snd_timer_close_locked() marks an instance with SNDRV_TIMER_IFLG_DEAD
and returns early when the flag is already set, but the flag is never
cleared again. A completed close ends in remove_slave_links(), which
leaves timeri->timer NULL, so a second close is already harmless through
the timer == NULL path; the early return can only be reached by an
instance that was opened again in between. For such an instance the
close unlinks nothing, so snd_timer_instance_free() frees an object that
is still on timer->open_list_head, still on snd_timer_master_list if it
was opened with a slave key, still owns any adopted slaves, and still
holds its timer and module references.
snd_seq_timer_open() reopens an instance exactly like that: it retries
its fallback open on the same object after a failure that has already
run snd_timer_close_locked() internally. An unprivileged user with
access to /dev/snd/timer and /dev/snd/seq can force that failure, since
snd_timer_check_master() returns -EBUSY when a pending slave matches the
new master's (slave_class, slave_id) key and the target timer has
reached max_instances, and SNDRV_TIMER_IOCTL_SELECT with dev_class =
SNDRV_TIMER_CLASS_SLAVE keeps the caller-supplied dev_sclass, so a
sequencer queue's key can be forged. The freed instance is afterwards
dereferenced by any further snd_timer_open() on that timer, by
snd_timer_check_slave(), and by /proc/asound/timers, which faults on the
stale ti->owner pointer.
The flag only has to be visible while the close is in progress, which is
all its other users need. Clear it in remove_slave_links(), under the
same timer->lock that sets it, once the instance is off every list.
Fixes: da3039e91d ("ALSA: timer: Forcibly close timer instances at closing")
Cc: stable@vger.kernel.org
Assisted-by: Claude:claude-opus-5
Signed-off-by: Norbert Szetei <norbert@doyensec.com>
Link: https://patch.msgid.link/CA41AA48-75BF-45E9-A36D-3A5D2F124F60@doyensec.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Infinix INBOOK X3 Slim (ALC269VB, subsystem 0x2782:0250) has its
internal speakers wired to pin 0x1b, but the BIOS pin configuration
table declares 0x1b as "no physical connection" (0x411111f0). It instead
declares pin 0x14 as the internal speaker, although nothing is connected
to 0x14.
As a result the internal speakers are silent under Linux while the
headphone jack works correctly. The codec output path to 0x14 is fully
open (DAC assigned and streaming, mixer and pin unmuted, EAPD asserted),
so the failure is silent with no error reported. The speakers work under
Windows, where the vendor driver supplies its own pin table.
Add a fixup that disables the unconnected pin 0x14 and declares pin
0x1b as the internal speaker.
Reusing the existing ALC269VC_FIXUP_INFINIX_Y4_MAX was tried first, since
it also remaps 0x1b to an internal speaker. It is not sufficient here: it
leaves 0x14 declared, so autoconfig finds two line_outs
line_outs=2 (0x14/0x1b/0x0/0x0/0x0) type:speaker
and binds the primary "Speaker" control to the unconnected pin 0x14 while
demoting the working speaker at 0x1b to "Bass Speaker". Audio is audible
that way, but the volume and mute controls a desktop actually uses end up
attached to a pin that is not wired to anything. Disabling 0x14 is what
produces a single correct Speaker output.
Verified on the affected machine: with the corrected pin
configuration the driver's autoconfig reports
line_outs=1 (0x1b/0x0/0x0/0x0/0x0) type:speaker
both channels play, headphone auto-mute switches correctly in both
directions, and audio survives codec runtime suspend (D3) and resume.
Signed-off-by: Gunal Seenivasagan <gunal2002@gmail.com>
Link: https://patch.msgid.link/20260726085715.229802-1-gunal2002@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_usbmidi_akai_output() computes its fill-loop bound
buf_end = ep->max_transfer - MAX_AKAI_SYSEX_LEN - 1;
as a signed int, so a small device-advertised bulk-OUT max_transfer
makes buf_end negative. The loop guard then compares the u32
urb->transfer_buffer_length against that negative int: the usual
arithmetic conversion turns buf_end into a large unsigned value, so the
guard stays true and each iteration keeps appending SysEx framing and
payload bytes past the end of the URB transfer buffer, which is only
max_transfer bytes long.
A USB device that advertises a tiny bulk-OUT endpoint can therefore
trigger an attacker-length- and content-controlled heap out-of-bounds
write when a process writes to the created /dev/snd/midiC*D* node.
Return early when there is no room for even one SysEx, so the loop is
never entered with a bound that would wrap. The loop is the last
statement of the function, so bailing out is equivalent to it not
running.
Discovered by XBOW, triaged by Baul Lee <baul.lee@xbow.com>
Fixes: 4434ade8c9 ("ALSA: usb-audio: add support for Akai MPD16")
Suggested-by: Takashi Iwai <tiwai@suse.de>
Reported-by: Federico Kirschbaum <federico.kirschbaum@xbow.com>
Reported-by: Baul Lee <baul.lee@xbow.com>
Cc: stable@vger.kernel.org
Signed-off-by: Baul Lee <baul.lee@xbow.com>
Link: https://patch.msgid.link/20260726074500.50145-1-baul.lee@xbow.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_rme_digiface_read_status() reads a four-word status block from the
device into an uninitialised on-stack __le32 buf[4] and, whenever the
vendor control-IN transfer does not return a negative error, copies all
four words into the caller's status[].
snd_usb_ctl_msg() copies the full requested size back into the caller's
buffer regardless of how many bytes the data stage actually delivered:
buf = kmemdup(data, size, GFP_KERNEL);
err = usb_control_msg(dev, pipe, request, requesttype,
value, index, buf, size, timeout);
memcpy(data, buf, size);
usb_control_msg() returns the transferred length on a short control-IN,
which is a non-negative value, and writes only that many bytes. The
remainder of the copy back is the kmemdup()ed image of the caller's
buffer, so a device answering with a short data stage leaves the
trailing words of buf[] holding leftover kernel stack. The only guard
in the caller is err < 0, so those words are stored into status[].
They then reach user space: snd_rme_digiface_get_status_val() selects a
16-bit halfword of status[] per the control's reg/mask, and the eight
Digiface status controls together expose the whole 16-byte frame to an
unprivileged reader of /dev/snd/controlC*.
Zero-initialise the buffer so a short read yields zeros instead of stack
residue. This mirrors snd_rme_get_status1(), which already clears its
output word before the same kind of vendor read.
Discovered by XBOW, triaged by Baul Lee <baul.lee@xbow.com>
Fixes: 611a96f6ac ("ALSA: usb-audio: Add mixer quirk for RME Digiface USB")
Reported-by: Federico Kirschbaum <federico.kirschbaum@xbow.com>
Reported-by: Baul Lee <baul.lee@xbow.com>
Cc: stable@vger.kernel.org
Signed-off-by: Baul Lee <baul.lee@xbow.com>
Link: https://patch.msgid.link/20260726065020.46070-1-baul.lee@xbow.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_ump_attach_legacy_rawmidi() allocates the legacy conversion array
ump->out_cvts and, on the snd_rawmidi_new() error path, frees it with
kfree() but leaves ump->out_cvts pointing at the freed memory. When the
endpoint is later torn down, snd_ump_endpoint_free() frees ump->out_cvts
a second time, resulting in a double free.
The host snd-usb-audio driver attaches the legacy rawmidi for any USB
MIDI 2.0 (UMP) device, so a device that makes snd_rawmidi_new() fail
reaches this path on enumeration.
Clear ump->out_cvts after freeing it on the error path so it is not
freed again during teardown.
Discovered by XBOW, triaged by Baul Lee <baul.lee@xbow.com>
Fixes: 33cd763078 ("ALSA: ump: Export MIDI1 / UMP conversion helpers")
Reported-by: Federico Kirschbaum <federico.kirschbaum@xbow.com>
Reported-by: Baul Lee <baul.lee@xbow.com>
Cc: stable@vger.kernel.org
Signed-off-by: Baul Lee <baul.lee@xbow.com>
Link: https://patch.msgid.link/20260726051633.41206-1-baul.lee@xbow.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
create_midi2_ump() registers a card-owned snd_ump_endpoint and stores a
back-pointer to its per-interface snd_usb_midi2_ump object in
ump->private_data, but it never installs an ump->private_free hook and
never clears that pointer.
If a later step of snd_usb_midi_v2_create() fails, its error path calls
free_all_midi2_umps(), which kfree()s the snd_usb_midi2_ump object while
the already-registered endpoint keeps pointing at it. The created
/dev/snd/umpC*D* node stays exposed, so the first operation of any UMP
open, ump_to_endpoint(), dereferences the dangling ump->private_data and
reads rmidi->eps[dir] out of freed memory.
A malicious USB MIDI 2.0 device that makes creation fail after the
endpoint is registered can thus trigger a slab use-after-free read on a
subsequent open of the UMP node.
Clear the endpoint's back-pointer before freeing the object, and let
ump_to_endpoint() tolerate a NULL private_data so the open/close/trigger
callbacks fail cleanly (their callers already handle a NULL endpoint)
instead of dereferencing a stale pointer.
Discovered by XBOW, triaged by Baul Lee <baul.lee@xbow.com>
Fixes: ff49d1df79 ("ALSA: usb-audio: USB MIDI 2.0 UMP support")
Reported-by: Federico Kirschbaum <federico.kirschbaum@xbow.com>
Reported-by: Baul Lee <baul.lee@xbow.com>
Cc: stable@vger.kernel.org
Signed-off-by: Baul Lee <baul.lee@xbow.com>
Link: https://patch.msgid.link/20260726051337.41124-1-baul.lee@xbow.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Acer Nitro 5 AN515-46 (SSID 1025:159e, Realtek ALC287) has a combo
headset jack whose microphone does not work out of the box: the BIOS
leaves pin 0x19 unconfigured, so no headset mic is created.
Apply ALC2XX_FIXUP_HEADSET_MIC, the same fixup already used by the
sibling models AN515-57 (1025:1539) and AN517-55 (1025:1597), which
makes the headset microphone work correctly.
Tested on an Acer Nitro 5 AN515-46 by overriding the model via a patch
firmware with model=alc2xx-fixup-headset-mic.
Signed-off-by: Marcos Paulo Medeiros <maarcospm1996@gmail.com>
Link: https://patch.msgid.link/20260725214131.25872-1-maarcospm1996@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Setting up the interface when suspended/resuming fails on this card.
Adding a reset and delay quirk will eliminate this problem.
Note: This device's VID conflicts with Apple's (0x05ac).
usb 1-1: New USB device found, idVendor=05ac, idProduct=110b
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 1-1: Product: Q2A
usb 1-1: Manufacturer: JKY Technology
usb 1-1: SerialNumber: 330270D2251225
Suggested-by: Rong Zhang <i@rong.moe>
Signed-off-by: Lianqin Hu <hulianqin@vivo.com>
Reviewed-by: Rong Zhang <i@rong.moe>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/TYUPR06MB6217566CFD33F57D3AE46816D2CF2@TYUPR06MB6217.apcprd06.prod.outlook.com
The lx6464es driver advertises both 16-bit and packed 24-bit PCM formats,
but lx_trigger_start() and lx_interrupt_request_new_buffer() calculate the
DMA period size as runtime->period_size * runtime->channels * 3. That is
only correct for the packed 24-bit formats.
For 16-bit streams the driver submits buffers that are 50% larger than the
actual ALSA period and advances the DMA address by the same wrong amount.
For example, with 2 channels, 256 frames and 4 periods, the third buffer
already extends beyond the ALSA buffer and the fourth buffer starts outside
it.
Use snd_pcm_lib_period_bytes() so the byte count matches the runtime
format, channel count and period size.
Fixes: 02bec49045 ("ALSA: lx6464es - driver for the digigram lx6464es interface")
Cc: stable@vger.kernel.org
Signed-off-by: Xu Rao <raoxu@uniontech.com>
Link: https://patch.msgid.link/8BB12E8D92A7CDBA+20260723085710.2567463-1-raoxu@uniontech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A userspace-driven ALSA timer (SND_UTIMER) lets an unprivileged user set
the backing snd_timer's hardware resolution to an arbitrary 64-bit value
via SNDRV_TIMER_IOCTL_CREATE. snd_utimer_create() only rejects zero.
When such a timer is bound to a sequencer queue, initialize_timer()
computes the tick period as
tmr->ticks = 1000000000 / (r * freq);
where r is that user-controlled resolution and freq is the sequencer
update rate in Hz, clamped to MIN_FREQUENCY..MAX_FREQUENCY (10..6250).
A resolution of 2^63 makes the 64-bit product r * freq wrap to zero for
any even freq, including DEFAULT_FREQUENCY (1000), so the division faults
with a divide-by-zero.
The division runs under tmr->lock with interrupts disabled, so the oops
leaves the spinlock held and hangs the CPU. It is reachable by an
unprivileged user with access to /dev/snd/timer and /dev/snd/seq.
Oops: divide error: 0000 [#1] SMP KASAN PTI
CPU: 7 UID: 1000 PID: 456 Comm: alsa_seq_utimer Not tainted 7.2.0-rc4+
RIP: 0010:initialize_timer.constprop.0+0x20a/0x2d0
snd_seq_timer_start+0x15e/0x2b0
snd_seq_control_queue+0x56f/0xba0
snd_seq_write+0x3e0/0x730
Reject an overflowing product with check_mul_overflow() and fall back to
a single tick, which also avoids feeding a wrapped-but-nonzero divisor
(e.g. 2^63 * 1000 mod 2^64 == 0, or other resolutions wrapping to a small
value) into the period computation.
Fixes: 37745918e0 ("ALSA: timer: Introduce virtual userspace-driven timers")
Cc: <stable@vger.kernel.org>
Assisted-by: Claude:claude-opus-4-8
Signed-off-by: Norbert Szetei <norbert@doyensec.com>
Link: https://patch.msgid.link/DF8A3844-AD5E-4B8A-9CFC-BD83C212BA38@doyensec.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Razer Barracuda X 2.4 GHz USB headset dongle (0x1532:0x0552)
reports a minimum volume register value of cval->min = -16800.
In UAC 1/256 dB units, -16800 corresponds to -65.625 dB. However,
stock ALSA misinterprets this raw integer as 1/100 dB units
(-168.00 dB), causing user-space audio servers (PipeWire /
PulseAudio) to map their volume curves against an incorrectly wide
range.
Add an explicit usbmix_dB_map entry overriding Unit 2 to -6562
(-65.62 dB) to accurately report the physical hardware
attenuation bounds.
Signed-off-by: Markus Lindner <lindner.markus@outlook.at>
Link: https://patch.msgid.link/AS8P195MB2142F4EFF83980BD02BA6566E1C12@AS8P195MB2142.EURP195.PROD.OUTLOOK.COM
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Setting up the interface when suspended/resuming fails on this card.
Adding a reset and delay quirk will eliminate this problem.
usb 1-1: New USB device found, idVendor=0124, idProduct=0c21
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 1-1: Product: USB Headphone
usb 1-1: Manufacturer: Generic
usb 1-1: SerialNumber: 20210726905926
Signed-off-by: Lianqin Hu <hulianqin@vivo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/TYUPR06MB6217CBB68C8F868C076A4353D2C12@TYUPR06MB6217.apcprd06.prod.outlook.com
tas2563_save_calibration() assigns cali_data.total_sz before it reads the
per-device calibration data from EFI, but its error paths return without
clearing it again. cali_data.cali_reg_array is left all zero, because the
function returns before the register addresses are assigned.
On the first playback tasdev_load_calibrated_data() does
if (!data || !cali_data->total_sz)
return;
which passes, since total_sz is still non-zero. It then issues five
4-byte bulk writes to p->r0_reg, p->r0_low_reg, p->invr0_reg, p->pow_reg
and p->tlimit_reg, all of which are 0. Register 0 decodes to book 0 /
page 0 / register 0x00, so the auto-incrementing block write zeroes
registers 0x00 to 0x03. Register 0x03 is PB_CFG1, which holds AMP_LEVEL,
so the amplifier gain is set to its minimum and the speaker stays silent.
This is reproducible on a Lenovo Yoga 7 14ARB7 (two TAS2563 on I2C,
ACPI INT8866) whose factory calibration was never written to UEFI, so the
EFI read fails with EFI_NOT_FOUND. The two woofers driven by the
amplifiers are silent while the tweeters driven directly by the ALC287
play. Reading the amplifier registers over i2c shows PWR_CTL = 0x00
(active) and the TDM slots correctly programmed by the RCA profile, but
PB_CFG1 = 0x00. With this change PB_CFG1 keeps its power-on default of
0x20 and both woofers play.
tas2781_save_calibration() in tas2781_hda.c already clears total_sz on
failure; do the same for the TAS2563 variant.
Signed-off-by: Philipp Oster <philippdev5396@outlook.de>
Link: https://patch.msgid.link/20260720-tas2781-calfix-v1-1-3a5fa6ad90bc@outlook.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The firmware on the Lunnen Ground 14 marks pin 0x1b as unused even
though the internal speakers are connected to it. As a result, the
speakers are not detected.
Add a pin configuration quirk for PCI subsystem ID 2782:a212 to configure
pin 0x1b as an internal speaker.
The pin configuration was tested on a Lunnen Ground 14 (DMI product LL4FA)
with an ALC269VC codec. The internal speakers and microphone work as
expected.
Cc: stable@vger.kernel.org
Signed-off-by: Nikita Maksimov <nickstogramm@yandex.ru>
Link: https://patch.msgid.link/20260720180214.73770-1-nickstogramm@yandex.ru
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_timer_close_locked() drains the closing instance's own in-flight
callback (IFLG_CALLBACK) before freeing it, but not its slaves'. When a
master instance is closed, remove_slave_links() clears each slave's
->timer; the slave's own close then reads timer == NULL and takes the
branch that skips the drain entirely (snd_timer_stop_slave() also no-ops
on a NULL timer). So a slave whose callback is still running when the
master is closed is freed underneath the live callback, leading to
use-after-free.
Drain the slaves too before remove_slave_links() severs them.
snd_timer_stop() has already taken this instance off the active list, so
no new slave callback can be queued. Take the slaves off the ack list so
a pending one can't fire either, then wait for any that is already in
flight.
Fixes: 37745918e0 ("ALSA: timer: Introduce virtual userspace-driven timers")
Cc: stable@vger.kernel.org
Assisted-by: Claude:claude-opus-4-8
Signed-off-by: Norbert Szetei <norbert@doyensec.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/D26598EB-DBF7-4D76-9F71-8E4BD59822D4@doyensec.com
The userspace-driven timer (utimer) TRIGGER ioctl calls
snd_timer_interrupt() directly with no serialization, so two threads
triggering the same utimer can run snd_timer_interrupt() on one
snd_timer concurrently.
snd_timer_process_callbacks() drops timer->lock around each instance
callback and marks the in-flight callback with the single
SNDRV_TIMER_IFLG_CALLBACK bit; snd_timer_close_locked() waits on that
bit to drain an in-flight callback before freeing the instance. The bit
cannot represent two concurrent callbacks: when a second interrupt
re-queues an instance whose callback is still running, both run at once,
the first to finish clears the bit, and the close-path drain then frees
the instance (and its callback_data) while the other callback is still
live - a use-after-free reachable by any user able to open
/dev/snd/timer, both via a user timer instance and via a sequencer queue
timer bound to the utimer.
snd_timer_interrupt() sets IFLG_CALLBACK before dropping timer->lock, so
a concurrent interrupt already observes it under the lock. Skip
re-queuing an instance (and its slaves) to the ack/sack list while its
callback is in flight; the accumulated pticks are delivered on the next
tick, so no event is lost.
Fixes: 37745918e0 ("ALSA: timer: Introduce virtual userspace-driven timers")
Cc: stable@vger.kernel.org
Suggested-by: Takashi Iwai <tiwai@suse.de>
Assisted-by: Claude:claude-opus-4-8
Signed-off-by: Norbert Szetei <norbert@doyensec.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/6F9B6501-8E65-4265-B02C-7EFB240D1664@doyensec.com
JBL Quantum650 Wireless (0ecb:2125) requires the same workaround that
was used for JBL Quantum610 and Quantum810 for limiting the sample rate.
Without it, the capture (microphone) stream fails to work. Setting the
QUIRK_FLAG_FIXED_RATE flag, as done for the sibling models, makes both
playback and capture work correctly.
Signed-off-by: Daniel C. Ribeiro <dcoutinho.96@gmail.com>
Link: https://patch.msgid.link/20260719090037.40149-1-dcoutinho.96@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In cs35l56_init() use complete_all() to signal init_completion instead
of complete().
cs35l56_init() was signaling init_completion using the complete() function.
This only releases ONE waiter.
If cs35l56_component_probe() was called multiple times the first time
would consume that one signal, then future calls would timeout waiting for
the completion. This could happen if:
- The component is probed, removed, then probed again without the cs35l56
module being removed.
- A call to component_probe() returns an error and ASoC calls it again
later.
It should use complete_all() so that after it has been signaled it will
allow any code that waits on it to continue immediately.
The one case where the driver must wait for initialization to run again is
when waiting for a reboot after firmware download, and here the code
correctly calls reinit_completion() first.
Fixes: e496112529 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260716132045.1469156-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On I2C/SPI call cs35l56_init() before calling
snd_soc_register_component() to prevent the potential for a deadlock
on init_completion.
For most buses all the hardware would be ready when probe() returns,
but on SoundWire, probe() must return before the SoundWire bus driver
will enumerate the device. All access to the registers must be deferred
until the driver receives an ATTACHED notification. But anything that
could return -EPROBE_DEFER must be called during probe, and that includes
snd_soc_register_component(). Because of that, on SoundWire the ASoC
component can be created before the registers are accssible, so
cs35l56_component_probe() waits for init_completion to signal that the
registers are accessible.
On I2C/SPI this 2-stage startup isn't required so their probe()
functions simply called cs35l56_common_probe() and then cs35l56_init().
The problem with this was that snd_soc_register_component() was still
called early. If this triggered ASoC to create the card, ASoC would call
cs35l56_component_probe() which waits on init_completion - but this would
be running inside the cs35l56 driver probe() so blocking it from reaching
the code that signals init_completion, causing a deadlock.
Fixes: e496112529 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Reported-by: Salman S. Tahir <salman.abusaad@gmail.com>
Closes: https://lore.kernel.org/linux-sound/95c21574-97d5-4311-9263-9e174d22d22c@opensource.cirrus.com/T/#u
Tested-by: Salman S. Tahir <salman.abusaad@gmail.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260716132045.1469156-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The fs210x driver currently puts the regmap cache into cache only mode
during suspend but at no point does it make it writable again, meaning
that device can't work after suspend other than basic powerup which
bypasses the cache.
Mark the cache as writable again when we resync the registers, it's not
clear to me that the cache only mode is needed at all but this is a
fairly minimal change. Since this is setting a flag it's safe to do
even if the cache was not previously in cache only mode.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20260715-asoc-fs210x-fix-resume-v1-1-dd81d9c62c05@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
HP ZBook 8 G2a 14 and 16 (SSIDs 0x103c:0x8f94, 0x103c:0x8f95) use the
Realtek ALC245 codec with a TAS2781 amplifier via I2C. They share the same
hardware configuration as the existing 0x8f40/0x8f41/0x8f42/0x8f62 models
but have inverted speaker mute LED polarity, so the existing
ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED quirk drives the LED backwards: off
when muted and on when unmuted.
Add a dedicated quirk with inverted COEF values. These are speaker-only
models without an HP pin, so the LED is driven directly through the
vmaster_mute hook; there is no need to probe the HP pin at runtime since
the configuration is static and known per SSID.
Signed-off-by: Chris Chiu <chris.chiu@canonical.com>
Link: https://patch.msgid.link/20260716023825.387532-1-chris.chiu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a keep-alive (KAE) silent stream is active on an Intel HDMI/DP
codec, opening a real PCM stream reprograms the converter format and the
audio infoframe in snd_hda_hdmi_generic_pcm_prepare(). Part of that
reprogramming - the converter channel count and the channel mapping in
snd_hda_hdmi_setup_audio_infoframe() - is not safe to do while a
keep-alive stream is active. This is most visible when switching to a
multichannel PCM configuration, where the active channel count actually
changes. In that case the newly opened PCM stream plays no sound.
Add an optional hdmi_ops .prepare hook, called at the start of the
PCM prepare sequence (before the format and infoframe are touched), and
implement it for HSW+ to release keep-alive. Keep-alive is then
re-enabled as before once the new stream has been set up, in the
setup_stream op.
Fixes: 15175a4f2b ("ALSA: hda/hdmi: add keep-alive support for ADL-P and DG2")
Reported-by: Alexander Kaplan <alexander.kaplan@sms-medipool.de>
Closes: https://gitlab.freedesktop.org/drm/xe/kernel/-/work_items/8412
Tested-by: Alexander Kaplan <alexander.kaplan@sms-medipool.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://patch.msgid.link/20260715180610.1371243-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The bt-sco-pcm-wb DAI uses the same stream_name strings as bt-sco-pcm
("Playback" and "Capture"). This causes duplicate DAPM AIF widget
names within the same component, leading to debugfs warnings:
debugfs: 'Playback' already exists in 'dapm'
debugfs: 'Capture' already exists in 'dapm'
Give the wideband DAI distinct stream names ("WB Playback" and
"WB Capture") and add corresponding DAPM AIF widgets and routes for
them.
Fixes: 5947e1b499 ("ASoC: bt-sco: extend rate and add a general compatible string")
Assisted-by: VeroCoder:claude-sonnet-4-5
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/20260715100620.1387159-1-shengjiu.wang@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cs35l41_get_acpi_mute_state() evaluates a _DSM method to get the ACPI
mute state and reads the first byte from the returned object.
However, the returned ACPI object is owned by the caller and is never
freed after use, so each successful query leaks the _DSM result object.
The code also assumes that the returned object is a buffer with at least
one byte. A malformed firmware response can return a different object
type or an empty buffer, and the direct ret->buffer.pointer dereference
can then access an invalid pointer.
Use the typed _DSM helper, validate that the returned buffer contains at
least one byte, and free the ACPI object after reading it.
Fixes: 447106e92a ("ALSA: hda: cs35l41: Support mute notifications for CS35L41 HDA")
Signed-off-by: Guangshuo Li <lgs201920130244@gmail.com>
Link: https://patch.msgid.link/20260708113625.752913-1-lgs201920130244@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Alienware x16 R2 has two pairs of speakers, but the BIOS
marks pin 0x17 as unused, so only the pin 0x14 pair plays and
audio is very quiet/dull.
Apply ALC289_FIXUP_DUAL_SPK like on other Dell machines to set
up pin 0x17 and route it to DAC1. Tested on my x16 R2 with
kernel 6.18.38, and now all speakers play at full volume.
Signed-off-by: Oliver Ohrt <oliver@theohrts.com>
Link: https://patch.msgid.link/20260715070409.42696-1-oliver@theohrts.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The sound and microphone mute LEDs do not function on this newer
revision of the board (8AB8) while they do on the older 880D models.
I have verified this on another laptop which was manufactured before
the one with the issue.
Added the ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED from a G9 model, which
uses the same codec, to make it work. Tested on kernel version 7.1.3 on
the aforementioned newer revision notebook.
Signed-off-by: Marcel Kłos <marcel@marmak.net.pl>
Link: https://patch.msgid.link/4dab5622-9100-4730-8c99-b58da939549b@marmak.net.pl
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The internal digital microphone on the MSI Vector A16 HX A8WIG is not
detected: the ACP platform devices are created, but snd_soc_acp6x_mach
never binds because the machine is missing from the DMI quirk table, so
no capture device shows up at all.
This is the same board as the already supported "Vector A16 HX A8WHG",
differing only in the trailing model code. Add the corresponding entry.
Signed-off-by: Antonio Ignacio Campos Ruiz <acamposruiz@gmail.com>
Link: https://patch.msgid.link/20260713165709.19489-1-acamposruiz@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In a full-duplex setup, when one direction (playback or capture) is
closed while the other is still running, imx_aif_shutdown() was
unconditionally calling snd_soc_dai_set_sysclk() with rate=0 for all
cpu/codec DAIs, which would disable the clock still needed by the
active stream.
Add snd_soc_dai_active() checks before clearing sysclk so that only
truly inactive DAIs have their clocks reset.
Fixes: 2260bc6ea8 ("ASoC: imx-card: Add WM8524 support")
Cc: stable@vger.kernel.org
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://patch.msgid.link/20260710031333.3491445-1-chancel.liu@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The H616 codec does not have capture capabilities. Set the
.playback_only quirks flag to denote this.
This was somehow missing from the original driver patch, even though
the patch prior to it in the series added this quirk.
Fixes: 9155c321a1 ("ASoC: sun4i-codec: support allwinner H616 codec")
Signed-off-by: Chen-Yu Tsai <wens@kernel.org>
Link: https://patch.msgid.link/20260714113304.270224-1-wens@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The TAS2781 firmware parser reads several variable-length description
strings with strlen() before checking that the string terminator is
present inside the firmware blob. A malformed firmware image without a
NUL terminator can therefore make the parser walk past the end of the
firmware buffer before the later size checks run.
Add a small bounded string-length helper and use it for all description
fields that are parsed from the firmware buffer. Keep the existing size
checks for the fixed bytes that follow each string.
Fixes: 915f5eadeb ("ASoC: tas2781: firmware lib")
Signed-off-by: Pengpeng Hou <pengpeng@iscas.ac.cn>
Link: https://patch.msgid.link/20260706144540.93929-1-pengpeng@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
queue_delete() closes the queue timer, then frees it. snd_seq_timer_close()
clears q->timer->timeri. snd_use_lock_sync() then drains borrowers, and
snd_seq_timer_delete() frees q->timer.
A borrower can re-open the timer inside that window. A SET_QUEUE_CLIENT
that took a queueptr() use_lock reference before the queue was unlinked
runs snd_seq_timer_open() after the close. Open refuses re-open only while
timeri is set, and the close just cleared it, so it re-opens timeri.
snd_seq_timer_delete() does not close that instance. Its snd_seq_timer_stop()
is a no-op, because running was cleared first. So it frees q->timer with the
instance still live. The queue is freed next.
The instance stays on the global timer with callback_data pointing at the
freed queue. A non-owner START on the unlocked queue arms it. The next tick
derefs the freed queue in snd_seq_timer_interrupt().
Reachable by an unprivileged user with access to /dev/snd/seq. No CAP and
no queue ownership required.
Close any lingering instance in the destructor. There, ->timeri can no
longer change: the queue is unlinked and all use_lock borrowers have
drained, so no snd_seq_queue_use() can re-open it. Close it before clearing
q->timer. snd_timer_close() waits for any in-flight snd_seq_timer_interrupt()
to finish, and that callback still reads q->timer (via snd_seq_check_queue()),
so q->timer must stay valid until it drains.
Fixes: 1da177e4c3 ("Linux-2.6.12-rc2")
Cc: stable@vger.kernel.org
Assisted-by: Claude:claude-opus-4-8
Signed-off-by: Norbert Szetei <norbert@doyensec.com>
Link: https://patch.msgid.link/422FDB81-2A68-47C7-A22D-2D3301E2E86D@doyensec.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some ASUS TUF ACP70-based systems expose ACP ACPI configuration flags
that select a non-working fallback audio path, similar to previously
affected ASUS platforms.
Add DMI-based overrides in snd_amd_acp_find_config() for the following
systems to skip ACP ACPI flag-based selection:
- ASUS TUF Gaming Vivobook 18
- ASUS TUF Gaming A14 FA401EA
This ensures the intended SoundWire-based machine driver is selected on
these platforms.
Signed-off-by: Syed Saba Kareem <Syed.SabaKareem@amd.com>
Link: https://patch.msgid.link/20260710102926.1633385-1-syed.sabakareem@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Audio control requests that sets sampling frequency sometimes fail on
this card. Adding delay between control messages eliminates that problem.
usb 1-1: New USB device found, idVendor=2fc6, idProduct=f0b5
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 1-1: Product: iBasso DC-Elite
usb 1-1: Manufacturer: iBasso
usb 1-1: SerialNumber: CTUA171130B
Signed-off-by: Lianqin Hu <hulianqin@vivo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/TYUPR06MB6217D8FF419F24378196FCEFD2FA2@TYUPR06MB6217.apcprd06.prod.outlook.com
Remove the mic bias current comparator threshold override (NID 0x1c,
verb 0x320, value 0x010) from Conexant codec driver.
This override was originally intended to support volume up/down controls on
headsets with inline remote controls, but it causes microphone detection
failures on some headsets with impedance less than 1k ohm.
After consulting with the vendor's engineers, it was confirmed that this
setting is board-specific and should be handled by BIOS/firmware rather
than the generic codec driver, especially since inline remote support
is not currently implemented.
Fixes: 7aeb259086 ("ALSA: hda/conexant: Fix headset auto detect fail in cx8070 and SN6140")
Cc: stable@vger.kernel.org
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260713100329.306892-1-zhangheng@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the BCLK divider ratio is 1:1, fsl_sai_set_bclk() enables bypass
mode by setting BYP, but never clears the bit. The BYP=1 value remains
in the regcache, and is restored by regcache_sync() on the next runtime
resume.
Since BYP=1 combined with BCD=1 immediately outputs the ungated MCLK
as BCLK without waiting for BCE/TE/RE to be enabled, the clock is
driven prematurely before the stream is fully configured, causing
noise on some codecs.
Fix this by clearing BYP and BCI in fsl_sai_hw_free() taking into
account sync mode and the opposite stream's state, so that the regcache
holds BYP=0 before runtime suspend and regcache_sync() on resume will
not restore bypass mode prematurely.
Fixes: a50b7926d0 ("ASoC: fsl_sai: implement 1:1 bclk:mclk ratio support")
Cc: stable@vger.kernel.org
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20260710070835.3749817-1-chancel.liu@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some units of the Lenovo Legion Pro 7 16ARX8H (82WS) report codec
subsystem ID 17aa:38a7 instead of 17aa:38a8. Since only 38a8 has a
codec SSID quirk, these machines fall through to the PCI SSID match
17aa:386f (Legion Pro 7i 16IAX7) and get ALC287_FIXUP_CS35L41_I2C_2,
which probes the Cirrus amplifiers of the Intel variant. The TI
TAS2781 amplifier (ACPI TIAS2781:00) present on this AMD variant is
never bound and the internal speakers remain silent.
Add a codec SSID quirk for 17aa:38a7 pointing to
ALC287_FIXUP_TAS2781_I2C, mirroring the existing 38a8 entry.
Tested on a Legion Pro 7 16ARX8H (82WS, BIOS LPCN62WW): with the codec
SSID overridden to 17aa:38a8 via the HDA patch loader, the TAS2781
amplifier binds and the internal speakers work.
Cc: <stable@vger.kernel.org>
Signed-off-by: Damien Laine <damien.laine@gmail.com>
Link: https://patch.msgid.link/20260712213708.1835469-1-damien.laine@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The internal speakers on the MECHREVO WUJIE Series are silent,
while the headphone output works correctly.
The BIOS reports NID 0x1b on the Realtek ALC233 codec as
unconnected with pin configuration 0x411111f0. However, the pin
is connected to an internal speaker.
Overriding NID 0x1b with 0x90170150 makes the HDA generic parser
detect both 0x14 and 0x1b as speaker outputs and restores internal
speaker playback.
Add a pin configuration fixup for the affected PCI SSID c011:1d05.
Tested on a MECHREVO WUJIE Series laptop with board
WUJIE Series-Lark4-LNL and firmware EM_LNL326_V1.0.23.
Signed-off-by: Chen Bowen <hicbowen@gmail.com>
Link: https://patch.msgid.link/20260710133409.16641-1-hicbowen@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fail cs35l56_hda_fw_load() and log an error if it didn't find a wmfw file
and the BIOS didn't patch the firmware already. Also remove a section of
code from cs35l56_hda_request_firmware_files() that attempted to find a
.bin if a .wmfw was not found.
The CS35L56 ROM can only provide default audio in SoundWire mode. A wmfw
is needed to enable I2S audio. Also none of the customer-specific .bin
files are compatible with the ROM firmware. So a .wmfw file is always
required.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260709161211.686498-1-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Salvador reported that the recent fix for applying the DSD quirk to
Musical Fidelity devices broke for his M6s DAC model (2772:0502).
Although this is basically a firmware bug, the model in question is
fairly old, and no further firmware update can be expected, so it'd be
better to address in the driver side.
As an ad hoc workaround, skip the DSD quirk for this device by adding
an empty quirk entry of 2772:0502; this essentially skips the later
DSD quirk entry by the match with the vendor 2772.
Fixes: da3a7efff6 ("ALSA: usb-audio: Update for native DSD support quirks")
Reported-by: Salvador Blaya <tiniebla6@gmail.com>
Closes: https://lore.kernel.org/CAOdyq+qFaqCh=tK_wNnA64hv5pQuA1Y09ANxQ=xK8yR-t4mf9Q@mail.gmail.com
Tested-by: Salvador Blaya <tiniebla6@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260709095614.1418838-1-tiwai@suse.de
