Commit Graph

56388 Commits

Author SHA1 Message Date
Takashi Iwai
a8934c2c6d ALSA: usb-audio: Fix sticky mixer regressions on M-Audio Fast Track Ultra
The recent fix for sticky mixer volumes caused regressions of M-audio
Fast Track Ultra device, where the mixer state is kept to the default
value.

Add the quirk entries to tolerate the broken mixer behavior.  As the
device is known to work in the implicit feedback mode, explicitly
enable the implicit feedback mode, too.

Since there are two FTU models that are almost identical, both entries
are added in this patch (0763:2080 and 0763:2081).

Fixes: 86aa1ea1f1 ("ALSA: usb-audio: Do not expose sticky mixers")
Link: https://bugzilla.suse.com/show_bug.cgi?id=1273166
Link: https://patch.msgid.link/20260807083418.1712585-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-08-07 10:35:31 +02:00
Takashi Iwai
d03cdb1c36 Merge tag 'asoc-fix-v7.2-rc6' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v7.2

A moderately large set of fixes, but mostly unremarkable:

 - A series of robustness fixes for the SOF code that came from testing.
 - Fixes for user visible issues with some of the Qualcomm controls.
 - Fixes to the register default tables in Cirrus drivers which ensure
   correcy syncing on resume.  This is a wider problem which will be
   fixed with a regmap core change but I'd already applied these, the
   rest of the fixes will come in the merge window.
 - Making the existing RT645 driver user selectable so it can be used
   with more machines.
 - Quite a few new AMD laptop quirks.
 - A MAINTAINERS update for SpacemiT.
2026-08-06 13:55:51 +02:00
Mark Brown
33c5aaf693 ASoC: cs35l41/cs35l45/cs4265: sort the reg_defaults tables
Peter Ujfalusi <peter.ujfalusi@linux.intel.com> says:

reg_defaults must be sorted by ascending register address as
regcache_lookup_reg() locates the entries in it with bsearch(), see commit
fd80df352b ("regcache: Add support for sorting defaults arrays").

These three tables have entries which are out of order, so the binary search
does not find part of them.  For those registers regcache_reg_needs_sync()
cannot compare the cached value against the default and reports that a sync
is needed, so they are written to the device on every regcache_sync() even
when they were never touched.

The patches only reorder the existing entries, the text of every entry is
kept verbatim and no default value is changed.  Each table was verified by
evaluating the register addresses and replaying lib/bsearch.c on them.

Entries not reachable by the binary search, per table:

  cs35l41_reg           2 (of 47)
  cs35l45_defaults     36 (of 73)
  cs4265_reg_defaults   3 (of 16)

For cs35l45 this is nearly half of the table: the DSP1_RX*_RATE and
DSP1_TX*_RATE registers sit in the middle of it while their addresses are
far above everything else, which cuts the search off from the whole
0x4c40 - 0xf010 range.

Found by an audit of all reg_defaults tables under sound/, the SoundWire
codec drivers are fixed by a separate series.

Link: https://patch.msgid.link/20260805082413.26174-1-peter.ujfalusi@linux.intel.com
2026-08-05 13:34:37 +01:00
Peter Ujfalusi
e4fe3e0465 ASoC: cs4265: sort the register default table
reg_defaults must be sorted by ascending register address, as
regcache_lookup_reg() locates entries in it with bsearch().  See commit
fd80df352b ("regcache: Add support for sorting defaults arrays").

cs4265_reg_defaults[] lists CS4265_INT_MASK (0x0e),
CS4265_STATUS_MODE_MSB (0x0f) and CS4265_STATUS_MODE_LSB (0x10) after
CS4265_SPDIF_CTL1 (0x11) and CS4265_SPDIF_CTL2 (0x12), so the binary search
does not find those three entries.  regcache_reg_needs_sync() then cannot
compare them against their default and reports that a sync is needed, so
they are written to the device on every regcache_sync() even when they were
never touched.

Sort the table by register address.

Fixes: fb6f806967 ("ASoC: Add support for the CS4265 CODEC")
Cc: stable@vger.kernel.org
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260805082413.26174-4-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-08-05 13:34:36 +01:00
Peter Ujfalusi
f39a68ed08 ASoC: cs35l45: sort the register default table
reg_defaults must be sorted by ascending register address, as
regcache_lookup_reg() locates entries in it with bsearch().  See commit
fd80df352b ("regcache: Add support for sorting defaults arrays").

cs35l45_defaults[] lists the DSP1_RX*_RATE and DSP1_TX*_RATE registers
(0x02b80080 - 0x02b802b8) in the middle of the table, ahead of entries with
much lower addresses, so the binary search does not find 36 of its 73
entries.  regcache_reg_needs_sync() then cannot compare those against their
default and reports that a sync is needed, so they are written to the
device on every regcache_sync() even when they were never touched.

Sort the table by register address.

Fixes: 74b14e2850 ("ASoC: cs35l45: DSP Support")
Cc: stable@vger.kernel.org
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260805082413.26174-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-08-05 13:34:35 +01:00
Peter Ujfalusi
d74aac116c ASoC: cs35l41: sort the register default table
reg_defaults must be sorted by ascending register address, as
regcache_lookup_reg() locates entries in it with bsearch().  See commit
fd80df352b ("regcache: Add support for sorting defaults arrays").

cs35l41_reg[] lists CS35L41_BSTCVRT_PEAK_CUR (0x3808) after
CS35L41_BSTCVRT_COEFF (0x3810) and CS35L41_BSTCVRT_SLOPE_LBST (0x3814), so
the binary search does not find those two entries.
regcache_reg_needs_sync() then cannot compare them against their default
and reports that a sync is needed, so they are written to the device on
every regcache_sync() even when they were never touched.

Sort the table by register address.

Fixes: 5f2f539901 ("ASoC: cs35l41: Correct handling of some registers in the cache")
Cc: stable@vger.kernel.org
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260805082413.26174-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-08-05 13:34:34 +01:00
Zhang Heng
1b0e5c7a4d ASoC: amd: yc: Add DMI quirk for MSI Raider A18 HX A7VHG
Add a DMI quirk for the MSI Raider A18 HX A7VHG fixing the
issue where the internal microphone was not detected.

Link: https://bugzilla.kernel.org/show_bug.cgi?id=221574
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260805091600.318018-2-zhangheng@kylinos.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-08-05 12:52:55 +01:00
Zhang Heng
bc734d167a ASoC: amd: yc: Add DMI quirk for Xiaomi RedmiBook 16 2025
Add a DMI quirk for the Xiaomi RedmiBook 16 2025 (AMD) fixing the
issue where the internal microphone was not detected.

Link: https://github.com/thesofproject/linux/issues/5860
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260805091600.318018-1-zhangheng@kylinos.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-08-05 12:52:54 +01:00
Baul Lee
2ca1eea3cd ALSA: usx2y: bound the hwdep mmap fault offset
snd_us428ctls_vm_fault() turns the faulting page offset into a kernel
address with no bound of any kind:

	offset = vmf->pgoff << PAGE_SHIFT;
	vaddr = (char *)(...)->us428ctls_sharedmem + offset;
	page = virt_to_page(vaddr);
	get_page(page);
	vmf->page = page;

	return 0;

snd_us428ctls_mmap() checks only the length of the mapping, never the
offset, and us428ctls_sharedmem is a single page from
alloc_pages_exact().  For a character device file_mmap_size_max()
returns ULONG_MAX, so the mm layer imposes no ceiling either.  Every page
offset above zero resolves to a struct page outside the object, and the
handler installs it into the caller's address space read-write; the vma
is not marked read-only.

The caller picks the page frame with a single mmap() argument and gets
read-write access to a page of kernel memory it does not own; an offset
that lands in an unpopulated vmemmap region oopses instead.

A process that can open the hwdep node of an attached US-X2Y reaches
this after loading the FPGA image through the same node; no capability
check is involved.

On 7.2.0-rc5 (arm64), mmap() with a large offset:

  Unable to handle kernel paging request at virtual address fffffdffc45d5ac8
  pc : snd_us428ctls_vm_fault+0x68/0x140 [snd_usb_usx2y]
  Call trace:
   snd_us428ctls_vm_fault+0x68/0x140 [snd_usb_usx2y]
   __do_fault
   __handle_mm_fault
   handle_mm_fault
   el0_da

Reject any offset outside the shared region.  The pcm hwdep handler in
usx2yhwdeppcm.c computes its address the same way and needs the same
bound.

Discovered by XBOW, triaged by Baul Lee <baul.lee@xbow.com>

Fixes: 1da177e4c3 ("Linux-2.6.12-rc2")
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/20260805013445.38283-1-baul.lee@xbow.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-08-05 09:37:27 +02:00
Baul Lee
69ee44e1a2 ALSA: usb-audio: fix OOB write on Type II inbound URBs
data_ep_set_params() sizes each URB transfer buffer before it adds the
Format Type II transfer delimiter:

	u->packets = urb_packs;
	u->buffer_size = maxsize * u->packets;

	if (fmt->fmt_type == UAC_FORMAT_TYPE_II)
		u->packets++; /* for transfer delimiter */
	u->urb = usb_alloc_urb(u->packets, GFP_KERNEL);

buffer_size is computed from the pre-increment packet count and never
recomputed, so for a Type II endpoint the buffer is one packet short of
the packet count the URB is built with.

prepare_inbound_urb() then lays out one iso frame per packet and never
consults buffer_size:

	offs = 0;
	for (i = 0; i < urb_ctx->packets; i++) {
		urb->iso_frame_desc[i].offset = offs;
		urb->iso_frame_desc[i].length = ep->curpacksize;
		offs += ep->curpacksize;
	}

	urb->transfer_buffer_length = offs;
	urb->number_of_packets = urb_ctx->packets;

The last descriptor therefore points one packet past the end of the
transfer buffer, where the host controller writes device data on every
inbound transfer.  prepare_silent_urb() and prepare_playback_urb() bound
their fill loops by ctx->buffer_size, so only capture is affected.

fmt_type comes from the device's audio streaming descriptors, so any
device advertising a Type II capture format hits this once userspace sets
hw_params on the stream.

KASAN on 7.2.0-rc5 (arm64) with a dummy_hcd/raw-gadget device, one report
per inbound transfer:

  BUG: KASAN: slab-out-of-bounds in dummy_timer
  Write of size 64 at addr ffff0000186171c0 by task cons02/166
   __asan_memcpy
   dummy_timer
   hrtimer_run_softirq
  Allocated by task 166:
   usb_alloc_coherent
   snd_usb_endpoint_set_params
  The buggy address is located 0 bytes to the right of
   allocated 64-byte region [ffff000018617180, ffff0000186171c0)

Compute buffer_size after the delimiter packet has been accounted for,
and bound the fill loop by buffer_size, as prepare_silent_urb() already
does on the outbound side.  This grows every Type II URB allocation by
one maxsize packet.

Discovered by XBOW, triaged by Baul Lee <baul.lee@xbow.com>

Fixes: 8fdff6a319 ("ALSA: snd-usb: implement new endpoint streaming model")
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/20260805013441.38245-1-baul.lee@xbow.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-08-05 09:35:28 +02:00
Baul Lee
2615f0fb90 ALSA: us144mkii: re-anchor capture URBs on resubmission
capture_urb_complete() resubmits each capture URB without anchoring it:

	usb_get_urb(urb);
	ret = usb_submit_urb(urb, GFP_ATOMIC);

Anchoring is a property of a submission, not of the URB.  The giveback
path calls usb_unanchor_urb() before urb->complete(), so an URB
resubmitted from its own completion handler is off the anchor.  The
capture URBs are anchored once, at stream start, so from the first
completion onward tascam->capture_anchor is empty.

tascam_free_urbs(), tascam_disconnect(), tascam_suspend() and the
stop-work path all call usb_kill_anchored_urbs(&tascam->capture_anchor)
to reap the capture URBs before anything is freed.  With the anchor empty
those calls return immediately and the URBs stay queued on the host
controller.

tascam_free_urbs() then returns the capture transfer buffers with
usb_free_coherent(), and snd_card_free() releases the snd_card
allocation that embeds tascam (card->private_data).  The controller
completes the queued URBs afterwards, writing device-supplied data into
the freed transfer buffer, and capture_urb_complete() dereferences the
freed driver object.

KASAN on 7.2.0-rc5 (arm64):

  BUG: KASAN: slab-use-after-free in dummy_timer
  Write of size 512 at addr ffff000015b62000
   __asan_memcpy
   dummy_timer
   hrtimer_run_softirq
  Allocated by task 64:
   usb_alloc_coherent
   tascam_alloc_urbs
   tascam_probe
  Freed by task 170:
   usb_free_coherent
   tascam_free_urbs
   tascam_disconnect
   usb_unbind_interface

  BUG: KASAN: slab-use-after-free in capture_urb_complete
  Read of size 4 at addr ffff0000170ee878
  Freed by task 170:
   release_card_device
   snd_card_free
   tascam_disconnect

Restore the usb_anchor_urb() between the reference count bump and the
resubmission.  That also makes the handler's usb_unanchor_urb() failure
arm meaningful again and restores usb_kill_anchored_urbs() as a barrier
on the disconnect, suspend and stop-work paths.

The anchoring was removed on the premise that the URB is already anchored
from the initial submission, which does not hold once the first giveback
has run.

Discovered by XBOW, triaged by Baul Lee <baul.lee@xbow.com>

Fixes: 5cff1529a2 ("ALSA: us144mkii: capture_urb_complete: redundant usb_anchor_urb corrupts anchor list on each resubmission")
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/20260804123625.91769-1-baul.lee@xbow.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-08-04 18:06:37 +02:00
Baul Lee
620f1e52a4 ALSA: FCP: fix OOB write in fcp_meter_ctl_get()
fcp_ioctl_set_meter_map() bounds the user-supplied Level Meter map size
by the driver's own limit of 255

	if (map.map_size < 1 || map.map_size > 255 ||
	    map.meter_slots < 1 || map.meter_slots > 255)
		return -EINVAL;

and passes it to fcp_add_new_ctl() as the control's channel count, where
it is stored as elem->channels.

Every control read writes into struct snd_ctl_elem_value, whose integer
array is declared long value[128], so the limit is 128, not 255.
fcp_meter_ctl_get() stores one 64-bit word per channel into that array
with no bound of its own:

	for (i = 0; i < elem->channels; i++) {
		int idx = private->meter_level_map[i];
		int value = idx < 0 ? 0 : le32_to_cpu(resp[idx]);

		ucontrol->value.integer.value[i] = value;
	}

snd_ctl_elem_read_user() serves that object from
memdup_user(_control, sizeof(*control)), 1224 bytes on LP64 out of
kmalloc-2048.  offsetof(struct snd_ctl_elem_value, value) is 72, so
element i is written at byte 72 + 8 * i and element 144 already lands
past the allocation.  At map_size 255 the last store ends at byte 2112,
888 bytes past the object and 64 bytes into the adjacent slab object.
The stored words come from the device and meter_level_map[] selects
which word lands in which slot, so extent and contents are both
controlled.

The core does not catch this.  snd_ctl_check_elem_info() is reached only
from __snd_ctl_elem_info(), which snd_ctl_elem_read() calls under
CONFIG_SND_CTL_DEBUG; without that option snd_ctl_skip_validation() is a
compile-time true.  __snd_ctl_add_replace() validates kcontrol->count and
never inspects elem->channels.

Installing an oversized map needs CAP_SYS_RAWIO, but the control outlives
the hwdep descriptor that created it, so the out-of-bounds stores are
issued by any process able to read controls on /dev/snd/controlC0.

KASAN on 7.2.0-rc5 (arm64), triggered by an unprivileged control read:

  BUG: KASAN: slab-out-of-bounds in fcp_meter_ctl_get
  Write of size 8 at addr ffff000017af04c8 by task fcp_trigger/185
   __asan_store8
   fcp_meter_ctl_get
   snd_ctl_elem_read
   snd_ctl_ioctl
  Allocated by task 185:
   memdup_user
   snd_ctl_ioctl
  The buggy address is located 0 bytes to the right of
   allocated 1224-byte region [ffff000017af0000, ffff000017af04c8)

Bound the map size by the ABI limit rather than by 255, and bound the
store loop at the sink so it cannot run past the value array whatever
elem->channels holds.

Discovered by XBOW, triaged by Baul Lee <baul.lee@xbow.com>

Fixes: 46757a3e7d ("ALSA: FCP: Add Focusrite Control Protocol driver")
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/20260804123611.91715-1-baul.lee@xbow.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-08-04 18:05:50 +02:00
Rudi Heitbaum
588852647b ASoC: rt5645: Make the Kconfig symbol user selectable
SND_SOC_RT5645 has no prompt, so it can only be turned on by a machine
driver that selects it. Every such driver is x86, MediaTek or Rockchip,
which leaves the codec unreachable on any other platform.

The part has a devicetree binding of its own,
Documentation/devicetree/bindings/sound/realtek,rt5645.yaml, and nothing
in the driver is machine specific, so a board that describes it with
simple-audio-card or audio-graph-card should be able to build it. Today
there is no configuration in which that is possible.

Give the symbol a prompt, matching SND_SOC_RT5640 immediately above it.
The machine drivers that select it are unaffected.

Signed-off-by: Rudi Heitbaum <rudi@heitbaum.com>
Link: https://patch.msgid.link/anCcciWmO6QOLTsl@5e001e58230e
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-08-03 19:45:03 +01:00
Rong Zhang
c973e61895 ALSA: usb-audio: Add QUIRK_FLAG_MIXER_GET_CUR_BROKEN for Logitech PRO X 2 LIGHTSPEED
The UAC mixer of the Logitech PRO X 2 LIGHTSPEED has broken mixer
GET_CUR behavior but otherwise works fine.

Add a quirk table entry matching VID/PID=0x046d/0x0af7 and apply the
MIXER_GET_CUR_BROKEN quirk flag to make the mixer usable again.

Quirky device sample (after applying the quirk flag):

  usb 3-2.1: New USB device found, idVendor=046d, idProduct=0af7, bcdDevice= 1.00
  usb 3-2.1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
  usb 3-2.1: Product: PRO X 2 LIGHTSPEED
  usb 3-2.1: Manufacturer: Logitech
  usb 3-2.1: SerialNumber: 0000000000000000
  usb 3-2.1: 2:0: broken mixer GET_CUR (-18944/0/256 => -2662)
  usb 3-2.1: 6:0: broken mixer GET_CUR (-18944/0/256 => 0)

Fixes: 86aa1ea1f1 ("ALSA: usb-audio: Do not expose sticky mixers")
Suggested-by: Brian van den Berg <faxuser@proton.me>
Reported-by: Brian van den Berg <faxuser@proton.me>
Link: https://lore.kernel.org/all/370007e6-b73b-4bfc-8410-a860781c7ad7@proton.me/
Signed-off-by: Rong Zhang <i@rong.moe>
Link: https://patch.msgid.link/20260731-uac-lg-pro-x-2-ls-v1-1-268eaefe66ab@rong.moe
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-08-03 09:37:12 +02:00
Xu Rao
8bec01c80e ALSA: hda/tas2781: fix ACPI reference handling
tas2781_read_acpi() gets a reference to the matching ACPI device and then
looks up its first physical device node. After taking a reference to the
physical device, it immediately drops the ACPI device reference.

However, every later failure jumps to an error path that drops the ACPI
device reference a second time. This unbalances the reference count and
may prematurely release the ACPI device.

In addition, acpi_get_first_physical_node() may return NULL. Without a
check, the driver passes the NULL physical device to the property helper
calls and may dereference it.

Return -ENODEV when no physical device is associated with the ACPI node,
and remove the duplicate acpi_dev_put() from the common error path.

Fixes: bb5f86ea50 ("ALSA: hda/tas2781: Add tas2781 hda SPI driver")
Cc: stable@vger.kernel.org
Signed-off-by: Xu Rao <raoxu@uniontech.com>
Link: https://patch.msgid.link/97EA8F29DA0D9AF7+20260731033554.949564-1-raoxu@uniontech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-08-03 08:56:42 +02:00
Mark Brown
ee145d35e4 ASoC: codecs: lpass-{tx,wsa}-macro: fix enum kcontrol accesses
Dawid Wróbel <me@dawidwrobel.com> says:

Both drivers access enumerated controls through value.integer.value[0]
instead of value.enumerated.item[0]. The same bug was fixed in rx-macro
and va-macro in 2022 (bcfe5f76cc, 0ea5eff7c6); tx-macro and
wsa-macro were missed.

On 64-bit kernels with CONFIG_SND_CTL_DEBUG this trips the elem value
sanity check, and every read of the affected controls fails with
-EINVAL.

Reproduced and fixed on a Xiaomi Mi Pad 5 Pro (SM8250) for tx-macro.
wsa-macro is compile-tested only — that codec is not instantiated on
this hardware.

Link: https://patch.msgid.link/20260730-worktree-lpass-tx-macro-enum-fix-v2-0-6d091c736116@dawidwrobel.com
2026-08-01 01:02:01 +01:00
Dawid Wróbel
56f24311fd ASoC: codecs: lpass-wsa-macro: Fix enum kcontrol accesses
EAR SPKR PA Gain" and the four "WSA RX* Mux" controls are enumerated,
but their get and put callbacks access the value through
ucontrol->value.integer.value[0] (a long) instead of
ucontrol->value.enumerated.item[0] (an unsigned int).

This same pattern was fixed in the sibling drivers by
commit bcfe5f76cc ("ASoC: codecs: rx-macro: fix accessing array
out of bounds for enum type") and
commit 0ea5eff7c6 ("ASoC: codecs: va-macro: fix accessing array
out of bounds for enum type"), but wsa-macro was missed.

On 64-bit kernels with CONFIG_SND_CTL_DEBUG this trips the elem value
sanity check and every read of these controls fails with -EINVAL.

Fixes: 809bcbcece ("ASoC: codecs: lpass-wsa-macro: Add support to WSA Macro")
Fixes: 2c4066e5d4 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route")
Assisted-by: Claude:claude-fable-5
Cc: stable@vger.kernel.org
Signed-off-by: Dawid Wróbel <me@dawidwrobel.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260730-worktree-lpass-tx-macro-enum-fix-v2-2-6d091c736116@dawidwrobel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-08-01 01:01:59 +01:00
Dawid Wróbel
1ba381759e ASoC: codecs: lpass-tx-macro: Fix enum kcontrol accesses
The "DEC0 MODE" to "DEC7 MODE" controls are enumerated, but
tx_macro_dec_mode_get() and tx_macro_dec_mode_put() access their
value through ucontrol->value.integer.value[0] (a long) instead of
ucontrol->value.enumerated.item[0] (an unsigned int).

This same pattern was fixed in the sibling drivers by
commit bcfe5f76cc ("ASoC: codecs: rx-macro: fix accessing array
out of bounds for enum type") and
commit 0ea5eff7c6 ("ASoC: codecs: va-macro: fix accessing array
out of bounds for enum type"), but tx-macro was missed.

On 64-bit kernels built with CONFIG_SND_CTL_DEBUG, the elem value
sanity check catches the 4 bytes written past the enumerated item
and every read of these controls fails with -EINVAL:

  snd-sm8250 sound: control 2:0:0:DEC0 MODE:0: access overflow

Fixes: c39667ddcf ("ASoC: codecs: lpass-tx-macro: add support for lpass tx macro")
Assisted-by: Claude:claude-fable-5
Cc: stable@vger.kernel.org
Signed-off-by: Dawid Wróbel <me@dawidwrobel.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260730-worktree-lpass-tx-macro-enum-fix-v2-1-6d091c736116@dawidwrobel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-08-01 01:01:58 +01:00
Peter Ujfalusi
17661c67b2 ASoC: SOF: ipc4-pcm: Continue the pipeline trigger in case of IPC timeout
Ignore IPC errors for pipeline state change if the firmware state is
crashed or the IPC has timed out.

If the firmware has crashed the kernel still needs to go through the state
changes to reset its internal to be able to correctly work the next time
the DSP is booted up.

The case with IPC timeout is a bit more problematic, but it has been
rootcaused to be the result of system scheduling blockage and the firmware
did actually received and handled the message, but the reply handling got
blocked by issues outside of the SOF stack.
So far the best way to handle this is to continue with setting the state.

Fixes: c40aad7c81 ("ASoC: SOF: ipc4-pcm: Workaround for crashed firmware on system suspend")
Cc: stable@vger.kernel.org
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://patch.msgid.link/20260730112343.26687-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-31 16:14:11 +01:00
Zhang Heng
38417f5fc8 ASoC: amd: yc: Add DMI quirk for HP Victus Laptop 16-e1xxx
Add DMI quirk to enable ACP6x sound card for HP Victus by HP Laptop
16-e1xxx, which fixes microphone not working issue.

Without this quirk, the DMIC on acp6x device is not properly enabled,
causing the microphone to not work.

Link: https://bugzilla.kernel.org/show_bug.cgi?id=218926
Link: b29ba27a0e
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Reported-by: CuerdOS Dev Team <cuerdoslinux@proton.me>
Signed-off-by: CuerdOS Dev Team <cuerdoslinux@proton.me>
Link: https://patch.msgid.link/20260730123920.104525-1-zhangheng@kylinos.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-31 15:55:58 +01:00
Bard Liao
992eb0dfbc ASoC/soundwire: Intel: reset the PCMSyCM registers in hda_sdw_bpt_close
Resetting the PCMSyCM registers is required for Intel SoundWire stream. The
same procedure is done in sdw_hda_dai_hw_params() for the normal
SoundWire stream, too.

Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Acked-by: Vinod Koul <vkoul@kernel.org>
Link: https://patch.msgid.link/20260730012518.2180906-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-31 13:23:10 +01:00
Peter Ujfalusi
e780e4917d ASoC: SOF: sof-audio: Fix error path in sof_widget_setup_unlocked()
If either tplg_ops->dai_config or widget_kcontrol_setup fail during widget
setup we would double decrement the use_count of the widget because the
sof_widget_free_unlocked() would be called twice, similarly the core_put
would be invoked twice as well.

Since the use_count and core_put() is handled within the widget_free
function we need to return without falling through the pipe_widget_free
label.

The fixes tag is picked to the last change around this part of the code
which is adequately old enough for backporting purposes.

Link: https://github.com/thesofproject/sof/issues/10826
Fixes: 31ed8da1c8 ("ASoC: SOF: sof-audio: Modify logic for enabling/disabling topology cores")
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>
Link: https://patch.msgid.link/20260730085914.27546-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-31 00:08:07 +01:00
Bard Liao
ae63720dd7 ASoC: SOF: topology: Use acpi mach from the machine driver
The parameters may be changed by the sof_sdw machine driver is the
SOC_SDW_PCH_DMIC quirk is set. Use the mach_params from the machine
driver to ensure the sof_sdw_get_tplg_files() function select the right
function topologies.

Fixes: 2fbeff3338 ("ASoC: Intel: add sof_sdw_get_tplg_files ops")
Cc: stable@vger.kernel.org
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://patch.msgid.link/20260730071724.22296-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-30 23:19:52 +01:00
Peter Ujfalusi
0c0e418dbc ASoC: SOF: ipc4-topology: Refresh copier IPC payload before widget setup
The ipc_config_data buffer for copier widgets is built once during
ipc_prepare (called from sof_pcm_setup_connected_widgets) and cached
for reuse. For host copiers this buffer contains the copier_data with
gtw_cfg.node_id (host DMA ID). For DAI copiers it additionally includes
a dma_config_tlv trailer with stream_id and dma_channel_id for HDA link
DMA.

On suspend/resume, both host and link DMA streams are released and
re-allocated with potentially different stream tags. The underlying
copier_data and dma_config_tlv structures are correctly updated by
host_config and sdw_hda_dai_hw_params respectively. However, since the
widget list (spcm->stream[].list) persists across suspend,
sof_pcm_hw_params skips sof_pcm_setup_connected_widgets and ipc_prepare
never runs again to rebuild ipc_config_data. The stale cached payload
is then sent to firmware with boot-time DMA channel assignments, causing
DMA channel conflicts that lead to firmware errors and crashes.

Fix this by refreshing copier_data and dma_config_tlv portions of
ipc_config_data in sof_ipc4_widget_setup right before the IPC message
is sent. This ensures the payload always reflects the current DMA state
regardless of whether ipc_prepare ran.

For DAI copiers, the gtw_cfg.config_length in copier_data is temporarily
inflated to include the TLV size (matching the ipc_config_data layout)
before copying, then restored, mirroring what
sof_ipc4_prepare_copier_module does when first building the buffer.

Fixes: e9c6b118de ("ASoC: SOF: make dma_config_tlv be an array")
Cc: stable@vger.kernel.org
Link: https://github.com/thesofproject/sof/issues/10700
Link: https://github.com/thesofproject/sof/issues/10955
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Link: https://patch.msgid.link/20260730054822.5913-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-30 20:14:29 +01:00
Takashi Iwai
2b6e56b848 Merge tag 'asoc-fix-v7.2-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v7.2

This is a relatively large set of updates, the biggest batch of things
is Charles' fixes for the SDCA code which have been through a number of
iterations on the list and deal with a bunch of issues that have been
seen as we get more real world usage of SDCA.  We also have the usual
device specific fix and quirk traffic that we tend to see, there's a
small pile of fixes for the tas2562 driver since I saw some bugs while
reviewing fixes sent by Haidar Lee but it's nothing too remarkable.
2026-07-30 14:02:28 +02:00
Shuming Fan
d0ecbedd6a ASoC: rt722: reset codec to fix abnormal sound
The audio output may become abnormal after a warm reboot from Windows.
Reset the codec once during hardware initialization to restore it to a known
state and prevent the issue.
BugLink: https://github.com/thesofproject/linux/issues/5845

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://patch.msgid.link/20260728090111.3676617-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-29 23:27:37 +01:00
André Pragosa
630c45e92d ALSA: hda/realtek: Add mute LED quirk for HP Victus 16-e0xxx (MB 88ED)
Add subsystem ID 103c:88ed to the existing HP Victus 16-e0xxx
mute LED quirk list.

The HP Victus 16-e0xxx with subsystem ID 103c:88ed uses the same
mute LED coefficient configuration as the already supported
103c:88eb variant.

The mute LED was verified by manually toggling coefficient index
0x0b (bit 3) using hda-verb. After adding the quirk, the LED is
registered as hda::mute and follows the audio mute state.

Signed-off-by: André Pragosa <pragosa512@gmail.com>
Link: https://patch.msgid.link/20260728221129.14680-2-pragosa512@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-07-29 08:32:12 +02:00
Mikhail Gavrilov
e93bb6f3ac ALSA: usb-audio: Add GET_SAMPLE_RATE quirk for C-Media CM6206
The C-Media CM6206 (0d8c:0102) truncates the three-byte sample rate it
returns for UAC_GET_CUR to its two low bytes.  After the rate has been
set to 96000 (0x017700) the device reports back 30464 (0x007700).

At probe time the driver initializes every altsetting to its maximum
rate, so altsetting 5 is set to 96000 and the warning appears on each
plug-in, before anything has opened the device:

  usb 3-1.3: 1:5 Set sample rate 96000, clock 0
  usb 3-1.3: current rate 30464 is different from the runtime rate 96000

That altsetting is the one parse_audio_format_rates_v1() already fixes
up for this chip, so this affects every CM6206.

Only the read-back is broken, the rate itself is applied: a 1 kHz sine
rendered at 96 kHz is recovered at 1000.2 Hz, and a silent fallback to
48000 would have been reported as 0x00bb80 rather than as the low half
of the requested rate.

Add a QUIRK_FLAG_GET_SAMPLE_RATE entry for the device so the read-back
is skipped.  Setting the same flag through the quirk_flags module
parameter makes the warning disappear while the 96000 init still
happens.

Signed-off-by: Mikhail Gavrilov <mikhail.v.gavrilov@gmail.com>
Link: https://patch.msgid.link/20260728222239.62749-1-mikhail.v.gavrilov@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-07-29 08:31:42 +02:00
Sonali Pradhan
8d7a30c50c ALSA: usb-audio: Clamp frame size in implicit-feedback mode
snd_usb_handle_sync_urb() scales received sync packet sizes by the sender's
stride and stores the result directly in out_packet->packet_size[i]. If a
connected USB device sends an oversized sync packet, this frame count can
exceed ep->maxframesize.

The un-clamped frame count then propagates to the playback endpoint queue,
potentially driving packet transfers beyond the endpoint's hardware frame
limits.

Cap the calculated frame count against ep->maxframesize in
snd_usb_handle_sync_urb() to prevent oversized packets from entering the
playback queue.

Fixes: 28acb12014 ("ALSA: usb-audio: use sender stride for implicit feedback")
Cc: stable@vger.kernel.org
Assisted-by: Jetski:Gemini-3.6-Flash
Signed-off-by: Sonali Pradhan <sonalipradhan@google.com>
Link: https://patch.msgid.link/20260728202432.2354994-1-sonalipradhan@google.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-07-29 08:25:27 +02:00
Sonali Pradhan
d0199ae166 ALSA: usb-audio: Fix DMA buffer out-of-bounds write when fill_max is set
When a USB audio endpoint requests full packet transfers via the fill_max
descriptor flag, data_ep_set_params() promotes ep->curpacksize to
ep->maxpacksize. However, maxsize is left at the original sample-rate
derived value.

Since u->buffer_size is allocated as maxsize * packets, the resulting
DMA buffer is far too small for the requested transfer length. When the
USB host controller streams up to curpacksize bytes per packet, it writes
past the end of the buffer via DMA, corrupting kernel heap memory.

Update maxsize to curpacksize when fill_max is set so that the allocated
DMA buffer size matches the actual transfer request size.

[ changed to reassign maxsize only when ep->fill_max is set -- tiwai ]

Fixes: 8fdff6a319 ("ALSA: snd-usb: implement new endpoint streaming model")
Cc: stable@vger.kernel.org
Assisted-by: Jetski:Gemini-3.6-Flash
Signed-off-by: Sonali Pradhan <sonalipradhan@google.com>
Link: https://patch.msgid.link/20260728201716.2347726-1-sonalipradhan@google.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-07-29 08:23:32 +02:00
Robert Abrahamse
cee0466796 ALSA: usb-audio: Add quirk for Corsair Virtuoso (later revision)
Add USB mixer mapping quirk for later revisions of the Corsair Virtuoso
headset with USB IDs 0x1b1c:0x0a43 (wired) and 0x1b1c:0x0a44
(wireless). These devices exhibit the same mixer label collision as
earlier Virtuoso variants: all controls are labelled "Headset", causing
applications like PulseAudio to move the sidetone control instead of
the main playback volume.

Signed-off-by: Robert Abrahamse <denobyte2@gmail.com>
Link: https://patch.msgid.link/20260728140314.11601-1-denobyte2@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-07-28 18:30:43 +02:00
Norbert Szetei
f495b6c4c8 ALSA: pcm: wake linked drain waiters on unlink
snd_pcm_drain() on a linked stream parks an on-stack wait entry on the
drained peer's runtime->sleep, and after schedule_timeout() removes it
only if that peer is still found in the caller's group.  If group
membership changes during the wait and the sleep ends by signal or
timeout (so autoremove_wake_function() does not run), finish_wait() is
skipped and snd_pcm_drain() returns with the entry still queued on that
stream's sleep list; a later wake_up() then walks a freed stack frame.
This is reachable by unlinking either the drained or the draining stream.

Unlike the close path (snd_pcm_drop() -> snd_pcm_post_stop()),
snd_pcm_unlink() never wakes the sleep queues.  Wake every group member
under the group lock before the membership change, so a linked drainer is
released and drops its entry while the streams are still grouped.

The window was opened when snd_pcm_link_rwsem stopped being held across
the wait and the removal became conditional on group membership (see
Fixes). The later switch to finish_wait() kept that conditional removal,
so the signal/timeout case remained.

Fixes: f57f3df03a ("ALSA: pcm: More fine-grained PCM link locking")
Cc: stable@vger.kernel.org
Assisted-by: Claude:claude-opus-5
Signed-off-by: Norbert Szetei <norbert@doyensec.com>
Link: https://patch.msgid.link/A0705100-D10B-4286-9980-0142ABEEAD51@doyensec.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-07-28 18:29:48 +02:00
Jackie Dong
9ad4c2331d ASoC: amd: acp: Add DMI quirk for Lenovo Legion 7 15ASH11
Lenovo Legion 7 15ASH11 with AMD RYZEN AI MAX+ 392 (Strix Halo, ACP
7.0) uses Realtek ALC287 series codec and no any DMIC connected by ACP.
All DMICs directly connet with ALC codec.

Without this quirk, Input Device of Gnome Sound settings shows Internal
Stereo Microphone and Digital Microphone by default. In fact, Digital
Microphone of ACP doesn't work due to no connecting with ALC287 codec,
the Internal Stereo Microphone as analog device based on snd_hda_intel
driver can work well.

Add a DMI quirk to override the flag to 0, consistent with the existing
entry for the Lenovo Yoga Pro 7 15ASH11.

Signed-off-by: Jackie Dong <xy-jackie@139.com>
Link: https://patch.msgid.link/20260727041452.16701-1-xy-jackie@139.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-28 00:14:01 +01:00
Surendra Singh Chouhan
f47064c970 ASoC: sophgo: return 1 on volume change in cv1800b_adc_volume_set()
cv1800b_adc_volume_set() serves as the .put callback for the "Internal
I2S Capture Volume" control.

ALSA mixer control callbacks must return 1 when the register value is
modified, 0 if unchanged, or a negative error code on failure. Returning
0 unconditionally causes ALSA core to assume the value was unchanged,
suppressing SNDRV_CTL_EVENT_MASK_VALUE change notifications to userspace
sound servers (e.g. PipeWire/PulseAudio).

Fix this by comparing the new register value with the existing register
value. If unchanged, return 0; otherwise, write the updated value and
return 1.

Fixes: 4cf8752a03 ("ASoC: sophgo: add CV1800B internal ADC codec driver")
Signed-off-by: Surendra Singh Chouhan <kr494167@gmail.com>
Link: https://patch.msgid.link/20260727053804.25599-1-kr494167@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-27 23:56:58 +01:00
wangdicheng
dd88cf6273 ASoC: tas2781: Use correct calibration data for SINEGAIN2 register
The SINEGAIN2_REG case in cali_reg_update() references t->sin_gn[]
rather than t->sin_gn2[], causing the second pilot tone gain
calibration to be programmed with the wrong register address.

These are distinct fields in struct fct_param_address and are
populated from separate firmware parameters by the parser in
tas2781-fmwlib.c.

Fixes: 84d6a465f2 ("ASoC: tas2781: Support dsp firmware Alpha and Beta seaies")
Signed-off-by: wangdicheng <wangdicheng@kylinos.cn>
Link: https://patch.msgid.link/20260720081616.631413-1-wangdich9700@163.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-27 18:48:21 +01:00
Mark Brown
fc810085f6 ASoC: Fix races on creation of SDCA jack detection
Charles Keepax <ckeepax@opensource.cirrus.com> says:

Currently there exists a couple races that can result in the DAPM graph
coming up in a state that doesn't match the hardware with respect to
SDCA jack detection. This series fixes these up by adding a component
level fixup_controls helper into the asoc core and shuffling around the
IRQ requests from the SDCA side.

The core creates DAPM widgets/routes quite a long time before
it creates the associated ALSA control, and the jack detection
IRQ is currently registered in component probe. At the time of
component probe, the DAPM widgets exist, shortly after this the
DAPM routes are added. At the time the DAPM routes are added the
register value for the control is checked and the appropriate path
is connected. The existing handling in the SDCA jack IRQ handles
the case the control doesn't exist and updates the registers
directly, which works until the DAPM routes are added.  After the
routes are added the DAPM graph has already set connected on a
particular DAPM path, which will not be updated until an IRQ is
received when the control is present. Thus those updates are
usually not reflected in the resulting DAPM graph which can lead
to the audio path being erroneously powered on/off.

Link: https://patch.msgid.link/20260721143636.361814-1-ckeepax@opensource.cirrus.com
2026-07-27 18:47:11 +01:00
Charles Keepax
bf1b7821f8 ASoC: SDCA: Move kcontrol search out of IRQ
Now that the IRQs are always registered after all the ALSA
controls are created it is possible to search for the control
at the point the IRQ is requested. Move the control search out
of the IRQ handler and do it at IRQ request time.

This also fixes a potential issue when the card was torn down
and reprobed without destroying the codec device, the kctl
pointer stored by the IRQ handler would not be updated to the
new control on the second probe.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260721143636.361814-8-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-27 18:47:09 +01:00
Charles Keepax
b8f71f1613 ASoC: SDCA: Switch to fixup_controls callback for IRQ registration
Currently there are some race conditions around the boot of SDCA
jack detection. The core creates DAPM widgets/routes quite a
long time before it creates the associated ALSA control, and
the jack detection IRQ is currently registered in component
probe. At the time of component probe, the DAPM widgets exist,
shortly after this the DAPM routes are added. At the time the DAPM
routes are added the register value for the control is checked
and the appropriate path is connected. The existing handling
in the SDCA jack IRQ handles the case the control doesn't exist
and updates the registers directly, which works until the DAPM
routes are added.  After the routes are added the DAPM graph has
already set connected on a particular DAPM path, which will not
be updated until an IRQ is received when the control is present.
Thus those updates are usually not reflected in the resulting
DAPM graph which can lead to the audio path being erroneously
powered on/off.

Switch to the new fixup_controls callback to register the
IRQs, this is guaranteed to run after all the controls have
been created. Which means we can avoid the aforementioned race
condition and as a bonus no longer need to concern ourselves
with a case where the IRQ handler runs and the ALSA control
is unavailable.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260721143636.361814-7-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-27 18:47:08 +01:00
Charles Keepax
3e81e2fb21 ASoC: Add a component fixup_controls callback
A card level fixup_controls callback was added in:

commit df4d27b19b ("ASoC: Introduce 'fixup_controls' card method")

This allowed the machine driver to take actions after all the
card controls have been added. However, there are times when a
codec driver would also want to do things like obtain references
to controls for later use, which require all the controls to be
present. Add a component level fixup_controls callback, echoing
the card level option.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260721143636.361814-6-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-27 18:47:07 +01:00
Charles Keepax
050406cbd6 ASoC: SDCA: Populate IRQ data earlier
Currently, the IRQ data (attached Entity/Control/etc) is populated
as the IRQ is requested. However, this can cause issues as
occasionally the setup process wants to access specifics of
an IRQ before the IRQ is actually enabled. To facilitate this
cache all the IRQ data during sdca_irq_populate_early() and make
sdca_irq_populate() simply request the outstanding IRQs. This
also has the advantage that sdca_irq_populate() can now just
iterate through the IRQ array which is much smaller/faster than
going through every Entity in the Function for Controls.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260721143636.361814-5-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-27 18:47:06 +01:00
Charles Keepax
0880082c27 ASoC: SDCA: Remove devm from primary IRQ cleanup
To provide greater flexibility on when the IRQs are requested for
client drivers don't use devm for the primary IRQ request/cleanup
helper functions.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260721143636.361814-4-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-27 18:47:05 +01:00
Charles Keepax
f18e97fa7f ASoC: SDCA: Add sdca_irq_cleanup_late()
The SDCA IRQs are split into two groups, those registered at bus probe
time (basically just FDL) and those registered at component time.
There currently exists only a single cleanup function, if the FDL IRQ
is freed at component time, then nothing would re-register it if the
component is probed again. But the IRQs depending on a component need
to be freed if the card is destroyed so they can't use stale
components.

Split the clean up into two functions one for the component level and
one for the bus level.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260721143636.361814-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-27 18:47:04 +01:00
Charles Keepax
4ecef577d1 ASoC: SDCA: Rename sdca_irq_allocate() to include devm
Make it more clear sdca_irq_allocate() uses devm allocations by adding
it into the name.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260721143636.361814-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2026-07-27 18:47:03 +01:00
Michael Diesen
bed0c80840 ALSA: hda/realtek: Add quirk for HP Dragonfly Folio G3 2-in-1 (103c:8a05)
The HP Dragonfly Folio G3 2-in-1 also ships with PCI SSID 103c:8a05.
On this unit the ALC245 codec reports subsystem id 103c:8a06 - the SSID
that is already covered by commit 0a10faad5c ("ALSA: hda/realtek: add
quirk for HP Dragonfly Folio G3 2-in-1") - while the PCI SSID that
SND_PCI_QUIRK matches against is 103c:8a05:

  snd_hda_codec_alc269 ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:8a05
  cs35l41-hda spi1-CSC3551:00-cs35l41-hda.0: CS35L41 Bound - SSID: 103C8A06

The existing entry therefore never applies here, the four CS35L41
amplifiers on SPI are not registered and the internal speakers stay
silent.

Add the same fixup that the 8a06 entry uses: the four amplifiers bind
and the speaker mute LED (codec GPIO 0x04) works.

Signed-off-by: Michael Diesen <michael.diesen@posteo.de>
Link: https://patch.msgid.link/20260727091920.4634-1-michael.diesen@posteo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-07-27 15:54:17 +02:00
Jackie Dong
27460bd80b ALSA:hda/realtek:ALC269 fixup for Legion 7 15ASH11 Mic Mute LED
Lenovo Legion 7 15ASH11 with AMD RYZEN AI MAX+ 392 (Strix Halo, ACP
7.0) uses Realtek ALC287 series codec. Its audio subsystem adopts a
hardware design similar to that of the Yoga Pro 7 15ASH11.

It shares PCI SSID 17aa:38f9 with Thinkbook 16P Gen5.
Therefore, use HDA_CODEC_QUIRK to apply ALC287_FIXUP_LENOVO_YOGA_PRO7
for identification.

After added the HDA_CODEC_QUIRK quirk special for Lenovo Legion 7
15ASH11, the mic mute LED works well.

Signed-off-by: Jackie Dong <xy-jackie@139.com>
Link: https://patch.msgid.link/20260727080048.13254-1-xy-jackie@139.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-07-27 15:53:21 +02:00
Norbert Szetei
c2744d5f3a ALSA: timer: Clear SNDRV_TIMER_IFLG_DEAD once the close completes
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>
2026-07-26 16:11:45 +02:00
Gunal Seenivasagan
fd71820e87 ALSA: hda/realtek: Add quirk for Infinix INBOOK X3 Slim
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>
2026-07-26 15:55:52 +02:00
Takashi Iwai
a54bf16965 ALSA: 6fire: Fix UAF at error handling during probe
Although 6fire driver had a few fixes for dealing with the early error
handling during the probe phase, it forgot a pending URB before
freeing the resources, which may lead to a UAF.

This patch addresses it by doing the almost same cleanup procedure
like the normal disconnect phase at the error path.

Reported-and-tested-by: Shuangpeng Bai <shuangpeng.kernel@gmail.com>
Closes: https://lore.kernel.org/20260724030900.1984491-1-shuangpeng.kernel@gmail.com
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20260726074821.2288158-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2026-07-26 10:02:01 +02:00
Baul Lee
0970274613 ALSA: usb-audio: fix OOB write in snd_usbmidi_akai_output()
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>
2026-07-26 09:49:26 +02:00
Baul Lee
441aaad150 ALSA: usb-audio: fix stack info leak in RME Digiface status
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>
2026-07-26 09:05:59 +02:00