New HP ZBook with Realtek HDA codec ALC3247 needs the quirk
ALC236_FIXUP_HP_GPIO_LED to fix the micmute LED.
Signed-off-by: Chris Chiu <chris.chiu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250520132101.120685-1-chris.chiu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for HP Agusta.
Laptops use 2 CS35L41 Amps with HDA, using Internal boost, with I2C
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250520124757.12597-1-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of driver specific fixes that built up over the past few
weeks, plus one core patch which fixes handling of topology files where
some DAI names are substrings of others.
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Merge tag 'asoc-fix-v6.15-rc6' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.15
A collection of driver specific fixes that built up over the past few
weeks, plus one core patch which fixes handling of topology files where
some DAI names are substrings of others.
This board need to shutdown Class-D amp to avoid EMI issue.
Restore the Auto-Mute mode item will off pin control when Auto-mute mode was enable.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Links: https://lore.kernel.org/ee8bbe5236464c369719d96269ba8ef8@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM OSS layer tries to clear the buffer with the silence data at
initialization (or reconfiguration) of a stream with the explicit call
of snd_pcm_format_set_silence() with runtime->dma_area. But this may
lead to a UAF because the accessed runtime->dma_area might be freed
concurrently, as it's performed outside the PCM ops.
For avoiding it, move the code into the PCM core and perform it inside
the buffer access lock, so that it won't be changed during the
operation.
Reported-by: syzbot+32d4647f551007595173@syzkaller.appspotmail.com
Closes: https://lore.kernel.org/68164d8e.050a0220.11da1b.0019.GAE@google.com
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250516080817.20068-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Microdia JP001 does not support reading the sample rate which leads to
many lines of "cannot get freq at ep 0x84".
This patch adds the USB ID to quirks.c and avoids those error messages.
usb 7-4: New USB device found, idVendor=0c45, idProduct=636b, bcdDevice= 1.00
usb 7-4: New USB device strings: Mfr=2, Product=1, SerialNumber=3
usb 7-4: Product: JP001
usb 7-4: Manufacturer: JP001
usb 7-4: SerialNumber: JP001
usb 7-4: 3:1: cannot get freq at ep 0x84
Cc: <stable@vger.kernel.org>
Signed-off-by: Nicolas Chauvet <kwizart@gmail.com>
Link: https://patch.msgid.link/20250515102132.73062-1-kwizart@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
hda_generic_machine_select() appends -idisp to the tplg filename by
allocating a new string with devm_kasprintf(), then stores the string
right back into the global variable snd_soc_acpi_intel_hda_machines.
When the module is unloaded, this memory is freed, resulting in a global
variable pointing to freed memory. Reloading the module then triggers
a use-after-free:
BUG: KFENCE: use-after-free read in string+0x48/0xe0
Use-after-free read at 0x00000000967e0109 (in kfence-#99):
string+0x48/0xe0
vsnprintf+0x329/0x6e0
devm_kvasprintf+0x54/0xb0
devm_kasprintf+0x58/0x80
hda_machine_select.cold+0x198/0x17a2 [snd_sof_intel_hda_generic]
sof_probe_work+0x7f/0x600 [snd_sof]
process_one_work+0x17b/0x330
worker_thread+0x2ce/0x3f0
kthread+0xcf/0x100
ret_from_fork+0x31/0x50
ret_from_fork_asm+0x1a/0x30
kfence-#99: 0x00000000198a940f-0x00000000ace47d9d, size=64, cache=kmalloc-64
allocated by task 333 on cpu 8 at 17.798069s (130.453553s ago):
devm_kmalloc+0x52/0x120
devm_kvasprintf+0x66/0xb0
devm_kasprintf+0x58/0x80
hda_machine_select.cold+0x198/0x17a2 [snd_sof_intel_hda_generic]
sof_probe_work+0x7f/0x600 [snd_sof]
process_one_work+0x17b/0x330
worker_thread+0x2ce/0x3f0
kthread+0xcf/0x100
ret_from_fork+0x31/0x50
ret_from_fork_asm+0x1a/0x30
freed by task 1543 on cpu 4 at 141.586686s (6.665010s ago):
release_nodes+0x43/0xb0
devres_release_all+0x90/0xf0
device_unbind_cleanup+0xe/0x70
device_release_driver_internal+0x1c1/0x200
driver_detach+0x48/0x90
bus_remove_driver+0x6d/0xf0
pci_unregister_driver+0x42/0xb0
__do_sys_delete_module+0x1d1/0x310
do_syscall_64+0x82/0x190
entry_SYSCALL_64_after_hwframe+0x76/0x7e
Fix it by copying the match array with devm_kmemdup_array() before we
modify it.
Fixes: 5458411d75 ("ASoC: SOF: Intel: hda: refactoring topology name fixup for HDA mach")
Suggested-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Tavian Barnes <tavianator@tavianator.com>
Link: https://patch.msgid.link/570b15570b274520a0d9052f4e0f064a29c950ef.1747229716.git.tavianator@tavianator.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The function snd_es1968_capture_open() calls the function
snd_pcm_hw_constraint_pow2(), but does not check its return
value. A proper implementation can be found in snd_cx25821_pcm_open().
Add error handling for snd_pcm_hw_constraint_pow2() and propagate its
error code.
Fixes: b942cf815b ("[ALSA] es1968 - Fix stuttering capture")
Cc: stable@vger.kernel.org # v2.6.22
Signed-off-by: Wentao Liang <vulab@iscas.ac.cn>
Link: https://patch.msgid.link/20250514092444.331-1-vulab@iscas.ac.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A user reported on the Arch Linux Forums that their device is emitting
the following message in the kernel journal, which is fixed by adding
the quirk as submitted in this patch:
> kernel: usb 1-2: current rate 8436480 is different from the runtime rate 48000
There also is an entry for this product line added long time ago.
Their specific device has the following ID:
$ lsusb | grep Audio
Bus 001 Device 002: ID 1101:0003 EasyPass Industrial Co., Ltd Audioengine D1
Link: https://bbs.archlinux.org/viewtopic.php?id=305494
Fixes: 93f9d1a4ac ("ALSA: usb-audio: Apply sample rate quirk for Audioengine D1")
Cc: stable@vger.kernel.org
Signed-off-by: Christian Heusel <christian@heusel.eu>
Link: https://patch.msgid.link/20250512-audioengine-quirk-addition-v1-1-4c370af6eff7@heusel.eu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The acpi_handle should be just a handle and not a pointer in
sdw_intel_acpi_scan() parameter list.
It is called with 'acpi_handle handle' as parameter and it is passing it to
acpi_walk_namespace, which also expects acpi_handle and not acpi_handle*
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://patch.msgid.link/20250508181207.22113-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an event with UMP message is sent to a UMP client, the EP port
receives always no matter where the event is sent to, as it's a
catch-all port. OTOH, if an event is sent to EP port, and if the
event has a certain UMP Group, it should have been delivered to the
associated UMP Group port, too, but this was ignored, so far.
This patch addresses the behavior. Now a UMP event sent to the
Endpoint port will be delivered to the subscribers of the UMP group
port the event is associated with.
The patch also does a bit of refactoring to simplify the code about
__deliver_to_subscribers().
Fixes: 177ccf811d ("ALSA: seq: Support MIDI 2.0 UMP Endpoint port")
Link: https://patch.msgid.link/20250511134528.6314-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The partial matching of DAI widget to link names, can cause problems if
one of the widget names is a substring of another. E.g. with names
"Foo1" and Foo10", it's not possible to correctly link up "Foo1".
Modify the logic so that if multiple DAI links match the widget stream
name, prioritize a full match if one is found.
Fixes: fe88788779 ("ASoC: SOF: topology: Use partial match for connecting DAI link and DAI widget")
Link: https://github.com/thesofproject/linux/issues/5308
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Cc: stable@vger.kernel.org
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://patch.msgid.link/20250509085318.13936-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Keep using the PIO mode for commands on ACE2+ platforms, similarly how
the legacy stack is configured.
Fixes: 05cf17f1bf ("ASoC: SOF: Intel: hda-bus: Use PIO mode for Lunar Lake")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Cc: stable@vger.kernel.org
Link: https://patch.msgid.link/20250509081308.13784-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The firmware does not provide any information for capture streams via the
shared pipeline registers.
To avoid reporting invalid delay value for capture streams to user space
we need to disable it.
Fixes: af74dbd0db ("ASoC: SOF: ipc4-pcm: allocate time info for pcm delay feature")
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>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Link: https://patch.msgid.link/20250509085951.15696-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The header.numid is set to scontrol->comp_id in bytes_ext_get and it is
ignored during bytes_ext_put.
The use of comp_id is not quite great as it is kernel internal
identification number.
Set the header.numid to SOF_CTRL_CMD_BINARY during get and validate the
numid during put to provide consistent and compatible identification
number as IPC3.
For IPC4 existing tooling also ignored the numid but with the use of
SOF_CTRL_CMD_BINARY the different handling of the blobs can be dropped,
providing better user experience.
Reported-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
Closes: https://github.com/thesofproject/linux/issues/5282
Fixes: a062c8899f ("ASoC: SOF: ipc4-control: Add support for bytes control get and put")
Cc: stable@vger.kernel.org
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Link: https://patch.msgid.link/20250509085633.14930-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 0116a7d84b ("ASoC: mediatek: mt6359: Add stub for
mt6359_accdet_enable_jack_detect") added a stub for
mt6359_accdet_enable_jack_detect() in order to allow the mt8188-mt6359
driver to be enabled without requiring the mt6359-accdet to also be
enabled, since it is not always needed.
However, in the case that CONFIG_SND_SOC_MT8188_MT6359=y and
CONFIG_SND_SOC_MT6359_ACCDET=m, a link error will happen, which commit
b19fa45715 ("ASoC: mediatek: mt8188-mt6359: select
CONFIG_SND_SOC_MT6359_ACCDET") solved by selecting
CONFIG_SND_SOC_MT6359_ACCDET.
In order to not require CONFIG_SND_SOC_MT6359_ACCDET as originally
intended, but also prevent the link error, depend on ACCDET being
enabled or disabled (which will force MT8188_MT6359=m if
MT6359_ACCDET=m).
Fixes: f35d834d67 ("ASoC: mediatek: mt8188-mt6359: Add accdet headset jack detect support")
Fixes: b19fa45715 ("ASoC: mediatek: mt8188-mt6359: select CONFIG_SND_SOC_MT6359_ACCDET")
Suggested-by: Arnd Bergmann <arnd@kernel.org>
Signed-off-by: Nícolas F. R. A. Prado <nfraprado@collabora.com>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Link: https://patch.msgid.link/20250507-mt8188-mt6359-accdet-depend-v1-1-aad70ce62964@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver support was added without selecting the codec, which leads to
a link failure:
aarch64-linux-ld: sound/soc/mediatek/mt8188/mt8188-mt6359.o: in function `mt8188_mt6359_init':
mt8188-mt6359.c:(.text+0x19f0): undefined reference to `mt6359_accdet_enable_jack_detect'
Fixes: f35d834d67 ("ASoC: mediatek: mt8188-mt6359: Add accdet headset jack detect support")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://patch.msgid.link/20250505052106.1811802-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
A moderately large batch of fixes for v6.15, many driver specific
including cleanups for the enabling of the Cirrus KUnit tests and a fix
for a nasty crash on resume on AMD systems. We also have one core fix,
for an ordering issue between DAPM and DPCM which could leave things
incorrectly unpowered.
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Merge tag 'asoc-fix-v6.15-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.15
A moderately large batch of fixes for v6.15, many driver specific
including cleanups for the enabling of the Cirrus KUnit tests and a fix
for a nasty crash on resume on AMD systems. We also have one core fix,
for an ordering issue between DAPM and DPCM which could leave things
incorrectly unpowered.
Merge series from Stefan Binding <sbinding@opensource.cirrus.com>:
Both CS35L56 and CS42L43 have maximum volumes above 0dB.
However, for many use cases, this can cause distorted audio, depending
various factors, such as other signal-processing elements in the chain,
for example if the audio passes through a gain control before reaching
the amp or the signal path has been tuned for a particular maximum
gain in the amp.
In the cases where systems use the soc_sdw_* drivers, audio above the
0dB volume will likely always be distorted, therefore apply a 0dB
limit to those devices.
Stefan Binding (2):
ASoC: intel/sdw_utils: Add volume limit to cs42l43 speakers
ASoC: intel/sdw_utils: Add volume limit to cs35l56 speakers
include/sound/soc_sdw_utils.h | 1 +
sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c | 4 ++++
sound/soc/sdw_utils/soc_sdw_cs42l43.c | 10 ++++++++
sound/soc/sdw_utils/soc_sdw_cs_amp.c | 24 ++++++++++++++++++++
4 files changed, 39 insertions(+)
--
2.43.0
Merge series from Olivier Moysan <olivier.moysan@foss.st.com>:
This patchset adds some checks on kernel minimum rate requirements.
This avoids potential clock rate misconfiguration, when setting the
kernel frequency on STM32MP2 SoCs.
If any Soundwire manager interrupt is reported, and wake interrupt
is not reported, in this scenario irq_flag will be set to zero,
which results in interrupt handler return status as IRQ_NONE.
Add new irq flag 'wake_irq_flag' check for SoundWire wake interrupt
handling to fix incorrect irq handling return status.
Fixes: 3898b18907 ("ASoC: amd: ps: add soundwire wake interrupt handling")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://patch.msgid.link/20250430195517.3065308-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Actually check if the passed pointers are valid, before writing to them.
This also fixes a USBAN warning:
UBSAN: invalid-load in ../sound/soc/fsl/imx-card.c:687:25
load of value 255 is not a valid value for type '_Bool'
This is because playback_only is uninitialized and is not written to, as
the playback-only property is absent.
Fixes: 844de7eebe ("ASoC: audio-graph-card2: expand dai_link property part")
Signed-off-by: Alexander Stein <alexander.stein@ew.tq-group.com>
Link: https://patch.msgid.link/20250429094910.1150970-1-alexander.stein@ew.tq-group.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The volume control for cs35l56 speakers has a maximum gain of +12 dB.
However, for many use cases, this can cause distorted audio, depending
various factors, such as other signal-processing elements in the chain,
for example if the audio passes through a gain control before reaching
the amp or the signal path has been tuned for a particular maximum
gain in the amp.
In the case of systems which use the soc_sdw_* driver, audio will
likely be distorted in all cases above 0 dB, therefore add a volume
limit of 400, which is 0 dB maximum volume inside this driver.
The volume limit should be applied to both soundwire and soundwire
bridge configurations.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://patch.msgid.link/20250430103134.24579-3-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The volume control for cs42l43 speakers has a maximum gain of +31.5 dB.
However, for many use cases, this can cause distorted audio, depending
various factors, such as other signal-processing elements in the chain,
for example if the audio passes through a gain control before reaching
the codec or the signal path has been tuned for a particular maximum
gain in the codec.
In the case of systems which use the soc_sdw_cs42l43 driver, audio will
likely be distorted in all cases above 0 dB, therefore add a volume
limit of 128, which is 0 dB maximum volume inside this driver.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20250430103134.24579-2-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On MP2 SoCs SAI kernel clock rate is managed through
stm32_sai_set_parent_rate() function.
If the kernel clock rate was set previously to a low frequency, this
frequency may be too low to support the newly requested audio stream rate.
However the stm32_sai_rate_accurate() will only check accuracy against
the maximum kernel clock rate. The function will return leaving the kernel
clock rate unchanged.
Add a check on minimal frequency requirement, to avoid this.
Fixes: 2cfe1ff225 ("ASoC: stm32: sai: add stm32mp25 support")
Signed-off-by: Olivier Moysan <olivier.moysan@foss.st.com>
Link: https://patch.msgid.link/20250430165210.321273-3-olivier.moysan@foss.st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
the frequency of the kernel clock must be greater than or equal to the
bitclock rate. When searching for a convenient kernel clock rate in
stm32_sai_set_parent_rate() function, it is useless to continue the loop
below bitclock rate, as it will result in a invalid kernel clock rate.
Change the loop output condition.
Fixes: 2cfe1ff225 ("ASoC: stm32: sai: add stm32mp25 support")
Signed-off-by: Olivier Moysan <olivier.moysan@foss.st.com>
Link: https://patch.msgid.link/20250430165210.321273-2-olivier.moysan@foss.st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
More HP EliteBook with Realtek HDA codec ALC3247 and combined CS35L56
Amplifiers need quirk ALC236_FIXUP_HP_GPIO_LED to fix the micmute LED.
Signed-off-by: Chris Chiu <chris.chiu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250430101843.150833-1-chris.chiu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few ASUS models use the ALC256_FIXUP_ASUS_HEADSET_MODE although they
have no built-in mic pin on NID 0x13, as found in the commit
c1732ede5e ("ALSA: hda/realtek - Fix headset and mic on several Asus
laptops with ALC256"). This was relatively harmless in the past as
NID 0x13 was assigned as the secondary mic. But since the fix for the
pin sort order, this pin became the primary one, hence user started
noticing the broken input, and we've fixed already for a few ASUS
models to switch to ALC256_FIXUP_ASUS_MIC_NO_PRESENCE.
This patch corrects the other ASUS models to use the right quirk entry
for fixing the built-in mic regression. Here we cover X541SA
(1043:12e0), X541UV (1043:12f0), Z550SA (1043:13bf0) and X555UB
(1043:1ccd).
Fixes: 3b4309546b ("ALSA: hda: Fix headset detection failure due to unstable sort")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=220058
Link: https://patch.msgid.link/20250430053210.31776-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Depending on the architecture __ffs() returns either an 'unsigned long'
or 'unsigned int' result. Compile-testing this driver on targets that
use the latter produces a warning:
sound/soc/intel/catpt/dsp.c: In function 'catpt_dsp_set_srampge':
sound/soc/intel/catpt/dsp.c:181:44: error: format '%ld' expects argument of type 'long int', but argument 4 has type 'u32' {aka 'unsigned int'} [-Werror=format=]
181 | dev_dbg(cdev->dev, "sanitize block %ld: off 0x%08x\n",
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Change the type of the local variable to match the format string and
avoid the warning on any architecture.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20250429073545.3558494-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The MIDI substream name string is constructed from the combination of
the card shortname (which is taken from USB iProduct) and the USB
iJack. The problem is that some devices put the product name to the
iJack field, too. For example, aplaymidi -l output on the Lanchkey MK
49 are like:
% aplaymidi -l
Port Client name Port name
44:0 Launchkey MK4 49 Launchkey MK4 49 Launchkey MK4
44:1 Launchkey MK4 49 Launchkey MK4 49 Launchkey MK4
where the actual iJack name can't be seen because it's truncated due
to the doubly words.
For resolving those situations, this patch compares the iJack string
with the card shortname, and drops if both start with the same words.
Then the result becomes like:
% aplaymidi -l
Port Client name Port name
40:0 Launchkey MK4 49 Launchkey MK4 49 MIDI In
40:1 Launchkey MK4 49 Launchkey MK4 49 DAW In
A caveat is that there are some pre-defined names for certain
devices in the driver code, and this workaround shouldn't be applied
to them. Similarly, when the iJack isn't specified, we should skip
this check, too. The patch added those checks in addition to the
string comparison.
Suggested-by: Paul Davis <paul@linuxaudiosystems.com>
Tested-by: Paul Davis <paul@linuxaudiosystems.com>
Link: https://lore.kernel.org/CAFa_cKmEDQWcJatbYWi6A58Zg4Ma9_6Nr3k5LhqwyxC-P_kXtw@mail.gmail.com
Link: https://patch.msgid.link/20250429183626.20773-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There seem to be multiple USB device IDs used for these;
the one I have reports as 0b0e:030c when powered on.
(When powered off, it reports as 0b0e:0311.)
Signed-off-by: Joachim Priesner <joachim.priesner@web.de>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250428053606.9237-1-joachim.priesner@web.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This hardware has ALC274 codec with speaker NID 0x17 and line out
NID 0x16 for audio output. The line out is routed correctly but
the speaker is not. Thus the volume can't be controlled.
This patch removes DAC NID 0x06 (without volume control) from the
connection list for speaker NID 0x17. Routing both speaker and line
out pins to DAC NID 0x02 which controls the output volume.
Signed-off-by: Chris Chiu <chris.chiu@canonical.com>
Link: https://patch.msgid.link/20250425103618.534951-1-chris.chiu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update chip data using dev_get_drvdata(dev->parent) to fix
NULL pointer deref in acp_i2s_set_tdm_slot.
Fixes: cd60dec899 ("ASoC: amd: acp: Refactor TDM slots selction based on acp revision id")
Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu@amd.com>
Link: https://patch.msgid.link/20250425060144.1773265-2-venkataprasad.potturu@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the latest kernel versions system crashes were noticed occasionally
during suspend/resume. This occurs because the RZ SSI suspend trigger
(called from snd_soc_suspend()) is executed after rz_ssi_pm_ops->suspend()
and it accesses IP registers. After the rz_ssi_pm_ops->suspend() is
executed the IP clocks are disabled and its reset line is asserted.
Since snd_soc_suspend() is invoked through snd_soc_pm_ops->suspend(),
snd_soc_pm_ops is associated with soc_driver (defined in
sound/soc/soc-core.c), and there is no parent-child relationship between
soc_driver and rz_ssi_driver the power management subsystem does not
enforce a specific suspend/resume order between the RZ SSI platform driver
and soc_driver.
To ensure that the suspend/resume function of rz-ssi is executed after
snd_soc_suspend(), use NOIRQ_SYSTEM_SLEEP_PM_OPS().
Fixes: 1fc778f7c8 ("ASoC: renesas: rz-ssi: Add suspend to RAM support")
Cc: stable@vger.kernel.org
Signed-off-by: Claudiu Beznea <claudiu.beznea.uj@bp.renesas.com>
Link: https://patch.msgid.link/20250410141525.4126502-1-claudiu.beznea.uj@bp.renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The headphone clamps cause fairly loud pops during type detect
because they sink current from the detection process itself. Disable
the clamps whilst the type detect runs, to improve the detection
pop performance.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20250423090944.1504538-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
During initialisation of Focusrite USB audio interfaces, -EPROTO is
sometimes returned from usb_set_interface(), which sometimes prevents
the device from working: subsequent usb_set_interface() and
uac_clock_source_is_valid() calls fail.
This patch adds up to 5 retries in endpoint_set_interface(), with a
delay starting at 5ms and doubling each time. 5 retries was chosen to
allow for longer than expected waits for the interface to start
responding correctly; in testing, a single 5ms delay was sufficient to
fix the issue.
Closes: https://github.com/geoffreybennett/fcp-support/issues/2
Cc: stable@vger.kernel.org
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://patch.msgid.link/Z//7s9dKsmVxHzY2@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The speaker doesn't mute when plugged headphone.
This platform support 4ch speakers.
The speaker pin 0x14 wasn't fill verb table.
After assigned model ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX.
The speaker can mute when headphone was plugged.
Fixes: aa8e3ef4fe ("ALSA: hda/realtek: Add quirks for various HP ENVY models")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/eb4c14a4d85740069c909e756bbacb0e@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
mic_name returned by devm_kasprintf() could be NULL.
Add a check for it.
Signed-off-by: Chenyuan Yang <chenyuan0y@gmail.com>
Fixes: bee2fe4467 ("ASoC: Intel: sof_sdw: use generic rtd_init function for Realtek SDW DMICs")
Link: https://patch.msgid.link/20250415194134.292830-1-chenyuan0y@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
This series fixes the KConfig for cs_dsp and cs-amp-lib tests so that
CONFIG_KUNIT_ALL_TESTS doesn't cause them to add modules to the build.
Issue:
When multiple audio streams share a common BE DAI, the BE DAI
widget can be powered up before its hardware parameters are configured.
This incorrect sequence leads to intermittent pcm_write errors.
For example, the below Tegra use-case throws an error:
aplay(2 streams) -> AMX(mux) -> ADX(demux) -> arecord(2 streams),
here, 'AMX TX' and 'ADX RX' are common BE DAIs.
For above usecase when failure happens below sequence is observed:
aplay(1) FE open()
- BE DAI callbacks added to the list
- BE DAI state = SND_SOC_DPCM_STATE_OPEN
aplay(2) FE open()
- BE DAI callbacks are not added to the list as the state is
already SND_SOC_DPCM_STATE_OPEN during aplay(1) FE open().
aplay(2) FE hw_params()
- BE DAI hw_params() callback ignored
aplay(2) FE prepare()
- Widget is powered ON without BE DAI hw_params() call
aplay(1) FE hw_params()
- BE DAI hw_params() is now called
Fix:
Add BE DAIs in the list if its state is either SND_SOC_DPCM_STATE_OPEN
or SND_SOC_DPCM_STATE_HW_PARAMS as well.
It ensures the widget is powered ON after BE DAI hw_params() callback.
Fixes: 0c25db3f76 ("ASoC: soc-pcm: Don't reconnect an already active BE")
Signed-off-by: Sheetal <sheetal@nvidia.com>
Link: https://patch.msgid.link/20250404105953.2784819-1-sheetal@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Incorrect types are used as sizeof() arguments in devm_kcalloc().
It should be sizeof(dai_link_data) for link_data instead of
sizeof(snd_soc_dai_link).
This is found by our static analysis tool.
Signed-off-by: Chenyuan Yang <chenyuan0y@gmail.com>
Link: https://patch.msgid.link/20250406210854.149316-1-chenyuan0y@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Depend on SND_SOC_CS_AMP_LIB instead of selecting it.
KUNIT_ALL_TESTS should only build tests for components that are
already being built, it should not cause other stuff to be added
to the build.
Fixes: 177862317a ("ASoC: cs-amp-lib: Add KUnit test for calibration helpers")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250411123608.1676462-3-rf@opensource.cirrus.com
Reviewed-by: David Gow <davidgow@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A set of small fixes, quirks and device ID additions that came in since
-rc1, none of them super stand out. There's also a change to Srini's
email address in MAINTAINERS.
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Merge tag 'asoc-fix-v6.15-rc1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.15
A set of small fixes, quirks and device ID additions that came in since
-rc1, none of them super stand out. There's also a change to Srini's
email address in MAINTAINERS.
