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: c8212df7bc ("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>
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
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: 1e3cb6c321 ("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>
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: 823dba5191 ("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>
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: ca2cd6bc66 ("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>
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: a9b17a638a ("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>
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: dfe0f98b8d ("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>
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: 272b5edd3b ("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>
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: 0c93c29132 ("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>
Commit cfca1637bc ("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: cfca1637bc ("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>
Protect IPC RX and FW state handler list unregister/dispatch paths with
client_event_handler_mutex to match list registration and locking
comments.
Fixes: 5c19da34df ("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>
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: 0390a102cc ("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>
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>
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
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: 898cd43bde ("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>
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>
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: 48279c53fd ("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>
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: 6cd4c6459e ("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>
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: 6cd4c6459e ("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>
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>
Commit 902f497a1f ("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 46188db080 ("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: 902f497a1f ("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>
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>
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>
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: c223f41c1a ("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>
Some Chuwi Hi8 tablets with model number CWI513 ship with a BIOS
(American Megatrends 5.11, dated 05/21/2016) whose DMI sys_vendor and
product_name are both "Default string". The tested unit is labelled
just "Hi8", but its model number is the same as the existing "Chuwi Hi8
Pro (CWI513)" entry, and it needs the same quirks. That entry matches on
sys_vendor "Hampoo" and product_name "X1D3_C806N", so it never applies
to these units.
Without its quirks the internal speaker, whose amplifier has a
differential input wired across LOUT left and right, cancels to a faint
distorted residual for any centred (L == R) content, and the headphone
channels are swapped.
Add an entry matching board_vendor "Hampoo" and board_name "Cherry Trail
CR", plus the BIOS date since those two are too generic on their own.
drivers/platform/x86/touchscreen_dmi.c already matches the Chuwi HiBook
the same way.
The quirks are those of the existing Hi8 Pro entry, except for the
over-current detection scale factor: 1.0 instead of the 0.75 from
BYT_RT5651_DEFAULT_QUIRKS. With 0.75 and a CTIA headset (inline mic and
one button), loud speech into the mic, and headphone playback coupling
onto the mic line, were reported as spurious KEY_PLAYPAUSE presses, one
of them held for a whole 8 second speech window. With 1.0 the same test
gave no spurious events, and 3 real button presses were reported as
exactly 3 press/release pairs.
Tested on a unit with the generic DMI strings, with a module built from
this patch and no quirk module parameter. The quirks were applied via
the new entry, the card components read "cfg-spk:1 cfg-mic:in2
cfg-hp:lrswap", and the following all work: speaker, headphone left/right
order, jack and headset-mic detection, headset button, and the internal
and headset mics.
The problem was diagnosed with the help of an AI coding assistant, which
also drafted this changelog.
Assisted-by: Claude Opus 5.5
Signed-off-by: Jasper Firth <jasper.m.f@hotmail.com>
Link: https://patch.msgid.link/20260925-hi8-quirk-v1-1-ffd00b0447f9@hotmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Maciej Strozek <mstrozek@opensource.cirrus.com> says:
The first corrects the point at which the suspended flag is set during
system suspend, the second balances the runtime PM reference taken in
probe when one of the later probe steps fails.
Link: https://patch.msgid.link/20260924095458.2683755-1-mstrozek@opensource.cirrus.com
Add a common error path in probe that balances the runtime PM reference.
Fixes: 3af1815a2f ("ASoC: SDCA: Add basic SDCA function driver")
Signed-off-by: Maciej Strozek <mstrozek@opensource.cirrus.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260924095458.2683755-3-mstrozek@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
drv->suspended path was executed before system suspension instead of
after, move it lower to correct it.
Fixes: 7a5214f769 ("ASoC: SDCA: Add basic system suspend support")
Signed-off-by: Maciej Strozek <mstrozek@opensource.cirrus.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260924095458.2683755-2-mstrozek@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Lenovo Yoga Pro 7 14ASP9 (83HN) has no digital microphone on the ACP
PDM interface; all internal mics are wired to the Realtek ALC287 codec.
Add 83HN to acp70_acpi_flag_override_table to prevent the legacy ACP
driver from binding and exposing a non-functional PDM capture device,
consistent with the sibling 83W5/83V9 entries.
Reported-by: Mikail <mikailb1998@gmail.com>
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=221886
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260922123029.534152-1-zhangheng@kylinos.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
As per specification, Sample Base Rate (bit 14)
shall be set to 1 for 44.1 kHz.
Fixes: c7e79b2b2d ("ASoC: rt274: add rt274 codec driver")
Signed-off-by: Amadeusz Sławiński <amade@asmblr.net>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20260918111745.3589393-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fat fingered the change when reorganizing the jack-detect
initialization. Equivalent changes for rt274 and rt298:
Commit a43b4394bb ("ASoC: codecs: rt274: Always init jack_detect_work")
Commit 1eb73102da ("ASoC: codecs: rt298: Reorganize jack detect handling")
carry no update to the delay value.
>From user perspective, no difference has been observed between 1250 and
50 values on Intel's SkyLake, KabyLake and AmberLake configurations.
Fixes: 3082afe097 ("ASoC: codecs: rt286: Reorganize jack detect handling")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20260918111745.3589393-1-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Stream6 Input Threshold control points at stream index 4, which is
already used by Stream5. The threshold get and put callbacks derive the
lane ID from the register offset, so the duplicate index makes Stream6
operate on lane 4 and leaves lane 5 inaccessible from userspace.
Use stream index 5 for Stream6.
Fixes: a2df8c2d5b ("ASoC: tegra: Add Tegra186 based ASRC driver")
Signed-off-by: Sheetal <sheetal@nvidia.com>
Reviewed-by: Thierry Reding <treding@nvidia.com>
Link: https://patch.msgid.link/20260921085704.1248920-2-sheetal@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The DRC_THRESH_QR field is bits 7 and 6 of DRC_1, but the control sits on
DRC_0. Its shift and max are right for the field, only the register is
wrong.
Those bits of DRC_0 belong to DRC_STARTUP_GAIN, which "DRC Startup
Volume" claims as bits 10 to 6. So the two controls share bits 7 and 6.
Writing either one moves the other, and the quick release threshold is
never reached at all.
The six fields of DRC_1 tile it exactly, so there is nowhere else the
threshold could live. Point the control at DRC_1.
Found by a sweep for two controls claiming the same bits of one
register. No board with this codec was to hand, the register map is
the driver's own wm8903.h.
Fixes: f1c0a02f32 ("ALSA: ASoC: Add WM8903 CODEC driver")
Signed-off-by: HyeongJun An <sammiee5311@gmail.com>
Assisted-by: Claude:claude-opus-5
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260920135327.3628704-1-sammiee5311@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Lenovo V15 G6 ARP (product name 83UU) uses the ACP6x PDM interface
for its internal digital microphone array. Its firmware exposes neither
the AcpDmicConnected _DSD property nor the _WOV method, so acp6x_probe()
falls through to the DMI check and returns -ENODEV, leaving
acp_yc_mach.0 unbound and no capture device registered.
Add a DMI quirk entry for this model. With it, the machine driver binds
and the DMIC capture device is created:
acp_yc_mach acp_yc_mach.0: Enabling ACP DMIC support via DMI
card 2: acp6x [acp6x], device 0: DMIC capture dmic-hifi-0
Capture was verified working with arecord and PipeWire.
Signed-off-by: Daniel Ortiz <letargopausado@gmail.com>
Link: https://patch.msgid.link/20260920011920.2482786-1-letargopausado@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
stream_config is not initialized before being passed to
sdw_stream_add_slave(). The type field may contain garbage and is
later copied to stream->type by sdw_config_stream().
Zero-initialize stream_config so type defaults to SDW_STREAM_PCM.
While at it, use snd_sdw_params_to_config() helper instead of
open-coding the same logic.
Fixes: 2b7aecd585 ("ASoC: rt1017: Add RT1017 SDCA amplifier driver")
Cc: stable@vger.kernel.org
Signed-off-by: Jiangshan Yi <yijiangshan@kylinos.cn>
Link: https://patch.msgid.link/20260914104551.377880-1-yijiangshan@kylinos.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
stream_config is not initialized before being passed to
sdw_stream_add_slave(). The type field may contain garbage and is
later copied to stream->type by sdw_config_stream().
Zero-initialize stream_config so type defaults to SDW_STREAM_PCM.
Fixes: 18c0af945f ("ASoC: max98363: add soundwire amplifier driver")
Cc: stable@vger.kernel.org
Signed-off-by: Jiangshan Yi <yijiangshan@kylinos.cn>
Link: https://patch.msgid.link/20260914104456.376666-1-yijiangshan@kylinos.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
The ACP7.0 SoundWire machine table did not cover boards that combine TI
TAS2783 smart amplifiers on SoundWire link 0 with a Realtek RT712-VB
codec on link 1. In addition, the TAS2783 SoundWire codec driver was
not selected by the AMD legacy SoundWire machine config, so it was not
built for these platforms.
Select SND_SOC_TAS2783_SDW from SND_SOC_AMD_LEGACY_SDW_MACH so the
TAS2783 SoundWire codec driver is built together with the AMD legacy
(no-DSP) SoundWire machine.
Add a new link/address map, acp70_tas2783a_l0_rt712_vb_l1, describing
the TAS2783 amplifiers on link 0 and the RT712-VB codec on link 1, and
register a corresponding entry in snd_soc_acpi_amd_acp70_sdw_machines[].
The entry uses snd_soc_acpi_amd_sdca_is_device_rt712_vb() as its
machine_check callback so it only matches platforms that actually
expose the RT712-VB device.
Signed-off-by: Syed Saba Kareem <Syed.SabaKareem@amd.com>
Link: https://patch.msgid.link/20260917165314.1880855-1-syed.sabakareem@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Like for SPI, the I2C driver needs an OF match table for the kernel to
be able to automatically load the driver when built as a module. Add
one.
Signed-off-by: Alvin Šipraga <alvin.sipraga@analog.com>
Reviewed-by: Nuno Sá <nuno.sa@analog.com>
Link: https://patch.msgid.link/20260914-asoc-adau1977-fixes-v1-3-aa2f0cabd728@analog.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Since commit 5ab23c7923 ("modpost: Create modalias for builtin
modules") MODULE_DEVICE_TABLE() is enough to reference a match table and
the data isn't discarded by the linker even when the driver is built-in
and CONFIG_OF is disabled. Drop the of_match_ptr() wrapping so that OF
matching keeps working regardless of CONFIG_OF. This also means we can
drop __maybe_unused since it's always used.
The entries in adau1977_spi_of_match were also erroneously indented with
spaces - replace the indentation with tabs to conform with coding style.
Signed-off-by: Alvin Šipraga <alvin.sipraga@analog.com>
Reviewed-by: Nuno Sá <nuno.sa@analog.com>
Link: https://patch.msgid.link/20260914-asoc-adau1977-fixes-v1-2-aa2f0cabd728@analog.com
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_ADAU1977_{SPI,I2C} are missing Kconfig text, so they don't show
up in menuconfig and can't be selected by a user - only by another
symbol such as a machine driver. Add the text to make these symbols
selectable and usable with generic machine drivers like the simple audio
card.
Signed-off-by: Alvin Šipraga <alvin.sipraga@analog.com>
Reviewed-by: Nuno Sá <nuno.sa@analog.com>
Link: https://patch.msgid.link/20260914-asoc-adau1977-fixes-v1-1-aa2f0cabd728@analog.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Vijendar Mukunda <Vijendar.Mukunda@amd.com> says:
This series fixes four defects in the AMD ACP SoundWire machine drivers
(acp-sdw-legacy-mach.c and acp-sdw-sof-mach.c).
A bounds check is added to validate the SoundWire link ID before it is
used as an array index in create_sdw_dailink(), preventing out-of-bounds
access when an unexpected link_mask value is encountered. The codec
config count in the SOF machine driver is refactored to use a dedicated
variable rather than reusing the endpoint-count variable for two
purposes, making the intent clearer and avoiding a stale value being
passed to the codec config array. An operator-precedence bug in the
ffs(link_mask - 1) expression is corrected to ffs(link_mask) - 1,
ensuring the link ID is derived from the correct bit position. Finally,
the SOF machine driver card name is shortened to fit within the 16-byte
snd_card driver[] field and eliminate a compile-time warning.
Link: https://patch.msgid.link/20260910161728.1452808-1-Vijendar.Mukunda@amd.com
The ALSA snd_card driver[] field is 16 bytes (including the NUL
terminator), leaving 15 usable characters. The SOF framework
prepends a "sof-" prefix when registering the card, so
card->name = "amd-soundwire" becomes driver name "sof-amd-soundwire"
which is 17 characters and overflows the driver[16] buffer, triggering
a kernel warning.
Fix by shortening the card name to "amd-sdw"; the resulting driver
name "sof-amd-sdw" fits within the 15-character limit.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Mario Limonciello (AMD) <superm1@kernel.org>
Link: https://patch.msgid.link/20260910161728.1452808-5-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ffs(link_mask - 1) computes ffs on (link_mask - 1) instead of
subtracting 1 from the result of ffs(link_mask). For a typical
power-of-2 link_mask this returns the wrong link ID, causing cpu_pin_id
lookup to select the incorrect SoundWire manager.
Fix the operator precedence to ffs(link_mask) - 1 in both
acp-sdw-sof-mach.c and acp-sdw-legacy-mach.c.
Fixes: 6d8348ddc5 ("ASoC: amd: acp: refactor SoundWire machine driver code")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Mario Limonciello (AMD) <superm1@kernel.org>
Link: https://patch.msgid.link/20260910161728.1452808-4-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
num_devs was used both as the endpoint count and as the output for
asoc_sdw_parse_sdw_endpoints(), which overwrites it with the codec
configuration count. Introduce a separate num_confs variable to hold
the codec conf count so the two values remain distinct across
codec_conf allocation and card->num_configs assignment.
Fixes: 6d8348ddc5 ("ASoC: amd: acp: refactor SoundWire machine driver code")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Mario Limonciello (AMD) <superm1@kernel.org>
Link: https://patch.msgid.link/20260910161728.1452808-3-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a bounds check in create_sdw_dailink() to validate that the
SoundWire link ID derived from link_mask does not exceed the maximum
supported by the platform. If the link ID is out of range or link_mask
is zero, log an error and return -EINVAL to prevent accessing invalid
CPU pin ID tables.
Applied to both acp-sdw-sof-mach.c and acp-sdw-legacy-mach.c.
Fixes: 6d8348ddc5 ("ASoC: amd: acp: refactor SoundWire machine driver code")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Mario Limonciello (AMD) <superm1@kernel.org>
Link: https://patch.msgid.link/20260910161728.1452808-2-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In cs_amp_alloc_get_efi_variable() the first call to
cs_amp_get_efi_variable() might return EFI_SUCCESS if the variable
exists with zero length. Trap this and return -ENOENT to prevent
returning an unexpected NULL pointer.
The first cs_amp_get_efi_variable() call was assumed to return
EFI_BUFFER_TOO_SMALL if the variable existed, but if instead it
returned EFI_SUCCESS this would be converted to 0 by
cs_amp_convert_efi_status() and then be returned as a NULL pointer.
Fixes: 00fd40bc7a ("ASoC: cs-amp-lib: Support Dell SSIDExV2 UEFI variable")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260914122611.2783563-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
