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

[PATCH AUTOSEL 6.8 48/98] ASoC: Intel: avs: Populate board selection with new I2S entries
by Sasha Levin 29 Mar '24

29 Mar '24

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit 5b417fe0cded0b5917683398e6519aae8045cd40 ] Update board selection with tables specifying supported I2S configurations. DMIC/HDAudio board selection require no update as dmic/hdaudio machine boards are generic and not tied to any specific codec. Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://msgid.link/r/20240220115035.770402-11-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/board_selection.c | 85 +++++++++++++++++++++++++++ 1 file changed, 85 insertions(+) diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index 8e91eece992db..8360ce557401c 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -236,6 +236,82 @@ static struct snd_soc_acpi_mach avs_gml_i2s_machines[] = { {}, }; +static struct snd_soc_acpi_mach avs_cnl_i2s_machines[] = { + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + { + .id = "10EC5682", + .drv_name = "avs_rt5682", + .mach_params = { + .i2s_link_mask = AVS_SSP(1), + }, + .tplg_filename = "rt5682-tplg.bin", + }, + {}, +}; + +static struct snd_soc_acpi_mach avs_icl_i2s_machines[] = { + { + .id = "INT343A", + .drv_name = "avs_rt298", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt298-tplg.bin", + }, + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + {}, +}; + +static struct snd_soc_acpi_mach avs_tgl_i2s_machines[] = { + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + { + .id = "10EC0298", + .drv_name = "avs_rt298", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt298-tplg.bin", + }, + { + .id = "10EC1308", + .drv_name = "avs_rt1308", + .mach_params = { + .i2s_link_mask = AVS_SSP(1), + }, + .tplg_filename = "rt1308-tplg.bin", + }, + { + .id = "ESSX8336", + .drv_name = "avs_es8336", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "es8336-tplg.bin", + }, + {}, +}; + static struct snd_soc_acpi_mach avs_test_i2s_machines[] = { { .drv_name = "avs_i2s_test", @@ -296,6 +372,15 @@ static const struct avs_acpi_boards i2s_boards[] = { AVS_MACH_ENTRY(HDA_KBL_LP, avs_kbl_i2s_machines), AVS_MACH_ENTRY(HDA_APL, avs_apl_i2s_machines), AVS_MACH_ENTRY(HDA_GML, avs_gml_i2s_machines), + AVS_MACH_ENTRY(HDA_CNL_LP, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_CNL_H, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_CML_LP, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_ICL_LP, avs_icl_i2s_machines), + AVS_MACH_ENTRY(HDA_TGL_LP, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_EHL_0, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_ADL_P, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_RPL_P_0, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_RPL_M, avs_tgl_i2s_machines), {}, }; -- 2.43.0

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[PATCH AUTOSEL 6.8 45/98] ALSA: firewire-lib: handle quirk to calculate payload quadlets as data block counter
by Sasha Levin 29 Mar '24

29 Mar '24

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 4a486439d2ca85752c46711f373b6ddc107bb35d ] Miglia Harmony Audio (OXFW970) has a quirk to put the number of accumulated quadlets in CIP payload into the dbc field of CIP header. This commit handles the quirk in the packet processing layer. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20240218074128.95210-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/amdtp-stream.c | 12 ++++++++---- sound/firewire/amdtp-stream.h | 4 ++++ 2 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 7be17bca257f0..c9f153f85ae6b 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -773,10 +773,14 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf, } else { unsigned int dbc_interval; - if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) - dbc_interval = s->ctx_data.tx.dbc_interval; - else - dbc_interval = *data_blocks; + if (!(s->flags & CIP_DBC_IS_PAYLOAD_QUADLETS)) { + if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) + dbc_interval = s->ctx_data.tx.dbc_interval; + else + dbc_interval = *data_blocks; + } else { + dbc_interval = payload_length / sizeof(__be32); + } lost = dbc != ((*data_block_counter + dbc_interval) & 0xff); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index b7ff44751ab93..a1ed2e80f91a7 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -37,6 +37,9 @@ * the value of current SYT_INTERVAL; e.g. initial value is not zero. * @CIP_UNAWARE_SYT: For outgoing packet, the value in SYT field of CIP is 0xffff. * For incoming packet, the value in SYT field of CIP is not handled. + * @CIP_DBC_IS_PAYLOAD_QUADLETS: Available for incoming packet, and only effective with + * CIP_DBC_IS_END_EVENT flag. The value of dbc field is the number of accumulated quadlets + * in CIP payload, instead of the number of accumulated data blocks. */ enum cip_flags { CIP_NONBLOCKING = 0x00, @@ -51,6 +54,7 @@ enum cip_flags { CIP_NO_HEADER = 0x100, CIP_UNALIGHED_DBC = 0x200, CIP_UNAWARE_SYT = 0x400, + CIP_DBC_IS_PAYLOAD_QUADLETS = 0x800, }; /** -- 2.43.0

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[PATCH AUTOSEL 6.8 27/98] ASoC: Intel: common: DMI remap for rebranded Intel NUC M15 (LAPRC710) laptops
by Sasha Levin 29 Mar '24

29 Mar '24

From: mosomate <mosomate(a)gmail.com> [ Upstream commit c13e03126a5be90781084437689724254c8226e1 ] Added DMI quirk to handle the rebranded variants of Intel NUC M15 (LAPRC710) laptops. The DMI matching is based on motherboard attributes. Link: https://github.com/thesofproject/linux/issues/4218 Signed-off-by: Máté Mosonyi <mosomate(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240208165545.93811-20-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/dmi-quirks.c | 8 ++++++++ sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 2 files changed, 19 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 9ebdd0cd0b1cf..91ab97a456fa9 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -130,6 +130,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_rooks_county, }, + { + /* quirk used for NUC15 LAPRC710 skew */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)intel_rooks_county, + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 300391fbc2fc2..cf14bd3053fc9 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -236,6 +236,17 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_SDW_PCH_DMIC | RT711_JD2_100K), }, + { + /* NUC15 LAPRC710 skews */ + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | + RT711_JD2_100K), + }, /* TigerLake-SDCA devices */ { .callback = sof_sdw_quirk_cb, -- 2.43.0

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[PATCH AUTOSEL 6.8 26/98] ASoC: Intel: sof_rt5682: dmi quirk cleanup for mtl boards
by Sasha Levin 29 Mar '24

29 Mar '24

From: Brent Lu <brent.lu(a)intel.com> [ Upstream commit 7a2a8730d51f95b263a1e8b888598dc6395220dc ] Some dmi quirks are duplicated since codec and amplifier type are removed from board quirk. Remove redundant quirks. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240208165545.93811-7-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 40 ----------------------------- 1 file changed, 40 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index cd50f26d1edbe..02705034e5713 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -154,46 +154,6 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, - { - .callback = sof_rt5682_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), - DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98360_ALC5682I_I2S"), - }, - .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) | - SOF_BT_OFFLOAD_SSP(1) | - SOF_SSP_BT_OFFLOAD_PRESENT - ), - }, - { - .callback = sof_rt5682_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), - DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98360_ALC5682I_DISCRETE_I2S_BT"), - }, - .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) | - SOF_BT_OFFLOAD_SSP(1) | - SOF_SSP_BT_OFFLOAD_PRESENT - ), - }, - { - .callback = sof_rt5682_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), - DMI_MATCH(DMI_OEM_STRING, "AUDIO-ALC1019_ALC5682I_I2S"), - }, - .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) - ), - }, { .callback = sof_rt5682_quirk_cb, .matches = { -- 2.43.0

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[PATCH 0/6] ASoC: rt-sdw: fix locking and improve error logs
by Pierre-Louis Bossart 29 Mar '24

29 Mar '24

Fix a set of problematic locking sequences and update error messages, tested on SOF/SoundWire platforms. Pierre-Louis Bossart (6): ASoC: rt5682-sdw: fix locking sequence ASoC: rt711-sdca: fix locking sequence ASoC: rt711-sdw: fix locking sequence ASoC: rt712-sdca-sdw: fix locking sequence ASoC: rt722-sdca-sdw: fix locking sequence ASoC: rt-sdw*: add __func__ to all error logs sound/soc/codecs/rt1316-sdw.c | 8 +++--- sound/soc/codecs/rt1318-sdw.c | 8 +++--- sound/soc/codecs/rt5682-sdw.c | 16 +++++------ sound/soc/codecs/rt700.c | 16 +++++------ sound/soc/codecs/rt711-sdca-sdw.c | 6 ++-- sound/soc/codecs/rt711-sdca.c | 18 ++++++------ sound/soc/codecs/rt711-sdw.c | 8 +++--- sound/soc/codecs/rt711.c | 16 +++++------ sound/soc/codecs/rt712-sdca-dmic.c | 24 +++++++++------- sound/soc/codecs/rt712-sdca-sdw.c | 7 +++-- sound/soc/codecs/rt712-sdca.c | 20 ++++++------- sound/soc/codecs/rt715-sdca-sdw.c | 2 +- sound/soc/codecs/rt715-sdca.c | 46 +++++++++++++++--------------- sound/soc/codecs/rt715-sdw.c | 4 +-- sound/soc/codecs/rt715.c | 24 ++++++++-------- sound/soc/codecs/rt722-sdca-sdw.c | 4 +-- sound/soc/codecs/rt722-sdca.c | 21 +++++++------- 17 files changed, 126 insertions(+), 122 deletions(-) -- 2.40.1

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[PATCH v5 0/2] ASoC: nau8325: Modify driver code and dtschema.
by Seven Lee 29 Mar '24

29 Mar '24

Revise properties description and use standard units in dtschema. The unit conversion driver based on the attribute must also be changed accordingly. 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. V4->V5: - Properties use standard unit suffixes for vref-impedance and dac-vref. - Revise the enum definition. - Modify the code related to properties usage in the NAU8325 codec driver. - Re-arrange header files in alphabetical order. Seven Lee (2): ASoC: dt-bindings: Added schema for "nuvoton,nau8325" ASoC: nau8325: new driver .../bindings/sound/nuvoton,nau8325.yaml | 80 ++ sound/soc/codecs/nau8325.c | 893 ++++++++++++++++++ sound/soc/codecs/nau8325.h | 391 ++++++++ 3 files changed, 1364 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

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[PATCH 00/18] ASoC: drop driver owner assignment
by Krzysztof Kozlowski 29 Mar '24

29 Mar '24

Core for several drivers already sets the driver.owner, so driver does not need to. Simplify the drivers. Patches are independent, no dependencies. Best regards, Krzysztof --- Krzysztof Kozlowski (18): ASoC: loongson: i2s/pci: drop driver owner assignment ASoC: tlv320aic32x4-spi: drop driver owner assignment ASoC: tlv320aic3x-spi: drop driver owner assignment ASoC: max98373-sdw: drop driver owner assignment ASoC: rt1017-sdca-sdw: drop driver owner assignment ASoC: rt1308-sdw: drop driver owner assignment ASoC: rt1316-sdw: drop driver owner assignment ASoC: rt1318-sdw: drop driver owner assignment ASoC: rt5682-sdw: drop driver owner assignment ASoC: rt700-sdw: drop driver owner assignment ASoC: rt711-sdca-sdw: drop driver owner assignment ASoC: rt711-sdw: drop driver owner assignment ASoC: rt712-sdca-dmic: drop driver owner assignment ASoC: rt712-sdca-sdw: drop driver owner assignment ASoC: rt715-sdca-sdw: drop driver owner assignment ASoC: rt715-sdw: drop driver owner assignment ASoC: rt722-sdca-sdw: drop driver owner assignment ASoC: sdw-mockup: drop driver owner assignment sound/soc/codecs/max98373-sdw.c | 1 - sound/soc/codecs/rt1017-sdca-sdw.c | 1 - sound/soc/codecs/rt1308-sdw.c | 1 - sound/soc/codecs/rt1316-sdw.c | 1 - sound/soc/codecs/rt1318-sdw.c | 1 - sound/soc/codecs/rt5682-sdw.c | 1 - sound/soc/codecs/rt700-sdw.c | 1 - sound/soc/codecs/rt711-sdca-sdw.c | 1 - sound/soc/codecs/rt711-sdw.c | 1 - sound/soc/codecs/rt712-sdca-dmic.c | 1 - sound/soc/codecs/rt712-sdca-sdw.c | 1 - sound/soc/codecs/rt715-sdca-sdw.c | 1 - sound/soc/codecs/rt715-sdw.c | 1 - sound/soc/codecs/rt722-sdca-sdw.c | 1 - sound/soc/codecs/sdw-mockup.c | 1 - sound/soc/codecs/tlv320aic32x4-spi.c | 1 - sound/soc/codecs/tlv320aic3x-spi.c | 1 - sound/soc/loongson/loongson_i2s_pci.c | 1 - 18 files changed, 18 deletions(-) --- base-commit: 1cadb9c1269d4e90ec23f3dc3e5373ee405462d1 change-id: 20240327-module-owner-var-2744d140fcdf Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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[PATCH 00/18] ASoC: Intel: boards: updates for 6.10 - part3
by Pierre-Louis Bossart 29 Mar '24

29 Mar '24

This last part is the continuation of Brent Lu's cleanups. Multiple quirks have been removed to use "default" configurations and ACPI-based detection of codecs and selection of topology files. This cleanup has been done in multiple steps/phases since Fall 2023, thanks Brent for this contribution! Brent Lu (18): ASoC: Intel: ssp-common: relocate source file ASoC: Intel: ssp-common: relocate header file ASoC: Intel: ssp-common: naming convention change ASoC: Intel: ssp-common: module name change ASoC: Intel: ssp-common: delete module ASoC: Intel: ssp-common: get codec tplg suffix function ASoC: SOF: Intel: support tplg suffix detection ASoC: Intel: sof_nau8825: mach cleanup for adl boards ASoC: Intel: sof_nau8825: mach cleanup for rpl boards ASoC: Intel: sof_nau8825: add mtl_nau8825_def for mtl boards ASoC: Intel: sof_rt5682: mach cleanup for tgl boards ASoC: Intel: sof_rt5682: mach cleanup for adl boards ASoC: Intel: sof_rt5682: mach cleanup for rpl boards ASoC: Intel: sof_rt5682: mach cleanup for mtl boards ASoC: Intel: sof_cs42l42: board id cleanup for adl boards ASoC: Intel: sof_cs42l42: mach cleanup for adl boards ASoC: Intel: sof_cs42l42: add rpl_cs42l42_def for rpl boards ASoC: Intel: sof_cs42l42: add mtl_cs42l42_def for mtl boards .../sound/soc-acpi-intel-ssp-common.h | 22 ++- include/sound/soc-acpi.h | 12 ++ sound/soc/intel/boards/Kconfig | 13 +- sound/soc/intel/boards/Makefile | 3 - sound/soc/intel/boards/sof_board_helpers.c | 15 +- sound/soc/intel/boards/sof_board_helpers.h | 6 +- sound/soc/intel/boards/sof_cirrus_common.h | 2 +- sound/soc/intel/boards/sof_cs42l42.c | 17 +- sound/soc/intel/boards/sof_maxim_common.h | 2 +- sound/soc/intel/boards/sof_nau8825.c | 7 + sound/soc/intel/boards/sof_nuvoton_common.h | 2 +- sound/soc/intel/boards/sof_realtek_common.h | 2 +- sound/soc/intel/boards/sof_ssp_common.c | 122 -------------- sound/soc/intel/common/Makefile | 2 + .../intel/common/soc-acpi-intel-adl-match.c | 139 ++++----------- .../intel/common/soc-acpi-intel-mtl-match.c | 52 +++--- .../intel/common/soc-acpi-intel-rpl-match.c | 90 ++++------ .../intel/common/soc-acpi-intel-ssp-common.c | 159 ++++++++++++++++++ .../intel/common/soc-acpi-intel-tgl-match.c | 50 ++---- sound/soc/sof/intel/hda.c | 82 +++++++++ 20 files changed, 423 insertions(+), 376 deletions(-) rename sound/soc/intel/boards/sof_ssp_common.h => include/sound/soc-acpi-intel-ssp-common.h (66%) delete mode 100644 sound/soc/intel/boards/sof_ssp_common.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-ssp-common.c -- 2.40.1

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[PATCH] ASoC: tas2781: mark dvc_tlv with __maybe_unused
by Gergo Koteles 29 Mar '24

29 Mar '24

Since we put dvc_tlv static variable to a header file it's copied to each module that includes the header. But not all of them are actually used it. Fix this W=1 build warning: include/sound/tas2781-tlv.h:18:35: warning: 'dvc_tlv' defined but not used [-Wunused-const-variable=] Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202403290354.v0StnRpc-lkp@intel.com/ Fixes: ae065d0ce9e3 ("ALSA: hda/tas2781: remove digital gain kcontrol") Signed-off-by: Gergo Koteles <soyer(a)irl.hu> --- include/sound/tas2781-tlv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 4038dd421150..1dc59005d241 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -15,7 +15,7 @@ #ifndef __TAS2781_TLV_H__ #define __TAS2781_TLV_H__ -static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); #endif base-commit: 2d0401ee38d43ab0e4cdd02dfc9d402befb2b5c8 -- 2.44.0

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[PATCH] dt-bindings: ti,pcm1681: Convert to dtschema
by Animesh Agarwal 29 Mar '24

29 Mar '24

Convert the Texas Instruments PCM1681 bindings to DT schema. Signed-off-by: Animesh Agarwal <animeshagarwal28(a)gmail.com> --- .../devicetree/bindings/sound/ti,pcm1681.txt | 15 -------- .../devicetree/bindings/sound/ti,pcm1681.yaml | 35 +++++++++++++++++++ 2 files changed, 35 insertions(+), 15 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt deleted file mode 100644 index 4df17185ab80..000000000000 --- a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt +++ /dev/null @@ -1,15 +0,0 @@ -Texas Instruments PCM1681 8-channel PWM Processor - -Required properties: - - - compatible: Should contain "ti,pcm1681". - - reg: The i2c address. Should contain <0x4c>. - -Examples: - - i2c_bus { - pcm1681@4c { - compatible = "ti,pcm1681"; - reg = <0x4c>; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml new file mode 100644 index 000000000000..4093d0ff654d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm1681.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments PCM1681 8-channel PWM Processor + +maintainers: + - Animesh Agarwal <animeshagarwal28(a)gmail.com> + +properties: + compatible: + const: ti,pcm1681 + + reg: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + pcm1681@4c { + compatible = "ti,pcm1681"; + reg = <0x4c>; + }; + }; -- 2.44.0

2 2

[PATCH] ASoC: ops: Fix wraparound for mask in snd_soc_get_volsw
by Stephen Lee 29 Mar '24

29 Mar '24

In snd_soc_info_volsw(), mask is generated by figuring out the index of the most significant bit set in max and converting the index to a bitmask through bit shift 1. Unintended wraparound occurs when max is an integer value with msb bit set. Since the bit shift value 1 is treated as an integer type, the left shift operation will wraparound and set mask to 0 instead of all 1's. In order to fix this, we type cast 1 as `1ULL` to prevent the wraparound. Fixes: 7077148fb50a ("ASoC: core: Split ops out of soc-core.c") Signed-off-by: Stephen Lee <slee08177(a)gmail.com> --- This seems to show up in quite a few places in the alsa subsystem, should they be addressed altogether? --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 2d25748ca706..b27e89ff6a16 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -263,7 +263,7 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; int min = mc->min; int sign_bit = mc->sign_bit; - unsigned int mask = (1 << fls(max)) - 1; + unsigned int mask = (1ULL << fls(max)) - 1; unsigned int invert = mc->invert; int val; int ret; -- 2.43.2

2 2

[PATCH 1/2] ASoC: amd: acp: fix for acp_init function error handling
by Vijendar Mukunda 28 Mar '24

28 Mar '24

If acp_init() fails, acp pci driver probe should return error. Add acp_init() function return value check logic. Fixes: e61b415515d3 ("ASoC: amd: acp: refactor the acp init and de-init sequence") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-pci.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 8c8b1dcac628..c1d4140f0746 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -115,7 +115,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id goto unregister_dmic_dev; } - acp_init(chip); + ret = acp_init(chip); + if (ret) + return ret; + res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL); if (!res) { ret = -ENOMEM; -- 2.34.1

3 4

[PATCH] ASoC: hdac_hda: improve error logs
by Pierre-Louis Bossart 28 Mar '24

28 Mar '24

We have a couple of duplicate logs and missing information, add __func__ consistently and make sure useful error codes are logged. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/codecs/hdac_hda.c | 44 ++++++++++++++++++++----------------- 1 file changed, 24 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 6aa3223985be..29c88de5508b 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -230,7 +230,8 @@ static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, format_val = snd_hdac_stream_format(params_channels(params), bits, params_rate(params)); if (!format_val) { dev_err(dai->dev, - "invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", + "%s: invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", + __func__, params_rate(params), params_channels(params), params_format(params), maxbps); @@ -266,14 +267,12 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct hda_pcm_stream *hda_stream; struct hdac_hda_priv *hda_pvt; - struct hdac_device *hdev; unsigned int format_val; struct hda_pcm *pcm; unsigned int stream; int ret = 0; hda_pvt = snd_soc_component_get_drvdata(component); - hdev = &hda_pvt->codec->core; pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); if (!pcm) return -EINVAL; @@ -286,7 +285,7 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, ret = snd_hda_codec_prepare(hda_pvt->codec, hda_stream, stream, format_val, substream); if (ret < 0) - dev_err(&hdev->dev, "codec prepare failed %d\n", ret); + dev_err(dai->dev, "%s: failed %d\n", __func__, ret); return ret; } @@ -298,6 +297,7 @@ static int hdac_hda_dai_open(struct snd_pcm_substream *substream, struct hdac_hda_priv *hda_pvt; struct hda_pcm_stream *hda_stream; struct hda_pcm *pcm; + int ret; hda_pvt = snd_soc_component_get_drvdata(component); pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); @@ -308,7 +308,11 @@ static int hdac_hda_dai_open(struct snd_pcm_substream *substream, hda_stream = &pcm->stream[substream->stream]; - return hda_stream->ops.open(hda_stream, hda_pvt->codec, substream); + ret = hda_stream->ops.open(hda_stream, hda_pvt->codec, substream); + if (ret < 0) + dev_err(dai->dev, "%s: failed %d\n", __func__, ret); + + return ret; } static void hdac_hda_dai_close(struct snd_pcm_substream *substream, @@ -367,7 +371,7 @@ static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt, pcm_name = "HDMI 3"; break; default: - dev_err(&hcodec->core.dev, "invalid dai id %d\n", dai->id); + dev_err(dai->dev, "%s: invalid dai id %d\n", __func__, dai->id); return NULL; } @@ -381,7 +385,7 @@ static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt, } } - dev_err(&hcodec->core.dev, "didn't find PCM for DAI %s\n", dai->name); + dev_err(dai->dev, "%s: didn't find PCM for DAI %s\n", __func__, dai->name); return NULL; } @@ -411,7 +415,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&hdev->dev, "hdac link not found\n"); + dev_err(&hdev->dev, "%s: hdac link not found\n", __func__); return -EIO; } @@ -429,7 +433,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) ret = snd_hda_codec_device_new(hcodec->bus, component->card->snd_card, hdev->addr, hcodec, true); if (ret < 0) { - dev_err(&hdev->dev, "failed to create hda codec %d\n", ret); + dev_err(&hdev->dev, "%s: failed to create hda codec %d\n", __func__, ret); goto error_no_pm; } @@ -446,7 +450,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (fw) { ret = snd_hda_load_patch(hcodec->bus, fw->size, fw->data); if (ret < 0) { - dev_err(&hdev->dev, "failed to load hda patch %d\n", ret); + dev_err(&hdev->dev, "%s: failed to load hda patch %d\n", __func__, ret); goto error_no_pm; } release_firmware(fw); @@ -470,13 +474,13 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) ret = snd_hda_codec_set_name(hcodec, hcodec->preset->name); if (ret < 0) { - dev_err(&hdev->dev, "name failed %s\n", hcodec->preset->name); + dev_err(&hdev->dev, "%s: name failed %s\n", __func__, hcodec->preset->name); goto error_pm; } ret = snd_hdac_regmap_init(&hcodec->core); if (ret < 0) { - dev_err(&hdev->dev, "regmap init failed\n"); + dev_err(&hdev->dev, "%s: regmap init failed\n", __func__); goto error_pm; } @@ -484,16 +488,16 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (patch) { ret = patch(hcodec); if (ret < 0) { - dev_err(&hdev->dev, "patch failed %d\n", ret); + dev_err(&hdev->dev, "%s: patch failed %d\n", __func__, ret); goto error_regmap; } } else { - dev_dbg(&hdev->dev, "no patch file found\n"); + dev_dbg(&hdev->dev, "%s: no patch file found\n", __func__); } ret = snd_hda_codec_parse_pcms(hcodec); if (ret < 0) { - dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret); + dev_err(&hdev->dev, "%s: unable to map pcms to dai %d\n", __func__, ret); goto error_patch; } @@ -501,8 +505,8 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (!is_hdmi_codec(hcodec)) { ret = snd_hda_codec_build_controls(hcodec); if (ret < 0) { - dev_err(&hdev->dev, "unable to create controls %d\n", - ret); + dev_err(&hdev->dev, "%s: unable to create controls %d\n", + __func__, ret); goto error_patch; } } @@ -548,7 +552,7 @@ static void hdac_hda_codec_remove(struct snd_soc_component *component) hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&hdev->dev, "hdac link not found\n"); + dev_err(&hdev->dev, "%s: hdac link not found\n", __func__); return; } @@ -624,7 +628,7 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) /* hold the ref while we probe */ hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&hdev->dev, "hdac link not found\n"); + dev_err(&hdev->dev, "%s: hdac link not found\n", __func__); return -EIO; } snd_hdac_ext_bus_link_get(hdev->bus, hlink); @@ -640,7 +644,7 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) ARRAY_SIZE(hdac_hda_dais)); if (ret < 0) { - dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret); + dev_err(&hdev->dev, "%s: failed to register HDA codec %d\n", __func__, ret); return ret; } -- 2.40.1

2 2

[tiwai-sound:for-linus 12/17] include/sound/tas2781-tlv.h:18:35: warning: 'dvc_tlv' defined but not used
by kernel test robot 28 Mar '24

28 Mar '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: 2d0401ee38d43ab0e4cdd02dfc9d402befb2b5c8 commit: ae065d0ce9e36ca4efdfb9b96ce3395bd1c19372 [12/17] ALSA: hda/tas2781: remove digital gain kcontrol config: i386-allmodconfig (https://download.01.org/0day-ci/archive/20240329/202403290354.v0StnRpc-lkp@…) compiler: gcc-12 (Ubuntu 12.3.0-9ubuntu2) 12.3.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240329/202403290354.v0StnRpc-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202403290354.v0StnRpc-lkp@intel.com/ All warnings (new ones prefixed by >>): In file included from include/sound/tlv.h:10, from sound/pci/hda/tas2781_hda_i2c.c:22: >> include/sound/tas2781-tlv.h:18:35: warning: 'dvc_tlv' defined but not used [-Wunused-const-variable=] 18 | static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); | ^~~~~~~ include/uapi/sound/tlv.h:53:22: note: in definition of macro 'SNDRV_CTL_TLVD_DECLARE_DB_SCALE' 53 | unsigned int name[] = { \ | ^~~~ include/sound/tas2781-tlv.h:18:14: note: in expansion of macro 'DECLARE_TLV_DB_SCALE' 18 | static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); | ^~~~~~~~~~~~~~~~~~~~ vim +/dvc_tlv +18 include/sound/tas2781-tlv.h 678f38eba1f2fe Shenghao Ding 2023-06-18 17 678f38eba1f2fe Shenghao Ding 2023-06-18 @18 static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); 678f38eba1f2fe Shenghao Ding 2023-06-18 19 static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); 678f38eba1f2fe Shenghao Ding 2023-06-18 20 :::::: The code at line 18 was first introduced by commit :::::: 678f38eba1f2fe33ff700e85390ac98393e609ef ASoC: tas2781: Add Header file for tas2781 driver :::::: TO: Shenghao Ding <13916275206(a)139.com> :::::: CC: Mark Brown <broonie(a)kernel.org> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH] soundwire: qcom: Convert to platform remove callback returning void
by Uwe Kleine-König 28 Mar '24

28 Mar '24

The .remove() callback for a platform driver returns an int which makes many driver authors wrongly assume it's possible to do error handling by returning an error code. However the value returned is ignored (apart from emitting a warning) and this typically results in resource leaks. To improve here there is a quest to make the remove callback return void. In the first step of this quest all drivers are converted to .remove_new(), which already returns void. Eventually after all drivers are converted, .remove_new() will be renamed to .remove(). Trivially convert this driver from always returning zero in the remove callback to the void returning variant. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- drivers/soundwire/qcom.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 3c4d6debab1f..fb70afe64fcc 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1636,14 +1636,12 @@ static int qcom_swrm_probe(struct platform_device *pdev) return ret; } -static int qcom_swrm_remove(struct platform_device *pdev) +static void qcom_swrm_remove(struct platform_device *pdev) { struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(&pdev->dev); sdw_bus_master_delete(&ctrl->bus); clk_disable_unprepare(ctrl->hclk); - - return 0; } static int __maybe_unused swrm_runtime_resume(struct device *dev) @@ -1769,7 +1767,7 @@ MODULE_DEVICE_TABLE(of, qcom_swrm_of_match); static struct platform_driver qcom_swrm_driver = { .probe = &qcom_swrm_probe, - .remove = &qcom_swrm_remove, + .remove_new = qcom_swrm_remove, .driver = { .name = "qcom-soundwire", .of_match_table = qcom_swrm_of_match, base-commit: 11afac187274a6177a7ac82997f8691c0f469e41 -- 2.43.0

2 1

[PATCH 0/7] Soundwire: clean up sysfs group creation
by Greg Kroah-Hartman 28 Mar '24

28 Mar '24

Note, this is a redone version of a very old series I wrote back in 2022: https://lore.kernel.org/r/20220824135951.3604059-1-gregkh@linuxfoundation.o… but everyone has forgotten about it now, and I've reworked it, so I'm considering it a "new" version, and not v2. Here's a series that adds the functionality to the driver core to hide entire attribute groups, in a much saner way than we have attempted in the past (i.e. dynamically figuring it out.) Many thanks to Dan for this patch. I'll also be taking this into my driver-core branch and creating a stable tag for anyone else to pull from to get it into their trees, as I think it will want to be in many for this development cycle. After the driver core change, there's cleanups to the soundwire core for how the attribute groups are created, to remove the "manual" creation of them, and allow the driver core to create them correctly, as needed, when needed, which makes things much smaller for the soundwire code to manage. Comments appreciated! thanks, greg k-h Dan Williams (1): sysfs: Introduce a mechanism to hide static attribute_groups Greg Kroah-Hartman (5): soundwire: sysfs: move sdw_slave_dev_attr_group into the existing list of groups soundwire: sysfs: cleanup the logic for creating the dp0 sysfs attributes soundwire: sysfs: have the driver core handle the creation of the device groups soundwire: sysfs: remove sdw_slave_sysfs_init() soundwire: sysfs: remove unneeded ATTRIBUTE_GROUPS() comments drivers/soundwire/bus_type.c | 5 ++- drivers/soundwire/sysfs_local.h | 4 +- drivers/soundwire/sysfs_slave.c | 64 ++++++++++++++--------------- drivers/soundwire/sysfs_slave_dpn.c | 3 ++ fs/sysfs/group.c | 45 ++++++++++++++++---- include/linux/sysfs.h | 63 ++++++++++++++++++++++------ 6 files changed, 126 insertions(+), 58 deletions(-) -- 2.43.0

5 22

[PATCH 1/2] soundwire: amd: use inline function for register update
by Vijendar Mukunda 28 Mar '24

28 Mar '24

Define common inline function for register update. Use this inline function for updating SoundWire Pad registers and enable/disable SoundWire interrupt control registers. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_init.c | 36 +++++++++++++++------------------ drivers/soundwire/amd_init.h | 8 ++++++++ drivers/soundwire/amd_manager.c | 13 ++++++------ 3 files changed, 30 insertions(+), 27 deletions(-) diff --git a/drivers/soundwire/amd_init.c b/drivers/soundwire/amd_init.c index e45dc8261ab1..4cd26f3a21f5 100644 --- a/drivers/soundwire/amd_init.c +++ b/drivers/soundwire/amd_init.c @@ -17,42 +17,38 @@ #define ACP_PAD_PULLDOWN_CTRL 0x0001448 #define ACP_SW_PAD_KEEPER_EN 0x0001454 -#define AMD_SDW_PAD_PULLDOWN_CTRL_ENABLE_MASK 0x7f9a -#define AMD_SDW0_PAD_PULLDOWN_CTRL_ENABLE_MASK 0x7f9f -#define AMD_SDW1_PAD_PULLDOWN_CTRL_ENABLE_MASK 0x7ffa -#define AMD_SDW0_PAD_EN_MASK 1 -#define AMD_SDW1_PAD_EN_MASK 0x10 -#define AMD_SDW_PAD_EN_MASK (AMD_SDW0_PAD_EN_MASK | AMD_SDW1_PAD_EN_MASK) +#define AMD_SDW0_PAD_CTRL_MASK 0x60 +#define AMD_SDW1_PAD_CTRL_MASK 5 +#define AMD_SDW_PAD_CTRL_MASK (AMD_SDW0_PAD_CTRL_MASK | AMD_SDW1_PAD_CTRL_MASK) +#define AMD_SDW0_PAD_EN 1 +#define AMD_SDW1_PAD_EN 0x10 +#define AMD_SDW_PAD_EN (AMD_SDW0_PAD_EN | AMD_SDW1_PAD_EN) static int amd_enable_sdw_pads(void __iomem *mmio, u32 link_mask, struct device *dev) { - u32 val; - u32 pad_keeper_en_mask, pad_pulldown_ctrl_mask; + u32 pad_keeper_en, pad_pulldown_ctrl_mask; switch (link_mask) { case 1: - pad_keeper_en_mask = AMD_SDW0_PAD_EN_MASK; - pad_pulldown_ctrl_mask = AMD_SDW0_PAD_PULLDOWN_CTRL_ENABLE_MASK; + pad_keeper_en = AMD_SDW0_PAD_EN; + pad_pulldown_ctrl_mask = AMD_SDW0_PAD_CTRL_MASK; break; case 2: - pad_keeper_en_mask = AMD_SDW1_PAD_EN_MASK; - pad_pulldown_ctrl_mask = AMD_SDW1_PAD_PULLDOWN_CTRL_ENABLE_MASK; + pad_keeper_en = AMD_SDW1_PAD_EN; + pad_pulldown_ctrl_mask = AMD_SDW1_PAD_CTRL_MASK; break; case 3: - pad_keeper_en_mask = AMD_SDW_PAD_EN_MASK; - pad_pulldown_ctrl_mask = AMD_SDW_PAD_PULLDOWN_CTRL_ENABLE_MASK; + pad_keeper_en = AMD_SDW_PAD_EN; + pad_pulldown_ctrl_mask = AMD_SDW_PAD_CTRL_MASK; break; default: dev_err(dev, "No SDW Links are enabled\n"); return -ENODEV; } - val = readl(mmio + ACP_SW_PAD_KEEPER_EN); - val |= pad_keeper_en_mask; - writel(val, mmio + ACP_SW_PAD_KEEPER_EN); - val = readl(mmio + ACP_PAD_PULLDOWN_CTRL); - val &= pad_pulldown_ctrl_mask; - writel(val, mmio + ACP_PAD_PULLDOWN_CTRL); + amd_updatel(mmio, ACP_SW_PAD_KEEPER_EN, pad_keeper_en, pad_keeper_en); + amd_updatel(mmio, ACP_PAD_PULLDOWN_CTRL, pad_pulldown_ctrl_mask, 0); + return 0; } diff --git a/drivers/soundwire/amd_init.h b/drivers/soundwire/amd_init.h index 928b0c707162..5e7b43836a37 100644 --- a/drivers/soundwire/amd_init.h +++ b/drivers/soundwire/amd_init.h @@ -10,4 +10,12 @@ int amd_sdw_manager_start(struct amd_sdw_manager *amd_manager); +static inline void amd_updatel(void __iomem *mmio, int offset, u32 mask, u32 val) +{ + u32 tmp; + + tmp = readl(mmio + offset); + tmp = (tmp & ~mask) | val; + writel(tmp, mmio + offset); +} #endif diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 7cd24bd8e224..1066d87aa011 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -89,9 +89,8 @@ static void amd_enable_sdw_interrupts(struct amd_sdw_manager *amd_manager) u32 val; mutex_lock(amd_manager->acp_sdw_lock); - val = readl(amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(amd_manager->instance)); - val |= sdw_manager_reg_mask_array[amd_manager->instance]; - writel(val, amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(amd_manager->instance)); + val = sdw_manager_reg_mask_array[amd_manager->instance]; + amd_updatel(amd_manager->acp_mmio, ACP_EXTERNAL_INTR_CNTL(amd_manager->instance), val, val); mutex_unlock(amd_manager->acp_sdw_lock); writel(AMD_SDW_IRQ_MASK_0TO7, amd_manager->mmio + @@ -103,12 +102,12 @@ static void amd_enable_sdw_interrupts(struct amd_sdw_manager *amd_manager) static void amd_disable_sdw_interrupts(struct amd_sdw_manager *amd_manager) { - u32 val; + u32 irq_mask; mutex_lock(amd_manager->acp_sdw_lock); - val = readl(amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(amd_manager->instance)); - val &= ~sdw_manager_reg_mask_array[amd_manager->instance]; - writel(val, amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(amd_manager->instance)); + irq_mask = sdw_manager_reg_mask_array[amd_manager->instance]; + amd_updatel(amd_manager->acp_mmio, ACP_EXTERNAL_INTR_CNTL(amd_manager->instance), + irq_mask, 0); mutex_unlock(amd_manager->acp_sdw_lock); writel(0x00, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_MASK_0TO7); -- 2.34.1

2 3

[PATCH] ALSA: hda: cs35l56: Add ACPI device match tables
by Simon Trimmer 28 Mar '24

28 Mar '24

Adding the ACPI HIDs to the match table triggers the cs35l56-hda modules to be loaded on boot so that Serial Multi Instantiate can add the devices to the bus and begin the driver init sequence. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") --- sound/pci/hda/cs35l56_hda_i2c.c | 13 +++++++++++-- sound/pci/hda/cs35l56_hda_spi.c | 13 +++++++++++-- 2 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/cs35l56_hda_i2c.c b/sound/pci/hda/cs35l56_hda_i2c.c index 13beee807308..40f2f97944d5 100644 --- a/sound/pci/hda/cs35l56_hda_i2c.c +++ b/sound/pci/hda/cs35l56_hda_i2c.c @@ -56,10 +56,19 @@ static const struct i2c_device_id cs35l56_hda_i2c_id[] = { {} }; +static const struct acpi_device_id cs35l56_acpi_hda_match[] = { + { "CSC3554", 0 }, + { "CSC3556", 0 }, + { "CSC3557", 0 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, cs35l56_acpi_hda_match); + static struct i2c_driver cs35l56_hda_i2c_driver = { .driver = { - .name = "cs35l56-hda", - .pm = &cs35l56_hda_pm_ops, + .name = "cs35l56-hda", + .acpi_match_table = cs35l56_acpi_hda_match, + .pm = &cs35l56_hda_pm_ops, }, .id_table = cs35l56_hda_i2c_id, .probe = cs35l56_hda_i2c_probe, diff --git a/sound/pci/hda/cs35l56_hda_spi.c b/sound/pci/hda/cs35l56_hda_spi.c index a3b2fa76663d..7f02155fe61e 100644 --- a/sound/pci/hda/cs35l56_hda_spi.c +++ b/sound/pci/hda/cs35l56_hda_spi.c @@ -56,10 +56,19 @@ static const struct spi_device_id cs35l56_hda_spi_id[] = { {} }; +static const struct acpi_device_id cs35l56_acpi_hda_match[] = { + { "CSC3554", 0 }, + { "CSC3556", 0 }, + { "CSC3557", 0 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, cs35l56_acpi_hda_match); + static struct spi_driver cs35l56_hda_spi_driver = { .driver = { - .name = "cs35l56-hda", - .pm = &cs35l56_hda_pm_ops, + .name = "cs35l56-hda", + .acpi_match_table = cs35l56_acpi_hda_match, + .pm = &cs35l56_hda_pm_ops, }, .id_table = cs35l56_hda_spi_id, .probe = cs35l56_hda_spi_probe, -- 2.34.1

2 1

[PATCH 0/3] ASoC: Use snd_ctl_find_id_mixer() instead of open-coding
by Richard Fitzgerald 28 Mar '24

28 Mar '24

The first two patches change snd_soc_card_get_kcontrol() to use the core snd_ctl_find_id_mixer() functionality instead of open-coding its own list walk. The last patch adds a KUnit test for this, which was tested on the original and modified code. Richard Fitzgerald (3): ALSA: control: Introduce snd_ctl_find_id_mixer_locked() ASoC: soc-card: Use snd_ctl_find_id_mixer() instead of open-coding ASoC: soc-card: Add KUnit test case for snd_soc_card_get_kcontrol include/sound/control.h | 23 +++++ sound/soc/Kconfig | 8 ++ sound/soc/Makefile | 4 + sound/soc/soc-card-test.c | 191 ++++++++++++++++++++++++++++++++++++++ sound/soc/soc-card.c | 21 +---- 5 files changed, 230 insertions(+), 17 deletions(-) create mode 100644 sound/soc/soc-card-test.c -- 2.39.2

2 4

Dummy output with ES8326 codec
by GitHub issues - opened 28 Mar '24

28 Mar '24

alsa-project/alsa-ucm-conf issue #407 was opened from Henrythek: ## Description During the process of using the ES8326 codec, I encountered a dummy output issue. killall pulseaudio do not work, even after masking and then unmasking the pulseaudio service and attempting to restart it, the issue persists. The dummy output occurs intermittently when plugging in or unplugging headphones, and rebooting can temporarily resolve the issue. I am using the following configuration: - Operating System: Debian GNU/Linux 12 - Kernel Version: 6.7.0 ## Error Messages E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="0" name="pci-0000_00_1f.3-platform-sof-essx8336" card_name="alsa_card.pci-0000_00_1f.3-platform-sof-essx8336" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:1f.3/sof-essx8336/sound/card0 (alsa_card.pci-0000_00_1f.3-platform-sof-essx8336) failed to load module. ## Additional Information The UCM (Use Case Manager) configuration used is from the following repository: [alsa-ucm-conf/ucm2/Intel/sof-essx8336](https://github.com/alsa-project/alsa… ## Steps to Reproduce 1. Use the ES8326 codec on a Debian GNU/Linux 12 system with Kernel 6.7.0. 2. Attempt to use the audio functionality. 3. Plug in or unplug headphones. 4. Observe the dummy output issue and the error messages mentioned above. ## Additional Context [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/14782974… [aplay.txt](https://github.com/alsa-project/alsa-ucm-conf/files/14782975/apl… [pulseaudio_v.txt](https://github.com/alsa-project/alsa-ucm-conf/files/14782… ![Screenshot from 2024-03-27 21-44-34](https://github.com/alsa-project/alsa-ucm-conf/assets/68213530/925… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/407 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 00/15] ASoC: replace dpcm_playback/capture to playback/capture_only
by Kuninori Morimoto 28 Mar '24

28 Mar '24

Hi Mark When we use DPCM, we need to set dpcm_playback/capture flag. If these flag are set, soc_get_playback_capture() will check its availability, but non DPCM case is checked without special flags. OTOH, it cares playback/capture_only flag. It is needed. This patch remove DPCM special flag, and replace it playback/capture_only flag if needed. Kuninori Morimoto (15): ASoC: soc-pcm.c: cleanup soc_get_playback_capture() ASoC: amd: replace dpcm_playback/capture to playback/capture_only ASoC: fsl: replace dpcm_playback/capture to playback/capture_only ASoC: sof: replace dpcm_playback/capture to playback/capture_only ASoC: meson: replace dpcm_playback/capture to playback/capture_only ASoC: Intel: replace dpcm_playback/capture to playback/capture_only ASoC: samsung: replace dpcm_playback/capture to playback/capture_only ASoC: mediatek: replace dpcm_playback/capture to playback/capture_only ASoC: Intel/avs: replace dpcm_playback/capture to playback/capture_only ASoC: soc-core.c: replace dpcm_playback/capture to playback/capture_only ASoC: soc-topology.c: replace dpcm_playback/capture to playback/capture_only ASoC: soc-compress.c: replace dpcm_playback/capture to playback/capture_only ASoC: remove snd_soc_dai_link_set_capabilities() ASoC: soc-pcm.c: remove dpcm_playback/capture ASoC: soc-pcm: indicate warning if DPCM BE Codec has no settings include/sound/soc-dai.h | 1 - include/sound/soc.h | 4 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++--- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 16 ++-- sound/soc/fsl/imx-audmix.c | 6 +- sound/soc/fsl/imx-card.c | 7 +- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 32 ++++--- sound/soc/intel/boards/bxt_rt298.c | 26 +++--- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 14 ++- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/glk_rt5682_max98357a.c | 30 +++---- sound/soc/intel/boards/hsw_rt5640.c | 10 +-- sound/soc/intel/boards/kbl_da7219_max98357a.c | 26 +++--- sound/soc/intel/boards/kbl_da7219_max98927.c | 54 +++++------- sound/soc/intel/boards/kbl_rt5660.c | 18 ++-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 44 +++++----- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 22 ++--- sound/soc/intel/boards/skl_hda_dsp_common.c | 14 ++- .../soc/intel/boards/skl_nau88l25_max98357a.c | 26 +++--- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 26 +++--- sound/soc/intel/boards/skl_rt286.c | 26 +++--- sound/soc/intel/boards/sof_board_helpers.c | 12 +-- sound/soc/intel/boards/sof_da7219.c | 33 +++---- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 8 +- sound/soc/intel/boards/sof_sdw.c | 4 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 ++--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +-- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +-- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++---- .../mt8186/mt8186-mt6366-da7219-max98357.c | 86 +++++++----------- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 86 +++++++----------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------ .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 ++++++++-------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 ++++++------- sound/soc/meson/axg-card.c | 9 +- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/samsung/odroid.c | 11 ++- sound/soc/soc-compress.c | 10 ++- sound/soc/soc-core.c | 20 +---- sound/soc/soc-dai.c | 38 -------- sound/soc/soc-pcm.c | 88 ++++++++----------- sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 4 - 91 files changed, 507 insertions(+), 876 deletions(-) -- 2.25.1

4 25

[PATCH 00/34] ASoC: Intel: boards: updates for 6.10 - part2
by Pierre-Louis Bossart 27 Mar '24

27 Mar '24

This second part provides SoundWire-related cleanups and extensions required by Realtek RT722 and Cirrus Logic codecs. Also included is a cleanup of the RT715-sdca DAI naming and new tables for ACPI-based board detections. Balamurugan C (1): ASoC: Intel: soc-acpi-intel-arl-match: Add rt711 sdca codec support Bard Liao (7): ASoC: Intel: sof_sdw: use generic rtd_init function for Realtek SDW DMICs ASoC: Intel: sof_sdw: remove unused rt dmic rtd_init ASoC: Intel: sof_sdw_rt722_sdca: set rtd_init in codec_info_list[] ASoC: Intel: sof_sdw_rt722_sdca: use rt_dmic_rtd_init ASoC: rt715-sdca: rename dai name with rt715-sdca prefix ASoC: Intel: sof_sdw: change rt715-sdca dai name ASoC: Intel: change cs35l56 name_prefix Charles Keepax (26): ASoC: intel: sof_sdw: Make find_codec_info_part() return a pointer ASoC: intel: sof_sdw: Make find_codec_info_acpi() return a pointer ASoC: intel: sof_sdw: Make find_codec_info_dai() return a pointer ASoC: intel: sof_sdw: Only pass codec_conf pointer around ASoC: intel: sof_sdw: Set channel map directly from endpoints ASoC: Intel: sof_sdw: Move get_codec_dai_by_name() into sof_sdw itself ASoC: Intel: sof_sdw: Move flags to private struct ASoC: Intel: sof_sdw: Only pass dai_link pointer around ASoC: Intel: sof_sdw: Use for_each_set_bit ASoC: Intel: sof_sdw: Factor out SSP DAI creation ASoC: Intel: sof_sdw: Factor out DMIC DAI creation. ASoC: Intel: sof_sdw: Factor out HDMI DAI creation ASoC: Intel: sof_sdw: Factor out BlueTooth DAI creation ASoC: Intel: sof_sdw: Factor out codec name generation ASoC: Intel: sof_sdw: Remove no longer supported quirk ASoC: intel: soc-acpi: Add missing cs42l43 endpoints ASoC: Intel: sof-sdw: Add new code for parsing the snd_soc_acpi structs ASoC: Intel: sof_sdw: Move counting and codec_conf to new parsing ASoC: Intel: sof_sdw: Move ignore_pch_dmic to new parsing ASoC: Intel: sof_sdw: Move append_dai_type to new parsing ASoC: Intel: sof_sdw: Move generation of DAI links to new parsing ASoC: intel: sof_sdw: Factor out SoundWire DAI creation ASoC: Intel: sof_sdw: Don't pass acpi_link_adr to init functions ASoC: Intel: sof_sdw: Remove redundant initialisations ASoC: Intel: sof_sdw: Add quirk for optional codec speakers ASoC: Intel: sof_sdw: Add support for cs42l43 optional speaker output sound/soc/codecs/rt715-sdca.c | 4 +- sound/soc/intel/boards/Kconfig | 1 - sound/soc/intel/boards/Makefile | 4 +- sound/soc/intel/boards/sof_board_helpers.c | 18 - sound/soc/intel/boards/sof_board_helpers.h | 3 - sound/soc/intel/boards/sof_sdw.c | 1118 ++++++++--------- sound/soc/intel/boards/sof_sdw_common.h | 42 +- sound/soc/intel/boards/sof_sdw_cs42l42.c | 1 - sound/soc/intel/boards/sof_sdw_cs42l43.c | 50 + sound/soc/intel/boards/sof_sdw_cs_amp.c | 1 - sound/soc/intel/boards/sof_sdw_maxim.c | 1 - sound/soc/intel/boards/sof_sdw_rt5682.c | 1 - sound/soc/intel/boards/sof_sdw_rt700.c | 1 - sound/soc/intel/boards/sof_sdw_rt711.c | 2 - sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 25 - sound/soc/intel/boards/sof_sdw_rt715.c | 26 - sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 26 - sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 38 +- sound/soc/intel/boards/sof_sdw_rt_amp.c | 1 - sound/soc/intel/boards/sof_sdw_rt_dmic.c | 54 + .../boards/sof_sdw_rt_sdca_jack_common.c | 4 +- sound/soc/intel/boards/sof_ssp_common.h | 7 - .../intel/common/soc-acpi-intel-arl-match.c | 24 + .../intel/common/soc-acpi-intel-mtl-match.c | 35 +- .../intel/common/soc-acpi-intel-tgl-match.c | 35 +- 25 files changed, 753 insertions(+), 769 deletions(-) delete mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c delete mode 100644 sound/soc/intel/boards/sof_sdw_rt715_sdca.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt_dmic.c -- 2.40.1

2 35

[PATCH 00/25] Introduce support of audio for Amlogic A1 SoC family
by Jan Dakinevich 27 Mar '24

27 Mar '24

This series includes the following: - new audio clock and reset controller data and adaptation for it of existing code (patches 0001..0004); - adaptation of existing audio components for A1 Soc (patches 0005..0021); - handy cosmetics for dai-link naming (patches 0022..0023); - integration of audio devices into common trees (patch 0024); - audio support bring up on Amlogic ad402 reference board (patch 0025). This patch is not actually checked on real hardware (because all ad402 that we had were burned out). This patch is based on ad402's schematics and on experience with our own hardware (which is very close to reference board); Dmitry Rokosov (2): ASoC: dt-bindings: meson: introduce link-name optional property ASoC: meson: implement link-name optional property in meson card utils Jan Dakinevich (23): clk: meson: a1: restrict an amount of 'hifi_pll' params clk: meson: axg: move reset controller's code to separate module dt-bindings: clock: meson: add A1 audio clock and reset controller bindings clk: meson: a1: add the audio clock controller driver ASoC: meson: codec-glue: add support for capture stream ASoC: meson: g12a-toacodec: fix "Lane Select" width ASoC: meson: g12a-toacodec: rework the definition of bits ASoC: dt-bindings: meson: g12a-toacodec: add support for A1 SoC family ASoC: meson: g12a-toacodec: add support for A1 SoC family ASoC: meson: t9015: prepare to adding new platforms ASoC: dt-bindings: meson: t9015: add support for A1 SoC family ASoC: meson: t9015: add support for A1 SoC family ASoC: dt-bindings: meson: axg-pdm: document 'sysrate' property ASoC: meson: axg-pdm: introduce 'sysrate' property pinctrl/meson: fix typo in PDM's pin name ASoC: dt-bindings: meson: meson-axg-audio-arb: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-fifo: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-pdm: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-sound-card: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-tdm-formatters: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-tdm-iface: claim support of A1 SoC family arm64: dts: meson: a1: add audio devices arm64: dts: ad402: enable audio .../bindings/clock/amlogic,a1-audio-clkc.yaml | 83 +++ .../reset/amlogic,meson-axg-audio-arb.yaml | 10 +- .../bindings/sound/amlogic,axg-fifo.yaml | 8 + .../bindings/sound/amlogic,axg-pdm.yaml | 5 + .../sound/amlogic,axg-sound-card.yaml | 12 +- .../sound/amlogic,axg-tdm-formatters.yaml | 22 +- .../bindings/sound/amlogic,axg-tdm-iface.yaml | 6 +- .../bindings/sound/amlogic,g12a-toacodec.yaml | 1 + .../bindings/sound/amlogic,gx-sound-card.yaml | 6 + .../bindings/sound/amlogic,t9015.yaml | 4 +- .../arm64/boot/dts/amlogic/meson-a1-ad402.dts | 126 ++++ arch/arm64/boot/dts/amlogic/meson-a1.dtsi | 471 +++++++++++++++ drivers/clk/meson/Kconfig | 18 + drivers/clk/meson/Makefile | 2 + drivers/clk/meson/a1-audio.c | 556 ++++++++++++++++++ drivers/clk/meson/a1-audio.h | 58 ++ drivers/clk/meson/a1-pll.c | 8 +- drivers/clk/meson/axg-audio.c | 95 +-- drivers/clk/meson/meson-audio-rstc.c | 109 ++++ drivers/clk/meson/meson-audio-rstc.h | 12 + drivers/pinctrl/meson/pinctrl-meson-a1.c | 6 +- .../dt-bindings/clock/amlogic,a1-audio-clkc.h | 122 ++++ .../reset/amlogic,meson-a1-audio-reset.h | 29 + .../dt-bindings/sound/meson-g12a-toacodec.h | 5 + sound/soc/meson/axg-pdm.c | 10 +- sound/soc/meson/g12a-toacodec.c | 298 ++++++++-- sound/soc/meson/meson-card-utils.c | 12 +- sound/soc/meson/meson-codec-glue.c | 174 ++++-- sound/soc/meson/meson-codec-glue.h | 23 + sound/soc/meson/t9015.c | 326 +++++++++- 30 files changed, 2394 insertions(+), 223 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/amlogic,a1-audio-clkc.yaml create mode 100644 drivers/clk/meson/a1-audio.c create mode 100644 drivers/clk/meson/a1-audio.h create mode 100644 drivers/clk/meson/meson-audio-rstc.c create mode 100644 drivers/clk/meson/meson-audio-rstc.h create mode 100644 include/dt-bindings/clock/amlogic,a1-audio-clkc.h create mode 100644 include/dt-bindings/reset/amlogic,meson-a1-audio-reset.h -- 2.34.1

8 84

[PATCH v2 0/4] ALSA: hda/tas2781: fixes for 6.9-rc1
by Gergo Koteles 27 Mar '24

27 Mar '24

Hi, This series removes the no longer needed digital gain kcontrol, which caused problems in tas2563, because the register does not exists there. This series also adds locking and debug statements to the other kcontrols. They sometimes ran in parallel with tasdev_fw_ready, and caused weird sound problems. https://github.com/tomsom/yoga-linux/issues/58 Regards, Gergo Changes in v2: - Do not remove dvc_tlv scale from tas2781-tlv.h as it is also used by sound/soc/codecs/tas2781-i2c.c - Add kcontrol name to debug statements - Add a new patch to remove a useless debug statement. [1]: https://lore.kernel.org/all/cover.1711401621.git.soyer@irl.hu/ Gergo Koteles (4): ALSA: hda/tas2781: remove digital gain kcontrol ALSA: hda/tas2781: add locks to kcontrols ALSA: hda/tas2781: add debug statements to kcontrols ALSA: hda/tas2781: remove useless dev_dbg from playback_hook sound/pci/hda/tas2781_hda_i2c.c | 118 +++++++++++++++++++++----------- 1 file changed, 77 insertions(+), 41 deletions(-) base-commit: 4cece764965020c22cff7665b18a012006359095 -- 2.44.0

2 5

How to add jack GPIOs to audio-graph-cards?
by John Watts 27 Mar '24

27 Mar '24

Hi there, audio-graph-card's device tree schema indicates it supports the following: - pa-gpios - hp-det-gpio - mic-det-gpio But only implements pa-gpios. audio-graph-card2 doesn't implement any of these but will still attempts to grab pa-gpios but never use it (possibly a bug?) When using audio graph cards, how are you intended to add detection GPIOs? Are there future plans to add a jacks { } section of some sort for this? John.

2 1

Writing documentation for simple device tree ASoC cards
by John Watts 27 Mar '24

27 Mar '24

Hello! I've spent way too much time digging through emails, documentation and kernel source code to form a mental model that lets me understand and use simple-sound-card, audio-graph-card and audio-graph-card2. There's really no introductory documentation geared towards people who are writing device trees intending to glue together multiple audio components in to a single sound card. I'm deciding I'll try and take up this task to provide some basic documentation for this. Before starting I would like to quickly go over some notes in case I'm actually just incredibly wrong. I'd rather find that out now than later. As a general ASoC refresher: - ALSA provides PCM interfaces and controls to a sound card - ASoC introduces the concept of components - Components are what you instantiate with the device tree - Components can be asked to set a clock or create a PCM stream - Components have one or more DAIs - DAIs represent capabilities of a stream - Components have DAPM widgets for power management - These can be related to physical pins on the board or DAI streams - CPU components are ones that ALSA can use directly to process data - Codec components provide information about DAIs or drive streams Most of this is already explained in the documentation. The generic sound cards you can create using simple-sound-card and audio-graph-card are machine drivers. Their jobs are to ensure that when using a DAI all the related components, clocking and power management is set up and ready for use. This is done by: - Linking multiple DAIs together - Connecting DAPM widgets together to power on components - Calculate and setting the shared clock (usually mclk) - Setting DAI properties (like TDM slots and width) - Adding widgets for jack detection DAI linking isn't explained in the documentation but was a core part of my confusion when trying to use these cards: I didn't understand if it represented some relationship between physical pins, PCM channels, etc. My fuzzy model of a DAI link is that it exists to group DAIs (and thus components) necessary to acheive a desired data flow, where that flow is left up to the components themselves to implement. There's a lot of examples for actually using the simple family of cards but I imagine documenting the above would help people troubleshoot or describe their problems better. Other minor notes: - sound-name-prefix is very useful and fits in somewhere here - audio-graph-cards don't support GPIO jacks from what I can see I also have noticed that with audio-graph-card you can't use a DAI multiple times in different cards. This makes sense but is this due to the way graphs and endpoints work or due to the underlying sound system? This problem and jacks seem to point towards maybe allowing specification of DAPM widgets in the device tree, and maybe kcontrols for selecting which codec to use? I'm not sure. It would also be useful to understand the scope of these simple cards. Complex audio needs like mine seem to be outside the scope at the moment and require writing a custom card driver. Perhaps it's worth taking that as an opportunity to write a driver using the simple card framework as an example? John.

1 0

[PATCH 0/3] ALSA: hda/tas2781: fixes for 6.9-rc1
by Gergo Koteles 27 Mar '24

27 Mar '24

Hi, This series removes the no longer needed digital gain kcontrol, which caused problems in tas2563, because the register does not exists there. This series also adds locking and debug statements to the other kcontrols. They sometimes ran in parallel with tasdev_fw_ready, and caused weird sound problems. https://github.com/tomsom/yoga-linux/issues/58 Regards, Gergo Gergo Koteles (3): ALSA: hda/tas2781: remove digital gain kcontrol ALSA: hda/tas2781: add locks to kcontrols ALSA: hda/tas2781: add debug statements to kcontrols include/sound/tas2781-tlv.h | 1 - sound/pci/hda/tas2781_hda_i2c.c | 103 +++++++++++++++++++++----------- 2 files changed, 68 insertions(+), 36 deletions(-) base-commit: 4cece764965020c22cff7665b18a012006359095 -- 2.44.0

5 9

[PATCH] ASoC: sti: uniperif: fix the undefined bitwise shift behavior problem
by Su Hui 27 Mar '24

27 Mar '24

Clang static checker(scan-build): sound/soc/sti/uniperif_player.c:1115:12: warning: The result of the left shift is undefined because the right operand is negative [core.UndefinedBinaryOperatorResult] When UNIPERIF_CONFIG_BACK_STALL_REQ_SHIFT(ip) equals to -1, the result of SET_UNIPERIF_CONFIG_BACK_STALL_REQ_DISABLE(ip) is undefined. Here are some results of using different compilers. 1UL >> -1 1UL << -1 gcc 10.2.1 0x2 0 gcc 11.4.0 0 0x8000000000000000 clang 14.0.0 0x64b8a45d72a0 0x64b8a45d72a0 clang 17.0.0 0x556c43b0f2a0 0x556c43b0f2a0 Add some macros to ensure that when right opreand is negative or other invalid values, the results of bitwise shift is zero. Fixes: e1ecace6a685 ("ASoC: sti: Add uniperipheral header file") Signed-off-by: Su Hui <suhui(a)nfschina.com> --- sound/soc/sti/uniperif.h | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 2a5de328501c..1cbff01fbff0 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -12,17 +12,28 @@ #include <sound/dmaengine_pcm.h> +#define SR_SHIFT(a, b) ({unsigned long __a = (a); \ + unsigned int __b = (b); \ + __b < BITS_PER_LONG ? \ + __a >> __b : 0; }) + +#define SL_SHIFT(a, b) ({unsigned long __a = (a); \ + unsigned int __b = (b); \ + __b < BITS_PER_LONG ? \ + __a << __b : 0; }) + /* * Register access macros */ #define GET_UNIPERIF_REG(ip, offset, shift, mask) \ - ((readl_relaxed(ip->base + offset) >> shift) & mask) + (SR_SHIFT(readl_relaxed(ip->base + offset), shift) & mask) #define SET_UNIPERIF_REG(ip, offset, shift, mask, value) \ writel_relaxed(((readl_relaxed(ip->base + offset) & \ - ~(mask << shift)) | (((value) & mask) << shift)), ip->base + offset) + ~SL_SHIFT(mask, shift)) | SL_SHIFT(((value) & mask), shift)),\ + ip->base + offset) #define SET_UNIPERIF_BIT_REG(ip, offset, shift, mask, value) \ - writel_relaxed((((value) & mask) << shift), ip->base + offset) + writel_relaxed(SL_SHIFT(((value) & mask), shift), ip->base + offset) /* * UNIPERIF_SOFT_RST reg -- 2.30.2

2 3

[PATCH v4 0/7] Add support for the internal RK3308 audio codec
by Luca Ceresoli 26 Mar '24

26 Mar '24

This series adds a driver for the internal audio codec of the Rockchip RK3308 SoC, along with some related patches. This codec is internally connected to the I2S peripherals on the same chip, and it has some peculiarities arising from that interconnection. For proper bidirectional operation with the internal codec at any possible combination of sampling rates, the I2S peripheral needs two clock sources (tx and rx), while connection with an external codec commonly needs only one. Since v5.16 there is a driver for the I2S in sound/soc/rockchip/rockchip_i2s_tdm.c, but in some cases it does not configure correctly the clocks, resulting in an unnecessarily inaccurate rate. Patch 1 fixes this. Patches 2-4 add the codec driver along with the bindings and a new helper macro. Patches 5-7 add to the SoC DT file two I2S controllers (those which are internally connected to the internal codec) and the codec itself and enable the driver in the ARM64 defconfig. Luca Signed-off-by: Luca Ceresoli <luca.ceresoli(a)bootlin.com> --- Changes in v4: - several cleanups in the codec probe function - Link to v3: https://lore.kernel.org/r/20240221-rk3308-audio-codec-v3-0-dfa34abfcef6@boo… Changes in v3: - Add the I2S clock fix patch and remove a previous fix which is now superseded - Codec driver: fix silent playback until a given amplitude of sigital value, seen at >= 96 kHz rate - various other changes, listed per-patch - Link to v2: https://lore.kernel.org/r/20231219-rk3308-audio-codec-v2-0-c70d06021946@boo… Changes in v2: - largely rewrote the codec driver to use DAPM and lots of improvements and cleanups - removed the RK3308 audio card and related patches - various other changes, listed per-patch - Link to v1: https://lore.kernel.org/all/20220907142124.2532620-1-luca.ceresoli@bootlin.… --- Luca Ceresoli (7): ASoC: rockchip: i2s-tdm: Fix inaccurate sampling rates ASoC: dt-bindings: Add Rockchip RK3308 internal audio codec ASoC: core: add SOC_DOUBLE_RANGE_TLV() helper macro ASoC: codecs: Add RK3308 internal audio codec driver arm64: defconfig: enable Rockchip RK3308 internal audio codec driver arm64: dts: rockchip: add i2s_8ch_2 and i2s_8ch_3 arm64: dts: rockchip: add the internal audio codec .../bindings/sound/rockchip,rk3308-codec.yaml | 98 +++ MAINTAINERS | 7 + arch/arm64/boot/dts/rockchip/rk3308.dtsi | 56 ++ arch/arm64/configs/defconfig | 1 + include/sound/soc.h | 12 + sound/soc/codecs/Kconfig | 11 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk3308_codec.c | 974 +++++++++++++++++++++ sound/soc/codecs/rk3308_codec.h | 579 ++++++++++++ sound/soc/rockchip/rockchip_i2s_tdm.c | 352 +------- 10 files changed, 1746 insertions(+), 346 deletions(-) --- base-commit: dfda120c512b3edca1436f770924e91b14f93a98 change-id: 20231219-rk3308-audio-codec-a5558ba8949d Best regards, -- Luca Ceresoli <luca.ceresoli(a)bootlin.com>

4 12

[RFC PATCH v2] ASoC: meson: g12a-toacodec: rework the definition of bits
by Jan Dakinevich 26 Mar '24

26 Mar '24

There is a lot of defines, but almost all of them are not used. Lets rework them: - keep separate the definition for different platforms to make easier checking that they match documentation. - use LSB/MSB sufixes for uniformity. - don't use hard-coded values for already declared defines. Signed-off-by: Jan Dakinevich <jan.dakinevich(a)salutedevices.com> --- Links: [1] https://lore.kernel.org/lkml/20240314232201.2102178-1-jan.dakinevich@salute… Changes v1 -> v2: - Detached from v1's series (patch 7). - Fixed my wrong understanding of SOC_SINGLE's input parameters. sound/soc/meson/g12a-toacodec.c | 79 ++++++++++++++++++++------------- 1 file changed, 49 insertions(+), 30 deletions(-) diff --git a/sound/soc/meson/g12a-toacodec.c b/sound/soc/meson/g12a-toacodec.c index 531bb8707a3e..22181f4bab72 100644 --- a/sound/soc/meson/g12a-toacodec.c +++ b/sound/soc/meson/g12a-toacodec.c @@ -20,26 +20,37 @@ #define G12A_TOACODEC_DRV_NAME "g12a-toacodec" #define TOACODEC_CTRL0 0x0 -#define CTRL0_ENABLE_SHIFT 31 -#define CTRL0_DAT_SEL_SM1_MSB 19 -#define CTRL0_DAT_SEL_SM1_LSB 18 -#define CTRL0_DAT_SEL_MSB 15 -#define CTRL0_DAT_SEL_LSB 14 -#define CTRL0_LANE_SEL_SM1 16 -#define CTRL0_LANE_SEL 12 -#define CTRL0_LRCLK_SEL_SM1_MSB 14 -#define CTRL0_LRCLK_SEL_SM1_LSB 12 -#define CTRL0_LRCLK_SEL_MSB 9 -#define CTRL0_LRCLK_SEL_LSB 8 -#define CTRL0_LRCLK_INV_SM1 BIT(10) -#define CTRL0_BLK_CAP_INV_SM1 BIT(9) -#define CTRL0_BLK_CAP_INV BIT(7) -#define CTRL0_BCLK_O_INV_SM1 BIT(8) -#define CTRL0_BCLK_O_INV BIT(6) -#define CTRL0_BCLK_SEL_SM1_MSB 6 -#define CTRL0_BCLK_SEL_MSB 5 -#define CTRL0_BCLK_SEL_LSB 4 -#define CTRL0_MCLK_SEL GENMASK(2, 0) + +/* Common bits */ +#define CTRL0_ENABLE_SHIFT 31 +#define CTRL0_MCLK_SEL GENMASK(2, 0) + +/* G12A bits */ +#define CTRL0_DAT_SEL_G12A_MSB 15 +#define CTRL0_DAT_SEL_G12A_LSB 14 +#define CTRL0_LANE_SEL_G12A_MSB 13 +#define CTRL0_LANE_SEL_G12A_LSB 12 +#define CTRL0_LANE_SEL_G12A_MAX 3 +#define CTRL0_LRCLK_SEL_G12A_MSB 9 +#define CTRL0_LRCLK_SEL_G12A_LSB 8 +#define CTRL0_BLK_CAP_INV_G12A BIT(7) +#define CTRL0_BCLK_O_INV_G12A BIT(6) +#define CTRL0_BCLK_SEL_G12A_MSB 5 +#define CTRL0_BCLK_SEL_G12A_LSB 4 + +/* SM1 bits */ +#define CTRL0_DAT_SEL_SM1_MSB 19 +#define CTRL0_DAT_SEL_SM1_LSB 18 +#define CTRL0_LANE_SEL_SM1_MSB 17 +#define CTRL0_LANE_SEL_SM1_LSB 16 +#define CTRL0_LANE_SEL_SM1_MAX 3 +#define CTRL0_LRCLK_SEL_SM1_MSB 14 +#define CTRL0_LRCLK_SEL_SM1_LSB 12 +#define CTRL0_LRCLK_INV_SM1 BIT(10) +#define CTRL0_BLK_CAP_INV_SM1 BIT(9) +#define CTRL0_BCLK_O_INV_SM1 BIT(8) +#define CTRL0_BCLK_SEL_SM1_MSB 6 +#define CTRL0_BCLK_SEL_SM1_LSB 4 #define TOACODEC_OUT_CHMAX 2 @@ -108,7 +119,7 @@ static int g12a_toacodec_mux_put_enum(struct snd_kcontrol *kcontrol, } static SOC_ENUM_SINGLE_DECL(g12a_toacodec_mux_enum, TOACODEC_CTRL0, - CTRL0_DAT_SEL_LSB, + CTRL0_DAT_SEL_G12A_LSB, g12a_toacodec_mux_texts); static SOC_ENUM_SINGLE_DECL(sm1_toacodec_mux_enum, TOACODEC_CTRL0, @@ -210,7 +221,7 @@ static int g12a_toacodec_component_probe(struct snd_soc_component *c) { /* Initialize the static clock parameters */ return snd_soc_component_write(c, TOACODEC_CTRL0, - CTRL0_BLK_CAP_INV); + CTRL0_BLK_CAP_INV_G12A); } static int sm1_toacodec_component_probe(struct snd_soc_component *c) @@ -229,11 +240,13 @@ static const struct snd_soc_dapm_route g12a_toacodec_routes[] = { }; static const struct snd_kcontrol_new g12a_toacodec_controls[] = { - SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL, 3, 0), + SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL_G12A_LSB, + CTRL0_LANE_SEL_G12A_MAX, 0), }; static const struct snd_kcontrol_new sm1_toacodec_controls[] = { - SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL_SM1, 3, 0), + SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL_SM1_LSB, + CTRL0_LANE_SEL_SM1_MAX, 0), }; static const struct snd_soc_component_driver g12a_toacodec_component_drv = { @@ -266,16 +279,22 @@ static const struct regmap_config g12a_toacodec_regmap_cfg = { static const struct g12a_toacodec_match_data g12a_toacodec_match_data = { .component_drv = &g12a_toacodec_component_drv, - .field_dat_sel = REG_FIELD(TOACODEC_CTRL0, 14, 15), - .field_lrclk_sel = REG_FIELD(TOACODEC_CTRL0, 8, 9), - .field_bclk_sel = REG_FIELD(TOACODEC_CTRL0, 4, 5), + .field_dat_sel = REG_FIELD(TOACODEC_CTRL0, CTRL0_DAT_SEL_G12A_LSB, + CTRL0_DAT_SEL_G12A_MSB), + .field_lrclk_sel = REG_FIELD(TOACODEC_CTRL0, CTRL0_LRCLK_SEL_G12A_LSB, + CTRL0_LRCLK_SEL_G12A_MSB), + .field_bclk_sel = REG_FIELD(TOACODEC_CTRL0, CTRL0_BCLK_SEL_G12A_LSB, + CTRL0_BCLK_SEL_G12A_MSB), }; static const struct g12a_toacodec_match_data sm1_toacodec_match_data = { .component_drv = &sm1_toacodec_component_drv, - .field_dat_sel = REG_FIELD(TOACODEC_CTRL0, 18, 19), - .field_lrclk_sel = REG_FIELD(TOACODEC_CTRL0, 12, 14), - .field_bclk_sel = REG_FIELD(TOACODEC_CTRL0, 4, 6), + .field_dat_sel = REG_FIELD(TOACODEC_CTRL0, CTRL0_DAT_SEL_SM1_LSB, + CTRL0_DAT_SEL_SM1_MSB), + .field_lrclk_sel = REG_FIELD(TOACODEC_CTRL0, CTRL0_LRCLK_SEL_SM1_LSB, + CTRL0_LRCLK_SEL_SM1_MSB), + .field_bclk_sel = REG_FIELD(TOACODEC_CTRL0, CTRL0_BCLK_SEL_SM1_LSB, + CTRL0_BCLK_SEL_SM1_MSB), }; static const struct of_device_id g12a_toacodec_of_match[] = { -- 2.34.1

2 2

[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