From b28d3d0841a6083f82dfa9fe940afcd748f2a1e2 Mon Sep 17 00:00:00 2001 From: Federico Lombardo Date: Thu, 2 Jul 2026 22:43:47 +0200 Subject: [PATCH 01/36] ALSA: hda/realtek: Enable mute LED quirk for HP Laptop 15-dw0xxx HP Laptop 15-dw0xxx with subsystem ID 103c:85f0 needs the ALC236_FIXUP_HP_MUTE_LED_COEFBIT2 quirk to correctly disable and enable the Audio Mute LED. Add the missing quirk entry. Signed-off-by: Federico Lombardo Link: https://patch.msgid.link/20260702204347.187524-1-lombardo.federico.lf@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index f7700713dc62..c54f55730fdc 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6992,6 +6992,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x856a, "HP Pavilion 15-cs1xxx", ALC295_FIXUP_HP_PAVILION_MUTE_LED_1B), SND_PCI_QUIRK(0x103c, 0x85c6, "HP Pavilion x360 Convertible 14-dy1xxx", ALC295_FIXUP_HP_MUTE_LED_COEFBIT11), SND_PCI_QUIRK(0x103c, 0x85de, "HP Envy x360 13-ar0xxx", ALC285_FIXUP_HP_ENVY_X360), + SND_PCI_QUIRK(0x103c, 0x85f0, "HP Laptop 15-dw0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8603, "HP Omen 17-cb0xxx", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x860c, "HP ZBook 17 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT), From cb89f0c1aed02eb233c4271f76f830b37e222ff6 Mon Sep 17 00:00:00 2001 From: Malaya Kumar Rout Date: Sat, 4 Jul 2026 16:27:36 +0530 Subject: [PATCH 02/36] selftests/alsa: Fix memory leak in find_controls error path In find_controls(), card_data is allocated with malloc() but when snd_ctl_open_lconf() fails, the code jumps to next_card without freeing the allocated memory. This results in a memory leak for each card where snd_ctl_open_lconf() fails. Add free(card_data) before goto next_card to ensure proper cleanup of the allocated memory in the error path. Fixes: 5aaf9efffc57 ("kselftest: alsa: Add simplistic test for ALSA mixer controls kselftest") Signed-off-by: Malaya Kumar Rout Link: https://patch.msgid.link/20260704105736.94874-1-malayarout91@gmail.com Signed-off-by: Takashi Iwai --- tools/testing/selftests/alsa/mixer-test.c | 1 + 1 file changed, 1 insertion(+) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index d4f845c32804..c5ae2a026ce9 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -84,6 +84,7 @@ static void find_controls(void) if (err < 0) { ksft_print_msg("Failed to get hctl for card %d: %s\n", card, snd_strerror(err)); + free(card_data); goto next_card; } From 742a87fa54ad7123bff41bd1aa149fef6929a7af Mon Sep 17 00:00:00 2001 From: Daniel Schaefer Date: Sun, 5 Jul 2026 16:07:13 +0800 Subject: [PATCH 03/36] ALSA: hda/realtek: add quirk for Framework F111:0010 Similar to commit 67c738152207 ("ALSA: hda/realtek: fix mic boost on Framework PTL") and previous quirks for Framework systems with Realtek codecs. 0010 is another platform with an ALC285 which needs the same quirk. Signed-off-by: Daniel Schaefer Link: https://patch.msgid.link/20260705-framework-audio-ssid-0x0010-v1-1-fffc8996ac18@frame.work Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index c54f55730fdc..b6b502450a95 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7923,6 +7923,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0xf111, 0x000c, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0xf111, 0x000f, "Framework Laptop 13 Pro PTL", ALC295_FIXUP_FRAMEWORK_LAPTOP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0xf111, 0x010f, "Framework Laptop 13 PTL", ALC295_FIXUP_FRAMEWORK_LAPTOP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0xf111, 0x0010, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_LIMIT_INT_MIC_BOOST), #if 0 /* Below is a quirk table taken from the old code. From aba30af07d4fe499b50209801eba9da8a815522f Mon Sep 17 00:00:00 2001 From: Pengpeng Hou Date: Sun, 5 Jul 2026 16:46:01 +0800 Subject: [PATCH 04/36] ALSA: usb-audio: caiaq: validate EP1 reply lengths usb_ep1_command_reply_dispatch() uses buf[0] as a command byte and then reads command-specific fixed items from the same URB buffer. Several paths use buf + 1, buf[1], buf[2], or buf + 3 without first proving that urb->actual_length contains those bytes. Add per-command length checks, use a payload length derived from the bytes after the command byte for the control-state copy, and reject short analog input payloads before the input helper reads fixed offsets from the EP1 reply. Signed-off-by: Pengpeng Hou Link: https://patch.msgid.link/20260705084601.56400-1-pengpeng@iscas.ac.cn Signed-off-by: Takashi Iwai --- sound/usb/caiaq/device.c | 17 ++++++++++++++--- sound/usb/caiaq/input.c | 6 ++++++ 2 files changed, 20 insertions(+), 3 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index b20aae0caf60..a16e59248480 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -134,14 +134,22 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb) struct device *dev = &urb->dev->dev; struct snd_usb_caiaqdev *cdev = urb->context; unsigned char *buf = urb->transfer_buffer; + unsigned int payload_len; + unsigned int copy_len; if (urb->status || !cdev) { dev_warn(dev, "received EP1 urb->status = %i\n", urb->status); return; } + if (urb->actual_length < 1) + return; + + payload_len = urb->actual_length - 1; switch(buf[0]) { case EP1_CMD_GET_DEVICE_INFO: + if (payload_len < sizeof(struct caiaq_device_spec)) + break; memcpy(&cdev->spec, buf+1, sizeof(struct caiaq_device_spec)); cdev->spec.fw_version = le16_to_cpu(cdev->spec.fw_version); dev_dbg(dev, "device spec (firmware %d): audio: %d in, %d out, " @@ -157,18 +165,21 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb) wake_up(&cdev->ep1_wait_queue); break; case EP1_CMD_AUDIO_PARAMS: + if (payload_len < 1) + break; cdev->audio_parm_answer = buf[1]; wake_up(&cdev->ep1_wait_queue); break; case EP1_CMD_MIDI_READ: + if (urb->actual_length < 3 || urb->actual_length - 3 < buf[2]) + break; snd_usb_caiaq_midi_handle_input(cdev, buf[1], buf + 3, buf[2]); break; case EP1_CMD_READ_IO: if (cdev->chip.usb_id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ)) { - if (urb->actual_length > sizeof(cdev->control_state)) - urb->actual_length = sizeof(cdev->control_state); - memcpy(cdev->control_state, buf + 1, urb->actual_length); + copy_len = min_t(unsigned int, payload_len, sizeof(cdev->control_state)); + memcpy(cdev->control_state, buf + 1, copy_len); wake_up(&cdev->ep1_wait_queue); break; } diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index eabbf41fdfb2..8d924330c54c 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -203,6 +203,8 @@ static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *cdev, switch (cdev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): + if (len < 6) + return; snd_caiaq_input_report_abs(cdev, ABS_X, buf, 2); snd_caiaq_input_report_abs(cdev, ABS_Y, buf, 0); snd_caiaq_input_report_abs(cdev, ABS_Z, buf, 1); @@ -210,11 +212,15 @@ static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *cdev, case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): + if (len < 6) + return; snd_caiaq_input_report_abs(cdev, ABS_X, buf, 0); snd_caiaq_input_report_abs(cdev, ABS_Y, buf, 1); snd_caiaq_input_report_abs(cdev, ABS_Z, buf, 2); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + if (len < 16) + return; snd_caiaq_input_report_abs(cdev, ABS_HAT0X, buf, 4); snd_caiaq_input_report_abs(cdev, ABS_HAT0Y, buf, 2); snd_caiaq_input_report_abs(cdev, ABS_HAT1X, buf, 6); From ea588e4c7484ea883d3dab308f1f2c2f0f51f7d6 Mon Sep 17 00:00:00 2001 From: Rong Zhang Date: Mon, 6 Jul 2026 02:19:31 +0800 Subject: [PATCH 05/36] ALSA: usb-audio: Fix imbalance per-channel volume of sticky mixers I accidentally made an off-by-a-line mistake when mimicking other code paths that set all channels. The mistake breaks sticky mixers with multiple channels. I didn't realize this mistake at that time, as my device's mixer is single-channel. Fix it, so that per-channel volume of sticky mixers is balanced. Fixes: aa2f4addab44 ("ALSA: usb-audio: Set the value of potential sticky mixers to maximum") Signed-off-by: Rong Zhang Link: https://patch.msgid.link/20260706-uac-sticky-channels-fix-v1-1-92741c538283@rong.moe Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b4c855c25eef..703c118f9d4e 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1473,8 +1473,8 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, if (!cval->cmask) { snd_usb_set_cur_mix_value(cval, 0, 0, cval->max); } else { + idx = 0; for (i = 0; i < MAX_CHANNELS; i++) { - idx = 0; if (cval->cmask & BIT(i)) { snd_usb_set_cur_mix_value(cval, i + 1, idx, cval->max); idx++; From c845febafd92b2056abc0af541c1ad85785f1353 Mon Sep 17 00:00:00 2001 From: Malaya Kumar Rout Date: Sun, 5 Jul 2026 22:40:38 +0530 Subject: [PATCH 06/36] selftests/alsa: Fix format specifier and function mismatch in mixer-test In the ctl_value_index_valid() function, when validating 64-bit integer controls (SND_CTL_ELEM_TYPE_INTEGER64), the error message for maximum value validation has two bugs: 1. Uses snd_ctl_elem_info_get_max() instead of the 64-bit variant snd_ctl_elem_info_get_max64() 2. Uses %ld format specifier instead of %lld for long long type This is inconsistent with the minimum value validation (line 335-339) which correctly uses get_min64() and %lld. The mismatch can cause: - Incorrect maximum values being reported on some architectures - Undefined behavior due to format specifier mismatch - Potential crashes when printing the error message Fix by using snd_ctl_elem_info_get_max64() and %lld format specifier to match the data type and be consistent with the minimum value check. Signed-off-by: Malaya Kumar Rout Link: https://patch.msgid.link/20260705171038.171966-1-malayarout91@gmail.com Signed-off-by: Takashi Iwai --- tools/testing/selftests/alsa/mixer-test.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index c5ae2a026ce9..a329f901c5ed 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -340,9 +340,9 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, } if (int64_val > snd_ctl_elem_info_get_max64(ctl->info)) { - ksft_print_msg("%s.%d value %lld more than maximum %ld\n", + ksft_print_msg("%s.%d value %lld more than maximum %lld\n", ctl->name, index, int64_val, - snd_ctl_elem_info_get_max(ctl->info)); + snd_ctl_elem_info_get_max64(ctl->info)); return false; } From 04f15d8f393037ed72876399882265604a370b2b Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Fri, 3 Jul 2026 14:02:10 +0100 Subject: [PATCH 07/36] 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 08/36] 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 09/36] 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 335f7860846732477a9ce24f0db04dd253ba268c Mon Sep 17 00:00:00 2001 From: Matthew Schwartz Date: Mon, 6 Jul 2026 17:56:14 -0700 Subject: [PATCH 10/36] ALSA: hda/realtek: Fix headphone output on ASUS ROG Ally X The ASUS ROG Ally X (RC72LA, PCI SSID 0x1043:0x1eb3) with an ALC294 codec produces no sound from the headphone jack, and unplugging the headphones can leave the speakers muted too. The device uses ALC287_FIXUP_TAS2781_I2C, which chains into the ThinkPad fixup chain. alc285_fixup_thinkpad_x1_gen7() makes the headphone (pin 0x21) and speaker (pin 0x17) share DAC 0x03. That is correct for a ThinkPad, but here the shared DAC is exposed as a "Line Out" control that defaults to zero, so the headphones stay silent when plugged in. Add a dedicated fixup chain for the Ally X that sets up the TAS2781 amplifier and the headset jack without the ThinkPad DAC sharing, and routes the speaker to DAC 0x02 so the headphone and speaker get independent DACs. Also, fixup the wrong product code (it's RC72LA, not RCLA72). Signed-off-by: Matthew Schwartz Link: https://patch.msgid.link/20260707005615.52889-1-matthew.schwartz@linux.dev Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index b6b502450a95..8d409ccacad4 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -4117,6 +4117,8 @@ enum { ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, ALC287_FIXUP_TAS2781_I2C, + ALC287_FIXUP_ASUS_ALLY_X, + ALC287_FIXUP_ASUS_ALLY_X_I2C, ALC295_FIXUP_DELL_TAS2781_I2C, ALC245_FIXUP_TAS2781_SPI_2, ALC287_FIXUP_TXNW2781_I2C, @@ -6476,6 +6478,18 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, }, + [ALC287_FIXUP_ASUS_ALLY_X] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + .chained = true, + .chain_id = ALC287_FIXUP_ASUS_ALLY_X_I2C, + }, + [ALC287_FIXUP_ASUS_ALLY_X_I2C] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_tias_i2c, + .chained = true, + .chain_id = ALC225_FIXUP_HEADSET_JACK, + }, [ALC245_FIXUP_TAS2781_SPI_2] = { .type = HDA_FIXUP_FUNC, .v.func = tas2781_fixup_spi, @@ -7461,7 +7475,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1e93, "ASUS ExpertBook B9403CVAR", ALC294_FIXUP_ASUS_HPE), - SND_PCI_QUIRK(0x1043, 0x1eb3, "ASUS Ally RCLA72", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x1eb3, "ASUS Ally RC72LA", ALC287_FIXUP_ASUS_ALLY_X), SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2), HDA_CODEC_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC285_FIXUP_ASUS_I2C_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), From 883d0cae6d5f840629c5bcacf029c6aa9a9106f4 Mon Sep 17 00:00:00 2001 From: Matthew Schwartz Date: Mon, 6 Jul 2026 17:56:15 -0700 Subject: [PATCH 11/36] ALSA: hda/realtek: Fix headset mic on ASUS ROG Ally X The ASUS ROG Ally X (RC72LA, PCI SSID 0x1043:0x1eb3) leaves the combo jack microphone pin (0x19) unconfigured, so a headset microphone is neither detected nor captured. Only the internal microphone is available. Add a pin override that configures node 0x19 as an external headset mic, the same way the original ROG Ally does. The headphone output fixup chain is preserved, so speaker and headphone routing are unchanged. Signed-off-by: Matthew Schwartz Link: https://patch.msgid.link/20260707005615.52889-2-matthew.schwartz@linux.dev Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 8d409ccacad4..984868e3d8bb 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -4118,6 +4118,7 @@ enum { ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, ALC287_FIXUP_TAS2781_I2C, ALC287_FIXUP_ASUS_ALLY_X, + ALC287_FIXUP_ASUS_ALLY_X_SPEAKER, ALC287_FIXUP_ASUS_ALLY_X_I2C, ALC295_FIXUP_DELL_TAS2781_I2C, ALC245_FIXUP_TAS2781_SPI_2, @@ -6479,6 +6480,15 @@ static const struct hda_fixup alc269_fixups[] = { .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, }, [ALC287_FIXUP_ASUS_ALLY_X] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, /* headset mic */ + { } + }, + .chained = true, + .chain_id = ALC287_FIXUP_ASUS_ALLY_X_SPEAKER, + }, + [ALC287_FIXUP_ASUS_ALLY_X_SPEAKER] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_speaker2_to_dac1, .chained = true, From dba0bfded433d412506eb674d6af77cdca82e259 Mon Sep 17 00:00:00 2001 From: Longlong Xia Date: Mon, 6 Jul 2026 12:26:39 +0900 Subject: [PATCH 12/36] ALSA: firewire: Drop redundant mod_devicetable.h includes The ALSA FireWire driver headers include , which already provides struct ieee1394_device_id via . Drop the now-redundant direct includes from the per-device headers. Signed-off-by: Longlong Xia Reviewed-by: Takashi Sakamoto Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20260706032639.273929-1-xialonglong2025@163.com --- sound/firewire/bebob/bebob.h | 1 - sound/firewire/dice/dice.h | 1 - sound/firewire/digi00x/digi00x.h | 1 - sound/firewire/fireface/ff.h | 1 - sound/firewire/fireworks/fireworks.h | 1 - sound/firewire/motu/motu.h | 1 - sound/firewire/oxfw/oxfw.h | 1 - sound/firewire/tascam/tascam.h | 1 - 8 files changed, 8 deletions(-) diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index 4d73ecb30d79..41e019b9e24e 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -13,7 +13,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index 7744ea6a0791..5abae26d2bba 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -17,7 +17,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h index 82b647d383c5..60d17a6fddd1 100644 --- a/sound/firewire/digi00x/digi00x.h +++ b/sound/firewire/digi00x/digi00x.h @@ -12,7 +12,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/firewire/fireface/ff.h b/sound/firewire/fireface/ff.h index 7e42f5778a8a..c9c35c67db27 100644 --- a/sound/firewire/fireface/ff.h +++ b/sound/firewire/fireface/ff.h @@ -12,7 +12,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index c8d5879efe28..a8aadf754e72 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -13,7 +13,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index c66be0a89ccf..7333a76b50aa 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -12,7 +12,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 39ea9a6dde33..aeb83e1595b7 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -9,7 +9,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index d07ffcb27be6..9c42c9b48f21 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -12,7 +12,6 @@ #include #include #include -#include #include #include #include From f67be28fdf8b5d31ac1cc1152bb17250f9f8f513 Mon Sep 17 00:00:00 2001 From: Xu Rao Date: Tue, 7 Jul 2026 21:24:19 +0800 Subject: [PATCH 13/36] ALSA: hda: Fix cached processing coefficient verbs Intel HD Audio defines Coefficient Index and Processing Coefficient as separate audio widget controls in the Audio Widget Verb Definitions: Coefficient Index selects the coefficient slot, while Processing Coefficient accesses the value at the selected slot. hda_reg_read_coef() selects the slot with AC_VERB_SET_COEF_INDEX, but then uses AC_VERB_GET_COEF_INDEX for the value read. That reads back the selected index instead of the coefficient value. hda_reg_write_coef() has the same issue and builds the value write from AC_VERB_GET_COEF_INDEX instead of AC_VERB_SET_PROC_COEF. This only affects the regmap coefficient cache path used by codecs that set codec->cache_coef. Direct coefficient helpers already use the normal SET_COEF_INDEX followed by GET_PROC_COEF or SET_PROC_COEF sequence, which is likely why this has not been noticed widely. Use AC_VERB_GET_PROC_COEF for cached coefficient reads and AC_VERB_SET_PROC_COEF for cached coefficient writes. Fixes: 40ba66a702b8 ("ALSA: hda - Add cache support for COEF read/write") Cc: stable@vger.kernel.org Signed-off-by: Xu Rao Link: https://patch.msgid.link/DB9023BF2920BA99+20260707132419.1731342-1-raoxu@uniontech.com Signed-off-by: Takashi Iwai --- sound/hda/core/regmap.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/hda/core/regmap.c b/sound/hda/core/regmap.c index e7b866fc52c1..d6eb17aa9e08 100644 --- a/sound/hda/core/regmap.c +++ b/sound/hda/core/regmap.c @@ -214,7 +214,7 @@ static int hda_reg_read_coef(struct hdac_device *codec, unsigned int reg, err = snd_hdac_exec_verb(codec, verb, 0, NULL); if (err < 0) return err; - verb = (reg & ~0xfffff) | (AC_VERB_GET_COEF_INDEX << 8); + verb = (reg & ~0xfffff) | (AC_VERB_GET_PROC_COEF << 8); return snd_hdac_exec_verb(codec, verb, 0, val); } @@ -232,7 +232,7 @@ static int hda_reg_write_coef(struct hdac_device *codec, unsigned int reg, err = snd_hdac_exec_verb(codec, verb, 0, NULL); if (err < 0) return err; - verb = (reg & ~0xfffff) | (AC_VERB_GET_COEF_INDEX << 8) | + verb = (reg & ~0xfffff) | (AC_VERB_SET_PROC_COEF << 8) | (val & 0xffff); return snd_hdac_exec_verb(codec, verb, 0, NULL); } From 6b59c53c8adc2b522327407af5e1793a65b67e4b Mon Sep 17 00:00:00 2001 From: Christian Hewitt Date: Sat, 27 Jun 2026 13:12:05 +0000 Subject: [PATCH 14/36] 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 15/36] 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 16/36] 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 17/36] 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 18/36] 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 2d184128f45afad7bfdc17d946339765555dbd38 Mon Sep 17 00:00:00 2001 From: Agustin Luzardo Date: Sun, 5 Jul 2026 15:42:27 -0300 Subject: [PATCH 19/36] ALSA: usb-audio: Add quirk for Redragon H510-PRO Wireless headset The device with USB ID 040b:0897 (Weltrend Semiconductor chipset, sold rebranded as the Redragon H510-PRO Wireless headset, reporting "XiiSound Technology Corporation" in its USB string descriptors) reports a constant value on GET_CUR for its PCM Playback Volume control while still supporting an actually tunable volume. This trips the sticky-value detection in check_sticky_volume_control(), which disables the mixer control entirely: usb 1-4: 5:0: sticky mixer values (0/100/1 => 80), disabling As a result, the device boots with playback volume effectively muted and provides no way to raise it through the normal ALSA/PipeWire mixer path. Apply QUIRK_FLAG_MIXER_GET_CUR_BROKEN so the sticky check marks the control as get_cur_broken and relies on the cached value instead of disabling the mixer control outright. Tested by backporting this quirk flag and the supporting get_cur_broken logic onto a Linux 7.1.2-zen kernel build that does not yet carry it, and confirming that after applying the flag the kernel log changes from usb 1-4: 5:0: sticky mixer values (0/100/1 => 80), disabling to usb 1-4: 5:0: broken mixer GET_CUR (0/100/1 => 80) with the control usable via the driver's cached value afterward. Signed-off-by: Agustin Luzardo Link: https://patch.msgid.link/20260705184227.113588-1-agustinluzardo09@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 1cb588691e16..2949a0d2d961 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2210,6 +2210,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_FORCE_IFACE_RESET | QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x03f0, 0x654a, /* HP 320 FHD Webcam */ QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_MIC_RES_16), + DEVICE_FLG(0x040b, 0x0897, /* Weltrend Semiconductor, sold as Redragon H510-PRO Wireless headset */ + QUIRK_FLAG_MIXER_GET_CUR_BROKEN), DEVICE_FLG(0x041e, 0x3000, /* Creative SB Extigy */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x041e, 0x4080, /* Creative Live Cam VF0610 */ From 5c3f4b583c25911fad9890b463632761291f65ab Mon Sep 17 00:00:00 2001 From: Rohit Sinha Date: Tue, 7 Jul 2026 02:48:34 +0530 Subject: [PATCH 20/36] ALSA: hda/realtek - Add quirk for HP Victus 15-fa0xxx (MB 8A50) The mute LED on HP Victus 15-fa0xxx (board ID 8A50) uses ALC245 but requires a quirk to work. This patch enables the existing ALC245_FIXUP_HP_MUTE_LED_COEFBIT quirk for the device. Tested on my HP Victus 15-fa0xxx. The LED behaviour works as intended. Signed-off-by: Rohit Sinha Link: https://patch.msgid.link/20260706211834.384022-1-rohitsinha12043@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 984868e3d8bb..5d9cbf5b7b40 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7146,6 +7146,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8a36, "HP Pavilion Plus 14-eh0xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8a3d, "HP Victus 15-fb0xxx (MB 8A3D)", ALC245_FIXUP_HP_MUTE_LED_V2_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8a4f, "HP Victus 15-fa0xxx (MB 8A4F)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), + SND_PCI_QUIRK(0x103c, 0x8a50, "HP Victus 15-fa0xxx (MB 8A50)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8a6e, "HP EDNA 360", ALC287_FIXUP_CS35L41_I2C_4), SND_PCI_QUIRK(0x103c, 0x8a74, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8a75, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), From 0752734163a20c9185a12c0d0786995751ab7426 Mon Sep 17 00:00:00 2001 From: Kshamendra Kumar Mishra Date: Tue, 7 Jul 2026 22:07:44 +0530 Subject: [PATCH 21/36] ALSA: hda/realtek: Add mic mute LED quirk for HP Laptop 15-fd0xxx The HP Laptop 15-fd0xxx uses GPIO bit 0 to control the mic mute LED, but currently the quirk only enables the speaker mute LED (via coef bit). Switch to the existing ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO fixup which additionally registers a GPIO-based mic mute LED, mapping to the same verbs (SET_GPIO_DATA/SET_GPIO_DIRECTION bit 0) that the platform firmware uses. Signed-off-by: Kshamendra Kumar Mishra Link: https://patch.msgid.link/DJSOWKFBTJH8.2Z7EPWX514QXM@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 5d9cbf5b7b40..02cc3f6b378d 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7294,7 +7294,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8da8, "HP 16 Piston OmniBook X", ALC245_FIXUP_HP_ENVY_X360_15_FH0XXX), SND_PCI_QUIRK(0x103c, 0x8dc9, "HP Laptop 15-fc0xxx", ALC236_FIXUP_HP_DMIC), SND_PCI_QUIRK(0x103c, 0x8dd4, "HP EliteStudio 8 AIO", ALC274_FIXUP_HP_AIO_BIND_DACS), - SND_PCI_QUIRK(0x103c, 0x8dd7, "HP Laptop 15-fd0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), + SND_PCI_QUIRK(0x103c, 0x8dd7, "HP Laptop 15-fd0xxx", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), SND_PCI_QUIRK(0x103c, 0x8de8, "HP Gemtree", ALC245_FIXUP_TAS2781_SPI_2), SND_PCI_QUIRK(0x103c, 0x8de9, "HP Gemtree", ALC245_FIXUP_TAS2781_SPI_2), SND_PCI_QUIRK(0x103c, 0x8dec, "HP EliteBook 640 G12", ALC236_FIXUP_HP_GPIO_LED), From bc0d4444f384d183e2113b72c34643069f93ab63 Mon Sep 17 00:00:00 2001 From: Ravi Hothi Date: Fri, 3 Jul 2026 13:04:32 +0530 Subject: [PATCH 22/36] 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 23/36] 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 24/36] 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 25/36] 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 26/36] 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 97272a5704bf13369bba43951ad9fe8f806ccc7f Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 9 Jul 2026 14:57:06 +0800 Subject: [PATCH 27/36] ALSA: hda/realtek - Fixed Headphone noise issue for Dell QCM1255 This platform booted with Ubuntu 24.04 with Pipewire audio server. So, it has pop noise with headphone. But it's normal with Pulseaudio server. This patch was the workaround. Connect the headphones to DAC 0x2. The popping sound will disappear. Signed-off-by: Kailang Yang Link: https://lore.kernel.org/34b990cb56914148ba02fa8e9d176479@realtek.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 02cc3f6b378d..2fc25be99ca6 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -4172,6 +4172,7 @@ enum { ALC256_FIXUP_HONOR_MRB_XXX_M1020_AUDIO, ALC245_FIXUP_HP_ENVY_X360_15_FH0XXX, ALC287_FIXUP_ACER_MICMUTE_LED, + ALC236_FIXUP_DELL_HP_POP_NOISE, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -6779,6 +6780,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC2XX_FIXUP_HEADSET_MIC, }, + [ALC236_FIXUP_DELL_HP_POP_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_invalidate_dacs, + }, }; static const struct hda_quirk alc269_fixup_tbl[] = { @@ -6930,6 +6935,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0cc3, "Dell Oasis 14 Low Weight MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0cc4, "Dell Oasis 16 MTL-H/U", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0cc5, "Dell Oasis 14", ALC289_FIXUP_RTK_AMP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0e6b, "Dell Pro QCM1255", ALC236_FIXUP_DELL_HP_POP_NOISE), + SND_PCI_QUIRK(0x1028, 0x0e6d, "Dell Pro Micro QCM1255", ALC236_FIXUP_DELL_HP_POP_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), From d595255241e5fec0c94adeebf2565524398e37c5 Mon Sep 17 00:00:00 2001 From: Eckhart Mohr Date: Wed, 8 Jul 2026 15:21:06 +0200 Subject: [PATCH 28/36] ALSA: hda/realtek: Add quirk for TongFang X6xx45xU Fix microphone detection on built in headphone jack for some devices. Signed-off-by: Eckhart Mohr Cc: stable@vger.kernel.org Signed-off-by: Werner Sembach Link: https://patch.msgid.link/20260708132135.102680-1-wse@tuxedocomputers.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 2fc25be99ca6..b47177d734c7 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7918,6 +7918,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d05, 0x300f, "TongFang X6AR5xxY", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1d05, 0x3019, "TongFang X6FR5xxY", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1d05, 0x3031, "TongFang X6AR55xU", ALC2XX_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1d05, 0x3034, "TongFang X6xx45xU", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1d17, 0x3288, "Haier Boyue G42", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), From f86a7c96406d406a66f3d7653eca86aa576fcf6b Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 9 Jul 2026 13:12:24 +0100 Subject: [PATCH 29/36] ALSA: hda/cirrus_scodec: Make Kconfig visible if KUNIT Make the Kconfig item for cirrus_scodec visible if CONFIG_KUNIT is enabled. This is so that its KUnit test can be enabled by KUnit scripts without requiring a large amount of irrelevant additional components. The general rule for KUNIT_ALL_TESTS is that it should only enable tests for components that are already selected. However, the UML environment does not support ACPI, which means the HDA codec drivers that use cirrus_scodec cannot be selected. But cirrus_scodec does not need ACPI. By making the Kconfig option visible if CONFIG_KUNIT, the KUnit test can be enabled with only the minimal set of functionality that is required for cirrus_scodec. This is still compliant with the KUNIT_ALL_TESTS rule "only tests for enabled modules" because by default cirrus_scodec will only be enabled if the drivers that use it are enabled. It must be intentionally enabled to force it to be included for testing. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260709121224.614350-1-rf@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/side-codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/codecs/side-codecs/Kconfig b/sound/hda/codecs/side-codecs/Kconfig index 2a2e8804bf9e..1cfd83e251e4 100644 --- a/sound/hda/codecs/side-codecs/Kconfig +++ b/sound/hda/codecs/side-codecs/Kconfig @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_HDA_CIRRUS_SCODEC - tristate + tristate "Cirrus side-codec library" if KUNIT config SND_HDA_CIRRUS_SCODEC_KUNIT_TEST tristate "KUnit test for Cirrus side-codec library" if !KUNIT_ALL_TESTS From 8ca0016a9b799d8366b32da7e729fcf56627516f Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 9 Jul 2026 13:22:11 +0100 Subject: [PATCH 30/36] ALSA: hda: MAINTAINERS: Fix missing cirrus* file reference When the HDA source was reorganized some of the cirrus* files were moved into a new 'side-codecs' subdirectory. But MAINTAINERS wasn't updated to add a cirrus* file reference to cover these moved files. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260709122211.615785-1-rf@opensource.cirrus.com Signed-off-by: Takashi Iwai --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 4a8b0fd665ce..bf59602eec39 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6209,6 +6209,7 @@ F: include/dt-bindings/sound/cs* F: include/linux/mfd/cs42l43* F: include/sound/cs* F: sound/hda/codecs/cirrus* +F: sound/hda/codecs/side-codecs/cirrus* F: sound/hda/codecs/side-codecs/cs* F: sound/hda/codecs/side-codecs/hda_component* F: sound/soc/codecs/cs* From 93b47e66cc6d6c6382d44b44f5e7f6fc3a7b38c3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Jul 2026 11:56:06 +0200 Subject: [PATCH 31/36] ALSA: usb-audio: Skip DSD quirk for Musical Fidelity M6s DAC Salvador reported that the recent fix for applying the DSD quirk to Musical Fidelity devices broke for his M6s DAC model (2772:0502). Although this is basically a firmware bug, the model in question is fairly old, and no further firmware update can be expected, so it'd be better to address in the driver side. As an ad hoc workaround, skip the DSD quirk for this device by adding an empty quirk entry of 2772:0502; this essentially skips the later DSD quirk entry by the match with the vendor 2772. Fixes: da3a7efff64e ("ALSA: usb-audio: Update for native DSD support quirks") Reported-by: Salvador Blaya Closes: https://lore.kernel.org/CAOdyq+qFaqCh=tK_wNnA64hv5pQuA1Y09ANxQ=xK8yR-t4mf9Q@mail.gmail.com Tested-by: Salvador Blaya Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20260709095614.1418838-1-tiwai@suse.de --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2949a0d2d961..d98825f9697c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2463,6 +2463,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x2708, 0x0002, /* Audient iD14 */ QUIRK_FLAG_IGNORE_CTL_ERROR), + DEVICE_FLG(0x2772, 0x0502, /* Musical Fidelity M6s DAC */ + 0), /* for avoiding QUIRK_FLAG_DSD_RAW with vendor match */ DEVICE_FLG(0x2912, 0x30c8, /* Audioengine D1 */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x2a70, 0x1881, /* OnePlus Technology (Shenzhen) Co., Ltd. BE02T */ From 07ae060d35900977a93d4a05ec995b06c85c2552 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 9 Jul 2026 17:12:06 +0100 Subject: [PATCH 32/36] ALSA: hda: cs35l56: Fail if wmfw file is missing Fail cs35l56_hda_fw_load() and log an error if it didn't find a wmfw file and the BIOS didn't patch the firmware already. Also remove a section of code from cs35l56_hda_request_firmware_files() that attempted to find a .bin if a .wmfw was not found. The CS35L56 ROM can only provide default audio in SoundWire mode. A wmfw is needed to enable I2S audio. Also none of the customer-specific .bin files are compatible with the ROM firmware. So a .wmfw file is always required. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260709161211.686498-1-rf@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/side-codecs/cs35l56_hda.c | 30 +++++++--------------- 1 file changed, 9 insertions(+), 21 deletions(-) diff --git a/sound/hda/codecs/side-codecs/cs35l56_hda.c b/sound/hda/codecs/side-codecs/cs35l56_hda.c index a0ea08eb96a9..78c2cf387a00 100644 --- a/sound/hda/codecs/side-codecs/cs35l56_hda.c +++ b/sound/hda/codecs/side-codecs/cs35l56_hda.c @@ -512,20 +512,6 @@ static void cs35l56_hda_request_firmware_files(struct cs35l56_hda *cs35l56, NULL, "bin"); return; } - - /* - * Check for system-specific bin files without wmfw before - * falling back to generic firmware - */ - if (amp_name) - cs35l56_hda_request_firmware_file(cs35l56, coeff_firmware, coeff_filename, - base_name, system_name, amp_name, "bin"); - if (!*coeff_firmware) - cs35l56_hda_request_firmware_file(cs35l56, coeff_firmware, coeff_filename, - base_name, system_name, NULL, "bin"); - - if (*coeff_firmware) - return; } ret = cs35l56_hda_request_firmware_file(cs35l56, wmfw_firmware, wmfw_filename, @@ -616,13 +602,15 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) &wmfw_firmware, &wmfw_filename, &coeff_firmware, &coeff_filename); - /* - * If the BIOS didn't patch the firmware a bin file is mandatory to - * enable the ASP· - */ - if (!coeff_firmware && firmware_missing) { - dev_err(cs35l56->base.dev, ".bin file required but not found\n"); - goto err_fw_release; + /* If the BIOS didn't patch the firmware a wmfw and bin file are mandatory */ + if (firmware_missing) { + if (!wmfw_firmware) { + dev_err(cs35l56->base.dev, ".%s file required but not found\n", "wmfw"); + goto err_fw_release; + } else if (!coeff_firmware) { + dev_err(cs35l56->base.dev, ".%s file required but not found\n", "bin"); + goto err_fw_release; + } } mutex_lock(&cs35l56->base.irq_lock); From 9064637fb2a80b43105900a47d414997630e5b6b Mon Sep 17 00:00:00 2001 From: Chen Bowen Date: Fri, 10 Jul 2026 21:34:09 +0800 Subject: [PATCH 33/36] ALSA: hda/realtek: Fix speakers on MECHREVO WUJIE Series The internal speakers on the MECHREVO WUJIE Series are silent, while the headphone output works correctly. The BIOS reports NID 0x1b on the Realtek ALC233 codec as unconnected with pin configuration 0x411111f0. However, the pin is connected to an internal speaker. Overriding NID 0x1b with 0x90170150 makes the HDA generic parser detect both 0x14 and 0x1b as speaker outputs and restores internal speaker playback. Add a pin configuration fixup for the affected PCI SSID c011:1d05. Tested on a MECHREVO WUJIE Series laptop with board WUJIE Series-Lark4-LNL and firmware EM_LNL326_V1.0.23. Signed-off-by: Chen Bowen Link: https://patch.msgid.link/20260710133409.16641-1-hicbowen@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index b47177d734c7..55d699d5afca 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -3942,6 +3942,7 @@ enum { ALC275_FIXUP_DELL_XPS, ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, + ALC233_FIXUP_WUJIE_SPEAKERS, ALC233_FIXUP_LENOVO_L2MH_LOW_ENLED, ALC255_FIXUP_DELL_SPK_NOISE, ALC225_FIXUP_DISABLE_MIC_VREF, @@ -4210,6 +4211,13 @@ static void alc287_fixup_lenovo_yoga_book_9i(struct hda_codec *codec, } static const struct hda_fixup alc269_fixups[] = { + [ALC233_FIXUP_WUJIE_SPEAKERS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x90170150 }, /* internal speaker */ + { } + }, + }, [ALC269_FIXUP_GPIO2] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_gpio2, @@ -7949,6 +7957,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC295_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0xc011, 0x1d05, "MECHREVO WUJIE Series", ALC233_FIXUP_WUJIE_SPEAKERS), SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0xf111, 0x0006, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0xf111, 0x0009, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), From d35dfb6329accfe1cfa0b57e35214b5cbbe0f9ae Mon Sep 17 00:00:00 2001 From: Damien Laine Date: Sun, 12 Jul 2026 23:37:08 +0200 Subject: [PATCH 34/36] ALSA: hda/realtek: Fix speakers on Legion Pro 7 16ARX8H with codec SSID 17aa:38a7 Some units of the Lenovo Legion Pro 7 16ARX8H (82WS) report codec subsystem ID 17aa:38a7 instead of 17aa:38a8. Since only 38a8 has a codec SSID quirk, these machines fall through to the PCI SSID match 17aa:386f (Legion Pro 7i 16IAX7) and get ALC287_FIXUP_CS35L41_I2C_2, which probes the Cirrus amplifiers of the Intel variant. The TI TAS2781 amplifier (ACPI TIAS2781:00) present on this AMD variant is never bound and the internal speakers remain silent. Add a codec SSID quirk for 17aa:38a7 pointing to ALC287_FIXUP_TAS2781_I2C, mirroring the existing 38a8 entry. Tested on a Legion Pro 7 16ARX8H (82WS, BIOS LPCN62WW): with the codec SSID overridden to 17aa:38a8 via the HDA patch loader, the TAS2781 amplifier binds and the internal speakers work. Cc: Signed-off-by: Damien Laine Link: https://patch.msgid.link/20260712213708.1835469-1-damien.laine@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 55d699d5afca..443bc92c5e4b 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7807,6 +7807,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { HDA_CODEC_QUIRK(0x17aa, 0x386e, "Legion Y9000X 2022 IAH7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x386e, "Yoga Pro 7 14ARP8", ALC285_FIXUP_SPEAKER2_TO_DAC1), HDA_CODEC_QUIRK(0x17aa, 0x38a8, "Legion Pro 7 16ARX8H", ALC287_FIXUP_TAS2781_I2C), /* this must match before PCI SSID 17aa:386f below */ + HDA_CODEC_QUIRK(0x17aa, 0x38a7, "Legion Pro 7 16ARX8H", ALC287_FIXUP_TAS2781_I2C), /* this must match before PCI SSID 17aa:386f below */ SND_PCI_QUIRK(0x17aa, 0x386f, "Legion Pro 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C), SND_PCI_QUIRK(0x17aa, 0x3874, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), From f52524da7084c1a54683ae9fbc73e93fff19dd64 Mon Sep 17 00:00:00 2001 From: Zhang Heng Date: Mon, 13 Jul 2026 18:03:29 +0800 Subject: [PATCH 35/36] ALSA: hda: conexant: Remove mic bias threshold override Remove the mic bias current comparator threshold override (NID 0x1c, verb 0x320, value 0x010) from Conexant codec driver. This override was originally intended to support volume up/down controls on headsets with inline remote controls, but it causes microphone detection failures on some headsets with impedance less than 1k ohm. After consulting with the vendor's engineers, it was confirmed that this setting is board-specific and should be handled by BIOS/firmware rather than the generic codec driver, especially since inline remote support is not currently implemented. Fixes: 7aeb25908648 ("ALSA: hda/conexant: Fix headset auto detect fail in cx8070 and SN6140") Cc: stable@vger.kernel.org Signed-off-by: Zhang Heng Link: https://patch.msgid.link/20260713100329.306892-1-zhangheng@kylinos.cn Signed-off-by: Takashi Iwai --- sound/hda/codecs/conexant.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/hda/codecs/conexant.c b/sound/hda/codecs/conexant.c index 3d92262763f6..40da2832ba66 100644 --- a/sound/hda/codecs/conexant.c +++ b/sound/hda/codecs/conexant.c @@ -162,9 +162,6 @@ static void cx_fixup_headset_recog(struct hda_codec *codec) { unsigned int mic_present; - /* fix some headset type recognize fail issue, such as EDIFIER headset */ - /* set micbias output current comparator threshold from 66% to 55%. */ - snd_hda_codec_write(codec, 0x1c, 0, 0x320, 0x010); /* set OFF voltage for DFET from -1.2V to -0.8V, set headset micbias register * value adjustment trim from 2.2K ohms to 2.0K ohms. */ From 5caf27a2bf7f86f92f03e851d252901b64ed498c Mon Sep 17 00:00:00 2001 From: Lianqin Hu Date: Mon, 13 Jul 2026 12:35:50 +0000 Subject: [PATCH 36/36] ALSA: usb-audio: Add delay quirk for iBasso DC-Elite Audio control requests that sets sampling frequency sometimes fail on this card. Adding delay between control messages eliminates that problem. usb 1-1: New USB device found, idVendor=2fc6, idProduct=f0b5 usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: iBasso DC-Elite usb 1-1: Manufacturer: iBasso usb 1-1: SerialNumber: CTUA171130B Signed-off-by: Lianqin Hu Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/TYUPR06MB6217D8FF419F24378196FCEFD2FA2@TYUPR06MB6217.apcprd06.prod.outlook.com --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d98825f9697c..41149561aa06 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2487,6 +2487,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_MIXER_GET_CUR_BROKEN), DEVICE_FLG(0x2fc6, 0xf06b, /* MOONDROP Moonriver2 Ti */ QUIRK_FLAG_CTL_MSG_DELAY), + DEVICE_FLG(0x2fc6, 0xf0b5, /* iBasso DC-Elite */ + QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x2fc6, 0xf0b7, /* iBasso DC07 Pro */ QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x30be, 0x0101, /* Schiit Hel */