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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.
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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>
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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>
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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>
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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: 28acb12014fb ("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>
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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: 8fdff6a319e7 ("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>
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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>
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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: f57f3df03a8e ("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>
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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>
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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: 4cf8752a03e6 ("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>
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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: 84d6a465f211 ("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>
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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
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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>
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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>
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A card level fixup_controls callback was added in:
commit df4d27b19b89 ("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>
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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>
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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>
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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>
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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>
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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 0a10faad5ca5 ("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>
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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>
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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: da3039e91d1f ("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>
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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>
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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>
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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: 4434ade8c933 ("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>
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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: 611a96f6acf2 ("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>
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snd_ump_attach_legacy_rawmidi() allocates the legacy conversion array
ump->out_cvts and, on the snd_rawmidi_new() error path, frees it with
kfree() but leaves ump->out_cvts pointing at the freed memory. When the
endpoint is later torn down, snd_ump_endpoint_free() frees ump->out_cvts
a second time, resulting in a double free.
The host snd-usb-audio driver attaches the legacy rawmidi for any USB
MIDI 2.0 (UMP) device, so a device that makes snd_rawmidi_new() fail
reaches this path on enumeration.
Clear ump->out_cvts after freeing it on the error path so it is not
freed again during teardown.
Discovered by XBOW, triaged by Baul Lee <baul.lee@xbow.com>
Fixes: 33cd7630782d ("ALSA: ump: Export MIDI1 / UMP conversion helpers")
Reported-by: Federico Kirschbaum <federico.kirschbaum@xbow.com>
Reported-by: Baul Lee <baul.lee@xbow.com>
Cc: stable@vger.kernel.org
Signed-off-by: Baul Lee <baul.lee@xbow.com>
Link: https://patch.msgid.link/20260726051633.41206-1-baul.lee@xbow.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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create_midi2_ump() registers a card-owned snd_ump_endpoint and stores a
back-pointer to its per-interface snd_usb_midi2_ump object in
ump->private_data, but it never installs an ump->private_free hook and
never clears that pointer.
If a later step of snd_usb_midi_v2_create() fails, its error path calls
free_all_midi2_umps(), which kfree()s the snd_usb_midi2_ump object while
the already-registered endpoint keeps pointing at it. The created
/dev/snd/umpC*D* node stays exposed, so the first operation of any UMP
open, ump_to_endpoint(), dereferences the dangling ump->private_data and
reads rmidi->eps[dir] out of freed memory.
A malicious USB MIDI 2.0 device that makes creation fail after the
endpoint is registered can thus trigger a slab use-after-free read on a
subsequent open of the UMP node.
Clear the endpoint's back-pointer before freeing the object, and let
ump_to_endpoint() tolerate a NULL private_data so the open/close/trigger
callbacks fail cleanly (their callers already handle a NULL endpoint)
instead of dereferencing a stale pointer.
Discovered by XBOW, triaged by Baul Lee <baul.lee@xbow.com>
Fixes: ff49d1df79ae ("ALSA: usb-audio: USB MIDI 2.0 UMP support")
Reported-by: Federico Kirschbaum <federico.kirschbaum@xbow.com>
Reported-by: Baul Lee <baul.lee@xbow.com>
Cc: stable@vger.kernel.org
Signed-off-by: Baul Lee <baul.lee@xbow.com>
Link: https://patch.msgid.link/20260726051337.41124-1-baul.lee@xbow.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Acer Nitro 5 AN515-46 (SSID 1025:159e, Realtek ALC287) has a combo
headset jack whose microphone does not work out of the box: the BIOS
leaves pin 0x19 unconfigured, so no headset mic is created.
Apply ALC2XX_FIXUP_HEADSET_MIC, the same fixup already used by the
sibling models AN515-57 (1025:1539) and AN517-55 (1025:1597), which
makes the headset microphone work correctly.
Tested on an Acer Nitro 5 AN515-46 by overriding the model via a patch
firmware with model=alc2xx-fixup-headset-mic.
Signed-off-by: Marcos Paulo Medeiros <maarcospm1996@gmail.com>
Link: https://patch.msgid.link/20260725214131.25872-1-maarcospm1996@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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TongFang X6KK45xU and X6SP45xU have actually different PCI IDs. This patch
Adds the missing PCI ID to fix headphone detection and clarifies the
naming.
Fixes: d595255241e5 ("ALSA: hda/realtek: Add quirk for TongFang X6xx45xU")
Signed-off-by: Eckhart Mohr <e.mohr@tuxedocomputers.com>
Cc: stable@vger.kernel.org
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com>
Link: https://patch.msgid.link/20260724190109.169889-1-wse@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Setting up the interface when suspended/resuming fails on this card.
Adding a reset and delay quirk will eliminate this problem.
Note: This device's VID conflicts with Apple's (0x05ac).
usb 1-1: New USB device found, idVendor=05ac, idProduct=110b
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 1-1: Product: Q2A
usb 1-1: Manufacturer: JKY Technology
usb 1-1: SerialNumber: 330270D2251225
Suggested-by: Rong Zhang <i@rong.moe>
Signed-off-by: Lianqin Hu <hulianqin@vivo.com>
Reviewed-by: Rong Zhang <i@rong.moe>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/TYUPR06MB6217566CFD33F57D3AE46816D2CF2@TYUPR06MB6217.apcprd06.prod.outlook.com
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This laptop with an ALC236 codec requires the
ALC236_FIXUP_HP_MUTE_LED_COEFBIT2
fixup for its mute LED to function correctly.
Add the subsystem ID 0x103c:0x86c8 to the quirk table to apply this
fixup.
Signed-off-by: Madhavender Singh <madhav@disroot.org>
Link: https://patch.msgid.link/20260723104736.23386-1-madhav@disroot.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The lx6464es driver advertises both 16-bit and packed 24-bit PCM formats,
but lx_trigger_start() and lx_interrupt_request_new_buffer() calculate the
DMA period size as runtime->period_size * runtime->channels * 3. That is
only correct for the packed 24-bit formats.
For 16-bit streams the driver submits buffers that are 50% larger than the
actual ALSA period and advances the DMA address by the same wrong amount.
For example, with 2 channels, 256 frames and 4 periods, the third buffer
already extends beyond the ALSA buffer and the fourth buffer starts outside
it.
Use snd_pcm_lib_period_bytes() so the byte count matches the runtime
format, channel count and period size.
Fixes: 02bec4904508 ("ALSA: lx6464es - driver for the digigram lx6464es interface")
Cc: stable@vger.kernel.org
Signed-off-by: Xu Rao <raoxu@uniontech.com>
Link: https://patch.msgid.link/8BB12E8D92A7CDBA+20260723085710.2567463-1-raoxu@uniontech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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A userspace-driven ALSA timer (SND_UTIMER) lets an unprivileged user set
the backing snd_timer's hardware resolution to an arbitrary 64-bit value
via SNDRV_TIMER_IOCTL_CREATE. snd_utimer_create() only rejects zero.
When such a timer is bound to a sequencer queue, initialize_timer()
computes the tick period as
tmr->ticks = 1000000000 / (r * freq);
where r is that user-controlled resolution and freq is the sequencer
update rate in Hz, clamped to MIN_FREQUENCY..MAX_FREQUENCY (10..6250).
A resolution of 2^63 makes the 64-bit product r * freq wrap to zero for
any even freq, including DEFAULT_FREQUENCY (1000), so the division faults
with a divide-by-zero.
The division runs under tmr->lock with interrupts disabled, so the oops
leaves the spinlock held and hangs the CPU. It is reachable by an
unprivileged user with access to /dev/snd/timer and /dev/snd/seq.
Oops: divide error: 0000 [#1] SMP KASAN PTI
CPU: 7 UID: 1000 PID: 456 Comm: alsa_seq_utimer Not tainted 7.2.0-rc4+
RIP: 0010:initialize_timer.constprop.0+0x20a/0x2d0
snd_seq_timer_start+0x15e/0x2b0
snd_seq_control_queue+0x56f/0xba0
snd_seq_write+0x3e0/0x730
Reject an overflowing product with check_mul_overflow() and fall back to
a single tick, which also avoids feeding a wrapped-but-nonzero divisor
(e.g. 2^63 * 1000 mod 2^64 == 0, or other resolutions wrapping to a small
value) into the period computation.
Fixes: 37745918e0e7 ("ALSA: timer: Introduce virtual userspace-driven timers")
Cc: <stable@vger.kernel.org>
Assisted-by: Claude:claude-opus-4-8
Signed-off-by: Norbert Szetei <norbert@doyensec.com>
Link: https://patch.msgid.link/DF8A3844-AD5E-4B8A-9CFC-BD83C212BA38@doyensec.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Razer Barracuda X 2.4 GHz USB headset dongle (0x1532:0x0552)
reports a minimum volume register value of cval->min = -16800.
In UAC 1/256 dB units, -16800 corresponds to -65.625 dB. However,
stock ALSA misinterprets this raw integer as 1/100 dB units
(-168.00 dB), causing user-space audio servers (PipeWire /
PulseAudio) to map their volume curves against an incorrectly wide
range.
Add an explicit usbmix_dB_map entry overriding Unit 2 to -6562
(-65.62 dB) to accurately report the physical hardware
attenuation bounds.
Signed-off-by: Markus Lindner <lindner.markus@outlook.at>
Link: https://patch.msgid.link/AS8P195MB2142F4EFF83980BD02BA6566E1C12@AS8P195MB2142.EURP195.PROD.OUTLOOK.COM
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Adding rt5682 I2S codec support for NVL platforms and entry in match table.
Signed-off-by: Balamurugan C <balamurugan.c@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20260723062121.869857-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Charles Keepax <ckeepax@opensource.cirrus.com> says:
Fix up some error path and size checking issues in the SDCA code.
Link: https://patch.msgid.link/20260722103500.872714-1-ckeepax@opensource.cirrus.com
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When reading the Ranges structure from an SDCA Control, ensure that the
read data is large enough to encompass the required header before
accessing it.
Fixes: 64fb5af1d1bb ("ASoC: SDCA: Add parsing for Control range structures")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Link: https://patch.msgid.link/20260722103500.872714-5-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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If message offset was larger than the buffer length the size
check will pass incorrectly. Refactor the check such that it is
more robust to invalid sizes.
Fixes: daab108504be ("ASoC: SDCA: Add UMP buffer helper functions")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Link: https://patch.msgid.link/20260722103500.872714-4-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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In the case a disk firmware exists but is invalid and no SWFT firmware
exists fdl_load_file() will return without calling release_firmware().
Update the code to call this to ensure the firmware is released on the
error path.
Fixes: 71f7990a34cd ("ASoC: SDCA: Add FDL library for XU entities")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Link: https://patch.msgid.link/20260722103500.872714-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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devm_acpi_table_put() takes a struct acpi_table_header * but the value
passed in is struct acpi_table_header ** so the value passed to
acpi_put_table() is actually the pointer not the table itself.
Remove the extra reference to correct the passed value.
Fixes: c4d096c3ca42 ("ASoC: SDCA: Add SDCA FDL data parsing")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Link: https://patch.msgid.link/20260722103500.872714-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The Acer Aspire AG14-22P has its internal microphone connected to the
ACP as a digital microphone, but the BIOS does not advertise it via the
AcpDmicConnected ACPI property, so the internal microphone does not work
out of the box. Add a DMI quirk to enable it.
Tested on an Aspire AG14-22P (board Dove2_MDU, BIOS V1.03): the acp6x
DMIC card is created and the internal microphone captures audio.
Assisted-by: Claude-Code:claude-fable-5
Signed-off-by: Jorn Baayen <jhbaayen@fastmail.com>
Link: https://patch.msgid.link/20260721151924.30316-1-jhbaayen@fastmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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This HP All-in-One desktop (model 27-ca1xxx) uses an ALC274 codec
and has no sound from either the internal speakers or the headphone
jack. The audio output requires a combination of a specific verb
setup and GPIO configuration to become functional. Additionally,
the power_save feature must be disabled; otherwise, audible clicks
or pops occur during power state transitions.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=220694
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260721113539.317561-1-zhangheng@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Setting up the interface when suspended/resuming fails on this card.
Adding a reset and delay quirk will eliminate this problem.
usb 1-1: New USB device found, idVendor=0124, idProduct=0c21
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 1-1: Product: USB Headphone
usb 1-1: Manufacturer: Generic
usb 1-1: SerialNumber: 20210726905926
Signed-off-by: Lianqin Hu <hulianqin@vivo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/TYUPR06MB6217CBB68C8F868C076A4353D2C12@TYUPR06MB6217.apcprd06.prod.outlook.com
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The MSI Crosshair A16 HX D7WFKG has an internal digital microphone
connected through AMD ACP6x, but the system does not expose the
AcpDmicConnected ACPI property. As a result, acp_yc_mach does not bind
and no DMIC capture device is created.
Add a DMI quirk for this model. This was tested on a system with board
MS-15PL, BIOS E15PLAMS.10B, and AMD ACP6x rev 0x62. After applying the
quirk, the internal microphone appears as an acp6x DMIC capture device
and records correctly.
Signed-off-by: 郑鸿源 <spark_psc@outlook.com>
Reviewed-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://patch.msgid.link/20260721-master-v1-1-6f56d5a54d96@outlook.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Vendor want to add more machine on this workaround.
Fixes: 97272a5704bf ("ALSA: hda/realtek - Fixed Headphone noise issue for Dell QCM1255")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/e13d08e96ac449b6994d56dfe6ce3f5c@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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tas2563_save_calibration() assigns cali_data.total_sz before it reads the
per-device calibration data from EFI, but its error paths return without
clearing it again. cali_data.cali_reg_array is left all zero, because the
function returns before the register addresses are assigned.
On the first playback tasdev_load_calibrated_data() does
if (!data || !cali_data->total_sz)
return;
which passes, since total_sz is still non-zero. It then issues five
4-byte bulk writes to p->r0_reg, p->r0_low_reg, p->invr0_reg, p->pow_reg
and p->tlimit_reg, all of which are 0. Register 0 decodes to book 0 /
page 0 / register 0x00, so the auto-incrementing block write zeroes
registers 0x00 to 0x03. Register 0x03 is PB_CFG1, which holds AMP_LEVEL,
so the amplifier gain is set to its minimum and the speaker stays silent.
This is reproducible on a Lenovo Yoga 7 14ARB7 (two TAS2563 on I2C,
ACPI INT8866) whose factory calibration was never written to UEFI, so the
EFI read fails with EFI_NOT_FOUND. The two woofers driven by the
amplifiers are silent while the tweeters driven directly by the ALC287
play. Reading the amplifier registers over i2c shows PWR_CTL = 0x00
(active) and the TDM slots correctly programmed by the RCA profile, but
PB_CFG1 = 0x00. With this change PB_CFG1 keeps its power-on default of
0x20 and both woofers play.
tas2781_save_calibration() in tas2781_hda.c already clears total_sz on
failure; do the same for the TAS2563 variant.
Signed-off-by: Philipp Oster <philippdev5396@outlook.de>
Link: https://patch.msgid.link/20260720-tas2781-calfix-v1-1-3a5fa6ad90bc@outlook.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add a codec SSID quirk for Samsung ELECTRONICS 750XBE/730XBE using
HDA_CODEC_QUIRK() instead of SND_PCI_QUIRK(), because the alsa-info
report from this device does not expose a PCI subsystem ID, only the
HDA codec subsystem ID (0x144d:0xc824) is available.
This applies ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET to fix sound
being very low and distorted on the headphone jack of this system.
Reported-by: Caio Ramos <caioramos97@gmail.com>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=208663
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260720123702.799474-1-zhangheng@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The firmware on the Lunnen Ground 14 marks pin 0x1b as unused even
though the internal speakers are connected to it. As a result, the
speakers are not detected.
Add a pin configuration quirk for PCI subsystem ID 2782:a212 to configure
pin 0x1b as an internal speaker.
The pin configuration was tested on a Lunnen Ground 14 (DMI product LL4FA)
with an ALC269VC codec. The internal speakers and microphone work as
expected.
Cc: stable@vger.kernel.org
Signed-off-by: Nikita Maksimov <nickstogramm@yandex.ru>
Link: https://patch.msgid.link/20260720180214.73770-1-nickstogramm@yandex.ru
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In max98090_probe(), the -EPROBE_DEFER check after devm_clk_get() is
broken due to a missing IS_ERR() guard.
The code intends to return -EPROBE_DEFER only when the clock lookup
fails with that specific error. However, without IS_ERR() the check:
if (PTR_ERR(max98090->mclk) == -EPROBE_DEFER)
is called unconditionally, including when devm_clk_get() succeeds and
returns a valid pointer. Calling PTR_ERR() on a valid pointer
reinterprets its address as a signed long; the result is arbitrary
and is almost never equal to -EPROBE_DEFER, so the check silently
does nothing in the success case. When devm_clk_get() fails with
any error other than -EPROBE_DEFER the check is also skipped, leaving
max98090->mclk holding an error pointer with no indication to the caller.
This means a deferred probe will never actually be triggered for this
device, and any non-EPROBE_DEFER clock error is silently swallowed with
the error pointer left in the mclk field.
Fix this by adding the missing IS_ERR() guard around the PTR_ERR() call,
matching the pattern already used in the sibling max98088 and wm8960
drivers.
Fixes: b10ab7b838bd ("ASoC: max98090: Add master clock handling")
Signed-off-by: Uday Khare <udaykhare77@gmail.com>
Link: https://patch.msgid.link/20260720104254.14948-1-udaykhare77@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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