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commit 3bfd7c0ba184de99e9f5083b29e5308f30767265 upstream.
When the UMP Endpoint is configured with static blocks, the block
configuration will never change, hence the unused ports will be
unchanged as well. Creating sequencer ports for those unused ports
is simply useless, and it might be rather confusing for users.
The idea behind the inactive ports was for allowing connections
from/to ports that can become usable later, but this will never
happen for inactive groups in static blocks.
Let's change the sequencer UMP binding to skip those unused ports when
the UMP EP is with static blocks.
Fixes: 81fd444aa371 ("ALSA: seq: Bind UMP device")
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20240717083322.25892-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 88e98af9f4b5b0d60c1fe7f7f2701b5467691e75 upstream.
When suspended, the DMA channel may enter PAUSE state if dmaengine_pause()
is supported by DMA.
At this state, dmaengine_synchronize() should not be called, otherwise
the DMA channel can't be resumed successfully.
Fixes: e8343410ddf0 ("ALSA: dmaengine: Synchronize dma channel after drop()")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/1721198693-27636-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d7063c08738573fc2f3296da6d31a22fa8aa843a upstream.
Samsung Galaxy Book Pro 360 (13" 2022 NT935QDB-KC71S) with codec SSID
144d:c1a4 requires the same workaround to enable the speaker amp
as other Samsung models with the ALC298 codec.
Signed-off-by: Seunghun Han <kkamagui@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20240718080908.8677-1-kkamagui@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 8fc1e8b230771442133d5cf5fa4313277aa2bb8b upstream.
Positivo SU C1400 is equipped with ALC256, and it needs
ALC269_FIXUP_ASPIRE_HEADSET_MIC quirk to make its headset mic work.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20240712180642.22564-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c66995ae403073212f5ba60d2079003866c6e130 upstream.
The "Speaker Volume" control was being defined using four hardcoded magic
numbers. There are #defines in the cs35l56.h header for these numbers, so
change the code to use the defined constants.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20240703095517.208077-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
A few fairly small fixes for ASoC, there's a relatively large set of
hardening changes for the cs_dsp firmware file parsing and a couple of
other small device specific fixes.
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Merge tag 'asoc-fix-v6.10-rc7' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.10
A few fairly small fixes for ASoC, there's a relatively large set of
hardening changes for the cs_dsp firmware file parsing and a couple of
other small device specific fixes.
HP 250 G7 has a mute LED that can be made to work using quirk
ALC269_FIXUP_HP_LINE1_MIC1_LED. Enable already existing quirk.
Signed-off-by: Nazar Bilinskyi <nbilinskyi@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20240709080546.18344-1-nbilinskyi@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The internal mic boost on the VAIO models VJFE-CL and VJFE-IL is too high.
Fix this by applying the ALC269_FIXUP_LIMIT_INT_MIC_BOOST fixup to the machine
to limit the gain.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20240705141012.5368-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HDaudio specification Section 3.6.2 limits the number of BDL entries to 256.
Make sure we don't allow more periods than this normative value.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://patch.msgid.link/20240704090106.371497-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Those registers will be used when JD source is RT711_JD2_1P8V_1PORT.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Shuming Fan <shumingf@realtek.com>
Link: https://patch.msgid.link/20240704092327.652609-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When system enters suspend with an active stream, SOF core
calls hw_params_upon_resume(). On Intel platforms with HDA DMA used
to manage the link DMA, this leads to call chain of
hda_dsp_set_hw_params_upon_resume()
-> hda_dsp_dais_suspend()
-> hda_dai_suspend()
-> hda_ipc4_post_trigger()
A bug is hit in hda_dai_suspend() as hda_link_dma_cleanup() is run first,
which clears hext_stream->link_substream, and then hda_ipc4_post_trigger()
is called with a NULL snd_pcm_substream pointer.
Fixes: 2b009fa0823c ("ASoC: SOF: Intel: hda: Unify DAI drv ops for IPC3 and IPC4")
Link: https://github.com/thesofproject/linux/issues/5080
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://patch.msgid.link/20240704085708.371414-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A relatively large batch of updates, largely due to the long interval
since I last sent fixes due to various travel and holidays. There's a
lot of driver specific fixes and quirks in here, none of them too major,
and also some fixes for recently introduced memory safety issues in the
topology code.
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Merge tag 'asoc-fix-v6.10-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.10
A relatively large batch of updates, largely due to the long interval
since I last sent fixes due to various travel and holidays. There's a
lot of driver specific fixes and quirks in here, none of them too major,
and also some fixes for recently introduced memory safety issues in the
topology code.
Modify register setting sequence of enabling inline command
to fix issue of random interrupt from push-button.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://patch.msgid.link/9a7a3a66cbcb426487ca6f558f45e922@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The conversion of SPP to MIDI2 UMP called a wrong function, and the
secondary argument wasn't taken. As a result, MSB of SPP was always
zero. Fix to call the right function.
Fixes: e9e02819a98a ("ALSA: seq: Automatic conversion of UMP events")
Link: https://patch.msgid.link/20240626145141.16648-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Vivobook S 16X IPS needs a quirks-table entry for the internal microphone to function properly.
Signed-off-by: Vyacheslav Frantsishko <itmymaill@gmail.com>
Reviewed-by: Mario Limonciello <mario.limonciello@amd.com>
Link: https://patch.msgid.link/20240626070334.45633-1-itmymaill@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The recent fix for Lenovo IdeaPad 330-17IKB replaced the quirk entry,
and this eventually breaks the existing quirk for Lenovo Yoga Duet 7
13ITL6 equipped with the same PCI SSID 17aa:3820.
For applying a proper quirk for each model, check the codec SSID
additionally. Fortunately Yoga Duet has a different codec SSID,
0x17aa3802.
(Interestingly, 17aa:3802 has another conflict of SSID between another
Yoga model vs 14IRP8 which we had to work around similarly.)
Fixes: b1fd0d1285b1 ("ALSA: hda/realtek: Enable headset mic on IdeaPad 330-17IKB 81DM")
Link: https://patch.msgid.link/20240625155217.18767-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When dmaengine supports pause function, in suspend state,
dmaengine_pause() is called instead of dmaengine_terminate_async(),
In end of playback stream, the runtime->state will go to
SNDRV_PCM_STATE_DRAINING, if system suspend & resume happen
at this time, application will not resume playback stream, the
stream will be closed directly, the dmaengine_terminate_async()
will not be called before the dmaengine_synchronize(), which
violates the call sequence for dmaengine_synchronize().
This behavior also happens for capture streams, but there is no
SNDRV_PCM_STATE_DRAINING state for capture. So use
dmaengine_tx_status() to check the DMA status if the status is
DMA_PAUSED, then call dmaengine_terminate_async() to terminate
dmaengine before dmaengine_synchronize().
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1718851218-27803-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This HP Laptop uses ALC236 codec with COEF 0x07 controlling
the mute LED. Enable existing quirk for this device.
Signed-off-by: Aivaz Latypov <reichaivaz@gmail.com>
Link: https://patch.msgid.link/20240625081217.1049-1-reichaivaz@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The conversion from the legacy event to MIDI2 UMP for RPN and NRPN
missed the setup of the channel number, resulting in always the
channel 0. Fix it.
Fixes: e9e02819a98a ("ALSA: seq: Automatic conversion of UMP events")
Link: https://patch.msgid.link/20240625095200.25745-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit e70b8dd26711 ("ASoC: mediatek: mt8195: Remove afe-dai component
and rework codec link") removed the codec entry for the ETDM1_OUT_BE
dai link entirely instead of replacing it with COMP_EMPTY(). This worked
by accident as the remaining COMP_EMPTY() platform entry became the codec
entry, and the platform entry became completely empty, effectively the
same as COMP_DUMMY() since snd_soc_fill_dummy_dai() doesn't do anything
for platform entries.
This causes a KASAN out-of-bounds warning in mtk_soundcard_common_probe()
in sound/soc/mediatek/common/mtk-soundcard-driver.c:
for_each_card_prelinks(card, i, dai_link) {
if (adsp_node && !strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF")))
dai_link->platforms->of_node = adsp_node;
else if (!dai_link->platforms->name && !dai_link->platforms->of_node)
dai_link->platforms->of_node = platform_node;
}
where the code expects the platforms array to have space for at least one entry.
Add an COMP_EMPTY() entry so that dai_link->platforms has space.
Fixes: e70b8dd26711 ("ASoC: mediatek: mt8195: Remove afe-dai component and rework codec link")
Signed-off-by: Chen-Yu Tsai <wenst@chromium.org>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://patch.msgid.link/20240624061257.3115467-1-wenst@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
priv->pdev pointer was set after being used in
fsl_asoc_card_audmux_init().
Move this assignment at the start of the probe function, so
sub-functions can correctly use pdev through priv.
fsl_asoc_card_audmux_init() dereferences priv->pdev to get access to the
dev struct, used with dev_err macros.
As priv is zero-initialised, there would be a NULL pointer dereference.
Note that if priv->dev is dereferenced before assignment but never used,
for example if there is no error to be printed, the driver won't crash
probably due to compiler optimisations.
Fixes: 708b4351f08c ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support")
Signed-off-by: Elinor Montmasson <elinor.montmasson@savoirfairelinux.com>
Link: https://patch.msgid.link/20240620132511.4291-2-elinor.montmasson@savoirfairelinux.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The ACPI IDs used in the CS35L56 HDA drivers are all handled by the
serial multi-instantiate driver which starts multiple Linux device
instances from a single ACPI Device() node.
As serial multi-instantiate is not an optional part of the system add it
as a dependency in Kconfig so that it is not overlooked.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Link: https://lore.kernel.org/20240619161602.117452-1-simont@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
chip->flag variable assignment will be skipped when acp platform device
creation is skipped. In this case chip>flag value will not be set.
chip->flag variable should be assigned along with other structure
variables for 'chip' structure. Move chip->flag variable assignment
prior to acp platform device creation.
Fixes: 3a94c8ad0aae ("ASoC: amd: acp: add code for scanning acp pdm controller")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240617072844.871468-3-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP supports different pin configurations for I2S IO. Checking ACP pin
configuration value against specific value breaks the functionality for
other I2S pin configurations. This check is no longer required in i2s dai
driver probe call as i2s configuration check will be verified during acp
platform device creation sequence.
Remove i2s_mode check in acp_i2s_probe() function.
Fixes: b24484c18b10 ("ASoC: amd: acp: ACP code generic to support newer platforms")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240617072844.871468-2-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When acp platform device creation is skipped, chip->chip_pdev value will
remain NULL. Add NULL check for chip->chip_pdev structure in
snd_acp_resume() function to avoid null pointer dereference.
Fixes: 088a40980efb ("ASoC: amd: acp: add pm ops support for acp pci driver")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240617072844.871468-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Dell SKU 0C64 has a single rt1318 amplifier.
The prefix name of control still needs to be set rt1318-1 corresponding to UCM config.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240612075740.1678082-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With ARCH=m68k, make allmodconfig && make W=1 C=1 reports:
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/oss/dmasound/dmasound_core.o
Add the missing invocation of the MODULE_DESCRIPTION() macro.
Signed-off-by: Jeff Johnson <quic_jjohnson@quicinc.com>
Reviewed-by: Geert Uytterhoeven <geert@linux-m68k.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/20240617-md-m68k-sound-oss-dmasound-v1-1-5c19306be930@quicinc.com
Similarly to other Lenovo laptops these also have a dual speaker
setup with a shared amplifier.
The model also seems to have a conflicting PCI SSID with the codec SSID for
the Legion Y9000X 2022 IAH7. Only tested on the Yoga Pro 7, as I don't have
access to the other laptop.
Signed-off-by: Gergely Meszaros <meszaros.gergely97@gmail.com>
Link: https://lore.kernel.org/r/20240616085233.16922-1-meszaros.gergely97@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is an issue around with error handling and graph management with
the exising code, none of the error paths close the graph, which result in
leaving the loaded graph in dsp, however the driver thinks otherwise.
This can have a nasty side effect specially when we try to load the same
graph to dsp, dsp returns error which leaves the board with no sound and
requires restart.
Fix this by properly closing the graph when we hit errors between
open and close.
Fixes: 30ad723b93ad ("ASoC: qdsp6: audioreach: add q6apm lpass dai support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Tested-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org> # X13s
Link: https://lore.kernel.org/r/20240613-q6apm-fixes-v1-1-d88953675ab3@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
If the ASP1 DAI is hooked up by the machine driver the ASP TX mixer
sources should be initialized to disconnected. There aren't currently
any available products using the ASP so this doesn't affect any
existing systems.
The cs35l56 does not have any fixed default for the mixer source
registers. When the cs35l56 boots, its firmware patches these registers
to setup a system-specific routing; this is so that Windows can use
generic SDCA drivers instead of needing knowledge of chip-specific
registers. The setup varies between end-products, which each have
customized firmware, and so the default register state varies between
end-products. It can also change if the firmware on an end-product is
upgraded - for example if a change was needed to the routing for Windows
use-cases. It must be emphasized that the settings applied by the
firmware are not internal magic tuning; they are statically implementing
use-case setup that on Linux would be done via ALSA controls.
The driver is currently syncing the mixer controls with whatever
initial state the firmware wrote to the registers, so that they report
the actual audio routing. But if the ASP DAI is hooked up this can create
a powered-up DAPM graph without anything intentionally setting up a path.
This can lead to parts of the audio system powering up unexpectedly.
For example when cs35l56 is connected to cs42l43 using a codec-codec link,
this can create a complete DAPM graph which then powers-up cs42l43. But
the cs42l43 can only be clocked from its SoundWire bus so this causes a
bunch of errors in the kernel log where cs42l43 is unexpectedly powered-up
without a clock.
If the host is taking ownership of the ASP (either directly or as a
codec-to-codec link) there is no need to keep the mixer settings that the
firmware wrote. The driver has ALSA controls for setting these using
standard Linux mechanisms. So if the machine driver hooks up the ASP the
ASP mixers are initialized to "None" (no input). This prevents unintended
DAPM-graph power-ups, and means the initial state of the mixers is
always going to be None.
Since the initial state of the mixers can vary from system to system and
potentially between firmware upgrades, no use-case manager can currently
assume that cs35l56 has a known initial state. The firmware could just as
easily default them to "None" as to any input source. So defaulting them
to "None" in the driver is not increasing the entropy of the system.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240613132527.46537-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The interface associated with the hda_component should be deactivated
before the driver is deconstructed during removal.
Fixes: 4e7914eb1dae ("ALSA: hda/tas2781: remove sound controls in unbind")
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240613133713.75550-4-simont@opensource.cirrus.com
The interface associated with the hda_component should be deactivated
before the driver is deconstructed during removal.
Fixes: 7b2f3eb492da ("ALSA: hda: cs35l41: Add support for CS35L41 in HDA systems")
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240613133713.75550-3-simont@opensource.cirrus.com
The interface associated with the hda_component should be deactivated
before the driver is deconstructed during removal.
Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240613133713.75550-2-simont@opensource.cirrus.com
dsp_driver=4 will force the AVS driver stack to be used, it is better to
docuement this.
Fixes: 1affc44ea5dd ("ASoC: Intel: avs: PCI driver implementation")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240607060021.11503-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for this laptop, which uses CS35L41 HDA amps.
The laptop does not contain valid _DSD for these amps, so requires
entries into the CS35L41 configuration table to function correctly.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240606130351.333495-5-sbinding@opensource.cirrus.com
Add support for this laptop, which uses CS35L41 HDA amps.
The laptop does not contain valid _DSD for these amps, so requires
entries into the CS35L41 configuration table to function correctly.
[ fixed to lower hex numbers in quirk entries -- tiwai ]
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240606130351.333495-4-sbinding@opensource.cirrus.com
The Framework Laptop 16 does not have a combination headphone/headset
3.5mm jack; however, applying the pincfg from the Laptop 13 (nid=0x19)
erroneously informs hda that the node is present.
Fixes: 8804fa04a492 ("ALSA: hda/realtek: Add Framework laptop 16 to quirks")
Signed-off-by: Dustin L. Howett <dustin@howett.net>
Reviewed-by: Mario Limonciello <mario.limonciello@amd.com>
Link: https://lore.kernel.org/r/20240605-alsa-hda-realtek-remove-framework-laptop-16-from-quirks-v1-1-11d47fe8ec4d@howett.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The internal mic boost on the N14AP7 is too high. Fix this by applying the
ALC269_FIXUP_LIMIT_INT_MIC_BOOST fixup to the machine to limit the gain.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240605153923.2837-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP ProBook 445/465 G11 needs ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF quirk to
make mic-mute/audio-mute working.
Signed-off-by: Andy Chi <andy.chi@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240605092243.41963-1-andy.chi@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
