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ASoC: Fixes for v7.1
A bigger batch of fixes than usual due to -next not happeing last week,
this is mostly stuff for laptops - a lot of quirks and small fixes,
mainly for x86 and SoundWire. Nothing too big or exciting individually,
just two week's worth.
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In the pcm512x chipset driver, pcm512x_overclock_xxx_put() is defined as
a general mixer kcontrol instead of a DAPM kcontrol, so struct
snd_soc_dapm_context must not be accessed via
snd_soc_dapm_kcontrol_to_dapm().
This causes a NULL pointer dereference, so it must be modified to use
snd_soc_component_to_dapm().
Cc: stable@kernel.org
Closes: https://github.com/raspberrypi/linux/issues/7242
Fixes: 02dbbb7e982a ("ASoC: codecs: pcm512x: convert to snd_soc_dapm_xxx()")
Signed-off-by: Jeongjun Park <aha310510@gmail.com>
Link: https://patch.msgid.link/20260521113712.227438-1-aha310510@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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For PTL onwards Cirrus are intending to rely on function topologies,
rather than using a match table for each system type. Remove this
unnecessary match table entry. Having the match entries can
mean that systems match when they should use function topologies
instead, resulting in incorrect audio configurations. Although,
admittedly this is not too likely with this 6x amp configuration
as those are quite rare, but best to follow best practice.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260520163631.3300102-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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For PTL onwards Cirrus are intending to rely on function
topologies, rather than using a match table for each system
type. Chrome systems tend to have custom magic in the topology
and thus need to load a specific file. This causes problems as
these system can have the same layout as generic laptops causing
the match to apply to other laptops. Add a DMI quirk that forces
these matches to only apply to specific devices.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260520163631.3300102-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Adding ES83x6 I2S codec support for NVL platforms and entry in match table.
Signed-off-by: Balamurugan C <balamurugan.c@intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20260520061143.2024963-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add an entry in the soundwire quirk table for novalake boards to support
NVL RVP
Signed-off-by: Jairaj Arava <jairaj.arava@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20260520060814.2024852-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Richard Fitzgerald <rf@opensource.cirrus.com> says:
This series fixes bugs and a typo in cs-amp-lib.
Link: https://patch.msgid.link/20260521122511.987322-1-rf@opensource.cirrus.com
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Fix the error message in cs_amp_read_cal_coeff() to say "Failed to read".
It was incorrectly "Failed to write", probably a copy-paste error.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260521122511.987322-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Rewrite cs_amp_create_debugfs() so that dput() will be called on
a valid dentry returned from debugfs_lookup().
The pointer returned from debugfs_lookup() must be released by dput().
The pointer returned from debugfs_create_dir() does not need to be
passed to dput().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: cdd27fa3298a ("ASoC: cs-amp-lib: Add helpers for factory calibration")
Link: https://patch.msgid.link/20260521122511.987322-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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When calculating data->count replace the incorrect sizeof(data) with use
of struct_offset().
The faulty sizeof(data) was incorrectly calculating the size of the
pointer instead of the size of the struct pointed to. As it happens, both
values are 8 on a 64-bit CPU. In the unlikely event of using this code on
a 32-bit CPU the number of available bytes would be calculated 4 larger
than is actually available.
Instead of changing to sizeof(*data) it has been replaced by
struct_offset() because it has better chance of detecting these sorts of
typos. Also the offset of the data[] array is actually what we want to know
here anyway.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2b62e66626f0 ("ASoC: cs-amp-lib: Add function to write calibration to UEFI")
Link: https://patch.msgid.link/20260521122511.987322-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Use flush_work() instead of cancel_work_sync() to terminate pending IRQ
work in cs35l56_sdw_remove(). And flush_work() again after masking the
interrupts to flush any queueing that was racing with the masking. This is
the same sequence as cs35l56_sdw_system_suspend().
cs35l56_sdw_interrupt() takes the pm_runtime to prevent the bus powering-
down before the interrupt status can be read and handled. The work releases
this pm_runtime. So cancelling it, instead of flushing, could leave an
unbalanced pm_runtime.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Link: https://patch.msgid.link/20260521123057.988732-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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seq_ump_process_event() borrows client->out_rfile.output without
synchronizing with the first-open and last-close transition in
seq_ump_client_open() and seq_ump_client_close().
The last output unuse can therefore drop opened[STR_OUT] to zero and
release the rawmidi file while an in-flight event_input callback is still
inside snd_rawmidi_kernel_write(). That leaves the rawmidi substream
runtime exposed to teardown before the write path has taken its own
buffer reference.
Add a per-client rwlock for the event_input-visible output file. Publish
a newly opened output file under the write side, and hold the read side
from the output lookup through snd_rawmidi_kernel_write(). The last
output close copies and clears the visible output file under the write
side, then drops the lock and releases the saved rawmidi file. Use
IRQ-safe rwlock guards because event_input can also be reached from
atomic sequencer delivery.
The buggy scenario involves two paths, with each column showing the
order within that path:
path A label: event_input path path B label: last unuse path
1. seq_ump_process_event() reads 1. seq_ump_client_close()
client->out_rfile.output. drops opened[STR_OUT] to zero.
2. snd_rawmidi_kernel_write1() 2. snd_rawmidi_kernel_release()
has not yet pinned runtime. closes the output file.
3. The writer continues using 3. close_substream() frees
the borrowed substream. substream->runtime.
This keeps the output substream and runtime alive for the full
event_input write while keeping rawmidi release outside the rwlock.
KASAN reproduced this as a slab-use-after-free in
snd_rawmidi_kernel_write1(), with allocation through
seq_ump_use()/snd_seq_port_connect() and free through
seq_ump_unuse()/snd_seq_port_disconnect().
Suggested-by: Takashi Iwai <tiwai@suse.de>
Validation reproduced this kernel report:
KASAN slab-use-after-free in snd_rawmidi_kernel_write1+0x9d/0x400
RIP: 0033:0x7f5528af837f
Read of size 8
Call trace:
dump_stack_lvl+0x73/0xb0 (?:?)
print_report+0xd1/0x650 (?:?)
srso_alias_return_thunk+0x5/0xfbef5 (?:?)
__virt_addr_valid+0x1a7/0x340 (?:?)
kasan_complete_mode_report_info+0x64/0x200 (?:?)
kasan_report+0xf7/0x130 (?:?)
snd_rawmidi_kernel_write1+0x9d/0x400 (?:?)
__asan_load8+0x82/0xb0 (?:?)
update_stack_state+0x1ef/0x2d0 (?:?)
snd_rawmidi_kernel_write+0x1a/0x20 (?:?)
seq_ump_process_event+0xd4/0x120 (sound/core/seq/seq_ump_client.c:82)
__snd_seq_deliver_single_event+0x8a/0xe0 (?:?)
snd_seq_deliver_from_ump+0x2b2/0xd60 (?:?)
lock_acquire+0x14e/0x2e0 (?:?)
find_held_lock+0x31/0x90 (?:?)
snd_seq_port_use_ptr+0xa6/0xe0 (?:?)
__kasan_check_write+0x18/0x20 (?:?)
do_raw_read_unlock+0x32/0xa0 (?:?)
_raw_read_unlock+0x26/0x50 (?:?)
snd_seq_deliver_single_event+0x45c/0x4b0 (?:?)
snd_seq_deliver_event+0x10d/0x1b0 (?:?)
snd_seq_client_enqueue_event+0x192/0x240 (?:?)
snd_seq_write+0x2cd/0x450 (?:?)
apparmor_file_permission+0x20/0x30 (?:?)
security_file_permission+0x51/0x60 (?:?)
vfs_write+0x1ce/0x850 (?:?)
__fget_files+0x12b/0x220 (?:?)
lock_release+0xc8/0x2a0 (?:?)
__rcu_read_unlock+0x74/0x2d0 (?:?)
__fget_files+0x135/0x220 (?:?)
ksys_write+0x15a/0x180 (?:?)
rcu_is_watching+0x24/0x60 (?:?)
__x64_sys_write+0x46/0x60 (?:?)
x64_sys_call+0x7d/0x20d0 (?:?)
do_syscall_64+0xc1/0x360 (arch/x86/entry/syscall_64.c:87)
entry_SYSCALL_64_after_hwframe+0x77/0x7f (?:?)
Fixes: 81fd444aa371 ("ALSA: seq: Bind UMP device")
Signed-off-by: Zhang Cen <rollkingzzc@gmail.com>
Link: https://patch.msgid.link/20260520103249.3048345-1-rollkingzzc@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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scarlett2_hwdep_write() rejects writes when offset + count is greater than
or equal to the selected flash segment size. That incorrectly treats a
write ending exactly at the end of the segment as out of space, although
the last byte written is still within the segment.
Split invalid argument checks from the segment-space check, keep
zero-length writes as no-ops, and compare count against the remaining
segment size. This permits exact-end writes and avoids relying on
offset + count before deciding whether the request is in bounds.
Fixes: 1abfbd3c9527 ("ALSA: scarlett2: Add support for uploading new firmware")
Cc: stable@vger.kernel.org
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260519-alsa-scarlett2-flash-write-boundary-v1-1-b550480e92da@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Like the HP ProBook 440 G6, the HP ProBook 430 G6 needs
the ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF quirk for its
mute and microphone mute LEDs.
Tested on a HP ProBook 430 G6.
Signed-off-by: Marius Hoch <mail@mariushoch.de>
Link: https://patch.msgid.link/20260519140248.4211-2-mail@mariushoch.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The pending irq work might be still floating while the assigned stream
has been already closed, which may lead to UAF, especially when
another async work for fasync is involved.
For addressing this, extend the hda_controller_ops for allowing the
extra cleanup procedure that is specific to the controller driver, and
make sure to cancel and sync the pending irq work at each PCM close
before releasing the resources.
Reported-by: Jake Lamberson <lamberson.jake@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260519121157.28477-2-tiwai@suse.de
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Currently, the delayed IRQ handling for PCM streams is managed in a
single work embedded in hda_intel, but this is basically a per-stream
thing. Due to the single work, we can't cancel the work properly at
closing each stream, for example.
For making the IRQ pending work to be stream-based, this patch changes
the following:
- An extended version of azx_dev (i.e. the hd-audio stream object) is
defined for snd-hda-intel
- The irq_pending flag and irq_pending_work are moved to
hda_intel_stream, so that they can be hda-intel stream specific
- The stream creation and assignment are refactored so that
snd-hda-intel can handle individually;
the snd-hda-intel specific workaround for stream tags is also moved
to snd-hda-intel itself instead of the common code
- The irq pending work is canceled properly at free / shutdown
While we're at it, changed the bit field flag to bool, as the bit
field doesn't help much in our case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260519121157.28477-1-tiwai@suse.de
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The default case in snd_soc_ret() use va_start without va_end to
cleanup "args" object which can cause undefined behavior. So, add
missing va_end to cleanup "args" object.
This is reported by Coverity Scan as "Missing varargs init or cleanup".
Fixes: 943116ba2a6a ("ASoC: add common snd_soc_ret() and use it")
Signed-off-by: Robertus Diawan Chris <robertusdchris@gmail.com>
Link: https://patch.msgid.link/20260519054024.274741-1-robertusdchris@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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find_format_descriptor() walks the class-specific interface extras by
advancing with bLength. It rejects descriptors that extend past the
remaining buffer, but it does not reject descriptor lengths smaller than
a USB descriptor header.
Reject too-short descriptors before using bLength to advance the local
scan. This keeps the UA-101 parser robust against malformed descriptor
data and matches the usual USB descriptor walking rules.
Fixes: 63978ab3e3e9 ("sound: add Edirol UA-101 support")
Cc: stable@vger.kernel.org
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260519-alsa-ua101-desc-len-v1-1-4307d1a5e054@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The ALC245 sound card on this machine requires the quirk
`ALC245_FIXUP_HP_ENVY_X360_15_FH0XXX` to fix the mic and mute LED.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=221509
Cc: <stable@vger.kernel.org>
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260519015535.891156-1-zhangheng@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_seq_create_port() walks client->ports_list_head looking for
the ordered insertion point and on loop fall-through passes
&p->list to list_add_tail():
list_for_each_entry(p, &client->ports_list_head, list) {
if (p->addr.port == port) {
kfree(new_port);
return -EBUSY;
}
if (p->addr.port > num)
break;
...
}
list_add_tail(&new_port->list, &p->list);
When the loop walks all entries without break (e.g., the new
port sorts last), p is past-the-end. &p->list aliases
&client->ports_list_head (the list head) via container_of offset
cancellation, so the insert lands at the list tail. That is the
intended behaviour, but the access is undefined per C11 even
though it works in practice.
Track an explicit insert_before pointer initialised to the list
head and overwritten to &p->list only when the loop breaks
early. The observable behaviour is unchanged.
Fixes: 9244b2c3079f ("[ALSA] alsa core: convert to list_for_each_entry*")
Signed-off-by: Maoyi Xie <maoyixie.tju@gmail.com>
Link: https://patch.msgid.link/20260518194023.1667857-3-maoyixie.tju@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_timer_dev_register() walks snd_timer_list looking for the
ordered insertion point and on loop fall-through passes
&timer1->device_list to list_add_tail():
list_for_each_entry(timer1, &snd_timer_list, device_list) {
...
break; /* on found-position */
...
}
list_add_tail(&timer->device_list, &timer1->device_list);
When the loop walks all entries without break, timer1 is
past-the-end. &timer1->device_list aliases &snd_timer_list (the
list head) via container_of offset cancellation, so the insert
lands at the list tail. That is the intended behaviour, but the
access is undefined per C11 even though it works in practice.
Track an explicit insert_before pointer initialised to the list
head and overwritten to &timer1->device_list only when the loop
breaks early. The observable behaviour is unchanged.
Fixes: 9244b2c3079f ("[ALSA] alsa core: convert to list_for_each_entry*")
Signed-off-by: Maoyi Xie <maoyixie.tju@gmail.com>
Link: https://patch.msgid.link/20260518194023.1667857-2-maoyixie.tju@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When the SAI peripheral is left in a running state by the bootloader,
the driver can experience an interrupt storm during probe that prevents
successful initialization. This occurs because the current code registers
the IRQ handler before resetting the hardware to a known state.
The issue manifests as:
- Continuous interrupts firing immediately after devm_request_irq()
- Driver probe failure or system hang
- Error messages about unhandled interrupts
This is particularly problematic on systems where U-Boot or other
bootloaders enable SAI for boot-time audio feedback or diagnostics
and don't properly disable it before handing control to Linux.
Fix this by reordering the probe sequence:
1. Add fsl_sai_reset_hw() to clear TCSR/RCSR control registers,
which disables the transmitter/receiver and all interrupt sources
2. Move devm_request_irq() to after hardware initialization
This ensures the SAI is in a clean reset state before the interrupt
handler can be invoked, preventing the storm while maintaining proper
error handling and cleanup paths.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/20260512065252.75859-1-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The HP Z66 G6 14 inch laptop uses the ALC236 codec with subsystem ID
0x103c:8df7. Without a quirk entry, the PCI SSID falls back to the
generic 0x103c:0000 fixup, which does not configure the mute/micmute
LED GPIOs correctly.
Add the SND_PCI_QUIRK entry for this model using
ALC236_FIXUP_HP_GPIO_LED, matching the surrounding HP EliteBook G12
entries (0x8dec-0x8dfe) which share the same ALC236 codec and GPIO LED
layout.
Signed-off-by: Minxi Hou <houminxi@gmail.com>
Link: https://patch.msgid.link/20260518031542.2899188-1-houminxi@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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At transition to the iov_iter for PCM data transfer, we blindly
applied the iov_iter setup also for silencing (i.e. data = NULL), and
it leads to a calculation of bogus iov_iter. Fortunately this didn't
cause troubles on most of architectures but it goes wrong on RISC-V
now, causing a NULL dereference.
Handle the NULL data case to treat the silencing in interleaved_copy()
for addressing the bug above. noninterleaved_copy() has already the
NULL data handling, so it doesn't need changes.
Reported-by: Jiakai Xu <xujiakai24@mails.ucas.ac.cn>
Closes: https://lore.kernel.org/20260515051516.3103036-1-xujiakai24@mails.ucas.ac.cn
Fixes: cf393babb37a ("ALSA: pcm: Add copy ops with iov_iter")
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20260517165121.31399-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The HP Pavilion Plus 14-eh0xxx with subsystem ID 103c:8a36 needs the
ALC245 COEF bit mute LED quirk for the mute LED to follow the audio mute
state.
Add the missing quirk entry.
Signed-off-by: Aryan Kushwaha <aryankushwaha3101@gmail.com>
Link: https://patch.msgid.link/20260516144436.35022-1-aryankushwaha3101@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Mirror of Mark Brown's ASoC: hdac_hdmi rate-limit patch (commit
[lkml.kernel.org/lkml/2025/6/13/1380]) for the generic snd_parse_eld()
helper used by ASoC hdmi-codec.
When a HDMI sink is disconnected (e.g. a board with two HDMI outputs and
only one cable), userspace audio servers like PipeWire keep probing the
disconnected card and trigger:
HDMI: Unknown ELD version 0
at every probe — easily 30+ messages per burst on rk3588. The same
applies to malformed ELD (MNL out of range). Both conditions are
expected when no sink is attached; rate-limit the dev_info() so the
kernel ring buffer does not fill up.
Signed-off-by: Francesco Saverio Pavone <pavone.lawyer@gmail.com>
Link: https://patch.msgid.link/20260516141244.21801-1-pavone.lawyer@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Heiko reported that BIOS on some recent machines doesn't set up PCI
SSID properly but leave with zero (e.g. on HP Dragonfly Folio 13.5
inch G3 with SSID 103c:8a05/8a06), which confuses the quirk table
matching and results in the non-functional state.
Fix it by skipping the PCI SSID matching when either vendor or device
ID is zero and falling back to the codec SSID that is supposed to be
more stable for those cases.
Reported-by: Heiko Schmid <heiko@future-machines.org>
Tested-by: Heiko Schmid <heiko@future-machines.org>
Closes: https://lore.kernel.org/20260514133110.12302-1-heiko@future-machines.org
Link: https://patch.msgid.link/20260515105700.276420-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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HP 250 15.6 inch G10 Notebook PC uses the same ALC236 codec
as the HP 255 15.6 inch G10 (103c:8b2f) and requires the same
fixup to enable the internal speaker EAPD and microphone routing.
Signed-off-by: Sergio Boglione <sboglione@gmail.com>
Link: https://patch.msgid.link/20260516131651.143109-1-sboglione@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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These devices were incorrectly using the ALC287_FIXUP_TAS2781_I2C quirk
leading to errors:
[ 18.765990] Serial bus multi instantiate pseudo device driver TXNW2781:00: error -ENXIO: IRQ index 0 not found
[ 18.768153] Serial bus multi instantiate pseudo device driver TXNW2781:00: error -ENXIO: IRQ index 0 not found
[ 18.768476] Serial bus multi instantiate pseudo device driver TXNW2781:00: error -ENXIO: IRQ index 0 not found
[ 18.768899] Serial bus multi instantiate pseudo device driver TXNW2781:00: Instantiated 3 I2C devices.
Use the ALC287_FIXUP_TXNW2781_I2C quirk instead to fix this and restore
speaker audio on affected devices.
Fixes: 1e9c708dc3ae ("ALSA: hda/tas2781: Add new quirk for Lenovo, ASUS, Dell projects")
Link: https://lore.kernel.org/59fd4aa4-76b9-4984-8db9-a60e55ec6e80@losource.net/
Closes: https://lore.kernel.org/CACB9z7kjs8rhLstEc8fV29BCTb5dd881JwGozoKdO5cwCb=YwQ@mail.gmail.com
Signed-off-by: Eric Naim <dnaim@cachyos.org>
Link: https://patch.msgid.link/20260516111532.111463-1-dnaim@cachyos.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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find_control() to retrieve a cached info accesses the array with the
given index blindly, which may lead to an OOB array access.
Add a sanity check for avoiding it.
Link: https://sashiko.dev/#/patchset/20260511230121.28606-1-rosenp%40gmail.com
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20260515085606.242284-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The ASUS Zenbook S16 (UM5606GA) with AMD Ryzen AI 9 465 (Strix Point,
ACP 7.0) has a BIOS that incorrectly sets the ACPI property
'acp-audio-config-flag' to 0x10 (FLAG_AMD_LEGACY_ONLY_DMIC) for the ACP
device. This prevents snd_pci_ps from probing the SoundWire bus, resulting
in no internal audio (dummy output only).
The hardware uses a Cirrus Logic CS42L43 (headphone/jack) and four CS35L56
smart amplifiers (speakers), all on SoundWire link 1. The corresponding
machine table entry (acp70_cs42l43_l1u0_cs35l56x4_l1u0123) already exists
in amd-acp70-acpi-match.c and correctly describes this topology.
Add a DMI quirk to override the flag to 0, consistent with the existing
entry for the HN7306EA.
Signed-off-by: Jasper Smet <josbeir@gmail.com>
Link: https://patch.msgid.link/20260513052137.56703-1-josbeir@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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devm_kasprintf() can fail and return NULL. The legacy AMD SoundWire
machine driver logs cpus->dai_name before checking the allocation result.
Move the debug print after the NULL check, matching the ordering used by
the SOF AMD SoundWire path after commit 5726b68473f7 ("ASoC: amd/sdw_utils:
avoid NULL deref when devm_kasprintf() fails").
Fixes: 2981d9b0789c ("ASoC: amd: acp: add soundwire machine driver for legacy stack")
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260511-asoc-amd-acp-sdw-legacy-dai-name-null-v1-1-dc6151b6da8a@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Some Old DSP firmware versions use 32-bit address arithmetic and size for
validating the PCM buffer address range. If a buffer is allocated near
the top of the 32-bit address space, arithmetic calculations involving
the end address can overflow and fail checks.
Work around this by increasing the preallocated PCM buffer size by one
page. The DSP is still passed the usable buffer size, excluding the extra
page, which prevents the firmware from seeing an end address that crosses
the 32-bit boundary.
This was not hit before because PCM buffer allocation and DSP-side
mapping happened at different points, and the size mapped on the DSP was
usually nperiods * period_size. Therefore the mapped size was unlikely to
match the full preallocated buffer size exactly, although the issue was
still possible. With early buffer mapping on the DSP, the full
preallocated buffer is mapped during PCM creation, making the failure
reproducible at boot.
Fixes: 8ea6e25c8536 ("ASoC: qcom: q6apm: Add support for early buffer mapping on DSP")
Cc: Stable@vger.kernel.org
Reported-by: Jens Glathe <jens.glathe@oldschoolsolutions.biz>
Closes: https://lore.kernel.org/all/7f10abbd-fb78-4c3a-ab90-7ca78239891a@oldschoolsolutions.biz/
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Tested-by: Jens Glathe <jens.glathe@oldschoolsolutions.biz>
Link: https://patch.msgid.link/20260514090607.2435484-1-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Commit 778031e1658d ("ALSA: hda/ca0132: Set HP/Speaker
auto-detect default from headphone pin verb") enables HP/Speaker
auto-detect by default when the headphone pin supports presence detect.
With auto-detect enabled, ca0132_select_out() and ca0132_alt_select_out()
choose the output from jack presence instead of the manual HP/Speaker
selection. This means selecting speaker output while headphones are
plugged in updates the control state, but audio still routes to the
headphones.
Treat an explicit manual output selection as a request to leave
auto-detect mode. Clear the HP/Speaker auto-detect switch before applying
the manual selection, and notify userspace so the auto-detect control
state is updated in mixers. Do this for both the normal HP/Speaker
Playback Switch and the alternate Output Select control used by desktop
cards.
This keeps auto-detect enabled by default for devices with jack presence
detection, while preserving the expected behavior that a manual output
choice takes effect immediately.
Fixes: 778031e1658d ("ALSA: hda/ca0132: Set HP/Speaker auto-detect default from headphone pin verb")
Signed-off-by: Matt DeVillier <matt.devillier@gmail.com>
Link: https://lore.kernel.org/CAFTm+6AfeXKf=b2frG4xC5yC4jjM9TkD6c8+dOWWFw6BDjDESw@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add a SND_PCI_QUIRK entry for the HP Pavilion Laptop 16-ag0xxx
(subsystem 0x103c:0x8cbc, Realtek ALC245). The
ALC245_FIXUP_HP_X360_MUTE_LEDS fixup is already used by the
neighbouring HP Pavilion Aero Laptop 13-bg0xxx (0x103c:0x8cbd);
it chains the master-mute COEF handler with the GPIO mic-mute
LED handler, which is what this machine needs.
Tested on the affected hardware: both the mute and mic-mute key
LEDs respond correctly to the keyboard hotkeys after this change.
Cc: <stable@vger.kernel.org>
Signed-off-by: Adrien Burnett <an.arctic.pigeon@gmail.com>
Link: https://patch.msgid.link/20260514165905.21175-1-an.arctic.pigeon@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Volume control for the speakers on the Lenovo Yoga Pro 7 15ASH11 laptop
doesn't work.
The DAC routing is the same as on the ThinkPad X1 Gen7 function, so reuse
the alc285_fixup_thinkpad_x1_gen7 to get it working.
Signed-off-by: Jackie Dong <xy-jackie@139.com>
Link: https://patch.msgid.link/20260514153940.7320-1-xy-jackie@139.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_hda_ctl_add() dereferences kctl->id.subdevice without checking
whether kctl is NULL. Multiple callers in sound/hda/codecs/ca0132.c
pass the return value of snd_ctl_new1() directly to snd_hda_ctl_add()
without a NULL check:
return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
snd_ctl_new1() returns NULL when the underlying snd_ctl_new() fails
on memory allocation (kzalloc_flex),which can occur under memory
pressure or via fault injection.
Add a NULL check at the entry of snd_hda_ctl_add(), matching the
pattern already used by snd_ctl_add_replace() at the same call
path (sound/core/control.c:515). Return -EINVAL to let callers
handle the error gracefully.
Fixes: 44f0c9782cc6 ("ALSA: hda/ca0132: Add tuning controls")
Signed-off-by: Quan Sun <2022090917019@std.uestc.edu.cn>
Link: https://patch.msgid.link/20260514132245.3062884-1-2022090917019@std.uestc.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Samsung Galaxy Book5 360 (NP750QHA, PCI subsystem ID 0x144d:0xc902)
has severe audio distortion on the 3.5mm headphone jack. Applying
ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET corrects the output path
configuration, consistent with fixes already applied to other Samsung
Galaxy Book models using the same ALC256 codec.
Cc: stable@vger.kernel.org
Link: https://github.com/thesofproject/linux/issues/5648
Signed-off-by: Markus Kramer <linux@markus-kramer.de>
Link: https://patch.msgid.link/20260513222818.14351-1-linux@markus-kramer.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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First of all, it has to be 'default n' (small letter n), otherwise
it looks for CONFIG_N which is absent and in case of appearance
will enable something unrelated. Second and most important is that
'n' *is* the default 'default' already. Hence just drop malformed
line.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260513162758.365972-1-andriy.shevchenko@linux.intel.com
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In addition to the mic jack fix, also need to avoid boosting the
internal mic too much, otherwise >50% input volume clips a lot.
Also add a second SSID. We have one for the classic chassis/speaker and
one for the new Pro chassis/speaker.
To: Jaroslav Kysela <perex@perex.cz>
To: Takashi Iwai <tiwai@suse.com>
To: linux-sound@vger.kernel.org
Cc: Dustin L. Howett <dustin@howett.net>
Cc: linux@frame.work
Signed-off-by: Daniel Schaefer <dhs@frame.work>
Link: https://patch.msgid.link/20260513155513.11683-1-dhs@frame.work
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The internal mic boost on the Positivo DN50E 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>
Link: https://patch.msgid.link/20260511181558.670563-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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uaudio_transfer_buffer_setup() calls dma_get_sgtable() and then passes
the sg_table to uaudio_iommu_map_xfer_buf() without checking whether sg
table construction succeeded. If dma_get_sgtable() fails, the sg_table
contents are not valid.
uaudio_iommu_map_pa() also ignores iommu_map() failures for the event and
transfer rings and still returns the allocated IOVA to the QMI response.
That can expose an unmapped IOVA to the audio DSP. For transfer rings,
the failed mapping also leaves the IOVA allocator state marked in use.
Check both operations. Free the coherent transfer buffer when sg table
construction fails, free the sg table when transfer-buffer IOMMU mapping
fails, and release the transfer-ring IOVA if iommu_map() fails. Also
return the existing event-ring IOVA when the event ring is already mapped,
matching the pre-split helper behavior.
Fixes: 326bbc348298 ("ALSA: usb-audio: qcom: Introduce QC USB SND offloading support")
Fixes: 44499ecb4f28 ("ALSA: usb: qcom: Fix false-positive address space check")
Cc: stable@vger.kernel.org
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260511-alsa-usb-qcom-offload-map-errors-v1-1-6502695e58bc@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Lenovo Legion 7 16ITHG6 uses codec SSID 17aa:3855, but its PCI
SSID is 17aa:3811. The latter is now also used by the Legion S7 15IMH05
quirk, which is matched before codec SSID fallback and incorrectly
routes Legion 7 16ITHG6 machines to ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS.
That fixup does not bind the CLSA0101 CS35L41 companion amplifiers,
making the built-in speakers silent even though playback appears to be
active.
Add a codec SSID quirk for 17aa:3855 before the conflicting PCI SSID
quirk so that the Legion 7 16ITHG6 uses ALC287_FIXUP_LEGION_16ITHG6.
This restores CS35L41 firmware loading and binds both speaker
amplifiers.
Fixes: 67f4c61a73e9 ("ALSA: hda/realtek: Add quirk for Legion S7 15IMH")
Cc: stable@vger.kernel.org
Tested-by: Nicholas Bonello <hadobedo@gmail.com>
Assisted-by: Codex:GPT-5
Signed-off-by: Nicholas Bonello <hadobedo@gmail.com>
Link: https://patch.msgid.link/20260508225507.47667-1-hadobedo@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Setting up the interface when suspended/resumeing fail on this card.
Adding a reset and delay quirk will eliminate this problem.
usb 1-1: new full-speed USB device number 2 using xhci-hcd
usb 1-1: New USB device found, idVendor=3302, idProduct=17c2
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 1-1: Product: USB-C Audio
usb 1-1: Manufacturer: TTGK Technology
usb 1-1: SerialNumber: 170120210706
Signed-off-by: Lianqin Hu <hulianqin@vivo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/TYUPR06MB621720E4E8F99A42E162FD51D23D2@TYUPR06MB6217.apcprd06.prod.outlook.com
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When initialise new control with scarlett2_add_new_ctl() function for
Autogain Status, scarlett2_add_new_ctl() might throw an error. So, add
error check after initialise new control for Autogain Status.
This is reported by Coverity Scan with CID 1598781 as UNUSED_VALUE.
Fixes: 0a995e38dc44 ("ALSA: scarlett2: Add support for software-controllable input gain")
Signed-off-by: Robertus Diawan Chris <robertusdchris@gmail.com>
Link: https://patch.msgid.link/20260508033914.111596-1-robertusdchris@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In fs210x_effect_scene_info(), a string was copied like this:
strscpy(DST, SRC, strlen(SRC) + 1);
A buffer overflow would happen if strlen(SRC) >= sizeof(DST).
Actually, strscpy() must be used this way:
strscpy(DST, SRC, sizeof(DST));
strscpy(DST, SRC); // defaults to sizeof(DST)
Fixes: 756117701779 ("ASoC: codecs: Add FourSemi FS2104/5S audio amplifier driver")
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Link: https://patch.msgid.link/20260513190852.196723-2-grandmaster@al2klimov.de
Signed-off-by: Mark Brown <broonie@kernel.org>
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The SoundWire slave update_status() callback can be invoked when the
status has not changed. To prevent large amounts of log noise with debug
enabled, log them only when the status changes. This also helps with
understanding them, because they now log an actual change in state.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260514151854.695145-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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clk_get_parent() returns NULL when the clock has no parent (or when the
input clk is NULL); it never returns an ERR_PTR. The current IS_ERR(mux)
check therefore never triggers - a NULL return falls through silently
to clk_set_parent(NULL, parent_clk), which simply fails with -EINVAL.
Use a NULL check so the dedicated error path runs and the prior
clk_get() reference is released via clk_put().
Signed-off-by: Ingyu Jang <ingyujang25@korea.ac.kr>
Acked-by: Sen Wang <sen@ti.com>
Link: https://patch.msgid.link/20260514185215.3753998-1-ingyujang25@korea.ac.kr
Signed-off-by: Mark Brown <broonie@kernel.org>
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First of all, it has to be 'default n' (small letter n), otherwise
it looks for CONFIG_N which is absent and in case of appearance
will enable something unrelated. Second and most important is that
'n' *is* the default 'default' already. Hence just drop malformed
line.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://patch.msgid.link/20260513162612.365729-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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devm_kasprintf() can fail while building the temporary speaker
component string. If that happens, spk_components is set to NULL, but
the current code can still pass it to strlen() on a later loop iteration
or after the loop when appending the speaker component list to
card->components.
Use NULL to represent the initial "no speaker components" state, and
return -ENOMEM immediately if building spk_components fails.
Fixes: 0f60ecffbfe3 ("ASoC: sdw_utils: generate combined spk components string")
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260512-asoc-sdw-utils-spk-components-alloc-v1-1-c9bbd6d2e123@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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