jack. + * @codec: the HDA codec + * @key_nid: key event is generated by this pin NID + * @keymap: map of key type and key code + * @jack_nid: key reports to the jack of this pin NID + * + * This function is used in the case of key is generated from one NID while is + * reported to the jack of another NID. + */ +int snd_hda_jack_bind_keymap(struct hda_codec *codec, hda_nid_t key_nid, + const struct hda_jack_keymap *keymap, + hda_nid_t jack_nid) +{ + const struct hda_jack_keymap *map; + struct hda_jack_tbl *key_gen = snd_hda_jack_tbl_get(codec, key_nid); + struct hda_jack_tbl *report_to = snd_hda_jack_tbl_get(codec, jack_nid); + + WARN_ON(codec->dp_mst); + + if (!key_gen || !report_to || !report_to->jack) + return -EINVAL; + + key_gen->key_report_jack = jack_nid; + + if (keymap) + for (map = keymap; map->type; map++) + snd_jack_set_key(report_to->jack, map->type, map->key); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hda_jack_bind_keymap); + /** * snd_hda_jack_report_sync - sync the states of all jacks and report if changed * @codec: the HDA codec @@ -651,7 +684,15 @@ void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res) } if (!event) return; - event->jack_dirty = 1; + + if (event->key_report_jack) { + struct hda_jack_tbl *report_to = + snd_hda_jack_tbl_get_mst(codec, event->key_report_jack, + event->dev_id); + if (report_to) + report_to->jack_dirty = 1; + } else + event->jack_dirty = 1; call_jack_callback(codec, res, event); snd_hda_jack_report_sync(codec); diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 8ceaf0ef5df1..339746dd9578 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -40,6 +40,7 @@ struct hda_jack_tbl { unsigned int block_report:1; /* in a transitional state - do not report to userspace */ hda_nid_t gating_jack; /* valid when gating jack plugged */ hda_nid_t gated_jack; /* gated is dependent on this jack */ + hda_nid_t key_report_jack; /* key reports to this jack */ int type; int button_state; struct snd_jack *jack; @@ -99,6 +100,10 @@ snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid); +int snd_hda_jack_bind_keymap(struct hda_codec *codec, hda_nid_t key_nid, + const struct hda_jack_keymap *keymap, + hda_nid_t jack_nid); + u32 snd_hda_jack_pin_sense(struct hda_codec *codec, hda_nid_t nid, int dev_id); /* the jack state returned from snd_hda_jack_detect_state() */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 52907506e16e..f381474f9403 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3102,6 +3102,16 @@ static void alc_headset_btn_callback(struct hda_codec *codec, if (jack->unsol_res & (7 << 10)) report |= SND_JACK_BTN_3; + if (jack->jack->key_report_jack) { + struct hda_jack_tbl *report_to = + snd_hda_jack_tbl_get(codec, jack->jack->key_report_jack); + + if (report_to) { + report_to->button_state = report; + return; + } + } + jack->jack->button_state = report; } @@ -3163,16 +3173,23 @@ static void alc_fixup_headset_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: spec->has_hs_key = 1; snd_hda_jack_detect_enable_callback(codec, 0x55, alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, - SND_JACK_HEADSET, alc_headset_btn_keymap); break; - case HDA_FIXUP_ACT_INIT: + case HDA_FIXUP_ACT_BUILD: + hp_pin = alc_get_hp_pin(spec); + if (!hp_pin || snd_hda_jack_bind_keymap(codec, 0x55, + alc_headset_btn_keymap, + hp_pin)) + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", + false, SND_JACK_HEADSET, + alc_headset_btn_keymap); + alc_enable_headset_jack_key(codec); break; } -- 2.25.1

2 2

[PATCH] ASoC: soc-core: Prevent warning if no DMI table is present
by Jon Hunter 03 Mar '21

03 Mar '21

Many systems do not provide a DMI table and on these systems a warning, such as the following, is printed on boot ... WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name! If DMI support is enabled in the kernel, there is no simple way to detect if a DMI table is table or not. Note that the variable 'dmi_available' is not exported and so cannot be used by kernel modules. It could be possible to have every ASoC sound card driver set the long name to avoid the above message, but it might be intentional for the long name, that we fall back to using the sound card name. Therefore, make this a debug print by default to avoid the warning. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f6d4e99b590c..f1189e7c1fcc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1576,7 +1576,7 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) /* make up dmi long name as: vendor-product-version-board */ vendor = dmi_get_system_info(DMI_BOARD_VENDOR); if (!vendor || !is_dmi_valid(vendor)) { - dev_warn(card->dev, "ASoC: no DMI vendor name!\n"); + dev_dbg(card->dev, "ASoC: no DMI vendor name!\n"); return 0; } -- 2.25.1

3 3

[PATCH v5 1/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Mar '21

03 Mar '21

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. CPU0 CPU1 dpcm_be_dai_trigger(): dpcm_be_disconnect(): for_each_dpcm_be(fe, stream, dpcm) { spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); struct snd_soc_pcm_runtime *be = dpcm->be; <-- Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_be_dai_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- include/sound/soc-dpcm.h | 5 ++++ sound/soc/soc-pcm.c | 59 +++++++++++++++++++++++++++++++++------- 2 files changed, 54 insertions(+), 10 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 0f6c50b17bba..599cd6054bc3 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -103,6 +103,11 @@ struct snd_soc_dpcm_runtime { int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */ }; +struct snd_soc_dpcm_rtd_list { + int num_rtds; + struct snd_soc_pcm_runtime *rtds[0]; +}; + #define for_each_dpcm_fe(be, stream, _dpcm) \ list_for_each_entry(_dpcm, &(be)->dpcm[stream].fe_clients, list_fe) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 14d85ca1e435..ccda0f77e827 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2028,15 +2028,51 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_be_list_create(struct snd_soc_pcm_runtime *fe, int stream, + struct snd_soc_dpcm_rtd_list **be_list) +{ + struct snd_soc_dpcm *dpcm; + unsigned long flags; + int size, i, ret = 0; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + + size = 0; + for_each_dpcm_be(fe, stream, dpcm) + size++; + + *be_list = kzalloc(struct_size(*be_list, rtds, size), GFP_ATOMIC); + if (*be_list) { + i = 0; + for_each_dpcm_be(fe, stream, dpcm) + (*be_list)->rtds[i++] = dpcm->be; + + if (i != size) + dev_err(fe->dev, "ASoC: abnormal change in be clients: %d -> %d\n", + size, i); + + (*be_list)->num_rtds = i; + } else { + ret = -ENOMEM; + } + + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} + int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { - struct snd_soc_dpcm *dpcm; - int ret = 0; + struct snd_soc_dpcm_rtd_list *be_list; + int i, ret = 0; - for_each_dpcm_be(fe, stream, dpcm) { + ret = dpcm_be_list_create(fe, stream, &be_list); + if (ret < 0) + return ret; - struct snd_soc_pcm_runtime *be = dpcm->be; + for (i = 0; i < be_list->num_rtds; i++) { + struct snd_soc_pcm_runtime *be = be_list->rtds[i]; struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); @@ -2056,7 +2092,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2066,7 +2102,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2076,7 +2112,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2090,7 +2126,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2103,7 +2139,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2116,13 +2152,16 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret < 0) + break; } + kfree(be_list); return ret; } EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); -- 2.21.0

2 1

[PATCH 0/3] ALSA: digigram: fix kernel-doc warnings
by Pierre-Louis Bossart 03 Mar '21

03 Mar '21

make W=1 reports these warnings with v5.12-rc1. Pierre-Louis Bossart (3): ALSA: vx: fix kernel-doc warning ALSA: pci: mixart: fix kernel-doc warning ALSA: pci: vx222: fix kernel-doc warning sound/drivers/vx/vx_core.c | 2 +- sound/pci/mixart/mixart_hwdep.c | 3 ++- sound/pci/vx222/vx222_ops.c | 10 +++++----- 3 files changed, 8 insertions(+), 7 deletions(-) -- 2.25.1

2 4

[PATCH] ALSA: hda/realtek: Add quirk for Intel NUC 10
by Werner Sembach 03 Mar '21

03 Mar '21

ALSA: hda/realtek: Add quirk for Intel NUC 10 This adds a new SND_PCI_QUIRK(...) and applies it to the Intel NUC 10 devices. This fixes the issue of the devices not having audio input and output on the headset jack because the kernel does not recognize when something is plugged in. The new quirk was inspired by the quirk for the Intel NUC 8 devices, but it turned out that the NUC 10 uses another pin. This information was acquired by black box testing likely pins. Co-developed-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- Resend of this patch with git send-email, because last patch got tabs replaced with spaces. >From d281364b8ca6c054a0e5ce20caa599bf7d08160d Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Fri, 26 Feb 2021 13:54:30 +0100 Subject: [PATCH] Fix Intel NUC10 no output and input on headset jack --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..c14d624dbaf1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6362,6 +6362,7 @@ enum { ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, ALC256_FIXUP_INTEL_NUC8_RUGGED, + ALC256_FIXUP_INTEL_NUC10, ALC255_FIXUP_XIAOMI_HEADSET_MIC, ALC274_FIXUP_HP_MIC, ALC274_FIXUP_HP_HEADSET_MIC, @@ -7744,6 +7745,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC256_FIXUP_INTEL_NUC10] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, [ALC255_FIXUP_XIAOMI_HEADSET_MIC] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -8183,6 +8193,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), + SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), #if 0 /* Below is a quirk table taken from the old code. -- 2.25.1

2 1

[PATCH v5 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Mar '21

03 Mar '21

On Wed, 2021-02-17 16:29, Takashi Iwai wrote: >On Wed, 17 Feb 2021 05:31:49 +0100, >Gyeongtaek Lee wrote: >> >> If stop by underrun and DPCM BE disconnection is run simultaneously, >> data abort can be occurred by the sequence below. >> >> CPU0 CPU1 >> dpcm_be_dai_trigger(): dpcm_be_disconnect(): >> >> for_each_dpcm_be(fe, stream, dpcm) { >> >> spin_lock_irqsave(&fe->card->dpcm_lock, flags); >> list_del(&dpcm->list_be); >> list_del(&dpcm->list_fe); >> spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); >> kfree(dpcm); >> >> struct snd_soc_pcm_runtime *be = dpcm->be; <-- Accessing freed memory >> >> To prevent this situation, dpcm_lock should be acquired during >> iteration of dpcm list in dpcm_be_dai_trigger(). > >I don't think we can apply spin lock there blindly. There is >non-atomic PCM that must not take a spin lock there, too. Hi, Takashi I fixed the patch to hold dpcm_lock during accessing dpcm related data only, not on the trigger callback of driver. Thank you. Gyeongtaek Lee > > >thanks, > >Takashi > Gyeongtaek Lee (1): ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger() include/sound/soc-dpcm.h | 5 ++++ sound/soc/soc-pcm.c | 59 +++++++++++++++++++++++++++++++++------- 2 files changed, 54 insertions(+), 10 deletions(-) base-commit: fe07bfda2fb9cdef8a4d4008a409bb02f35f1bd8 -- 2.21.0

1 0

Re: [PATCH] ASoC: wm8960: Remove bitclk relax condition
by S.j. Wang 03 Mar '21

03 Mar '21

> > > > Using a higher bitclk then expected doesn't always work. > > Here is an example: > > > > aplay -Dhw:0,0 -d 5 -r 48000 -f S24_LE -c 2 audio48k24b2c.wav > > > > In this case, the required bitclk is 48000 * 24 * 2 = 2304000 but the > > closest bitclk that can be derived is 3072000. Since the clock is > > faster than expected, it will start to send bytes from the next > > channel so the sound will be corrupted. > > > > Fixes: 82bab88910ee ("ASoC: codec: wm8960: Relax bit clock computation > > when using PLL") > > Fixes: 3c01b9ee2ab9 ("ASoC: codec: wm8960: Relax bit clock > > computation") > > Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > I think this is probably going to need a much more involved fix. > The problem is that there are systems that depend on this behaviour, so you > can't just flat out revert it. And to be fair the I2S specification says that bit > clock can run at a higher rate than required for the data, so the behaviour is > correct as well. > > Probably the best solution here is to add additional contraints from the > machine driver on which rates/bit depths/channels are supported. > Just double check the issue, the real reason is in below: The call sequence in wm8960_configure_clocking is ret = wm8960_configure_sysclk(); if (ret >= 0) goto configure_clock; .... ret = wm8960_configure_pll(); configure_clock: ... wm8960_configure_sysclk is called before wm8960_configure_pll, as there is bitclk relax on both functions, so wm8960_configure_sysclk always return success, then wm8960_configure_pll() never be called. So bitclk relax condition should be removed in wm8960_configure_sysclk, That wm8960_configure_pll can be called, and there is also bitclk relax function in wm8960_configure_pll. I will update and resend this patch. Best regards Wang shengjiu

1 0

Fiio M3K: Failed to find a working profile
by GitHub issues - edited 03 Mar '21

03 Mar '21

alsa-project/alsa-lib issue #123 was edited from x1unix: Hello, I have a [Fiio M3K](https://fiio.com/m3k) player which I often use as an external USB DAC. It worked well with Pipewire, but I had some other issues with Pipewire so I had to rollback to PulseAudio. Unfortunately, PulseAudio and module-alsa-card cannot setup this device, so it doesn't work. Here is a small portion of PulseAudio log (full log is available at link below): ``` E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" card_name="alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 (alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00) failed to load module. D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC2 is accessible: yes D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 is busy: no ``` I'm using ArchLinux with Linux 5.11.2 and PulseAudio 14.2. **Logs and debug info** USB device in `lsusb` output: ``` Bus 005 Device 022: ID 2972:0044 FiiO Electronics Technology FiiO M3K ``` aplay, pulseaudio debug log and other info I've posted as gist: https://gist.github.com/x1unix/dae3fb810ac499cc63362a805220968f Issue URL : https://github.com/alsa-project/alsa-lib/issues/123 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Fiio M3K: Failed to find a working profile
by GitHub issues - edited 03 Mar '21

03 Mar '21

alsa-project/alsa-lib issue #123 was edited from x1unix: Hello, I have a [Fiio M3K](https://fiio.com/m3k) player which I often use as an external USB DAC. It worked well with Pipewire, but I had some other issues with Pipewire so I had to rollback to PulseAudio. Unfortunately, PulseAudio and module-alsa-card cannot setup this device, so it doesn't work. Here is a small portion of PulseAudio log (full log is available at link below): ``` E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" card_name="alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 (alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00) failed to load module. D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC2 is accessible: yes D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 is busy: no ``` I'm using ArchLinux with Linux 5.11.2 and PulseAudio 14.2. **Logs and debug info** aplay, pulseaudio debug log and other info I've posted as gist: https://gist.github.com/x1unix/dae3fb810ac499cc63362a805220968f Issue URL : https://github.com/alsa-project/alsa-lib/issues/123 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Fiio M3K: Failed to find a working profile
by GitHub issues - edited 03 Mar '21

03 Mar '21

alsa-project/alsa-lib issue #123 was edited from x1unix: Hello, I have a [Fiio M3K](https://fiio.com/m3k) player which I often use as an external USB DAC. It worked well with Pipewire, but I had some other issues with Pipewire so I had to rollback to PulseAudio. Unfortunately, PulseAudio and module-alsa-card cannot setup this device, so it doesn't work. Here is a small portion of PulseAudio log (full log is available at link below): ``` E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" card_name="alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 (alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00) failed to load module. D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC2 is accessible: yes D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 is busy: no ``` I'm using ArchLinux and Linux 5.11.2 and PulseAudio 14.2. **Logs and debug info** aplay, pulseaudio debug log and other info I've posted as gist: https://gist.github.com/x1unix/dae3fb810ac499cc63362a805220968f Issue URL : https://github.com/alsa-project/alsa-lib/issues/123 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Fiio M3K: Failed to find a working profile
by GitHub issues - opened 03 Mar '21

03 Mar '21

alsa-project/alsa-lib issue #123 was opened from x1unix: Hello, I have a [Fiio M3K](https://fiio.com/m3k) player which I often use as an external USB DAC. It worked well with Pipewire, but I had some other issues with Pipewire so I had to rollback to PulseAudio. Unfortunately, PulseAudio and module-alsa-card cannot setup this device, so it doesn't work. Here is a small portion of PulseAudio log (full log is available at link below): ``` E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" card_name="alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 (alsa_card.usb-GuangZhou_FiiO_Electronics_Technology_Co.__Ltd._FiiO_M3K_FiiO_M3K-00) failed to load module. D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC2 is accessible: yes D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/usb5/5-1/5-1:1.0/sound/card2 is busy: no ``` I'm using ArchLinux and Linux 5.11.2. **Logs and debug info** aplay, pulseaudio debug log and other info I've posted as gist: https://gist.github.com/x1unix/dae3fb810ac499cc63362a805220968f Issue URL : https://github.com/alsa-project/alsa-lib/issues/123 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 00/15] Report jack and button detection + Capture Support
by Lucas Tanure 03 Mar '21

03 Mar '21

Hi All, Here is a patch series for reporting to user space jack and button events and add the support for Capture. With some cleanups and fixes along the way. Regards, Lucas Tanure Lucas Tanure (12): ASoC: cs42l42: Don't enable/disable regulator at Bias Level ASoC: cs42l42: Always wait at least 3ms after reset ASoC: cs42l42: Remove power if the driver is being removed ASoC: cs42l42: Disable regulators if probe fails ASoC: cs42l42: Fix Bitclock polarity inversion ASoC: cs42l42: Provide finer control on playback path ASoC: cs42l42: Set clock source for both ways of stream ASoC: cs42l42: Fix channel width support ASoC: cs42l42: Add Capture Support ASoC: cs42l42: Report jack and button detection ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called ASoC: cs42l42: Fix mixer volume control Richard Fitzgerald (3): ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT ASoC: cs42l42: Only start PLL if it is needed ASoC: cs42l42: Wait for PLL to lock before switching to it sound/soc/codecs/cs42l42.c | 497 +++++++++++++++++++++---------------- sound/soc/codecs/cs42l42.h | 42 +++- 2 files changed, 317 insertions(+), 222 deletions(-) -- 2.30.1

3 24

[PATCH] ALSA: hda/realtek: Add quirk for Intel NUC 10
by Werner Sembach 02 Mar '21

02 Mar '21

ALSA: hda/realtek: Add quirk for Intel NUC 10 This adds a new SND_PCI_QUIRK(...) and applies it to the Intel NUC 10 devices. This fixes the issue of the devices not having audio input and output on the headset jack because the kernel does not recognize when something is plugged in. The new quirk was inspired by the quirk for the Intel NUC 8 devices, but it turned out that the NUC 10 uses another pin. This information was acquired by black box testing likely pins. Co-developed-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- Forgot to add the "From"-line >From d281364b8ca6c054a0e5ce20caa599bf7d08160d Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Fri, 26 Feb 2021 13:54:30 +0100 Subject: [PATCH] Fix Intel NUC10 no output and input on headset jack --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..c14d624dbaf1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6362,6 +6362,7 @@ enum { ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, ALC256_FIXUP_INTEL_NUC8_RUGGED, + ALC256_FIXUP_INTEL_NUC10, ALC255_FIXUP_XIAOMI_HEADSET_MIC, ALC274_FIXUP_HP_MIC, ALC274_FIXUP_HP_HEADSET_MIC, @@ -7744,6 +7745,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC256_FIXUP_INTEL_NUC10] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, [ALC255_FIXUP_XIAOMI_HEADSET_MIC] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -8183,6 +8193,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), + SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), #if 0 /* Below is a quirk table taken from the old code. -- 2.25.1

1 0

[PATCH] ALSA: hda/realtek: Add quirk for Clevo NH55RZQ
by Werner Sembach 02 Mar '21

02 Mar '21

From: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> ALSA: hda/realtek: Add quirk for Clevo NH55RZQ This applies a SND_PCI_QUIRK(...) to the Clevo NH55RZQ barebone. This fixes the issue of the device not recognizing a pluged in microphone. The device has both, a microphone only jack, and a speaker + microphone combo jack. The combo jack already works. The microphone-only jack does not recognize when a device is pluged in without this patch. Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Co-developed-by: Werner Sembach <wse(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> --- This is a resend of the patch because I missed that the editor I used to write the commit message replaced tabs with spaces. >From 2835edd753fd19c72a644dccb7e941cfc0ecdf8e Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Fri, 26 Feb 2021 13:50:15 +0100 Subject: [PATCH] Fix device detection on microphone jack of Clevo NH55RZQ --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..8014e80d72c3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8089,6 +8089,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8551, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[5|7][0-9]RZ[Q]", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.25.1

2 2

[PATCH] ALSA: hda/realtek: Add quirk for Clevo NH55RZQ
by Werner Sembach 02 Mar '21

02 Mar '21

From: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> ALSA: hda/realtek: Add quirk for Clevo NH55RZQ This applies a SND_PCI_QUIRK(...) to the Clevo NH55RZQ barebone. This fixes the issue of the device not recognizing a pluged in microphone. The device has both, a microphone only jack, and a speaker + microphone combo jack. The combo jack already works. The microphone-only jack does not recognize when a device is pluged in without this patch. Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Co-developed-by: Werner Sembach <wse(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- Third time's the charm, now using git send-email, I'm really sorry for the spam. >From 2835edd753fd19c72a644dccb7e941cfc0ecdf8e Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Fri, 26 Feb 2021 13:50:15 +0100 Subject: [PATCH] Fix device detection on microphone jack of Clevo NH55RZQ --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..8014e80d72c3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8089,6 +8089,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8551, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[5|7][0-9]RZ[Q]", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.25.1

1 0

[PATCH v3] ALSA: hda/hdmi: let new platforms assign the pcm slot dynamically
by Hui Wang 02 Mar '21

02 Mar '21

If the platform set the dyn_pcm_assign to true, it will call hdmi_find_pcm_slot() to find a pcm slot when hdmi/dp monitor is connected and need to create a pcm. So far only intel_hsw_common_init() and patch_nvhdmi() set the dyn_pcm_assign to true, here we let tgl platforms assign the pcm slot dynamically first, if the driver runs for a period of time and there is no regression reported, we could set no_fixed_assgin to true in the intel_hsw_common_init(), and then set it to true in the patch_nvhdmi(). This change comes from the discussion between Takashi and Kai Vehmanen. Please refer to: https://github.com/alsa-project/alsa-lib/pull/118 Suggested-and-reviewed-by: Takashi Iwai <tiwai(a)suse.de> Suggested-and-reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e405be7929e3..e6d0843ee9df 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -157,6 +157,7 @@ struct hdmi_spec { bool dyn_pin_out; bool dyn_pcm_assign; + bool dyn_pcm_no_legacy; bool intel_hsw_fixup; /* apply Intel platform-specific fixups */ /* * Non-generic VIA/NVIDIA specific @@ -1345,6 +1346,12 @@ static int hdmi_find_pcm_slot(struct hdmi_spec *spec, { int i; + /* on the new machines, try to assign the pcm slot dynamically, + * not use the preferred fixed map (legacy way) anymore. + */ + if (spec->dyn_pcm_no_legacy) + goto last_try; + /* * generic_hdmi_build_pcms() may allocate extra PCMs on some * platforms (with maximum of 'num_nids + dev_num - 1') @@ -1374,6 +1381,7 @@ static int hdmi_find_pcm_slot(struct hdmi_spec *spec, return i; } + last_try: /* the last try; check the empty slots in pins */ for (i = 0; i < spec->num_nids; i++) { if (!test_bit(i, &spec->pcm_bitmap)) @@ -2987,8 +2995,16 @@ static int patch_i915_tgl_hdmi(struct hda_codec *codec) * the index indicate the port number. */ static const int map[] = {0x4, 0x6, 0x8, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf}; + int ret; - return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); + ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); + if (!ret) { + struct hdmi_spec *spec = codec->spec; + + spec->dyn_pcm_no_legacy = true; + } + + return ret; } /* Intel Baytrail and Braswell; with eld notifier */ -- 2.25.1

2 1

[PATCH] ALSA: hda/realtek: Add quirk for Clevo NH55RZQ
by Werner Sembach 02 Mar '21

02 Mar '21

From: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> ALSA: hda/realtek: Add quirk for Clevo NH55RZQ This applies a SND_PCI_QUIRK(...) to the Clevo NH55RZQ barebone. This fixes the issue of the device not recognizing a pluged in microphone. The device has both, a microphone only jack, and a speaker + microphone combo jack. The combo jack already works. The microphone-only jack does not recognize when a device is pluged in without this patch. Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Co-developed-by: Werner Sembach <wse(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> --- Hi, this is my first ever submitted kernel patch, feel free to criticise me if I made an error or missed a best practise bullet point. Also: I'm unsure if this would fit the requirements for stable(a)vger.kernel.org and/or trivial(a)kernel.org, but I think not (correct me if I'm wrong). Kind regards Werner Sembach >From 2835edd753fd19c72a644dccb7e941cfc0ecdf8e Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Fri, 26 Feb 2021 13:50:15 +0100 Subject: [PATCH] Fix device detection on microphone jack of Clevo NH55RZQ --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..8014e80d72c3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8089,6 +8089,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8551, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[5|7][0-9]RZ[Q]", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.25.1

2 2

[PATCH] ASoC: simple-card: Add dummy dai support simple sound card
by Shengjiu Wang 02 Mar '21

02 Mar '21

If sound card doesn't need specific codec device, just dummy codec is enough, then we can link the dummy component directly. In this case, user needs to specify below setting in devicetree. Previously the sound-dai is a node of codec, now we check if it is zero before parsing the node, zero means dummy component is specified. simple-audio-card,codec { sound-dai = <0>; }; Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/generic/simple-card.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index c262281bc64d..8b0cabd1bad1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -34,10 +34,19 @@ static int asoc_simple_parse_dai(struct device_node *node, { struct of_phandle_args args; int ret; + u32 val; if (!node) return 0; + ret = of_property_read_u32(node, DAI, &val); + if (val == 0) { + dlc->of_node = NULL; + dlc->dai_name = "snd-soc-dummy-dai"; + dlc->name = "snd-soc-dummy"; + return 0; + } + /* * Get node via "sound-dai = <&phandle port>" * it will be used as xxx_of_node on soc_bind_dai_link() -- 2.27.0

3 3

[RFC PATCH 0/5] Flexible sysclk/pll configuration
by Sameer Pujar 02 Mar '21

02 Mar '21

This series attempts to add flexible system clock (sysclk) and pll configuration for the DAI controller from simple card or audio graph card. Sysclk configuration ==================== Motivation: ----------- The problem this tries to address is, some Codecs may have multiple clocks and multiple sources for its sysclk. For example sysclk may depend on MCLK provided by SoC or it may used an internal pll. The simple card or audio graph card driver is generic and can be used on multiple platforms. So if a platform using this driver needs a generic configuration, then the Codec specific stuff cannot be hardcoded in the driver. Hence it would be better if the info is driven from DT, especially from a Codec DAI or endpoint subnode. Solution: --------- "system-clock-id" and "system-clock-source" DT properties can be exposed for a flexible sysclk configuration. These properties go into a Codec DAI or endpoint subnode. Please note that other sysclk related properties like "system-clock-frequency" and "system-clock-direction-out" are already present. Currently snd_soc_dai_set_sysclk() does not have an argument for source. This series adds this additional field and updates all the required drivers. This is needed because ideally sysclk "clk_id" and "source" fields are different. Some drivers are making use of "clk_id" field to program source information. **May be this needs to be corrected going ahead** Pll configuration ================= Motivation: ----------- Similar to sysclk configuration, pll configuration also needs little more flexibility. If a Codec has internal pll and it can be sourced from multiple clocks a specific configuration may be required for a given platform. Some of the cases are: - MCLK (supplied by SoC) --> Codec pll -> Codec sysclk - SoC I2S bit clock (BCLK) --> Codec pll -> Codec sysclk Solution: --------- Approach is similar to what is done for extending sysclk flexibility. Following DT properties are added to address the same. "pll-id" "pll-source" "pll-input-reference" "pll-output-reference" "pll-input-frequency" "pll-output-frequency" All these are optional properties. Simple card or audio graph card drivers can use above info and call snd_soc_dai_set_pll() for necessary configuration. Sameer Pujar (5): ASoC: soc-component: Fix return value of snd_soc_component_set_pll() ASoC: soc-dai: Add sysclk source to snd_soc_dai_set_sysclk() ASoC: audio-graph-card: Add bindings for sysclk and pll ASoC: simple-card-utils: Parse sysclk id and source ASoC: simple-card-utils: Support pll configuration .../bindings/sound/audio-graph-port.yaml | 71 +++++++++ include/sound/simple_card_utils.h | 35 +++++ include/sound/soc-dai.h | 4 +- sound/soc/amd/acp-da7219-max98357a.c | 2 +- sound/soc/amd/acp-rt5645.c | 2 +- sound/soc/amd/acp3x-rt5682-max9836.c | 4 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 2 +- sound/soc/atmel/mikroe-proto.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-i2s.c | 2 +- sound/soc/cirrus/snappercl15.c | 4 +- sound/soc/codecs/88pm860x-codec.c | 2 +- sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adau17x1.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4613.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/arizona.c | 2 +- sound/soc/codecs/cpcap.c | 4 +- sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 2 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs4265.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/cs42l42.c | 2 +- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/cs42l56.c | 2 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/cs42xx8.c | 2 +- sound/soc/codecs/cs43130.c | 2 +- sound/soc/codecs/cs53l30.c | 2 +- sound/soc/codecs/cx2072x.c | 2 +- sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/da7218.c | 2 +- sound/soc/codecs/da7219.c | 2 +- sound/soc/codecs/da732x.c | 2 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/es7134.c | 2 +- sound/soc/codecs/es7241.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/es8328.c | 2 +- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/madera.c | 3 +- sound/soc/codecs/max98088.c | 3 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max98095.c | 2 +- sound/soc/codecs/max98390.c | 2 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/max98925.c | 2 +- sound/soc/codecs/max98927.c | 2 +- sound/soc/codecs/mc13783.c | 6 +- sound/soc/codecs/ml26124.c | 2 +- sound/soc/codecs/nau8810.c | 2 +- sound/soc/codecs/nau8822.c | 2 +- sound/soc/codecs/pcm186x.c | 2 +- sound/soc/codecs/pcm3060.c | 2 +- sound/soc/codecs/pcm3168a.c | 2 +- sound/soc/codecs/rt1011.c | 2 +- sound/soc/codecs/rt1305.c | 2 +- sound/soc/codecs/rt274.c | 2 +- sound/soc/codecs/rt286.c | 2 +- sound/soc/codecs/rt298.c | 2 +- sound/soc/codecs/rt5514.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5660.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/sta350.c | 2 +- sound/soc/codecs/sti-sas.c | 2 +- sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/tscs454.c | 2 +- sound/soc/codecs/twl4030.c | 4 +- sound/soc/codecs/twl6040.c | 2 +- sound/soc/codecs/uda1334.c | 2 +- sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8524.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/fsl/eukrea-tlv320.c | 4 +- sound/soc/fsl/fsl-asoc-card.c | 8 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_micfil.c | 2 +- sound/soc/fsl/fsl_sai.c | 8 +- sound/soc/fsl/imx-audmix.c | 2 +- sound/soc/fsl/imx-hdmi.c | 2 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/generic/simple-card-utils.c | 173 +++++++++++++++++++-- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 4 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 4 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 4 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- sound/soc/intel/boards/sof_realtek_common.c | 2 +- sound/soc/intel/boards/sof_rt5682.c | 4 +- sound/soc/intel/boards/sof_sdw_rt1308.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 2 +- sound/soc/jz4740/jz4740-i2s.c | 2 +- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 4 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 4 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 8 +- sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 2 +- sound/soc/mediatek/mt8183/mt8183-dai-tdm.c | 2 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 6 +- sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 2 +- sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 2 +- .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 8 +- sound/soc/meson/aiu-encoder-i2s.c | 2 +- sound/soc/meson/axg-tdm-interface.c | 2 +- sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/mxs/mxs-saif.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 4 +- sound/soc/pxa/brownstone.c | 4 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/imote2.c | 4 +- sound/soc/pxa/magician.c | 4 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/z2.c | 4 +- sound/soc/pxa/zylonite.c | 2 +- sound/soc/qcom/apq8096.c | 2 +- sound/soc/qcom/lpass-cpu.c | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- sound/soc/qcom/sc7180.c | 6 +- sound/soc/qcom/sdm845.c | 22 +-- sound/soc/qcom/sm8250.c | 2 +- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 +- sound/soc/rockchip/rk3399_gru_sound.c | 14 +- sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_max98090.c | 4 +- sound/soc/rockchip/rockchip_rt5645.c | 4 +- sound/soc/samsung/aries_wm8994.c | 6 +- sound/soc/samsung/arndale.c | 8 +- sound/soc/samsung/bells.c | 6 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 6 +- sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/littlemill.c | 14 +- sound/soc/samsung/midas_wm1811.c | 10 +- sound/soc/samsung/neo1973_wm8753.c | 4 +- sound/soc/samsung/odroid.c | 2 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 4 +- sound/soc/samsung/s3c24xx_uda134x.c | 4 +- sound/soc/samsung/smartq_wm8987.c | 6 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 4 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/spdif.c | 2 +- sound/soc/samsung/speyside.c | 8 +- sound/soc/samsung/tm2_wm5110.c | 6 +- sound/soc/samsung/tobermory.c | 6 +- sound/soc/sh/migor.c | 4 +- sound/soc/sh/siu_dai.c | 2 +- sound/soc/sh/ssi.c | 2 +- sound/soc/soc-component.c | 2 +- sound/soc/soc-dai.c | 9 +- sound/soc/sti/uniperif_player.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/stm/stm32_i2s.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 2 +- sound/soc/tegra/trimslice.c | 2 +- sound/soc/ti/davinci-evm.c | 4 +- sound/soc/ti/davinci-mcasp.c | 2 +- sound/soc/ti/j721e-evm.c | 8 +- sound/soc/ti/n810.c | 2 +- sound/soc/ti/omap-abe-twl6040.c | 6 +- sound/soc/ti/omap-dmic.c | 2 +- sound/soc/ti/omap-mcbsp.c | 2 +- sound/soc/ti/omap3pandora.c | 4 +- sound/soc/ti/rx51.c | 2 +- sound/soc/uniphier/aio-cpu.c | 4 +- sound/soc/ux500/ux500_msp_dai.c | 2 +- 271 files changed, 639 insertions(+), 389 deletions(-) -- 2.7.4

3 9

[PATCH v6 0/9] ALSA: add virtio sound driver
by Anton Yakovlev 02 Mar '21

02 Mar '21

This series implements a driver part of the virtio sound device specification v8 [1]. The driver supports PCM playback and capture substreams, jack and channel map controls. A message-based transport is used to write/read PCM frames to/from a device. As a device part was used OpenSynergy proprietary implementation. v6 changes: - Code readability improvements suggested by Takashi. - Added additional comments about PCM message lifetime. - Temporarily removed DEVICE_NEEDS_RESET device status handling. Some kind of assistance from the virtio bus/level is required. In turn, this requires a separate discussion of the possibility of implementing such handling for virtio devices whose configuration may change (which may require re-probing, like in case of the sound device). [1] https://lists.oasis-open.org/archives/virtio-dev/202003/msg00185.html Anton Yakovlev (9): uapi: virtio_ids: add a sound device type ID from OASIS spec ALSA: virtio: add virtio sound driver ALSA: virtio: handling control messages ALSA: virtio: build PCM devices and substream hardware descriptors ALSA: virtio: handling control and I/O messages for the PCM device ALSA: virtio: PCM substream operators ALSA: virtio: introduce jack support ALSA: virtio: introduce PCM channel map support ALSA: virtio: introduce device suspend/resume support MAINTAINERS | 9 + include/uapi/linux/virtio_ids.h | 1 + include/uapi/linux/virtio_snd.h | 334 +++++++++++++++++++++ sound/Kconfig | 2 + sound/Makefile | 3 +- sound/virtio/Kconfig | 10 + sound/virtio/Makefile | 13 + sound/virtio/virtio_card.c | 410 ++++++++++++++++++++++++++ sound/virtio/virtio_card.h | 111 +++++++ sound/virtio/virtio_chmap.c | 219 ++++++++++++++ sound/virtio/virtio_ctl_msg.c | 310 ++++++++++++++++++++ sound/virtio/virtio_ctl_msg.h | 78 +++++ sound/virtio/virtio_jack.c | 233 +++++++++++++++ sound/virtio/virtio_pcm.c | 498 ++++++++++++++++++++++++++++++++ sound/virtio/virtio_pcm.h | 118 ++++++++ sound/virtio/virtio_pcm_msg.c | 417 ++++++++++++++++++++++++++ sound/virtio/virtio_pcm_ops.c | 478 ++++++++++++++++++++++++++++++ 17 files changed, 3243 insertions(+), 1 deletion(-) create mode 100644 include/uapi/linux/virtio_snd.h create mode 100644 sound/virtio/Kconfig create mode 100644 sound/virtio/Makefile create mode 100644 sound/virtio/virtio_card.c create mode 100644 sound/virtio/virtio_card.h create mode 100644 sound/virtio/virtio_chmap.c create mode 100644 sound/virtio/virtio_ctl_msg.c create mode 100644 sound/virtio/virtio_ctl_msg.h create mode 100644 sound/virtio/virtio_jack.c create mode 100644 sound/virtio/virtio_pcm.c create mode 100644 sound/virtio/virtio_pcm.h create mode 100644 sound/virtio/virtio_pcm_msg.c create mode 100644 sound/virtio/virtio_pcm_ops.c -- 2.30.1

2 31

[PATCH] ALSA: hda: ignore invalid NHLT table
by Mark Pearson 02 Mar '21

02 Mar '21

On some Lenovo systems if the microphone is disabled in the BIOS only the NHLT table header is created, with no data. This means the endpoints field is not correctly set to zero - leading to an unintialised variable and hence invalid descriptors are parsed leading to page faults. The Lenovo firmware team is addressing this, but adding a check preventing invalid tables being parsed is worthwhile. Tested on a Lenovo T14. Tested-by: Philipp Leskovitz <philipp.leskovitz(a)secunet.com> Reported-by: Philipp Leskovitz <philipp.leskovitz(a)secunet.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> --- sound/hda/intel-nhlt.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 059aaf04f..0889f2cc5 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -37,6 +37,11 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) if (!nhlt) return 0; + if (nhlt->header.length <= sizeof(struct acpi_table_header)) { + dev_warn(dev, "Invalid DMIC description table\n"); + return 0; + } + epnt = (struct nhlt_endpoint *)nhlt->desc; for (j = 0; j < nhlt->endpoint_count; j++) { -- 2.29.2

1 0

[PATCH] ASoC: codec: Omit superfluous error message in jz4760_codec_probe()
by Tang Bin 02 Mar '21

02 Mar '21

The function devm_platform_ioremap_resource has already contained error message, so remove the redundant dev_err here. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/codecs/jz4760.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/jz4760.c b/sound/soc/codecs/jz4760.c index e8f28ccc145a..c2d8a107f159 100644 --- a/sound/soc/codecs/jz4760.c +++ b/sound/soc/codecs/jz4760.c @@ -843,7 +843,6 @@ static int jz4760_codec_probe(struct platform_device *pdev) codec->base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(codec->base)) { ret = PTR_ERR(codec->base); - dev_err(dev, "Failed to ioremap mmio memory: %d\n", ret); return ret; } -- 2.18.2

4 3

[PATCH] ASoC: wm8960: Remove bitclk relax condition
by Shengjiu Wang 02 Mar '21

02 Mar '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> Using a higher bitclk then expected doesn't always work. Here is an example: aplay -Dhw:0,0 -d 5 -r 48000 -f S24_LE -c 2 audio48k24b2c.wav In this case, the required bitclk is 48000 * 24 * 2 = 2304000 but the closest bitclk that can be derived is 3072000. Since the clock is faster than expected, it will start to send bytes from the next channel so the sound will be corrupted. Fixes: 82bab88910ee ("ASoC: codec: wm8960: Relax bit clock computation when using PLL") Fixes: 3c01b9ee2ab9 ("ASoC: codec: wm8960: Relax bit clock computation") Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8960.c | 29 +++-------------------------- 1 file changed, 3 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index df351519a3a6..368bec1dfbdb 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -608,10 +608,6 @@ static const int bclk_divs[] = { * - lrclk = sysclk / dac_divs * - 10 * bclk = sysclk / bclk_divs * - * If we cannot find an exact match for (sysclk, lrclk, bclk) - * triplet, we relax the bclk such that bclk is chosen as the - * closest available frequency greater than expected bclk. - * * @wm8960: codec private data * @mclk: MCLK used to derive sysclk * @sysclk_idx: sysclk_divs index for found sysclk @@ -629,7 +625,7 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, { int sysclk, bclk, lrclk; int i, j, k; - int diff, closest = mclk; + int diff; /* marker for no match */ *bclk_idx = -1; @@ -653,12 +649,6 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, *bclk_idx = k; break; } - if (diff > 0 && closest > diff) { - *sysclk_idx = i; - *dac_idx = j; - *bclk_idx = k; - closest = diff; - } } if (k != ARRAY_SIZE(bclk_divs)) break; @@ -676,10 +666,6 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, * - freq_out = sysclk * sysclk_divs * - 10 * sysclk = bclk * bclk_divs * - * If we cannot find an exact match for (sysclk, lrclk, bclk) - * triplet, we relax the bclk such that bclk is chosen as the - * closest available frequency greater than expected bclk. - * * @component: component structure * @freq_in: input frequency used to derive freq out via PLL * @sysclk_idx: sysclk_divs index for found sysclk @@ -697,14 +683,12 @@ int wm8960_configure_pll(struct snd_soc_component *component, int freq_in, { struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component); int sysclk, bclk, lrclk, freq_out; - int diff, closest, best_freq_out; + int diff; int i, j, k; bclk = wm8960->bclk; lrclk = wm8960->lrclk; - closest = freq_in; - best_freq_out = -EINVAL; *sysclk_idx = *dac_idx = *bclk_idx = -1; for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { @@ -725,18 +709,11 @@ int wm8960_configure_pll(struct snd_soc_component *component, int freq_in, *bclk_idx = k; return freq_out; } - if (diff > 0 && closest > diff) { - *sysclk_idx = i; - *dac_idx = j; - *bclk_idx = k; - closest = diff; - best_freq_out = freq_out; - } } } } - return best_freq_out; + return -EINVAL; } static int wm8960_configure_clocking(struct snd_soc_component *component) { -- 2.27.0

2 1

[PATCH] ALSA: hda/realtek: Add quirk for Intel NUC 10
by Werner Sembach 02 Mar '21

02 Mar '21

From: Werner Sembach <wse(a)tuxedocomputers.com> ALSA: hda/realtek: Add quirk for Intel NUC 10 This adds a new SND_PCI_QUIRK(...) and applies it to the Intel NUC 10 devices. This fixes the issue of the devices not having audio input and output on the headset jack because the kernel does not recognize when something is plugged in. The new quirk was inspired by the quirk for the Intel NUC 8 devices, but it turned out that the NUC 10 uses another pin. This information was acquired by black box testing likely pins. Co-developed-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> --- Hi, this is my second ever submitted kernel patch with the first one send just some minutes ago, feel free to criticise me if I made an error or missed a best practice bullet point. Also: I'm unsure if this would fit the requirements for stable(a)vger.kernel.org and/or trivial(a)kernel.org, but I think not (correct me if I'm wrong). Kind regards Werner Sembach >From d281364b8ca6c054a0e5ce20caa599bf7d08160d Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Fri, 26 Feb 2021 13:54:30 +0100 Subject: [PATCH] Fix Intel NUC10 no output and input on headset jack --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..c14d624dbaf1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6362,6 +6362,7 @@ enum { ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, ALC256_FIXUP_INTEL_NUC8_RUGGED, + ALC256_FIXUP_INTEL_NUC10, ALC255_FIXUP_XIAOMI_HEADSET_MIC, ALC274_FIXUP_HP_MIC, ALC274_FIXUP_HP_HEADSET_MIC, @@ -7744,6 +7745,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC256_FIXUP_INTEL_NUC10] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, [ALC255_FIXUP_XIAOMI_HEADSET_MIC] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -8183,6 +8193,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), + SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), #if 0 /* Below is a quirk table taken from the old code. -- 2.25.1

1 0

[PATCH] ALSA: hda: intel-nhlt: verify config type
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

Multiple bug reports report issues with the SOF and SST drivers when dealing with single microphone cases. We currently read the DMIC array information unconditionally but we don't check that the configuration type is actually a mic array. When the DMIC link does not rely on a mic array configuration, the recommendation is to check the format information to infer the maximum number of channels, and map this to the number of microphones. This leaves a potential for a mismatch between actual microphones available in hardware and what the ACPI table contains, but we have no other source of information. Note that single microphone configurations can alternatively be handled with a 'mic array' configuration along with a 'vendor-defined' geometry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201251 BugLink: https://github.com/thesofproject/linux/issues/2725 Fixes: 7a33ea70e1868 ('ALSA: hda: intel-nhlt: handle NHLT VENDOR_DEFINED DMIC geometry') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/intel-nhlt.h | 5 ++++ sound/hda/intel-nhlt.c | 54 +++++++++++++++++++++++++++++++------- 2 files changed, 50 insertions(+), 9 deletions(-) diff --git a/include/sound/intel-nhlt.h b/include/sound/intel-nhlt.h index 743c2f442280..d0574805865f 100644 --- a/include/sound/intel-nhlt.h +++ b/include/sound/intel-nhlt.h @@ -112,6 +112,11 @@ struct nhlt_vendor_dmic_array_config { /* TODO add vendor mic config */ } __packed; +enum { + NHLT_CONFIG_TYPE_GENERIC = 0, + NHLT_CONFIG_TYPE_MIC_ARRAY = 1 +}; + enum { NHLT_MIC_ARRAY_2CH_SMALL = 0xa, NHLT_MIC_ARRAY_2CH_BIG = 0xb, diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 059aaf04f536..d053beccfaec 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -31,18 +31,44 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) struct nhlt_endpoint *epnt; struct nhlt_dmic_array_config *cfg; struct nhlt_vendor_dmic_array_config *cfg_vendor; + struct nhlt_fmt *fmt_configs; unsigned int dmic_geo = 0; - u8 j; + u16 max_ch = 0; + u8 i, j; if (!nhlt) return 0; - epnt = (struct nhlt_endpoint *)nhlt->desc; + for (j = 0, epnt = nhlt->desc; j < nhlt->endpoint_count; j++, + epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length)) { - for (j = 0; j < nhlt->endpoint_count; j++) { - if (epnt->linktype == NHLT_LINK_DMIC) { - cfg = (struct nhlt_dmic_array_config *) - (epnt->config.caps); + if (epnt->linktype != NHLT_LINK_DMIC) + continue; + + cfg = (struct nhlt_dmic_array_config *)(epnt->config.caps); + fmt_configs = (struct nhlt_fmt *)(epnt->config.caps + epnt->config.size); + + /* find max number of channels based on format_configuration */ + if (fmt_configs->fmt_count) { + dev_dbg(dev, "%s: found %d format definitions\n", + __func__, fmt_configs->fmt_count); + + for (i = 0; i < fmt_configs->fmt_count; i++) { + struct wav_fmt_ext *fmt_ext; + + fmt_ext = &fmt_configs->fmt_config[i].fmt_ext; + + if (fmt_ext->fmt.channels > max_ch) + max_ch = fmt_ext->fmt.channels; + } + dev_dbg(dev, "%s: max channels found %d\n", __func__, max_ch); + } else { + dev_dbg(dev, "%s: No format information found\n", __func__); + } + + if (cfg->device_config.config_type != NHLT_CONFIG_TYPE_MIC_ARRAY) { + dmic_geo = max_ch; + } else { switch (cfg->array_type) { case NHLT_MIC_ARRAY_2CH_SMALL: case NHLT_MIC_ARRAY_2CH_BIG: @@ -59,13 +85,23 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) dmic_geo = cfg_vendor->nb_mics; break; default: - dev_warn(dev, "undefined DMIC array_type 0x%0x\n", - cfg->array_type); + dev_warn(dev, "%s: undefined DMIC array_type 0x%0x\n", + __func__, cfg->array_type); + } + + if (dmic_geo > 0) { + dev_dbg(dev, "%s: Array with %d dmics\n", __func__, dmic_geo); + } + if (max_ch > dmic_geo) { + dev_dbg(dev, "%s: max channels %d exceed dmic number %d\n", + __func__, max_ch, dmic_geo); } } - epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); } + dev_dbg(dev, "%s: dmic number %d max_ch %d\n", + __func__, dmic_geo, max_ch); + return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); -- 2.25.1

2 1

[PATCH v2] ALSA: hda: fix kernel-doc warnings
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

v5.12-rc1 flags new warnings with make W=1, fix missing or broken function descriptors. sound/pci/hda/hda_codec.c:3492: warning: expecting prototype for snd_hda_input_mux_info_info(). Prototype was for snd_hda_input_mux_info() instead sound/pci/hda/hda_codec.c:3521: warning: expecting prototype for snd_hda_input_mux_info_put(). Prototype was for snd_hda_input_mux_put() instead sound/pci/hda/hda_codec.c:3958: warning: expecting prototype for _snd_hda_pin_ctl(). Prototype was for _snd_hda_set_pin_ctl() instead sound/pci/hda/hda_jack.c:223: warning: expecting prototype for snd_hda_set_dirty_all(). Prototype was for snd_hda_jack_set_dirty_all() instead sound/pci/hda/hda_jack.c:309: warning: expecting prototype for snd_hda_jack_detect_enable_mst(). Prototype was for snd_hda_jack_detect_enable_callback_mst() instead sound/pci/hda/hda_generic.c:3933: warning: expecting prototype for snd_dha_gen_add_mute_led_cdev(). Prototype was for snd_hda_gen_add_mute_led_cdev() instead sound/pci/hda/hda_generic.c:4093: warning: expecting prototype for snd_dha_gen_add_micmute_led_cdev(). Prototype was for snd_hda_gen_add_micmute_led_cdev() instead sound/pci/hda/patch_ca0132.c:2357: warning: expecting prototype for Prepare and send the SCP message to DSP(). Prototype was for dspio_scp() instead sound/pci/hda/patch_ca0132.c:2883: warning: expecting prototype for Allocate router ports(). Prototype was for dsp_allocate_router_ports() instead sound/pci/hda/patch_ca0132.c:3202: warning: expecting prototype for Write a block of data into DSP code or data RAM using pre(). Prototype was for dspxfr_one_seg() instead sound/pci/hda/patch_ca0132.c:3397: warning: expecting prototype for data overlay to DSP memories(). Prototype was for dspxfr_image() instead sound/hda/hdac_regmap.c:393: warning: expecting prototype for snd_hdac_regmap_init(). Prototype was for snd_hdac_regmap_exit() instead sound/hda/ext/hdac_ext_controller.c:142: warning: expecting prototype for snd_hdac_ext_bus_get_link_index(). Prototype was for snd_hdac_ext_bus_get_link() instead sound/hda/ext/hdac_ext_stream.c:140: warning: expecting prototype for snd_hdac_ext_linkstream_start(). Prototype was for snd_hdac_ext_link_stream_start() instead Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/ext/hdac_ext_controller.c | 2 +- sound/hda/ext/hdac_ext_stream.c | 2 +- sound/hda/hdac_regmap.c | 2 +- sound/pci/hda/hda_codec.c | 6 +++--- sound/pci/hda/hda_generic.c | 4 ++-- sound/pci/hda/hda_jack.c | 4 ++-- sound/pci/hda/patch_ca0132.c | 9 ++++----- 7 files changed, 14 insertions(+), 15 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index a9bd39b93697..b2df7b4f9227 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -133,7 +133,7 @@ void snd_hdac_link_free_all(struct hdac_bus *bus) EXPORT_SYMBOL_GPL(snd_hdac_link_free_all); /** - * snd_hdac_ext_bus_get_link_index - get link based on codec name + * snd_hdac_ext_bus_get_link - get link based on codec name * @bus: the pointer to HDAC bus object * @codec_name: codec name */ diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index c4d54a838773..0c005d67fa89 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -133,7 +133,7 @@ void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_decouple); /** - * snd_hdac_ext_linkstream_start - start a stream + * snd_hdac_ext_link_stream_start - start a stream * @stream: HD-audio ext core stream to start */ void snd_hdac_ext_link_stream_start(struct hdac_ext_stream *stream) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index d75f31eb9d78..fe3587547cfe 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -386,7 +386,7 @@ int snd_hdac_regmap_init(struct hdac_device *codec) EXPORT_SYMBOL_GPL(snd_hdac_regmap_init); /** - * snd_hdac_regmap_init - Release the regmap from HDA codec + * snd_hdac_regmap_exit - Release the regmap from HDA codec * @codec: the codec object */ void snd_hdac_regmap_exit(struct hdac_device *codec) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9b755062d841..2026f1ccaf5a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3483,7 +3483,7 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power); */ /** - * snd_hda_input_mux_info_info - Info callback helper for the input-mux enum + * snd_hda_input_mux_info - Info callback helper for the input-mux enum * @imux: imux helper object * @uinfo: pointer to get/store the data */ @@ -3506,7 +3506,7 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux, EXPORT_SYMBOL_GPL(snd_hda_input_mux_info); /** - * snd_hda_input_mux_info_put - Put callback helper for the input-mux enum + * snd_hda_input_mux_put - Put callback helper for the input-mux enum * @codec: the HDA codec * @imux: imux helper object * @ucontrol: pointer to get/store the data @@ -3941,7 +3941,7 @@ unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_correct_pin_ctl); /** - * _snd_hda_pin_ctl - Helper to set pin ctl value + * _snd_hda_set_pin_ctl - Helper to set pin ctl value * @codec: the HDA codec * @pin: referred pin NID * @val: pin control value to set diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 5e40944e7342..8b7c5508f368 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3923,7 +3923,7 @@ static void vmaster_update_mute_led(void *private_data, int enabled) } /** - * snd_dha_gen_add_mute_led_cdev - Create a LED classdev and enable as vmaster mute LED + * snd_hda_gen_add_mute_led_cdev - Create a LED classdev and enable as vmaster mute LED * @codec: the HDA codec * @callback: the callback for LED classdev brightness_set_blocking */ @@ -4074,7 +4074,7 @@ static int add_micmute_led_hook(struct hda_codec *codec) } /** - * snd_dha_gen_add_micmute_led_cdev - Create a LED classdev and enable as mic-mute LED + * snd_hda_gen_add_micmute_led_cdev - Create a LED classdev and enable as mic-mute LED * @codec: the HDA codec * @callback: the callback for LED classdev brightness_set_blocking * diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index b8b568046592..ac00866d8032 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -213,7 +213,7 @@ static void jack_detect_update(struct hda_codec *codec, } /** - * snd_hda_set_dirty_all - Mark all the cached as dirty + * snd_hda_jack_set_dirty_all - Mark all the cached as dirty * @codec: the HDA codec * * This function sets the dirty flag to all entries of jack table. @@ -293,7 +293,7 @@ find_callback_from_list(struct hda_jack_tbl *jack, } /** - * snd_hda_jack_detect_enable_mst - enable the jack-detection + * snd_hda_jack_detect_enable_callback_mst - enable the jack-detection * @codec: the HDA codec * @nid: pin NID to enable * @func: callback function to register diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 7e62aed172a9..c966f49fa942 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2338,7 +2338,7 @@ static int dspio_send_scp_message(struct hda_codec *codec, } /** - * Prepare and send the SCP message to DSP + * dspio_scp - Prepare and send the SCP message to DSP * @codec: the HDA codec * @mod_id: ID of the DSP module to send the command * @src_id: ID of the source @@ -2865,7 +2865,7 @@ static int dsp_dma_stop(struct hda_codec *codec, } /** - * Allocate router ports + * dsp_allocate_router_ports - Allocate router ports * * @codec: the HDA codec * @num_chans: number of channels in the stream @@ -3178,8 +3178,7 @@ static int dspxfr_hci_write(struct hda_codec *codec, } /** - * Write a block of data into DSP code or data RAM using pre-allocated - * DMA engine. + * dspxfr_one_seg - Write a block of data into DSP code or data RAM using pre-allocated DMA engine. * * @codec: the HDA codec * @fls: pointer to a fast load image @@ -3376,7 +3375,7 @@ static int dspxfr_one_seg(struct hda_codec *codec, } /** - * Write the entire DSP image of a DSP code/data overlay to DSP memories + * dspxfr_image - Write the entire DSP image of a DSP code/data overlay to DSP memories * * @codec: the HDA codec * @fls_data: pointer to a fast load image -- 2.25.1

2 1

[PATCH 0/4]: ALSA: usb-audio: Add Pioneer DJM-850 support
by Nicolas MURE 02 Mar '21

02 Mar '21

Hello, This patches serie adds support for the Pioneer DJM-850 mixer. The mixer has a 24bits (little endian) sound card, which features 4 stereo channels for capture and 4 stereo channels for playback. The mixer also has controls to specify the capture source for each channels, and also a control to set the sample rate (44.1kHz, 48kHz, 96kHz). Thank you :) Signed-off-by: Nicolas MURE <nicolas.mure2019(a)gmail.com>

2 11

[PATCH 0/4] drop unneeded snd_soc_dai_set_drvdata
by Julia Lawall 02 Mar '21

02 Mar '21

snd_soc_dai_set_drvdata is not needed when the set data comes from snd_soc_dai_get_drvdata or dev_get_drvdata. --- sound/soc/fsl/fsl_micfil.c | 2 -- sound/soc/fsl/fsl_sai.c | 2 -- sound/soc/fsl/fsl_xcvr.c | 1 - sound/soc/mxs/mxs-saif.c | 10 ---------- sound/soc/pxa/mmp-sspa.c | 1 - sound/soc/sunxi/sun4i-i2s.c | 2 -- 6 files changed, 18 deletions(-)

3 6

[PATCH][next] ASoC: codecs: lpass-rx-macro: remove redundant initialization of variable hph_pwr_mode
by Colin King 02 Mar '21

02 Mar '21

From: Colin Ian King <colin.king(a)canonical.com> The variable hph_pwr_mode is being initialized with a value that is never read and it is being updated later with a new value. The initialization is redundant and can be removed. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/lpass-rx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 8c04b3b2c907..76909c50d7b9 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -2038,7 +2038,7 @@ static int rx_macro_load_compander_coeff(struct snd_soc_component *component, { u16 comp_coeff_lsb_reg, comp_coeff_msb_reg; int i; - int hph_pwr_mode = HPH_LOHIFI; + int hph_pwr_mode; if (!rx->comp_enabled[comp]) return 0; -- 2.30.0

3 2

[PATCH 0/6] ASoC: core: remove cppcheck warnings
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

This is the first batch of cleanups to make cppcheck more usable, currently we have way too many warnings that drown real issues. Pierre-Louis Bossart (6): ASoC: soc-ops: remove useless assignment ASoC: soc-pcm: remove redundant assignment ASoC: soc-pcm: remove shadowing variable ASoC: soc-pcm: add error log ASoC: soc-topology: clarify expression ASoC: generic: simple-card-utils: remove useless assignment sound/soc/generic/simple-card-utils.c | 2 +- sound/soc/soc-ops.c | 2 +- sound/soc/soc-pcm.c | 4 ++-- sound/soc/soc-topology.c | 16 ++++++++-------- 4 files changed, 12 insertions(+), 12 deletions(-) -- 2.25.1

2 7

[PATCH 00/10] ASoC: qcom: remove cppcheck warnings
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

Second batch of cleanups for Qualcomm SOCs and codecs. The only functional change is the addition of a missing error check in the last patch. Pierre-Louis Bossart (10): ASoC: qcom: lpass-hdmi: remove useless return ASoC: qcom: lpass-platform: remove useless assignment ASoC: qcom: q6dsp-dai: clarify expression ASoC: qcom: q6afe: remove useless assignments ASoC: qcom: q6afe: align function prototype ASoC: qcom: q6asm: align function prototypes ASoC: wcd-clsh-v2: align function prototypes ASoC: wcd9335: clarify return value ASoC: wcd934x: remove useless return ASoC: lpass-wsa-macro: add missing test sound/soc/codecs/lpass-wsa-macro.c | 2 ++ sound/soc/codecs/wcd-clsh-v2.h | 6 +++--- sound/soc/codecs/wcd9335.c | 2 +- sound/soc/codecs/wcd934x.c | 2 -- sound/soc/qcom/lpass-hdmi.c | 4 ---- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- sound/soc/qcom/qdsp6/q6afe.c | 5 ++--- sound/soc/qcom/qdsp6/q6afe.h | 2 +- sound/soc/qcom/qdsp6/q6asm.h | 6 +++--- 10 files changed, 14 insertions(+), 19 deletions(-) -- 2.25.1

3 12

[PATCH 0/6] ASoC: samsung: remove cppcheck warnings
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

No functional changes except for patch 2 and 3 where missing error checks were added for consistency. Pierre-Louis Bossart (6): ASoC: samsung: i2s: remove unassigned variable ASoC: samsung: s3c24xx_simtec: add missing error check ASoC: samsung: smdk_wm8994: add missing return ASoC: samsung: snow: remove useless test ASoC: samsung: tm2_wm5110: check of_parse return value ASoC: samsung: tm2_wm5510: remove shadowing variable sound/soc/samsung/i2s.c | 3 +-- sound/soc/samsung/s3c24xx_simtec.c | 5 +++++ sound/soc/samsung/smdk_wm8994.c | 1 + sound/soc/samsung/snow.c | 5 +---- sound/soc/samsung/tm2_wm5110.c | 3 +-- 5 files changed, 9 insertions(+), 8 deletions(-) -- 2.25.1

4 21

[PATCH 0/8] ASoC: sh: remove cppcheck warnings
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

No functional changes except for patch 4 where a missing error check was added for consistency. Pierre-Louis Bossart (8): ASoC: sh: dma-sh7760: remove unused variable ASoC: sh: rcar: align function prototypes ASoC: sh: rcar: simplify return ASoC: sh: rcar: core: rename shadowing variables ASoC: sh: rcar: ctu: add missing error check ASoC: sh: rcar: ssi: remove redundant assignment ASoC: sh: siu_pcm: remove useless assignment ASoC: sh: siu_pcm: remove unused variable sound/soc/sh/dma-sh7760.c | 1 - sound/soc/sh/rcar/core.c | 14 +++++++------- sound/soc/sh/rcar/ctu.c | 2 ++ sound/soc/sh/rcar/rsnd.h | 6 +++--- sound/soc/sh/rcar/ssi.c | 1 - sound/soc/sh/siu_pcm.c | 6 +----- 6 files changed, 13 insertions(+), 17 deletions(-) -- 2.25.1

3 13

[PATCH 0/9] ASoC: fsl: remove cppcheck warnings
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

Nothing critical and no functional changes. The only change that needs attention if the 'fsl_ssi: remove unnecessary tests' patch, where variables are to zero, then tested to set register fields. Either the tests are indeed redundant or the entire programming sequence is incorrect. Pierre-Louis Bossart (9): ASoC: fsl: fsl_asrc: remove useless assignment ASoC: fsl: fsl_dma: remove unused variable ASoC: fsl: fsl_easrc: remove useless assignments ASoC: fsl: fsl_esai: clarify expression ASoC: fsl: fsl_ssi: remove unnecessary tests ASoC: fsl: imx-hdmi: remove unused structure members ASoC: fsl: mpc5200: signed parameter in snprintf format ASoC: fsl: mpc8610: remove useless assignment ASoC: fsl: p1022_ds: remove useless assignment sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_dma.c | 3 --- sound/soc/fsl/fsl_easrc.c | 6 +++--- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_ssi.c | 5 ++--- sound/soc/fsl/imx-hdmi.c | 4 ---- sound/soc/fsl/mpc5200_dma.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- 9 files changed, 10 insertions(+), 18 deletions(-) -- 2.25.1

2 10

[PATCH] ASoC: constify of_phandle_args in snd_soc_get_dai_name()
by Krzysztof Kozlowski 02 Mar '21

02 Mar '21

The pointer to of_phandle_args passed to snd_soc_get_dai_name() and of_xlate_dai_name() implementations is not modified. Since it is being used only to translate passed OF node to a DAI name, it should not be modified, so mark it as const for correctness and safer code. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- include/sound/soc-component.h | 4 ++-- include/sound/soc.h | 2 +- sound/soc/meson/aiu-acodec-ctrl.c | 2 +- sound/soc/meson/aiu-codec-ctrl.c | 2 +- sound/soc/meson/aiu.c | 4 ++-- sound/soc/meson/aiu.h | 2 +- sound/soc/qcom/lpass-cpu.c | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- sound/soc/soc-component.c | 2 +- sound/soc/soc-core.c | 2 +- 10 files changed, 12 insertions(+), 12 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5b47768222b7..7dc75b39287f 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -101,7 +101,7 @@ struct snd_soc_component_driver { /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name); int (*of_xlate_dai_id)(struct snd_soc_component *comment, struct device_node *endpoint); @@ -450,7 +450,7 @@ void snd_soc_component_remove(struct snd_soc_component *component); int snd_soc_component_of_xlate_dai_id(struct snd_soc_component *component, struct device_node *ep); int snd_soc_component_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name); int snd_soc_component_compr_open(struct snd_compr_stream *cstream); void snd_soc_component_compr_free(struct snd_compr_stream *cstream, diff --git a/include/sound/soc.h b/include/sound/soc.h index bd38015d6c6d..78609ab331c8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1219,7 +1219,7 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, struct device_node **bitclkmaster, struct device_node **framemaster); int snd_soc_get_dai_id(struct device_node *ep); -int snd_soc_get_dai_name(struct of_phandle_args *args, +int snd_soc_get_dai_name(const struct of_phandle_args *args, const char **dai_name); int snd_soc_of_get_dai_name(struct device_node *of_node, const char **dai_name); diff --git a/sound/soc/meson/aiu-acodec-ctrl.c b/sound/soc/meson/aiu-acodec-ctrl.c index 7078197e0cc5..27a6d3259c50 100644 --- a/sound/soc/meson/aiu-acodec-ctrl.c +++ b/sound/soc/meson/aiu-acodec-ctrl.c @@ -159,7 +159,7 @@ static const struct snd_kcontrol_new aiu_acodec_ctrl_controls[] = { }; static int aiu_acodec_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name) { return aiu_of_xlate_dai_name(component, args, dai_name, AIU_ACODEC); diff --git a/sound/soc/meson/aiu-codec-ctrl.c b/sound/soc/meson/aiu-codec-ctrl.c index 4b773d3e8b07..c3ea733fce91 100644 --- a/sound/soc/meson/aiu-codec-ctrl.c +++ b/sound/soc/meson/aiu-codec-ctrl.c @@ -125,7 +125,7 @@ static const struct snd_soc_dapm_route aiu_hdmi_ctrl_routes[] = { }; static int aiu_hdmi_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name) { return aiu_of_xlate_dai_name(component, args, dai_name, AIU_HDMI); diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index dc35ca79021c..ba15d5762b0b 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -42,7 +42,7 @@ static const struct snd_soc_dapm_route aiu_cpu_dapm_routes[] = { }; int aiu_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name, unsigned int component_id) { @@ -72,7 +72,7 @@ int aiu_of_xlate_dai_name(struct snd_soc_component *component, } static int aiu_cpu_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name) { return aiu_of_xlate_dai_name(component, args, dai_name, AIU_CPU); diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h index 87aa19ac4af3..393b6c2307e4 100644 --- a/sound/soc/meson/aiu.h +++ b/sound/soc/meson/aiu.h @@ -45,7 +45,7 @@ struct aiu { SNDRV_PCM_FMTBIT_S24_LE) int aiu_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name, unsigned int component_id); diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c642e5f8f28c..4762286b33fe 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -340,7 +340,7 @@ int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai) EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_probe); static int asoc_qcom_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name) { struct lpass_data *drvdata = snd_soc_component_get_drvdata(component); diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 4e1f101281e7..e8915519f427 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1315,7 +1315,7 @@ static struct snd_soc_dai_driver q6afe_dais[] = { }; static int q6afe_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name) { int id = args->args[0]; diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 159bf88b9f8c..8415e9bd2932 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -370,7 +370,7 @@ int snd_soc_component_of_xlate_dai_id(struct snd_soc_component *component, } int snd_soc_component_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, + const struct of_phandle_args *args, const char **dai_name) { if (component->driver->of_xlate_dai_name) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f6d4e99b590c..50190654edda 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2994,7 +2994,7 @@ int snd_soc_get_dai_id(struct device_node *ep) } EXPORT_SYMBOL_GPL(snd_soc_get_dai_id); -int snd_soc_get_dai_name(struct of_phandle_args *args, +int snd_soc_get_dai_name(const struct of_phandle_args *args, const char **dai_name) { struct snd_soc_component *pos; -- 2.25.1

2 1

[PATCH 0/5] soc-pcm: tidyup snd_pcm_hardware setup for FE/BE
by Kuninori Morimoto 02 Mar '21

02 Mar '21

Hi Mark These patches tidyup snd_pcm_hardware setup for FE/BE. [1/5] changes behavior, but I think it is bug-fix. Kuninori Morimoto (5): ASoC: soc-pcm: remove strange format storing ASoC: soc-pcm: unpack dpcm_init_runtime_hw() ASoC: soc-pcm: add dpcm_runtime_setup_fe() ASoC: soc-pcm: add dpcm_runtime_setup() ASoC: soc-pcm: unpack dpcm_set_fe_runtime() sound/soc/soc-pcm.c | 79 +++++++++++++++++++++------------------------ 1 file changed, 36 insertions(+), 43 deletions(-) -- 2.25.1

2 6