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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 and data registers, the device can
write interrupts to a bogus address and cause an IOMMU IO_PAGE_FAULT.
Clear the MSI Enable bit before reinitializing ACP hardware on system
resume. Keep runtime resume separate because firmware does not restore
PCI configuration on that path.
Fixes: c8212df7bc0f ("ASoC: amd: add acp6x pci driver pm ops")
Reported-by: Marek Borůvka <boruvka.marek@gmail.com>
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=222119
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Link: https://patch.msgid.link/20261013125425.3541206-1-mario.limonciello@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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HyeongJun An <sammiee5311@gmail.com> says:
In the es8316 thread Mark suggested checking at registration time for
controls whose register fields overlap [1]. Before adding such a check,
I wanted to see what it would find in the tree today.
A script expanded the control tables of every ASoC driver built by
allmodconfig and looked for two controls in one table whose register
fields overlap. It found 80 pairs in 33 drivers. Most are real bugs.
The rest are deliberate, where two names drive the same field, such as
the AC and DRC mode thresholds in sta32x.
This series fixes six that I could confirm against the vendor
datasheet. In each one, writing a control changes a different control,
and the bits it should own cannot be reached. Most date back to the
first version of the driver.
One more is left out on purpose. The cs42l56 ADC Mute Switch writes the
ADC invert bits. The real mute bits are also written by the
cs42l56_mute() callback on every stream start and stop, so moving the
control there would let the stream mute override it. I am not sure
whether you would rather move it or drop it.
I have none of this hardware, so these are checked against the
datasheets only. More fixes, checked against the register headers,
and the registration time check would follow.
[1] https://lore.kernel.org/all/0fefbfe1-021b-46b2-8945-15956e75e37f@sirena.org.uk/
Link: https://patch.msgid.link/20261008130953.2446692-1-sammiee5311@gmail.com
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The DAC and ADC 128x Oversampling Switches use bit 5 of R10 and R14.
The NAU8822 datasheet register map puts DACOS and ADCOS at bit 3 of
those registers.
In R14, bit 5 is inside HPF[6:4], which the High Pass Cut Off control
owns. So turning on ADC oversampling changes the high pass cut-off, and
the oversampling bit cannot be set. In R10, bit 5 is reserved, so the
DAC switch has no effect.
Use bit 3 for both.
Fixes: 1e3cb6c321be ("ASoC: nau8822: new codec driver")
Assisted-by: Claude:claude-opus-5-5
Signed-off-by: HyeongJun An <sammiee5311@gmail.com>
Link: https://patch.msgid.link/20261008130953.2446692-7-sammiee5311@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The DAC and ADC 128x Oversampling Switches use bit 5 of R10 and R14.
Table 66 of the WM8978 datasheet puts DACOSR128 and ADCOSR128 at bit 3
of those registers.
In R14, bit 5 is inside HPFCUT[6:4], which the High Pass Cut Off
control owns. So turning on ADC oversampling changes the high pass
cut-off, and the oversampling bit cannot be set. In R10, bit 5 is
reserved, so the DAC switch has no effect.
Use bit 3 for both.
Fixes: 823dba519122 ("ASoC: Fix broken bitfield definitions in WM8978")
Assisted-by: Claude:claude-opus-5-5
Signed-off-by: HyeongJun An <sammiee5311@gmail.com>
Link: https://patch.msgid.link/20261008130953.2446692-6-sammiee5311@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Register 0x31 holds two 4-bit fields. The datasheet puts SPK_GAIN_MIN
at bits 7:4 and DHT_VROT_PNT at bits 3:0. The Speaker DHT Min Volume
and Speaker DHT Rotation Volume controls both use bits 3:0.
So setting the minimum speaker gain changes the DHT rotation point, and
SPK_GAIN_MIN cannot be set.
Move Speaker DHT Min Volume to bits 7:4.
Fixes: ca2cd6bc6663 ("ASoC: Add max98371 codec driver")
Assisted-by: Claude:claude-opus-5-5
Signed-off-by: HyeongJun An <sammiee5311@gmail.com>
Link: https://patch.msgid.link/20261008130953.2446692-5-sammiee5311@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The DAC Zero Flag Polarity enum uses PCM3168A_DAC_ATSPDA_SHIFT, which
is bit 6 of register 70. That bit is ATSPDA, the attenuation speed,
which the DAC Volume Rate Multiplier enum already owns. The datasheet
puts ZREV, the zero flag polarity, at bit 0. The header defines
PCM3168A_DAC_ZREV_SHIFT for it, but nothing uses it.
So changing the zero flag polarity changes the attenuation speed, and
ZREV cannot be set.
Use PCM3168A_DAC_ZREV_SHIFT.
Fixes: a9b17a638af5 ("ASoC: pcm3168a: Add driver for pcm3168a codec")
Link: https://www.ti.com/lit/ds/symlink/pcm3168a.pdf
Assisted-by: Claude:claude-opus-5-5
Signed-off-by: HyeongJun An <sammiee5311@gmail.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20261008130953.2446692-4-sammiee5311@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The NG Type Switch enum selects between "Apply Specific" and "Apply
All", which is the NGALL bit. The datasheet puts NGALL at bit 7 of the
Noise Gate Control register. The enum uses bit 6, which is NG, the
noise gate enable owned by the NG Enable Switch. cs42l52.h names it
CS42L52_NG_ENABLE_SHIFT. cs42l56 has the same Noise Gate register and
puts its NG All Switch at bit 7.
So changing the noise gate type turns the noise gate on or off, and
NGALL cannot be set.
Use bit 7.
Fixes: dfe0f98b8d2e ("ASoC: Add support for CS42L52 Codec")
Link: https://statics.cirrus.com/pubs/proDatasheet/CS42L52_F2.pdf
Assisted-by: Claude:claude-opus-5-5
Signed-off-by: HyeongJun An <sammiee5311@gmail.com>
Link: https://patch.msgid.link/20261008130953.2446692-3-sammiee5311@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The Limiter Cushion control writes bits 4:2 of the ALC Thresholds
register. Those bits are ALCMIN, which the ALC MIN control owns. The
datasheet puts CUSH at bits 4:2 of the Limit Thresholds Control
register, next to LMAX, which the Limiter MAX control already uses.
So setting the limiter cushion changes the ALC minimum threshold, and
the cushion itself cannot be set.
Use CS42L56_LIM_THRESHOLD_CTL.
Fixes: 272b5edd3b8f ("ASoC: Add support for CS42L56 CODEC")
Link: https://statics.cirrus.com/pubs/proDatasheet/CS42L56_F2.pdf
Assisted-by: Claude:claude-opus-5-5
Signed-off-by: HyeongJun An <sammiee5311@gmail.com>
Link: https://patch.msgid.link/20261008130953.2446692-2-sammiee5311@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The interrupt handler uses a logical AND to check whether SPDIFRX
was in the SYNC state before clearing errors. Since both SYNC (1)
and ENABLE (3) are nonzero, this also treats ENABLE as SYNC (and
any other non-zero value that might be added).
The logical && basically just converts this to check for != 0.
So errors occurring while SPDIFRX is enabled cause the driver
to retry synch instead of disconnecting the PCM stream.
Fix is to change the logical && to equality (==) and only retry
when exactly in sync state.
Fixes: 0c93c291321f2 ("ASoC: stm32: spdifrx: retry synchronization in sync state")
Signed-off-by: Avi Weiss <thnkslprpt@gmail.com>
Reviewed-by: Olivier Moysan <olivier.moysan@foss.st.com>
Link: https://patch.msgid.link/20261011035459.2289403-1-thnkslprpt@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Commit cfca1637bc2b ("ASoC: Intel: avs: Unprepare a stream when XRUN
occurs") resets the HDA stream when ->prepare() is called in the XRUN
state. User space does not always recover that way. PipeWire's ALSA
backend recovers from an underrun with snd_pcm_drop(), snd_pcm_prepare()
and a new start, so ->prepare() runs in the SETUP state,
hdac_stream->prepared is still set and the reset is skipped. Every
recovery attempt then fails the same way:
spa.alsa: hw:avsmax98927,1p: snd_pcm_avail after recover: Broken pipe
On a Google Pixelbook (Kaby Lake, max98927 speakers) this happens after
an occasional underrun during normal playback and silences the speakers
for one to four minutes, until the PCM is closed. It can be provoked by
stopping PipeWire with SIGSTOP for 0.5 s during playback: in one run
that single underrun led to 1297 failed recoveries over two minutes,
until the sink was suspended. A plain alsa-lib program that recovers
with snd_pcm_drop() and snd_pcm_prepare() hits the same loop, while
snd_pcm_prepare() straight from XRUN recovers at once.
Skip the reset only when the stream is already PREPARED, the repeated
->prepare() case the check exists for, and reset it otherwise. With
this change both kinds of recovery succeed with a single restart.
Fixes: cfca1637bc2b ("ASoC: Intel: avs: Unprepare a stream when XRUN occurs")
Assisted-by: Claude:claude-opus-5-5
Signed-off-by: Sergey Tiraspolsky <stiraspo@gmail.com>
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20261010103753.16969-1-stiraspo@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Protect IPC RX and FW state handler list unregister/dispatch paths with
client_event_handler_mutex to match list registration and locking
comments.
Fixes: 5c19da34df02 ("ASoC: SOF: Use guard()/scoped_guard() for mutex locks where it makes sense")
Cc: stable@vger.kernel.org
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20261008090143.15158-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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When ch_count is smaller than blob->alh_cfg.device_count (e.g. a mono
capture stream, such as a SmartMic, aggregated across multiple SDW
links), step = ch_count / blob->alh_cfg.device_count truncates to 0.
The subsequent GENMASK(step - 1, 0) then underflows step - 1 (u32) to
UINT_MAX, causing GENMASK()'s internal shift to use an out-of-bounds
exponent:
UBSAN: shift-out-of-bounds in sound/soc/sof/ipc4-topology.c:2363:13
shift exponent 18446744069414584384 is too large for 64-bit type 'long unsigned int'
Treat ch_count < device_count the same way as the existing "channels
equal to output reference channels" case: apply the same ch_mask to
every aggregated device instead of trying to split channels that
can't be evenly divided.
Fixes: 0390a102cc18 ("ASoC: SOF: ipc4-topology: use different channel mask for each sdw amp feedback")
Cc: stable@vger.kernel.org
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20261008090130.15135-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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ASoC: Fixes for v7.3
A fairly small collection of fixes and a few device quirks, nothing
especially remarkable unless you have the affected system.
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This adds the following SoundWire configuration:
cs42l43b link 0 UID 1
cs35l57 link 0 UID 0
cs35l57 link 0 UID 1
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Link: https://patch.msgid.link/20261007130857.102432-1-simont@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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A BIOS for this SKU has been observed with an incorrect
acp-audio-config-flag, add the override so that it is forced to the
correct value.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Link: https://patch.msgid.link/20261007130835.102325-1-simont@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Richard Fitzgerald <rf@opensource.cirrus.com> says:
The amp feedback path (AEC) was incorrectly using SoundWire DP3.
The firmware outputs SDCA OT25 feedback on DP4. DP3 is reserved for
SDCA companion amp.
This series adds a DAI for OT25 and switches the sdw machine driver
to use the new DAI for the feedback path.
Link: https://patch.msgid.link/20261005155235.1386525-1-rf@opensource.cirrus.com
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Change the CS35L56/57/62/63 entries in codec_info_list[] to use
the OT25 DAI for amp feedback.
The AMP feedback was incorrectly using a DAI connected to DP3,
but that is reserved for SDCA companion amp. The correct amp
output is DP4, which is the SDCA OT25 feedback output.
Fixes: 898cd43bde307 ("ASoC: intel: sof_sdw: Add support for CS35L63 into machine driver")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20261005155235.1386525-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add a DAI for the SDCA OT25 stream on SoundWire DP4.
The SDCA-defined OT25 stream is an amp reference feedback, typically used
for AEC. The firmware outputs this on DP4. The firmware owns all control
registers for this stream so there are no mixer controls or any other
configuration options.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20261005155235.1386525-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The probe function puts the device into the runtime PM active state with
only the "iis" bus clock enabled, but it also sets priv->op_clk to the
parent of the RCLK_SRC mux without enabling it. The runtime PM suspend
callback disables both clocks, so the first runtime suspend after probe
disables op_clk once more than it was enabled.
On Exynos5250 Arndale both clocks resolve to the same EXYNOS_I2S_BUS gate,
so this triggers the following warning during boot:
WARNING: drivers/clk/clk.c:1270 at clk_core_disable+0xe4/0x23c, CPU#0: kworker/u8:3/45
i2s_bus already disabled
...
Workqueue: pm pm_runtime_work
Call trace:
...
clk_core_disable from clk_disable+0x28/0x34
clk_disable from i2s_runtime_suspend+0x58/0x68 [snd_soc_i2s]
i2s_runtime_suspend [snd_soc_i2s] from genpd_runtime_suspend+0xc4/0x298
genpd_runtime_suspend from __rpm_callback+0x48/0x18c
__rpm_callback from rpm_callback+0x4c/0x50
rpm_callback from rpm_suspend+0xf4/0x718
rpm_suspend from pm_runtime_work+0x9c/0xa0
...
Fix this by enabling op_clk in probe right after getting it, so the clock
state matches the runtime PM active state, and by disabling it in remove.
Fixes: 48279c53fd1d ("ASoC: samsung: i2s: Prevent external abort on exynos5433 I2S1 access")
Assisted-by: Claude Opus 5.5 <noreply@anthropic.com>
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Link: https://patch.msgid.link/20261002092116.585466-1-m.szyprowski@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The chip has strange power-on reset values for the registers that
control the output ASRC and DAC source muxes - the values are actually
marked "Reserved" in the datasheet [1]. This means that until the
underlying kcontrols are modified, the codec will not function as
expected.
The registers in question are:
register / field reset value valid values
ASRCO_SOURCE_0_1 0x10
ASRC_OUT_SOURCE0 0 4 .. 16
ASRC_OUT_SOURCE1 1 4 .. 16
ASRCO_SOURCE_2_3 0x10
ASRC_OUT_SOURCE2 2 4 .. 16
ASRC_OUT_SOURCE3 3 4 .. 16
DAC_SOURCE_0_1 0x10
DAC_SOURCE_0 0 12 .. 13
DAC_SOURCE_1 1 12 .. 13
Set some sane defaults during the driver probe:
ADC Decimator0 -> Output ASRC0
ADC Decimator1 -> Output ASRC1
ADC Decimator2 -> Output ASRC2
ADC Decimator3 -> Output ASRC3
Input ASRC0 -> DAC0
Input ASRC1 -> DAC1
With this, the chip will behave as advertised by the default kcontrol
state exposed by the driver.
[1] https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1372.pdf
Fixes: 6cd4c6459e47 ("ASoC: Add ADAU1372 audio CODEC support")
Signed-off-by: Alvin Šipraga <alvin.sipraga@analog.com>
Acked-by: Nuno Sá <nuno.sa@analog.com>
Link: https://patch.msgid.link/20261006-asoc-adau1372-fixes-v1-2-599cc17fb5f6@analog.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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gpiod_set_value() emits a WARN when the GPIO provider indicates that it
can sleep, as is typical for I2C controlled GPIO expanders. Since the
powerdown GPIO is only manipulated in process context, use the _cansleep
variant instead. This lets the driver use sleepy GPIOs to power off the
chip.
Fixes: 6cd4c6459e47 ("ASoC: Add ADAU1372 audio CODEC support")
Signed-off-by: Alvin Šipraga <alvin.sipraga@analog.com>
Acked-by: Nuno Sá <nuno.sa@analog.com>
Link: https://patch.msgid.link/20261006-asoc-adau1372-fixes-v1-1-599cc17fb5f6@analog.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The Acer Aspire Go 15 (AG15-21P, board MDC Herbag2_MDU) has two PDM
DMICs connected to the ACP. The BIOS does not set the AcpDmicConnected
_DSD property, so the acp6x machine driver never registers the DMIC
card. The existing Herbag_MDU entry does not match this board name, so
add a DMI quirk for it.
Tested on BIOS V1.11: the acp6x DMIC capture device is registered and
records audio on both channels.
Cc: stable@vger.kernel.org
Signed-off-by: Michael Garland <minimelkav@gmail.com>
Link: https://patch.msgid.link/20261005-acp6x-herbag2-quirk-v1-1-a1f2683d758f@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The mute LED on the HP Pavilion Laptop 15-eg3xxx (PCI SSID 103c:8bfd,
Realtek ALC287) does not work because the machine has no entry in the
quirk table. No fixup is applied, so the mute LED is never driven, even
though the mute function itself works.
The LED is controlled via GPIO pin 4 (mask 0x10), which is exactly what
ALC287_FIXUP_HP_GPIO_LED configures. This was verified on the hardware
with hda-verb: setting the GPIO data bit lights the mute LED, and
clearing it turns the LED off.
hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x10
hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION 0x10
hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x10 # LED on
hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x00 # LED off
Add a quirk entry for 103c:8bfd using ALC287_FIXUP_HP_GPIO_LED.
Signed-off-by: Jaidheer Sirigineedi <jaidheer7@gmail.com>
Link: https://patch.msgid.link/20261002185951.33855-1-jaidheer7@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some Lenovo BIOSes (e.g. Legion Pro 7 16IRX9H, SSID 17aa:38cd) store
valid V1 calibration data in the CALI_DATA EFI variable but leave the
TimeStamp and CRC fields zeroed. The driver rejects such data, so the
amplifiers always run with default parameters on these machines.
The rejection was silent until
commit 4fe238513407 ("ALSA: hda/tas2781: Move and unified the
calibrated-data getting function for SPI and I2C into the tas2781_hda lib")
turned it into "V1 CRC error".
The stored values are consistent (R0 = 3.82/3.83 ohm, 1/R0 matching
within 1e-4), so accept V1 data when both TimeStamp and CRC are zero
and the first device entry is non-zero, and emit a warning.
Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Cc: stable@vger.kernel.org
Signed-off-by: Mohamed Yassine Jebabli <med.jebabli@gmail.com>
Link: https://patch.msgid.link/20261001-tas2781-v1-nocrc-v1-1-ff0c1032ce76@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Commit e576e7843c0d ("ALSA: ctxfi: Simplify dao_clear_{left,right}_input() functions")
centralized the DAO input mapper clearing logic in
dao_clear_input(), but omitted the kfree() calls.
It removes the mapper entries from the list and sets their pointers
to NULL without freeing the allocated array. This leaks memory each
time dao_clear_input() is called.
Store the base pointer of the allocation before clearing the mapper
entries and free the array after removing entries from the list.
Fixes: e576e7843c0d ("ALSA: ctxfi: Simplify dao_clear_{left,right}_input() functions")
Cc: stable@vger.kernel.org
Signed-off-by: Harin Lee <me@harin.net>
Link: https://patch.msgid.link/20261001144501.3050667-2-me@harin.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix headphone jack detection for some devices
Signed-off-by: Eckhart Mohr <e.mohr@tuxedocomputers.com>
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com>
Link: https://patch.msgid.link/20261001183403.103990-1-wse@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Cambridge Audio streamer advertises the UAC2 RAW_DATA format on
altsetting 2 of its playback interface, but without the DSD_RAW quirk
flag the driver exposes it only as a SPECIAL format that cannot be
used.
Add QUIRK_FLAG_DSD_RAW for the device (USB ID 22e8:ca10), so the
altsetting is exposed as DSD_U32_BE, enabling native DSD64/128/256
playback.
Tested with: gst-launch-1.0 filesrc location=file.dsf ! \
avdemux_dsf ! dsdconvert ! \
alsasink device=hw:2,0
Signed-off-by: Matus Gajdos <matuszpd@gmail.com>
Link: https://patch.msgid.link/20261002125237.82188-1-matuszpd@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ASoC: Fixes for v7.3
A bigger collection of fixes than usual due to your vacation but nothing
hugely remarkable here, just fairly standard quirks and driver specific
bugfixes.
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We used the auto-cleanup via snd_card_unref() for errors at probe of
ice1712 driver, but this may be problematic in a subtle way when an
error happens at snd_card_register(); since snd_card_unref() skips the
snd_card_disconnect() call, it may miss some resource clearance.
For fixing the issue, use the new __free(snd_card_free) instead, which
does call snd_card_free() explicitly at errors for avoiding such a
pitfall.
Fixes: d736eba9c453 ("ALSA: ice1712: Fix the card leak at probe error with the auto-cleanup")
Link: https://patch.msgid.link/20261001151039.592033-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The ALC235 (0x10ec0235) is registered as the ALC255 codec variant in
patch_alc269(), but the headset mode functions were missing a case
for it, so the CTIA/OMTP jack type detection and routing never ran
on this codec. As a result, a plugged-in headset is detected (jack
presence works) but the microphone signal is never routed, leaving
the capture stream permanently silent.
Verified on an ASUS M5451GA laptop (HDA:10ec0235,10431814): with the
ALC255 coef sequence applied, the headset microphone works correctly.
Signed-off-by: Zeyu Li <13776528859@163.com>
Link: https://patch.msgid.link/20260926140336.27896-1-13776528859@163.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Acer Swift SFG16-72 (1025:173b) uses the ALC245 codec, but its
mic-mute LED is not enabled by default. Apply the existing
ALC245_FIXUP_ACER_MICMUTE_LED quirk to enable the mic-mute LED.
Assisted-by: LLM
Cc: stable@vger.kernel.org
Signed-off-by: Jakub Gawlik <jakubgawlik13@gmail.com>
Link: https://patch.msgid.link/20260930145839.374762-1-jakubgawlik13@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The HP ENVY x360 2-in-1 Laptop 15-ey0xxx (SSID 103c:8a31, ALC245 with
CS35L41 I2C amps) currently only gets ALC287_FIXUP_CS35L41_I2C_2, so the
mute and micmute LEDs on the keyboard never light up.
On this machine the mute LED is driven by COEF 0x0b bits 2-3 and the
micmute LED by GPIO 0x04 (active low), which is exactly what
ALC245_FIXUP_HP_X360_MUTE_LEDS sets up. Switch the quirk to
ALC287_FIXUP_CS35L41_I2C_2_HP_MUTE_LEDS, which keeps the CS35L41 setup
and chains to that LED fixup.
Verified by driving the COEF and GPIO values by hand with hda-verb,
following Master Playback Switch and Mic ACP LED Capture Switch.
Assisted-by: Claude:claude-opus-5-5
Signed-off-by: Harrison Lucas <tiiowsalt@gmail.com>
Link: https://patch.msgid.link/20260929174348.167537-1-tiiowsalt@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The HP Pavilion 14-dv1001TU (subsystem ID 103c:8987) uses a Realtek
ALC287 codec. The mute LED on this machine is controlled by GPIO bit
0x10, which was verified via hda-verb, but the subsystem ID is missing
from the quirk table. As a result, the ALC287_FIXUP_HP_GPIO_LED fixup
that drives GPIO 0x10 is never applied and the mute LED stays off even
though the associated LED class device toggles correctly.
Add the missing SND_PCI_QUIRK entry to map this subsystem ID to
ALC287_FIXUP_HP_GPIO_LED so the mute LED works as expected.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=221810
Signed-off-by: renpanpan <renpanpan@kylinos.cn>
Link: https://patch.msgid.link/20260930065504.24151-1-rppxj178543@163.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Commit 4b490e0d103c ("ALSA: ctxfi: Add ADC helper functions for GPIO")
moved the GPIO_DATA read into hw_adc_stop(), but the Titanium HD
(SB1270) setup in hw_adc_init() uses the local data variable holding
the GPIO_CTRL value.
Writing this value to GPIO_DATA sets the output mute bits. Playback
runs without errors, but the outputs stay muted.
Re-read GPIO_DATA before setting the speed mode bits. Also, add the
delay after writing the mode bits, as in the original sequence.
On affected kernels, switching the output selection was reported to
restore clean playback.
Fixes: 4b490e0d103c ("ALSA: ctxfi: Add ADC helper functions for GPIO")
Cc: stable@vger.kernel.org
Reported-by: Dany Patoine <dany_patoine@hotmail.com>
Closes: https://lore.kernel.org/linux-sound/DM6PR02MB5433A42E81FEFAC400DD1905EA8B2@DM6PR02MB5433.namprd02.prod.outlook.com/
Signed-off-by: Harin Lee <me@harin.net>
Link: https://patch.msgid.link/20260930115824.739412-1-me@harin.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Faaris reported a problem with Audient EVO4 device and it turned out
to be a regression by the recent fix commit 87a6f2fa6e6c ("ALSA:
usb-audio: Propagate write errors in generic mixer put callbacks").
It exhibited that the hardware gives errors at accessing the mixer
unit 10 on certain channels, and the change above made it a fatal
error.
We had already a quirk for Audient iD14 to work around such errors
from the mixer controls, and we can simply apply the same for EVO4.
OTOH, it's highly possible that other Audient devices suffer from the
same issue; they must be using similar firmware, after all.
So, in this patch, we apply the quirk generically to all devices with
the vendor ID Audient (2708), instead. Ignoring the control error
isn't usually less critical than overreaction to the firmware
misbehavior.
Fixes: 87a6f2fa6e6c ("ALSA: usb-audio: Propagate write errors in generic mixer put callbacks")
Reported-and-tested-by: Faaris Ansari <faarisansari@googlemail.com>
Closes: https://lore.kernel.org/CANBVYRCL=8QdLxGg4S6qrahrFtwJxhv-aSGpW7-1S=+iOe4ZGA@mail.gmail.com
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20260929144132.1521617-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Jan reported that a few Behringer devices became broken since the
recent optimization to avoid usb_string() call at probe time at the
commit b364a0d23cae ("ALSA: usb-audio: Use strings in struct usb_dev
for manufacturer & co"). Interestingly, the devices seem requiring
the explicit descriptor read at probing time, and the optimization
above dropped it.
There is already a boot quirk for another model, Behringer CM1A
(1397:1234), that adds a device descriptor read, and this seems
working for them, too. As the quirk is safe and cheap, just apply the
same boot quirk to all Behringer devices for avoiding the pitfall
again.
Fixes: b364a0d23cae ("ALSA: usb-audio: Use strings in struct usb_dev for manufacturer & co")
Reported-by: Jan Lentfer <jan.lentfer@web.de>
Closes: https://lore.kernel.org/e7087d42-5e74-4d85-b1c5-b11eff235d41@web.de
Link: https://patch.msgid.link/20260929122938.1471867-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Commit 902f497a1ff5 ("ASoC: codecs: lpass-wsa-macro: remove useless
gain read/write sequence") removed the read and write of the digital
volume register in wsa_macro_enable_interpolator(), on the grounds that
writing back the value just read does nothing. The comment above it,
"apply gain after int clk is enabled", was left in place.
On the Dell XPS 13 9345 (X1E80100, four WSA8845 amplifiers on two WSA
macros) the write does something: a volume change made while the path
is idle does not take effect when playback starts. Lowering the digital
volume from 81 to 63 with nothing playing, then playing a test tone,
gave about the same level as before the change. With the rewrite
restored, lowering it from 81 to 69 while idle played audibly quieter,
and restoring 81 while idle brought the level back. Changes made during
playback take effect with or without the rewrite.
The register already holds the new value when this happens. Without the
rewrite, after an idle change from 69 to 81, playback stayed at the old
level while the register, read from the hardware through /dev/mem, held
the new one (0xfd). Writing that same value back through /dev/mem
brought the level up at once.
This is the behaviour described in commit 46188db080bd ("ASoC: codecs:
lpass-wsa-macro: fix compander volume hack"): "the volume registers
still need to be written after enabling clocks in order for any prior
updates to take effect." The value read comes from the register cache,
so the write pushes the last requested volume to the hardware once its
clock runs.
Restore the rewrite in the interpolator's POST_PMU event only. The mix
path event removed later in the same series is not brought back.
Fixes: 902f497a1ff5 ("ASoC: codecs: lpass-wsa-macro: remove useless gain read/write sequence")
Cc: stable@vger.kernel.org
Assisted-by: Claude:claude-opus-5-5
Signed-off-by: Liviu Nicoara <lnicoara@thinkoid.org>
Reviewed-by: Johan Hovold <johan@kernel.org>
Link: https://patch.msgid.link/20260929115755.6096-1-lnicoara@thinkoid.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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The ASUS ExpertBook PM3406CHA uses an ALC256 codec with subsystem
ID 1043:1664. On Linux boot, the internal speakers are silent even
though playback is routed to the codec and the speaker pin is
configured for output.
The codec starts with coefficient 0x10 set to 0x1d20. Writing 0x7f20
to coefficient 0x10 immediately restores internal speaker output.
This was reproduced on the affected hardware from a fresh boot using
hda-verb. Reducing a larger initialization sequence showed that this
coefficient write alone is sufficient; no GPIO, EAPD, pin-control, or
other coefficient changes are required. After applying the coefficient
in userspace, speaker output also remains functional across suspend
and resume.
Add a machine-specific HDA fixup that writes the required coefficient.
The fixup has been compile-tested, but could not be boot-tested because
the affected hardware is no longer available.
Closes: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/2162718
Signed-off-by: Muhtasham Nawr al-Mahmud <muhtaseem2005@gmail.com>
Link: https://patch.msgid.link/20260924144512.3989-1-muhtaseem2005@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add ALC225_FIXUP_HP_AIO_SPK to enable the internal speaker on
HP Pavilion AiO 24 (103c:85ba).
Link: https://bugzilla.kernel.org/show_bug.cgi?id=221691
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260929091109.400313-1-zhangheng@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The HP Laptop 15-fd2xxx (PCI SSID 103c:8e43, Realtek ALC236) has a
speaker mute LED on the F5 key and a microphone mute LED on the F8 key.
Neither lights up, because the model has no quirk entry and only the
generic HP vendor fallback (ALC269_FIXUP_HP_MUTE_LED) is applied.
Probing the codec with hda-verb showed how the LEDs are wired:
- mute LED: COEF index 0x07, bit 0, on NID 0x20 (set = LED on)
- mic mute LED: codec GPIO0, active low (driven low = LED on)
Driving GPIO1 and GPIO2 did not light the mute LED. This wiring is
exactly what the existing ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO fixup
implements (COEF 0x07 bit 0 for the mute LED, GPIO0 with inverted
polarity for the micmute LED), so use it for this model. The sibling
HP Laptop 15-fd0xxx (103c:8dd7) already uses the same fixup.
Tested on this machine running Ubuntu 7.0.0-34-generic (SOF driver, the
quirk built into snd-hda-codec-alc269 from Ubuntu's 7.0.0-34.34 source
tree): the hda::mute and hda::micmute LEDs are created, follow the
Master mute state and the Dmic0 capture switch respectively, and the
F5 and F8 keys toggle them correctly. Muting the microphone also
silences the capture stream. The LEDs also kept the correct state across
an s2idle suspend/resume cycle (speaker muted in one run, microphone in
another) and still responded to mute changes afterwards; long suspends
were not tested. The desktop session was SwayFX (wlroots) with PipeWire,
which only changes the mute state through the normal mixer controls; no
other desktop was tried. Not tested on a mainline kernel tree directly;
the change is only the quirk table entry.
AI assistance: this patch, the hardware analysis behind it (probing the
codec with hda-verb and choosing the existing fixup) and this commit
message were written by Claude Code (Anthropic, model claude-sonnet-5-5).
The submitter ran the privileged commands and reboots and reported the
LED behavior; Claude Code ran the tests and read the kernel logs.
Assisted-by: Claude:claude-sonnet-5-5
Signed-off-by: Preston Lam <plamlam99@gmail.com>
Link: https://patch.msgid.link/20260928231621.1672704-1-plamlam99@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add ASUS Zenbook S14 UM5406GA (Strix Point, ACP 7.0) to
acp70_acpi_flag_override_table so that the broken BIOS
acp-audio-config-flag is ignored and the default SoundWire
enumeration path is used. Without this, no ASoC machine driver
matches and internal speakers/mic fail to probe.
Reported-by: Jannick Tobler <business@jtobler.net>
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=222009
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260928083925.266471-1-zhangheng@kylinos.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
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The HP Pavilion 15-eh3174nw with an ALC287 codec requires the
ALC287_FIXUP_HP_GPIO_LED fixup for its mute LED to work correctly.
Add subsystem ID 0x103c:0x8bc7 to the quirk table to apply the
existing fixup.
Tested on an HP Pavilion 15-eh3174nw; the mute LED now follows the
speaker mute state.
The HP Laptop 15-dw1xxx with an ALC236 codec requires the
ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO fixup for its mute LED to work correctly.
Add subsystem ID 0x103c:0x85f1 to the quirk table to apply the existing fixup.
Tested on an HP Laptop 15-dw1xxx; the mute LED now follows the speaker
mute state.
Signed-off-by: Bartosz Brom <bartosz.brom06@gmail.com>
Link: https://patch.msgid.link/20260928134335.1875-1-bartosz.brom06@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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A variant of the Lenovo Yoga Slim 7 exists on Lunar Lake that
doesn't have a physical jack socket. Cirrus codecs still use match
tables on LNL but the match table makes no distinction between a system
with or without a jack. This leads to the system failing probe as the
jack in the topology file is not matched by a DAI link as the machine
driver is constructed.
Add a new quirk to block using match tables for specific devices, and
rely on function topologies which can handle this difference seamlessly.
This is slightly preferred over moving everything to function topologies
due to the lower chance of causing regressions on other systems.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260928105610.242687-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The HP Victus 15-fb2xxx (PCI SSID 103c:8c2f) with the ALC245 codec
does not turn on the mute LED when audio is muted.
Add a quirk entry to enable it.
Reported-by: erph.briones27@gmail.com
Link: https://bugzilla.kernel.org/show_bug.cgi?id=221946
Signed-off-by: Krish Gulati <krishgulati7@gmail.com>
Tested-by: Raphael Briones <erph.briones27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260921131728.8999-1-krishgulati7@gmail.com
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The Acer Nitro AN515-45 requires ALC2XX_FIXUP_HEADSET_MIC to make the
headset microphone work properly.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=221925
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260921124450.639601-2-zhangheng@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix Bass Speaker volume control on Lenovo ThinkBook G8+ by adding
quirk 17aa:38dc with ALC285_FIXUP_SPEAKER2_TO_DAC1.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=221799
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260921124450.639601-1-zhangheng@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It lost its headset microphone functionality, and this patch will restore it.
Fixes: 97272a5704bf ("ALSA: hda/realtek - Fixed Headphone noise issue for Dell QCM1255")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/3c251e40a2944ae4ae479e4c2e4e63a4@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This workaround was originally added for Tegra194, but no longer gets
applied for it. Commit 615d43540043 ("ALSA: hda/tegra: fix tegra-hda on
tegra30 soc") changed the guard to tegra30-hda, which worked because all
affected archs had the fallback compatible. However, commit 7f0ea5acfc19
("arm64: tegra: Use correct compatible string for Tegra194 HDA") removed
the fallback compatible for Tegra194, causing this to break.
Fixes: 7f0ea5acfc19 ("arm64: tegra: Use correct compatible string for Tegra194 HDA")
Cc: stable@vger.kernel.org
Assisted-by: LLM
Signed-off-by: Aaron Kling <webgeek1234@gmail.com>
Link: https://patch.msgid.link/20260926-tegra194-hda-sdo-v1-1-f7a636a07a58@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The DAI ID is a hardware port ID and is not necessarily the index of
the corresponding entry in variant->dai_driver.
When configuring a DAI with LPAIF_I2SCTL_MODE_QUAD01, the probe code
currently uses dai_id to index variant->dai_driver. This is incorrect
for platforms where DAI IDs are sparse. For example, the IPQ806x MI2S
DAI has ID 4 while it is the only entry in the DAI driver table. If
that DAI is configured for QUAD01 and probe reaches this branch, the
code writes beyond that single entry via dai_driver[4] instead of
updating the DAI being processed.
Use the loop index i when updating the current DAI's playback channel
constraints, while retaining dai_id for indexing the hardware-port
specific playback SD-line mode array.
This fixes the incorrect DAI table access for platforms where the DAI
ID does not match its position in the driver table.
Fixes: c223f41c1a52 ("ASoC: qcom: Add four speaker support on MI2S secondary")
Cc: stable@vger.kernel.org
Signed-off-by: Zimeng Li <me@lizi.moe>
Assisted-by: LLM-assisted source analysis
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260927010911.51980-1-me@lizi.moe
Signed-off-by: Mark Brown <broonie@kernel.org>
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setup_card() registers the input device, whose name and phys point into
struct snd_usb_caiaqdev, and then goes on to snd_card_register() and
snd_usb_caiaq_control_init(). If either fails, snd_probe() calls
snd_card_free(), which frees the device state but leaves the input
device registered: card_free() only clears the pointer, and the input
device is unregistered from snd_disconnect() alone. Reading its "uevent"
attribute afterwards dereferences freed memory:
BUG: KASAN: slab-use-after-free in string+0x4a9/0x4f0
Read of size 1 at addr ffff8880208f5043 by task caiaq/4967
add_uevent_var+0x183/0x3a0
input_dev_uevent+0x162/0x900
dev_uevent+0x2f1/0x870
uevent_show+0x1ca/0x3a0
...
Unregister the input device on the probe error path, as
snd_disconnect() does. snd_usb_caiaq_input_disconnect() is a no-op when
no input device was registered.
Reproduced with a raw-gadget Audio Kontrol 1 and a temporary hack that
makes snd_usb_caiaq_control_init() fail.
Fixes: 28abd224db4a ("ALSA: caiaq: Handle probe errors properly")
Cc: stable@vger.kernel.org
Reported-by: syzbot+2a123f6269da57ffefaa@syzkaller.appspotmail.com
Closes: https://syzkaller.appspot.com/bug?extid=2a123f6269da57ffefaa
Signed-off-by: Nguyen Ngoc Thang <ngocthang2710.1999@gmail.com>
Link: https://patch.msgid.link/20260925170717.22462-1-ngocthang2710.1999@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Mario Limonciello
Mark Brown
HyeongJun An
Avi Weiss
Sergey Tiraspolsky
Peter Ujfalusi
Takashi Iwai
Simon Trimmer
Richard Fitzgerald
Marek Szyprowski
Alvin Šipraga
Michael Garland
Jaidheer Sirigineedi
Mohamed Yassine Jebabli
Harin Lee
Eckhart Mohr
Matus Gajdos
Zeyu Li
Jakub Gawlik
Harrison Lucas
renpanpan
Liviu Nicoara
Muhtasham Nawr al-Mahmud
Zhang Heng
Preston Lam
Bartosz Brom
Charles Keepax
Krish Gulati
Kailang Yang
Aaron Kling
Zimeng Li
Nguyen Ngoc Thang