gusclassic and gusextreme still leave their ISA PM callbacks disabled
because the shared GF1 core only provides probe-time startup and full
shutdown paths.
Those helpers are not suitable for suspend and resume. They reset software
handlers and tear down runtime state such as the DRAM allocator, timer
state, DMA queues, PCM state and UART setup. Resume instead needs a
narrower recovery path that rebuilds the GF1 hardware state without
rerunning probe-only detection or discarding the bookkeeping kept by the
card instance.
Add shared GF1 suspend and resume helpers for that recovery path. Suspend
now quiesces GF1 PCM, aborts queued GF1 DMA work, resets the UART and
powers the chip down without tearing down allocator, timer or rawmidi
bookkeeping. Resume rebuilds the GF1 hardware state, restores timer and
UART handlers, and brings the chip back to a usable post-resume state for
the ISA front-ends.
The scope is limited to restoring post-resume usability. It does not
attempt transparent continuation of active GF1 PCM or synth state across
suspend, and userspace may still need to reprepare streams or reload
onboard sample data after resume. Open rawmidi substreams are restored
only to a usable post-resume state.
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260406-b4-alsa-gus-isa-pm-v1-1-b6829a7457cd@gmail.com
spdif_passthru_playback_get_resources() uses atc->pll_rate as the RSR
for the MSR calculation loop. However, pll_rate is only updated in
atc_pll_init() and not in hw_pll_init(), so it remains 0 after the
card init.
When spdif_passthru_playback_setup() skips atc_pll_init() for
32000 Hz, (rsr * desc.msr) always becomes 0, causing the loop to spin
indefinitely.
Add fallback to use atc->rsr when atc->pll_rate is 0. This reflects
the hardware state, since hw_card_init() already configures the PLL
to the default RSR.
Fixes: 8cc7236148 ("ALSA: SB X-Fi driver merge")
Cc: stable@vger.kernel.org
Signed-off-by: Harin Lee <me@harin.net>
Link: https://patch.msgid.link/20260406074913.217374-1-me@harin.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 391e69143d increased CT_PTP_NUM from 1 to 4 to support 256
playback streams, but the additional pages are not used by the card
correctly. The CT20K2 hardware already has multiple VMEM_PTPAL
registers, but using them separately would require refactoring the
entire virtual memory allocation logic.
ct_vm_map() always uses PTEs in vm->ptp[0].area regardless of
CT_PTP_NUM. On AMD64 systems, a single PTP covers 512 PTEs (2M). When
aggregate memory allocations exceed this limit, ct_vm_map() tries to
access beyond the allocated space and causes a page fault:
BUG: unable to handle page fault for address: ffffd4ae8a10a000
Oops: Oops: 0002 [#1] SMP PTI
RIP: 0010:ct_vm_map+0x17c/0x280 [snd_ctxfi]
Call Trace:
atc_pcm_playback_prepare+0x225/0x3b0
ct_pcm_playback_prepare+0x38/0x60
snd_pcm_do_prepare+0x2f/0x50
snd_pcm_action_single+0x36/0x90
snd_pcm_action_nonatomic+0xbf/0xd0
snd_pcm_ioctl+0x28/0x40
__x64_sys_ioctl+0x97/0xe0
do_syscall_64+0x81/0x610
entry_SYSCALL_64_after_hwframe+0x76/0x7e
Revert CT_PTP_NUM to 1. The 256 SRC_RESOURCE_NUM and playback_count
remain unchanged.
Fixes: 391e69143d ("ALSA: ctxfi: Bump playback substreams to 256")
Cc: stable@vger.kernel.org
Signed-off-by: Harin Lee <me@harin.net>
Link: https://patch.msgid.link/20260406074857.216034-1-me@harin.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A "-Wmissing-field-initializers" warning was emitted when compiling the
module using the W=2 option. There is a sentinel initializer field
missing in the end of scarlett2_devices[]. Tested using a
Scarlett Solo 4th gen.
Fixes: d98cc48902 ("ALSA: scarlett2: Move USB IDs out from device_info struct")
Signed-off-by: Panagiotis Petrakopoulos <npetrakopoulos2003@gmail.com>
Link: https://patch.msgid.link/20260405222548.8903-1-npetrakopoulos2003@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
onyx_prepare() accepts 32/44.1/48 kHz PCM playback, but it leaves the
Onyx IEC958 sample-rate status bits at the driver's initial 44.1 kHz
setting in DIG_INFO3. As a result, 32 kHz and 48 kHz PCM streams
advertise a stale IEC958 sample rate unless userspace rewrites IEC958
Playback Default first.
Update only the consumer sample-frequency bits in DIG_INFO3 from the PCM
runtime during prepare, resolving the long-standing FIXME in the PCM
playback path while leaving the other user-controlled IEC958 status bits
unchanged.
Mark IEC958 Playback Default as volatile as well, since prepare() now
changes the exposed register contents outside the control put callback.
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260403-onyx-spdif-pcm-rate-v1-1-dcfaf931cf83@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "IEC958 Default PCM Playback Switch" control is backed directly by
mout->share_spdif. The share-switch callbacks currently access that state
without serialization, and spdif_share_sw_put() always returns 0, so
normal userspace writes never emit the standard ALSA control value
notification.
snd_hda_multi_out_analog_open() may also clear mout->share_spdif when the
analog PCM capabilities and the SPDIF capabilities no longer intersect.
That fallback is still needed to avoid creating an impossible hw
constraint set, but it changes the mixer backing value without notifying
subscribers.
Protect the share-switch callbacks with spdif_mutex like the other SPDIF
control handlers, return the actual change value from spdif_share_sw_put(),
and notify the cached control when the open path forcibly disables
shared SPDIF mode after dropping spdif_mutex.
This keeps the existing auto-disable behavior while making switch state
changes visible to userspace.
Fixes: 9a08160bdb ("[ALSA] hda-codec - Add "IEC958 Default PCM" switch")
Fixes: 022b466fc3 ("ALSA: hda - Avoid invalid formats and rates with shared SPDIF")
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260403-hda-spdif-share-notify-v3-1-4eb1356b0f17@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It(ID 31b2:0111 JU Jiu) reports a MIN value -12800 for volume control, but
will mute when setting it less than -10880.
Thanks to my girlfriend Kagura for reporting this issue.
Cc: Kagura <me@mail.kagurach.uk>
Cc: stable@vger.kernel.org
Signed-off-by: Cryolitia PukNgae <cryolitia.pukngae@linux.dev>
Link: https://patch.msgid.link/20260402-syy-v1-1-068d3bc30ddc@linux.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently we have a return code on the driver pointer operation but the
core ignores that. Let's start paying attention.
Reported-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260401-alsa-unconfigured-tstamp-v1-2-694c2cb5f71d@kernel.org
There are a number of mechanisms, including the userspace accessible
timestamp and buffer availability ioctl()s, which allow us to trigger
a timestamp update on a stream before it has been configured. Since
drivers might rely on stream configuration for reporting of pcm_io_frames,
including potentially doing a division by the number of channels, and
these operations are not meaningful for an unconfigured stream reject
attempts to read timestamps before any configuration is done.
Signed-off-by: Mark Brown <broonie@kernel.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260401-alsa-unconfigured-tstamp-v1-1-694c2cb5f71d@kernel.org
The ISA ES1688 driver still carries a disabled suspend/resume block in
its isa_driver definition, while the same file already provides minimal
power-management handling for the PnP ES968 path.
Add ISA-specific PM callbacks and factor the existing ES1688 suspend and
resume sequence into common card-level helpers shared by both probe
paths. Suspend moves the card to D3hot. Resume reinitializes the chip
with snd_es1688_reset() and restores the card to D0, propagating reset
failures to the caller.
This wires up power-management callbacks for the ISA path and keeps the
PM handling consistent between the ISA and PnP probe paths.
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260401-alsa-es1688-pm-v1-1-510767628fe6@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the capability checks in the SRC and SRCIMP allocation loops
with a precomputed loop bound. Cards with a dedicated mic input
(SB1270, OK0010) allocate all NUM_ATC_SRCS entries, otherwise stop
at 4.
Signed-off-by: Harin Lee <me@harin.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260401090159.2404387-4-me@harin.net
Skip the unused DAIO type per model (SPDIFIO on CTSB073X, SPDIFI_BAY
on all others) and use the correct DAIO type directly as da_desc
type. This removes the mismatch and misleading between the actual
DAIO resource and the da_desc type like SPDIFI_BAY (formerly SPDIFI1).
Update related functions accordingly, and drop the unreachable
SPDIFI_BAY case from the hw20k2 daio_device_index().
Signed-off-by: Harin Lee <me@harin.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260401090159.2404387-3-me@harin.net
Rename the SPDIFI1 enum value to SPDIFI_BAY to better reflect its
purpose as the S/PDIF input on the internal drive bay, as opposed to
the S/PDIF input via Flexijack or optical (SPDIFIO; not SPDIFI-zero).
Signed-off-by: Harin Lee <me@harin.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260401090159.2404387-2-me@harin.net
The i2sbus PCM code uses pi->active to constrain the sibling stream to
an already prepared duplex format and rate in i2sbus_pcm_open().
That state is set from i2sbus_pcm_prepare(), but the current code only
clears it on close. As a result, the sibling stream can inherit stale
constraints after the prepared state has been torn down.
Clear pi->active when hw_params() or hw_free() tears down the prepared
state, and set it again only after prepare succeeds.
Replace the stale FIXME in the duplex constraint comment with a description
of the current driver behavior: i2sbus still programs a single shared
transport configuration for both directions, so mixed formats are not
supported in duplex mode.
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202604010125.AvkWBYKI-lkp@intel.com/
Fixes: f3d9478b2c ("[ALSA] snd-aoa: add snd-aoa")
Cc: stable@vger.kernel.org
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260331-aoa-i2sbus-clear-stale-active-v2-1-3764ae2889a1@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Lenovo Yoga Pro 7 14IMH9 (DMI: 83E2) shares PCI SSID 17aa:3847
with the Legion 7 16ACHG6, but has a different codec subsystem ID
(17aa:38cf). The existing SND_PCI_QUIRK for 17aa:3847 applies
ALC287_FIXUP_LEGION_16ACHG6, which attempts to initialize an external
I2C amplifier (CLSA0100) that is not present on the Yoga Pro 7 14IMH9.
As a result, pin 0x17 (bass speakers) is connected to DAC 0x06 which
has no volume control, making hardware volume adjustment completely
non-functional. Audio is either silent or at maximum volume regardless
of the slider position.
Add a HDA_CODEC_QUIRK entry using the codec subsystem ID (17aa:38cf)
to correctly identify the Yoga Pro 7 14IMH9 and apply
ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN, which redirects pin 0x17 to
DAC 0x02 and restores proper volume control. The existing Legion entry
is preserved unchanged.
This follows the same pattern used for 17aa:386e, where Legion Y9000X
and Yoga Pro 7 14ARP8 share a PCI SSID but are distinguished via
HDA_CODEC_QUIRK.
Link: https://github.com/nomad4tech/lenovo-yoga-pro-7-linux
Tested-by: Alexander Savenko <alex.sav4387@gmail.com>
Signed-off-by: Alexander Savenko <alex.sav4387@gmail.com>
Link: https://patch.msgid.link/20260331082929.44890-1-alex.sav4387@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent refactoring of xfi driver changed the assignment of
atc->daios[] at atc_get_resources(); now it loops over all enum
DAIOTYP entries while it looped formerly only a part of them.
The problem is that the last entry, SPDIF1, is a special type that
is used only for hw20k1 CTSB073X model (as a replacement of SPDIFIO),
and there is no corresponding definition for hw20k2. Due to the lack
of the info, it caused a kernel crash on hw20k2, which was already
worked around by the commit b045ab3dff ("ALSA: ctxfi: Fix missing
SPDIFI1 index handling").
This patch addresses the root cause of the regression above properly,
simply by skipping the incorrect SPDIF1 type in the parser loop.
For making the change clearer, the code is slightly arranged, too.
Fixes: a2dbaeb5c6 ("ALSA: ctxfi: Refactor resource alloc for sparse mappings")
Cc: <stable@vger.kernel.org>
Link: https://bugzilla.suse.com/show_bug.cgi?id=1259925
Link: https://patch.msgid.link/20260331081227.216134-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mention complains about this coding style:
ERROR: code indent should use tabs where possible
#6640: FILE: sound/hda/codecs/realtek/alc269.c:6640:
+ [ALC233_FIXUP_LENOVO_GPIO2_MIC_HOTKEY] = {$
fix it up.
Fixes: 5de5db3535 ("ALSA: hda/realtek - Enable Mute LED for Lenovo platform")
Signed-off-by: Lei Huang <huanglei@kylinos.cn>
Link: https://patch.msgid.link/20260331024036.30782-1-huanglei814@163.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Pin Complex 0x17 (bass/woofer speakers) is incorrectly reported as
unconnected in the BIOS (pin default 0x411111f0 = N/A). This causes the
kernel to configure speaker_outs=0, meaning only the tweeters (pin 0x14)
are used. The result is very low, tinny audio with no bass.
The existing quirk ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN (already present
in patch_realtek.c for SSID 0x17aa3801) fixes the issue completely.
Reported-by: Garcicasti <andresgarciacastilla@gmail.com>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=221298
Signed-off-by: songxiebing <songxiebing@kylinos.cn>
Link: https://patch.msgid.link/20260331033650.285601-1-songxiebing@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for laptops:
- ASUS PM5406CGA
- ASUS PM5606CGA
- ASUS P5406CCA
- ASUS P5606CCA
Laptops use 2 CS35L41 Amps with HDA, using Internal boost, with I2C or
SPI.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://patch.msgid.link/20260330134651.443439-3-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP Dragonfly 13.5 inch G4 (SSID 103C8B63) has _DSD properties in
ACPI firmware with valid reset-gpios and cs-gpios for the four CS35L41
amplifiers on SPI.
However, the _DSD specifies cirrus,boost-type as Internal (0), while
the hardware requires External Boost. With Internal Boost configured,
the amplifiers trigger "Amp short error" when audio is played at
moderate-to-high volume, eventually shutting down entirely.
Add a configuration table entry to override the boost type to
External, similar to the existing workaround for 103C89C6. All GPIO
indices are set to -1 since the _DSD provides valid reset-gpios and
cs-gpios.
Confirmed on BIOS V90 01.11.00 (January 2026), the latest available.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=219520
Originally-by: Nicholas Wang <me@nicho1as.wang>
Signed-off-by: Leonard Lausen <leonard@lausen.nl>
Link: https://patch.msgid.link/db84dcf91bc8dbd217b35572b177d967655ff903@lausen.nl
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is another Book2 Pro model (NP950QED) that seems equipped with
the same speaker module as the non-360 model, which requires
ALC298_FIXUP_SAMSUNG_AMP_V2_2_AMPS quirk.
Reported-by: Throw <zakkabj@gmail.com>
Link: https://patch.msgid.link/20260330162249.147665-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
i2sbus_add_dev() keeps the matched "sound" child pointer after
for_each_child_of_node() has dropped the iterator reference. Take an
extra reference before saving that node and drop it after the
layout-id/device-id lookup is complete.
The function also stores np in dev->sound.ofdev.dev.of_node without
taking a reference for the embedded soundbus device. Since i2sbus
overrides the embedded platform device release callback, balance that
reference explicitly in the local error path and in i2sbus_release_dev().
Fixes: f3d9478b2c ("[ALSA] snd-aoa: add snd-aoa")
Cc: stable@vger.kernel.org
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260330-aoa-i2sbus-ofnode-lifetime-v1-1-51c309f4ff06@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error message in cs8409_i2c_bulk_read() incorrectly says "I2C Bulk
Write Failed" when it should say "I2C Bulk Read Failed". This is a
copy-paste error from cs8409_i2c_bulk_write().
Signed-off-by: wangdicheng <wangdicheng@kylinos.cn>
Link: https://patch.msgid.link/20260330054131.434994-1-wangdich9700@163.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Same issue that the Scarlett 2i2 1st Gen had:
QUIRK_FLAG_SKIP_IFACE_SETUP causes distorted audio on the
Scarlett Solo 1st Gen (1235:801c).
Fixes: 38c322068a ("ALSA: usb-audio: Add QUIRK_FLAG_SKIP_IFACE_SETUP")
Reported-by: Dag Smedberg <dag@dsmedberg.se>
Tested-by: Dag Smedberg <dag@dsmedberg.se>
Signed-off-by: Dag Smedberg <dag@dsmedberg.se>
Link: https://patch.msgid.link/20260329170420.4122-1-dag@dsmedberg.se
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The loop creates a whitespace-stripped copy of the card shortname
where `len < sizeof(card->id)` is used for the bounds check. Since
sizeof(card->id) is 16 and the local id buffer is also 16 bytes,
writing 16 non-space characters fills the entire buffer,
overwriting the terminating nullbyte.
When this non-null-terminated string is later passed to
snd_card_set_id() -> copy_valid_id_string(), the function scans
forward with `while (*nid && ...)` and reads past the end of the
stack buffer, reading the contents of the stack.
A USB device with a product name containing many non-ASCII, non-space
characters (e.g. multibyte UTF-8) will reliably trigger this as follows:
BUG: KASAN: stack-out-of-bounds in copy_valid_id_string
sound/core/init.c:696 [inline]
BUG: KASAN: stack-out-of-bounds in snd_card_set_id_no_lock+0x698/0x74c
sound/core/init.c:718
The off-by-one has been present since commit bafeee5b1f ("ALSA:
snd_usb_caiaq: give better shortname") from June 2009 (v2.6.31-rc1),
which first introduced this whitespace-stripping loop. The original
code never accounted for the null terminator when bounding the copy.
Fix this by changing the loop bound to `sizeof(card->id) - 1`,
ensuring at least one byte remains as the null terminator.
Fixes: bafeee5b1f ("ALSA: snd_usb_caiaq: give better shortname")
Cc: stable@vger.kernel.org
Cc: Andrey Konovalov <andreyknvl@gmail.com>
Reported-by: Berk Cem Goksel <berkcgoksel@gmail.com>
Signed-off-by: Berk Cem Goksel <berkcgoksel@gmail.com>
Link: https://patch.msgid.link/20260329133825.581585-1-berkcgoksel@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit ab949d5196 ("ALSA: hda - Fix deadlock of controller device
lock at unbinding") added a temporary device_unlock()/device_lock()
pair around probe-work cancellation to avoid a deadlock between
controller unbind and codec probe.
That deadlock depended on the driver core taking both a device lock and
its parent lock during bind and unbind. Since commit 8c97a46af0
("driver core: hold dev's parent lock when needed") and follow-up
fixes, the parent lock is only taken when bus->need_parent_lock is set.
The HDA bus does not set that flag, so codec binding no longer locks
the controller device as the codec's parent.
Keep cancel_delayed_work_sync(), since the async probe/remove race
still needs to be serialized, but drop the stale unlock/relock
workaround and its outdated FIXME comment. Keeping it around only
opens an unnecessary unlocked window inside azx_remove().
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260328-hda-intel-drop-obsolete-probe-workaround-v1-1-bc43aeafc98b@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use macros to make the AF16Rig quirk table smaller.
Add a disabled block containing the theoretical quirks for the other
clock sources that the AF16Rig has. It's disabled because I cannot test
it.
Fixes: 0da18c2dd1 ("ALSA: usb-audio: Add quirks for Arturia AF16Rig")
Tested-By: Phil Willoughby <willerz@gmail.com>
Signed-off-by: Phil Willoughby <willerz@gmail.com>
Link: https://patch.msgid.link/20260328160326.23665-1-willerz@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've added the definitions of the missing GPI and GPO verbs for
reading in the previous commit, but the counter-part for setting
values is missing.
Add the definitions of missing verbs for comprehensiveness.
Link: https://patch.msgid.link/20260328134319.207482-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
print_gpio() prints the GPIO capability header and the bidirectional
GPIO state, but it never reports the separate GPI and GPO pins even
though AC_PAR_GPIO_CAP exposes their counts.
The HD-audio specification defines dedicated GPI and GPO verbs
alongside the GPIO ones, so codecs with input-only or output-only
general-purpose pins currently lose that state from
/proc/asound/card*/codec#* altogether.
Add the missing read verb definitions and extend print_gpio() to dump
the GPI and GPO pins, too, while leaving the existing IO[] output
unchanged.
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260328-hda-proc-gpi-gpo-v1-1-fabb36564bee@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The AF16Rig supports 34 channels at 44.1k/48k, 18 channels at 88.2k/96k
and 10 channels at 176.4k/192k.
This quirks is necessary because the automatic probing process we would
otherwise use fails. The root cause of that is that the AF16Rig clock is
not readable (its descriptor says that it is but the reads fail).
Except as described below, the values in the audio format quirks were
copied from the USB descriptors of the device. The rate information is
from the datasheet of the device. The clock is the internal clock of the
AF16Rig.
Tested-By: Phil Willoughby <willerz@gmail.com>
I have tested all the configurations enabled by this patch.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Signed-off-by: Phil Willoughby <willerz@gmail.com>
Link: https://patch.msgid.link/20260328112426.14816-1-willerz@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
asihpi_ctl_init() builds mixer control names in the fixed 44-byte
hpi_ctl->name buffer with sprintf().
This is not only a defensive cleanup. The current in-tree name tables and
format strings can already exceed 44 bytes. For example,
"Bitstream 0 Internal 0 Monitor Playback Volume"
is 46 characters before the trailing NUL, so the current sprintf() call
writes past the end of hpi_ctl->name.
The generated control name is used as the ALSA control element key, so
blindly truncating it is not sufficient. Switch the formatting to
snprintf() and emit an error if truncation happens, showing the
truncated name while still keeping the write bounded to hpi_ctl->name.
Signed-off-by: Pengpeng Hou <pengpeng@iscas.ac.cn>
Link: https://patch.msgid.link/20260328102808.33969-1-pengpeng@iscas.ac.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without CONFIG_SND_DYNAMIC_MINORS, ALSA reserves only two fixed minors
for compress devices on each card: comprD0 and comprD1.
snd_find_free_minor() currently computes the compress minor as
type + dev without validating dev first, so device numbers greater than
1 spill into the HWDEP minor range instead of failing registration.
ASoC passes rtd->id to snd_compress_new(), so this can happen on real
non-dynamic-minor builds.
Add a dedicated fixed-minor check for SNDRV_DEVICE_TYPE_COMPRESS in
snd_find_free_minor() and reject out-of-range device numbers with
-EINVAL before constructing the minor.
Also remove the stale TODO in compress_offload.c that still claims
multiple compress nodes are missing.
Fixes: 3eafc959b3 ("ALSA: core: add support for compressed devices")
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260325-alsa-compress-static-minors-v1-1-0628573bee1c@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a mute led quirck for HP Victus 15-fb0xxx (103c:8a3d) model
- As it used 0x8(full bright)/0x7f(little dim) for mute led on and other
values as 0ff (0x0, 0x4, ...)
- So, use ALC245_FIXUP_HP_MUTE_LED_V2_COEFBIT insted for safer approach
Cc: <stable@vger.kernel.org>
Signed-off-by: Sourav Nayak <nonameblank007@gmail.com>
Link: https://patch.msgid.link/20260327142805.17139-1-nonameblank007@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This motherboard uses USB audio instead, causing this driver to complain
about "no codecs found!".
Add it to the denylist to silence the warning.
The first attempt only matched on the PCI device, but this caused issues
for some laptops, so DMI match against the board as well.
Signed-off-by: Stuart Hayhurst <stuart.a.hayhurst@gmail.com>
Link: https://patch.msgid.link/20260327155737.21818-2-stuart.a.hayhurst@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar to commit 7b509910b3 ("ALSA hda/realtek: Add quirk for
Framework F111:000C") and previous quirks for Framework systems with
Realtek codecs.
000F is another new platform with an ALC285 which needs the same quirk.
Signed-off-by: Dustin L. Howett <dustin@howett.net>
Link: https://patch.msgid.link/20260327-framework-alsa-000f-v1-1-74013aba1c00@howett.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The NeuralDSP Quad Cortex does not support DSD playback. We need
this product-specific entry with zero quirks because otherwise it
falls through to the vendor-specific entry which marks it as
supporting DSD playback.
Cc: Yue Wang <yuleopen@gmail.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Signed-off-by: Phil Willoughby <willerz@gmail.com>
Link: https://patch.msgid.link/20260328080921.3310-1-willerz@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cássio Gabriel