September 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASOC: Add ADAU7118 8 Channel PDM-to-I2S/TDM Converter driver
by Nuno Sá 02 Oct '19

02 Oct '19

This patch adds support for the 8 channel PDM-to-I2S/TDM converter. The ADAU7118 converts four stereo pulse density modulation (PDM) bitstreams into one pulse code modulation (PCM) output stream. The source for the PDM data can be eight microphones or other PDM sources. The PCM audio data is output on a serial audio interface port in either inter-IC serial (I2S) or time domain multiplexed (TDM) format. Signed-off-by: Nuno Sá <nuno.sa(a)analog.com> --- MAINTAINERS | 1 + sound/soc/codecs/Kconfig | 28 ++ sound/soc/codecs/Makefile | 6 + sound/soc/codecs/adau7118-hw.c | 43 +++ sound/soc/codecs/adau7118-i2c.c | 82 +++++ sound/soc/codecs/adau7118.c | 621 ++++++++++++++++++++++++++++++++ sound/soc/codecs/adau7118.h | 24 ++ 7 files changed, 805 insertions(+) create mode 100644 sound/soc/codecs/adau7118-hw.c create mode 100644 sound/soc/codecs/adau7118-i2c.c create mode 100644 sound/soc/codecs/adau7118.c create mode 100644 sound/soc/codecs/adau7118.h diff --git a/MAINTAINERS b/MAINTAINERS index a50e97a63bc8..77c920559d82 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -985,6 +985,7 @@ F: drivers/media/i2c/adv7842* ANALOG DEVICES INC ASOC CODEC DRIVERS M: Lars-Peter Clausen <lars(a)metafoo.de> +M: Nuno Sá <nuno.sa(a)analog.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://wiki.analog.com/ W: http://ez.analog.com/community/linux-device-drivers diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9f89a5346299..006b0686776c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -34,6 +34,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU1977_I2C if I2C select SND_SOC_ADAU1701 if I2C select SND_SOC_ADAU7002 + select SND_SOC_ADAU7118_I2C if I2C + select SND_SOC_ADAU7118_HW select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4118 if I2C @@ -392,6 +394,32 @@ config SND_SOC_ADAU1977_I2C config SND_SOC_ADAU7002 tristate "Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter" +config SND_SOC_ADAU7118 + tristate + +config SND_SOC_ADAU7118_HW + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - HW Mode" + select SND_SOC_ADAU7118 + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter. In this mode, the device works in standalone mode which + means that there is no bus to comunicate with it. Stereo mode is not + supported in this mode. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-hw. + +config SND_SOC_ADAU7118_I2C + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - I2C" + select SND_SOC_ADAU7118 + select REGMAP_I2C + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter over I2C. This gives full support over the device. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-i2c. + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5b4bb8cf4325..974dc1eda797 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,9 @@ snd-soc-adau1977-objs := adau1977.o snd-soc-adau1977-spi-objs := adau1977-spi.o snd-soc-adau1977-i2c-objs := adau1977-i2c.o snd-soc-adau7002-objs := adau7002.o +snd-soc-adau7118-objs := adau7118.o +snd-soc-adau7118-i2c-objs := adau7118-i2c.o +snd-soc-adau7118-hw-objs := adau7118-hw.o snd-soc-adav80x-objs := adav80x.o snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o @@ -301,6 +304,9 @@ obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o obj-$(CONFIG_SND_SOC_ADAU7002) += snd-soc-adau7002.o +obj-$(CONFIG_SND_SOC_ADAU7118) += snd-soc-adau7118.o +obj-$(CONFIG_SND_SOC_ADAU7118_I2C) += snd-soc-adau7118-i2c.o +obj-$(CONFIG_SND_SOC_ADAU7118_HW) += snd-soc-adau7118-hw.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o diff --git a/sound/soc/codecs/adau7118-hw.c b/sound/soc/codecs/adau7118-hw.c new file mode 100644 index 000000000000..499dd48ea01b --- /dev/null +++ b/sound/soc/codecs/adau7118-hw.c @@ -0,0 +1,43 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter Standalone Hw + * driver + * + * Copyright 2019 Analog Devices Inc. + */ +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/platform_device.h> + +#include "adau7118.h" + +static int adau7118_probe_hw(struct platform_device *pdev) +{ + return adau7118_probe(&pdev->dev, NULL, true); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct platform_device_id adau7118_id[] = { + { .name = "adau7118" }, + { } +}; +MODULE_DEVICE_TABLE(platform, adau7118_id); + +static struct platform_driver adau7118_driver_hw = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_hw, + .id_table = adau7118_id, +}; +module_platform_driver(adau7118_driver_hw); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver for standalone hw mode"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118-i2c.c b/sound/soc/codecs/adau7118-i2c.c new file mode 100644 index 000000000000..1fd0b0287699 --- /dev/null +++ b/sound/soc/codecs/adau7118-i2c.c @@ -0,0 +1,82 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C + * + * Copyright 2019 Analog Devices Inc. + */ +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "adau7118.h" + +static const struct reg_default adau7118_reg_defaults[] = { + { ADAU7118_REG_VENDOR_ID, 0x41 }, + { ADAU7118_REG_DEVICE_ID1, 0x71 }, + { ADAU7118_REG_DEVICE_ID2, 0x18 }, + { ADAU7118_REG_REVISION_ID, 0x00 }, + { ADAU7118_REG_ENABLES, 0x3F }, + { ADAU7118_REG_DEC_RATIO_CLK_MAP, 0xC0 }, + { ADAU7118_REG_HPF_CONTROL, 0xD0 }, + { ADAU7118_REG_SPT_CTRL1, 0x41 }, + { ADAU7118_REG_SPT_CTRL2, 0x00 }, + { ADAU7118_REG_SPT_CX(0), 0x01 }, + { ADAU7118_REG_SPT_CX(1), 0x11 }, + { ADAU7118_REG_SPT_CX(2), 0x21 }, + { ADAU7118_REG_SPT_CX(3), 0x31 }, + { ADAU7118_REG_SPT_CX(4), 0x41 }, + { ADAU7118_REG_SPT_CX(5), 0x51 }, + { ADAU7118_REG_SPT_CX(6), 0x61 }, + { ADAU7118_REG_SPT_CX(7), 0x71 }, + { ADAU7118_REG_DRIVE_STRENGTH, 0x2a }, + { ADAU7118_REG_RESET, 0x00 }, +}; + +static const struct regmap_config adau7118_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .reg_defaults = adau7118_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau7118_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .max_register = ADAU7118_REG_RESET, +}; + +static int adau7118_probe_i2c(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *map; + + map = devm_regmap_init_i2c(i2c, &adau7118_regmap_config); + if (IS_ERR(map)) { + dev_err(&i2c->dev, "Failed to init regmap %ld\n", PTR_ERR(map)); + return PTR_ERR(map); + } + + return adau7118_probe(&i2c->dev, map, false); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct i2c_device_id adau7118_id[] = { + {"adau7118", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, adau7118_id); + +static struct i2c_driver adau7118_driver = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_i2c, + .id_table = adau7118_id, +}; +module_i2c_driver(adau7118_driver); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c new file mode 100644 index 000000000000..d6b90e31042f --- /dev/null +++ b/sound/soc/codecs/adau7118.c @@ -0,0 +1,621 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver + * + * Copyright 2019 Analog Devices Inc. + */ +#include <linux/bitfield.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "adau7118.h" + +#define ADAU7118_DEC_RATIO_MASK GENMASK(1, 0) +#define ADAU7118_DEC_RATIO(x) FIELD_PREP(ADAU7118_DEC_RATIO_MASK, x) +#define ADAU7118_CLK_MAP_MASK GENMASK(7, 4) +#define ADAU7118_SLOT_WIDTH_MASK GENMASK(5, 4) +#define ADAU7118_SLOT_WIDTH(x) FIELD_PREP(ADAU7118_SLOT_WIDTH_MASK, x) +#define ADAU7118_TRISTATE_MASK BIT(6) +#define ADAU7118_TRISTATE(x) FIELD_PREP(ADAU7118_TRISTATE_MASK, x) +#define ADAU7118_DATA_FMT_MASK GENMASK(3, 1) +#define ADAU7118_DATA_FMT(x) FIELD_PREP(ADAU7118_DATA_FMT_MASK, x) +#define ADAU7118_SAI_MODE_MASK BIT(0) +#define ADAU7118_SAI_MODE(x) FIELD_PREP(ADAU7118_SAI_MODE_MASK, x) +#define ADAU7118_LRCLK_BCLK_POL_MASK GENMASK(1, 0) +#define ADAU7118_LRCLK_BCLK_POL(x) \ + FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) +#define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) +#define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_FULL_SOFT_R_MASK BIT(1) +#define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) + +struct adau7118_data { + struct regmap *map; + struct device *dev; + struct regulator *iovdd; + struct regulator *dvdd; + u32 slot_width; + u32 slots; + bool hw_mode; + bool right_j; +}; + +/* Input Enable */ +static const struct snd_kcontrol_new adau7118_dapm_pdm_control[4] = { + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 0, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 1, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 2, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 3, 1, 0), +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_sw[] = { + /* Input Enable Switches */ + SND_SOC_DAPM_SWITCH("PDM0", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[0]), + SND_SOC_DAPM_SWITCH("PDM1", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[1]), + SND_SOC_DAPM_SWITCH("PDM2", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[2]), + SND_SOC_DAPM_SWITCH("PDM3", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[3]), + + /* PDM Clocks */ + SND_SOC_DAPM_SUPPLY("PDM_CLK0", ADAU7118_REG_ENABLES, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM_CLK1", ADAU7118_REG_ENABLES, 5, 0, NULL, 0), + + /* Output channels */ + SND_SOC_DAPM_AIF_OUT("AIF1TX1", "Capture", 0, ADAU7118_REG_SPT_CX(0), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX2", "Capture", 0, ADAU7118_REG_SPT_CX(1), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX3", "Capture", 0, ADAU7118_REG_SPT_CX(2), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX4", "Capture", 0, ADAU7118_REG_SPT_CX(3), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX5", "Capture", 0, ADAU7118_REG_SPT_CX(4), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX6", "Capture", 0, ADAU7118_REG_SPT_CX(5), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX7", "Capture", 0, ADAU7118_REG_SPT_CX(6), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX8", "Capture", 0, ADAU7118_REG_SPT_CX(7), + 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_sw[] = { + { "PDM0", "Capture Switch", "PDM_DAT0" }, + { "PDM1", "Capture Switch", "PDM_DAT1" }, + { "PDM2", "Capture Switch", "PDM_DAT2" }, + { "PDM3", "Capture Switch", "PDM_DAT3" }, + { "AIF1TX1", NULL, "PDM0" }, + { "AIF1TX2", NULL, "PDM0" }, + { "AIF1TX3", NULL, "PDM1" }, + { "AIF1TX4", NULL, "PDM1" }, + { "AIF1TX5", NULL, "PDM2" }, + { "AIF1TX6", NULL, "PDM2" }, + { "AIF1TX7", NULL, "PDM3" }, + { "AIF1TX8", NULL, "PDM3" }, + { "Capture", NULL, "PDM_CLK0" }, + { "Capture", NULL, "PDM_CLK1" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_hw[] = { + SND_SOC_DAPM_AIF_OUT("AIF1TX", "Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_hw[] = { + { "AIF1TX", NULL, "PDM_DAT0" }, + { "AIF1TX", NULL, "PDM_DAT1" }, + { "AIF1TX", NULL, "PDM_DAT2" }, + { "AIF1TX", NULL, "PDM_DAT3" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets[] = { + SND_SOC_DAPM_INPUT("PDM_DAT0"), + SND_SOC_DAPM_INPUT("PDM_DAT1"), + SND_SOC_DAPM_INPUT("PDM_DAT2"), + SND_SOC_DAPM_INPUT("PDM_DAT3"), +}; + +static int adau7118_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int chan, ret; + + dev_dbg(st->dev, "Set channel map, %d", tx_num); + + for (chan = 0; chan < tx_num; chan++) { + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CX(chan), + ADAU7118_SPT_SLOT_MASK, + ADAU7118_SPT_SLOT(tx_slot[chan])); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set format, fmt:%d\n", fmt); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(0)); + break; + case SND_SOC_DAIFMT_LEFT_J: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(1)); + break; + case SND_SOC_DAIFMT_RIGHT_J: + st->right_j = true; + break; + default: + dev_err(st->dev, "Invalid format %d", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + if (ret < 0) + return ret; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(0); + break; + case SND_SOC_DAIFMT_NB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(2); + break; + case SND_SOC_DAIFMT_IB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(1); + break; + case SND_SOC_DAIFMT_IB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(3); + break; + default: + dev_err(st->dev, "Invalid Inv mask %d", + fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL2, + ADAU7118_LRCLK_BCLK_POL_MASK, + regval); + + return ret < 0 ? ret : 0; +} + +static int adau7118_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret; + + dev_dbg(st->dev, "Set tristate, %d\n", tristate); + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_TRISTATE_MASK, + ADAU7118_TRISTATE(tristate)); + + return ret < 0 ? ret : 0; +} + +static int adau7118_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set tdm, slots:%d width:%d\n", slots, slot_width); + + switch (slot_width) { + case 32: + regval = ADAU7118_SLOT_WIDTH(0); + break; + case 24: + regval = ADAU7118_SLOT_WIDTH(2); + break; + case 16: + regval = ADAU7118_SLOT_WIDTH(1); + break; + default: + dev_err(st->dev, "Invalid slot width:%d\n", slot_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SLOT_WIDTH_MASK, regval); + if (ret < 0) + return ret; + + st->slot_width = slot_width; + st->slots = slots; + + return 0; +} + +static int adau7118_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + u32 data_width = params_width(params), slots_width; + int ret; + u32 regval; + + if (!st->slots) { + /* set stereo mode */ + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SAI_MODE_MASK, + ADAU7118_SAI_MODE(0)); + if (ret < 0) + return ret; + + slots_width = 32; + } else { + slots_width = st->slot_width; + } + + if (data_width > slots_width) { + dev_err(st->dev, "Invalid data_width:%d, slots_width:%d", + data_width, slots_width); + return -EINVAL; + } + + if (st->right_j) { + switch (slots_width - data_width) { + case 8: + /* delay bclck by 8 */ + regval = ADAU7118_DATA_FMT(2); + break; + case 12: + /* delay bclck by 12 */ + regval = ADAU7118_DATA_FMT(3); + break; + case 16: + /* delay bclck by 16 */ + regval = ADAU7118_DATA_FMT(4); + break; + default: + dev_err(st->dev, + "Cannot set right_j setting, slot_w:%d, data_w:%d\n", + slots_width, data_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + regval); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + int ret = 0; + + dev_dbg(st->dev, "Set bias level %d\n", level); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + if (!st->iovdd) + return 0; + + /* power on */ + ret = regulator_enable(st->iovdd); + if (ret) + return ret; + + /* there's no timing constraints before enabling dvdd */ + ret = regulator_enable(st->dvdd); + if (ret) { + regulator_disable(st->iovdd); + return ret; + } + + if (st->hw_mode) + return 0; + + regcache_cache_only(st->map, false); + /* sync cache */ + ret = snd_soc_component_cache_sync(component); + } + break; + case SND_SOC_BIAS_OFF: + /* power off */ + if (!st->iovdd) + return 0; + + ret = regulator_disable(st->dvdd); + if (ret) + return ret; + + ret = regulator_disable(st->iovdd); + if (ret) + return ret; + + if (st->hw_mode) + return 0; + + /* cache only */ + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + + break; + } + + return ret; +} + +static int adau7118_component_probe(struct snd_soc_component *component) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + int ret = 0; + + if (st->hw_mode) { + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_hw, + ARRAY_SIZE(adau7118_widgets_hw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_hw, + ARRAY_SIZE(adau7118_routes_hw)); + } else { + snd_soc_component_init_regmap(component, st->map); + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_sw, + ARRAY_SIZE(adau7118_widgets_sw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_sw, + ARRAY_SIZE(adau7118_routes_sw)); + } + + return ret; +} + +#ifdef CONFIG_PM +static int adau7118_suspend(struct snd_soc_component *component) +{ + return snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); +} + +static int adau7118_resume(struct snd_soc_component *component) +{ + return snd_soc_component_force_bias_level(component, + SND_SOC_BIAS_STANDBY); +} +#else +#define adau7118_suspend NULL +#define adau7118_resume NULL +#endif + +static const struct snd_soc_dai_ops adau7118_ops = { + .hw_params = adau7118_hw_params, + .set_channel_map = adau7118_set_channel_map, + .set_fmt = adau7118_set_fmt, + .set_tdm_slot = adau7118_set_tdm_slot, + .set_tristate = adau7118_set_tristate, +}; + +static struct snd_soc_dai_driver adau7118_dai = { + .name = "adau7118-hifi-capture", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 8, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S20_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 192000, + .sig_bits = 24, + }, +}; + +static const struct snd_soc_component_driver adau7118_component_driver = { + .probe = adau7118_component_probe, + .set_bias_level = adau7118_set_bias_level, + .suspend = adau7118_suspend, + .resume = adau7118_resume, + .dapm_widgets = adau7118_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau7118_widgets), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static void adau7118_regulator_disable(void *data) +{ + struct adau7118_data *st = data; + int ret; + /* + * If we fail to disable DVDD, don't bother in trying IOVDD. We + * actually don't want to be left in the situation where DVDD + * is enabled and IOVDD is disabled. + */ + ret = regulator_disable(st->dvdd); + if (ret) + return; + + regulator_disable(st->iovdd); +} + +static int adau7118_regulator_setup(struct adau7118_data *st) +{ + int ret = 0; + + st->iovdd = devm_regulator_get_optional(st->dev, "IOVDD"); + if (!IS_ERR(st->iovdd)) { + /* + * Only get dvdd if iovdd is present. Furthermore, + * if iovdd is present, dvdd must also be given. The two + * supplies should be controlled together since dvdd should + * not be applied before iovdd. + */ + st->dvdd = devm_regulator_get(st->dev, "DVDD"); + if (IS_ERR(st->dvdd)) { + dev_err(st->dev, "Could not get dvdd: %ld\n", + PTR_ERR(st->dvdd)); + return PTR_ERR(st->dvdd); + } + /* just assume the device is in reset */ + if (!st->hw_mode) { + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + } + + ret = devm_add_action_or_reset(st->dev, + adau7118_regulator_disable, st); + } else { + if (PTR_ERR(st->iovdd) == -ENODEV) { + st->iovdd = NULL; + } else { + dev_err(st->dev, "Failed to get IOVDD %ld\n", + PTR_ERR(st->iovdd)); + return PTR_ERR(st->iovdd); + } + } + + return ret; +} + +static int adau7118_parset_dt(const struct adau7118_data *st) +{ + int ret; + u32 dec_ratio = 0; + /* 4 inputs */ + u32 clk_map[4], regval; + + if (st->hw_mode) + return 0; + + ret = device_property_read_u32(st->dev, "adi,decimation-ratio", + &dec_ratio); + if (!ret) { + switch (dec_ratio) { + case 64: + regval = ADAU7118_DEC_RATIO(0); + break; + case 32: + regval = ADAU7118_DEC_RATIO(1); + break; + case 16: + regval = ADAU7118_DEC_RATIO(2); + break; + default: + dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_DEC_RATIO_MASK, regval); + if (ret) + return ret; + } + + ret = device_property_read_u32_array(st->dev, "adi,pdm-clk-map", + clk_map, ARRAY_SIZE(clk_map)); + if (!ret) { + int pdm; + u32 _clk_map = 0; + + for (pdm = 0; pdm < ARRAY_SIZE(clk_map); pdm++) + _clk_map |= (clk_map[pdm] << (pdm + 4)); + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_CLK_MAP_MASK, _clk_map); + if (ret) + return ret; + } + + return 0; +} + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode) +{ + struct adau7118_data *st; + int ret; + + st = devm_kzalloc(dev, sizeof(*st), GFP_KERNEL); + if (!st) + return -ENOMEM; + + st->dev = dev; + st->hw_mode = hw_mode; + dev_set_drvdata(dev, st); + + if (!hw_mode) { + st->map = map; + adau7118_dai.ops = &adau7118_ops; + /* + * Perform a full soft reset. This will set all register's + * with their reset values. + */ + ret = regmap_update_bits(map, ADAU7118_REG_RESET, + ADAU7118_FULL_SOFT_R_MASK, + ADAU7118_FULL_SOFT_R(1)); + if (ret) + return ret; + } + + ret = adau7118_parset_dt(st); + if (ret) + return ret; + + ret = adau7118_regulator_setup(st); + if (ret) + return ret; + + return devm_snd_soc_register_component(dev, + &adau7118_component_driver, + &adau7118_dai, 1); +} +EXPORT_SYMBOL_GPL(adau7118_probe); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.h b/sound/soc/codecs/adau7118.h new file mode 100644 index 000000000000..c65679a4dff1 --- /dev/null +++ b/sound/soc/codecs/adau7118.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef _LINUX_ADAU7118_H +#define _LINUX_ADAU7118_H + +struct regmap; +struct device; + +/* register map */ +#define ADAU7118_REG_VENDOR_ID 0x00 +#define ADAU7118_REG_DEVICE_ID1 0x01 +#define ADAU7118_REG_DEVICE_ID2 0x02 +#define ADAU7118_REG_REVISION_ID 0x03 +#define ADAU7118_REG_ENABLES 0x04 +#define ADAU7118_REG_DEC_RATIO_CLK_MAP 0x05 +#define ADAU7118_REG_HPF_CONTROL 0x06 +#define ADAU7118_REG_SPT_CTRL1 0x07 +#define ADAU7118_REG_SPT_CTRL2 0x08 +#define ADAU7118_REG_SPT_CX(num) (0x09 + (num)) +#define ADAU7118_REG_DRIVE_STRENGTH 0x11 +#define ADAU7118_REG_RESET 0x12 + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode); + +#endif -- 2.23.0

3 8

[alsa-devel] alsa acting up
by liebrecht＠grossmann-venter.com 02 Oct '19

02 Oct '19

See images in links. 1) "http://grossmann-venter.com/visitors/ug/alsa/alsa-pulse-181vsl_unplug.png" 2) "http://grossmann-venter.com/visitors/ug/alsa/alsa-dont recognize1818VSLpng" 1) As soon as pulseaudio is started Alsa says 1818VSL is unplugged....? 2) Clearly Presonus 1818VSL is found as lsusb output says, but alsa has no clue about it. How do I get alsa to recognise 181VSL as it is found by lsusb so it is presnet on the system. It used to work fine but alsa just stopped working with 1818VSL recently for no reason. I know this is a developers group, but please at least answer where I can get help. Alsa help is absolutely non-existing. I also posted a bug with no response in another thread. Thanks.

2 3

[alsa-devel] [PATCH 00/13] ASoC: SOF: fixes and improvements
by Pierre-Louis Bossart 01 Oct '19

01 Oct '19

No new features in this patchset but fixes and improvements identified during validation/product hardening/SoundWire development. I realize we are at the end of the merge window so will let Takashi/Mark decide if this series can be provided as fixes for v5.4 or queued for later. Kai Vehmanen (2): ASoC: SOF: pcm: fix resource leak in hw_free ASoC: SOF: Intel: hda: fix warnings during FW load Keyon Jie (1): ASoC: SOF: topology: fix parse fail issue for byte/bool tuple types Liam Girdwood (3): ASoC: SOF: ipc: retain DSP context after FW exception. ASoC: SOF: Intel: initialise and verify FW crash dump data. ASoC: SOF: Intel: Add context data to any IPC timeout. Marcin Rajwa (2): ASoC: SOF: ipc: introduce no_stream_position in sof_ipc_stream_params struct ASoC: SOF: Intel: hda: fix reset of host_period_bytes Pan Xiuli (1): ASoC: SOF: pcm: harden PCM STOP sequence Pierre-Louis Bossart (2): ASoC: SOF: loader: fix kernel oops on firmware boot failure ASoC: SOF: trace: move to opt-in with Kconfig and module parameter Ranjani Sridharan (2): ASoC: SOF: core: check for mandatory fw_ready op during SOF probe ASoC: SOF: Intel: hda: Disable DMI L1 entry during capture include/sound/sof/stream.h | 4 +-- sound/soc/sof/Kconfig | 16 ++++++++++++ sound/soc/sof/core.c | 28 +++++++++++++++----- sound/soc/sof/debug.c | 16 ++++++++++++ sound/soc/sof/intel/Kconfig | 10 +++++++ sound/soc/sof/intel/bdw.c | 29 +++++++++++++++++++- sound/soc/sof/intel/byt.c | 29 +++++++++++++++++++- sound/soc/sof/intel/hda-ctrl.c | 12 +++------ sound/soc/sof/intel/hda-loader.c | 1 + sound/soc/sof/intel/hda-pcm.c | 15 ++++++++--- sound/soc/sof/intel/hda-stream.c | 45 +++++++++++++++++++++++++++----- sound/soc/sof/intel/hda.c | 7 +++++ sound/soc/sof/intel/hda.h | 5 +++- sound/soc/sof/ipc.c | 10 +------ sound/soc/sof/loader.c | 4 +-- sound/soc/sof/pcm.c | 35 +++++++++++++++++-------- sound/soc/sof/sof-priv.h | 17 ++++++++---- sound/soc/sof/topology.c | 4 ++- sound/soc/sof/trace.c | 17 +++++++++++- 19 files changed, 245 insertions(+), 59 deletions(-) -- 2.20.1

2 26

[alsa-devel] [PATCH 0/4] ASoC: Intel: boards: improvements for RT5651/82-based platforms
by Pierre-Louis Bossart 01 Oct '19

01 Oct '19

RT5651/RT5682 boards are used for SOF validation. This patchset corrects problems identified during module load/reload tests and an error for DMIC support in some configurations. These fixes are not critical and can be queued for v5.5 Bard Liao (1): ASoC: intel: sof_rt5682: use separate route map for dmic Jaska Uimonen (3): ASoC: intel: bytcr_rt5651: add null check to support_button_press ASoC: rt5682: add NULL handler to set_jack function ASoC: intel: sof_rt5682: add remove function to disable jack sound/soc/codecs/rt5651.c | 3 ++ sound/soc/codecs/rt5682.c | 12 +++++-- sound/soc/intel/boards/sof_rt5682.c | 51 ++++++++++++++++++++++++++--- 3 files changed, 60 insertions(+), 6 deletions(-) -- 2.20.1

2 8

[alsa-devel] [PATCH] ASoC: max98373: check for device node before parsing
by Sathyanarayana Nujella 01 Oct '19

01 Oct '19

Below Oops is caused in a system which uses ACPI instead of device node: of_get_named_gpiod_flags: can't parse 'maxim,reset-gpio' property of node '(null)[0]' BUG: kernel NULL pointer dereference, address: 0000000000000010 This patch avoids NULL pointer deferencing by adding a check before parsing and initializes to make reset-gpio pin as invalid. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Jairaj Arava <jairaj.arava(a)intel.com> --- sound/soc/codecs/max98373.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index e609abcf3220..eb709d528259 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -901,16 +901,20 @@ static void max98373_slot_config(struct i2c_client *i2c, max98373->i_slot = value & 0xF; else max98373->i_slot = 1; - - max98373->reset_gpio = of_get_named_gpio(dev->of_node, + if (dev->of_node) { + max98373->reset_gpio = of_get_named_gpio(dev->of_node, "maxim,reset-gpio", 0); - if (!gpio_is_valid(max98373->reset_gpio)) { - dev_err(dev, "Looking up %s property in node %s failed %d\n", - "maxim,reset-gpio", dev->of_node->full_name, - max98373->reset_gpio); + if (!gpio_is_valid(max98373->reset_gpio)) { + dev_err(dev, "Looking up %s property in node %s failed %d\n", + "maxim,reset-gpio", dev->of_node->full_name, + max98373->reset_gpio); + } else { + dev_dbg(dev, "maxim,reset-gpio=%d", + max98373->reset_gpio); + } } else { - dev_dbg(dev, "maxim,reset-gpio=%d", - max98373->reset_gpio); + /* this makes reset_gpio as invalid */ + max98373->reset_gpio = -1; } if (!device_property_read_u32(dev, "maxim,spkfb-slot-no", &value)) -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: Intel: cht_bsw_rt5645: Add quirk for boards using pmc_plt_clk_0
by Sam McNally 01 Oct '19

01 Oct '19

As of commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL"), the cht_bsw_rt5645 driver needs to enable the clock it's using for the codec's mclk. It does this from commit 7735bce05a9c ("ASoC: Intel: boards: use devm_clk_get() unconditionally"), enabling pmc_plt_clk_3. However, Strago family Chromebooks use pmc_plt_clk_0 for the codec mclk, resulting in white noise with some digital microphones. Add a DMI-based quirk for Strago family Chromebooks to use pmc_plt_clk_0 instead - mirroring the changes made to cht_bsw_max98090_ti in commit a182ecd3809c ("ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0") and making use of the existing dmi_check_system() call and related infrastructure added in commit 22af29114eb4 ("ASoC: Intel: cht-bsw-rt5645: add quirks for SSP0/AIF1/AIF2 routing"). Signed-off-by: Sam McNally <sammc(a)chromium.org> --- Changes in v2: - Added to the description references to more related commits (similar cht_bsw_max98090_ti clock quirks inspiring this change and and the cht_bsw_rt5645 quirks setup) sound/soc/intel/boards/cht_bsw_rt5645.c | 26 +++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 8879c3be29d5..c68a5b85a4a0 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -48,6 +48,7 @@ struct cht_mc_private { #define CHT_RT5645_SSP2_AIF2 BIT(16) /* default is using AIF1 */ #define CHT_RT5645_SSP0_AIF1 BIT(17) #define CHT_RT5645_SSP0_AIF2 BIT(18) +#define CHT_RT5645_PMC_PLT_CLK_0 BIT(19) static unsigned long cht_rt5645_quirk = 0; @@ -59,6 +60,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP0_AIF1 enabled"); if (cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2) dev_info(dev, "quirk SSP0_AIF2 enabled"); + if (cht_rt5645_quirk & CHT_RT5645_PMC_PLT_CLK_0) + dev_info(dev, "quirk PMC_PLT_CLK_0 enabled"); } static int platform_clock_control(struct snd_soc_dapm_widget *w, @@ -226,15 +229,21 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, return 0; } -/* uncomment when we have a real quirk static int cht_rt5645_quirk_cb(const struct dmi_system_id *id) { cht_rt5645_quirk = (unsigned long)id->driver_data; return 1; } -*/ static const struct dmi_system_id cht_rt5645_quirk_table[] = { + { + /* Strago family Chromebooks */ + .callback = cht_rt5645_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Intel_Strago"), + }, + .driver_data = (void *)CHT_RT5645_PMC_PLT_CLK_0, + }, { }, }; @@ -526,6 +535,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) int dai_index = 0; int ret_val = 0; int i; + const char *mclk_name; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) @@ -662,11 +672,15 @@ static int snd_cht_mc_probe(struct platform_device *pdev) if (ret_val) return ret_val; - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (cht_rt5645_quirk & CHT_RT5645_PMC_PLT_CLK_0) + mclk_name = "pmc_plt_clk_0"; + else + mclk_name = "pmc_plt_clk_3"; + + drv->mclk = devm_clk_get(&pdev->dev, mclk_name); if (IS_ERR(drv->mclk)) { - dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); + dev_err(&pdev->dev, "Failed to get MCLK from %s: %ld\n", + mclk_name, PTR_ERR(drv->mclk)); return PTR_ERR(drv->mclk); } -- 2.23.0.237.gc6a4ce50a0-goog

3 2

[alsa-devel] [PATCH V2 1/2] ASoC: fsl_mqs: add DT binding documentation
by Shengjiu Wang 01 Oct '19

01 Oct '19

Add the DT binding documentation for NXP MQS driver Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in v2 -refine the comments for properties .../devicetree/bindings/sound/fsl,mqs.txt | 36 +++++++++++++++++++ 1 file changed, 36 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,mqs.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.txt b/Documentation/devicetree/bindings/sound/fsl,mqs.txt new file mode 100644 index 000000000000..40353fc30255 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.txt @@ -0,0 +1,36 @@ +fsl,mqs audio CODEC + +Required properties: + - compatible : Must contain one of "fsl,imx6sx-mqs", "fsl,codec-mqs" + "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs". + - clocks : A list of phandles + clock-specifiers, one for each entry in + clock-names + - clock-names : "mclk" - must required. + "core" - required if compatible is "fsl,imx8qm-mqs", it + is for register access. + - gpr : A phandle of General Purpose Registers in IOMUX Controller. + Required if compatible is "fsl,imx6sx-mqs". + +Required if compatible is "fsl,imx8qm-mqs": + - power-domains: A phandle of PM domain provider node. + - reg: Offset and length of the register set for the device. + +Example: + +mqs: mqs { + compatible = "fsl,imx6sx-mqs"; + gpr = <&gpr>; + clocks = <&clks IMX6SX_CLK_SAI1>; + clock-names = "mclk"; + status = "disabled"; +}; + +mqs: mqs@59850000 { + compatible = "fsl,imx8qm-mqs"; + reg = <0x59850000 0x10000>; + clocks = <&clk IMX8QM_AUD_MQS_IPG>, + <&clk IMX8QM_AUD_MQS_HMCLK>; + clock-names = "core", "mclk"; + power-domains = <&pd_mqs0>; + status = "disabled"; +}; -- 2.21.0

4 6

[alsa-devel] [PATCH] ASoC: dt-bindings: mt8183: add missing update
by Tzung-Bi Shih 01 Oct '19

01 Oct '19

Headset codec is optional. Add missing update to DT binding document. Fixes: a962a809e5e4 ("ASoC: mediatek: mt8183: make headset codec optional") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../bindings/sound/mt8183-mt6358-ts3a227-max98357.txt | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt b/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt index d6d5207fa996..17ff3892f439 100644 --- a/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt +++ b/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt @@ -2,9 +2,11 @@ MT8183 with MT6358, TS3A227 and MAX98357 CODECS Required properties: - compatible : "mediatek,mt8183_mt6358_ts3a227_max98357" -- mediatek,headset-codec: the phandles of ts3a227 codecs - mediatek,platform: the phandle of MT8183 ASoC platform +Optional properties: +- mediatek,headset-codec: the phandles of ts3a227 codecs + Example: sound { -- 2.23.0.351.gc4317032e6-goog

2 1

[alsa-devel] [PATCH v2] ASoC: soc-component.h: remove GPL explanation from header
by Kuninori Morimoto 01 Oct '19

01 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc-component.h already has SPDX License, thus, GPL explanation is not needed. This patch removes it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark - this is v2 of "[PATCH] ASoC: soc-component.h: tidyup for soc-component.h" - Subject was renamed - keep "soc.h" - do nothing to header-check include/sound/soc-component.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5d80b2e..1a832695 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -3,10 +3,6 @@ * soc-component.h * * Copyright (c) 2019 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __SOC_COMPONENT_H #define __SOC_COMPONENT_H -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: xlnx: Use devm_platform_ioremap_resource() in xlnx_formatter_pcm_probe()
by Markus Elfring 01 Oct '19

01 Oct '19

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 19 Sep 2019 17:27:57 +0200 Simplify this function implementation by using a known wrapper function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/xilinx/xlnx_formatter_pcm.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 48970efe7838..fb652e73abeb 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -564,7 +564,6 @@ static int xlnx_formatter_pcm_probe(struct platform_device *pdev) int ret; u32 val; struct xlnx_pcm_drv_data *aud_drv_data; - struct resource *res; struct device *dev = &pdev->dev; aud_drv_data = devm_kzalloc(dev, sizeof(*aud_drv_data), GFP_KERNEL); @@ -584,13 +583,7 @@ static int xlnx_formatter_pcm_probe(struct platform_device *pdev) return ret; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(dev, "audio formatter node:addr to resource failed\n"); - ret = -ENXIO; - goto clk_err; - } - aud_drv_data->mmio = devm_ioremap_resource(dev, res); + aud_drv_data->mmio = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(aud_drv_data->mmio)) { dev_err(dev, "audio formatter ioremap failed\n"); ret = PTR_ERR(aud_drv_data->mmio); -- 2.23.0

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