The dummy sequencer port forwards events by copying an incoming
struct snd_seq_event into a stack temporary, rewriting source and
destination, and dispatching the temporary to subscribers. That legacy
event storage is smaller than struct snd_seq_ump_event.
When a UMP event reaches the dummy client, the copy leaves the UMP flag
set but only provides legacy-sized stack storage. The subscriber
delivery path then uses snd_seq_event_packet_size() and copies a
UMP-sized packet from that stack object, reading past the end of the
temporary.
Use the existing union __snd_seq_event storage and copy the packet size
reported for the incoming event before rewriting the common routing
fields. This preserves the full UMP packet for UMP events while keeping
legacy event handling unchanged.
Fixes: 32cb23a0f9 ("ALSA: seq: dummy: Allow UMP conversion")
Signed-off-by: Kyle Zeng <kylebot@openai.com>
Link: https://patch.msgid.link/20260605080204.32045-1-kylebot@openai.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Setting up the interface when suspended/resumeing fail on this card.
Adding a reset and delay quirk will eliminate this problem.
usb 1-1: new full-speed USB device number 2 using xhci-hcd
usb 1-1: New USB device found, idVendor=3c20, idProduct=3d21
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 1-1: Product: AB13X USB Audio
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/TYUPR06MB62174610061C213260E1A992D2102@TYUPR06MB6217.apcprd06.prod.outlook.com
snd_pcm_drain() uses init_waitqueue_entry which does not clear
entry.prev/next, and add_wait_queue with a conditional
remove_wait_queue that is skipped when to_check is no longer
in the group after concurrent UNLINK. The orphaned wait entry
remains on the unlinked substream sleep queue. On the next
drain iteration, add_wait_queue adds the entry to a new queue
while still linked on the old one, corrupting both lists. A
subsequent wake_up dereferences NULL at the func pointer
(mapped from the spinlock at offset 0 of the misinterpreted
wait_queue_head_t), causing a kernel panic.
Replace init_waitqueue_entry/add_wait_queue/conditional
remove_wait_queue with init_wait_entry/prepare_to_wait/
finish_wait. init_wait_entry clears prev/next via
INIT_LIST_HEAD on each iteration and sets
autoremove_wake_function which auto-removes the entry on
wake-up. finish_wait safely handles both the already-removed
and still-queued cases.
Fixes: 9b1dbd69ba ("ALSA: pcm: fix use-after-free on linked stream runtime in snd_pcm_drain")
Signed-off-by: Ji'an Zhou <eilaimemedsnaimel@gmail.com>
Link: https://patch.msgid.link/20260604142559.3840881-1-eilaimemedsnaimel@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: Fixes for v7.1
There's only one actual fix here, for the TDM configuration on the
Freescale SAI controller, everytihng else is DMI quirks for AMD systems.
One of those is relatively large as it adds a bunch of different structs
but it's all data.
The ASUS Vivobook 18 M1807GA (AMD ACP7.x, PCI 1022:15e2, subsystem
1043:3531) exposes a single Realtek RT721 SDCA codec on SoundWire link 1.
The BIOS reports the ACP audio config flag as 0 (SoundWire mode), so
snd_pci_ps claims the device, brings up the SoundWire managers and
enumerates the RT721 peripheral (sdw:0:1:025d:0721:01); the rt721-sdca
codec driver binds successfully.
No sound card is created, however: acp63_sdw_machine_select() walks
snd_soc_acpi_amd_acp70_sdw_machines[] and finds no entry whose declared
SoundWire peripherals are all present on the bus. The only existing RT721
entry, acp70_rt721_l1u0_tas2783x2_l1u8b, additionally requires two
TAS2783 amplifiers and deliberately exposes the RT721 as jack + DMIC
only. This M1807GA variant has no external amplifiers - the RT721's
internal AIF2 amplifier path drives the speakers - so that entry never
matches and no machine device is registered.
Add a standalone RT721 machine entry for link 1 exposing all three RT721
endpoints (jack/AIF1, speaker amplifier/AIF2, DMIC/AIF3), mirroring the
standalone RT722 configuration. Place it after the TAS2783 combo entry so
platforms that do have the external amplifiers continue to match the more
specific entry first.
ACPI _ADR of the codec: 0x000130025D072101
(link_id=1 version=3 mfg_id=0x025d Realtek part_id=0x0721 class=0x01).
Verified on the hardware: with the entry present the amd_sdw machine
binds, an "amd-soundwire" card is registered exposing the rt721-sdca
AIF1 (SimpleJack) and AIF2 (SmartAmp) PCM devices, and audio plays out
of the built-in speakers.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=221282
Signed-off-by: Aleksander Pshenitsyn <brains.fatman@gmail.com>
Link: https://patch.msgid.link/20260531101159.14241-1-brains.fatman@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The MSI Raider A18 HX A9WJG has an internal digital microphone
connected through AMD ACP6x, but this machine does not expose the
AcpDmicConnected ACPI property, so acp_yc_mach does not bind.
Add a DMI quirk for this model.
This was tested on an MSI Raider A18 HX A9WJG with board MS-182L,
BIOS E182LAMS.31A, AMD ACP6x rev 0x62, and Realtek ALC274. After
applying the quirk, the internal microphone appears as an acp6x DMIC
capture device and records correctly.
Signed-off-by: David Glushkov <david.glushkov@sntiq.com>
Link: https://patch.msgid.link/20260531214512.170716-1-david.glushkov@sntiq.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When configuring 32 slots TDM (channels == slots == 32), the xMR
(Mask Register) write used:
~0UL - ((1 << min(channels, slots)) - 1)
The literal "1" is a signed 32-bit int. Shifting it by 32 positions is
undefined behaviour which may set this register to 0xFFFFFFFF, masking
all 32 slots.
Use GENMASK_U32() macro instead. For 32 slots this produces a zero mask:
~GENMASK_U32(31, 0) = ~0xFFFFFFFF = 0x00000000
Behaviour for fewer than 32 slots is unchanged.
Fixes: 770f58d7d2 ("ASoC: fsl_sai: Support multiple data channel enable bits")
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/20260601083327.1535185-1-chancel.liu@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The MSI Bravo 17 C7VF routes its internal digital microphone through
the ACP6x. The machine driver only enables the DMIC for boards present
in the DMI quirk table, so on this model the internal mic is never
detected and no capture device is created.
Add a quirk entry matching the board's DMI identifiers so the DMIC is
enabled and the internal microphone works.
Signed-off-by: João Miguel <jmiguel.ghp@gmail.com>
Link: https://patch.msgid.link/20260523213548.5219-1-jmiguel.ghp@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Lenovo Yoga Pro 7 15ASH11 with AMD RYZEN AI MAX+ 388 (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 HN7306EA.
Signed-off-by: Jackie Dong <xy-jackie@139.com>
Link: https://patch.msgid.link/20260527102005.58528-1-xy-jackie@139.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It makes no sense to restore the saved value for a sticky mixer, since
setting any value is a no-op.
However, in some rare cases, SET_CUR is effective despite GET_CUR always
returns a constant value. These mixers are not sticky, but there's no
way to distinguish them. Without any additional information, the best
thing we can do is to set the mixer value to the maximum before bailing
out, so that a soft mixer can still reach the maximum hardware volume if
the mixer turns out to be non-sticky. Meanwhile, all channels must be
synchronized to prevent imbalance volume.
Fixes: 86aa1ea1f1 ("ALSA: usb-audio: Do not expose sticky mixers")
Signed-off-by: Rong Zhang <i@rong.moe>
Link: https://patch.msgid.link/20260531-uac-sticky-error-path-v1-1-12c2329d17ef@rong.moe
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: Fixes for v7.1
This round of fixes is mostly Sirini's Qualcomm cleanups that have been
in review for a while, we also have a couple of small fixes from Cássio.
simple_mux_control_put() rejects values greater than e->items, but
enum control values are zero based. For the two-entry mux used by this
driver, valid values are 0 and 1, so value 2 must be rejected as well.
Accepting e->items can store an invalid mux state, pass it to the GPIO
setter, and pass it on to the DAPM mux update path where it is used as
an index into the enum text array.
Use the same >= e->items check used by the ASoC enum helpers.
Fixes: 342fbb7578 ("ASoC: add simple-mux")
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260527-asoc-simple-mux-enum-bounds-v1-1-3f805b9fc671@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Setting up the interface when suspended/resumeing fail on this card.
Adding a reset and delay quirk will eliminate this problem.
usb 1-1: new full-speed USB device number 2 using xhci-hcd
usb 1-1: New USB device found, idVendor=25aa, idProduct=600b
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 1-1: Product: TAE1159
usb 1-1: Manufacturer: Generic
usb 1-1: SerialNumber: 20210726905926
Signed-off-by: Lianqin Hu <hulianqin@vivo.com>
Link: https://patch.msgid.link/TYUPR06MB621736D7C85D43200E54E740D2082@TYUPR06MB6217.apcprd06.prod.outlook.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 21.5" Retina 4K iMac (Late 2015, DMI product name "iMac16,1") ships
with a Cirrus Logic CS4208 codec wired to an external speaker amplifier
enabled through codec GPIO0 -- the same arrangement as the late-2013
MacBookPro 11,x. Without a matching entry in cs4208_mac_fixup_tbl[] the
fixup picker logs:
snd_hda_codec_cs420x hdaudioC1D0: CS4208: picked fixup for codec SSID 106b:0000
i.e. an empty fixup name, GPIO0 stays low, the external amp is never
powered up, and the internal speakers are silent on a stock kernel.
The codec SSID reported by hardware is 0x106b:0x7f00. Reusing CS4208_MBP11
(GPIO0 + SPDIF switch fixup) makes the internal speakers and S/PDIF
output work out of the box, removing the need for users to set
`options snd_hda_intel model=mbp11` via /etc/modprobe.d/.
Tested on iMac16,1 (kernel 6.17.0): four internal drivers
(Left tweeter, Left woofer, Right tweeter, Right woofer, exposed as the
4 channels of the analog-surround-40 ALSA profile) produce audio after
the fixup is applied.
Signed-off-by: Jakub Pisarczyk <pisarz77@gmail.com>
Link: https://patch.msgid.link/20260526201830.34097-1-pisarz77@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> says:
Here is the set of patches, that fixes one of the isssue reported by
Richard Acayan, while doing fix for the reported issue, found various
other issues in the existing code.
This set contains some of those cleanups along with few trivial coding
style patches which looked uncomfortable to read.
Patch 1 should be enough to fix the issue reported.
Tested this is on UNO-Q.
Link: https://patch.msgid.link/20260518092347.3446946-1-srinivas.kandagatla@oss.qualcomm.com
q6asm_dai_close() and q6asm_dai_compr_free() currently issue CMD_CLOSE
whenever prtd->state is non-zero.
After prepare() closes an existing stream, the state is updated to
Q6ASM_STREAM_STOPPED. Since this state is also non-zero, the close and
free paths can send CMD_CLOSE again for a stream that has already been
closed.
Restrict CMD_CLOSE to the Q6ASM_STREAM_RUNNING state so the command is
sent only when the ASM stream is still active.
Fixes: 2a9e92d371 ("ASoC: qdsp6: q6asm: Add q6asm dai driver")
Cc: Stable@vger.kernel.org
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260518092347.3446946-3-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The q6asm-dai stream state is used by prepare() to decide whether an
existing stream setup needs to be closed before opening/configuring a new
one. Updating the state from trigger or asynchronous DSP callbacks can make
that state stale or incorrect relative to the actual setup lifetime.
In particular, setting Q6ASM_STREAM_STOPPED on STOP or EOS completion can
make prepare() believe there is no active setup to close, which can result
in opening/configuring the same stream more than once.
Keep stream state updates tied to prepare(), where the stream is actually
closed and reopened, and stop changing it from trigger and EOS callbacks.
Fixes: bfbb12dfa1 ("ASoC: qcom: q6asm-dai: perform correct state check before closing")
Cc: Stable@vger.kernel.org
Closes: https://lore.kernel.org/all/afS7rTHdc9TyIeLx@rdacayan/
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260518092347.3446946-2-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
byt_cht_es8316_init() enables MCLK before configuring the codec sysclk
and creating the headset jack. If either of those later steps fails, the
function returns without disabling MCLK, leaving the clock enabled after
card registration fails.
Track whether this driver enabled MCLK and disable it on the init error
paths. Add the matching DAI link exit callback so the same clock enable
is also balanced when ASoC cleans up a successfully initialized link.
Fixes: a03bdaa565 ("ASoC: Intel: add machine driver for BYT/CHT + ES8316")
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260519-asoc-bytcht-es8316-mclk-leak-v1-1-b4a11cdc2afd@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Firmware 2417 for the Scarlett 4th Gen 2i2 moved the direct monitor
gain parameter by 4 bytes, from offset 0x2a0 to 0x2a4, breaking the
"Direct Monitor X Mix Y" controls.
Special-case the offset in the get/set config helpers when the
running firmware is 2417 or later.
Fixes: 4e809a2996 ("ALSA: scarlett2: Add support for Solo, 2i2, and 4i4 Gen 4")
Cc: <stable@vger.kernel.org>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://patch.msgid.link/ahIWTueUlWA5xiV+@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
cs35l56_hda_read_acpi() gets an allocated ACPI _SUB string from
acpi_get_subsystem_id(). On success, that string is used to create the
firmware system name.
Several error paths after the _SUB lookup can return without releasing
the allocated string. This includes speaker ID lookup errors other than
-ENOENT, and errors after a firmware system name has been allocated.
Use scoped cleanup for the temporary _SUB string and make
cs35l56->system_name device-managed. This releases the temporary _SUB
string on every error path and lets devres release the firmware system
name on probe failure and device removal.
Fixes: 6f03b446cb ("ALSA: hda: cs35l56: Add support for speaker id")
Fixes: 40b1c2f9b2 ("ALSA: hda/cs35l56: Workaround bad dev-index on Lenovo Yoga Book 9i GenX")
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260522-alsa-cs35l56-system-name-leak-v4-1-a6154dd09cd9@gmail.com
The BIOS on the Lenovo Yoga Slim 7 14AGP11 (AMD Ryzen AI / Kraken
Point chassis; board LNVNB161216, product 83QS) programs the PCI
subsystem ID of the HDA function as 17aa:0000. As a result no entry
in alc269_fixup_tbl[] matches via SND_PCI_QUIRK, the fixup falls back
to the generic auto-routing path, and the bass speaker pin is left
mis-routed. Laptop speakers sound noticeably thin.
The codec's own internal subsystem ID register reports 0x17aa394c
correctly, so an HDA_CODEC_QUIRK entry (which matches on the codec
SSID rather than on the PCI SSID) binds the chassis to the existing
ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN fixup. This mirrors the same
workaround already in place for the closely-related Yoga 7 2-in-1
14AKP10 and 16AKP10 entries earlier in the table.
With this change the kernel log goes from
ALC287: picked fixup for PCI SSID 17aa:0000
to
ALC287: picked fixup alc287-yoga9-bass-spk-pin
and speaker routing matches what the firmware intended. Verified by
the reporter against the equivalent modprobe override
(model=,alc287-yoga9-bass-spk-pin).
Link: https://bugzilla.kernel.org/show_bug.cgi?id=221438
Signed-off-by: Kris Kater <kris@kater.nu>
Link: https://patch.msgid.link/20260522060902.9423-1-kris@kater.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The comment for the pin configuration 0x21 in the fixup
ALC299_FIXUP_PREDATOR_SPK states "use as headset mic, without its own
jack detect", but the fixup name and the actual usage indicate that the
pin is meant to be used as internal speaker. Correct the comment to
avoid confusion.
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260522060742.1384390-1-zhangheng@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The register DSP event queue is updated under parser->lock, but
snd_motu_register_dsp_message_parser_count_event() reads pull_pos and
push_pos without the lock.
snd_motu_register_dsp_message_parser_copy_event() also reads both queue
positions before taking the lock.
Protect these accesses with parser->lock as well. This keeps the hwdep
poll/read path consistent with the producer side and with the cached
meter/parameter accessors.
Fixes: 634ec0b290 ("ALSA: firewire-motu: notify event for parameter change in register DSP model")
Cc: stable@vger.kernel.org
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260521-alsa-firewire-motu-event-locking-v1-1-708e1c2b5e56@gmail.com
ASoC: Fixes for v7.1
A bigger batch of fixes than usual due to -next not happeing last week,
this is mostly stuff for laptops - a lot of quirks and small fixes,
mainly for x86 and SoundWire. Nothing too big or exciting individually,
just two week's worth.
For PTL onwards Cirrus are intending to rely on function topologies,
rather than using a match table for each system type. Remove this
unnecessary match table entry. Having the match entries can
mean that systems match when they should use function topologies
instead, resulting in incorrect audio configurations. Although,
admittedly this is not too likely with this 6x amp configuration
as those are quite rare, but best to follow best practice.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260520163631.3300102-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For PTL onwards Cirrus are intending to rely on function
topologies, rather than using a match table for each system
type. Chrome systems tend to have custom magic in the topology
and thus need to load a specific file. This causes problems as
these system can have the same layout as generic laptops causing
the match to apply to other laptops. Add a DMI quirk that forces
these matches to only apply to specific devices.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260520163631.3300102-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Rewrite cs_amp_create_debugfs() so that dput() will be called on
a valid dentry returned from debugfs_lookup().
The pointer returned from debugfs_lookup() must be released by dput().
The pointer returned from debugfs_create_dir() does not need to be
passed to dput().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: cdd27fa329 ("ASoC: cs-amp-lib: Add helpers for factory calibration")
Link: https://patch.msgid.link/20260521122511.987322-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When calculating data->count replace the incorrect sizeof(data) with use
of struct_offset().
The faulty sizeof(data) was incorrectly calculating the size of the
pointer instead of the size of the struct pointed to. As it happens, both
values are 8 on a 64-bit CPU. In the unlikely event of using this code on
a 32-bit CPU the number of available bytes would be calculated 4 larger
than is actually available.
Instead of changing to sizeof(*data) it has been replaced by
struct_offset() because it has better chance of detecting these sorts of
typos. Also the offset of the data[] array is actually what we want to know
here anyway.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2b62e66626 ("ASoC: cs-amp-lib: Add function to write calibration to UEFI")
Link: https://patch.msgid.link/20260521122511.987322-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use flush_work() instead of cancel_work_sync() to terminate pending IRQ
work in cs35l56_sdw_remove(). And flush_work() again after masking the
interrupts to flush any queueing that was racing with the masking. This is
the same sequence as cs35l56_sdw_system_suspend().
cs35l56_sdw_interrupt() takes the pm_runtime to prevent the bus powering-
down before the interrupt status can be read and handled. The work releases
this pm_runtime. So cancelling it, instead of flushing, could leave an
unbalanced pm_runtime.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e496112529 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Link: https://patch.msgid.link/20260521123057.988732-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
seq_ump_process_event() borrows client->out_rfile.output without
synchronizing with the first-open and last-close transition in
seq_ump_client_open() and seq_ump_client_close().
The last output unuse can therefore drop opened[STR_OUT] to zero and
release the rawmidi file while an in-flight event_input callback is still
inside snd_rawmidi_kernel_write(). That leaves the rawmidi substream
runtime exposed to teardown before the write path has taken its own
buffer reference.
Add a per-client rwlock for the event_input-visible output file. Publish
a newly opened output file under the write side, and hold the read side
from the output lookup through snd_rawmidi_kernel_write(). The last
output close copies and clears the visible output file under the write
side, then drops the lock and releases the saved rawmidi file. Use
IRQ-safe rwlock guards because event_input can also be reached from
atomic sequencer delivery.
The buggy scenario involves two paths, with each column showing the
order within that path:
path A label: event_input path path B label: last unuse path
1. seq_ump_process_event() reads 1. seq_ump_client_close()
client->out_rfile.output. drops opened[STR_OUT] to zero.
2. snd_rawmidi_kernel_write1() 2. snd_rawmidi_kernel_release()
has not yet pinned runtime. closes the output file.
3. The writer continues using 3. close_substream() frees
the borrowed substream. substream->runtime.
This keeps the output substream and runtime alive for the full
event_input write while keeping rawmidi release outside the rwlock.
KASAN reproduced this as a slab-use-after-free in
snd_rawmidi_kernel_write1(), with allocation through
seq_ump_use()/snd_seq_port_connect() and free through
seq_ump_unuse()/snd_seq_port_disconnect().
Suggested-by: Takashi Iwai <tiwai@suse.de>
Validation reproduced this kernel report:
KASAN slab-use-after-free in snd_rawmidi_kernel_write1+0x9d/0x400
RIP: 0033:0x7f5528af837f
Read of size 8
Call trace:
dump_stack_lvl+0x73/0xb0 (?:?)
print_report+0xd1/0x650 (?:?)
srso_alias_return_thunk+0x5/0xfbef5 (?:?)
__virt_addr_valid+0x1a7/0x340 (?:?)
kasan_complete_mode_report_info+0x64/0x200 (?:?)
kasan_report+0xf7/0x130 (?:?)
snd_rawmidi_kernel_write1+0x9d/0x400 (?:?)
__asan_load8+0x82/0xb0 (?:?)
update_stack_state+0x1ef/0x2d0 (?:?)
snd_rawmidi_kernel_write+0x1a/0x20 (?:?)
seq_ump_process_event+0xd4/0x120 (sound/core/seq/seq_ump_client.c:82)
__snd_seq_deliver_single_event+0x8a/0xe0 (?:?)
snd_seq_deliver_from_ump+0x2b2/0xd60 (?:?)
lock_acquire+0x14e/0x2e0 (?:?)
find_held_lock+0x31/0x90 (?:?)
snd_seq_port_use_ptr+0xa6/0xe0 (?:?)
__kasan_check_write+0x18/0x20 (?:?)
do_raw_read_unlock+0x32/0xa0 (?:?)
_raw_read_unlock+0x26/0x50 (?:?)
snd_seq_deliver_single_event+0x45c/0x4b0 (?:?)
snd_seq_deliver_event+0x10d/0x1b0 (?:?)
snd_seq_client_enqueue_event+0x192/0x240 (?:?)
snd_seq_write+0x2cd/0x450 (?:?)
apparmor_file_permission+0x20/0x30 (?:?)
security_file_permission+0x51/0x60 (?:?)
vfs_write+0x1ce/0x850 (?:?)
__fget_files+0x12b/0x220 (?:?)
lock_release+0xc8/0x2a0 (?:?)
__rcu_read_unlock+0x74/0x2d0 (?:?)
__fget_files+0x135/0x220 (?:?)
ksys_write+0x15a/0x180 (?:?)
rcu_is_watching+0x24/0x60 (?:?)
__x64_sys_write+0x46/0x60 (?:?)
x64_sys_call+0x7d/0x20d0 (?:?)
do_syscall_64+0xc1/0x360 (arch/x86/entry/syscall_64.c:87)
entry_SYSCALL_64_after_hwframe+0x77/0x7f (?:?)
Fixes: 81fd444aa3 ("ALSA: seq: Bind UMP device")
Signed-off-by: Zhang Cen <rollkingzzc@gmail.com>
Link: https://patch.msgid.link/20260520103249.3048345-1-rollkingzzc@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
scarlett2_hwdep_write() rejects writes when offset + count is greater than
or equal to the selected flash segment size. That incorrectly treats a
write ending exactly at the end of the segment as out of space, although
the last byte written is still within the segment.
Split invalid argument checks from the segment-space check, keep
zero-length writes as no-ops, and compare count against the remaining
segment size. This permits exact-end writes and avoids relying on
offset + count before deciding whether the request is in bounds.
Fixes: 1abfbd3c95 ("ALSA: scarlett2: Add support for uploading new firmware")
Cc: stable@vger.kernel.org
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260519-alsa-scarlett2-flash-write-boundary-v1-1-b550480e92da@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pending irq work might be still floating while the assigned stream
has been already closed, which may lead to UAF, especially when
another async work for fasync is involved.
For addressing this, extend the hda_controller_ops for allowing the
extra cleanup procedure that is specific to the controller driver, and
make sure to cancel and sync the pending irq work at each PCM close
before releasing the resources.
Reported-by: Jake Lamberson <lamberson.jake@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260519121157.28477-2-tiwai@suse.de
Currently, the delayed IRQ handling for PCM streams is managed in a
single work embedded in hda_intel, but this is basically a per-stream
thing. Due to the single work, we can't cancel the work properly at
closing each stream, for example.
For making the IRQ pending work to be stream-based, this patch changes
the following:
- An extended version of azx_dev (i.e. the hd-audio stream object) is
defined for snd-hda-intel
- The irq_pending flag and irq_pending_work are moved to
hda_intel_stream, so that they can be hda-intel stream specific
- The stream creation and assignment are refactored so that
snd-hda-intel can handle individually;
the snd-hda-intel specific workaround for stream tags is also moved
to snd-hda-intel itself instead of the common code
- The irq pending work is canceled properly at free / shutdown
While we're at it, changed the bit field flag to bool, as the bit
field doesn't help much in our case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260519121157.28477-1-tiwai@suse.de
find_format_descriptor() walks the class-specific interface extras by
advancing with bLength. It rejects descriptors that extend past the
remaining buffer, but it does not reject descriptor lengths smaller than
a USB descriptor header.
Reject too-short descriptors before using bLength to advance the local
scan. This keeps the UA-101 parser robust against malformed descriptor
data and matches the usual USB descriptor walking rules.
Fixes: 63978ab3e3 ("sound: add Edirol UA-101 support")
Cc: stable@vger.kernel.org
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260519-alsa-ua101-desc-len-v1-1-4307d1a5e054@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_seq_create_port() walks client->ports_list_head looking for
the ordered insertion point and on loop fall-through passes
&p->list to list_add_tail():
list_for_each_entry(p, &client->ports_list_head, list) {
if (p->addr.port == port) {
kfree(new_port);
return -EBUSY;
}
if (p->addr.port > num)
break;
...
}
list_add_tail(&new_port->list, &p->list);
When the loop walks all entries without break (e.g., the new
port sorts last), p is past-the-end. &p->list aliases
&client->ports_list_head (the list head) via container_of offset
cancellation, so the insert lands at the list tail. That is the
intended behaviour, but the access is undefined per C11 even
though it works in practice.
Track an explicit insert_before pointer initialised to the list
head and overwritten to &p->list only when the loop breaks
early. The observable behaviour is unchanged.
Fixes: 9244b2c307 ("[ALSA] alsa core: convert to list_for_each_entry*")
Signed-off-by: Maoyi Xie <maoyixie.tju@gmail.com>
Link: https://patch.msgid.link/20260518194023.1667857-3-maoyixie.tju@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
