March 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH 00/21] ASoC: Intel: boards: updates for 6.10 - part1
by Pierre-Louis Bossart 26 Mar '24

26 Mar '24

We have quite a few machine driver patches, spread in 3 series of about 20 patches each. This first part from Brent Lu adds common helpers and board configurations to reduce the number of quirks. Brent Lu (21): ASoC: Intel: board_helpers: support DAI link ID customization ASoC: Intel: sof_ssp_amp: use common module for DAI link generation ASoC: Intel: board_helpers: change dai link helpers to static function ASoC: Intel: sof_da7219: add rpl_mx98360_da7219 board config ASoC: Intel: sof_rt5682: support ALC5650 on RPL boards ASoC: Intel: sof_cs42l42: rename BT offload quirk ASoC: Intel: board_helpers: support sof_card_private initialization ASoC: Intel: sof_cs42l42: use common module for sof_card_private initialization ASoC: Intel: sof_nau8825: use common module for sof_card_private initialization ASoC: Intel: sof_rt5682: use common module for sof_card_private initialization ASoC: Intel: sof_ssp_amp: use common module for sof_card_private initialization ASoC: Intel: sof_da7219: use common module for DAI link generation ASoC: Intel: sof_da7219: add codec exit function ASoC: Intel: sof_da7219: add SOF_DA7219_MCLK_EN quirk ASoC: Intel: sof_da7219: board id cleanup for jsl boards ASoC: Intel: sof_da7219: board id cleanup for adl boards ASoC: Intel: sof_da7219: board id cleanup for rpl boards ASoC: Intel: sof_rt5682: remove unnecessary idisp HDMI quirk ASoC: Intel: sof_ssp_amp: remove unnecessary idisp HDMI quirk ASoC: Intel: sof_nau8825: remove sof_nau8825 board id ASoC: Intel: sof_rt5682: board id cleanup for cml boards sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_board_helpers.c | 146 ++++--- sound/soc/intel/boards/sof_board_helpers.h | 97 ++++- sound/soc/intel/boards/sof_cs42l42.c | 72 +-- sound/soc/intel/boards/sof_da7219.c | 410 ++++++------------ sound/soc/intel/boards/sof_nau8825.c | 92 +--- sound/soc/intel/boards/sof_rt5682.c | 222 ++++------ sound/soc/intel/boards/sof_ssp_amp.c | 264 ++++------- .../intel/common/soc-acpi-intel-adl-match.c | 4 +- .../intel/common/soc-acpi-intel-cml-match.c | 6 +- .../intel/common/soc-acpi-intel-jsl-match.c | 4 +- .../intel/common/soc-acpi-intel-rpl-match.c | 12 + 12 files changed, 518 insertions(+), 812 deletions(-) -- 2.40.1

2 22

[PATCH v1 0/2] ASoC: codecs: ES8326: Reducin powerconsumption and
by Zhang Yi 26 Mar '24

26 Mar '24

We changed the configuration related to hibernation. and delete the REG_SUPPLY to cover mute issue. Zhang Yi (2): ASoC: codecs: ES8326: Reducing power consumption ASoC: codecs: ES8326: Delete unused REG_SUPPLY sound/soc/codecs/es8326.c | 41 ++++++++++++++++++++++++++++++--------- 1 file changed, 32 insertions(+), 9 deletions(-) -- 2.17.1

2 3

[PATCH v3 00/22] SoC: Cleanup MediaTek soundcard machine drivers
by AngeloGioacchino Del Regno 26 Mar '24

26 Mar '24

Changes in v3: - Added audio-routing names in enum in all yaml files - Added mention of disallowing old and new properties together in commit message of bindings patches - Fixed validation errors with sound-card-common.yaml inclusion due to missing model property in examples - Removed `else` enforcing headset-codec/speaker-codecs requirement if xxx-dai-link not present to avoid future commit noise as the deprecated statement will disallow deprecated properties as required Changes in v2: - Bindings: Changed link-name/codec/clk-provider to remove `items` and leave just the enum - Moved .*-dai-link pattern additionalProperties after `type: object` - Added ref to sound-card-common.yaml - Fixed dai-link-xxx -> xxx-dai-link typo in example comment This series performs a cleanup of most of the MediaTek AFE drivers and soundcard machine drivers, reducing code duplication and setting a base to be consistent with their devicetree bindings, as many of those are using different properties and nodes for no good reason. Summarizing: - Commonizes functions and ops where possible - Adds a common probe mechanism, increasing maintainability of soundcard drivers for older MediaTek SoCs - Migrates all drivers to support the new bindings - Obviously retains compatibility with old device trees - Reduces machine-specific parameters hardcoding in drivers - Can now set machine-specific params in device tree - Uses the `audio-routing` and `dai-link` nodes like some other non-MediaTek SoC sound drivers - Imposes consistency between MediaTek ASoC machine soundcard drivers bindings - Reduces code size and greatly reduces the amount of code that will be required for newer drivers (retaining compatibility with the old bindings was costly in terms of code size, otherwise this series would've removed ~1000 more lines, or something along that line). This series was (manually) tested on MT8173, MT8192, MT8195 and MT8186 Chromebooks. AngeloGioacchino Del Regno (22): ASoC: mediatek: mt8192-afe-pcm: Convert to devm_pm_runtime_enable() ASoC: mediatek: mt8192-afe-pcm: Simplify with dev_err_probe() ASoC: mediatek: Commonize ADDA rate transform functions and enums ASoC: mediatek: Assign dummy when codec not specified for a DAI link ASoC: mediatek: Add common machine soundcard driver probe mechanism ASoC: mediatek: common: Constify struct mtk_sof_priv ASoC: mediatek: mt8188: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8195: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8192: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8186: Migrate to mtk_soundcard_common_probe ASoC: mediatek: Add common snd_soc_ops .startup() callback ASoC: mediatek: mt8195: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8192: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8186-rt1019: Migrate to the common mtk_soundcard_startup ASoC: mediatek: Add common mtk_afe_component_probe callback ASoC: mediatek: Use common mtk_afe_pcm_platform with common probe cb ASoC: mediatek: mt8186: Unify mt8186-mt6366 machine drivers ASoC: dt-bindings: mt8195: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8192: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8186: Document audio-routing and dai-link subnode arm64: dts: mediatek: mt8195-cherry: Specify sound DAI links and routing arm64: dts: mediatek: mt8186-corsola: Specify sound DAI links and routing .../sound/mt8186-mt6366-da7219-max98357.yaml | 131 +- .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 120 +- .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 139 +- .../bindings/sound/mt8195-mt6359.yaml | 134 ++ .../boot/dts/mediatek/mt8186-corsola.dtsi | 42 +- .../boot/dts/mediatek/mt8195-cherry.dtsi | 45 + sound/soc/mediatek/Kconfig | 24 +- sound/soc/mediatek/common/Makefile | 2 + .../mediatek/common/mtk-afe-platform-driver.c | 18 + .../soc/mediatek/common/mtk-dai-adda-common.c | 70 + .../soc/mediatek/common/mtk-dai-adda-common.h | 45 + .../soc/mediatek/common/mtk-dsp-sof-common.c | 15 +- .../soc/mediatek/common/mtk-dsp-sof-common.h | 1 - sound/soc/mediatek/common/mtk-soc-card.h | 7 +- .../mediatek/common/mtk-soundcard-driver.c | 205 ++- .../mediatek/common/mtk-soundcard-driver.h | 42 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 14 +- sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 85 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 90 +- sound/soc/mediatek/mt8186/Makefile | 3 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 19 +- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 94 +- .../mt8186/mt8186-mt6366-da7219-max98357.c | 1189 ----------------- ...t6366-rt1019-rt5682s.c => mt8186-mt6366.c} | 578 ++++---- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 21 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 89 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 203 +-- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 114 +- sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 90 +- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 301 ++--- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 21 +- sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 90 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 487 +++---- 35 files changed, 1781 insertions(+), 2775 deletions(-) create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.c create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.h delete mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c rename sound/soc/mediatek/mt8186/{mt8186-mt6366-rt1019-rt5682s.c => mt8186-mt6366.c} (72%) -- 2.44.0

3 28

[PATCH v2 0/5] ASoC: Harden DAPM route checks and Intel fixes
by Cezary Rojewski 26 Mar '24

26 Mar '24

Set of loosely connected patches. Most impactful change is dropping any permisiveness when snd_soc_dapm_add_routes() fails in soc-topology.c To do it safely, disable route checks for all skylake-driver boards. Relevant background: Since commit daa480bde6b3 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()") route checks are no longer permissive. Probe failures for Intel boards have been partially addressed by commit a22ae72b86a4 ("ASoC: soc-core: isable route checks for legacy devices") and its follow up but only skl_nau88l25_ssm4567.c is patched. The rest of the boards still need fixing. After that, removal of copy-pastas found in ssm4567.c and redundant code in i2s_test.c for avs-boards. Changes in v2: - glk_rt5682_max98357a.c and skl_hda_dsp_generic.c now disable route checks only for the skylake-drvier - asoc now logs failures of snd_soc_dapm_add_routes() in soc-topology.c Amadeusz Sławiński (1): ASoC: Intel: avs: i2s_test: Remove redundant dapm routes Cezary Rojewski (4): ASoC: Intel: Disable route checks for Skylake boards ASoC: topology: Do not ignore route checks when parsing graphs ASoC: Intel: avs: ssm4567: Do not ignore route checks ASoC: Intel: avs: ssm4567: Board cleanup sound/soc/intel/avs/boards/i2s_test.c | 79 ------------------- sound/soc/intel/avs/boards/ssm4567.c | 5 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 1 + sound/soc/intel/boards/bxt_rt298.c | 1 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 + sound/soc/intel/boards/kbl_da7219_max98357a.c | 1 + sound/soc/intel/boards/kbl_da7219_max98927.c | 4 + sound/soc/intel/boards/kbl_rt5660.c | 1 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 + .../intel/boards/kbl_rt5663_rt5514_max98927.c | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 + .../soc/intel/boards/skl_nau88l25_max98357a.c | 1 + sound/soc/intel/boards/skl_rt286.c | 1 + sound/soc/soc-topology.c | 11 ++- 14 files changed, 27 insertions(+), 85 deletions(-) -- 2.25.1

3 8

[PATCH v3 0/5] ASoC: fsl: Support register and unregister rpmsg sound card through remoteproc
by Chancel Liu 26 Mar '24

26 Mar '24

echo /lib/firmware/fw.elf > /sys/class/remoteproc/remoteproc0/firmware (A) echo start > /sys/class/remoteproc/remoteproc0/state (B) echo stop > /sys/class/remoteproc/remoteproc0/state The rpmsg sound card is registered in (A) and unregistered in (B). After "start", imx-audio-rpmsg registers devices for ASoC platform driver and machine driver. Then sound card is registered. After "stop", imx-audio-rpmsg unregisters devices for ASoC platform driver and machine driver. Then sound card is unregistered. changes in v2 - Fix build errors reported by kernel test robot changes in v3 - Add a new patch for fsl_rpmsg to register CPU DAI with rpmsg channel name - Update imx-rpmsg.c to get DT node of ASoC CPU DAI device with rpmsg channel name instead of using undocumented bindings Chancel Liu (5): ASoC: fsl: imx-pcm-rpmsg: Register component with rpmsg channel name ASoC: fsl: imx-audio-rpmsg: Register device with rpmsg channel name ASoC: fsl: Let imx-audio-rpmsg register platform device for card ASoC: fsl: fsl_rpmsg: Register CPU DAI with name of rpmsg channel ASoC: fsl: imx-rpmsg: Update to correct DT node sound/soc/fsl/fsl_rpmsg.c | 43 ++++++++++++++++++++------------- sound/soc/fsl/imx-audio-rpmsg.c | 21 +++++++++++++--- sound/soc/fsl/imx-pcm-rpmsg.c | 11 ++++++--- sound/soc/fsl/imx-rpmsg.c | 28 ++++++++++++++------- 4 files changed, 71 insertions(+), 32 deletions(-) -- 2.43.0

3 7

[PATCH 0/7] ASoC: mediatek: AFE drivers cleanups
by AngeloGioacchino Del Regno 26 Mar '24

26 Mar '24

This series converts MediaTek AFE drivers' probe functions to use dev_err_probe() and devm functions where possible and, in some cases, dropping the .remove_new() callback, reducing the size. Cheers! AngeloGioacchino Del Regno (7): ASoC: mediatek: mt8173-afe-pcm: Convert to devm_pm_runtime_enable() ASoC: mediatek: mt8173-afe-pcm: Use devm_snd_soc_register_component() ASoC: mediatek: mt8183-afe-pcm: Convert to devm_pm_runtime_enable() ASoC: mediatek: mt8183-afe-pcm: Simplify with dev_err_probe() ASoC: mediatek: mt8192-afe-pcm: Convert to devm_pm_runtime_enable() ASoC: mediatek: mt8192-afe-pcm: Simplify with dev_err_probe() ASoC: mediatek: mt8195-afe-pcm: Drop .remove_new() callback sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 78 +++------------ sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 108 +++++++-------------- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 95 ++++++------------ sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 12 +-- 4 files changed, 81 insertions(+), 212 deletions(-) -- 2.43.0

5 15

[PATCH v11] ASoc: tas2783: Add tas2783 codec driver
by Shenghao Ding 26 Mar '24

26 Mar '24

The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp speaker protection algorithm. The integrated speaker voltage and current sense provides for real-time monitoring of loudspeakers. The ASoC component provides the majority of the functionality of the device, all the audio functions. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v11: - simplify tasdevice_set_sdw_stream - fixed some Linux coding style - fixed the spelling mistakes - Select left/right channel based on unique id - a longer description has been added - remove unused crc8 in KCONFIG - Invert of RE here means reciprocal of RE - correct indentation - correct the wrong pow calibrated register - correct comment on RESET register - fixed usually lines don't end with an open parenthesis - fixed tasdevice_sdw_shutdown - use defines for function, entity, etc. for readable - remove unnecessary codec_lock in component_probe() - remove struct sdw_stream_data - remove unused macro definitions - use BIT(2) and BIT(1) instead of 0x2 and 0x1 - implement pm_runtime - reg_start is u8, reg_start + i only adds i byte, it should be reg_start + i * 4, because the calibration values - fixed total_sz in tas2783_calibration - add a define for TASDEVICE_REG(0, 0, 0x0a) - rework macro style in header file - Add select CRC32 - rewrite the mute and the return value handling - Add comments on bin file - Add link_id into the fw name to distinguish the same sdw id on different sdw id. - remove unnecessary mutex_lock - use set_active() in tasdevice_io_init and move all other parts in the probe. - move pm_runtime_get_noresume to the probe - fix the return value of putvol function - add put_noidle in tasdevice_sdw_remove - remove unused header file(of.h) - correct the style of the multi-line comment - Set the readable registers in tas2783_readable_register for debugfs - Use new PM macros - Use default put and get inline SOC_SINGLE_RANGE_TLV and SOC_SINGLE_SX_TLV - Use respective %pt to log the calibrated time - Remove unnecessary struct tm - Add trailing comma tasdevice_dapm_widgets and tasdevice_audio_map - Rewrite the return value handling in tasdevice_sdw_probe - use dev_num instead of id.unique_id - rework the log - Add code to check the size of calibrated data in UEFI - Add description of Layout of calibrated Data in UEFI - switch back link_id and unique_id to identify the devcie - unique_id offset as the device index of the calibrated data - update calibrated data layout stored in UEFI, add link_id and unique_id into the calibrated data in case of tas2783 sitting in different SDQW buses. - correct the wrong unique_id - set the unique_id as hex style in dsp bin file name - use __func__ in the dev_err message in two identical logs intas2783_apply_calibv1/v2. - In tas2783_apply_calibv2, the very least move initializations to a different line. - No need for separate j variable, reuse i in tasdevice_read_prop. - remove unused "struct sdw_bus_params params", unsigned char dev_name[32] pstream, and chip_id - define dspfw_binaryname as a local variable in tasdevice_comp_probe. - reduce a lot of indentation in tasdevice_mute and spilte tasdevice_mute into tasdevice_enable_echo_ref and tasdevice_playback - remove tas2783_reg_defaults - use REGCACHE_MAPLE instead of REGCACHE_RBTREE - define fw_node_sz to store the node size --- sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 893 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 107 ++++ 4 files changed, 1018 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f78ea2f86fa6..939384837d5e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -248,6 +248,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1880,6 +1881,21 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP + select REGMAP_SOUNDWIRE + select CRC32 + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + TAS2783 supports the MIPI SoundWire v1.2 compatible interface + for audio and control data. At the end of calibrated data appends + the crc data with crc32_sarwate for error detection. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7c075539dc47..6f27f810affc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -287,6 +287,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -672,6 +673,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..2844cc5189dc --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,893 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Kevin Lu <kevin-lu(a)ti.com> +// Shenghao Ding <shenghao-ding(a)ti.com> +// + +#include <asm/unaligned.h> +#include <linux/acpi.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include <sound/tas2781-tlv.h> +#include "tas2783.h" + +struct calibdatav2_info { + unsigned int number_of_devices; + unsigned int crc32_indx; + unsigned int byt_sz; + unsigned int *cali_data; +}; + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, /* Resistance */ + TAS2783_CALIBRATION_RE_LOW, /* Low limitation of RE */ + TAS2783_CALIBRATION_INV_RE, /* Reciprocal of RE */ + TAS2783_CALIBRATION_POW, /* RMS Power */ + TAS2783_CALIBRATION_TLIMIT /* Temperature limitation */ +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + /* Page 0 */ + case 0x8000 ... 0x807F: + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + /* + * Only software reset register was volatiled. + * Bit is self clearing. + * 0b = Don't reset + * 1b = reset + */ + case 0x8001: + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x44ffffff, + .cache_type = REGCACHE_MAPLE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, dvc_tlv), + SOC_SINGLE_SX_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, amp_vol_tlv), +}; + +/* + * Load the calibration data, including speaker impedance, f0, etc. + * Calibration is done by the manufacturer in the factory. The calibration + * data are used by the algorithm for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * The DSP will work with default data values if calibrated data are + * missing or are invalid. + * Layout of calibrated Data V1 in UEFI(total 256 bytes): + * Calibrated Data of Dev 0(unique_id offset 0) (20 bytes) + * Calibrated Data of Dev 1(unique_id offset 1) (20 bytes) + * Calibrated Data of Dev 2(unique_id offset 2) (20 bytes) + * Calibrated Data of Dev 3(unique_id offset 3) (20 bytes) + * Calibrated Data of Dev 4(unique_id offset 4) (20 bytes) + * Calibrated Data of Dev 5(unique_id offset 5) (20 bytes) + * Calibrated Data of Dev 6(unique_id offset 6) (20 bytes) + * Calibrated Data of Dev 7(unique_id offset 7) (20 bytes) + * TimeStamp of Calibration (4 bytes) + * CRC (4 bytes) + * Reserved (88 bytes) + */ +static void tas2783_apply_calibv1(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *cali_start; + u16 offset; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + offset = tas_dev->sdw_peripheral->id.unique_id - + TAS2783_UNIQUE_ID_BASE; + if (offset >= TAS2783_MAX_DEV_NUM) { + dev_err(tas_dev->dev, "%s: offset(%u) is out of range.\n", + __func__, offset); + return; + } + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)(cali_data + offset * sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + &cali_start[4 * i], 4); + if (ret) { + dev_err(tas_dev->dev, "%s: Cali failed %x:%d\n", + __func__, tas2783_cali_reg[i], ret); + break; + } + } +} + +/* + * New features were added in calibrated Data V2: + * 1. Added some the fields to store the link_id and uniqie_id for + * multi-link solutions + * 2. Support flexible number of devices instead of fixed one in V1. + * Layout of calibrated data V2 in UEFI(total 256 bytes): + * ChipID (2783, 4 bytes) + * Device-Sum (4 bytes) + * TimeStamp of Calibration (4 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info () { + * SDW link id (2bytes) + * SDW unique_id (2bytes) + * } + * Calibrated Data of Device #i (20 bytes) + * } + * CRC (4 bytes) + * Reserved (the rest) + */ +static void tas2783_apply_calibv2(struct tasdevice_priv *tas_dev, + const struct calibdatav2_info *p) +{ + const unsigned int arr_size = ARRAY_SIZE(tas2783_cali_reg); + struct regmap *map = tas_dev->regmap; + unsigned int i, j, k; + u8 *cali_start; + u16 dev_indx; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + dev_indx = tas_dev->sdw_peripheral->bus->link_id | + tas_dev->sdw_peripheral->id.unique_id << 16; + + for (i = 0; i < p->number_of_devices; i++) { + k = i * (arr_size + 1); + + if (dev_indx != p->cali_data[k]) + continue; + + /* skip 4 byte dev-indx */ + cali_start = (u8 *)&p->cali_data[k + 1]; + + for (j = 0; j < arr_size; j++) { + k = 4 * j; + ret = regmap_bulk_write(map, tas2783_cali_reg[j], + &cali_start[k], 4); + if (ret) { + dev_err(tas_dev->dev, + "%s: Cali failed %x:%d\n", __func__, + tas2783_cali_reg[j], ret); + break; + } + } + return; + } +} + +static void tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct calibration_data cali_data; + unsigned int *tmp_val; + unsigned int crc; + efi_status_t status; + + cali_data.total_sz = 0; + + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, NULL); + if (status == EFI_BUFFER_TOO_SMALL + && cali_data.total_sz < TAS2783_MAX_CALIDATA_SIZE) { + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, + cali_data.data); + dev_dbg(tas_dev->dev, "%s: cali get %lx bytes result:%ld\n", + __func__, cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return; + } + + tmp_val = (unsigned int *)cali_data.data; + + /* Check Calibrated Data V1 */ + crc = crc32(~0, cali_data.data, TAS2783_CALIDATAV1_BYTE_SIZE) ^ ~0; + if (crc == tmp_val[TAS2783_CALIDATAV1_CRC32_INDX]) { + /* Date and time of when calibration was done. */ + tas2783_apply_calibv1(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "V1: %ptTs", + &tmp_val[TAS2783_CALIDATAV1_TIMESTAMP_INDX]); + return; + } + + /* Check Calibrated Data V2 */ + if (tmp_val[0] == 2783) { + const struct calibdatav2_info calib_info = { + .number_of_devices = tmp_val[1], + .crc32_indx = 3 + tmp_val[1] * 6, + .byt_sz = 12 + tmp_val[1] * 24, + .cali_data = &tmp_val[3] + }; + + if (calib_info.number_of_devices > TAS2783_MAX_DEV_NUM || + calib_info.number_of_devices == 0) { + dev_dbg(tas_dev->dev, "No dev in calibrated data V2."); + return; + } + crc = crc32(~0, cali_data.data, calib_info.byt_sz) + ^ ~0; + if (crc == tmp_val[calib_info.crc32_indx]) { + tas2783_apply_calibv2(tas_dev, &calib_info); + dev_dbg(tas_dev->dev, "V2: %ptTs", + &tmp_val[TAS2783_CALIDATAV2_TIMESTAMP_INDX]); + } else { + dev_dbg(tas_dev->dev, + "V2: CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[calib_info.crc32_indx]); + } + } else { + dev_err(tas_dev->dev, "Non-2783 chip\n"); + } +} + +static void tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + struct regmap *map = tas_dev->regmap; + struct tas2783_firmware_node *p; + unsigned char *buf; + int fw_node_sz = sizeof(struct tas2783_firmware_node); + int img_sz, ret; + int offset = 0; + + if (!fmw || !fmw->data) { + dev_warn(tas_dev->dev, + "%s: failed to read fw, work in bypass-dsp mode.\n", + __func__); + return; + } + buf = (unsigned char *)fmw->data; + + img_sz = get_unaligned_le32(&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_warn(tas_dev->dev, "%s: size not matching, %d %u.", + __func__, (int)fmw->size, img_sz); + return; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + fw_node_sz, p->length); + else + ret = regmap_write(map, p->download_addr, + *(buf + offset + fw_node_sz)); + + if (ret != 0) { + dev_dbg(tas_dev->dev, + "%s: load FW fail: %d, work in bypass.\n", + __func__, ret); + return; + } + offset += fw_node_sz + p->length; + } + + tas2783_calibration(tas_dev); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC"), +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"}, +}; + +static int tasdevice_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + int ret; + + dev_dbg(dai->dev, "%s: dai_name %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); + return ret; + } + + dev_dbg(dai->dev, "%s: format: %i rate: %i\n", __func__, + params_format(params), params_rate(params)); + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, sdw_stream); + + return 0; +} + +/* Capture stream is the echo ref data for voice mode.*/ +static int tasdevice_enable_echo_ref(struct tasdevice_priv *tas_dev, + int mute) +{ + struct regmap *map = tas_dev->regmap; + unsigned int val; + int ret; + + if (mute) + val = TAS2783_REG_AEF_INACTIVE; + else + val = TAS2783_REG_AEF_ACTIVE; + + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, TAS2783_REG_AEF_INACTIVE); + if (ret) + dev_err(tas_dev->dev, "%s: Disable AEF failed.\n", __func__); + + return ret; +} + +static int tasdevice_playback(struct tasdevice_priv *tas_dev, int mute) +{ + struct regmap *map = tas_dev->regmap; + int ret; + + if (mute) { + /* FU23 mute (0x40400108) */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 1); + if (ret) { + dev_err(tas_dev->dev, "%s: FU23 mute failed.\n", + __func__); + goto out; + } + + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK | + TAS2783_REG_AEF_MASK, + TAS2783_REG_PWR_MODE_ACTIVE | + TAS2783_REG_PWR_MODE_SW_PWD); + if (ret) { + dev_err(tas_dev->dev, "%s: shutdown failed.\n", + __func__); + } + } else { + /* FU23 Unmute, 0x40400108. */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 Unmute failed.\n", __func__); + goto out; + } + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK, + TAS2783_REG_PWR_MODE_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR Unmute failed.\n", __func__); + } + } +out: + return ret; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + + int ret; + + dev_dbg(tas_dev->dev, "%s: %d.\n", __func__, mute); + + if (direction == SNDRV_PCM_STREAM_CAPTURE) + ret = tasdevice_enable_echo_ref(tas_dev, mute); + else + ret = tasdevice_playback(tas_dev, mute); + + if (ret) + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_comp_probe(struct snd_soc_component *comp) +{ + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(comp); + unsigned char dspfw_binaryname[TAS2783_DSPFW_FILENAME_LEN]; + acpi_handle handle = ACPI_HANDLE(tas_dev->dev); + const struct firmware *fw_entry = NULL; + const char *sub = NULL; + int ret; + + if (handle) { + sub = acpi_get_subsystem_id(handle); + if (IS_ERR(sub)) + sub = NULL; + } + + tas_dev->component = comp; + + /* + * Each tas2783 in the system has its own dspfw. + */ + if (comp->name_prefix) { + /* + * name_prefix.bin stores the dsp firmware including speaker + * protection algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied into + * firmware folder. + */ + scnprintf(dspfw_binaryname, TAS2783_DSPFW_FILENAME_LEN, + "%s-tas2783.bin", comp->name_prefix); + } else { + /* Compatible with the previous naming rule */ + if (sub) { + /* + * subsys_id-link_id[0,1,...,N]-unique_id + * [0x8,...,0xf].bin stores the dsp firmware including + * speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, + * it must be copied into firmware folder. + */ + scnprintf(dspfw_binaryname, TAS2783_DSPFW_FILENAME_LEN, + "%s-%d-0x%x.bin", sub, + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } else { + /* + * tas2783-link_id[0,1,...,N]-unique_id + * [0x8,...,0xf].bin stores the dsp firmware including + * speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, + * it must be copied into firmware folder. + */ + scnprintf(dspfw_binaryname, TAS2783_DSPFW_FILENAME_LEN, + "tas2783-%d-0x%x.bin", + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } + } + + ret = request_firmware(&fw_entry, dspfw_binaryname, tas_dev->dev); + if (ret) { + dev_err(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", __func__, + tas_dev->sdw_peripheral->id.unique_id, ret); + goto out; + } + + tasdevice_dspfw_ready(fw_entry, tas_dev); + +out: + if (fw_entry) + release_firmware(fw_entry); + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_comp_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, tasdevice_dai_driver, + ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + return ret; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + + return 0; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + struct sdw_dpn_prop *dpn; + unsigned long addr; + int nval, i; + u32 bit; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = BIT(2); /* BITMAP: 00000100 */ + prop->sink_ports = BIT(1); /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, + struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(tas_priv->regmap, false); + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + * Update count of parent 'active' children. + */ + pm_runtime_set_active(&slave->dev); + } + + /* sw reset */ + ret = regmap_write(tas_priv->regmap, TAS2873_REG_SWRESET, + TAS2873_REG_SWRESET_RESET); + if (ret) { + dev_err(tas_priv->dev, "Reset failed.\n"); + goto out; + } + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, false); + regcache_mark_dirty(tas_priv->regmap); + } + + tas_priv->first_hw_init = true; + tas_priv->hw_init = true; + +out: + return ret; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + /* + * Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return -ENOMEM; + + tas_dev->dev = dev; + tas_dev->sdw_peripheral = peripheral; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) + return dev_err_probe(dev, PTR_ERR(tas_dev->regmap), + "Failed devm_regmap_init_sdw."); + + return tasdevice_init(tas_dev); +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev->first_hw_init) + pm_runtime_disable(tas_dev->dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {} +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static int __maybe_unused tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (!tas_priv->hw_init) + return 0; + + regcache_cache_only(tas_priv->regmap, true); + + return 0; +} + +#define TAS2783_PROBE_TIMEOUT_MS 5000 + +static int __maybe_unused tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long time; + + if (!tas_priv->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT_MS)); + if (!time) { + dev_err(&slave->dev, "Init not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(tas_priv->regmap, false); + regcache_sync(tas_priv->regmap); + + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume) + RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume, NULL) +}; + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = &tas2783_sdca_pm, + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..de72e07602bc --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,107 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2023 - 2024 Texas Instruments Incorporated + * https://www.ti.com + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + * Shenghao Ding <shenghao-ding(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +/* Unique id start */ +#define TAS2783_UNIQUE_ID_BASE 0x08 +#define TAS2783_MAX_DEV_NUM 8 + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 + \ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0x00, 0x01) +#define TAS2873_REG_SWRESET_MASK BIT(0) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +#define TAS2873_REG_PWR_CTRL TASDEVICE_REG(0x0, 0x00, 0x02) +#define TAS2783_REG_AEF_MASK GENMASK(4, 3) +#define TAS2783_REG_AEF_ACTIVE 0 +#define TAS2783_REG_AEF_INACTIVE GENMASK(4, 3) +#define TAS2783_REG_PWR_MODE_MASK GENMASK(2, 0) +#define TAS2783_REG_PWR_MODE_ACTIVE 0 +#define TAS2783_REG_PWR_MODE_SW_PWD BIT(1) + +#define TAS2783_REG_TDM_RX_CFG TASDEVICE_REG(0x0, 0x00, 0x0a) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1a) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x13, 0x70) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +/* TAS2783 SDCA Control - function number */ +#define TAS2783_FUNC_TYPE_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_FU_VOLUME 0x02 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 +#define TAS2783_DSPFW_FILENAME_LEN 64 + +#define TAS2783_MAX_CALIDATA_SIZE 256 + +#define TAS2783_CALIDATAV1_TIMESTAMP_INDX 40 +#define TAS2783_CALIDATAV1_CRC32_INDX 41 +/* 8 devs * 20 bytes calibrated data/dev + 4 bytes Timestamp */ +#define TAS2783_CALIDATAV1_BYTE_SIZE (8 * 20 + 4) + +#define TAS2783_CALIDATAV2_TIMESTAMP_INDX 2 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct snd_soc_component *component; + struct sdw_slave *sdw_peripheral; + struct regmap *regmap; + struct device *dev; + bool hw_init; + bool first_hw_init; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

3 3

[PATCH] ASoC: cs42l43: Correct extraction of data pointer in suspend/resume
by Charles Keepax 26 Mar '24

26 Mar '24

The current code is pulling the wrong pointer causing it to disable the wrong IRQ. Correct the code to pull the correct cs42l43 core data pointer. Fixes: 64353af49fec ("ASoC: cs42l43: Add system suspend ops to disable IRQ") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l43.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 860d5cda67bf..94685449f0f4 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2364,7 +2364,8 @@ static int cs42l43_codec_runtime_resume(struct device *dev) static int cs42l43_codec_suspend(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; disable_irq(cs42l43->irq); @@ -2373,7 +2374,8 @@ static int cs42l43_codec_suspend(struct device *dev) static int cs42l43_codec_suspend_noirq(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; enable_irq(cs42l43->irq); @@ -2382,7 +2384,8 @@ static int cs42l43_codec_suspend_noirq(struct device *dev) static int cs42l43_codec_resume(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; enable_irq(cs42l43->irq); @@ -2391,7 +2394,8 @@ static int cs42l43_codec_resume(struct device *dev) static int cs42l43_codec_resume_noirq(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; disable_irq(cs42l43->irq); -- 2.39.2

2 1

[PATCH v1 1/1] ASoC: fsl: imx-es8328: Remove leftover gpio initialisation
by Andy Shevchenko 26 Mar '24

26 Mar '24

The gpio field is not used anymore, remove the leftover. This also fixes the compilation error after the ... Fixes: 9855f05e5536 ("ASoC: fsl: imx-es8328: Switch to using gpiod API") Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/fsl/imx-es8328.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index d67b8a149bff..5b9648f3b087 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -28,7 +28,6 @@ struct imx_es8328_data { static struct snd_soc_jack_gpio headset_jack_gpios[] = { { - .gpio = -1, .name = "headset-gpio", .report = SND_JACK_HEADSET, .invert = 0, -- 2.43.0.rc1.1.gbec44491f096

2 1

[PATCH v2][next] ALSA: firewire-lib: Avoid -Wflex-array-member-not-at-end warning
by Gustavo A. R. Silva 26 Mar '24

26 Mar '24

Use the `DEFINE_FLEX()` helper for an on-stack definition of a flexible structure where the size of the flexible-array member is known at compile-time, and refactor the rest of the code, accordingly. So, with these changes, fix the following warning: sound/firewire/amdtp-stream.c:1184:46: warning: structure containing a flexible array member is not at the end of another structure [-Wflex-array-member-not-at-end] Link: https://github.com/KSPP/linux/issues/202 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- Changes in v2: - Use DEFINE_FLEX() helper instead of a new tagged struct. v1: - Link: https://lore.kernel.org/linux-hardening/d3a764eb76909b16b8a22d9ff530e5edf0e… sound/firewire/amdtp-stream.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index c9f153f85ae6..d35d0a420ee0 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1180,13 +1180,11 @@ static void process_rx_packets(struct fw_iso_context *context, u32 tstamp, size_ (void)fw_card_read_cycle_time(fw_parent_device(s->unit)->card, &curr_cycle_time); for (i = 0; i < packets; ++i) { - struct { - struct fw_iso_packet params; - __be32 header[CIP_HEADER_QUADLETS]; - } template = { {0}, {0} }; + DEFINE_FLEX(struct fw_iso_packet, template, header, + header_length, CIP_HEADER_QUADLETS); bool sched_irq = false; - build_it_pkt_header(s, desc->cycle, &template.params, pkt_header_length, + build_it_pkt_header(s, desc->cycle, template, pkt_header_length, desc->data_blocks, desc->data_block_counter, desc->syt, i, curr_cycle_time); @@ -1198,7 +1196,7 @@ static void process_rx_packets(struct fw_iso_context *context, u32 tstamp, size_ } } - if (queue_out_packet(s, &template.params, sched_irq) < 0) { + if (queue_out_packet(s, template, sched_irq) < 0) { cancel_stream(s); return; } -- 2.34.1

3 2

Scan hardware to find speakers not detected by OS
by Jacek Blocki 26 Mar '24

26 Mar '24

Hello, In Alsa-devel Digest, Vol 205, Issue 136 there is my message on internal spekaker and headphones not working on Huawei D14 laptop. SImilar issue was reported in Vol 205, Issue 142 (No sound "dummy output" - snd_hda_intel on Pop OS 22.04 LTS on Acer chromebook). Is there any tool/troubleshooting guide allowing to diagnose above problem? In my cases (Huawei laptop) i guess OS does not detect speakers. Is there any way to scan hardware to find components undetected by OS? Regards, Jacek Blocki

1 0

ucm2: MediaTek: mt8195-sof: Add support for Tomato RT5682s variant
by GitHub pull_request - opened 26 Mar '24

26 Mar '24

alsa-project/alsa-ucm-conf pull request #406 was opened from kholk: A later revision of the MT8195 Cherry Tomato Chromebook (Acer Chromebook Spin 513 CP513-2H, Revision 3 and 4) are using the RT5682s codec instead of RT5682i. The differences are only about a couple of missing switches, where the 'i' variant had a switch for the L/R Headphone output and a configurable DAC L/R Mux, while the 's' one misses the mux control and solely relies on the main Headphone Switch. This configuration has been tested on Arch Linux with Pipewire 10.03 + WirePlumber. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/406 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/406.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] ALSA/ASoC/SOF: SSP dev_type support
by Peter Ujfalusi 26 Mar '24

26 Mar '24

Hi, SOF always passed 0 as dev_type to intel_nhlt_get_endpoint_blob() when looking up the blob from the NHLT table. This causes issues since alsa-utils commit 3a47ef2487ed ("topology: nhlt: intel: support more device types and directions") The dev_type is no longer always 0 in the topology embedded NHLT table resulting lookup failures for analog codecs since they will have dev_type=4. With SOF the dev_type is not used, we always use the SSP port index for looking up the configuration blob. The solution for the issue is to fetch the dev_type for the SSP port and feed it back to the intel_nhlt_get_endpoint_blob() to be able to find the configuration regardless of the assigned dev_type. Regards, Peter --- Brent Lu (2): ALSA: hda: intel-nhlt: add intel_nhlt_ssp_device_type() function ASoC: SOF: ipc4-topology: support NHLT device type include/sound/intel-nhlt.h | 10 ++++++++++ sound/hda/intel-nhlt.c | 26 ++++++++++++++++++++++++++ sound/soc/sof/ipc4-topology.c | 19 ++++++++++++++++--- 3 files changed, 52 insertions(+), 3 deletions(-) -- 2.43.0

3 6

[PATCH] ASoC: cs-amp-lib: Check for no firmware controls when writing calibration
by Richard Fitzgerald 25 Mar '24

25 Mar '24

From: Simon Trimmer <simont(a)opensource.cirrus.com> When a wmfw file has not been loaded the firmware control descriptions necessary to write a stored calibration are not present. In this case print a more descriptive error message. The message is logged at info level because it is not fatal, and does not necessarily imply that anything is broken. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs-amp-lib.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c index 01ef4db5407d..287ac01a3873 100644 --- a/sound/soc/codecs/cs-amp-lib.c +++ b/sound/soc/codecs/cs-amp-lib.c @@ -56,6 +56,11 @@ static int _cs_amp_write_cal_coeffs(struct cs_dsp *dsp, dev_dbg(dsp->dev, "Calibration: Ambient=%#x, Status=%#x, CalR=%d\n", data->calAmbient, data->calStatus, data->calR); + if (list_empty(&dsp->ctl_list)) { + dev_info(dsp->dev, "Calibration disabled due to missing firmware controls\n"); + return -ENOENT; + } + ret = cs_amp_write_cal_coeff(dsp, controls, controls->ambient, data->calAmbient); if (ret) return ret; -- 2.39.2

2 1

[RESEND] ASoC: amd: simplify soundwire dependencies for legacy stack
by Vijendar Mukunda 25 Mar '24

25 Mar '24

The CONFIG_SOUNDWIRE_AMD is a user-visible option, it should be never selected by another driver. So replace the extra complexity with a normal Kconfig dependency in SND_SOC_AMD_SOUNDWIRE. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/Kconfig | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index fa74635cee08..3508f5a96b75 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -134,15 +134,14 @@ config SND_SOC_AMD_RPL_ACP6x config SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE tristate - select SOUNDWIRE_AMD if SND_SOC_AMD_SOUNDWIRE != n select SND_AMD_SOUNDWIRE_ACPI if ACPI config SND_SOC_AMD_SOUNDWIRE tristate "Support for SoundWire based AMD platforms" default SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE depends on SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE - depends on ACPI && SOUNDWIRE - depends on !(SOUNDWIRE=m && SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE=y) + depends on ACPI + depends on SOUNDWIRE_AMD help This adds support for SoundWire for AMD platforms. Say Y if you want to enable SoundWire links with SOF. -- 2.34.1

2 1

[PATCH v2 1/1] ASoC: fsl: imx-es8328: Switch to using gpiod API
by Andy Shevchenko 25 Mar '24

25 Mar '24

This updates the driver to gpiod API, and removes yet another use of of_get_named_gpio(). Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- v2: fixed compilation error sound/soc/fsl/imx-es8328.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index 6f0d031c1d5f..d67b8a149bff 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -3,7 +3,7 @@ // Copyright 2012 Freescale Semiconductor, Inc. // Copyright 2012 Linaro Ltd. -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> #include <linux/of_platform.h> @@ -23,7 +23,7 @@ struct imx_es8328_data { struct snd_soc_card card; char codec_dai_name[DAI_NAME_SIZE]; char platform_name[DAI_NAME_SIZE]; - int jack_gpio; + struct gpio_desc *jack_gpiod; }; static struct snd_soc_jack_gpio headset_jack_gpios[] = { @@ -54,8 +54,8 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) struct imx_es8328_data, card); int ret = 0; - /* Headphone jack detection */ - if (gpio_is_valid(data->jack_gpio)) { + if (data->jack_gpiod) { + /* Headphone jack detection */ ret = snd_soc_card_jack_new_pins(rtd->card, "Headphone", SND_JACK_HEADSET | SND_JACK_BTN_0, &headset_jack, @@ -64,7 +64,7 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - headset_jack_gpios[0].gpio = data->jack_gpio; + headset_jack_gpios[0].desc = data->jack_gpiod; ret = snd_soc_jack_add_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios), headset_jack_gpios); @@ -174,7 +174,11 @@ static int imx_es8328_probe(struct platform_device *pdev) data->dev = dev; - data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0); + data->jack_gpiod = devm_gpiod_get_optional(dev, "jack", GPIOD_IN); + if (IS_ERR(data->jack_gpiod)) { + ret = PTR_ERR(data->jack_gpiod); + goto put_device; + } /* * CPU == Platform -- 2.43.0.rc1.1.gbec44491f096

2 2

[PATCH 1/1] ASoC: fsl: fsl_ssi: Add dev_err_probe if PCM DMA init fails
by Alexander Stein 25 Mar '24

25 Mar '24

This happens especially if this driver is built-in, but SDMA driver is configured as module. Signed-off-by: Alexander Stein <alexander.stein(a)ew.tq-group.com> --- sound/soc/fsl/fsl_ssi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ab6ec1974807..4ca3a16f7ac0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1401,8 +1401,10 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, goto error_pcm; } else { ret = imx_pcm_dma_init(pdev); - if (ret) + if (ret) { + dev_err_probe(dev, ret, "Failed to init PCM DMA\n"); goto error_pcm; + } } return 0; -- 2.34.1

3 2

[PATCH] ALSA: hda: cs35l56: Set the init_done flag before component_add()
by Richard Fitzgerald 25 Mar '24

25 Mar '24

From: Simon Trimmer <simont(a)opensource.cirrus.com> Initialization is completed before adding the component as that can start the process of the device binding and trigger actions that check init_done. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") --- sound/pci/hda/cs35l56_hda.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index f3c5715f5e02..1a3f84599cb5 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -1045,14 +1045,14 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) pm_runtime_mark_last_busy(cs35l56->base.dev); pm_runtime_enable(cs35l56->base.dev); + cs35l56->base.init_done = true; + ret = component_add(cs35l56->base.dev, &cs35l56_hda_comp_ops); if (ret) { dev_err(cs35l56->base.dev, "Register component failed: %d\n", ret); goto pm_err; } - cs35l56->base.init_done = true; - return 0; pm_err: -- 2.39.2

2 1

[PATCH] ALSA: hda: cs35l56: Raise device name message log level
by Richard Fitzgerald 25 Mar '24

25 Mar '24

From: Simon Trimmer <simont(a)opensource.cirrus.com> The system and amplifier names influence which firmware and tuning files are downloaded to the device; log these values to aid end-user system support. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/pci/hda/cs35l56_hda.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 41974b3897a7..f3c5715f5e02 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -1024,8 +1024,8 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) goto err; } - dev_dbg(cs35l56->base.dev, "DSP system name: '%s', amp name: '%s'\n", - cs35l56->system_name, cs35l56->amp_name); + dev_info(cs35l56->base.dev, "DSP system name: '%s', amp name: '%s'\n", + cs35l56->system_name, cs35l56->amp_name); regmap_multi_reg_write(cs35l56->base.regmap, cs35l56_hda_dai_config, ARRAY_SIZE(cs35l56_hda_dai_config)); -- 2.39.2

2 1

aplay: fix S24_LE wav header
by GitHub pull_request - edited 25 Mar '24

25 Mar '24

alsa-project/alsa-utils pull request #260 was edited from psi-sue: S24_LE is 32 bits in width storing 24 bits of data and 8 bits of padding So wav header needs to be 32 bits not 24 Request URL : https://github.com/alsa-project/alsa-utils/pull/260 Patch URL : https://github.com/alsa-project/alsa-utils/pull/260.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplay: fix S24_LE wav header
by GitHub pull_request - opened 25 Mar '24

25 Mar '24

alsa-project/alsa-utils pull request #260 was opened from psi-sue: S24_LE is 32 bits in width storing 24 bits of data and 8 bytes of padding So wav header needs to be 32 bits not 24 Request URL : https://github.com/alsa-project/alsa-utils/pull/260 Patch URL : https://github.com/alsa-project/alsa-utils/pull/260.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v4 0/2] ASoC: nau8325: Modify driver code and dtschema.
by Seven Lee 25 Mar '24

25 Mar '24

Modify properties description and use standard units in dtschema. Change: V1 -> V2: - Revise the driver description part for the C++ comment. - In the nau8325_clkdet_put function, modify the max variable to hard code. - Removed "Clock Detection" switch control. - modify the "ALC Enable" switch name. - Revise the dtschema for "nuvoton,dac-vref". V2 -> V3: - Properties use standard unit suffixes. - Modify the enum definition. - Driver code should be used dev_dbg(). V3 -> V4: - Properties use standard unit suffixes. - Modify the enum definition. Seven Lee (2): ASoC: dt-bindings: Added schema for "nuvoton,nau8325" ASoC: nau8325: new driver .../bindings/sound/nuvoton,nau8325.yaml | 90 ++ sound/soc/codecs/nau8325.c | 856 ++++++++++++++++++ sound/soc/codecs/nau8325.h | 391 ++++++++ 3 files changed, 1337 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml create mode 100644 sound/soc/codecs/nau8325.c create mode 100644 sound/soc/codecs/nau8325.h -- 2.25.1

4 6

[PATCH v3 00/32] spi: get rid of some legacy macros
by Uwe Kleine-König 25 Mar '24

25 Mar '24

Changes since v2 (https://lore.kernel.org/linux-spi/cover.1705944943.git.u.kleine-koenig@peng…): - Drop patch "mtd: rawnand: fsl_elbc: Let .probe retry if local bus is missing" which doesn't belong into this series. - Fix a build failure noticed by the kernel build bot in drivers/spi/spi-au1550.c. (I failed to catch this because this driver is mips only, but not enabled in a mips allmodconfig. That's a bit unfortunate, but not easily fixable.) - Add the Reviewed-by: and Acked-by: tags I received for v2. Mark already announced for v2 that he is willing to apply the whole series to his spi tree. Assuming no other show stoper are found in this v3, I assume that's the plan still for this series now. Thanks Uwe Uwe Kleine-König (32): fpga: ice40-spi: Follow renaming of SPI "master" to "controller" ieee802154: ca8210: Follow renaming of SPI "master" to "controller" iio: adc: ad_sigma_delta: Follow renaming of SPI "master" to "controller" Input: pxspad - follow renaming of SPI "master" to "controller" Input: synaptics-rmi4 - follow renaming of SPI "master" to "controller" media: mgb4: Follow renaming of SPI "master" to "controller" media: netup_unidvb: Follow renaming of SPI "master" to "controller" media: usb/msi2500: Follow renaming of SPI "master" to "controller" media: v4l2-subdev: Follow renaming of SPI "master" to "controller" misc: gehc-achc: Follow renaming of SPI "master" to "controller" mmc: mmc_spi: Follow renaming of SPI "master" to "controller" mtd: dataflash: Follow renaming of SPI "master" to "controller" net: ks8851: Follow renaming of SPI "master" to "controller" net: vertexcom: mse102x: Follow renaming of SPI "master" to "controller" platform/chrome: cros_ec_spi: Follow renaming of SPI "master" to "controller" spi: bitbang: Follow renaming of SPI "master" to "controller" spi: cadence-quadspi: Don't emit error message on allocation error spi: cadence-quadspi: Follow renaming of SPI "master" to "controller" spi: cavium: Follow renaming of SPI "master" to "controller" spi: geni-qcom: Follow renaming of SPI "master" to "controller" spi: loopback-test: Follow renaming of SPI "master" to "controller" spi: slave-mt27xx: Follow renaming of SPI "master" to "controller" spi: spidev: Follow renaming of SPI "master" to "controller" staging: fbtft: Follow renaming of SPI "master" to "controller" staging: greybus: spi: Follow renaming of SPI "master" to "controller" tpm_tis_spi: Follow renaming of SPI "master" to "controller" usb: gadget: max3420_udc: Follow renaming of SPI "master" to "controller" video: fbdev: mmp: Follow renaming of SPI "master" to "controller" wifi: libertas: Follow renaming of SPI "master" to "controller" spi: fsl-lib: Follow renaming of SPI "master" to "controller" spi: Drop compat layer from renaming "master" to "controller" Documentation: spi: Update documentation for renaming "master" to "controller" .../driver-api/driver-model/devres.rst | 2 +- Documentation/spi/spi-summary.rst | 74 +++++++++---------- drivers/char/tpm/tpm_tis_spi_main.c | 4 +- drivers/fpga/ice40-spi.c | 4 +- drivers/iio/adc/ad_sigma_delta.c | 14 ++-- drivers/input/joystick/psxpad-spi.c | 4 +- drivers/input/rmi4/rmi_spi.c | 2 +- drivers/media/pci/mgb4/mgb4_core.c | 14 ++-- .../media/pci/netup_unidvb/netup_unidvb_spi.c | 48 ++++++------ drivers/media/usb/msi2500/msi2500.c | 38 +++++----- drivers/media/v4l2-core/v4l2-spi.c | 4 +- drivers/misc/gehc-achc.c | 8 +- drivers/mmc/host/mmc_spi.c | 6 +- drivers/mtd/devices/mtd_dataflash.c | 2 +- drivers/net/ethernet/micrel/ks8851_spi.c | 4 +- drivers/net/ethernet/vertexcom/mse102x.c | 2 +- drivers/net/ieee802154/ca8210.c | 2 +- .../net/wireless/marvell/libertas/if_spi.c | 2 +- drivers/platform/chrome/cros_ec_spi.c | 8 +- drivers/spi/spi-ath79.c | 4 +- drivers/spi/spi-au1550.c | 2 +- drivers/spi/spi-bitbang.c | 64 ++++++++-------- drivers/spi/spi-butterfly.c | 6 +- drivers/spi/spi-cadence-quadspi.c | 7 +- drivers/spi/spi-cavium.c | 6 +- drivers/spi/spi-cavium.h | 2 +- drivers/spi/spi-davinci.c | 6 +- drivers/spi/spi-fsl-lib.c | 14 ++-- drivers/spi/spi-geni-qcom.c | 2 +- drivers/spi/spi-gpio.c | 2 +- drivers/spi/spi-lm70llp.c | 6 +- drivers/spi/spi-loopback-test.c | 4 +- drivers/spi/spi-oc-tiny.c | 6 +- drivers/spi/spi-omap-uwire.c | 4 +- drivers/spi/spi-sh-sci.c | 10 +-- drivers/spi/spi-slave-mt27xx.c | 2 +- drivers/spi/spi-xilinx.c | 4 +- drivers/spi/spi-xtensa-xtfpga.c | 2 +- drivers/spi/spi.c | 2 +- drivers/spi/spidev.c | 2 +- drivers/staging/fbtft/fbtft-core.c | 4 +- drivers/staging/greybus/spilib.c | 66 ++++++++--------- drivers/usb/gadget/udc/max3420_udc.c | 2 +- drivers/video/fbdev/mmp/hw/mmp_spi.c | 26 +++---- include/linux/spi/spi.h | 20 +---- include/linux/spi/spi_bitbang.h | 2 +- include/media/v4l2-common.h | 6 +- 47 files changed, 253 insertions(+), 272 deletions(-) base-commit: b9b98f594b6f4c0b0fb2da4493453aef183bca4b -- 2.43.0

3 4

No sound "dummy output" - snd_hda_intel on Pop OS 22.04 LTS on Acer chromebook plus 514 (Google Markath rev 3)
by GitHub issues - edited 23 Mar '24

23 Mar '24

alsa-project/alsa-lib issue #391 was edited from judicandus: Installed a fresh Pop OS installation on an Acer Chromebook Plus 514, but audio is not working with just one interface available "Dummy Output". Alsa-info attached: [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/14733718/alsa… **1) What I already tried to do** All the instructions in: https://support.system76.com/articles/audio/ , specifically: i. reinstalling pipewire and deleting configurations: systemctl --user restart wireplumber pipewire pipewire-pulse rm -r ~/.config/pulse/* rm -r ~/.local/state/wireplumber/* ii. reinstalling apps sudo apt install --reinstall alsa-base alsa-utils linux-sound-base libasound2 sudo apt reinstall libpipewire-0.3-0 libpipewire-0.3-common libpipewire-0.3-modules pipewire pipewire-audio-client-libraries pipewire-bin pipewire-pulse sudo alsa force-reload iii. snh_hda_intel specific bug fix echo "options snd-hda-intel dmic_detect=0" | sudo tee -a /etc/modprobe.d/alsa-base.conf echo "blacklist snd_soc_skl" | sudo tee -a /etc/modprobe.d/blacklist.conf (reboot) iv. installing SOF driver according to https://github.com/thesofproject/sof-bin from the tarball Nothing worked. **2) Output** ``` aplay -l **** Liste des périphériques matériels PLAYBACK **** carte 0 : Generic [HD-Audio Generic], périphérique 3 : HDMI 0 [HDMI 0] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 7 : HDMI 1 [HDMI 1] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 8 : HDMI 2 [HDMI 2] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 9 : HDMI 3 [HDMI 3] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 ``` ``` arecord -l **** Liste des périphériques matériels CAPTURE **** ``` ``` lsmod | grep snd_hda_intel snd_hda_intel 61440 1 snd_intel_dspcfg 36864 2 snd_hda_intel,snd_sof snd_hda_codec 221184 2 snd_hda_codec_hdmi,snd_hda_intel snd_hda_core 151552 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_pcm 196608 14 snd_sof_amd_acp,snd_hda_codec_hdmi,snd_pci_acp6x,snd_hda_intel,snd_hda_codec,snd_sof,snd_compress,snd_soc_core,snd_sof_utils,snd_hda_core,snd_soc_rt1019,snd_pci_ps,snd_soc_rt5682s,snd_pcm_dmaengine snd 147456 16 snd_seq,snd_seq_device,snd_hda_codec_hdmi,snd_hwdep,snd_hda_intel,snd_hda_codec,snd_sof,snd_timer,snd_compress,snd_soc_core,snd_pcm,snd_rawmidi ``` ``` lspci -nnk | grep -A2 Audio 04:00.1 Audio device [0403]: Advanced Micro Devices, Inc. [AMD/ATI] Device [1002:1640] Subsystem: Advanced Micro Devices, Inc. [AMD/ATI] Device [1002:1640] Kernel driver in use: snd_hda_intel -- 04:00.5 Multimedia controller [0480]: Advanced Micro Devices, Inc. [AMD] Raven/Raven2/FireFlight/Renoir Audio Processor [1022:15e2] (rev 6f) DeviceName: Multimedia controller Subsystem: Advanced Micro Devices, Inc. [AMD] Raven/Raven2/FireFlight/Renoir Audio Processor [1022:15e2] Kernel driver in use: snd_sof_amd_rembrandt Kernel modules: snd_pci_acp3x, snd_rn_pci_acp3x, snd_pci_acp5x, snd_pci_acp6x, snd_acp_pci, snd_rpl_pci_acp6x, snd_pci_ps, snd_sof_amd_renoir, snd_sof_amd_rembrandt, snd_sof_amd_vangogh ``` ``` cat /proc/asound/cards 0 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xd06c0000 irq 77 ``` ``` sudo dmesg | grep -i -e audio -e error -e fail [ 0.282615] ACPI: _OSC evaluation for CPUs failed, trying _PDC [ 0.629902] ACPI Error: Could not enable SleepButton event (20230628/evxfevnt-182) [ 6.100083] RAS: Correctable Errors collector initialized. [ 6.327488] hub 6-0:1.0: config failed, hub doesn't have any ports! (err -19) [ 8.787783] system76_acpi: module verification failed: signature and/or required key missing - tainting kernel [ 10.353860] snd_hda_intel 0000:04:00.1: bound 0000:04:00.0 (ops amdgpu_dm_audio_component_bind_ops [amdgpu]) [ 10.454926] input: HD-Audio Generic HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input10 [ 10.461777] input: HD-Audio Generic HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input11 [ 10.491432] input: HD-Audio Generic HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input12 [ 10.518137] input: HD-Audio Generic HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input13 [ 10.659161] snd_sof_amd_rembrandt 0000:04:00.5: Direct firmware load for amd/sof/community/sof-rmb.ri failed with error -2 [ 10.659196] snd_sof_amd_rembrandt 0000:04:00.5: error: sof firmware file is missing, you might need to [ 10.659208] snd_sof_amd_rembrandt 0000:04:00.5: error: failed to load DSP firmware -2 [ 10.659441] snd_sof_amd_rembrandt 0000:04:00.5: error: sof_probe_work failed err: -2 ``` ``` systemctl --user status pipewire pipewire.service - PipeWire Multimedia Service Loaded: loaded (/usr/lib/systemd/user/pipewire.service; enabled; vendor preset: enabled) Active: active (running) since Sat 2024-03-23 22:15:36 CET; 58min ago TriggeredBy: ● pipewire.socket Main PID: 1320 (pipewire) Tasks: 3 (limit: 9239) Memory: 6.6M CPU: 335ms CGroup: /user.slice/user-1000.slice/user(a)1000.service/session.slice/pipewire.service └─1320 /usr/bin/pipewire mars 23 22:15:36 pop-os systemd[1310]: Started PipeWire Multimedia Service. mars 23 22:15:36 pop-os pipewire[1320]: mod.jackdbus-detect: Failed to receive jackdbus reply: org.freedesktop.DBus.Error.ServiceUnknown: The name is not activatable ``` I am not sure if this is a bug but since I tried all solutions I could find on the internet, I thought I might try my luck here. Thank you beforehand for any assistance or insight. Issue URL : https://github.com/alsa-project/alsa-lib/issues/391 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

No sound "dummy output" - snd_hda_intel on Pop OS 22.04 LTS on Acer chromebook plus 514 (Google Markath rev 3)
by GitHub issues - opened 23 Mar '24

23 Mar '24

alsa-project/alsa-lib issue #391 was opened from judicandus: Installed a fresh Pop OS installation on an Acer Chromebook Plus 514, but audio is not working with just one interface available "Dummy Output". Alsa-info attached: [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/14733718/alsa… **1) What I already tried to do** All the instructions in: https://support.system76.com/articles/audio/ , specifically: i. reinstalling pipewire and deleting configurations: systemctl --user restart wireplumber pipewire pipewire-pulse rm -r ~/.config/pulse/* rm -r ~/.local/state/wireplumber/* ii. reinstalling apps sudo apt install --reinstall alsa-base alsa-utils linux-sound-base libasound2 sudo apt reinstall libpipewire-0.3-0 libpipewire-0.3-common libpipewire-0.3-modules pipewire pipewire-audio-client-libraries pipewire-bin pipewire-pulse sudo alsa force-reload iii. snh_hda_intel specific bug fix echo "options snd-hda-intel dmic_detect=0" | sudo tee -a /etc/modprobe.d/alsa-base.conf echo "blacklist snd_soc_skl" | sudo tee -a /etc/modprobe.d/blacklist.conf (reboot) iv. installing SOF driver according to https://github.com/thesofproject/sof-bin from the tarball Nothing worked. **2) Output** aplay -l **** Liste des périphériques matériels PLAYBACK **** carte 0 : Generic [HD-Audio Generic], périphérique 3 : HDMI 0 [HDMI 0] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 7 : HDMI 1 [HDMI 1] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 8 : HDMI 2 [HDMI 2] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 9 : HDMI 3 [HDMI 3] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 arecord -l **** Liste des périphériques matériels CAPTURE **** lsmod | grep snd_hda_intel snd_hda_intel 61440 1 snd_intel_dspcfg 36864 2 snd_hda_intel,snd_sof snd_hda_codec 221184 2 snd_hda_codec_hdmi,snd_hda_intel snd_hda_core 151552 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_pcm 196608 14 snd_sof_amd_acp,snd_hda_codec_hdmi,snd_pci_acp6x,snd_hda_intel,snd_hda_codec,snd_sof,snd_compress,snd_soc_core,snd_sof_utils,snd_hda_core,snd_soc_rt1019,snd_pci_ps,snd_soc_rt5682s,snd_pcm_dmaengine snd 147456 16 snd_seq,snd_seq_device,snd_hda_codec_hdmi,snd_hwdep,snd_hda_intel,snd_hda_codec,snd_sof,snd_timer,snd_compress,snd_soc_core,snd_pcm,snd_rawmidi lspci -nnk | grep -A2 Audio 04:00.1 Audio device [0403]: Advanced Micro Devices, Inc. [AMD/ATI] Device [1002:1640] Subsystem: Advanced Micro Devices, Inc. [AMD/ATI] Device [1002:1640] Kernel driver in use: snd_hda_intel -- 04:00.5 Multimedia controller [0480]: Advanced Micro Devices, Inc. [AMD] Raven/Raven2/FireFlight/Renoir Audio Processor [1022:15e2] (rev 6f) DeviceName: Multimedia controller Subsystem: Advanced Micro Devices, Inc. [AMD] Raven/Raven2/FireFlight/Renoir Audio Processor [1022:15e2] Kernel driver in use: snd_sof_amd_rembrandt Kernel modules: snd_pci_acp3x, snd_rn_pci_acp3x, snd_pci_acp5x, snd_pci_acp6x, snd_acp_pci, snd_rpl_pci_acp6x, snd_pci_ps, snd_sof_amd_renoir, snd_sof_amd_rembrandt, snd_sof_amd_vangogh cat /proc/asound/cards 0 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xd06c0000 irq 77 sudo dmesg | grep -i -e audio -e error -e fail [ 0.282615] ACPI: _OSC evaluation for CPUs failed, trying _PDC [ 0.629902] ACPI Error: Could not enable SleepButton event (20230628/evxfevnt-182) [ 6.100083] RAS: Correctable Errors collector initialized. [ 6.327488] hub 6-0:1.0: config failed, hub doesn't have any ports! (err -19) [ 8.787783] system76_acpi: module verification failed: signature and/or required key missing - tainting kernel [ 10.353860] snd_hda_intel 0000:04:00.1: bound 0000:04:00.0 (ops amdgpu_dm_audio_component_bind_ops [amdgpu]) [ 10.454926] input: HD-Audio Generic HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input10 [ 10.461777] input: HD-Audio Generic HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input11 [ 10.491432] input: HD-Audio Generic HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input12 [ 10.518137] input: HD-Audio Generic HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input13 [ 10.659161] snd_sof_amd_rembrandt 0000:04:00.5: Direct firmware load for amd/sof/community/sof-rmb.ri failed with error -2 [ 10.659196] snd_sof_amd_rembrandt 0000:04:00.5: error: sof firmware file is missing, you might need to [ 10.659208] snd_sof_amd_rembrandt 0000:04:00.5: error: failed to load DSP firmware -2 [ 10.659441] snd_sof_amd_rembrandt 0000:04:00.5: error: sof_probe_work failed err: -2 systemctl --user status pipewire pipewire.service - PipeWire Multimedia Service Loaded: loaded (/usr/lib/systemd/user/pipewire.service; enabled; vendor preset: enabled) Active: active (running) since Sat 2024-03-23 22:15:36 CET; 58min ago TriggeredBy: ● pipewire.socket Main PID: 1320 (pipewire) Tasks: 3 (limit: 9239) Memory: 6.6M CPU: 335ms CGroup: /user.slice/user-1000.slice/user(a)1000.service/session.slice/pipewire.service └─1320 /usr/bin/pipewire mars 23 22:15:36 pop-os systemd[1310]: Started PipeWire Multimedia Service. mars 23 22:15:36 pop-os pipewire[1320]: mod.jackdbus-detect: Failed to receive jackdbus reply: org.freedesktop.DBus.Error.ServiceUnknown: The name is not activatable I am not sure if this is a bug but since I tried all solutions I could find on the internet, I thought I might try my luck here. Thank you beforehand for any assistance or insight. Issue URL : https://github.com/alsa-project/alsa-lib/issues/391 Repository URL: https://github.com/alsa-project/alsa-lib

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