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Vendor want to add more machine on this workaround.
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/e13d08e96ac449b6994d56dfe6ce3f5c@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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tas2563_save_calibration() assigns cali_data.total_sz before it reads the
per-device calibration data from EFI, but its error paths return without
clearing it again. cali_data.cali_reg_array is left all zero, because the
function returns before the register addresses are assigned.
On the first playback tasdev_load_calibrated_data() does
if (!data || !cali_data->total_sz)
return;
which passes, since total_sz is still non-zero. It then issues five
4-byte bulk writes to p->r0_reg, p->r0_low_reg, p->invr0_reg, p->pow_reg
and p->tlimit_reg, all of which are 0. Register 0 decodes to book 0 /
page 0 / register 0x00, so the auto-incrementing block write zeroes
registers 0x00 to 0x03. Register 0x03 is PB_CFG1, which holds AMP_LEVEL,
so the amplifier gain is set to its minimum and the speaker stays silent.
This is reproducible on a Lenovo Yoga 7 14ARB7 (two TAS2563 on I2C,
ACPI INT8866) whose factory calibration was never written to UEFI, so the
EFI read fails with EFI_NOT_FOUND. The two woofers driven by the
amplifiers are silent while the tweeters driven directly by the ALC287
play. Reading the amplifier registers over i2c shows PWR_CTL = 0x00
(active) and the TDM slots correctly programmed by the RCA profile, but
PB_CFG1 = 0x00. With this change PB_CFG1 keeps its power-on default of
0x20 and both woofers play.
tas2781_save_calibration() in tas2781_hda.c already clears total_sz on
failure; do the same for the TAS2563 variant.
Signed-off-by: Philipp Oster <philippdev5396@outlook.de>
Link: https://patch.msgid.link/20260720-tas2781-calfix-v1-1-3a5fa6ad90bc@outlook.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add a codec SSID quirk for Samsung ELECTRONICS 750XBE/730XBE using
HDA_CODEC_QUIRK() instead of SND_PCI_QUIRK(), because the alsa-info
report from this device does not expose a PCI subsystem ID, only the
HDA codec subsystem ID (0x144d:0xc824) is available.
This applies ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET to fix sound
being very low and distorted on the headphone jack of this system.
Reported-by: Caio Ramos <caioramos97@gmail.com>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=208663
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260720123702.799474-1-zhangheng@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The firmware on the Lunnen Ground 14 marks pin 0x1b as unused even
though the internal speakers are connected to it. As a result, the
speakers are not detected.
Add a pin configuration quirk for PCI subsystem ID 2782:a212 to configure
pin 0x1b as an internal speaker.
The pin configuration was tested on a Lunnen Ground 14 (DMI product LL4FA)
with an ALC269VC codec. The internal speakers and microphone work as
expected.
Cc: stable@vger.kernel.org
Signed-off-by: Nikita Maksimov <nickstogramm@yandex.ru>
Link: https://patch.msgid.link/20260720180214.73770-1-nickstogramm@yandex.ru
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_timer_close_locked() drains the closing instance's own in-flight
callback (IFLG_CALLBACK) before freeing it, but not its slaves'. When a
master instance is closed, remove_slave_links() clears each slave's
->timer; the slave's own close then reads timer == NULL and takes the
branch that skips the drain entirely (snd_timer_stop_slave() also no-ops
on a NULL timer). So a slave whose callback is still running when the
master is closed is freed underneath the live callback, leading to
use-after-free.
Drain the slaves too before remove_slave_links() severs them.
snd_timer_stop() has already taken this instance off the active list, so
no new slave callback can be queued. Take the slaves off the ack list so
a pending one can't fire either, then wait for any that is already in
flight.
Fixes: 37745918e0e7 ("ALSA: timer: Introduce virtual userspace-driven timers")
Cc: stable@vger.kernel.org
Assisted-by: Claude:claude-opus-4-8
Signed-off-by: Norbert Szetei <norbert@doyensec.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/D26598EB-DBF7-4D76-9F71-8E4BD59822D4@doyensec.com
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The userspace-driven timer (utimer) TRIGGER ioctl calls
snd_timer_interrupt() directly with no serialization, so two threads
triggering the same utimer can run snd_timer_interrupt() on one
snd_timer concurrently.
snd_timer_process_callbacks() drops timer->lock around each instance
callback and marks the in-flight callback with the single
SNDRV_TIMER_IFLG_CALLBACK bit; snd_timer_close_locked() waits on that
bit to drain an in-flight callback before freeing the instance. The bit
cannot represent two concurrent callbacks: when a second interrupt
re-queues an instance whose callback is still running, both run at once,
the first to finish clears the bit, and the close-path drain then frees
the instance (and its callback_data) while the other callback is still
live - a use-after-free reachable by any user able to open
/dev/snd/timer, both via a user timer instance and via a sequencer queue
timer bound to the utimer.
snd_timer_interrupt() sets IFLG_CALLBACK before dropping timer->lock, so
a concurrent interrupt already observes it under the lock. Skip
re-queuing an instance (and its slaves) to the ack/sack list while its
callback is in flight; the accumulated pticks are delivered on the next
tick, so no event is lost.
Fixes: 37745918e0e7 ("ALSA: timer: Introduce virtual userspace-driven timers")
Cc: stable@vger.kernel.org
Suggested-by: Takashi Iwai <tiwai@suse.de>
Assisted-by: Claude:claude-opus-4-8
Signed-off-by: Norbert Szetei <norbert@doyensec.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/6F9B6501-8E65-4265-B02C-7EFB240D1664@doyensec.com
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JBL Quantum650 Wireless (0ecb:2125) requires the same workaround that
was used for JBL Quantum610 and Quantum810 for limiting the sample rate.
Without it, the capture (microphone) stream fails to work. Setting the
QUIRK_FLAG_FIXED_RATE flag, as done for the sibling models, makes both
playback and capture work correctly.
Signed-off-by: Daniel C. Ribeiro <dcoutinho.96@gmail.com>
Link: https://patch.msgid.link/20260719090037.40149-1-dcoutinho.96@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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HP Pavilion x360 sets a bogus PCI SSID (103c:0000), hence the driver
picks up a wrong quirk entry, resulting in an almost silent output.
And yet, the existing quirk for x390 doesn't seem sufficing, and we
need the extra setup for the amp.
This patch adds the quirk entry for the codec SSID (103c:8486) to
initialize the amp via COEF verbs and chains to the existing quirk for
another x360 model to address the silent output.
Closes: https://lore.kernel.org/CAF2ktaUW2oaWwGazGtJQ3o1JyE2R4O2xPd-Dchr=qqi7_QRruQ@mail.gmail.com
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260716064916.540616-1-tiwai@suse.de
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v7.2
A fairly standard collection of device specific fixes and quirks,
nothing particularly remarkable.
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A couple of new Dell laptops are shipping using the sidecar amp
configuration. Add the required kernel quirk to enable.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260716144209.2411523-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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In cs35l56_init() use complete_all() to signal init_completion instead
of complete().
cs35l56_init() was signaling init_completion using the complete() function.
This only releases ONE waiter.
If cs35l56_component_probe() was called multiple times the first time
would consume that one signal, then future calls would timeout waiting for
the completion. This could happen if:
- The component is probed, removed, then probed again without the cs35l56
module being removed.
- A call to component_probe() returns an error and ASoC calls it again
later.
It should use complete_all() so that after it has been signaled it will
allow any code that waits on it to continue immediately.
The one case where the driver must wait for initialization to run again is
when waiting for a reboot after firmware download, and here the code
correctly calls reinit_completion() first.
Fixes: e496112529006 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260716132045.1469156-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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On I2C/SPI call cs35l56_init() before calling
snd_soc_register_component() to prevent the potential for a deadlock
on init_completion.
For most buses all the hardware would be ready when probe() returns,
but on SoundWire, probe() must return before the SoundWire bus driver
will enumerate the device. All access to the registers must be deferred
until the driver receives an ATTACHED notification. But anything that
could return -EPROBE_DEFER must be called during probe, and that includes
snd_soc_register_component(). Because of that, on SoundWire the ASoC
component can be created before the registers are accssible, so
cs35l56_component_probe() waits for init_completion to signal that the
registers are accessible.
On I2C/SPI this 2-stage startup isn't required so their probe()
functions simply called cs35l56_common_probe() and then cs35l56_init().
The problem with this was that snd_soc_register_component() was still
called early. If this triggered ASoC to create the card, ASoC would call
cs35l56_component_probe() which waits on init_completion - but this would
be running inside the cs35l56 driver probe() so blocking it from reaching
the code that signals init_completion, causing a deadlock.
Fixes: e496112529006 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Reported-by: Salman S. Tahir <salman.abusaad@gmail.com>
Closes: https://lore.kernel.org/linux-sound/95c21574-97d5-4311-9263-9e174d22d22c@opensource.cirrus.com/T/#u
Tested-by: Salman S. Tahir <salman.abusaad@gmail.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260716132045.1469156-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The fs210x driver currently puts the regmap cache into cache only mode
during suspend but at no point does it make it writable again, meaning
that device can't work after suspend other than basic powerup which
bypasses the cache.
Mark the cache as writable again when we resync the registers, it's not
clear to me that the cache only mode is needed at all but this is a
fairly minimal change. Since this is setting a flag it's safe to do
even if the cache was not previously in cache only mode.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20260715-asoc-fs210x-fix-resume-v1-1-dd81d9c62c05@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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HP ZBook 8 G2a 14 and 16 (SSIDs 0x103c:0x8f94, 0x103c:0x8f95) use the
Realtek ALC245 codec with a TAS2781 amplifier via I2C. They share the same
hardware configuration as the existing 0x8f40/0x8f41/0x8f42/0x8f62 models
but have inverted speaker mute LED polarity, so the existing
ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED quirk drives the LED backwards: off
when muted and on when unmuted.
Add a dedicated quirk with inverted COEF values. These are speaker-only
models without an HP pin, so the LED is driven directly through the
vmaster_mute hook; there is no need to probe the HP pin at runtime since
the configuration is static and known per SSID.
Signed-off-by: Chris Chiu <chris.chiu@canonical.com>
Link: https://patch.msgid.link/20260716023825.387532-1-chris.chiu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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: 15175a4f2bbb ("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>
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The bt-sco-pcm-wb DAI uses the same stream_name strings as bt-sco-pcm
("Playback" and "Capture"). This causes duplicate DAPM AIF widget
names within the same component, leading to debugfs warnings:
debugfs: 'Playback' already exists in 'dapm'
debugfs: 'Capture' already exists in 'dapm'
Give the wideband DAI distinct stream names ("WB Playback" and
"WB Capture") and add corresponding DAPM AIF widgets and routes for
them.
Fixes: 5947e1b4992e ("ASoC: bt-sco: extend rate and add a general compatible string")
Assisted-by: VeroCoder:claude-sonnet-4-5
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/20260715100620.1387159-1-shengjiu.wang@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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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: 447106e92a0c ("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>
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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>
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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>
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The internal digital microphone on the MSI Vector A16 HX A8WIG is not
detected: the ACP platform devices are created, but snd_soc_acp6x_mach
never binds because the machine is missing from the DMI quirk table, so
no capture device shows up at all.
This is the same board as the already supported "Vector A16 HX A8WHG",
differing only in the trailing model code. Add the corresponding entry.
Signed-off-by: Antonio Ignacio Campos Ruiz <acamposruiz@gmail.com>
Link: https://patch.msgid.link/20260713165709.19489-1-acamposruiz@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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In a full-duplex setup, when one direction (playback or capture) is
closed while the other is still running, imx_aif_shutdown() was
unconditionally calling snd_soc_dai_set_sysclk() with rate=0 for all
cpu/codec DAIs, which would disable the clock still needed by the
active stream.
Add snd_soc_dai_active() checks before clearing sysclk so that only
truly inactive DAIs have their clocks reset.
Fixes: 2260bc6ea8bd ("ASoC: imx-card: Add WM8524 support")
Cc: stable@vger.kernel.org
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://patch.msgid.link/20260710031333.3491445-1-chancel.liu@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The H616 codec does not have capture capabilities. Set the
.playback_only quirks flag to denote this.
This was somehow missing from the original driver patch, even though
the patch prior to it in the series added this quirk.
Fixes: 9155c321a1d0 ("ASoC: sun4i-codec: support allwinner H616 codec")
Signed-off-by: Chen-Yu Tsai <wens@kernel.org>
Link: https://patch.msgid.link/20260714113304.270224-1-wens@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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The TAS2781 firmware parser reads several variable-length description
strings with strlen() before checking that the string terminator is
present inside the firmware blob. A malformed firmware image without a
NUL terminator can therefore make the parser walk past the end of the
firmware buffer before the later size checks run.
Add a small bounded string-length helper and use it for all description
fields that are parsed from the firmware buffer. Keep the existing size
checks for the fixed bytes that follow each string.
Fixes: 915f5eadebd2 ("ASoC: tas2781: firmware lib")
Signed-off-by: Pengpeng Hou <pengpeng@iscas.ac.cn>
Link: https://patch.msgid.link/20260706144540.93929-1-pengpeng@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
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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: 1da177e4c3f4 ("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>
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Some ASUS TUF ACP70-based systems expose ACP ACPI configuration flags
that select a non-working fallback audio path, similar to previously
affected ASUS platforms.
Add DMI-based overrides in snd_amd_acp_find_config() for the following
systems to skip ACP ACPI flag-based selection:
- ASUS TUF Gaming Vivobook 18
- ASUS TUF Gaming A14 FA401EA
This ensures the intended SoundWire-based machine driver is selected on
these platforms.
Signed-off-by: Syed Saba Kareem <Syed.SabaKareem@amd.com>
Link: https://patch.msgid.link/20260710102926.1633385-1-syed.sabakareem@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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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
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Remove the mic bias current comparator threshold override (NID 0x1c,
verb 0x320, value 0x010) from Conexant codec driver.
This override was originally intended to support volume up/down controls on
headsets with inline remote controls, but it causes microphone detection
failures on some headsets with impedance less than 1k ohm.
After consulting with the vendor's engineers, it was confirmed that this
setting is board-specific and should be handled by BIOS/firmware rather
than the generic codec driver, especially since inline remote support
is not currently implemented.
Fixes: 7aeb25908648 ("ALSA: hda/conexant: Fix headset auto detect fail in cx8070 and SN6140")
Cc: stable@vger.kernel.org
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260713100329.306892-1-zhangheng@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When the BCLK divider ratio is 1:1, fsl_sai_set_bclk() enables bypass
mode by setting BYP, but never clears the bit. The BYP=1 value remains
in the regcache, and is restored by regcache_sync() on the next runtime
resume.
Since BYP=1 combined with BCD=1 immediately outputs the ungated MCLK
as BCLK without waiting for BCE/TE/RE to be enabled, the clock is
driven prematurely before the stream is fully configured, causing
noise on some codecs.
Fix this by clearing BYP and BCI in fsl_sai_hw_free() taking into
account sync mode and the opposite stream's state, so that the regcache
holds BYP=0 before runtime suspend and regcache_sync() on resume will
not restore bypass mode prematurely.
Fixes: a50b7926d015 ("ASoC: fsl_sai: implement 1:1 bclk:mclk ratio support")
Cc: stable@vger.kernel.org
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20260710070835.3749817-1-chancel.liu@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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17aa:38a7
Some units of the Lenovo Legion Pro 7 16ARX8H (82WS) report codec
subsystem ID 17aa:38a7 instead of 17aa:38a8. Since only 38a8 has a
codec SSID quirk, these machines fall through to the PCI SSID match
17aa:386f (Legion Pro 7i 16IAX7) and get ALC287_FIXUP_CS35L41_I2C_2,
which probes the Cirrus amplifiers of the Intel variant. The TI
TAS2781 amplifier (ACPI TIAS2781:00) present on this AMD variant is
never bound and the internal speakers remain silent.
Add a codec SSID quirk for 17aa:38a7 pointing to
ALC287_FIXUP_TAS2781_I2C, mirroring the existing 38a8 entry.
Tested on a Legion Pro 7 16ARX8H (82WS, BIOS LPCN62WW): with the codec
SSID overridden to 17aa:38a8 via the HDA patch loader, the TAS2781
amplifier binds and the internal speakers work.
Cc: <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>
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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>
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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>
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Salvador reported that the recent fix for applying the DSD quirk to
Musical Fidelity devices broke for his M6s DAC model (2772:0502).
Although this is basically a firmware bug, the model in question is
fairly old, and no further firmware update can be expected, so it'd be
better to address in the driver side.
As an ad hoc workaround, skip the DSD quirk for this device by adding
an empty quirk entry of 2772:0502; this essentially skips the later
DSD quirk entry by the match with the vendor 2772.
Fixes: da3a7efff64e ("ALSA: usb-audio: Update for native DSD support quirks")
Reported-by: Salvador Blaya <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
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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.
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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>
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Fix microphone detection on built in headphone jack for some devices.
Signed-off-by: Eckhart Mohr <e.mohr@tuxedocomputers.com>
Cc: stable@vger.kernel.org
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com>
Link: https://patch.msgid.link/20260708132135.102680-1-wse@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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In tas2562_parse_dt(), the fallback lookup for the deprecated
"shut-down" GPIO property is broken due to a missing pair of braces.
The code intends to reset sdz_gpio to NULL only when the lookup
returns an error that is not -EPROBE_DEFER (so the driver gracefully
continues without a GPIO). However, without braces the statement:
tas2562->sdz_gpio = NULL;
falls outside the IS_ERR() check and is executed unconditionally
for every path through the if block, including a successful GPIO
lookup.
This means any device using the deprecated 'shut-down' DT property
will always have sdz_gpio == NULL after probe, making the GPIO
completely non-functional.
Fix this by adding the missing braces to scope the NULL assignment
inside the IS_ERR() branch, matching the pattern already used for
the primary 'shutdown' GPIO lookup above.
Fixes: f78a97003b8b ("ASoC: tas2562: Update shutdown GPIO property")
Signed-off-by: Uday Khare <udaykhare77@gmail.com>
Link: https://patch.msgid.link/20260706153109.10953-1-udaykhare77@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Currently if the jack is forced to the microphone mode, it will report
as line in. Correct the report to microphone.
Fixes: fc918cbe874e ("ASoC: cs42l43: Add support for the cs42l43")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260708103430.1395207-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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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
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Add compatible for sound card on Qualcomm Eliza boards.
Signed-off-by: Ravi Hothi <ravi.hothi@oss.qualcomm.com>
Link: https://patch.msgid.link/20260703073434.2589657-4-ravi.hothi@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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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>
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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>
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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>
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This series contains three bug fixes for the AMD common ACP PCI driver
(pci-ps.c) covering ACP6.3, ACP7.0, ACP7.1 and ACP7.2 platforms.
Link: https://patch.msgid.link/20260707060130.2514138-1-Vijendar.Mukunda@amd.com
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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: 7f91f012c1df0 ("ASoC: amd: ps: fix for irq handler return status")
Link: https://patch.msgid.link/20260707060130.2514138-4-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The driver handles ACP6.3/7.0/7.1/7.2 platforms but the region was
claimed with the stale name "AMD ACP6.2 audio" left over from the
original ACP6.2 driver. Correct it to "AMD ACP6.3 audio".
Fixes: 95e43a170bb1 ("ASoC: amd: add Pink Sardine ACP PCI driver")
Signed-off-by: Vijendar Mukunda <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>
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BIOS/firmware may re-enable MSI in PCI config space during system
level resume even though this driver only uses legacy INTx interrupts.
If MSI is left enabled with stale address/data registers, the device
will write interrupts to a bogus address causing IOMMU IO_PAGE_FAULT
and interrupt delivery failure.
Clear the MSI Enable bit before reinitializing the ACP hardware on
system level resume.
Fixes: 491628388005 ("ASoC: amd: ps: add callback functions for acp pci driver pm ops")
Signed-off-by: Vijendar Mukunda <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>
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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>
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The I2S FIFO soft-resets its fast domain on start (AIU_RST_SOFT bit 0 +
AIU_I2S_SYNC read in aiu_fifo_i2s_trigger), mirroring the downstream
vendor driver's audio_out_i2s_enable(). The S/PDIF FIFO has no equivalent:
it only toggles the IEC958 DCU, so a stale datapath FIFO can be replayed,
producing the "machine gun noise" buffer underrun - on start when switching
outputs, and on stop when playback ends. The latter is audible on devices
with an always-on S/PDIF-fed DAC (e.g. the ES7144 on the WeTek Play2).
The vendor driver resets the IEC958 fast domain (AIU_RST_SOFT bit 2) on
both enable and disable (audio_hw_958_enable), and when reconfiguring
(audio_hw_958_reset clears AIU_958_DCU_FF_CTRL then resets). Do the same:
reset before enabling the DCU on start, and before disabling on stop.
Fixes: 6ae9ca9ce986bf ("ASoC: meson: aiu: add i2s and spdif support")
Signed-off-by: Christian Hewitt <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>
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Intel HD Audio defines Coefficient Index and Processing Coefficient as
separate audio widget controls in the Audio Widget Verb Definitions:
Coefficient Index selects the coefficient slot, while Processing
Coefficient accesses the value at the selected slot.
hda_reg_read_coef() selects the slot with AC_VERB_SET_COEF_INDEX, but
then uses AC_VERB_GET_COEF_INDEX for the value read. That reads back the
selected index instead of the coefficient value. hda_reg_write_coef()
has the same issue and builds the value write from AC_VERB_GET_COEF_INDEX
instead of AC_VERB_SET_PROC_COEF.
This only affects the regmap coefficient cache path used by codecs that
set codec->cache_coef. Direct coefficient helpers already use the normal
SET_COEF_INDEX followed by GET_PROC_COEF or SET_PROC_COEF sequence, which
is likely why this has not been noticed widely.
Use AC_VERB_GET_PROC_COEF for cached coefficient reads and
AC_VERB_SET_PROC_COEF for cached coefficient writes.
Fixes: 40ba66a702b8 ("ALSA: hda - Add cache support for COEF read/write")
Cc: stable@vger.kernel.org
Signed-off-by: Xu Rao <raoxu@uniontech.com>
Link: https://patch.msgid.link/DB9023BF2920BA99+20260707132419.1731342-1-raoxu@uniontech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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