From 04f15d8f393037ed72876399882265604a370b2b Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Fri, 3 Jul 2026 14:02:10 +0100 Subject: [PATCH 01/23] ASoC: sof_sdw: Add a quirk for 0x17aa383c laptop using sidecar amps with cs42l43 Add a quirk for a Lenovo laptop (SSID: 0x17aa383c) to allow using sidecar CS35L56 amps with CS42L43 codec. Signed-off-by: Maciej Strozek Link: https://patch.msgid.link/20260703130210.21991-1-mstrozek@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index d43daf9b025d..59eb0328bbb5 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -855,6 +855,7 @@ static const struct snd_pci_quirk sof_sdw_ssid_quirk_table[] = { SND_PCI_QUIRK(0x17aa, 0x2348, "Lenovo P16", SOC_SDW_CODEC_MIC), SND_PCI_QUIRK(0x17aa, 0x2349, "Lenovo P1", SOC_SDW_CODEC_MIC), SND_PCI_QUIRK(0x17aa, 0x3821, "Lenovo 0x3821", SOC_SDW_SIDECAR_AMPS), + SND_PCI_QUIRK(0x17aa, 0x383c, "Lenovo 0x383c", SOC_SDW_SIDECAR_AMPS), {} }; From 4c9854ad3cdc7ce408453207f153bb910cc6f3c1 Mon Sep 17 00:00:00 2001 From: Tianze Shao Date: Wed, 1 Jul 2026 16:56:06 +0200 Subject: [PATCH 02/23] ASoC: rt712-sdca: reset codec at io_init to fix silent headphone On ThinkPad X1 Carbon Gen 13 (Lunar Lake, RT712-SDCA version VA) the 3.5mm headphone jack is silent after rebooting from Windows. rt712_sdca_va_io_init() gates rt712_sdca_calibration() on the persisted vendor SW_CONFIG1 flag, and io_init writes SW_CONFIG1=1 at the end regardless of whether the calibration succeeded. Across a warm reboot the codec keeps power, so SW_CONFIG1 stays unchanged, the calibration may be skipped, and the retained state can be invalid, leaving the headphone amp disabled. This patch mimics the reset sequence in rt711-sdca.c, it adds an rt712_sdca_reset() helper, and calls it from io_init so the codec is reset before initialization. RT712_PARA_VERB_CTL, RT712_HIDDEN_REG_SW_RESET and RT712_HDA_LEGACY_RESET_CTL are already defined but were unused. The reset clears SW_CONFIG1 and the analog state so rt712_sdca_calibration() runs from a clean state and completes. Problem reproducible: boot Windows (headphone is good) -> reboot to Linux (silent). The reproducibility may depend on Windows' behaviour. Signed-off-by: Tianze Shao Link: https://patch.msgid.link/TYCPR01MB872223CB999D2099A5E0F58DC4F62@TYCPR01MB8722.jpnprd01.prod.outlook.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt712-sdca.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/rt712-sdca.c b/sound/soc/codecs/rt712-sdca.c index 4796fce084ff..d6353af07380 100644 --- a/sound/soc/codecs/rt712-sdca.c +++ b/sound/soc/codecs/rt712-sdca.c @@ -1843,6 +1843,15 @@ static void rt712_sdca_vb_io_init(struct rt712_sdca_priv *rt712) } } +static void rt712_sdca_reset(struct rt712_sdca_priv *rt712) +{ + rt712_sdca_index_update_bits(rt712, RT712_VENDOR_REG, + RT712_PARA_VERB_CTL, RT712_HIDDEN_REG_SW_RESET, + RT712_HIDDEN_REG_SW_RESET); + rt712_sdca_index_update_bits(rt712, RT712_VENDOR_HDA_CTL, + RT712_HDA_LEGACY_RESET_CTL, 0x1, 0x1); +} + int rt712_sdca_io_init(struct device *dev, struct sdw_slave *slave) { struct rt712_sdca_priv *rt712 = dev_get_drvdata(dev); @@ -1870,6 +1879,8 @@ int rt712_sdca_io_init(struct device *dev, struct sdw_slave *slave) pm_runtime_get_noresume(&slave->dev); + rt712_sdca_reset(rt712); + rt712_sdca_index_read(rt712, RT712_VENDOR_REG, RT712_JD_PRODUCT_NUM, &val); rt712->hw_id = (val & 0xf000) >> 12; rt712->version_id = (val & 0x0f00) >> 8; From dbbb5bc5176e36b13aa22e2174ab4779c5ae1dca Mon Sep 17 00:00:00 2001 From: Syed Saba Kareem Date: Fri, 3 Jul 2026 18:02:47 +0530 Subject: [PATCH 03/23] ASoC: amd: acp: Fix linker error with SDCA quirks Fix undefined reference to `snd_soc_acpi_amd_sdca_is_device_rt712_vb` linker error when CONFIG_SND_SOC_ACPI_AMD_MATCH=y and CONFIG_SND_SOC_ACPI_AMD_SDCA_QUIRKS=m, which causes built-in code to reference a symbol only available in a module. Fix this by changing SND_SOC_ACPI_AMD_SDCA_QUIRKS from tristate to bool and compiling the quirks code directly into snd-soc-acpi-amd-match rather than as a separate module. This ensures the quirks symbols are always available at link time when the match tables reference them. Fixes: 10d366a846be ("ASoC: amd: acp: Fix Kconfig dependencies for SND_SOC_ACPI_AMD_SDCA_QUIRKS") Reported-by: Arnd Bergmann Tested-by: Arnd Bergmann Signed-off-by: Syed Saba Kareem Reviewed-by: Vijendar Mukunda Link: https://patch.msgid.link/20260703123314.147977-1-syed.sabakareem@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/Kconfig | 10 +++++++--- sound/soc/amd/acp/Makefile | 3 +-- sound/soc/amd/acp/amd-acp70-acpi-match.c | 2 +- sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c | 4 ---- 4 files changed, 9 insertions(+), 10 deletions(-) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 977e4f2a7a70..a04c301df4d6 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -15,13 +15,17 @@ config SND_SOC_AMD_ACP_COMMON config SND_SOC_ACPI_AMD_MATCH tristate - select SND_SOC_ACPI_AMD_SDCA_QUIRKS if SND_SOC_SDCA select SND_SOC_ACPI if ACPI config SND_SOC_ACPI_AMD_SDCA_QUIRKS - tristate - depends on ACPI + bool "AMD ACPI SDCA quirks" + depends on SND_SOC_ACPI_AMD_MATCH depends on SND_SOC_SDCA + depends on SND_SOC_ACPI_AMD_MATCH = m || SND_SOC_SDCA = y + default y + help + Enable SDCA quirk support for AMD ACPI match tables. + This is compiled into the snd-soc-acpi-amd-match module. if SND_SOC_AMD_ACP_COMMON diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile index 81d23aded348..ab5f9dc871a6 100644 --- a/sound/soc/amd/acp/Makefile +++ b/sound/soc/amd/acp/Makefile @@ -24,10 +24,10 @@ snd-acp-mach-y := acp-mach-common.o snd-acp-legacy-mach-y := acp-legacy-mach.o acp3x-es83xx/acp3x-es83xx.o snd-acp-sof-mach-y := acp-sof-mach.o snd-soc-acpi-amd-match-y := amd-acp63-acpi-match.o amd-acp70-acpi-match.o +snd-soc-acpi-amd-match-$(CONFIG_SND_SOC_ACPI_AMD_SDCA_QUIRKS) += soc-acpi-amd-sdca-quirks.o snd-acp-sdw-mach-y := acp-sdw-mach-common.o snd-acp-sdw-sof-mach-y += acp-sdw-sof-mach.o snd-acp-sdw-legacy-mach-y += acp-sdw-legacy-mach.o -snd-soc-acpi-amd-sdca-quirks-y += soc-acpi-amd-sdca-quirks.o obj-$(CONFIG_SND_SOC_AMD_ACP_PCM) += snd-acp-pcm.o obj-$(CONFIG_SND_SOC_AMD_ACP_I2S) += snd-acp-i2s.o @@ -41,7 +41,6 @@ obj-$(CONFIG_SND_AMD_ASOC_REMBRANDT) += snd-acp-rembrandt.o obj-$(CONFIG_SND_AMD_ASOC_ACP63) += snd-acp63.o obj-$(CONFIG_SND_AMD_ASOC_ACP70) += snd-acp70.o -obj-$(CONFIG_SND_SOC_ACPI_AMD_SDCA_QUIRKS) += snd-soc-acpi-amd-sdca-quirks.o obj-$(CONFIG_SND_AMD_SOUNDWIRE_ACPI) += snd-amd-sdw-acpi.o obj-$(CONFIG_SND_SOC_AMD_MACH_COMMON) += snd-acp-mach.o obj-$(CONFIG_SND_SOC_AMD_LEGACY_MACH) += snd-acp-legacy-mach.o diff --git a/sound/soc/amd/acp/amd-acp70-acpi-match.c b/sound/soc/amd/acp/amd-acp70-acpi-match.c index 18f2918d4ada..9e97a805d0df 100644 --- a/sound/soc/amd/acp/amd-acp70-acpi-match.c +++ b/sound/soc/amd/acp/amd-acp70-acpi-match.c @@ -775,5 +775,5 @@ MODULE_DESCRIPTION("AMD ACP7.0 & ACP7.1 tables and support for ACPI enumeration" MODULE_LICENSE("GPL"); MODULE_AUTHOR("Vijendar.Mukunda@amd.com"); #if IS_ENABLED(CONFIG_SND_SOC_ACPI_AMD_SDCA_QUIRKS) -MODULE_IMPORT_NS("SND_SOC_ACPI_AMD_SDCA_QUIRKS"); +MODULE_IMPORT_NS("SND_SOC_SDCA"); #endif diff --git a/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c b/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c index 63bf9e3c0ae1..9248b6d09e76 100644 --- a/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c +++ b/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c @@ -35,8 +35,4 @@ bool snd_soc_acpi_amd_sdca_is_device_rt712_vb(void *arg) return false; } -EXPORT_SYMBOL_NS(snd_soc_acpi_amd_sdca_is_device_rt712_vb, "SND_SOC_ACPI_AMD_SDCA_QUIRKS"); -MODULE_DESCRIPTION("ASoC ACPI AMD SDCA quirks"); -MODULE_LICENSE("GPL"); -MODULE_IMPORT_NS("SND_SOC_SDCA"); From 6b59c53c8adc2b522327407af5e1793a65b67e4b Mon Sep 17 00:00:00 2001 From: Christian Hewitt Date: Sat, 27 Jun 2026 13:12:05 +0000 Subject: [PATCH 04/23] ASoC: meson: aiu: fifo-spdif: soft reset the S/PDIF datapath on start/stop The I2S FIFO soft-resets its fast domain on start (AIU_RST_SOFT bit 0 + AIU_I2S_SYNC read in aiu_fifo_i2s_trigger), mirroring the downstream vendor driver's audio_out_i2s_enable(). The S/PDIF FIFO has no equivalent: it only toggles the IEC958 DCU, so a stale datapath FIFO can be replayed, producing the "machine gun noise" buffer underrun - on start when switching outputs, and on stop when playback ends. The latter is audible on devices with an always-on S/PDIF-fed DAC (e.g. the ES7144 on the WeTek Play2). The vendor driver resets the IEC958 fast domain (AIU_RST_SOFT bit 2) on both enable and disable (audio_hw_958_enable), and when reconfiguring (audio_hw_958_reset clears AIU_958_DCU_FF_CTRL then resets). Do the same: reset before enabling the DCU on start, and before disabling on stop. Fixes: 6ae9ca9ce986bf ("ASoC: meson: aiu: add i2s and spdif support") Signed-off-by: Christian Hewitt Reviewed-by: Martin Blumenstingl Link: https://patch.msgid.link/20260627131205.808800-1-christianshewitt@gmail.com Signed-off-by: Mark Brown --- sound/soc/meson/aiu-fifo-spdif.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/meson/aiu-fifo-spdif.c b/sound/soc/meson/aiu-fifo-spdif.c index e0e00ec026dc..a9861c5d6637 100644 --- a/sound/soc/meson/aiu-fifo-spdif.c +++ b/sound/soc/meson/aiu-fifo-spdif.c @@ -24,6 +24,7 @@ #define AIU_MEM_IEC958_CONTROL_MODE_16BIT BIT(7) #define AIU_MEM_IEC958_CONTROL_MODE_LINEAR BIT(8) #define AIU_MEM_IEC958_BUF_CNTL_INIT BIT(0) +#define AIU_RST_SOFT_958_FAST BIT(2) #define AIU_FIFO_SPDIF_BLOCK 8 @@ -68,11 +69,15 @@ static int fifo_spdif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + snd_soc_component_write(component, AIU_RST_SOFT, + AIU_RST_SOFT_958_FAST); fifo_spdif_dcu_enable(component, true); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_STOP: + snd_soc_component_write(component, AIU_RST_SOFT, + AIU_RST_SOFT_958_FAST); fifo_spdif_dcu_enable(component, false); break; default: From dedf4ccbcbd249e2435344d137247d94c0aa10ee Mon Sep 17 00:00:00 2001 From: Baojun Xu Date: Thu, 2 Jul 2026 18:44:29 +0800 Subject: [PATCH 05/23] ASoC: amd: acp: amd-acp70-acpi-match: Add tas2783 support Add driver data and ACPI match table entries for the TAS2783 on ACP7.0 and ACP7.1 platforms using SoundWire link 0. Signed-off-by: Baojun Xu Link: https://patch.msgid.link/20260702104429.1157-1-baojun.xu@ti.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/amd-acp70-acpi-match.c | 45 ++++++++++++++++++++++++ 1 file changed, 45 insertions(+) diff --git a/sound/soc/amd/acp/amd-acp70-acpi-match.c b/sound/soc/amd/acp/amd-acp70-acpi-match.c index 9e97a805d0df..ccd01152c87d 100644 --- a/sound/soc/amd/acp/amd-acp70-acpi-match.c +++ b/sound/soc/amd/acp/amd-acp70-acpi-match.c @@ -553,6 +553,46 @@ static const struct snd_soc_acpi_link_adr acp70_rt722_l0_rt1320_l1[] = { {} }; +static const struct snd_soc_acpi_adr_device tas2783_2_adr[] = { + { + /* left */ + .adr = 0x00003c0102000001ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "tas2783-1", + }, + { + /* right */ + .adr = 0x00003d0102000001ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "tas2783-2", + }, + { + /* left */ + .adr = 0x0000390102000001ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "tas2783-3", + }, + { + /* right */ + .adr = 0x00003a0102000001ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "tas2783-4", + }, +}; + +static const struct snd_soc_acpi_link_adr acp70_tas2783_2[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(tas2783_2_adr), + .adr_d = tas2783_2_adr, + }, + {} +}; + static const struct snd_soc_acpi_adr_device rt1320_0_single_adr[] = { { .adr = 0x000030025D132001ull, @@ -659,6 +699,11 @@ static const struct snd_soc_acpi_link_adr acp70_rt721_only[] = { }; struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_sdw_machines[] = { + { + .link_mask = BIT(0), + .links = acp70_tas2783_2, + .drv_name = "amd_sdw", + }, { .link_mask = BIT(0) | BIT(1), .links = acp70_rt1320_l0_rt722_l1, From 5893013efabb056399a01e267f410cf76eba25eb Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Tue, 7 Jul 2026 11:29:36 +0530 Subject: [PATCH 06/23] ASoC: amd: ps: disable MSI on resume in ACP PCI driver BIOS/firmware may re-enable MSI in PCI config space during system level resume even though this driver only uses legacy INTx interrupts. If MSI is left enabled with stale address/data registers, the device will write interrupts to a bogus address causing IOMMU IO_PAGE_FAULT and interrupt delivery failure. Clear the MSI Enable bit before reinitializing the ACP hardware on system level resume. Fixes: 491628388005 ("ASoC: amd: ps: add callback functions for acp pci driver pm ops") Signed-off-by: Vijendar Mukunda Reviewed-by: Mario Limonciello (AMD) Link: https://patch.msgid.link/20260707060130.2514138-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/ps/pci-ps.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 4ecda224157b..635832da45f9 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -693,8 +693,37 @@ static int snd_acp_runtime_resume(struct device *dev) return acp_hw_runtime_resume(dev); } +static void acp_disable_msi_on_resume(struct pci_dev *pdev) +{ + u16 control; + + if (!pdev->msi_cap) + return; + + pci_read_config_word(pdev, pdev->msi_cap + PCI_MSI_FLAGS, &control); + if (control & PCI_MSI_FLAGS_ENABLE) { + dev_warn(&pdev->dev, + "ACP: MSI unexpectedly enabled after resume (flags=0x%04x), disabling\n", + control); + control &= ~PCI_MSI_FLAGS_ENABLE; + pci_write_config_word(pdev, pdev->msi_cap + PCI_MSI_FLAGS, control); + } +} + static int snd_acp_resume(struct device *dev) { + struct pci_dev *pdev = to_pci_dev(dev); + + /* + * BIOS/firmware may re-enable MSI in PCI config space during + * system resume even though this driver only uses legacy INTx + * interrupts. If MSI is left enabled with stale address/data + * registers, the device will write interrupts to a bogus address + * causing IOMMU IO_PAGE_FAULT and interrupt delivery failure. + * Explicitly clear the MSI Enable bit before reinitializing + * the ACP hardware. + */ + acp_disable_msi_on_resume(pdev); return acp_hw_resume(dev); } From f7697ecf6eab9d4887dd731038b3dc405c7e755e Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Tue, 7 Jul 2026 11:29:37 +0530 Subject: [PATCH 07/23] ASoC: amd: ps: fix wrong ACP version string in pci_request_regions() The driver handles ACP6.3/7.0/7.1/7.2 platforms but the region was claimed with the stale name "AMD ACP6.2 audio" left over from the original ACP6.2 driver. Correct it to "AMD ACP6.3 audio". Fixes: 95e43a170bb1 ("ASoC: amd: add Pink Sardine ACP PCI driver") Signed-off-by: Vijendar Mukunda Reviewed-by: Mario Limonciello (AMD) Link: https://patch.msgid.link/20260707060130.2514138-3-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/ps/pci-ps.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 635832da45f9..1162d13d8505 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -602,7 +602,7 @@ static int snd_acp63_probe(struct pci_dev *pci, return -ENODEV; } - ret = pci_request_regions(pci, "AMD ACP6.2 audio"); + ret = pci_request_regions(pci, "AMD ACP6.3 audio"); if (ret < 0) { dev_err(&pci->dev, "pci_request_regions failed\n"); goto disable_pci; From dec5aaa27603e1d7b426ce3504af6d1a62e4d444 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Tue, 7 Jul 2026 11:29:38 +0530 Subject: [PATCH 08/23] ASoC: amd: ps: replace bitwise OR with logical OR in IRQ return check The condition 'irq_flag | wake_irq_flag' uses bitwise OR to combine two integer flags that are used as booleans. Replace with logical OR '||' to correctly express the intended boolean check. Signed-off-by: Vijendar Mukunda Reviewed-by: Mario Limonciello (AMD) Fixes: 7f91f012c1df0 ("ASoC: amd: ps: fix for irq handler return status") Link: https://patch.msgid.link/20260707060130.2514138-4-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/ps/pci-ps.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 1162d13d8505..729f9aaba69e 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -248,7 +248,7 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) if (sdw_dma_irq_flag) return IRQ_WAKE_THREAD; - if (irq_flag | wake_irq_flag) + if (irq_flag || wake_irq_flag) return IRQ_HANDLED; else return IRQ_NONE; From bc0d4444f384d183e2113b72c34643069f93ab63 Mon Sep 17 00:00:00 2001 From: Ravi Hothi Date: Fri, 3 Jul 2026 13:04:32 +0530 Subject: [PATCH 09/23] ASoC: dt-bindings: qcom: Add Eliza LPASS macro codecs Add bindings for Qualcomm Eliza LPASS RX, TX, VA and WSA macro codecs, which are compatible with the existing SM8550 generation. Signed-off-by: Ravi Hothi Link: https://patch.msgid.link/20260703073434.2589657-2-ravi.hothi@oss.qualcomm.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml | 1 + Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml | 1 + Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml | 1 + .../devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml | 1 + 4 files changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml index 2eed2277511f..4988e7ed6e34 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml @@ -21,6 +21,7 @@ properties: - qcom,sc8280xp-lpass-rx-macro - items: - enum: + - qcom,eliza-lpass-rx-macro - qcom,kaanapali-lpass-rx-macro - qcom,sm8650-lpass-rx-macro - qcom,sm8750-lpass-rx-macro diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml index e5e65e226a02..d8682ff2e82c 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml @@ -21,6 +21,7 @@ properties: - qcom,sc8280xp-lpass-tx-macro - items: - enum: + - qcom,eliza-lpass-tx-macro - qcom,kaanapali-lpass-tx-macro - qcom,sm8650-lpass-tx-macro - qcom,sm8750-lpass-tx-macro diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml index 5c42b2b323ee..aea31fbdad37 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml @@ -21,6 +21,7 @@ properties: - qcom,sc8280xp-lpass-va-macro - items: - enum: + - qcom,eliza-lpass-va-macro - qcom,glymur-lpass-va-macro - qcom,kaanapali-lpass-va-macro - qcom,sm8650-lpass-va-macro diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml index d5f22b5cf021..9fedd80532e2 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml @@ -20,6 +20,7 @@ properties: - qcom,sc8280xp-lpass-wsa-macro - items: - enum: + - qcom,eliza-lpass-wsa-macro - qcom,glymur-lpass-wsa-macro - qcom,kaanapali-lpass-wsa-macro - qcom,sm8650-lpass-wsa-macro From 4214220c314435bc1b8ca492b7657ae5ad52145e Mon Sep 17 00:00:00 2001 From: Ravi Hothi Date: Fri, 3 Jul 2026 13:04:33 +0530 Subject: [PATCH 10/23] ASoC: dt-bindings: qcom,sm8250: Add Eliza sound card Add bindings for Eliza sound card, which is compatible with the existing SM8450. Signed-off-by: Ravi Hothi Acked-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20260703073434.2589657-3-ravi.hothi@oss.qualcomm.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index 15f38622b98b..dae440ecab59 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -23,6 +23,7 @@ properties: - const: qcom,sdm845-sndcard - items: - enum: + - qcom,eliza-sndcard - qcom,kaanapali-sndcard - qcom,sm8550-sndcard - qcom,sm8650-sndcard From 0d037985725a2123b9d92dbf98ccb5f81e2bb8b4 Mon Sep 17 00:00:00 2001 From: Ravi Hothi Date: Fri, 3 Jul 2026 13:04:34 +0530 Subject: [PATCH 11/23] ASoC: qcom: sc8280xp: Add support for Eliza Add compatible for sound card on Qualcomm Eliza boards. Signed-off-by: Ravi Hothi Link: https://patch.msgid.link/20260703073434.2589657-4-ravi.hothi@oss.qualcomm.com Signed-off-by: Mark Brown --- sound/soc/qcom/sc8280xp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index 7925aa3f63ba..98b15a527e37 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -164,6 +164,7 @@ static int sc8280xp_platform_probe(struct platform_device *pdev) } static const struct of_device_id snd_sc8280xp_dt_match[] = { + {.compatible = "qcom,eliza-sndcard", "eliza"}, {.compatible = "qcom,kaanapali-sndcard", "kaanapali"}, {.compatible = "qcom,qcm6490-idp-sndcard", "qcm6490"}, {.compatible = "qcom,qcs615-sndcard", "qcs615"}, From f74e6e15485b68b92b2807071e822db6309b7e38 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 8 Jul 2026 11:34:30 +0100 Subject: [PATCH 12/23] ASoC: cs42l43: Correct report for forced microphone jack Currently if the jack is forced to the microphone mode, it will report as line in. Correct the report to microphone. Fixes: fc918cbe874e ("ASoC: cs42l43: Add support for the cs42l43") Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20260708103430.1395207-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43-jack.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c index 934666295ee3..dbf45cabfffe 100644 --- a/sound/soc/codecs/cs42l43-jack.c +++ b/sound/soc/codecs/cs42l43-jack.c @@ -310,6 +310,7 @@ irqreturn_t cs42l43_bias_detect_clamp(int irq, void *data) #define CS42L43_JACK_ABSENT 0x0 #define CS42L43_JACK_OPTICAL (SND_JACK_MECHANICAL | SND_JACK_AVOUT) +#define CS42L43_JACK_MICROPHONE (SND_JACK_MECHANICAL | SND_JACK_MICROPHONE) #define CS42L43_JACK_HEADPHONE (SND_JACK_MECHANICAL | SND_JACK_HEADPHONE) #define CS42L43_JACK_HEADSET (SND_JACK_MECHANICAL | SND_JACK_HEADSET) #define CS42L43_JACK_LINEOUT (SND_JACK_MECHANICAL | SND_JACK_LINEOUT) @@ -871,7 +872,7 @@ static const struct cs42l43_jack_override_mode { .hsdet_mode = CS42L43_JACK_3_POLE_SWITCHES, .mic_ctrl = (0x3 << CS42L43_JACK_STEREO_CONFIG_SHIFT) | CS42L43_HS1_BIAS_EN_MASK | CS42L43_HS2_BIAS_EN_MASK, - .report = CS42L43_JACK_LINEIN, + .report = CS42L43_JACK_MICROPHONE, }, [CS42L43_JACK_RAW_OPTICAL] = { .hsdet_mode = CS42L43_JACK_3_POLE_SWITCHES, From 3238c634725afbb2a137fdda762208510828f71d Mon Sep 17 00:00:00 2001 From: Uday Khare Date: Mon, 6 Jul 2026 21:01:09 +0530 Subject: [PATCH 13/23] ASoC: tas2562: fix deprecated 'shut-down' GPIO always cleared after lookup In tas2562_parse_dt(), the fallback lookup for the deprecated "shut-down" GPIO property is broken due to a missing pair of braces. The code intends to reset sdz_gpio to NULL only when the lookup returns an error that is not -EPROBE_DEFER (so the driver gracefully continues without a GPIO). However, without braces the statement: tas2562->sdz_gpio = NULL; falls outside the IS_ERR() check and is executed unconditionally for every path through the if block, including a successful GPIO lookup. This means any device using the deprecated 'shut-down' DT property will always have sdz_gpio == NULL after probe, making the GPIO completely non-functional. Fix this by adding the missing braces to scope the NULL assignment inside the IS_ERR() branch, matching the pattern already used for the primary 'shutdown' GPIO lookup above. Fixes: f78a97003b8b ("ASoC: tas2562: Update shutdown GPIO property") Signed-off-by: Uday Khare Link: https://patch.msgid.link/20260706153109.10953-1-udaykhare77@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2562.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 2f7cfc2be970..e1d62f30418a 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -675,11 +675,12 @@ static int tas2562_parse_dt(struct tas2562_data *tas2562) if (tas2562->sdz_gpio == NULL) { tas2562->sdz_gpio = devm_gpiod_get_optional(dev, "shut-down", GPIOD_OUT_HIGH); - if (IS_ERR(tas2562->sdz_gpio)) + if (IS_ERR(tas2562->sdz_gpio)) { if (PTR_ERR(tas2562->sdz_gpio) == -EPROBE_DEFER) return -EPROBE_DEFER; - tas2562->sdz_gpio = NULL; + tas2562->sdz_gpio = NULL; + } } if (tas2562->model_id == TAS2110) From d091132889c1378dd0944a72f86eae3e4da1e4fa Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Fri, 10 Jul 2026 16:08:35 +0900 Subject: [PATCH 14/23] ASoC: fsl_sai: Fix spurious BCLK on resume by clearing BYP When the BCLK divider ratio is 1:1, fsl_sai_set_bclk() enables bypass mode by setting BYP, but never clears the bit. The BYP=1 value remains in the regcache, and is restored by regcache_sync() on the next runtime resume. Since BYP=1 combined with BCD=1 immediately outputs the ungated MCLK as BCLK without waiting for BCE/TE/RE to be enabled, the clock is driven prematurely before the stream is fully configured, causing noise on some codecs. Fix this by clearing BYP and BCI in fsl_sai_hw_free() taking into account sync mode and the opposite stream's state, so that the regcache holds BYP=0 before runtime suspend and regcache_sync() on resume will not restore bypass mode prematurely. Fixes: a50b7926d015 ("ASoC: fsl_sai: implement 1:1 bclk:mclk ratio support") Cc: stable@vger.kernel.org Signed-off-by: Chancel Liu Reviewed-by: Shengjiu Wang Link: https://patch.msgid.link/20260710070835.3749817-1-chancel.liu@oss.nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 29 +++++++++++++++++++++++++---- 1 file changed, 25 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 9661602b53c5..d232c8f53061 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -808,6 +808,8 @@ static int fsl_sai_hw_free(struct snd_pcm_substream *substream, struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; unsigned int ofs = sai->soc_data->reg_offset; + int adir = tx ? RX : TX; + int dir = tx ? TX : RX; /* Clear xMR to avoid channel swap with mclk_with_tere enabled case */ regmap_write(sai->regmap, FSL_SAI_xMR(tx), 0); @@ -815,10 +817,29 @@ static int fsl_sai_hw_free(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), FSL_SAI_CR3_TRCE_MASK, 0); - if (!sai->is_consumer_mode[tx] && - sai->mclk_streams & BIT(substream->stream)) { - clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[tx]]); - sai->mclk_streams &= ~BIT(substream->stream); + if (!sai->is_consumer_mode[tx]) { + bool adir_active = !!(sai->mclk_streams & BIT(!substream->stream)); + /* + * If opposite stream provides clocks for synchronous mode and + * it is inactive, Clear BYP and BCI + */ + if (fsl_sai_dir_is_synced(sai, adir) && !adir_active) + regmap_update_bits(sai->regmap, FSL_SAI_xCR2(!tx, ofs), + FSL_SAI_CR2_BCI | FSL_SAI_CR2_BYP, 0); + /* + * Clear BYP and BCI of current stream if either of: + * 1. current stream doesn't provide clocks for synchronous mode + * 2. current stream provides clocks for synchronous mode but no + * more stream is active. + */ + if (!fsl_sai_dir_is_synced(sai, dir) || !adir_active) + regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx, ofs), + FSL_SAI_CR2_BCI | FSL_SAI_CR2_BYP, 0); + + if (sai->mclk_streams & BIT(substream->stream)) { + clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[tx]]); + sai->mclk_streams &= ~BIT(substream->stream); + } } return 0; From 27d090f3ccd4cade897fb7e0c8c0a6e2fd3a95a0 Mon Sep 17 00:00:00 2001 From: Syed Saba Kareem Date: Fri, 10 Jul 2026 15:59:19 +0530 Subject: [PATCH 15/23] ASoC: amd: acp: add ACP70 DMI override for new ASUS TUF platforms Some ASUS TUF ACP70-based systems expose ACP ACPI configuration flags that select a non-working fallback audio path, similar to previously affected ASUS platforms. Add DMI-based overrides in snd_amd_acp_find_config() for the following systems to skip ACP ACPI flag-based selection: - ASUS TUF Gaming Vivobook 18 - ASUS TUF Gaming A14 FA401EA This ensures the intended SoundWire-based machine driver is selected on these platforms. Signed-off-by: Syed Saba Kareem Link: https://patch.msgid.link/20260710102926.1633385-1-syed.sabakareem@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-config.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c index 0d977f4f758d..051946562725 100644 --- a/sound/soc/amd/acp-config.c +++ b/sound/soc/amd/acp-config.c @@ -44,6 +44,18 @@ static const struct dmi_system_id acp70_acpi_flag_override_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "83W5"), }, }, + { + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC"), + DMI_MATCH(DMI_PRODUCT_NAME, "TUF Gaming A14 FA401EA"), + }, + }, + { + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC"), + DMI_MATCH(DMI_PRODUCT_NAME, "Vivobook 18 M1807GA"), + }, + }, {} }; From bc889dfcea9294a1eae7f8e2f3573a90764ae4d0 Mon Sep 17 00:00:00 2001 From: Pengpeng Hou Date: Mon, 6 Jul 2026 22:45:39 +0800 Subject: [PATCH 16/23] ASoC: tas2781: bound firmware description string parsing The TAS2781 firmware parser reads several variable-length description strings with strlen() before checking that the string terminator is present inside the firmware blob. A malformed firmware image without a NUL terminator can therefore make the parser walk past the end of the firmware buffer before the later size checks run. Add a small bounded string-length helper and use it for all description fields that are parsed from the firmware buffer. Keep the existing size checks for the fixed bytes that follow each string. Fixes: 915f5eadebd2 ("ASoC: tas2781: firmware lib") Signed-off-by: Pengpeng Hou Link: https://patch.msgid.link/20260706144540.93929-1-pengpeng@iscas.ac.cn Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-fmwlib.c | 63 ++++++++++++++++++++++++++++--- 1 file changed, 57 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index bd16d5326a23..dcbeb9618195 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -12,6 +12,7 @@ #include #include #include +#include #include #include #include @@ -1100,13 +1101,42 @@ static int tasdevice_load_block_kernel( return 0; } +static int tasdevice_fw_strnlen(const struct firmware *fmw, int offset) +{ + const u8 *start; + const u8 *nul; + size_t remaining; + size_t len; + + if (offset < 0 || offset >= fmw->size) + return -EINVAL; + + start = fmw->data + offset; + remaining = fmw->size - offset; + nul = memchr(start, '\0', remaining); + if (!nul) + return -EINVAL; + + len = nul - start; + if (len > INT_MAX) + return -EOVERFLOW; + + return len; +} + static int fw_parse_variable_hdr(struct tasdevice_priv *tas_priv, struct tasdevice_dspfw_hdr *fw_hdr, const struct firmware *fmw, int offset) { const unsigned char *buf = fmw->data; - int len = strlen((char *)&buf[offset]); + int len; + len = tasdevice_fw_strnlen(fmw, offset); + if (len < 0) { + dev_err(tas_priv->dev, "%s: Description error\n", __func__); + offset = len; + goto out; + } len++; if (offset + len + 8 > fmw->size) { @@ -1238,7 +1268,12 @@ static int fw_parse_data(struct tasdevice_fw *tas_fmw, memcpy(img_data->name, &data[offset], 64); offset += 64; - n = strlen((char *)&data[offset]); + n = tasdevice_fw_strnlen(fmw, offset); + if (n < 0) { + dev_err(tas_fmw->dev, "%s: Description error\n", __func__); + offset = n; + goto out; + } n++; if (offset + n + 2 > fmw->size) { dev_err(tas_fmw->dev, "%s: Description error\n", __func__); @@ -1309,7 +1344,12 @@ static int fw_parse_program_data(struct tasdevice_priv *tas_priv, } offset += 64; - n = strlen((char *)&buf[offset]); + n = tasdevice_fw_strnlen(fmw, offset); + if (n < 0) { + dev_err(tas_priv->dev, "Description err\n"); + offset = n; + goto out; + } /* skip '\0' and 5 unused bytes */ n += 6; if (offset + n > fmw->size) { @@ -1372,7 +1412,12 @@ static int fw_parse_configuration_data( memcpy(config->name, &data[offset], 64); offset += 64; - n = strlen((char *)&data[offset]); + n = tasdevice_fw_strnlen(fmw, offset); + if (n < 0) { + dev_err(tas_priv->dev, "Description err\n"); + offset = n; + goto out; + } n += 15; if (offset + n > fmw->size) { dev_err(tas_priv->dev, "Description err\n"); @@ -2134,7 +2179,8 @@ static int fw_parse_calibration_data(struct tasdevice_priv *tas_priv, { struct tasdevice_calibration *calibration; unsigned char *data = (unsigned char *)fmw->data; - unsigned int i, n; + unsigned int i; + int n; if (offset + 2 > fmw->size) { dev_err(tas_priv->dev, "%s: Calibrations error\n", __func__); @@ -2166,7 +2212,12 @@ static int fw_parse_calibration_data(struct tasdevice_priv *tas_priv, calibration = &(tas_fmw->calibrations[i]); offset += 64; - n = strlen((char *)&data[offset]); + n = tasdevice_fw_strnlen(fmw, offset); + if (n < 0) { + dev_err(tas_priv->dev, "Description err\n"); + offset = n; + goto out; + } /* skip '\0' and 2 unused bytes */ n += 3; if (offset + n > fmw->size) { From d9e96f859de3ea3e99bce927a988449a1816483c Mon Sep 17 00:00:00 2001 From: Chen-Yu Tsai Date: Tue, 14 Jul 2026 19:33:03 +0800 Subject: [PATCH 17/23] ASoC: sun4i-codec: Set quirks.playback_only for H616 codec The H616 codec does not have capture capabilities. Set the .playback_only quirks flag to denote this. This was somehow missing from the original driver patch, even though the patch prior to it in the series added this quirk. Fixes: 9155c321a1d0 ("ASoC: sun4i-codec: support allwinner H616 codec") Signed-off-by: Chen-Yu Tsai Link: https://patch.msgid.link/20260714113304.270224-1-wens@kernel.org Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index f4e22af594fa..93f511fb6190 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -2237,6 +2237,7 @@ static const struct sun4i_codec_quirks sun50i_h616_codec_quirks = { .reg_dac_fifoc = REG_FIELD(SUN50I_H616_CODEC_DAC_FIFOC, 0, 31), .reg_dac_txdata = SUN8I_H3_CODEC_DAC_TXDATA, .has_reset = true, + .playback_only = true, .dma_max_burst = SUN4I_DMA_MAX_BURST, }; From 9f86aea992568c2b4db78c80ff9508af9e050ff7 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Fri, 10 Jul 2026 12:13:33 +0900 Subject: [PATCH 18/23] ASoC: fsl: imx-card: Skip sysclk reset for active DAIs in shutdown In a full-duplex setup, when one direction (playback or capture) is closed while the other is still running, imx_aif_shutdown() was unconditionally calling snd_soc_dai_set_sysclk() with rate=0 for all cpu/codec DAIs, which would disable the clock still needed by the active stream. Add snd_soc_dai_active() checks before clearing sysclk so that only truly inactive DAIs have their clocks reset. Fixes: 2260bc6ea8bd ("ASoC: imx-card: Add WM8524 support") Cc: stable@vger.kernel.org Signed-off-by: Chancel Liu Link: https://patch.msgid.link/20260710031333.3491445-1-chancel.liu@oss.nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-card.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index a4518fefad69..43438af1e1c6 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -496,11 +496,15 @@ static void imx_aif_shutdown(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai; int i; - for_each_rtd_cpu_dais(rtd, i, cpu_dai) - snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_OUT); + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + if (!snd_soc_dai_active(cpu_dai)) + snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_OUT); + } - for_each_rtd_codec_dais(rtd, i, codec_dai) - snd_soc_dai_set_sysclk(codec_dai, 0, 0, SND_SOC_CLOCK_IN); + for_each_rtd_codec_dais(rtd, i, codec_dai) { + if (!snd_soc_dai_active(codec_dai)) + snd_soc_dai_set_sysclk(codec_dai, 0, 0, SND_SOC_CLOCK_IN); + } } static const struct snd_soc_ops imx_aif_ops = { From d39e4e27962aa1fb5b8f0df057e1079504d369eb Mon Sep 17 00:00:00 2001 From: Antonio Ignacio Campos Ruiz Date: Mon, 13 Jul 2026 18:57:09 +0200 Subject: [PATCH 19/23] ASoC: amd: yc: Add DMI quirk for MSI Vector A16 HX A8WIG The internal digital microphone on the MSI Vector A16 HX A8WIG is not detected: the ACP platform devices are created, but snd_soc_acp6x_mach never binds because the machine is missing from the DMI quirk table, so no capture device shows up at all. This is the same board as the already supported "Vector A16 HX A8WHG", differing only in the trailing model code. Add the corresponding entry. Signed-off-by: Antonio Ignacio Campos Ruiz Link: https://patch.msgid.link/20260713165709.19489-1-acamposruiz@gmail.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index d6df7de70b27..af1dce9c9b8a 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -521,6 +521,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Vector A16 HX A8WHG"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Micro-Star International Co., Ltd."), + DMI_MATCH(DMI_PRODUCT_NAME, "Vector A16 HX A8WIG"), + } + }, { .driver_data = &acp6x_card, .matches = { From 70d0d7b24d8606effdcbc0b0dec6955759ed36c7 Mon Sep 17 00:00:00 2001 From: bui duc phuc Date: Wed, 15 Jul 2026 12:11:12 +0700 Subject: [PATCH 20/23] ASoC: tegra: tegra20_das: Use dev_err_probe() for error handling Replace the existing dev_err() and PTR_ERR() sequence with dev_err_probe(), preserving the original error code while simplifying the error handling. Signed-off-by: bui duc phuc Link: https://patch.msgid.link/20260715051115.17385-2-phucduc.bui@gmail.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_das.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index b48cc4a6967b..ea8bc9bcfa67 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -167,10 +167,9 @@ static int tegra20_das_probe(struct platform_device *pdev) das->regmap = devm_regmap_init_mmio(&pdev->dev, regs, &tegra20_das_regmap_config); - if (IS_ERR(das->regmap)) { - dev_err(&pdev->dev, "regmap init failed\n"); - return PTR_ERR(das->regmap); - } + if (IS_ERR(das->regmap)) + return dev_err_probe(&pdev->dev, PTR_ERR(das->regmap), + "regmap init failed\n"); tegra20_das_connect_dap_to_dac(das, TEGRA20_DAS_DAP_ID_1, TEGRA20_DAS_DAP_SEL_DAC1); From b82384bffb27467e2e3966831d83eff3a6e1fbbb Mon Sep 17 00:00:00 2001 From: bui duc phuc Date: Wed, 15 Jul 2026 12:11:13 +0700 Subject: [PATCH 21/23] ASoC: tegra: tegra210_adx: Return the original error directly snd_soc_add_component_controls() already reports failures internally. Return its error code directly instead of logging the error again in the component probe callback. Also remove the now unnecessary local error variable. Signed-off-by: bui duc phuc Reviewed-by: Thierry Reding Link: https://patch.msgid.link/20260715051115.17385-3-phucduc.bui@gmail.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_adx.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/tegra/tegra210_adx.c b/sound/soc/tegra/tegra210_adx.c index 15a94196ee1a..e75bf50f8fb0 100644 --- a/sound/soc/tegra/tegra210_adx.c +++ b/sound/soc/tegra/tegra210_adx.c @@ -496,16 +496,12 @@ static struct snd_kcontrol_new tegra264_adx_controls[] = { static int tegra210_adx_component_probe(struct snd_soc_component *component) { struct tegra210_adx *adx = snd_soc_component_get_drvdata(component); - int err = 0; - if (adx->soc_data->num_controls) { - err = snd_soc_add_component_controls(component, adx->soc_data->controls, - adx->soc_data->num_controls); - if (err) - dev_err(component->dev, "can't add ADX controls, err: %d\n", err); - } + if (adx->soc_data->num_controls) + return snd_soc_add_component_controls(component, adx->soc_data->controls, + adx->soc_data->num_controls); - return err; + return 0; } static const struct snd_soc_component_driver tegra210_adx_cmpnt = { From f7cbc51424cd22098b0e6c09cbe338d7cd6137a7 Mon Sep 17 00:00:00 2001 From: bui duc phuc Date: Wed, 15 Jul 2026 12:11:14 +0700 Subject: [PATCH 22/23] ASoC: tegra: tegra210_amx: Return the original error directly snd_soc_add_component_controls() already reports failures internally. Return its error code directly instead of logging the error again in the component probe callback. Also remove the now unnecessary local error variable. Signed-off-by: bui duc phuc Reviewed-by: Thierry Reding Link: https://patch.msgid.link/20260715051115.17385-4-phucduc.bui@gmail.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_amx.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/tegra/tegra210_amx.c b/sound/soc/tegra/tegra210_amx.c index cc1f9c158191..6a67ce6128b4 100644 --- a/sound/soc/tegra/tegra210_amx.c +++ b/sound/soc/tegra/tegra210_amx.c @@ -504,16 +504,12 @@ static struct snd_kcontrol_new tegra264_amx_controls[] = { static int tegra210_amx_component_probe(struct snd_soc_component *component) { struct tegra210_amx *amx = snd_soc_component_get_drvdata(component); - int err = 0; - if (amx->soc_data->num_controls) { - err = snd_soc_add_component_controls(component, amx->soc_data->controls, - amx->soc_data->num_controls); - if (err) - dev_err(component->dev, "can't add AMX controls, err: %d\n", err); - } + if (amx->soc_data->num_controls) + return snd_soc_add_component_controls(component, amx->soc_data->controls, + amx->soc_data->num_controls); - return err; + return 0; } static const struct snd_soc_component_driver tegra210_amx_cmpnt = { From 226791fa6dbb1e1c6d4c8ac4d10b6c73f0dc32e1 Mon Sep 17 00:00:00 2001 From: bui duc phuc Date: Wed, 15 Jul 2026 12:11:15 +0700 Subject: [PATCH 23/23] ASoC: tegra: tegra210: Use devm_clk_get_optional() for sync_input clock The existing comment states that the sync_input clock is only needed when another I/O is configured to use the current I2S instance as its input clock, and the current code does not treat its absence as an error. Use devm_clk_get_optional() to match the existing behaviour while still reporting real failures via dev_err_probe(). Update the comment to describe the optional nature of the clock rather than the previous error handling. Signed-off-by: bui duc phuc Reviewed-by: Thierry Reding Link: https://patch.msgid.link/20260715051115.17385-5-phucduc.bui@gmail.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_i2s.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index ff8c72fc38c5..e7e26e917291 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -1078,13 +1078,13 @@ static int tegra210_i2s_probe(struct platform_device *pdev) "can't retrieve I2S bit clock\n"); /* - * Not an error, as this clock is needed only when some other I/O - * requires input clock from current I2S instance, which is - * configurable from DT. + * This clock is optional and is only needed when another I/O uses + * the current I2S instance as its input clock, as configured in DT. */ - i2s->clk_sync_input = devm_clk_get(dev, "sync_input"); + i2s->clk_sync_input = devm_clk_get_optional(dev, "sync_input"); if (IS_ERR(i2s->clk_sync_input)) - dev_dbg(dev, "can't retrieve I2S sync input clock\n"); + return dev_err_probe(dev, PTR_ERR(i2s->clk_sync_input), + "can't retrieve I2S sync input clock\n"); regs = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(regs))