When a keep-alive (KAE) silent stream is active on an Intel HDMI/DP
codec, opening a real PCM stream reprograms the converter format and the
audio infoframe in snd_hda_hdmi_generic_pcm_prepare(). Part of that
reprogramming - the converter channel count and the channel mapping in
snd_hda_hdmi_setup_audio_infoframe() - is not safe to do while a
keep-alive stream is active. This is most visible when switching to a
multichannel PCM configuration, where the active channel count actually
changes. In that case the newly opened PCM stream plays no sound.
Add an optional hdmi_ops .prepare hook, called at the start of the
PCM prepare sequence (before the format and infoframe are touched), and
implement it for HSW+ to release keep-alive. Keep-alive is then
re-enabled as before once the new stream has been set up, in the
setup_stream op.
Fixes: 15175a4f2b ("ALSA: hda/hdmi: add keep-alive support for ADL-P and DG2")
Reported-by: Alexander Kaplan <alexander.kaplan@sms-medipool.de>
Closes: https://gitlab.freedesktop.org/drm/xe/kernel/-/work_items/8412
Tested-by: Alexander Kaplan <alexander.kaplan@sms-medipool.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://patch.msgid.link/20260715180610.1371243-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
cs35l41_get_acpi_mute_state() evaluates a _DSM method to get the ACPI
mute state and reads the first byte from the returned object.
However, the returned ACPI object is owned by the caller and is never
freed after use, so each successful query leaks the _DSM result object.
The code also assumes that the returned object is a buffer with at least
one byte. A malformed firmware response can return a different object
type or an empty buffer, and the direct ret->buffer.pointer dereference
can then access an invalid pointer.
Use the typed _DSM helper, validate that the returned buffer contains at
least one byte, and free the ACPI object after reading it.
Fixes: 447106e92a ("ALSA: hda: cs35l41: Support mute notifications for CS35L41 HDA")
Signed-off-by: Guangshuo Li <lgs201920130244@gmail.com>
Link: https://patch.msgid.link/20260708113625.752913-1-lgs201920130244@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Alienware x16 R2 has two pairs of speakers, but the BIOS
marks pin 0x17 as unused, so only the pin 0x14 pair plays and
audio is very quiet/dull.
Apply ALC289_FIXUP_DUAL_SPK like on other Dell machines to set
up pin 0x17 and route it to DAC1. Tested on my x16 R2 with
kernel 6.18.38, and now all speakers play at full volume.
Signed-off-by: Oliver Ohrt <oliver@theohrts.com>
Link: https://patch.msgid.link/20260715070409.42696-1-oliver@theohrts.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The sound and microphone mute LEDs do not function on this newer
revision of the board (8AB8) while they do on the older 880D models.
I have verified this on another laptop which was manufactured before
the one with the issue.
Added the ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED from a G9 model, which
uses the same codec, to make it work. Tested on kernel version 7.1.3 on
the aforementioned newer revision notebook.
Signed-off-by: Marcel Kłos <marcel@marmak.net.pl>
Link: https://patch.msgid.link/4dab5622-9100-4730-8c99-b58da939549b@marmak.net.pl
Signed-off-by: Takashi Iwai <tiwai@suse.de>
queue_delete() closes the queue timer, then frees it. snd_seq_timer_close()
clears q->timer->timeri. snd_use_lock_sync() then drains borrowers, and
snd_seq_timer_delete() frees q->timer.
A borrower can re-open the timer inside that window. A SET_QUEUE_CLIENT
that took a queueptr() use_lock reference before the queue was unlinked
runs snd_seq_timer_open() after the close. Open refuses re-open only while
timeri is set, and the close just cleared it, so it re-opens timeri.
snd_seq_timer_delete() does not close that instance. Its snd_seq_timer_stop()
is a no-op, because running was cleared first. So it frees q->timer with the
instance still live. The queue is freed next.
The instance stays on the global timer with callback_data pointing at the
freed queue. A non-owner START on the unlocked queue arms it. The next tick
derefs the freed queue in snd_seq_timer_interrupt().
Reachable by an unprivileged user with access to /dev/snd/seq. No CAP and
no queue ownership required.
Close any lingering instance in the destructor. There, ->timeri can no
longer change: the queue is unlinked and all use_lock borrowers have
drained, so no snd_seq_queue_use() can re-open it. Close it before clearing
q->timer. snd_timer_close() waits for any in-flight snd_seq_timer_interrupt()
to finish, and that callback still reads q->timer (via snd_seq_check_queue()),
so q->timer must stay valid until it drains.
Fixes: 1da177e4c3 ("Linux-2.6.12-rc2")
Cc: stable@vger.kernel.org
Assisted-by: Claude:claude-opus-4-8
Signed-off-by: Norbert Szetei <norbert@doyensec.com>
Link: https://patch.msgid.link/422FDB81-2A68-47C7-A22D-2D3301E2E86D@doyensec.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 <hulianqin@vivo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/TYUPR06MB6217D8FF419F24378196FCEFD2FA2@TYUPR06MB6217.apcprd06.prod.outlook.com
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: 7aeb259086 ("ALSA: hda/conexant: Fix headset auto detect fail in cx8070 and SN6140")
Cc: stable@vger.kernel.org
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260713100329.306892-1-zhangheng@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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: <stable@vger.kernel.org>
Signed-off-by: Damien Laine <damien.laine@gmail.com>
Link: https://patch.msgid.link/20260712213708.1835469-1-damien.laine@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 <hicbowen@gmail.com>
Link: https://patch.msgid.link/20260710133409.16641-1-hicbowen@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260709161211.686498-1-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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: da3a7efff6 ("ALSA: usb-audio: Update for native DSD support quirks")
Reported-by: Salvador Blaya <tiniebla6@gmail.com>
Closes: https://lore.kernel.org/CAOdyq+qFaqCh=tK_wNnA64hv5pQuA1Y09ANxQ=xK8yR-t4mf9Q@mail.gmail.com
Tested-by: Salvador Blaya <tiniebla6@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260709095614.1418838-1-tiwai@suse.de
A fairly standard set of device specific fixes and quriks for new
devices, nothing too remarkable here.
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Merge tag 'asoc-fix-v7.2-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v7.2
A fairly standard set of device specific fixes and quriks for new
devices, nothing too remarkable here.
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 <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260709122211.615785-1-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260709121224.614350-1-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 <kailang@realtek.com>
Link: https://lore.kernel.org/34b990cb56914148ba02fa8e9d176479@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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: f78a97003b ("ASoC: tas2562: Update shutdown GPIO property")
Signed-off-by: Uday Khare <udaykhare77@gmail.com>
Link: https://patch.msgid.link/20260706153109.10953-1-udaykhare77@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently if the jack is forced to the microphone mode, it will report
as line in. Correct the report to microphone.
Fixes: fc918cbe87 ("ASoC: cs42l43: Add support for the cs42l43")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260708103430.1395207-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Ravi Hothi <ravi.hothi@oss.qualcomm.com> says:
This series adds audio support for the Qualcomm Eliza SoC. Eliza uses
the same LPASS macro codec hardware as SM8550, so the existing SM8550
fallback compatible is reused for all four macro codecs (RX, TX, VA,
WSA). The sound card is compatible with the SM8450 generation machine
driver.
Patch 1 documents the Eliza LPASS macro codec compatibles in the
dt-bindings for RX, TX, VA and WSA macros.
Patch 2 documents the Eliza sound card compatible.
Patch 3 adds the Eliza sound card entry to the machine driver.
Link: https://patch.msgid.link/20260703073434.2589657-1-ravi.hothi@oss.qualcomm.com
Add bindings for Eliza sound card, which is compatible with
the existing SM8450.
Signed-off-by: Ravi Hothi <ravi.hothi@oss.qualcomm.com>
Acked-by: Krzysztof Kozlowski <krzysztof.kozlowski@oss.qualcomm.com>
Link: https://patch.msgid.link/20260703073434.2589657-3-ravi.hothi@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <ravi.hothi@oss.qualcomm.com>
Link: https://patch.msgid.link/20260703073434.2589657-2-ravi.hothi@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <kshamendrakumarmishra@gmail.com>
Link: https://patch.msgid.link/DJSOWKFBTJH8.2Z7EPWX514QXM@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 <rohitsinha12043@gmail.com>
Link: https://patch.msgid.link/20260706211834.384022-1-rohitsinha12043@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 <agustinluzardo09@gmail.com>
Link: https://patch.msgid.link/20260705184227.113588-1-agustinluzardo09@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 <Vijendar.Mukunda@amd.com>
Reviewed-by: Mario Limonciello (AMD) <superm1@kernel.org>
Fixes: 7f91f012c1 ("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 <broonie@kernel.org>
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: 95e43a170b ("ASoC: amd: add Pink Sardine ACP PCI driver")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Mario Limonciello (AMD) <superm1@kernel.org>
Link: https://patch.msgid.link/20260707060130.2514138-3-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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: 4916283880 ("ASoC: amd: ps: add callback functions for acp pci driver pm ops")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Mario Limonciello (AMD) <superm1@kernel.org>
Link: https://patch.msgid.link/20260707060130.2514138-2-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <baojun.xu@ti.com>
Link: https://patch.msgid.link/20260702104429.1157-1-baojun.xu@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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: 6ae9ca9ce9 ("ASoC: meson: aiu: add i2s and spdif support")
Signed-off-by: Christian Hewitt <christianshewitt@gmail.com>
Reviewed-by: Martin Blumenstingl <martin.blumenstingl@googlemail.com>
Link: https://patch.msgid.link/20260627131205.808800-1-christianshewitt@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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: 40ba66a702 ("ALSA: hda - Add cache support for COEF read/write")
Cc: stable@vger.kernel.org
Signed-off-by: Xu Rao <raoxu@uniontech.com>
Link: https://patch.msgid.link/DB9023BF2920BA99+20260707132419.1731342-1-raoxu@uniontech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ALSA FireWire driver headers include <linux/firewire.h>, which
already provides struct ieee1394_device_id via
<linux/device-id/ieee1394.h>.
Drop the now-redundant direct <linux/mod_devicetable.h> includes
from the per-device headers.
Signed-off-by: Longlong Xia <xialonglong2025@163.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260706032639.273929-1-xialonglong2025@163.com
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 <matthew.schwartz@linux.dev>
Link: https://patch.msgid.link/20260707005615.52889-2-matthew.schwartz@linux.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 <matthew.schwartz@linux.dev>
Link: https://patch.msgid.link/20260707005615.52889-1-matthew.schwartz@linux.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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: 10d366a846 ("ASoC: amd: acp: Fix Kconfig dependencies for SND_SOC_ACPI_AMD_SDCA_QUIRKS")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Tested-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Syed Saba Kareem <Syed.SabaKareem@amd.com>
Reviewed-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://patch.msgid.link/20260703123314.147977-1-syed.sabakareem@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <shaotianze@outlook.com>
Link: https://patch.msgid.link/TYCPR01MB872223CB999D2099A5E0F58DC4F62@TYCPR01MB8722.jpnprd01.prod.outlook.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <malayarout91@gmail.com>
Link: https://patch.msgid.link/20260705171038.171966-1-malayarout91@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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: aa2f4addab ("ALSA: usb-audio: Set the value of potential sticky mixers to maximum")
Signed-off-by: Rong Zhang <i@rong.moe>
Link: https://patch.msgid.link/20260706-uac-sticky-channels-fix-v1-1-92741c538283@rong.moe
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CPUs when Intel SNC (Sub-NUMA Clustering) is enabled
(Reinette Chatre)
Signed-off-by: Ingo Molnar <mingo@kernel.org>
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Merge tag 'x86-urgent-2026-07-05' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip
Pull x86 fix from Ingo Molnar:
- Prevent OOB access in the resctrl code while offlining
CPUs when Intel SNC (Sub-NUMA Clustering) is enabled
(Reinette Chatre)
* tag 'x86-urgent-2026-07-05' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip:
x86,fs/resctrl: Prevent out-of-bounds access while offlining CPU when SNC enabled
- Fix a perf_event_attr::remove_on_exec bug for group
events (Taeyang Lee)
- Fix uprobes CALL emulation interaction with shadow stacks,
and add a testcase for this (David Windsor)
- Fix uprobes unregister bug (Jiri Olsa)
Signed-off-by: Ingo Molnar <mingo@kernel.org>
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Merge tag 'perf-urgent-2026-07-05' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip
Pull perf events fixes from Ingo Molnar:
- Fix a perf_event_attr::remove_on_exec bug for group events
(Taeyang Lee)
- Fix uprobes CALL emulation interaction with shadow stacks, and
add a testcase for this (David Windsor)
- Fix uprobes unregister bug (Jiri Olsa)
* tag 'perf-urgent-2026-07-05' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip:
uprobes/x86: Use proper mm_struct in __in_uprobe_trampoline
selftests/x86: Add shadow stack uprobe CALL test
x86/uprobes: Keep shadow stack in sync for emulated CALLs
perf/core: Detach event groups during remove_on_exec
- Fix a resource leak in the RISC-V imsic-early driver
(Haoxiang Li)
- Fix an OF node reference leak in the ARM gic-v3-its driver
(Yuho Choi)
- Fix a dangling handler function on module removal bug in
the TS-4800 ARM board irqchip driver (Qingshuang Fu)
Signed-off-by: Ingo Molnar <mingo@kernel.org>
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Merge tag 'irq-urgent-2026-07-05' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip
Pull irq fixes from Ingo Molnar:
"Misc irqchip driver fixes:
- Fix a resource leak in the RISC-V imsic-early driver (Haoxiang Li)
- Fix an OF node reference leak in the ARM gic-v3-its driver (Yuho
Choi)
- Fix a dangling handler function on module removal bug in the
TS-4800 ARM board irqchip driver (Qingshuang Fu)"
* tag 'irq-urgent-2026-07-05' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip:
irqchip/ts4800: Fix missing chained handler cleanup on remove
irqchip/gic-v3-its: Fix OF node reference leak
irqchip/irq-riscv-imsic-early: Fix fwnode leak on state setup failure
A standard set of driver-specific fixes and quirks accumulated since
the merge window:
ASoC:
- SOF: Sanity check to prevent OOB reads
- rsnd: Fix clock leak and double-disable issues with PM
- tas675x: Misc fixes for register fields, etc
- lpass-va-macro: Correct codec version for Qualcomm SC7280
- amd-yc: DMIC quirk for Alienware m15 R7 AMD
Others:
- us144mkii: Fix a UAF on disconnect and anchor list corruption
- HD-audio: Realtek quirks for HP models
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Merge tag 'sound-7.2-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A standard set of driver-specific fixes and quirks accumulated since
the merge window:
ASoC:
- SOF: Sanity check to prevent OOB reads
- rsnd: Fix clock leak and double-disable issues with PM
- tas675x: Misc fixes for register fields, etc
- lpass-va-macro: Correct codec version for Qualcomm SC7280
- amd-yc: DMIC quirk for Alienware m15 R7 AMD
Others:
- us144mkii: Fix a UAF on disconnect and anchor list corruption
- HD-audio: Realtek quirks for HP models"
* tag 'sound-7.2-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: rsnd: src: Add missing scu_supply clock to suspend/resume
Documentation: sound: tas675x: Fix temperature range and impedance documentation
ASoC: codecs: tas675x: Fix CHx temperature range register bit fields
ASoC: codecs: tas675x: use READ_ONCE for params to be used concurrently
ASoC: rsnd: adg: make rsnd_adg_clk_control() idempotent
ASoC: SOF: validate probe info element counts
ALSA: usx2y: us144mkii: fix work UAF on disconnect
ASoC: amd: yc: Add Alienware m15 R7 AMD to DMIC quirk table
ALSA: hda/realtek: Add quirk for HP Victus 16-e0xxx (88EE) to enable mute LED
MAINTAINERS: ASoC: SOF: add AMD reviewer for Sound Open Firmware
ASoC: codecs: lpass-va-macro: Fix LPASS Codec Version for SC7280
ALSA: us144mkii: capture_urb_complete: redundant usb_anchor_urb corrupts anchor list on each resubmission
A small set of fixes that came in since -rc1, we have one core fix for
shutting down target mode properly if the system suspends while it's
running plus a small set of fairly unremarkable device specific fixes.
There's also a couple of pure DT binding changes for Renesas SoCs, the
power domains one allows some SoCs to be correctly described with
existing code.
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Merge tag 'spi-fix-v7.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/spi
Pull spi fixes from Mark Brown:
"A small set of fixes that came in since -rc1, we have one core fix for
shutting down target mode properly if the system suspends while it's
running plus a small set of fairly unremarkable device specific fixes.
There's also a couple of pure DT binding changes for Renesas SoCs, the
power domains one allows some SoCs to be correctly described with
existing code"
* tag 'spi-fix-v7.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/spi:
spi: rzv2h-rspi: Fix DMA transfer error handling for signal interruption
spi: dt-bindings: snps,dw-apb-ssi: add 'power-domains' property
spi: dt-bindings: snps,dw-apb-ssi: drop superfluous RZ/N1 entry
spi: dw: use the correct error msg if request_irq() fails
spi: dw: fix first spi transfer with dma always fallback to PIO
spi: core: Abort active target transfer on controller suspend
spi: sh-msiof: abort transfers when reset times out
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 <pengpeng@iscas.ac.cn>
Link: https://patch.msgid.link/20260705084601.56400-1-pengpeng@iscas.ac.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar to commit 67c7381522 ("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 <dhs@frame.work>
Link: https://patch.msgid.link/20260705-framework-audio-ssid-0x0010-v1-1-fffc8996ac18@frame.work
Signed-off-by: Takashi Iwai <tiwai@suse.de>
