November 2025 - Alsa-devel - mailman.alsa-project.org

[PATCH v1] ASoC: tas2781: correct the wrong period
by Shenghao Ding 22 Nov '25

22 Nov '25

A wrong preiod at the end of the sentence was reported by one of my customers. Their thorough code review is greatly appreciated. Fixes: 49e2e353fb0d ("ASoC: tas2781: Add Calibration Kcontrols for Chromebook") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 0a86964dfcfd..d1c76ab0144d 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -1400,7 +1400,7 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) /* * Alloc kcontrol via devm_kzalloc(), which don't manually - * free the kcontrol。 + * free the kcontrol. */ cali_ctrls = devm_kcalloc(priv->dev, nctrls, sizeof(cali_ctrls[0]), GFP_KERNEL); -- 2.43.0

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Re: [PATCH] add the SND quirk for HP pavilion aero laptop 13z-be200
by Takashi Iwai 21 Nov '25

21 Nov '25

On Fri, 21 Nov 2025 02:57:12 +0100, Jacob Zhong wrote: > > The laptop uses ALC287 chip (as shown in /proc/asound/card1/codec#0). > It seems that every HP pavilion laptop in the table uses the same quirk, > so I just copied them. I have verified that the mute LED on my laptop > works with this patch. > > For reference, here's the alsa-info of my laptop: > https://alsa-project.org/db/?f=2d5f297087708610bc01816ab12052abdd4a17c0 > > Signed-off-by: Jacob Zhong <cmpute(a)qq.com> Applied now. Thanks. Takashi

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[PATCH v7 1/2] ASoC: rt5575: Add the codec driver for the ALC5575
by Oder Chiou 21 Nov '25

21 Nov '25

The ALC5575 integrates an audio DSP that typically loads its firmware from an external flash via its own SPI host interface. In certain hardware configurations, the firmware can alternatively be loaded through the SPI client interface. The driver provides basic mute and volume control functions. When the SPI client interface is enabled, firmware loading is handled by the SPI driver. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/rt5575-spi.c | 86 ++++++++ sound/soc/codecs/rt5575-spi.h | 16 ++ sound/soc/codecs/rt5575.c | 370 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5575.h | 54 +++++ 6 files changed, 540 insertions(+) create mode 100644 sound/soc/codecs/rt5575-spi.c create mode 100644 sound/soc/codecs/rt5575-spi.h create mode 100644 sound/soc/codecs/rt5575.c create mode 100644 sound/soc/codecs/rt5575.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a0dfef57200c..a3ea5febd1e0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -211,6 +211,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT1305 imply SND_SOC_RT1308 imply SND_SOC_RT5514 + imply SND_SOC_RT5575 imply SND_SOC_RT5616 imply SND_SOC_RT5631 imply SND_SOC_RT5640 @@ -1767,6 +1768,15 @@ config SND_SOC_RT5514_SPI_BUILTIN bool # force RT5514_SPI to be built-in to avoid link errors default SND_SOC_RT5514=y && SND_SOC_RT5514_SPI=m +config SND_SOC_RT5575 + tristate "Realtek ALC5575 Codec - I2C" + depends on I2C + +config SND_SOC_RT5575_SPI + tristate "Realtek ALC5575 Codec - SPI" + depends on SPI_MASTER + select SND_SOC_RT5575 + config SND_SOC_RT5616 tristate "Realtek RT5616 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 39138d96a720..82f660cbe8ec 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -252,6 +252,8 @@ snd-soc-rt286-y := rt286.o snd-soc-rt298-y := rt298.o snd-soc-rt5514-y := rt5514.o snd-soc-rt5514-spi-y := rt5514-spi.o +snd-soc-rt5575-y := rt5575.o +snd-soc-rt5575-spi-y := rt5575-spi.o snd-soc-rt5616-y := rt5616.o snd-soc-rt5631-y := rt5631.o snd-soc-rt5640-y := rt5640.o @@ -684,6 +686,8 @@ obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o obj-$(CONFIG_SND_SOC_RT5514_SPI) += snd-soc-rt5514-spi.o obj-$(CONFIG_SND_SOC_RT5514_SPI_BUILTIN) += snd-soc-rt5514-spi.o +obj-$(CONFIG_SND_SOC_RT5575) += snd-soc-rt5575.o +obj-$(CONFIG_SND_SOC_RT5575_SPI) += snd-soc-rt5575-spi.o obj-$(CONFIG_SND_SOC_RT5616) += snd-soc-rt5616.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/rt5575-spi.c b/sound/soc/codecs/rt5575-spi.c new file mode 100644 index 000000000000..9dbc8170fb76 --- /dev/null +++ b/sound/soc/codecs/rt5575-spi.c @@ -0,0 +1,86 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * rt5575-spi.c -- ALC5575 SPI driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#include <linux/of.h> +#include <linux/spi/spi.h> + +#include "rt5575-spi.h" + +#define RT5575_SPI_CMD_BURST_WRITE 5 +#define RT5575_SPI_BUF_LEN 240 + +struct rt5575_spi_burst_write { + u8 cmd; + u32 addr; + u8 data[RT5575_SPI_BUF_LEN]; + u8 dummy; +} __packed; + +bool rt5575_spi_ready; +static struct spi_device *rt5575_spi; + +/** + * rt5575_spi_burst_write - Write data to SPI by rt5575 address. + * @addr: Start address. + * @txbuf: Data buffer for writing. + * @len: Data length. + * + */ +int rt5575_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) +{ + struct rt5575_spi_burst_write buf = { + .cmd = RT5575_SPI_CMD_BURST_WRITE + }; + unsigned int end, offset = 0; + + while (offset < len) { + if (offset + RT5575_SPI_BUF_LEN <= len) + end = RT5575_SPI_BUF_LEN; + else + end = len % RT5575_SPI_BUF_LEN; + + buf.addr = cpu_to_le32(addr + offset); + + memcpy(&buf.data, &txbuf[offset], end); + + spi_write(rt5575_spi, &buf, sizeof(buf)); + + offset += RT5575_SPI_BUF_LEN; + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5575_spi_burst_write); + +static int rt5575_spi_probe(struct spi_device *spi) +{ + rt5575_spi = spi; + + rt5575_spi_ready = true; + + return 0; +} + +static const struct of_device_id rt5575_of_match[] = { + { .compatible = "realtek,rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(of, rt5575_of_match); + +static struct spi_driver rt5575_spi_driver = { + .driver = { + .name = "rt5575", + .of_match_table = of_match_ptr(rt5575_of_match), + }, + .probe = rt5575_spi_probe, +}; +module_spi_driver(rt5575_spi_driver); + +MODULE_DESCRIPTION("ALC5575 SPI driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5575-spi.h b/sound/soc/codecs/rt5575-spi.h new file mode 100644 index 000000000000..cafe49a7ecc2 --- /dev/null +++ b/sound/soc/codecs/rt5575-spi.h @@ -0,0 +1,16 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5575-spi.h -- ALC5575 SPI driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#ifndef __RT5575_SPI_H__ +#define __RT5575_SPI_H__ + +extern bool rt5575_spi_ready; + +int rt5575_spi_burst_write(u32 addr, const u8 *txbuf, size_t len); + +#endif /* __RT5575_SPI_H__ */ diff --git a/sound/soc/codecs/rt5575.c b/sound/soc/codecs/rt5575.c new file mode 100644 index 000000000000..58508d643273 --- /dev/null +++ b/sound/soc/codecs/rt5575.c @@ -0,0 +1,370 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * rt5575.c -- ALC5575 ALSA SoC audio component driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#include <linux/firmware.h> +#include <linux/i2c.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "rt5575.h" +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) +#include "rt5575-spi.h" +#endif + +static bool rt5575_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5575_ID: + case RT5575_ID_1: + case RT5575_MIXL_VOL: + case RT5575_MIXR_VOL: + case RT5575_PROMPT_VOL: + case RT5575_SPK01_VOL: + case RT5575_SPK23_VOL: + case RT5575_MIC1_VOL: + case RT5575_MIC2_VOL: + case RT5575_WNC_CTRL: + case RT5575_MODE_CTRL: + case RT5575_I2S_RATE_CTRL: + case RT5575_SLEEP_CTRL: + case RT5575_ALG_BYPASS_CTRL: + case RT5575_PINMUX_CTRL_2: + case RT5575_GPIO_CTRL_1: + case RT5575_DSP_BUS_CTRL: + case RT5575_SW_INT: + case RT5575_DSP_BOOT_ERR: + case RT5575_DSP_READY: + case RT5575_DSP_CMD_ADDR: + case RT5575_EFUSE_DATA_2: + case RT5575_EFUSE_DATA_3: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(ob_tlv, -9525, 75, 0); + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) +static void rt5575_fw_load_by_spi(const struct firmware *fw, void *context) +{ + struct rt5575_priv *rt5575 = context; + struct i2c_client *i2c = rt5575->i2c; + const struct firmware *firmware; + static const char * const fw_path[] = { + "realtek/rt5575/rt5575_fw2.bin", + "realtek/rt5575/rt5575_fw3.bin", + "realtek/rt5575/rt5575_fw4.bin" + }; + static const u32 fw_addr[] = { 0x5f600000, 0x5f7fe000, 0x5f7ff000 }; + int i, ret; + + regmap_write(rt5575->dsp_regmap, 0xfafafafa, 0x00000004); + regmap_write(rt5575->dsp_regmap, 0x18008064, 0x00000000); + regmap_write(rt5575->dsp_regmap, 0x18008068, 0x0002ffff); + + rt5575_spi_burst_write(0x5f400000, fw->data, fw->size); + release_firmware(fw); + + for (i = 0; i < ARRAY_SIZE(fw_addr); i++) { + ret = request_firmware(&firmware, fw_path[i], &i2c->dev); + if (!ret) { + rt5575_spi_burst_write(fw_addr[i], firmware->data, firmware->size); + release_firmware(firmware); + } else { + dev_err(&i2c->dev, "Request firmware failure: %d\n", ret); + return; + } + } + + regmap_write(rt5575->dsp_regmap, 0x18000000, 0x00000000); + + regmap_update_bits(rt5575->regmap, RT5575_SW_INT, 1, 1); + + regmap_read_poll_timeout(rt5575->regmap, RT5575_SW_INT, ret, !ret, 100000, 10000000); + + if (ret) + dev_err(&i2c->dev, "Run firmware failure: %d\n", ret); +} +#endif + +static const struct snd_kcontrol_new rt5575_snd_controls[] = { + SOC_DOUBLE("Speaker CH-01 Playback Switch", RT5575_SPK01_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Speaker CH-01 Playback Volume", RT5575_SPK01_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Speaker CH-23 Playback Switch", RT5575_SPK23_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Speaker CH-23 Playback Volume", RT5575_SPK23_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Mic1 Capture Switch", RT5575_MIC1_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Mic1 Capture Volume", RT5575_MIC1_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Mic2 Capture Switch", RT5575_MIC2_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Mic2 Capture Volume", RT5575_MIC2_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE_R("Mix Playback Switch", RT5575_MIXL_VOL, RT5575_MIXR_VOL, 31, 1, 1), + SOC_DOUBLE_R_TLV("Mix Playback Volume", RT5575_MIXL_VOL, RT5575_MIXR_VOL, 1, 127, 0, + ob_tlv), + SOC_DOUBLE("Prompt Playback Switch", RT5575_PROMPT_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Prompt Playback Volume", RT5575_PROMPT_VOL, 17, 1, 167, 0, ob_tlv), +}; + +static const struct snd_soc_dapm_widget rt5575_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3RX", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3TX", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF4RX", "AIF4 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF4TX", "AIF4 Capture", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_INPUT("INPUT"), + SND_SOC_DAPM_OUTPUT("OUTPUT"), +}; + +static const struct snd_soc_dapm_route rt5575_dapm_routes[] = { + { "AIF1TX", NULL, "INPUT" }, + { "AIF2TX", NULL, "INPUT" }, + { "AIF3TX", NULL, "INPUT" }, + { "AIF4TX", NULL, "INPUT" }, + { "OUTPUT", NULL, "AIF1RX" }, + { "OUTPUT", NULL, "AIF2RX" }, + { "OUTPUT", NULL, "AIF3RX" }, + { "OUTPUT", NULL, "AIF4RX" }, +}; + +static long long rt5575_get_priv_id(struct rt5575_priv *rt5575) +{ + int priv_id_low, priv_id_high; + + regmap_write(rt5575->regmap, RT5575_EFUSE_PID, 0xa0000000); + regmap_read(rt5575->regmap, RT5575_EFUSE_DATA_2, &priv_id_low); + regmap_read(rt5575->regmap, RT5575_EFUSE_DATA_3, &priv_id_high); + regmap_write(rt5575->regmap, RT5575_EFUSE_PID, 0); + + return ((long long)priv_id_high << 32) | (long long)priv_id_low; +} + +static int rt5575_probe(struct snd_soc_component *component) +{ + struct rt5575_priv *rt5575 = snd_soc_component_get_drvdata(component); + + rt5575->component = component; + + dev_info(component->dev, "Private ID: %llx\n", rt5575_get_priv_id(rt5575)); + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) + request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, "realtek/rt5575/rt5575_fw1.bin", + component->dev, GFP_KERNEL, rt5575, rt5575_fw_load_by_spi); +#endif + + return 0; +} + +#define RT5575_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5575_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver rt5575_dai[] = { + { + .name = "rt5575-aif1", + .id = RT5575_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif2", + .id = RT5575_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif3", + .id = RT5575_AIF3, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif4", + .id = RT5575_AIF4, + .playback = { + .stream_name = "AIF4 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF4 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, +}; + +static const struct snd_soc_component_driver rt5575_soc_component_dev = { + .probe = rt5575_probe, + .controls = rt5575_snd_controls, + .num_controls = ARRAY_SIZE(rt5575_snd_controls), + .dapm_widgets = rt5575_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5575_dapm_widgets), + .dapm_routes = rt5575_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5575_dapm_routes), + .use_pmdown_time = 1, + .endianness = 1, +}; + +static const struct regmap_config rt5575_dsp_regmap = { + .name = "dsp", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 2, +}; + +static int rt5575_i2c_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct rt5575_priv *rt5575 = i2c_get_clientdata(client); + + regmap_read(rt5575->dsp_regmap, reg | RT5575_DSP_MAPPING, val); + + return 0; +} + +static int rt5575_i2c_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + struct rt5575_priv *rt5575 = i2c_get_clientdata(client); + + regmap_write(rt5575->dsp_regmap, reg | RT5575_DSP_MAPPING, val); + + return 0; +} + +static const struct regmap_config rt5575_regmap = { + .reg_bits = 16, + .val_bits = 32, + .reg_stride = 4, + .max_register = 0xfffc, + .readable_reg = rt5575_readable_register, + .reg_read = rt5575_i2c_read, + .reg_write = rt5575_i2c_write, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct i2c_device_id rt5575_i2c_id[] = { + { "rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5575_i2c_id); + +static int rt5575_i2c_probe(struct i2c_client *i2c) +{ + struct rt5575_priv *rt5575; + int ret, val; + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) + if (!rt5575_spi_ready) + return -EPROBE_DEFER; +#endif + + rt5575 = devm_kzalloc(&i2c->dev, sizeof(struct rt5575_priv), + GFP_KERNEL); + if (rt5575 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5575); + + rt5575->i2c = i2c; + + rt5575->dsp_regmap = devm_regmap_init_i2c(i2c, &rt5575_dsp_regmap); + if (IS_ERR(rt5575->dsp_regmap)) { + ret = PTR_ERR(rt5575->dsp_regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); + return ret; + } + + rt5575->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt5575_regmap); + if (IS_ERR(rt5575->regmap)) { + ret = PTR_ERR(rt5575->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); + return ret; + } + + regmap_read(rt5575->regmap, RT5575_ID, &val); + if (val != RT5575_DEVICE_ID) { + dev_err(&i2c->dev, "Device with ID register %08x is not rt5575\n", val); + return -ENODEV; + } + + regmap_read(rt5575->regmap, RT5575_ID_1, &val); + if (!val) { + dev_err(&i2c->dev, "This is not formal version\n"); + return -ENODEV; + } + + return devm_snd_soc_register_component(&i2c->dev, &rt5575_soc_component_dev, rt5575_dai, + ARRAY_SIZE(rt5575_dai)); +} + +static const struct of_device_id rt5575_of_match[] = { + { .compatible = "realtek,rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(of, rt5575_of_match); + +static struct i2c_driver rt5575_i2c_driver = { + .driver = { + .name = "rt5575", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(rt5575_of_match), + }, + .probe = rt5575_i2c_probe, + .id_table = rt5575_i2c_id, +}; +module_i2c_driver(rt5575_i2c_driver); + +MODULE_DESCRIPTION("ASoC ALC5575 driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5575.h b/sound/soc/codecs/rt5575.h new file mode 100644 index 000000000000..11149612274a --- /dev/null +++ b/sound/soc/codecs/rt5575.h @@ -0,0 +1,54 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5575.h -- ALC5575 ALSA SoC audio driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#ifndef __RT5575_H__ +#define __RT5575_H__ + +#define RT5575_DEVICE_ID 0x10ec5575 +#define RT5575_DSP_MAPPING 0x18000000 + +#define RT5575_ID 0x8008 +#define RT5575_ID_1 0x800c +#define RT5575_MIXL_VOL 0x8a14 +#define RT5575_MIXR_VOL 0x8a18 +#define RT5575_PROMPT_VOL 0x8a84 +#define RT5575_SPK01_VOL 0x8a88 +#define RT5575_SPK23_VOL 0x8a8c +#define RT5575_MIC1_VOL 0x8a98 +#define RT5575_MIC2_VOL 0x8a9c +#define RT5575_WNC_CTRL 0x80ec +#define RT5575_MODE_CTRL 0x80f0 +#define RT5575_I2S_RATE_CTRL 0x80f4 +#define RT5575_SLEEP_CTRL 0x80f8 +#define RT5575_ALG_BYPASS_CTRL 0x80fc +#define RT5575_PINMUX_CTRL_2 0x81a4 +#define RT5575_GPIO_CTRL_1 0x8208 +#define RT5575_DSP_BUS_CTRL 0x880c +#define RT5575_SW_INT 0x0018 +#define RT5575_DSP_BOOT_ERR 0x8e14 +#define RT5575_DSP_READY 0x8e24 +#define RT5575_DSP_CMD_ADDR 0x8e28 +#define RT5575_EFUSE_DATA_2 0xc638 +#define RT5575_EFUSE_DATA_3 0xc63c +#define RT5575_EFUSE_PID 0xc660 + +enum { + RT5575_AIF1, + RT5575_AIF2, + RT5575_AIF3, + RT5575_AIF4, + RT5575_AIFS, +}; + +struct rt5575_priv { + struct i2c_client *i2c; + struct snd_soc_component *component; + struct regmap *dsp_regmap, *regmap; +}; + +#endif /* __RT5575_H__ */ -- 2.51.2

1 1

SplitPCMDevice macro broken with two or more channels
by GitHub issues - opened 21 Nov '25

21 Nov '25

alsa-project/alsa-ucm-conf issue #646 was opened from karcsesz: I have a Behringer XR18 that I am currently trying to write a custom UCM profile for that splits its 18x18 audio interface into separate mono and stereo virtual devices. I am able to create mono devices using the `SplitPCM` and `SplitPCMDevice` macros, however only when none of the channels higher than Channel0 are defined, as the created Capture/PlaybackPCM configuration values have the channel IDs duplicated: ``` $ alsaucm -c3 dump text [snip] Verb."18x18 Stereo" { Comment "9x9 Stereo Duplex" Device."Ch1 In" { Comment "Ch1 from" Values { CaptureCTL "_ucm0001.hw:XR18" CaptureChannels 2 CapturePCM "_ucm0001.behringer_xr18_stereo_in:XR18,0,0,1,1" PlaybackCTL "_ucm0001.hw:XR18" } } Device."Ch2 In" { Comment "Ch2 from" Values { CaptureCTL "_ucm0001.hw:XR18" CaptureChannels 2 CapturePCM "_ucm0001.behringer_xr18_stereo_in:XR18,0,2,3,3" PlaybackCTL "_ucm0001.hw:XR18" } } [snip] ``` `spa-acp-tool` also fails to find the PCM devices due to this issue ``` $ spa-acp-tool -vvv -c3 [snip] I alsa-ucm.c:2567 Probing profile 18x18 Stereo I alsa-ucm.c:2571 Set ucm verb to 18x18 Stereo D alsa-util.c:714 Trying _ucm0001.behringer_xr18_stereo_in:XR18,0,16,17,17 with SND_PCM_NO_AUTO_FORMAT ... I conf.c:5541 Unknown parameter 4 I conf.c:5712 Parse arguments error: No such file or directory I pcm.c:2722 Unknown PCM behringer_xr18_stereo_in:XR18,0,16,17,17 I alsa-util.c:721 Error opening PCM device _ucm0001.behringer_xr18_stereo_in:XR18,0,16,17,17: No such file or directory I alsa-ucm.c:2611 Profile '18x18 Stereo' mapping '18x18 Stereo: Ch9 In: source': input PCM open failed I alsa-ucm.c:2621 Profile 18x18 Stereo not supported ``` Appropriate snippet from the verb configuration: ``` Include.pcm_split.File "/common/pcm/split.conf" Macro [ [snip] { SplitPCM { Name "behringer_xr18_stereo_in" Direction Capture Format S24_3LE Channels 2 HWChannels 18 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR HWChannelPos6 FL HWChannelPos7 FR HWChannelPos8 FL HWChannelPos9 FR HWChannelPos10 FL HWChannelPos11 FR HWChannelPos12 FL HWChannelPos13 FR HWChannelPos14 FL HWChannelPos15 FR HWChannelPos16 FL HWChannelPos17 FR } } [snip] ] [snip] SectionDevice."Ch1 In" { Comment "Ch1 from" Macro.pcm_split.SplitPCMDevice { Name "behringer_xr18_stereo_in" Direction Capture HwChannels 18 Channels 2 Channel0 0 Channel1 1 ChannelPos0 FL ChannelPos1 FR } } SectionDevice."Ch2 In" { Comment "Ch2 from" Macro.pcm_split.SplitPCMDevice { Name "behringer_xr18_stereo_in" Direction Capture HwChannels 18 Channels 2 Channel0 2 Channel1 3 ChannelPos0 FL ChannelPos1 FR } } [snip] ``` My current hunch is the append within the `SplitPCMDevice_addchn` macro somehow getting called twice (maybe due to [evaluation restart](https://www.alsa-project.org/alsa-doc/alsa-lib/group__ucm__conf.ht…) [alsa-info.txt](https://github.com/user-attachments/files/23665421/alsa-info… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/646 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1 1/8] ASoC: tas2783A: sdw_utils: support ch 3 & 4
by Niranjan H Y 20 Nov '25

20 Nov '25

Currently the machine driver for tas2783A can only support 2 channels. This patch adds support for 2 channel playback with 4 device setup. Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> --- sound/soc/sdw_utils/soc_sdw_ti_amp.c | 4 ++++ sound/soc/sdw_utils/soc_sdw_utils.c | 22 ++++++++++++++++++---- 2 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/sdw_utils/soc_sdw_ti_amp.c b/sound/soc/sdw_utils/soc_sdw_ti_amp.c index cbd60faec..488ef2ef4 100644 --- a/sound/soc/sdw_utils/soc_sdw_ti_amp.c +++ b/sound/soc/sdw_utils/soc_sdw_ti_amp.c @@ -58,6 +58,10 @@ int asoc_sdw_ti_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, strscpy(speaker, "Left Spk", sizeof(speaker)); } else if (!strncmp(prefix, "tas2783-2", strlen("tas2783-2"))) { strscpy(speaker, "Right Spk", sizeof(speaker)); + } else if (!strncmp(prefix, "tas2783-3", strlen("tas2783-3"))) { + strscpy(speaker, "Left Spk2", sizeof(speaker)); + } else if (!strncmp(prefix, "tas2783-4", strlen("tas2783-4"))) { + strscpy(speaker, "Right Spk2", sizeof(speaker)); } else { ret = -EINVAL; dev_err(card->dev, "unhandled prefix %s", prefix); diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index 824fb613c..ec00f13a6 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -40,11 +40,25 @@ static const struct snd_soc_dapm_widget lr_spk_widgets[] = { SND_SOC_DAPM_SPK("Right Spk", NULL), }; +static const struct snd_soc_dapm_widget lr_4spk_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_SPK("Left Spk2", NULL), + SND_SOC_DAPM_SPK("Right Spk2", NULL), +}; + static const struct snd_kcontrol_new lr_spk_controls[] = { SOC_DAPM_PIN_SWITCH("Left Spk"), SOC_DAPM_PIN_SWITCH("Right Spk"), }; +static const struct snd_kcontrol_new lr_4spk_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), + SOC_DAPM_PIN_SWITCH("Left Spk2"), + SOC_DAPM_PIN_SWITCH("Right Spk2"), +}; + static const struct snd_soc_dapm_widget rt700_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_MIC("AMIC", NULL), @@ -69,10 +83,10 @@ struct asoc_sdw_codec_info codec_info_list[] = { .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, .init = asoc_sdw_ti_amp_init, .rtd_init = asoc_sdw_ti_spk_rtd_init, - .controls = lr_spk_controls, - .num_controls = ARRAY_SIZE(lr_spk_controls), - .widgets = lr_spk_widgets, - .num_widgets = ARRAY_SIZE(lr_spk_widgets), + .controls = lr_4spk_controls, + .num_controls = ARRAY_SIZE(lr_4spk_controls), + .widgets = lr_4spk_widgets, + .num_widgets = ARRAY_SIZE(lr_4spk_widgets), }, }, .dai_num = 1, -- 2.25.1

2 8

Support Tas2781 and mute led for HP platform
by Kailang 20 Nov '25

20 Nov '25

Hi Takashi, Attach patch was supported Tas2781 and mute led for HP platform. BR, Kailang

2 2

[PATCH] ASoC: wm8994: Prevent access to invalid VU register bits on WM1811
by Sylwester Nawrocki 20 Nov '25

20 Nov '25

The ADC2 and DAC2 are not available on WM1811 device. This patch moves the ADC2, DAC2 VU bitfields to a separate array so we can skip accessing them and avoid unreadable register access on WM1811. This allows to get rid of warnings during boot like: wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5 Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/codecs/wm8994.c | 60 ++++++++++++++++++++++++++++----------- 1 file changed, 44 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a84ae879d37e..038be667c1a6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -43,10 +43,12 @@ #define WM8994_NUM_DRC 3 #define WM8994_NUM_EQ 3 -static struct { +struct wm8994_reg_mask { unsigned int reg; unsigned int mask; -} wm8994_vu_bits[] = { +}; + +static struct wm8994_reg_mask wm8994_vu_bits[] = { { WM8994_LEFT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, { WM8994_RIGHT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, { WM8994_LEFT_LINE_INPUT_3_4_VOLUME, WM8994_IN2_VU }, @@ -60,14 +62,10 @@ static struct { { WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1DAC1_VU }, { WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU }, - { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, - { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, { WM8994_AIF2_DAC_LEFT_VOLUME, WM8994_AIF2DAC_VU }, { WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU }, { WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1ADC1_VU }, { WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU }, - { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, - { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, { WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2ADC_VU }, { WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, { WM8994_DAC1_LEFT_VOLUME, WM8994_DAC1_VU }, @@ -76,6 +74,14 @@ static struct { { WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU }, }; +/* VU bitfields for ADC2, DAC2 not available on WM1811 */ +static struct wm8994_reg_mask wm8994_adc2_dac2_vu_bits[] = { + { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, + { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, +}; + static int wm8994_drc_base[] = { WM8994_AIF1_DRC1_1, WM8994_AIF1_DRC2_1, @@ -1030,6 +1036,26 @@ static bool wm8994_check_class_w_digital(struct snd_soc_component *component) return true; } +static void wm8994_update_vu_bits(struct snd_soc_component *component) +{ + struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); + struct wm8994 *control = wm8994->wm8994; + int i; + + for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) + snd_soc_component_write(component, wm8994_vu_bits[i].reg, + snd_soc_component_read(component, + wm8994_vu_bits[i].reg)); + if (control->type == WM1811) + return; + + for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++) + snd_soc_component_write(component, + wm8994_adc2_dac2_vu_bits[i].reg, + snd_soc_component_read(component, + wm8994_adc2_dac2_vu_bits[i].reg)); +} + static int aif_mclk_set(struct snd_soc_component *component, int aif, bool enable) { struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); @@ -1076,7 +1102,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); struct wm8994 *control = wm8994->wm8994; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; - int ret, i; + int ret; int dac; int adc; int val; @@ -1144,10 +1170,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) - snd_soc_component_write(component, wm8994_vu_bits[i].reg, - snd_soc_component_read(component, - wm8994_vu_bits[i].reg)); + wm8994_update_vu_bits(component); break; case SND_SOC_DAPM_PRE_PMD: @@ -1181,7 +1204,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - int ret, i; + int ret; int dac; int adc; int val; @@ -1237,10 +1260,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) - snd_soc_component_write(component, wm8994_vu_bits[i].reg, - snd_soc_component_read(component, - wm8994_vu_bits[i].reg)); + wm8994_update_vu_bits(component); break; case SND_SOC_DAPM_PRE_PMD: @@ -4346,6 +4366,14 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994_vu_bits[i].mask, wm8994_vu_bits[i].mask); + if (control->type != WM1811) { + for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++) + snd_soc_component_update_bits(component, + wm8994_adc2_dac2_vu_bits[i].reg, + wm8994_adc2_dac2_vu_bits[i].mask, + wm8994_adc2_dac2_vu_bits[i].mask); + } + /* Set the low bit of the 3D stereo depth so TLV matches */ snd_soc_component_update_bits(component, WM8994_AIF1_DAC1_FILTERS_2, 1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT, -- 2.17.1

4 3

No soundcards recognised by ALSA after upgrading to Linux 6.17.8 from Linux 6.17.1 (Lenovo Yoga Slim 7 14ILL10, cs42l43)
by GitHub issues - opened 20 Nov '25

20 Nov '25

alsa-project/alsa-ucm-conf issue #645 was opened from emfol: After upgrading to Linux 6.17.8 (from Linux 6.17.1), the sound card is no longer recognized. I ran the `alsa-info.sh` script after booting each one of the kernels and the results follow: - 6.17.1: https://alsa-project.org/db/?f=5887d318c6cd7866365a0f134eeaf5721f33b82d - 6.17.8: https://alsa-project.org/db/?f=a5ca95f07b9a1844c102f63e6c0e36cd2e5755ef #### DMESG 6.17.8 ``` Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: SoundWire enabled on CannonLake+ platform, using SOF driver Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: enabling device (0000 -> 0002) Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if 0x040100 Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: bound 0000:00:02.0 (ops intel_audio_component_bind_ops [xe]) Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: use msi interrupt mode Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: hda codecs found, mask 4 Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: NHLT device BT(0) detected, ssp_mask 0x4 Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: BT link detected in NHLT tables: 0x4 Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: DMICs detected in NHLT tables: 4 Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Firmware paths/files for ipc type 1: Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Firmware file: intel/sof-ipc4/lnl/sof-lnl.ri Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Firmware lib path: intel/sof-ipc4-lib/lnl Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Topology file: intel/sof-ipc4-tplg/sof-lnl-cs42l43-l0-4ch.tplg Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Loaded firmware library: ADSPFW, version: 2.13.0.1 Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Booted firmware version: 2.13.0.1 ``` #### DMESG 6.17.1 ``` Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: SoundWire enabled on CannonLake+ platform, using SOF driver Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: enabling device (0000 -> 0002) Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if 0x040100 Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: bound 0000:00:02.0 (ops intel_audio_component_bind_ops [xe]) Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: use msi interrupt mode Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: hda codecs found, mask 4 Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: NHLT device BT(0) detected, ssp_mask 0x4 Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: BT link detected in NHLT tables: 0x4 Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: DMICs detected in NHLT tables: 4 Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Firmware paths/files for ipc type 1: Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Firmware file: intel/sof-ipc4/lnl/sof-lnl.ri Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Firmware lib path: intel/sof-ipc4-lib/lnl Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Topology file: intel/sof-ipc4-tplg/sof-lnl-cs42l43-l0-4ch.tplg Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Loaded firmware library: ADSPFW, version: 2.13.0.1 Nov 19 22:22:15 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Booted firmware version: 2.13.0.1 Nov 19 22:22:15 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Topology: ABI 3:29:1 Kernel ABI 3:23:1 Nov 19 22:22:15 ys7i14-em kernel: sof_sdw sof_sdw: ASoC: Parent card not yet available, widget card binding deferred Nov 19 22:22:15 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Loaded firmware library: ADSPFW, version: 2.13.0.1 Nov 19 22:22:21 ys7i14-em kernel: sof_sdw sof_sdw: Setting CS42L43 Speaker volume limit to 128 Nov 19 22:22:21 ys7i14-em kernel: sof_sdw sof_sdw: hda_dsp_hdmi_build_controls: no PCM in topology for HDMI converter 3 Nov 19 22:22:21 ys7i14-em kernel: input: sof-soundwire Jack as /devices/pci0000:00/0000:00:1f.3/sof_sdw/sound/card0/input14 Nov 19 22:22:21 ys7i14-em kernel: input: sof-soundwire HDMI/DP,pcm=5 as /devices/pci0000:00/0000:00:1f.3/sof_sdw/sound/card0/input15 Nov 19 22:22:21 ys7i14-em kernel: input: sof-soundwire HDMI/DP,pcm=6 as /devices/pci0000:00/0000:00:1f.3/sof_sdw/sound/card0/input16 Nov 19 22:22:21 ys7i14-em kernel: input: sof-soundwire HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:1f.3/sof_sdw/sound/card0/input17 Nov 19 22:33:07 ys7i14-em systemd[1]: sys-devices-pci0000:00-0000:00:1f.3-sof_sdw-sound-card0-controlC0.device: Failed to enqueue SYSTEMD_WANTS job, ignoring: Transaction for sound.target/start is destructive (reboot.target has 'start' job queued, but 'stop' is included in transaction). ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/645 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: sof-soundwire: Update cs42l45 JackControls
by GitHub pull_request - opened 19 Nov '25

19 Nov '25

alsa-project/alsa-ucm-conf pull request #644 was opened from mstrozek: Update JackControls to follow machine driver changes [1] [1] https://github.com/thesofproject/linux/pull/5590 Keeping this PR as a draft until the SOF PR is finalized and merged Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/644 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/644.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

HDA: route speaker volume to Post Mixer Analog
by GitHub pull_request - opened 19 Nov '25

19 Nov '25

alsa-project/alsa-ucm-conf pull request #643 was opened from Lyapsus: On certain laptops (e.g., Lenovo Legion Pro 7 16IAX10H), the live speaker gain is exposed through the Post Mixer Analog controls. The default UCM entries point at Speaker/Master, so software volume never touches the woofer amps. Routing the Speaker device to the Post Mixer Analog mixer and volume elements restores proper volume control without affecting other paths Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/643 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/643.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

fix typo in error.h to avoid compile error when gcc <= 2.95
by GitHub pull_request - opened 19 Nov '25

19 Nov '25

alsa-project/alsa-lib pull request #483 was opened from versaloon: a typo in include/error.h Request URL : https://github.com/alsa-project/alsa-lib/pull/483 Patch URL : https://github.com/alsa-project/alsa-lib/pull/483.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Lenovo Yoga Pro 7 built-in speakers volume not controlled by GNOME/PipeWire
by GitHub issues - edited 18 Nov '25

18 Nov '25

alsa-project/alsa-ucm-conf issue #642 was edited from 2O77: Device: Lenovo Yoga Pro 7 14IRH8 Sound card: sof-hda-dsp Problem: ALSA Post-Mixer / Pre-Mixer controls work via alsamixer, but GNOME volume slider or PipeWire sink volume does not affect hardware output. Likely an issue in the UCM profile for this device. Steps to reproduce: 1. Boot any Linux distribution. 2. Attempt to change volume via GNOME/KDE slider or `pactl` / PipeWire controls. 3. Observe that output volume of laptop speakers does not change. 4. Testing with headphones (AirPods, Beats) or external speakers works correctly. Notes: - This issue is specific to the built-in laptop speakers. - Different desktop environments (GNOME, KDE) and distributions (Fedora GNOME, Fedora KDE, openSUSE Tumbleweed) were tested and exhibit the same problem. - ALSA `Post-Mixer` / `Pre-Mixer` controls work correctly when adjusted via `alsamixer`. Expected behavior: - GNOME / PipeWire slider should affect the actual hardware output volume of the laptop speakers. Suggested area: - UCM profile for sof-hda-dsp on this Lenovo model may need adjustment. I first create an issue on pipewire and they directed me here with this outcome: Which mixer controls are used for the volume is controlled by the card UCM profile, so this is probably bug in the UCM profile. Its issue tracker is here: https://github.com/alsa-project/alsa-ucm-conf/issues Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/642 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo Yoga Pro 7 built-in speakers volume not controlled by GNOME/PipeWire
by GitHub issues - edited 18 Nov '25

18 Nov '25

alsa-project/alsa-ucm-conf issue #642 was edited from 2O77: Device: Lenovo Yoga Pro 7 14IRH8 Sound card: sof-hda-dsp Problem: ALSA Post-Mixer / Pre-Mixer controls work via alsamixer, but GNOME volume slider or PipeWire sink volume does not affect hardware output. Likely an issue in the UCM profile for this device. Steps to reproduce: 1. Boot any Linux distribution. 2. Attempt to change volume via GNOME/KDE slider or `pactl` / PipeWire controls. 3. Observe that output volume of laptop speakers does not change. 4. Testing with headphones (AirPods, Beats) or external speakers works correctly. Notes: - This issue is specific to the built-in laptop speakers. - Different desktop environments (GNOME, KDE) and distributions (Fedora GNOME, Fedora KDE, openSUSE Tumbleweed) were tested and exhibit the same problem. - ALSA `Post-Mixer` / `Pre-Mixer` controls work correctly when adjusted via `alsamixer`. Expected behavior: - GNOME / PipeWire slider should affect the actual hardware output volume of the laptop speakers. Suggested area: - UCM profile for sof-hda-dsp on this Lenovo model may need adjustment. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/642 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo Yoga Pro 7 built-in speakers volume not controlled by GNOME/PipeWire
by GitHub issues - edited 18 Nov '25

18 Nov '25

alsa-project/alsa-ucm-conf issue #642 was edited from 2O77: Device: Lenovo Yoga Pro 14 14IMH9 Sound card: sof-hda-dsp Problem: ALSA Post-Mixer / Pre-Mixer controls work via alsamixer, but GNOME volume slider or PipeWire sink volume does not affect hardware output. Likely an issue in the UCM profile for this device. Steps to reproduce: 1. Boot any Linux distribution. 2. Attempt to change volume via GNOME/KDE slider or `pactl` / PipeWire controls. 3. Observe that output volume of laptop speakers does not change. 4. Testing with headphones (AirPods, Beats) or external speakers works correctly. Notes: - This issue is specific to the built-in laptop speakers. - Different desktop environments (GNOME, KDE) and distributions (Fedora GNOME, Fedora KDE, openSUSE Tumbleweed) were tested and exhibit the same problem. - ALSA `Post-Mixer` / `Pre-Mixer` controls work correctly when adjusted via `alsamixer`. Expected behavior: - GNOME / PipeWire slider should affect the actual hardware output volume of the laptop speakers. Suggested area: - UCM profile for sof-hda-dsp on this Lenovo model may need adjustment. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/642 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo Yoga Pro 14 built-in speakers volume not controlled by GNOME/PipeWire
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-ucm-conf issue #642 was edited from 2O77: Device: Lenovo Yoga Pro 14 14IMH9 Sound card: sof-hda-dsp Problem: ALSA Post-Mixer / Pre-Mixer controls work via alsamixer, but GNOME volume slider or PipeWire sink volume does not affect hardware output. Likely an issue in the UCM profile for this device. Steps to reproduce: 1. Boot any Linux distribution. 2. Attempt to change volume via GNOME/KDE slider or `pactl` / PipeWire controls. 3. Observe that output volume of laptop speakers does not change. 4. Testing with headphones (AirPods, Beats) or external speakers works correctly. Notes: - This issue is specific to the built-in laptop speakers. - Different desktop environments (GNOME, KDE) and distributions (Fedora GNOME, Fedora KDE, openSUSE Tumbleweed) were tested and exhibit the same problem. - ALSA `Post-Mixer` / `Pre-Mixer` controls work correctly when adjusted via `alsamixer`. Expected behavior: - GNOME / PipeWire slider should affect the actual hardware output volume of the laptop speakers. Suggested area: - UCM profile for sof-hda-dsp on this Lenovo model may need adjustment. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/642 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo Yoga Pro 14 built-in speakers volume not controlled by GNOME/PipeWire
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-ucm-conf issue #642 was edited from 2O77: Device: Lenovo Yoga Pro 14 14IMH9 Sound card: sof-hda-dsp Problem: ALSA Post-Mixer / Pre-Mixer controls work via alsamixer, but GNOME volume slider or PipeWire sink volume does not affect hardware output. Likely an issue in the UCM profile for this device. Steps to reproduce: 1. Boot any Linux distribution. 2. Attempt to change volume via GNOME slider or `pactl` / PipeWire controls. 3. Observe that output volume of laptop speakers does not change. 4. Testing with headphones (AirPods, Beats) or external speakers works correctly. Notes: - This issue is specific to the built-in laptop speakers. - Different desktop environments (GNOME, KDE) and distributions (Fedora GNOME, Fedora KDE, openSUSE Tumbleweed) were tested and exhibit the same problem. - ALSA `Post-Mixer` / `Pre-Mixer` controls work correctly when adjusted via `alsamixer`. Expected behavior: - GNOME / PipeWire slider should affect the actual hardware output volume of the laptop speakers. Suggested area: - UCM profile for sof-hda-dsp on this Lenovo model may need adjustment. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/642 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo Yoga Pro 14 built-in speakers volume not controlled by GNOME/PipeWire
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-ucm-conf issue #642 was edited from 2O77: Device: Lenovo Yoga Pro 14 14IMH9 Sound card: sof-hda-dsp Problem: ALSA Post-Mixer / Pre-Mixer controls work via alsamixer, but GNOME volume slider or PipeWire sink volume does not affect hardware output. Likely an issue in the UCM profile for this device. Steps to reproduce: 1. Boot any Linux distribution. I tested on Fedora (GNOME and KDE) and openSUSE Tumbleweed, but it will probably occur on other distros or desktop environments as well. 2. Attempt to change volume via GNOME slider or `pactl` / PipeWire controls. 3. Observe that output volume of laptop speakers does not change. 4. Testing with headphones (AirPods, Beats) or external speakers works correctly. Notes: - This issue is specific to the built-in laptop speakers. - Different desktop environments (GNOME, KDE) and distributions (Fedora GNOME, Fedora KDE, openSUSE Tumbleweed) were tested and exhibit the same problem. - ALSA `Post-Mixer` / `Pre-Mixer` controls work correctly when adjusted via `alsamixer`. Expected behavior: - GNOME / PipeWire slider should affect the actual hardware output volume of the laptop speakers. Suggested area: - UCM profile for sof-hda-dsp on this Lenovo model may need adjustment. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/642 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo Yoga Pro 14 built-in speakers volume not controlled by GNOME/PipeWire
by GitHub issues - opened 17 Nov '25

17 Nov '25

alsa-project/alsa-ucm-conf issue #642 was opened from 2O77: Device: Lenovo Yoga Pro 14 14IMH9 Sound card: sof-hda-dsp Problem: ALSA Post-Mixer / Pre-Mixer controls work via alsamixer, but GNOME volume slider or PipeWire sink volume does not affect hardware output. Likely an issue in the UCM profile for this device. Steps to reproduce: 1. Boot Fedora (GNOME or KDE) or openSUSE Tumbleweed. 2. Attempt to change volume via GNOME slider or `pactl` / PipeWire controls. 3. Observe that output volume of laptop speakers does not change. 4. Testing with headphones (AirPods, Beats) or external speakers works correctly. Notes: - This issue is specific to the built-in laptop speakers. - Different desktop environments (GNOME, KDE) and distributions (Fedora GNOME, Fedora KDE, openSUSE Tumbleweed) were tested and exhibit the same problem. - ALSA `Post-Mixer` / `Pre-Mixer` controls work correctly when adjusted via `alsamixer`. Expected behavior: - GNOME / PipeWire slider should affect the actual hardware output volume of the laptop speakers. Suggested area: - UCM profile for sof-hda-dsp on this Lenovo model may need adjustment. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/642 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v2] ASoC: SDCA: bug fix while parsing mipi-sdca-control-cn-list
by Niranjan H Y 17 Nov '25

17 Nov '25

"struct sdca_control" declares "values" field as integer array. But the memory allocated to it is of char array. This causes crash for sdca_parse_function API. This patch addresses the issue by allocating correct data size. Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> --- v2: - use kcalloc instead of kzalloc --- sound/soc/sdca/sdca_functions.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sdca/sdca_functions.c b/sound/soc/sdca/sdca_functions.c index b2e3fab9b..49b98fe2d 100644 --- a/sound/soc/sdca/sdca_functions.c +++ b/sound/soc/sdca/sdca_functions.c @@ -950,7 +950,8 @@ static int find_sdca_entity_control(struct device *dev, struct sdca_entity *enti return ret; } - control->values = devm_kzalloc(dev, hweight64(control->cn_list), GFP_KERNEL); + control->values = devm_kcalloc(dev, hweight64(control->cn_list), + sizeof(int), GFP_KERNEL); if (!control->values) return -ENOMEM; -- 2.25.1

4 3

[PATCH 5.10] ASoC: SOF: Intel: hda: Fix potential buffer overflow by snprintf()
by Alexandra Diupina 17 Nov '25

17 Nov '25

From: Takashi Iwai <tiwai(a)suse.de> commit 94c1ceb043c1a002de9649bb630c8e8347645982 upstream. snprintf() returns the would-be-filled size when the string overflows the given buffer size, hence using this value may result in the buffer overflow (although it's unrealistic). This patch replaces with a safer version, scnprintf() for papering over such a potential issue. Fixes: 29c8e4398f02 ("ASoC: SOF: Intel: hda: add extended rom status dump to error log") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20220801165420.25978-4-tiwai@suse.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Alexandra Diupina <adiupina(a)astralinux.ru> --- Backport fix for CVE-2022-50050 sound/soc/sof/intel/hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index b4cc72483137..1d879c2b81e1 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -437,7 +437,7 @@ static void hda_dsp_dump_ext_rom_status(struct snd_sof_dev *sdev) for (i = 0; i < HDA_EXT_ROM_STATUS_SIZE; i++) { value = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_ROM_STATUS + i * 0x4); - len += snprintf(msg + len, sizeof(msg) - len, " 0x%x", value); + len += scnprintf(msg + len, sizeof(msg) - len, " 0x%x", value); } sof_dev_dbg_or_err(sdev->dev, hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS, -- 2.30.2

1 0

snd-loopback does not work
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-lib issue #482 was edited from Sameguyy: ``` $pactl list short sources 80 alsa_output.platform-snd_aloop.0.analog-stereo.monitor PipeWire s32le 2ch 48000Hz RUNNING 81 alsa_input.platform-snd_aloop.0.analog-stereo PipeWire s32le 2ch 48000Hz RUNNING ``` These are two interfaces that Pipewire automatically pulled in, and the sound goes into 3.0 and doesn't leave from 3.1 That is, I play a sound on hw:3.0 and when I try to read something from hw:3.1 I get silence aplay\arecord records silence $ uname -r 6.17.8-arch1-1 Wayland PipeWire gnome ```$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 3: HDMI 0 [SAMSUNG] Subdevices: 0/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 3: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 ``` ``` $pacman -Q | grep alsa alsa-card-profiles 1:1.4.9-1 alsa-lib 1.2.14-2 alsa-plugins 1:1.2.12-5 alsa-topology-conf 1.2.5.1-4 alsa-ucm-conf 1.2.14-2 alsa-utils 1.2.14-1 lib32-alsa-lib 1.2.14-2 lib32-alsa-plugins 1.2.12-1 vlc-plugin-alsa 3.0.21-31 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/482 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd-loopback does not work
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-lib issue #482 was edited from Sameguyy: ``` $pactl list short sources 80 alsa_output.platform-snd_aloop.0.analog-stereo.monitor PipeWire s32le 2ch 48000Hz RUNNING 81 alsa_input.platform-snd_aloop.0.analog-stereo PipeWire s32le 2ch 48000Hz RUNNING ``` These are two interfaces that Pipewire automatically pulled in, and the sound goes into 3.0 and doesn't leave from 3.1 That is, I play a sound on hw:3.0 and when I try to read something from hw:3.1 I get silence aplay\arecord records silence $ uname -r 6.17.8-arch1-1 Wayland PipeWire gnome ```$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 3: HDMI 0 [SAMSUNG] Subdevices: 0/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 3: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/482 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd-loopback does not work
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-lib issue #482 was edited from Sameguyy: ``` $pactl list short sources 80 alsa_output.platform-snd_aloop.0.analog-stereo.monitor PipeWire s32le 2ch 48000Hz RUNNING 81 alsa_input.platform-snd_aloop.0.analog-stereo PipeWire s32le 2ch 48000Hz RUNNING ``` These are two interfaces that Pipewire automatically pulled in, and the sound goes into 3.0 and doesn't leave from 3.1 That is, I play a sound on hw:3.0 and when I try to read something from hw:3.1 I get silence aplay\arecord records silence $ uname -r 6.17.8-arch1-1 Wayland PipeWire gnome ```$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 3: HDMI 0 [SAMSUNG] Subdevices: 0/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 3: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 ``` I don't understand what I'm doing wrong can sm1 guess Issue URL : https://github.com/alsa-project/alsa-lib/issues/482 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd-loopback does not work
by GitHub issues - opened 17 Nov '25

17 Nov '25

alsa-project/alsa-lib issue #482 was opened from Sameguyy: ``` $pactl list short sources 80 alsa_output.platform-snd_aloop.0.analog-stereo.monitor PipeWire s32le 2ch 48000Hz RUNNING 81 alsa_input.platform-snd_aloop.0.analog-stereo PipeWire s32le 2ch 48000Hz RUNNING ``` These are two interfaces that Pipewire automatically pulled in, and the sound goes into 3.0 and doesn't leave from 3.1 That is, I play a sound on hw:3.0 and when I try to read something from hw:3.1 I get silence aplay\arecord records silence $ uname -r 6.17.8-arch1-1 Wayland PipeWire gnome ```$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 3: HDMI 0 [SAMSUNG] Subdevices: 0/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 3: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 I don't understand what I'm doing wrong can sm1 guess Issue URL : https://github.com/alsa-project/alsa-lib/issues/482 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v6.18-rc5
by Mark Brown 14 Nov '25

14 Nov '25

The following changes since commit 6146a0f1dfae5d37442a9ddcba012add260bceb0: Linux 6.18-rc4 (2025-11-02 11:28:02 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.18-rc5 for you to fetch changes up to 360b3730f8eab6c4467c6cca4cb0e30902174a63: ASoC: rsnd: fix OF node reference leak in rsnd_ssiu_probe() (2025-11-13 00:36:01 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.18 A small collection of fixes, all driver specific and none especially remarkable unless you have the hardware (many not even then). ---------------------------------------------------------------- Claudiu Beznea (1): ASoC: da7213: Use component driver suspend/resume Haotian Zhang (3): ASoC: cs4271: Fix regulator leak on probe failure ASoC: codecs: va-macro: fix resource leak in probe error path ASoC: rsnd: fix OF node reference leak in rsnd_ssiu_probe() Miaoqian Lin (1): ASoC: sdw_utils: fix device reference leak in is_sdca_endpoint_present() Niranjan H Y (2): ASoC: tas2783A: Fix issues in firmware parsing ASoC: SDCA: bug fix while parsing mipi-sdca-control-cn-list Richard Fitzgerald (1): ASoC: doc: cs35l56: Update firmware filename description for B0 silicon Shenghao Ding (1): ASoC: tas2781: fix getting the wrong device number Documentation/sound/codecs/cs35l56.rst | 9 ++--- sound/soc/codecs/cs4271.c | 10 +++-- sound/soc/codecs/da7213.c | 69 ++++++++++++++++++++++------------ sound/soc/codecs/da7213.h | 1 + sound/soc/codecs/lpass-va-macro.c | 2 +- sound/soc/codecs/tas2781-i2c.c | 9 ++++- sound/soc/codecs/tas2783-sdw.c | 20 +++++++++- sound/soc/renesas/rcar/ssiu.c | 3 +- sound/soc/sdca/sdca_functions.c | 3 +- sound/soc/sdw_utils/soc_sdw_utils.c | 20 +++++++--- 10 files changed, 99 insertions(+), 47 deletions(-)

2 1

[PATCH alsa-utils] amidi: Ignore inactive MIDI ports as default at listing
by Takashi Iwai 13 Nov '25

13 Nov '25

When listing the devices, currently we show all MIDI ports including inactive ones. But those inactive ports are rarely useful, and it'd be more convenient to filter them out. This patch introduces the filtering of inactive ports at listing devices via amidi -l option. When user needs to scan all MIDI ports including inactive ports, pass the new option -x in addition. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- amidi/amidi.1 | 5 +++++ amidi/amidi.c | 11 ++++++++++- 2 files changed, 15 insertions(+), 1 deletion(-) diff --git a/amidi/amidi.1 b/amidi/amidi.1 index 5bc24ba3ad54..d3d321f16558 100644 --- a/amidi/amidi.1 +++ b/amidi/amidi.1 @@ -52,6 +52,11 @@ Prints the current version. .I \-l, \-\-list\-devices Prints a list of all hardware MIDI ports. +.TP +.I \-x, \-\-list\-inactive +Use together with \fI\-l\fP option. +Print all MIDI ports including inactive ports. + .TP .I \-L, \-\-list\-rawmidis Prints all RawMIDI definitions. diff --git a/amidi/amidi.c b/amidi/amidi.c index 75fb8c0a42a3..9a2aaf7205f0 100644 --- a/amidi/amidi.c +++ b/amidi/amidi.c @@ -57,6 +57,7 @@ static int stop; static int sysex_interval; static snd_rawmidi_t *input, **inputp; static snd_rawmidi_t *output, **outputp; +static int list_all; static void error(const char *format, ...) { @@ -76,6 +77,7 @@ static void usage(void) "-h, --help this help\n" "-V, --version print current version\n" "-l, --list-devices list all hardware ports\n" + "-x, --list-inactive list inactive ports, too\n" "-L, --list-rawmidis list all RawMIDI definitions\n" "-p, --port=name select port by name\n" "-s, --send=file send the contents of a (.syx) file\n" @@ -151,6 +153,9 @@ static void list_device(snd_ctl_t *ctl, int card, int device) card, device, sub, snd_strerror(err)); return; } + if (!list_all && + (snd_rawmidi_info_get_flags(info) & SNDRV_RAWMIDI_INFO_STREAM_INACTIVE)) + continue; name = snd_rawmidi_info_get_name(info); sub_name = snd_rawmidi_info_get_subdevice_name(info); if (sub == 0 && sub_name[0] == '\0') { @@ -471,11 +476,12 @@ static void add_send_hex_data(const char *str) int main(int argc, char *argv[]) { - static const char short_options[] = "hVlLp:s:r:S::dt:aci:T:"; + static const char short_options[] = "hVlxLp:s:r:S::dt:aci:T:"; static const struct option long_options[] = { {"help", 0, NULL, 'h'}, {"version", 0, NULL, 'V'}, {"list-devices", 0, NULL, 'l'}, + {"list-inactive", 0, NULL, 'x'}, {"list-rawmidis", 0, NULL, 'L'}, {"port", 1, NULL, 'p'}, {"send", 1, NULL, 's'}, @@ -508,6 +514,9 @@ int main(int argc, char *argv[]) case 'l': do_device_list = 1; break; + case 'x': + list_all = 1; + break; case 'L': do_rawmidi_list = 1; break; -- 2.51.1

1 0

[BUG] Realtek ALC236 Internal Mic not working on HP 255 G10 (Subsystem ID 103c:8b2f)
by Pape Samba Mbodj Sarr 13 Nov '25

13 Nov '25

Bonjour, J'ai un problème avec le microphone interne (intégré) qui n'est pas détecté par le système (affiché comme "non branché" / "not available") sur mon ordinateur portable HP. J'ai épuisé toutes les options de configuration utilisateur (modèle ALSA, patchs manuels). Ceci semble être un "quirk" (correctif) manquant pour le pilote. **Informations du système :** * **Ordinateur :** HP 255 15.6 inch G10 Notebook PC * **Codec Audio :** Realtek ALC236 * **Subsystem ID :** 103c:8b2f (Hewlett-Packard Company) * **Noyau Linux :** 6.14.0-35-generic (Version très récente) **URL du rapport alsa-info.sh (Codec Dump) :**http://alsa-project.org/db/?f=55da44f60f5deb6073c9edfe2f49c3b1cb03b53e L'entrée microphone (analog-input-mic) est marquée comme "not available" (non disponible) dans pactl. Je soupçonne que le pilote ne parvient pas à initialiser le microphone numérique (DMIC) ou confond l'entrée interne avec le jack externe. Merci d'avance de votre aide pour l'intégration de ce correctif. Cordialement, SARR Mbodj

1 0

aplay: cannot play S24_LE wav file recorded with arecord
by GitHub issues - opened 13 Nov '25

13 Nov '25

alsa-project/alsa-utils issue #308 was opened from yoshihisa-yamamoto-re: My environment is alsa-utils version : 1.2.14 (Debian trixie) I tried to record and play PCM audio as S24_LE format over alsa as follows: ``` arecord -f S24_LE out.wav aplay out.wav ``` However, the audio turned into noise and did not seem to play properly. This problem does **not** seem to occur when using alsa-utils version 1.2.9 (Ubuntu 24.04 LTS). Searching through the commits, I found the closed issue https://github.com/alsa-project/alsa-utils/pull/260. From reading the source, it appears that the header information has been rewritten, so I dumped it to check, and it looks like this: hexdump of wav created with alsa-utils version 1.2.14 : ``` 00000000 52 49 46 46 24 e2 04 00 57 41 56 45 66 6d 74 20 |RIFF$...WAVEfmt | 00000010 10 00 00 00 01 00 01 00 40 1f 00 00 00 7d 00 00 |........@....}..| 00000020 04 00 20 00 64 61 74 61 00 e2 04 00 e6 67 8d ff |.. .data.....g..| 00000030 c8 fd 8e ff 0c 8e 90 ff c6 18 92 ff 0a 9e 93 ff |................| 00000040 ec 1d 95 ff 7e 98 96 ff d2 0d 98 ff fd 7d 99 ff |....~........}..| 00000050 10 e9 9a ff 1c 4f 9c ff 35 b0 9d ff 6b 0c 9f ff |.....O..5...k...| 00000060 cf 63 a0 ff 74 b6 a1 ff 68 04 a3 ff be 4d a4 ff |.c..t...h....M..| 00000070 86 92 a5 ff cf d2 a6 ff a9 0e a8 ff 25 46 a9 ff |............%F..| ``` hexdump of wav created with alsa-utils version 1.2.9 : ``` 00000000 52 49 46 46 24 e2 04 00 57 41 56 45 66 6d 74 20 |RIFF$...WAVEfmt | 00000010 10 00 00 00 01 00 01 00 40 1f 00 00 00 7d 00 00 |........@....}..| 00000020 04 00 18 00 64 61 74 61 00 e2 04 00 17 3d fe ff |....data.....=..| 00000030 3e 3d fe ff 86 41 fe ff cb 47 fe ff 3b 46 fe ff |>=...A...G..;F..| 00000040 10 47 fe ff e3 4d fe ff 66 4f fe ff 8d 50 fe ff |.G...M..fO...P..| 00000050 be 4f fe ff 40 58 fe ff e7 5b fe ff 27 60 fe ff |.O..@X...[..'`..| 00000060 f9 60 fe ff ce 61 fe ff ce 63 fe ff d3 68 fe ff |.`...a...c...h..| 00000070 83 6f fe ff 84 6c fe ff 22 73 fe ff 91 6d fe ff |.o...l.."s...m..| ``` As the change in Commit 9cdcbe0e932b97fac88a6195f87647159a6c7194, the header information has changed BitsPerSample (bit_p_spl) from 24bit to 32bit (0x22-0x23). If I change this header info and treat the 1.2.14 wav as 24-bit per sample as before, the audio seems to play correctly. I don't have the correct information for S24_LE format, so I don't know if the header information is wrong or if I should modify the arecord to match this 32-bit. Issue URL : https://github.com/alsa-project/alsa-utils/issues/308 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] MAINTAINERS: update Tzung-Bi's email address
by Tzung-Bi Shih 11 Nov '25

11 Nov '25

Use kernel.org account instead. Signed-off-by: Tzung-Bi Shih <tzungbi(a)kernel.org> --- .../devicetree/bindings/sound/google,cros-ec-codec.yaml | 2 +- Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 2 +- MAINTAINERS | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index c3e9f3485449..dea293f403d9 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -8,7 +8,7 @@ title: Audio codec controlled by ChromeOS EC maintainers: - Cheng-Yi Chiang <cychiang(a)chromium.org> - - Tzung-Bi Shih <tzungbi(a)google.com> + - Tzung-Bi Shih <tzungbi(a)kernel.org> description: | Google's ChromeOS EC codec is a digital mic codec provided by the diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml index 1d73204451b1..ea7d4900ee4a 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Realtek rt1015p codec devicetree bindings maintainers: - - Tzung-Bi Shih <tzungbi(a)google.com> + - Tzung-Bi Shih <tzungbi(a)kernel.org> description: | Rt1015p is a rt1015 variant which does not support I2C and diff --git a/MAINTAINERS b/MAINTAINERS index cf0f18502372..f9749afc0b9d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4906,7 +4906,7 @@ F: drivers/platform/chrome/ CHROMEOS EC CODEC DRIVER M: Cheng-Yi Chiang <cychiang(a)chromium.org> -M: Tzung-Bi Shih <tzungbi(a)google.com> +M: Tzung-Bi Shih <tzungbi(a)kernel.org> R: Guenter Roeck <groeck(a)chromium.org> L: chrome-platform(a)lists.linux.dev S: Maintained -- 2.38.0.135.g90850a2211-goog

3 3

ucm2: IO-Boards: Toradex: aquila: add support
by GitHub pull_request - opened 10 Nov '25

10 Nov '25

alsa-project/alsa-ucm-conf pull request #641 was opened from hiagofranco: Add support for Toradex Aquila Development board, using the WM8904 audio codec. This is a carrier board for the Toradex Aquila family, where any Toradex Aquila SoM can be connected to it, therefore this is being added to the IO-Boards instead of a specific hardware vendor. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/641 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/641.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v1] ASoC: SDCA: bug fix while parsing mipi-sdca-control-cn-list
by Niranjan H Y 10 Nov '25

10 Nov '25

"struct sdca_control" declares "values" field as integer array. But the memory allocated to it is of char array. This causes crash for sdca_parse_function API. This patch addresses the issue by allocating correct data size. Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> --- sound/soc/sdca/sdca_functions.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sdca/sdca_functions.c b/sound/soc/sdca/sdca_functions.c index b2e3fab9b..6975c71a9 100644 --- a/sound/soc/sdca/sdca_functions.c +++ b/sound/soc/sdca/sdca_functions.c @@ -950,7 +950,8 @@ static int find_sdca_entity_control(struct device *dev, struct sdca_entity *enti return ret; } - control->values = devm_kzalloc(dev, hweight64(control->cn_list), GFP_KERNEL); + control->values = devm_kzalloc(dev, hweight64(control->cn_list) * sizeof(int), + GFP_KERNEL); if (!control->values) return -ENOMEM; -- 2.25.1

1 0

PreSonus revelator io44 reports [ALSA UCM error]
by GitHub issues - opened 10 Nov '25

10 Nov '25

alsa-project/alsa-ucm-conf issue #640 was opened from taiwanese-eirin: `wpctl status` shows messages like: ``` Audio ├─ Devices: │ 43. Built-in Audio [alsa] │ 58. Revelator IO 44 [ALSA UCM error] [alsa] ``` So far the playback seems to be working. Not yet tested the recording. Attached `alsa-info.sh` output for more details: [alsa-info.txt](https://github.com/user-attachments/files/23451692/alsa-info… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/640 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Regression on Lenovo Yoga 7i Aura Slim 14ILL10
by GitHub issues - edited 09 Nov '25

09 Nov '25

alsa-project/alsa-ucm-conf issue #639 was edited from nikola-maric-aula: I did a bisect and found that after https://github.com/alsa-project/alsa-ucm-conf/commit/76c14b89db6a45f479e466… my audio configuration wouldn't recognize the audio controller properly. Let me know what other details you want me to include. I'm on Zen kernel mostly, but have also tried LTS and mainline. <img width="400" src="https://github.com/user-attachments/assets/6bac78bc-ffad-47a4-a40f-5bc89a03…" /> Before: [alsa-info.sh - Before](http://alsa-project.org/db/?f=a0bac2a3b8b5fdc64fc691a12d75f7ac19cc3… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 46. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ 57. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 3 Output [vol: 1.00] │ 58. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 2 Output [vol: 1.00] │ 59. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 1 Output [vol: 1.00] │ * 60. Lunar Lake-M HD Audio Controller Speaker [vol: 0.48] │ ├─ Sources: │ 61. Lunar Lake-M HD Audio Controller Headset Microphone [vol: 1.00] │ * 62. Lunar Lake-M HD Audio Controller Digital Microphone [vol: 0.48] │ ├─ Filters: │ λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn ~ 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel alsaucm>> listcards 0: hw:0 LENOVO-83JX-YogaSlim714ILL10-LNVNB161216 ``` After: [alsa-info.sh - After](http://alsa-project.org/db/?f=afac4f3766910d0f3812216ab3a9433b846bf7… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 49. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ * 48. Dummy Output [vol: 1.00] │ ├─ Sources: │ ├─ Filters: │ └─ Streams: Video ├─ Devices: │ 57. Integrated RGB Camera [v4l2] λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel alsaucm>> listcards ALSA lib parser.c:1799:(verb_dev_list_add) error: unable to find device 'Headphones' ALSA lib parser.c:2100:(parse_verb_file) error: device management error in verb 'HiFi' ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -2 list is empty ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/639 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Regression on Lenovo Yoga 7i Aura Slim 14ILL10
by GitHub issues - edited 09 Nov '25

09 Nov '25

alsa-project/alsa-ucm-conf issue #639 was edited from nikola-maric-aula: I did a bisect and found that after https://github.com/alsa-project/alsa-ucm-conf/pull/628 my audio configuration wouldn't recognize the audio controller properly. Let me know what other details you want me to include. I'm on Zen kernel mostly, but have also tried LTS and mainline. <img width="400" src="https://github.com/user-attachments/assets/6bac78bc-ffad-47a4-a40f-5bc89a03…" /> Before: [alsa-info.sh - Before](http://alsa-project.org/db/?f=a0bac2a3b8b5fdc64fc691a12d75f7ac19cc3… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 46. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ 57. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 3 Output [vol: 1.00] │ 58. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 2 Output [vol: 1.00] │ 59. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 1 Output [vol: 1.00] │ * 60. Lunar Lake-M HD Audio Controller Speaker [vol: 0.48] │ ├─ Sources: │ 61. Lunar Lake-M HD Audio Controller Headset Microphone [vol: 1.00] │ * 62. Lunar Lake-M HD Audio Controller Digital Microphone [vol: 0.48] │ ├─ Filters: │ λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn ~ 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel alsaucm>> listcards 0: hw:0 LENOVO-83JX-YogaSlim714ILL10-LNVNB161216 ``` After: [alsa-info.sh - After](http://alsa-project.org/db/?f=afac4f3766910d0f3812216ab3a9433b846bf7… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 49. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ * 48. Dummy Output [vol: 1.00] │ ├─ Sources: │ ├─ Filters: │ └─ Streams: Video ├─ Devices: │ 57. Integrated RGB Camera [v4l2] λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel alsaucm>> listcards ALSA lib parser.c:1799:(verb_dev_list_add) error: unable to find device 'Headphones' ALSA lib parser.c:2100:(parse_verb_file) error: device management error in verb 'HiFi' ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -2 list is empty ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/639 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Regression on Lenovo Yoga 7i Aura Slim 14ILL10
by GitHub issues - edited 09 Nov '25

09 Nov '25

alsa-project/alsa-ucm-conf issue #639 was edited from nikola-maric-aula: I did a bisect and found that after https://github.com/alsa-project/alsa-ucm-conf/pull/628 my audio configuration wouldn't recognize the audio controller properly. Let me know what other details you want me to include. I'm on Zen kernel mostly, but have also tried LTS and mainline. <img width="400" src="https://github.com/user-attachments/assets/6bac78bc-ffad-47a4-a40f-5bc89a03…" /> Before: [alsa-info.sh - After](http://alsa-project.org/db/?f=a0bac2a3b8b5fdc64fc691a12d75f7ac19cc3e… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 46. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ 57. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 3 Output [vol: 1.00] │ 58. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 2 Output [vol: 1.00] │ 59. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 1 Output [vol: 1.00] │ * 60. Lunar Lake-M HD Audio Controller Speaker [vol: 0.48] │ ├─ Sources: │ 61. Lunar Lake-M HD Audio Controller Headset Microphone [vol: 1.00] │ * 62. Lunar Lake-M HD Audio Controller Digital Microphone [vol: 0.48] │ ├─ Filters: │ λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn ~ 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel alsaucm>> listcards 0: hw:0 LENOVO-83JX-YogaSlim714ILL10-LNVNB161216 ``` After: [alsa-info.sh - After](http://alsa-project.org/db/?f=afac4f3766910d0f3812216ab3a9433b846bf7… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 49. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ * 48. Dummy Output [vol: 1.00] │ ├─ Sources: │ ├─ Filters: │ └─ Streams: Video ├─ Devices: │ 57. Integrated RGB Camera [v4l2] λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel alsaucm>> listcards ALSA lib parser.c:1799:(verb_dev_list_add) error: unable to find device 'Headphones' ALSA lib parser.c:2100:(parse_verb_file) error: device management error in verb 'HiFi' ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -2 list is empty ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/639 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Regression on Lenovo Yoga 7i Aura Slim 14ILL10
by GitHub issues - edited 09 Nov '25

09 Nov '25

alsa-project/alsa-ucm-conf issue #639 was edited from nikola-maric-aula: I did a bisect and found that after https://github.com/alsa-project/alsa-ucm-conf/pull/628 my audio configuration wouldn't recognize the audio controller properly. Let me know what other details you want me to include. I'm on Zen kernel mostly, but have also tried LTS and mainline. <img width="400" src="https://github.com/user-attachments/assets/6bac78bc-ffad-47a4-a40f-5bc89a03…" /> Before: ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 46. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ 57. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 3 Output [vol: 1.00] │ 58. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 2 Output [vol: 1.00] │ 59. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 1 Output [vol: 1.00] │ * 60. Lunar Lake-M HD Audio Controller Speaker [vol: 0.48] │ ├─ Sources: │ 61. Lunar Lake-M HD Audio Controller Headset Microphone [vol: 1.00] │ * 62. Lunar Lake-M HD Audio Controller Digital Microphone [vol: 0.48] │ ├─ Filters: │ λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn ~ 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel ``` After: [alsa-info.sh - After](http://alsa-project.org/db/?f=afac4f3766910d0f3812216ab3a9433b846bf7… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 49. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ * 48. Dummy Output [vol: 1.00] │ ├─ Sources: │ ├─ Filters: │ └─ Streams: Video ├─ Devices: │ 57. Integrated RGB Camera [v4l2] λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel alsaucm>> listcards ALSA lib parser.c:1799:(verb_dev_list_add) error: unable to find device 'Headphones' ALSA lib parser.c:2100:(parse_verb_file) error: device management error in verb 'HiFi' ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -2 list is empty ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/639 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Regression on Lenovo Yoga 7i Aura Slim 14ILL10
by GitHub issues - edited 09 Nov '25

09 Nov '25

alsa-project/alsa-ucm-conf issue #639 was edited from nikola-maric-aula: I did a bisect and found that after https://github.com/alsa-project/alsa-ucm-conf/pull/628 my audio configuration wouldn't recognize the audio controller properly. Let me know what other details you want me to include. I'm on Zen kernel mostly, but have also tried LTS and mainline. <img width="400" src="https://github.com/user-attachments/assets/6bac78bc-ffad-47a4-a40f-5bc89a03…" /> Before: ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 46. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ 57. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 3 Output [vol: 1.00] │ 58. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 2 Output [vol: 1.00] │ 59. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 1 Output [vol: 1.00] │ * 60. Lunar Lake-M HD Audio Controller Speaker [vol: 0.48] │ ├─ Sources: │ 61. Lunar Lake-M HD Audio Controller Headset Microphone [vol: 1.00] │ * 62. Lunar Lake-M HD Audio Controller Digital Microphone [vol: 0.48] │ ├─ Filters: │ λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn ~ 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel ``` After: [alsa-info.sh - After](http://alsa-project.org/db/?f=afac4f3766910d0f3812216ab3a9433b846bf7… ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 49. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ * 48. Dummy Output [vol: 1.00] │ ├─ Sources: │ ├─ Filters: │ └─ Streams: Video ├─ Devices: │ 57. Integrated RGB Camera [v4l2] λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/639 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Regression on Lenovo Yoga 7i Aura Slim 14ILL10
by GitHub issues - edited 09 Nov '25

09 Nov '25

alsa-project/alsa-ucm-conf issue #639 was edited from nikola-maric-aula: I did a bisect and found that after https://github.com/alsa-project/alsa-ucm-conf/pull/628 my audio configuration wouldn't recognize the audio controller properly. Let me know what other details you want me to include. I'm on Zen kernel mostly, but have also tried LTS and mainline. <img width="400" src="https://github.com/user-attachments/assets/6bac78bc-ffad-47a4-a40f-5bc89a03…" /> Before: ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 46. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ 57. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 3 Output [vol: 1.00] │ 58. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 2 Output [vol: 1.00] │ 59. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 1 Output [vol: 1.00] │ * 60. Lunar Lake-M HD Audio Controller Speaker [vol: 0.48] │ ├─ Sources: │ 61. Lunar Lake-M HD Audio Controller Headset Microphone [vol: 1.00] │ * 62. Lunar Lake-M HD Audio Controller Digital Microphone [vol: 0.48] │ ├─ Filters: │ λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn ~ 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel ``` After: ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 49. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ * 48. Dummy Output [vol: 1.00] │ ├─ Sources: │ ├─ Filters: │ └─ Streams: Video ├─ Devices: │ 57. Integrated RGB Camera [v4l2] λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/639 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Regression on Lenovo Yoga 7i Aura Slim 14ILL10
by GitHub issues - opened 09 Nov '25

09 Nov '25

alsa-project/alsa-ucm-conf issue #639 was opened from nikola-maric-aula: I did a bisect and found that after https://github.com/alsa-project/alsa-ucm-conf/pull/628 my audio configuration wouldn't recognize the audio controller properly. Let me know what other details you want me to include. I'm on Zen kernel mostly, but have also tried LTS and mainline. Before: ```bash λ › wpctl status | grep -A15 'Audio$' Audio ├─ Devices: │ 46. Lunar Lake-M HD Audio Controller [alsa] │ ├─ Sinks: │ 57. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 3 Output [vol: 1.00] │ 58. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 2 Output [vol: 1.00] │ 59. Lunar Lake-M HD Audio Controller HDMI / DisplayPort 1 Output [vol: 1.00] │ * 60. Lunar Lake-M HD Audio Controller Speaker [vol: 0.48] │ ├─ Sources: │ 61. Lunar Lake-M HD Audio Controller Headset Microphone [vol: 1.00] │ * 62. Lunar Lake-M HD Audio Controller Digital Microphone [vol: 0.48] │ ├─ Filters: │ λ › aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sofsoundwire [sof-soundwire], device 0: Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 2: Speaker (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 5: HDMI1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 6: HDMI2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 7: HDMI3 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 λ › lspci -s 00:1f.3 -knn ~ 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem: Lenovo Device [17aa:3838] Kernel driver in use: sof-audio-pci-intel-lnl Kernel modules: snd_sof_pci_intel_lnl, snd_hda_intel ``` <img width="400" src="https://github.com/user-attachments/assets/6bac78bc-ffad-47a4-a40f-5bc89a03…" /> After: ```bash just a moment, rebooting ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/639 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[REGRESSION?] No sound from internal speakers on Teclast T.BOLT with Realtek ALC256
by Marco Aurelio Osorio De León 08 Nov '25

08 Nov '25

Subject: [REGRESSION?] No sound from internal speakers on Teclast T.BOLT with Realtek ALC256 Hello everyone, I have a Teclast T.BOLT laptop with a Realtek ALC256 audio codec. The problem is that the sound from the internal speakers isn't working. The audio works perfectly through headphones, but when I disconnect them, there's no sound from the speakers. In the graphical interface (PulseAudio/PipeWire), the volume bar moves as if the sound were playing correctly, but there's no physical audio output. I've tried numerous solutions without success: - I checked `alsamixer` to ensure nothing is muted and that "Auto-Mute" is enabled. - - Tested multiple `model` parameters for `snd-hda-intel` in `/etc/modprobe.d/alsa-base.conf` and also as a kernel parameter in GRUB (dell-xps-13-9360, dell-headset-multi, lenovo-legion-15imhg05, alc255-dell1, etc.). - Tried reconfiguring the pins with `hdajackretask`, forcing pin 0x14 to "Internal Speaker (LFE)" and testing different configurations for pin 0x21. - Attempted several `hda-verb` commands to force the codec state, both with `SET_COEF_DATA` and `SET_PIN_WIDGET_CONTROL`, but without success. The hardware is working correctly, as the speakers function without problems on a Windows 11 installation on the same laptop. This appears to be a configuration issue with the generic Linux driver for this specific hardware. I've switched from the standard Ubuntu 25.10 kernel to Liquorix, but there's no change. Linux XXXXX 6.17.7-1-liquorix-amd64 #1 ZEN SMP PREEMPT_DYNAMIC liquorix 6.17-7ubuntu1~questing (2025 x86_64 GNU/Linux) I generated a full report using `alsa-info.sh`. The URL is: https://alsa-project.org/db/?f=96176a5abcc90780bbba4a67dac946be6b793c2c Any ideas on what else I could try, or if it might be a bug that requires a specific fix for this laptop? Thank you for your time and effort. Regards.

1 0

[PATCH v1] ASoC: tas2781: fix getting the wrong device number
by Shenghao Ding 07 Nov '25

07 Nov '25

The return value of device_property_read_u32_array used for getting the property is the status instead of the number of the property. Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8f853310649c..7bd98ff5a250 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -1958,7 +1958,8 @@ static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) { struct i2c_client *client = (struct i2c_client *)tas_priv->client; unsigned int dev_addrs[TASDEVICE_MAX_CHANNELS]; - int i, ndev = 0; + int ndev = 0; + int i, rc; if (tas_priv->isacpi) { ndev = device_property_read_u32_array(&client->dev, @@ -1969,8 +1970,12 @@ static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) } else { ndev = (ndev < ARRAY_SIZE(dev_addrs)) ? ndev : ARRAY_SIZE(dev_addrs); - ndev = device_property_read_u32_array(&client->dev, + rc = device_property_read_u32_array(&client->dev, "ti,audio-slots", dev_addrs, ndev); + if (rc != 0) { + ndev = 1; + dev_addrs[0] = client->addr; + } } tas_priv->irq = -- 2.43.0

2 1

ucm2: Qualcomm: add TUXEDO Elite 14 support
by GitHub pull_request - edited 07 Nov '25

07 Nov '25

alsa-project/alsa-ucm-conf pull request #638 was edited from ektor5: TUXEDO Elite 14 Gen1 (Snapdragon X Elite) supports: - 4 speakers - 2 dmics - Headset - DisplayPort (USB) - HDMI Adding support to all above with this patchset. Signed-off-by: Ettore Chimenti <ettore.chimenti(a)linaro.org> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/638 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/638.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: Qualcomm: add TUXEDO Elite 14 support
by GitHub pull_request - opened 07 Nov '25

07 Nov '25

alsa-project/alsa-ucm-conf pull request #638 was opened from ektor5: TUXEDO Elite 14 Gen1 (Snapdragon X Elite) supports: - 4 speakers - 2 dmics - Headset - DisplayPort (USB) - HDMI Adding support to all above with this patchset. Signed-off-by: Ettore Chimenti <ettore.chimenti(a)linaro.org> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/638 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/638.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v6 1/2] ASoC: rt5575: Add the codec driver for the ALC5575
by Oder Chiou 07 Nov '25

07 Nov '25

The ALC5575 integrates an audio DSP that typically loads its firmware from an external flash via its own SPI host interface. In certain hardware configurations, the firmware can alternatively be loaded through the SPI client interface. The driver provides basic mute and volume control functions. When the SPI client interface is enabled, firmware loading is handled by the SPI driver. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/rt5575-spi.c | 95 +++++++++ sound/soc/codecs/rt5575-spi.h | 16 ++ sound/soc/codecs/rt5575.c | 366 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5575.h | 54 +++++ 6 files changed, 545 insertions(+) create mode 100644 sound/soc/codecs/rt5575-spi.c create mode 100644 sound/soc/codecs/rt5575-spi.h create mode 100644 sound/soc/codecs/rt5575.c create mode 100644 sound/soc/codecs/rt5575.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a0dfef57200c..cb4e14013cc8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -211,6 +211,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT1305 imply SND_SOC_RT1308 imply SND_SOC_RT5514 + imply SND_SOC_RT5575 imply SND_SOC_RT5616 imply SND_SOC_RT5631 imply SND_SOC_RT5640 @@ -1767,6 +1768,15 @@ config SND_SOC_RT5514_SPI_BUILTIN bool # force RT5514_SPI to be built-in to avoid link errors default SND_SOC_RT5514=y && SND_SOC_RT5514_SPI=m +config SND_SOC_RT5575 + tristate + depends on I2C + +config SND_SOC_RT5575_SPI + tristate + depends on SPI_MASTER + select SND_SOC_RT5575 + config SND_SOC_RT5616 tristate "Realtek RT5616 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 39138d96a720..82f660cbe8ec 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -252,6 +252,8 @@ snd-soc-rt286-y := rt286.o snd-soc-rt298-y := rt298.o snd-soc-rt5514-y := rt5514.o snd-soc-rt5514-spi-y := rt5514-spi.o +snd-soc-rt5575-y := rt5575.o +snd-soc-rt5575-spi-y := rt5575-spi.o snd-soc-rt5616-y := rt5616.o snd-soc-rt5631-y := rt5631.o snd-soc-rt5640-y := rt5640.o @@ -684,6 +686,8 @@ obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o obj-$(CONFIG_SND_SOC_RT5514_SPI) += snd-soc-rt5514-spi.o obj-$(CONFIG_SND_SOC_RT5514_SPI_BUILTIN) += snd-soc-rt5514-spi.o +obj-$(CONFIG_SND_SOC_RT5575) += snd-soc-rt5575.o +obj-$(CONFIG_SND_SOC_RT5575_SPI) += snd-soc-rt5575-spi.o obj-$(CONFIG_SND_SOC_RT5616) += snd-soc-rt5616.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/rt5575-spi.c b/sound/soc/codecs/rt5575-spi.c new file mode 100644 index 000000000000..cf30d22e8a8f --- /dev/null +++ b/sound/soc/codecs/rt5575-spi.c @@ -0,0 +1,95 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * rt5575-spi.c -- ALC5575 SPI driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#include <linux/of.h> +#include <linux/spi/spi.h> + +#include "rt5575-spi.h" + +#define RT5575_SPI_BUF_LEN 240 + +/* SPI Command */ +enum { + RT5575_SPI_CMD_16_READ = 0, + RT5575_SPI_CMD_16_WRITE, + RT5575_SPI_CMD_32_READ, + RT5575_SPI_CMD_32_WRITE, + RT5575_SPI_CMD_BURST_READ, + RT5575_SPI_CMD_BURST_WRITE, +}; + +struct rt5575_spi_burst_write { + u8 cmd; + u32 addr; + u8 data[RT5575_SPI_BUF_LEN]; + u8 dummy; +} __packed; + +bool rt5575_spi_ready; +static struct spi_device *rt5575_spi; + +/** + * rt5575_spi_burst_write - Write data to SPI by rt5575 address. + * @addr: Start address. + * @txbuf: Data buffer for writng. + * @len: Data length. + * + */ +int rt5575_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) +{ + struct rt5575_spi_burst_write buf = { + .cmd = RT5575_SPI_CMD_BURST_WRITE + }; + unsigned int end, offset = 0; + + while (offset < len) { + if (offset + RT5575_SPI_BUF_LEN <= len) + end = RT5575_SPI_BUF_LEN; + else + end = len % RT5575_SPI_BUF_LEN; + + buf.addr = cpu_to_le32(addr + offset); + + memcpy(&buf.data, &txbuf[offset], end); + + spi_write(rt5575_spi, &buf, sizeof(struct rt5575_spi_burst_write)); + + offset += RT5575_SPI_BUF_LEN; + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5575_spi_burst_write); + +static int rt5575_spi_probe(struct spi_device *spi) +{ + rt5575_spi = spi; + + rt5575_spi_ready = true; + + return 0; +} + +static const struct of_device_id rt5575_of_match[] = { + { .compatible = "realtek,rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(of, rt5575_of_match); + +static struct spi_driver rt5575_spi_driver = { + .driver = { + .name = "rt5575", + .of_match_table = of_match_ptr(rt5575_of_match), + }, + .probe = rt5575_spi_probe, +}; +module_spi_driver(rt5575_spi_driver); + +MODULE_DESCRIPTION("ALC5575 SPI driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5575-spi.h b/sound/soc/codecs/rt5575-spi.h new file mode 100644 index 000000000000..cafe49a7ecc2 --- /dev/null +++ b/sound/soc/codecs/rt5575-spi.h @@ -0,0 +1,16 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5575-spi.h -- ALC5575 SPI driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#ifndef __RT5575_SPI_H__ +#define __RT5575_SPI_H__ + +extern bool rt5575_spi_ready; + +int rt5575_spi_burst_write(u32 addr, const u8 *txbuf, size_t len); + +#endif /* __RT5575_SPI_H__ */ diff --git a/sound/soc/codecs/rt5575.c b/sound/soc/codecs/rt5575.c new file mode 100644 index 000000000000..58cceaac0705 --- /dev/null +++ b/sound/soc/codecs/rt5575.c @@ -0,0 +1,366 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * rt5575.c -- ALC5575 ALSA SoC audio component driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#include <linux/firmware.h> +#include <linux/i2c.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "rt5575.h" +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) +#include "rt5575-spi.h" +#endif + +static bool rt5575_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5575_ID: + case RT5575_ID_1: + case RT5575_MIXL_VOL: + case RT5575_MIXR_VOL: + case RT5575_PROMPT_VOL: + case RT5575_SPK01_VOL: + case RT5575_SPK23_VOL: + case RT5575_MIC1_VOL: + case RT5575_MIC2_VOL: + case RT5575_WNC_CTRL: + case RT5575_MODE_CTRL: + case RT5575_I2S_RATE_CTRL: + case RT5575_SLEEP_CTRL: + case RT5575_ALG_BYPASS_CTRL: + case RT5575_PINMUX_CTRL_2: + case RT5575_GPIO_CTRL_1: + case RT5575_DSP_BUS_CTRL: + case RT5575_SW_INT: + case RT5575_DSP_BOOT_ERR: + case RT5575_DSP_READY: + case RT5575_DSP_CMD_ADDR: + case RT5575_EFUSE_DATA_2: + case RT5575_EFUSE_DATA_3: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(ob_tlv, -9525, 75, 0); + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) +static void rt5575_spi_fw_loaded(const struct firmware *fw, void *context) +{ + struct rt5575_priv *rt5575 = context; + struct i2c_client *i2c = rt5575->i2c; + const struct firmware *firmware; + static const char * const fw_path[] = { + "realtek/rt5575/rt5575_fw2.bin", "realtek/rt5575/rt5575_fw3.bin", + "realtek/rt5575/rt5575_fw4.bin" + }; + static const u32 fw_addr[] = { 0x5f600000, 0x5f7fe000, 0x5f7ff000 }; + int i, ret; + + regmap_write(rt5575->dsp_regmap, 0xfafafafa, 0x00000004); + regmap_write(rt5575->dsp_regmap, 0x18008064, 0x00000000); + regmap_write(rt5575->dsp_regmap, 0x18008068, 0x0002ffff); + + rt5575_spi_burst_write(0x5f400000, fw->data, fw->size); + release_firmware(fw); + + for (i = 0; i < ARRAY_SIZE(fw_addr); i++) { + ret = request_firmware(&firmware, fw_path[i], &i2c->dev); + if (ret == 0) { + rt5575_spi_burst_write(fw_addr[i], firmware->data, firmware->size); + release_firmware(firmware); + } + } + + regmap_write(rt5575->dsp_regmap, 0x18000000, 0x00000000); + + regmap_update_bits(rt5575->regmap, RT5575_SW_INT, 1, 1); + + regmap_read_poll_timeout(rt5575->regmap, RT5575_SW_INT, ret, !ret, 100000, 10000000); + + if (ret) + dev_err(&i2c->dev, "Firmware failure\n"); +} +#endif + +static const struct snd_kcontrol_new rt5575_snd_controls[] = { + SOC_DOUBLE("Speaker01 Playback Switch", RT5575_SPK01_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Speaker01 Playback Volume", RT5575_SPK01_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Speaker23 Playback Switch", RT5575_SPK23_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Speaker23 Playback Volume", RT5575_SPK23_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Mic1 Capture Switch", RT5575_MIC1_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Mic1 Capture Volume", RT5575_MIC1_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Mic2 Capture Switch", RT5575_MIC2_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Mic2 Capture Volume", RT5575_MIC2_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE_R("Mix Playback Switch", RT5575_MIXL_VOL, RT5575_MIXR_VOL, 31, 1, 1), + SOC_DOUBLE_R_TLV("Mix Playback Volume", RT5575_MIXL_VOL, RT5575_MIXR_VOL, 1, 127, 0, + ob_tlv), + SOC_DOUBLE("Prompt Playback Switch", RT5575_PROMPT_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Prompt Playback Volume", RT5575_PROMPT_VOL, 17, 1, 167, 0, ob_tlv), +}; + +static const struct snd_soc_dapm_widget rt5575_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3RX", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3TX", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF4RX", "AIF4 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF4TX", "AIF4 Capture", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_INPUT("INPUT"), + SND_SOC_DAPM_OUTPUT("OUTPUT"), +}; + +static const struct snd_soc_dapm_route rt5575_dapm_routes[] = { + { "AIF1TX", NULL, "INPUT" }, + { "AIF2TX", NULL, "INPUT" }, + { "AIF3TX", NULL, "INPUT" }, + { "AIF4TX", NULL, "INPUT" }, + { "OUTPUT", NULL, "AIF1RX" }, + { "OUTPUT", NULL, "AIF2RX" }, + { "OUTPUT", NULL, "AIF3RX" }, + { "OUTPUT", NULL, "AIF4RX" }, +}; + +static long long rt5575_get_priv_id(struct rt5575_priv *rt5575) +{ + int priv_id_low, priv_id_high; + + regmap_write(rt5575->regmap, RT5575_EFUSE_PID, 0xa0000000); + regmap_read(rt5575->regmap, RT5575_EFUSE_DATA_2, &priv_id_low); + regmap_read(rt5575->regmap, RT5575_EFUSE_DATA_3, &priv_id_high); + regmap_write(rt5575->regmap, RT5575_EFUSE_PID, 0); + + return ((long long)priv_id_high << 32) | (long long)priv_id_low; +} + +static int rt5575_probe(struct snd_soc_component *component) +{ + struct rt5575_priv *rt5575 = snd_soc_component_get_drvdata(component); + + rt5575->component = component; + + dev_info(component->dev, "Private ID: %llx\n", rt5575_get_priv_id(rt5575)); + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) + request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, "realtek/rt5575/rt5575_fw1.bin", + component->dev, GFP_KERNEL, rt5575, rt5575_spi_fw_loaded); +#endif + + return 0; +} + +#define RT5575_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5575_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver rt5575_dai[] = { + { + .name = "rt5575-aif1", + .id = RT5575_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif2", + .id = RT5575_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif3", + .id = RT5575_AIF3, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif4", + .id = RT5575_AIF4, + .playback = { + .stream_name = "AIF4 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF4 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, +}; + +static const struct snd_soc_component_driver rt5575_soc_component_dev = { + .probe = rt5575_probe, + .controls = rt5575_snd_controls, + .num_controls = ARRAY_SIZE(rt5575_snd_controls), + .dapm_widgets = rt5575_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5575_dapm_widgets), + .dapm_routes = rt5575_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5575_dapm_routes), + .use_pmdown_time = 1, + .endianness = 1, +}; + +static const struct regmap_config rt5575_dsp_regmap = { + .name = "dsp", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 2, +}; + +static int rt5575_i2c_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct rt5575_priv *rt5575 = i2c_get_clientdata(client); + + regmap_read(rt5575->dsp_regmap, reg | RT5575_DSP_MAPPING, val); + + return 0; +} + +static int rt5575_i2c_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + struct rt5575_priv *rt5575 = i2c_get_clientdata(client); + + regmap_write(rt5575->dsp_regmap, reg | RT5575_DSP_MAPPING, val); + + return 0; +} + +static const struct regmap_config rt5575_regmap = { + .reg_bits = 16, + .val_bits = 32, + .reg_stride = 4, + .max_register = 0xfffc, + .readable_reg = rt5575_readable_register, + .reg_read = rt5575_i2c_read, + .reg_write = rt5575_i2c_write, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct i2c_device_id rt5575_i2c_id[] = { + { "rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5575_i2c_id); + +static int rt5575_i2c_probe(struct i2c_client *i2c) +{ + struct rt5575_priv *rt5575; + int ret, val; + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) + if (!rt5575_spi_ready) + return -EPROBE_DEFER; +#endif + + rt5575 = devm_kzalloc(&i2c->dev, sizeof(struct rt5575_priv), + GFP_KERNEL); + if (rt5575 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5575); + + rt5575->i2c = i2c; + + rt5575->dsp_regmap = devm_regmap_init_i2c(i2c, &rt5575_dsp_regmap); + if (IS_ERR(rt5575->dsp_regmap)) { + ret = PTR_ERR(rt5575->dsp_regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); + return ret; + } + + rt5575->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt5575_regmap); + if (IS_ERR(rt5575->regmap)) { + ret = PTR_ERR(rt5575->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); + return ret; + } + + regmap_read(rt5575->regmap, RT5575_ID, &val); + if (val != RT5575_DEVICE_ID) { + dev_err(&i2c->dev, "Device with ID register %08x is not rt5575\n", val); + return -ENODEV; + } + + regmap_read(rt5575->regmap, RT5575_ID_1, &val); + if (!val) { + dev_err(&i2c->dev, "This is not formal version\n"); + return -ENODEV; + } + + return devm_snd_soc_register_component(&i2c->dev, &rt5575_soc_component_dev, rt5575_dai, + ARRAY_SIZE(rt5575_dai)); +} + +static const struct of_device_id rt5575_of_match[] = { + { .compatible = "realtek,rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(of, rt5575_of_match); + +static struct i2c_driver rt5575_i2c_driver = { + .driver = { + .name = "rt5575", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(rt5575_of_match), + }, + .probe = rt5575_i2c_probe, + .id_table = rt5575_i2c_id, +}; +module_i2c_driver(rt5575_i2c_driver); + +MODULE_DESCRIPTION("ASoC ALC5575 driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5575.h b/sound/soc/codecs/rt5575.h new file mode 100644 index 000000000000..11149612274a --- /dev/null +++ b/sound/soc/codecs/rt5575.h @@ -0,0 +1,54 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5575.h -- ALC5575 ALSA SoC audio driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#ifndef __RT5575_H__ +#define __RT5575_H__ + +#define RT5575_DEVICE_ID 0x10ec5575 +#define RT5575_DSP_MAPPING 0x18000000 + +#define RT5575_ID 0x8008 +#define RT5575_ID_1 0x800c +#define RT5575_MIXL_VOL 0x8a14 +#define RT5575_MIXR_VOL 0x8a18 +#define RT5575_PROMPT_VOL 0x8a84 +#define RT5575_SPK01_VOL 0x8a88 +#define RT5575_SPK23_VOL 0x8a8c +#define RT5575_MIC1_VOL 0x8a98 +#define RT5575_MIC2_VOL 0x8a9c +#define RT5575_WNC_CTRL 0x80ec +#define RT5575_MODE_CTRL 0x80f0 +#define RT5575_I2S_RATE_CTRL 0x80f4 +#define RT5575_SLEEP_CTRL 0x80f8 +#define RT5575_ALG_BYPASS_CTRL 0x80fc +#define RT5575_PINMUX_CTRL_2 0x81a4 +#define RT5575_GPIO_CTRL_1 0x8208 +#define RT5575_DSP_BUS_CTRL 0x880c +#define RT5575_SW_INT 0x0018 +#define RT5575_DSP_BOOT_ERR 0x8e14 +#define RT5575_DSP_READY 0x8e24 +#define RT5575_DSP_CMD_ADDR 0x8e28 +#define RT5575_EFUSE_DATA_2 0xc638 +#define RT5575_EFUSE_DATA_3 0xc63c +#define RT5575_EFUSE_PID 0xc660 + +enum { + RT5575_AIF1, + RT5575_AIF2, + RT5575_AIF3, + RT5575_AIF4, + RT5575_AIFS, +}; + +struct rt5575_priv { + struct i2c_client *i2c; + struct snd_soc_component *component; + struct regmap *dsp_regmap, *regmap; +}; + +#endif /* __RT5575_H__ */ -- 2.51.1

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by * Neustradamus * 07 Nov '25

07 Nov '25

Dear ALSA team, Can you solve the problem of e-mails always detected in "Junk Emails"/"Spams" folder? SPF/DKIM/DMARC For example, a part of several e-mail logs: ... Authentication-Results: spf=pass (sender IP is XXX.XXX.XXX.XXX) smtp.mailfrom=alsa-project.org; dkim=pass (signature was verified) header.d=alsa-project.org;dkim=timeout (key query timeout) header.d=leemhuis.info;dmarc=temperror action=none header.from=leemhuis.info; Received-SPF: Pass (protection.outlook.com: domain of alsa-project.org designates XXX.XXX.XXX.XXX as permitted sender) receiver=protection.outlook.com; client-ip=XXX.XXX.XXX.XXX; helo=alsa0.perex.cz; pr=C ... X-SID-Result: FAIL ... ------------------ ... Authentication-Results: spf=pass (sender IP is XXX.XXX.XXX.XXX) smtp.mailfrom=alsa-project.org; dkim=pass (signature was verified) header.d=alsa-project.org;dmarc=fail action=none header.from=gmail.com;compauth=fail reason=001 Received-SPF: Pass (protection.outlook.com: domain of alsa-project.org designates XXX.XXX.XXX.XXX as permitted sender) receiver=protection.outlook.com; client-ip=XXX.XXX.XXX.XXX; helo=alsa0.perex.cz; pr=C ... ------------------ ... ARC-Seal: i=2; a=rsa-sha256; s=arcselector10001; d=microsoft.com; cv=fail; ... ARC-Authentication-Results: i=2; mx.microsoft.com 1; spf=pass (sender ip is XXX.XXX.XXX.XXX) smtp.rcpttodomain=hotmail.com smtp.mailfrom=alsa-project.org; dmarc=pass (p=none sp=none pct=100) action=none header.from=intel.com; dkim=pass (signature was verified) header.d=alsa-project.org; dkim=pass (signature was verified) header.d=intel.com; arc=fail (48) ... Authentication-Results: spf=pass (sender IP is XXX.XXX.XXX.XXX) smtp.mailfrom=alsa-project.org; dkim=pass (signature was verified) header.d=alsa-project.org;dkim=pass (signature was verified) header.d=intel.com;dmarc=pass action=none header.from=intel.com;compauth=pass reason=100 Received-SPF: Pass (protection.outlook.com: domain of alsa-project.org designates XXX.XXX.XXX.XXX as permitted sender) receiver=protection.outlook.com; client-ip=XXX.XXX.XXX.XXX; helo=alsa0.perex.cz; pr=C ... ---------------------- Thanks in advance. Regards, Neustradamus

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[PATCH v1] ASoc: tas2783A: Fix issues in firmware parsing
by Niranjan H Y 07 Nov '25

07 Nov '25

During firmware download, if the size of the firmware is too small, it wrongly assumes the firmware download is successful. If there is size mismatch with chunk's header, invalid memory is accessed. Fix these issues by throwing error during these cases. Fixes: 4cc9bd8d7b32 (ASoc: tas2783A: Add soundwire based codec driver) Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <error27(a)gmail.com> Closes: https://lore.kernel.org/r/202510291226.2R3fbYNh-lkp@intel.com/ Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> --- sound/soc/codecs/tas2783-sdw.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c index c8f925e5473d..43b779873b93 100644 --- a/sound/soc/codecs/tas2783-sdw.c +++ b/sound/soc/codecs/tas2783-sdw.c @@ -761,10 +761,17 @@ static void tas2783_fw_ready(const struct firmware *fmw, void *context) goto out; } - mutex_lock(&tas_dev->pde_lock); img_sz = fmw->size; buf = fmw->data; offset += FW_DL_OFFSET; + if (offset >= (img_sz - FW_FL_HDR)) { + dev_err(tas_dev->dev, + "firmware is too small"); + ret = -EINVAL; + goto out; + } + + mutex_lock(&tas_dev->pde_lock); while (offset < (img_sz - FW_FL_HDR)) { memset(&hdr, 0, sizeof(hdr)); offset += read_header(&buf[offset], &hdr); @@ -775,6 +782,14 @@ static void tas2783_fw_ready(const struct firmware *fmw, void *context) /* size also includes the header */ file_blk_size = hdr.length - FW_FL_HDR; + /* make sure that enough data is there */ + if (offset + file_blk_size > img_sz) { + ret = -EINVAL; + dev_err(tas_dev->dev, + "corrupt firmware file"); + break; + } + switch (hdr.file_id) { case 0: ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, @@ -807,7 +822,8 @@ static void tas2783_fw_ready(const struct firmware *fmw, void *context) break; } mutex_unlock(&tas_dev->pde_lock); - tas2783_update_calibdata(tas_dev); + if (!ret) + tas2783_update_calibdata(tas_dev); out: if (!ret) -- 2.25.1

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Fwd: [Bug 220109] New: Audio disappears on HP 15-fc000 after warm boot again (ALC256 code path)
by Ruslan Nikolaev 06 Nov '25

06 Nov '25

Hi, Forwarding my regression bug report for HP 15-fc000 / ALC236 (ALC256 code path). #regzbot introduced: v6.12.2 https://bugzilla.kernel.org/show_bug.cgi?id=220109 Sincerely, Ruslan Forwarded Message (see logs in the bug report above): There was a similar bug in the past (Bug 217440), which was fixed for this laptop. (I believe the code path is for ALC256, see below.) The same issue is occurring again as of kernel v.6.12.2. The symptoms are very similar - initially audio works but after a warm reboot, the audio completely disappears until the computer is powered off (there is no audio output at all). The issue is caused by a different change now. By bisecting different kernel versions, I found that reverting cc3d0b5dd989d3238d456f9fd385946379a9c13d in patch_realtek.c ( see https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git/commit/sou… ) restores the sound and it works fine after the reboot.

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ucm2: Add setting LED Mode in SetLED macro
by GitHub pull_request - opened 06 Nov '25

06 Nov '25

alsa-project/alsa-ucm-conf pull request #637 was opened from mstrozek: Add 'Mode' argument to SetLED macro in order to allow setting a different LED mode. This will allow mute switches with inverted values to control the LEDs with 'follow-route' option instead of default 'follow-mute'. Update all calls to SetLED with appropriate Mode argument. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/637 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/637.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ALSA: hda/senary: Replace magic numbers with defined constants
by wangdich9700＠163.com 06 Nov '25

06 Nov '25

From: wangdicheng <wangdich9700(a)163.com> Replace hardcoded GPIO node value with a defined constant for better code readability and maintainability. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/hda/codecs/senarytech.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/hda/codecs/senarytech.c b/sound/hda/codecs/senarytech.c index 99af8d5e51db..d61d57538bc9 100644 --- a/sound/hda/codecs/senarytech.c +++ b/sound/hda/codecs/senarytech.c @@ -19,6 +19,9 @@ #include "hda_jack.h" #include "generic.h" +/* GPIO node ID */ +#define SENARY_GPIO_NODE 0x01 + struct senary_spec { struct hda_gen_spec gen; @@ -131,11 +134,11 @@ static void senary_init_gpio_led(struct hda_codec *codec) 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, + snd_hda_codec_write(codec, SENARY_GPIO_NODE, 0, AC_VERB_SET_GPIO_MASK, mask); - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, + snd_hda_codec_write(codec, SENARY_GPIO_NODE, 0, AC_VERB_SET_GPIO_DIRECTION, mask); - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + snd_hda_codec_write(codec, SENARY_GPIO_NODE, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_led); } } -- 2.25.1

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[alsa-devel] MOTU Midi Express XT
by holger＠dehnhardt.org 06 Nov '25

06 Nov '25

I own a MOTU Midi Express XT 8x8 interface which is connected through the parallel port. Recently I upgraded my system and the new motherboard does not have an ISA parallel port anymore... So I bought a PCI parallel port extension card - hoping that it will work with the ALSA driver (mtpav). So I changed the interrupt an port address in modules.conf but the driver is no longer recognized as an ALSA driver even though it loads without errors. As the device still perfectly fits my needs I would like to modify the driver to get it working again. Can anyone give me advice what should be changed in the drivers code to make it work on a pci port? Thanks Holger

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