The current 6fire code tries to release the resources right after the
call of usb6fire_chip_abort(). But at this moment, the card object
might be still in use (as we're calling snd_card_free_when_closed()).
For avoid potential UAFs, move the release of resources to the card's
private_free instead of the manual call of usb6fire_chip_destroy() at
the USB disconnect callback.
Fixes: c6d43ba816 ("ALSA: usb/6fire - Driver for TerraTec DMX 6Fire USB")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20241113111042.15058-6-tiwai@suse.de
The USB disconnect callback is supposed to be short and not too-long
waiting. OTOH, the current code uses snd_card_free() at
disconnection, but this waits for the close of all used fds, hence it
can take long. It eventually blocks the upper layer USB ioctls, which
may trigger a soft lockup.
An easy workaround is to replace snd_card_free() with
snd_card_free_when_closed(). This variant returns immediately while
the release of resources is done asynchronously by the card device
release at the last close.
This patch also splits the code to the disconnect and the free phases;
the former is called immediately at the USB disconnect callback while
the latter is called from the card destructor.
Fixes: 523f1dce37 ("[ALSA] Add Native Instrument usb audio device support")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20241113111042.15058-5-tiwai@suse.de
us122l.mmap_count field was used for counting the hwdep mmap opens and
syncing at disconnection. But such a manual sync isn't needed, as the
refcount check is done in the ALSA core side. So let's drop it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20241113111042.15058-4-tiwai@suse.de
The USB disconnect callback is supposed to be short and not too-long
waiting. OTOH, the current code uses snd_card_free() at
disconnection, but this waits for the close of all used fds, hence it
can take long. It eventually blocks the upper layer USB ioctls, which
may trigger a soft lockup.
An easy workaround is to replace snd_card_free() with
snd_card_free_when_closed(). This variant returns immediately while
the release of resources is done asynchronously by the card device
release at the last close.
The loop of us122l->mmap_count check is dropped as well. The check is
useless for the asynchronous operation with *_when_closed().
Fixes: 030a07e441 ("ALSA: Add USB US122L driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20241113111042.15058-3-tiwai@suse.de
The USB disconnect callback is supposed to be short and not too-long
waiting. OTOH, the current code uses snd_card_free() at
disconnection, but this waits for the close of all used fds, hence it
can take long. It eventually blocks the upper layer USB ioctls, which
may trigger a soft lockup.
An easy workaround is to replace snd_card_free() with
snd_card_free_when_closed(). This variant returns immediately while
the release of resources is done asynchronously by the card device
release at the last close.
Fixes: 230cd5e248 ("[ALSA] prevent oops & dead keyboard on usb unplugging while the device is be ing used")
Reported-by: syzbot+73582d08864d8268b6fd@syzkaller.appspotmail.com
Closes: https://syzkaller.appspot.com/bug?extid=73582d08864d8268b6fd
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20241113111042.15058-2-tiwai@suse.de
snd_pcm_mmap_data_open() and _close() are defined as inline functions
in the public sound/pcm.h, but those are used only locally in
pcm_native.c, hence they should be better placed there.
Also, those are referred as callbacks, the useless inline is dropped.
Link: https://patch.msgid.link/20241113111628.17069-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
runtime is not used as seen with W=1 :
sound/core/compress_offload.c: In function ‘snd_compr_poll’:
sound/core/compress_offload.c:409:35: error: variable ‘runtime’ set but not used [-Werror=unused-but-set-variable]
409 | struct snd_compr_runtime *runtime;
| ^~~~~~~
Instead of dropping the runtime, use it in the function in place of
stream->runtime
Fixes: 04177158cf ("ALSA: compress_offload: introduce accel operation mode")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://patch.msgid.link/20241106075312.15601-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that many webcams have buggy firmware and don't expose the
mic capture volume with the proper resolution. We have workarounds in
mixer.c, but judging from the numbers, those can be better managed as
global quirk flags.
Link: https://patch.msgid.link/20241105120220.5740-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's no need to free/re-request the interrupt on system suspend.
PCI core takes care, using functions like pci_restore_msi_state().
Signed-off-by: Heiner Kallweit <hkallweit1@gmail.com>
Link: https://patch.msgid.link/1b7e109b-eb69-4542-8022-4ac8f9116474@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If amdtp_stream_init() fails in amdtp_tscm_init(), the latter returns zero,
though it's supposed to return error code, which is checked inside
init_stream() in file tascam-stream.c.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Fixes: 47faeea25e ("ALSA: firewire-tascam: add data block processing layer")
Signed-off-by: Murad Masimov <m.masimov@maxima.ru>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20241101185517.1819-1-m.masimov@maxima.ru
The legacy rawmidi tries to enumerate all possible UMP groups
belonging to the UMP endpoint. But currently it shows all 16 ports
when the UMP endpoint is configured with static blocks, although most
of them may be unused.
There was already a fix for the sequencer client side to ignore such
groups in the commit 3bfd7c0ba1 ("ALSA: seq: ump: Skip useless ports
for static blocks"), and this commit is a similar fix for UMP
rawmidi devices; it adds simply the check for the validity of each
group that has been already parsed. (Note that the group info was
moved to snd_ump_endpoint.groups[] by the commit 0642a3c5ca
("ALSA: ump: Update substream name from assigned FB names")).
Link: https://patch.msgid.link/20241104100735.16127-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 4f61c8fe35 ("ALSA: hda/conexant: Mute speakers at suspend /
shutdown") mutes speakers on system shutdown or whenever HDA controller
is suspended by PM; this however interacts badly with Thinkpad's ACPI
firmware behavior which uses beeps to signal various events (enter/leave
suspend or hibernation, AC power connect/disconnect, low battery, etc.);
now those beeps are either muted altogether (for suspend/hibernate/
shutdown related events) or work more or less randomly (eg. AC
plug/unplug is only audible when you are playing music at the moment,
because HDA device is likely in suspend mode otherwise).
Since the original bug report mentioned in 4f61c8fe35 complained about
Lenovo's Thinkpad laptop - revert this commit altogether.
Fixes: 4f61c8fe35 ("ALSA: hda/conexant: Mute speakers at suspend / shutdown")
Signed-off-by: Jarosław Janik <jaroslaw.janik@gmail.com>
Link: https://patch.msgid.link/20241030171813.18941-2-jaroslaw.janik@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After commit 0edb555a65 ("platform: Make platform_driver::remove()
return void") .remove() is (again) the right callback to implement for
platform drivers.
Convert all platform drivers below sound to use .remove(), with
the eventual goal to drop struct platform_driver::remove_new(). As
.remove() and .remove_new() have the same prototypes, conversion is done
by just changing the structure member name in the driver initializer.
On the way do a few whitespace changes to make indention consistent.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@baylibre.com>
Link: https://patch.msgid.link/20241029073748.508077-2-u.kleine-koenig@baylibre.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WD19 family of docks has the same audio chipset as the WD15. This
change enables jack detection on the WD19.
We don't need the dell_dock_mixer_init quirk for the WD19. It is only
needed because of the dell_alc4020_map quirk for the WD15 in
mixer_maps.c, which disables the volume controls. Even for the WD15,
this quirk was apparently only needed when the dock firmware was not
updated.
Signed-off-by: Jan Schär <jan@jschaer.ch>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20241029221249.15661-1-jan@jschaer.ch
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The biggest set of changes here is Hans' fixes and quirks for various
Baytrail based platforms with RT5640 CODECs, and there's one core fix
for a missed length assignment for __counted_by() checking. Otherwise
it's small device specific fixes, several of them in the DT bindings.
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Merge tag 'asoc-fix-v6.12-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.12
The biggest set of changes here is Hans' fixes and quirks for various
Baytrail based platforms with RT5640 CODECs, and there's one core fix
for a missed length assignment for __counted_by() checking. Otherwise
it's small device specific fixes, several of them in the DT bindings.
Merge series from Alexey Klimov <alexey.klimov@linaro.org>:
This sent as RFC because of the following:
- regarding the LO switch patch. I've got info about that from two persons
independently hence not sure what tags to put there and who should be
the author. Please let me know if that needs to be corrected.
- the wcd937x pdm watchdog is a problem for audio playback and needs to be
fixed. The minimal fix would be to at least increase timeout value but
it will still trigger in case of plenty of dbg messages or other
delay-generating things. Unfortunately, I can't test HPHL/R outputs hence
the patch is only for AUX. The other options would be introducing
module parameter for debugging and using HOLD_OFF bit for that or
adding Kconfig option.
Alexey Klimov (2):
ASoC: codecs: wcd937x: add missing LO Switch control
ASoC: codecs: wcd937x: relax the AUX PDM watchdog
sound/soc/codecs/wcd937x.c | 12 ++++++++++--
sound/soc/codecs/wcd937x.h | 4 ++++
2 files changed, 14 insertions(+), 2 deletions(-)
--
2.45.2
On a system with wcd937x, rxmacro and Qualcomm audio DSP, which is pretty
common set of devices on Qualcomm platforms, and due to the order of how
DAPM widgets are powered on (they are sorted), there is a small time window
when wcd937x chip is online and expects the flow of incoming data but
rxmacro is not yet online. When wcd937x is programmed to receive data
via AUX port then its AUX PDM watchdog is enabled in
wcd937x_codec_enable_aux_pa(). If due to some reasons the rxmacro and
soundwire machinery are delayed to start streaming data, then there is
a chance for this AUX PDM watchdog to reset the wcd937x codec. Such event
is not logged as a message and only wcd937x IRQ counter is increased
however there could be a lot of other reasons for that IRQ.
There is a similar opportunity for such delay during DAPM widgets power
down sequence.
If wcd937x codec reset happens on the start of the playback, then there
will be no sound and if such reset happens at the end of a playback then
it may generate additional clicks and pops noises.
On qrb4210 RB2 board without any debugging bits the wcd937x resets are
sometimes observed at the end of a playback though not always.
With some debugging messages or with some tracing enabled the AUX PDM
watchdog resets the wcd937x codec at the start of a playback and there
is no sound output at all.
In this patch:
- TIMEOUT_SEL bit in PDM_WD_CTL2 register is set to increase the watchdog
reset delay to 100ms which eliminates the AUX PDM watchdog IRQs on
qrb4210 RB2 board completely and decreases the number of unwanted clicks
noises;
- HOLD_OFF bit postpones triggering such watchdog IRQ till wcd937x codec
reset which usually happens at the end of a playback. This allows to
actually output some sound in case of debugging.
Cc: Adam Skladowski <a39.skl@gmail.com>
Cc: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Cc: Prasad Kumpatla <quic_pkumpatl@quicinc.com>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Alexey Klimov <alexey.klimov@linaro.org>
Link: https://patch.msgid.link/20241022033132.787416-3-alexey.klimov@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The wcd937x supports also AUX input but the control that sets correct
soundwire port for this is missing. This control is required for audio
playback, for instance, on qrb4210 RB2 board as well as on other
SoCs.
Reported-by: Adam Skladowski <a39.skl@gmail.com>
Reported-by: Prasad Kumpatla <quic_pkumpatl@quicinc.com>
Suggested-by: Adam Skladowski <a39.skl@gmail.com>
Suggested-by: Prasad Kumpatla <quic_pkumpatl@quicinc.com>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Signed-off-by: Alexey Klimov <alexey.klimov@linaro.org>
Link: https://patch.msgid.link/20241022033132.787416-2-alexey.klimov@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix DTB warnings when rk3308-codec used with audio-graph-card by
documenting port property:
codec@ff560000: 'port' does not match any of the regexes: 'pinctrl-[0-9]+'
Signed-off-by: Dmitry Yashin <dmt.yashin@gmail.com>
Reviewed-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://patch.msgid.link/20241028213314.476776-2-dmt.yashin@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This also includes a change to the capture level as they differ with
this device, as well as additional values for prefader channels.
Acked-by: Olivia Mackintosh <livvy@base.nu>
Signed-off-by: Sarah Grant <s@srd.tw>
Link: https://patch.msgid.link/20241025163927.12662-1-s@srd.tw
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Infinix ZERO BOOK 13 has a 2+2 speaker system which isn't probed correctly.
This patch adds a quirk with the proper pin connections.
Also The mic in this laptop suffers too high gain resulting in mostly
fan noise being recorded,
This patch Also limit mic boost.
HW Probe for device; https://linux-hardware.org/?probe=a2e892c47b
Test: All 4 speaker works, Mic has low noise.
Signed-off-by: Piyush Raj Chouhan <piyushchouhan1598@gmail.com>
Link: https://patch.msgid.link/20241028155516.15552-1-piyuschouhan1598@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The widgets array in the snd_soc_dapm_widget_list has a __counted_by
attribute attached to it, which points to the num_widgets variable. This
attribute is used in bounds checking, and if it is not set before the
array is filled, then the bounds sanitizer will issue a warning or a
kernel panic if CONFIG_UBSAN_TRAP is set.
This patch sets the size of the widgets list calculated with
list_for_each as the initial value for num_widgets as it is used for
allocating memory for the array. It is updated with the actual number of
added elements after the array is filled.
Signed-off-by: Aleksei Vetrov <vvvvvv@google.com>
Fixes: 80e698e2df ("ASoC: soc-dapm: Annotate struct snd_soc_dapm_widget_list with __counted_by")
Link: https://patch.msgid.link/20241028-soc-dapm-bounds-checker-fix-v1-1-262b0394e89e@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Stephen Rothwell reports multiple htmldocs indentation warnings when
merging sound tree for linux-next:
Documentation/sound/designs/compress-accel.rst:67: ERROR: Unexpected indentation.
Documentation/sound/designs/compress-accel.rst:67: WARNING: Blank line required after table.
Documentation/sound/designs/compress-accel.rst:68: WARNING: Line block ends without a blank line.
Documentation/sound/designs/compress-accel.rst:72: WARNING: Block quote ends without a blank line; unexpected unindent.
Documentation/sound/designs/compress-accel.rst:73: WARNING: Block quote ends without a blank line; unexpected unindent.
Fix these above by wrapping passthrough audio stream state machine
flowchart in a literal code block.
Fixes: 04177158cf ("ALSA: compress_offload: introduce accel operation mode")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Closes: https://lore.kernel.org/linux-next/20241028193242.11597640@canb.auug.org.au/
Signed-off-by: Bagas Sanjaya <bagasdotme@gmail.com>
Link: https://patch.msgid.link/20241028111647.17378-2-bagasdotme@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If devm_gpiod_get_optional() fails, we need to disable previously enabled
regulators, as done in the other error handling path of the function.
Also, gpiod_set_value_cansleep(, 1) needs to be called to undo a
potential gpiod_set_value_cansleep(, 0).
If the "reset" gpio is not defined, this additional call is just a no-op.
This behavior is the same as the one already in the .remove() function.
Fixes: 11b9cd748e ("ASoC: cs42l51: add reset management")
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/a5e5f4b9fb03f46abd2c93ed94b5c395972ce0d1.1729975570.git.christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Some old Bay Trail tablets which shipped with Android as factory OS
have the SST/LPE audio engine described by an ACPI device with a
HID (Hardware-ID) of LPE0F28 instead of 80860F28.
Add support for this. Note this uses a new sst_res_info for just
the LPE0F28 case because it has a different layout for the IO-mem ACPI
resources then the 80860F28.
An example of a tablet which needs this is the Vexia EDU ATLA 10 tablet,
which has been distributed to schools in the Spanish Andalucía region.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://patch.msgid.link/20241025090221.52198-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a requirement to expose the audio hardware that accelerates various
tasks for user space such as sample rate converters, compressed
stream decoders, etc.
This is description for the API extension for the compress ALSA API which
is able to handle "tasks" that are not bound to real-time operations
and allows for the serialization of operations.
For details, refer to "compress-accel.rst" document.
Cc: Mark Brown <broonie@kernel.org>
Cc: Shengjiu Wang <shengjiu.wang@gmail.com>
Cc: Nicolas Dufresne <nicolas@ndufresne.ca>
Cc: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Cc: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Tested-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20241002093904.1809799-1-perex@perex.cz
The Vexia Edu Atla 10 tablet mostly uses the BYTCR tablet defaults,
but as happens on more models it is using IN1 instead of IN3 for
its internal mic and JD_SRC_JD2_IN4N instead of JD_SRC_JD1_IN4P
for jack-detection.
Add a DMI quirk for this to fix the internal-mic and jack-detection.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://patch.msgid.link/20241024211615.79518-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On some x86 Bay Trail tablets which shipped with Android as factory OS,
the DSDT is so broken that the codec needs to be manually instantatiated
by the special x86-android-tablets.ko "fixup" driver for cases like this.
This means that the codec-dev cannot be retrieved through its ACPI fwnode,
add support to the bytcr_rt5640 machine driver for such manually
instantiated rt5640 i2c_clients.
An example of a tablet which needs this is the Vexia EDU ATLA 10 tablet,
which has been distributed to schools in the Spanish Andalucía region.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://patch.msgid.link/20241024211615.79518-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
An uncomfortably large set of fixes due to me not getting round to
sending them for longer than I should due to travel and illness. This
is mostly smaller driver specific changes, but there are a couple of
generic changes:
- Bumping the minimal topology ABI we check for during validation, the
code had support for v4 removed previously but the update of the
define used for initial validation was missed.
- Fix the assumption that DAPM structs will be embedded in a component
which isn't true for card widgets when doing name comparisons, though
fortunately this is rarely triggered.
We've pulled in one Soundwire fix which was part of a larger series
fixing cleanup issues in on Intel Soundwire systems.
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Merge tag 'asoc-fix-v6.12-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.12
An uncomfortably large set of fixes due to me not getting round to
sending them for longer than I should due to travel and illness. This
is mostly smaller driver specific changes, but there are a couple of
generic changes:
- Bumping the minimal topology ABI we check for during validation, the
code had support for v4 removed previously but the update of the
define used for initial validation was missed.
- Fix the assumption that DAPM structs will be embedded in a component
which isn't true for card widgets when doing name comparisons, though
fortunately this is rarely triggered.
We've pulled in one Soundwire fix which was part of a larger series
fixing cleanup issues in on Intel Soundwire systems.
Old procedure has a chance to meet Headphone no output.
Fixes: c2d6af53a4 ("ALSA: hda/realtek - Add default procedure for suspend and resume state")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/17b717a0a0b04a77aea4a8ec820cba13@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 15c7fab0e0 ("ASoC: qcom: Move Soundwire runtime stream alloc to
soundcards") moved the allocation of Soundwire stream runtime from the
Qualcomm Soundwire driver to each individual machine sound card driver,
except that it forgot to update SC7280 card.
Just like for other Qualcomm sound cards using Soundwire, the card
driver should allocate and release the runtime. Otherwise sound
playback will result in a NULL pointer dereference or other effect of
uninitialized memory accesses (which was confirmed on SDM845 having
similar issue).
Cc: stable@vger.kernel.org
Cc: Alexey Klimov <alexey.klimov@linaro.org>
Cc: Steev Klimaszewski <steev@kali.org>
Fixes: 15c7fab0e0 ("ASoC: qcom: Move Soundwire runtime stream alloc to soundcards")
Link: https://lore.kernel.org/r/20241010054109.16938-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20241012101108.129476-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
On some platforms, for example i.MX93, there is only one
audio PLL source, so some sample rate can't be supported.
If the PLL source is used for 8kHz series rates, then 11kHz
series rates can't be supported.
So add constraints according to the frequency of available
clock sources, then alsa-lib will help to convert the
unsupported rate for the driver.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1728884313-6778-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the kconfig option for the tas2781 HDA driver to select CRC32 rather
than CRC32_SARWATE. CRC32_SARWATE is an option from the kconfig
'choice' that selects the specific CRC32 implementation. Selecting a
'choice' option seems to have no effect, but even if it did work, it
would be incorrect for a random driver to override the user's choice.
CRC32 is the correct option to select for crc32() to be available.
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Cc: stable@vger.kernel.org
Signed-off-by: Eric Biggers <ebiggers@google.com>
Link: https://patch.msgid.link/20241020175624.7095-1-ebiggers@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
