Cen Zhang [Thu, 18 Jun 2026 17:00:10 +0000 (01:00 +0800)]
ALSA: usb-audio: Kill MIDI 2.0 URBs before freeing endpoints
MIDI 2.0 input URBs are started during snd_usb_midi_v2_create(). A
later setup failure can still jump to snd_usb_midi_v2_free(), which
currently frees each endpoint and its coherent URB buffers without first
stopping the submitted URBs. A completion can then dereference the
embedded URB context and endpoint state after they have been freed, or
try to resubmit from the stale endpoint.
This was observed as a KASAN slab-use-after-free in
input_urb_complete().
The buggy scenario involves two paths, with each column showing the order
within that path:
probe error path: USB completion path:
1. start_input_streams() submits 1. The HCD still owns a
input URBs. submitted input URB.
2. A later setup helper returns 2. input_urb_complete() runs
an error. with urb->context in ep.
3. snd_usb_midi_v2_free() frees 3. The completion reads ep
endpoint storage and URB buffers. state and can requeue URBs.
Make the endpoint destructor follow the same teardown ordering used for
disconnect when the endpoint has not already been disconnected: publish
ep->disconnected, kill the URBs synchronously, and drain the endpoint
before freeing URB buffers and endpoint storage. The guard avoids
repeating the stop sequence after the normal
snd_usb_midi_v2_disconnect_all() path, while still synchronizing the
direct MIDI 2.0 create-error free path.
This quirk is identical to the one for the Yamaha Steinberg UR22, here
applied to a CD player that also uses the Steinberg USB interface.
This quirk is necessary to avoid sporadic "clic" noise when using the DAC
of the player.
ALSA: usb-audio: qcom: clear opened when stream enable fails
On enable, subs->opened is set before the service_interval is validated;
an invalid interval jumps to the response label without clearing it, so
the substream is wedged at -EBUSY until a disable or disconnect.
ALSA: usb-audio: qcom: reject stream disable with no active interface
handle_uaudio_stream_req() resolves an interface index with
info_idx_from_ifnum(), which returns -EINVAL when no interface matches.
The enable branch and the response: cleanup label both guard against a
negative index, but the disable branch does not: it forms
info = &uadev[pcm_card_num].info[info_idx] and dereferences it.
uadev[].info is a pointer allocated only when a stream is first enabled,
so a negative info_idx on the disable path is unsafe in two ways:
- If the card was never enabled, .info is NULL and &info[-EINVAL] is a
wild pointer; reading info->data_ep_pipe faults (kernel oops).
- If the card was enabled at least once (.info allocated) and the
disable names an interface that does not match, &info[-EINVAL] points
before the allocation; info->data_ep_pipe / info->sync_ep_pipe are an
out-of-bounds slab read and, when non-zero, an out-of-bounds 4-byte
write (both pipe fields are cleared to 0). That is memory corruption,
not just a NULL dereference.
The request is reachable from unprivileged local userspace over
AF_QIPCRTR. Reject a disable request with no resolved interface, matching
the guard the enable path already has.
Haowen Tu [Thu, 18 Jun 2026 07:54:31 +0000 (15:54 +0800)]
ALSA: hda/realtek: Add headset mic quirk for Acer S40-54
Acer S40-54 with ALC256 does not restore headset mic detection properly
after S4 resume. After resume, headset plug events may no longer update
the headset mic state, leaving the headset microphone unavailable.
The system uses subsystem ID 1025:161f. Applying the existing
ALC256_FIXUP_ACER_MIC_NO_PRESENCE fixup restores headset mic detection
on this machine.
Maoyi Xie [Thu, 18 Jun 2026 06:03:15 +0000 (14:03 +0800)]
ALSA: caiaq: bound the length in the EP1 input parsers
snd_caiaq_input_read_erp() and snd_caiaq_input_read_io() can be reached
from snd_usb_caiaq_input_dispatch(). They read fixed byte offsets from
the reply buffer without checking the reported length. On a short reply
they decode stale bytes left from a previous, longer report and feed them
to the input layer.
This is not an out-of-bounds access. Every offset is a compile-time
driver constant. The largest is buf[21] in the Maschine ERP case. The
EP1 transfer buffer ep1_in_buf is EP1_BUFSIZE (64) bytes, and the USB
core caps actual_length at 64, so a short reply only reads in-bounds
stale data. Acting on data the device did not send is still wrong, so
bail out per usb_id case when the reply is shorter than the bytes that
case consumes.
read_erp: AK1 needs 2 bytes, Kore needs 16, Maschine needs 22.
read_io: the Kore case needs 5 bytes (buf[4]) and the Traktor Kontrol
X1 case needs 7 (buf[5]/buf[6]). The preceding key bit loop
is already bounded by "i < len * 8" and is left untouched.
snd_caiaq_input_read_analog() and snd_usb_caiaq_maschine_dispatch() are
not changed. Their callers already floor the reply length.
Maoyi Xie [Thu, 18 Jun 2026 06:03:15 +0000 (14:03 +0800)]
ALSA: caiaq: fix out-of-bounds read in the Traktor Kontrol S4 input parser
snd_usb_caiaq_tks4_dispatch() decodes the Traktor Kontrol S4 input
stream in fixed 16-byte (TKS4_MSGBLOCK_SIZE) message blocks. On every
iteration it advances buf and subtracts the block size while looping on
"while (len)".
len is urb->actual_length. That value is supplied by the device and is
not guaranteed to be a multiple of 16. When a final short block leaves
len between 1 and 15, the loop runs once more, reads up to buf[15], and
then does "len -= TKS4_MSGBLOCK_SIZE". As len is unsigned this underflows
to a huge value. The loop then keeps iterating and walking buf far past
the end of the 512-byte ep4_in_buf, reading out of bounds until a bogus
block id happens to be hit.
Iterate only while a full message block is available. This stops the
unsigned underflow and silently drops any trailing partial block, which
carries no complete control value anyway.
The sibling endpoint-4 parsers are not affected. The Traktor Kontrol X1
and Maschine arms in snd_usb_caiaq_ep4_reply_dispatch() floor
urb->actual_length before dispatching.
Ben Dooks [Wed, 17 Jun 2026 10:29:43 +0000 (11:29 +0100)]
ALSA: pcm: fix __le32 cast warning in snd_pcm_set_sync_per_card
In snd_pcm_set_sync_per_card() the le32 value is written to an u32
instead of an __le32 pointer. Fix the following warning by fixing
the type:
sound/soc/soc-pcm.c:2166:9: warning: incorrect type in argument 7 (different base types)
sound/soc/soc-pcm.c:2166:9: expected int
sound/soc/soc-pcm.c:2166:9: got restricted snd_pcm_format_t
Ai Chao [Wed, 17 Jun 2026 02:52:34 +0000 (10:52 +0800)]
ALSA: usb-audio: Add quirk flags for SC13A
The SC13A ( VID 0x1ff7, PID 0x0f81) not support reading the current
sample rate and results in an error message printed to kmsg. Set
QUIRK_FLAG_GET_SAMPLE_RATE to skip the sample rate check.
Quirky device sample:
usb 3-5.2.4.1: new high-speed USB device number 11 using xhci_hcd
usb 3-5.2.4.1: New USB device found, idVendor=1ff7, idProduct=0f81
usb 3-5.2.4.1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 3-5.2.4.1: Product: SC13A
usb 3-5.2.4.1: Manufacturer: Linux Foundation
usb 3-5.2.4.1: SerialNumber: 000002
usb 3-5.2.4.1: Found UVC 1.50 device SC13A (1ff7:0f81)
usb 3-5.2.4.1: 3:1: cannot get freq at ep 0x86
usb 3-5.2.4.1: Warning! Unlikely big volume range (=4096),
cval->res is probably wrong.
usb 3-5.2.4.1: [5] FU [Mic Capture Volume] ch = 1, val = 0/4096/1
usbcore: registered new interface driver snd-usb-audio
usb 3-5.2.4.1: 3:1: cannot get freq at ep 0x86
usb 3-5.2.4.1: 3:1: cannot get freq at ep 0x86
The Qualcomm USB audio offload driver obtains an endpoint transfer-ring
table by calling xhci_sideband_get_endpoint_buffer(). This getter passes
the endpoint ring to xhci_ring_to_sgtable(), which allocates the outer
struct sg_table with kzalloc_obj(*sgt). The event-ring path is
equivalent: xhci_sideband_get_event_buffer() also returns the result of
xhci_ring_to_sgtable().
Inside xhci_ring_to_sgtable(), sg_alloc_table_from_pages() separately
allocates the scatterlist storage referenced by sgt->sgl. The returned
object therefore has two allocation layers: the outer struct sg_table
and its internal scatterlist storage.
The Qualcomm caller only invokes sg_free_table(sgt). sg_free_table()
releases the scatterlist storage owned by the table, but it does not
free the separately allocated outer struct sg_table. The local sgt
pointer is then discarded, so every successful endpoint or event-ring
query leaks the outer object.
Call kfree(sgt) after sg_free_table(sgt) in both setup paths, after the
required page and DMA addresses have been copied out.
The Lenovo IdeaPad Slim 5 16AKP10 (PCI SSID 17aa:38b6) uses the
Conexant SN6140 codec. The internal microphone is on pin 0x1a but
the BIOS configures it with pin default 0x95a60120, which includes
a jack detection bit that causes the kernel to treat it as an
unplugged external mic rather than a fixed internal mic.
Add a pin config quirk that overrides pin 0x1a to 0x95a60130,
setting the connectivity bits to indicate a fixed/always-connected
device. This allows the internal microphone to be correctly
identified and used.
Cássio Gabriel [Mon, 15 Jun 2026 13:37:26 +0000 (10:37 -0300)]
ALSA: compress: Fix task creation error unwind
snd_compr_task_new() allocates the driver task before validating the
returned DMA buffers and reserving file descriptors. When either of
those later steps fails, the core frees its task wrapper and DMA-buffer
references without calling the driver's task_free() callback. Any
driver resources allocated by task_create() are therefore leaked.
The dual-fd allocation path also jumps to cleanup without storing the
negative get_unused_fd_flags() result in retval. Since retval still
contains the successful task_create() return value, TASK_CREATE can
incorrectly report success although the task was discarded.
Preserve the fd allocation errors and call task_free() when failure
occurs after a successful task_create() callback.
Takashi Iwai [Mon, 15 Jun 2026 18:19:22 +0000 (20:19 +0200)]
Merge tag 'asoc-v7.2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v7.2
There's been quite a lot of framework improvements this time around,
though mainly cleanups and robustness rather than user visible features.
The same pattern is seen with a lot of the driver work that's going on,
there are new features but a huge proportion of this is bug fixing and
cleanup work. We also have a good selectio of new device support.
- Improvements to SDCA jack handling from Charles Keepax.
- Use of device links to make suspend handling more robust from Richard
Fitzgerald.
- Use of a new helper to factor out a common pattern in SoundWire
enmeration from Charles Keepax.
- Slimming down of the component from Kuninori Morimoto.
- Simplification of format auto selection from Kuninori Morimoto.
- Lots of conversions to guard() from Bui Duc Phuc.
- Addition of a simple-amplifier driver supporting more featureful GPIO
controller amplifiers than the previous basic driver from Herve
Codina.
- Support for AMD ACP 7.x, Cirrus Logic CS42448/CS42888, Everest Semi
ES9356, Mediatek MT2701 and MT8196, Renesas RZ/G3E, Spacemit K3,
Texas Instruments TAC5xx2 and TAS67524.
Lianqin Hu [Mon, 15 Jun 2026 12:05:01 +0000 (12:05 +0000)]
ALSA: usb-audio: Add iface reset and delay quirk for XIBERIA K03S
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 USB device found, idVendor=36f9, idProduct=c009
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0
usb 1-1: Product: XIBERIA K03S
usb 1-1: Manufacturer: Actions
usb 1-1: usb_probe_device
Viktor Menshin [Mon, 15 Jun 2026 09:25:15 +0000 (18:25 +0900)]
ALSA: hda/realtek: Add quirk for Lenovo Xiaoxin 14 GT
The Lenovo Xiaoxin 14 GT (Chinese market model, AMD Ryzen AI 9 365)
produces constant electrical hissing and crackling noise from both
internal speakers and 3.5mm headphone jack during audio playback.
Audio works correctly on Windows.
The PCI SSID 17aa:3912 is not present in the quirk list. The device
shares the same AMD platform and ALC287 codec as neighboring Lenovo
14" AMD models (17aa:3911, 17aa:390d), so apply the same fixup.
Note: the fixup selection is based on similarity with neighboring
models and has not been verified by testing a compiled kernel.
Guidance from maintainers on the correct fixup is welcome.
Colton Jones [Mon, 15 Jun 2026 03:36:20 +0000 (03:36 +0000)]
ALSA: hda/realtek: Add CS35L41 I2C quirk for ASUS UM3405GA
The ASUS Zenbook 14 UM3405GA uses a Realtek ALC294 codec with two
Cirrus Logic CS35L41 speaker amplifiers exposed through the CSC3551 ACPI
device. The machine reports the Realtek subsystem ID 1043:19f4.
Without a PCI quirk, the codec falls back to generic pin matching and the
internal speakers remain silent even though PCM playback completes.
Add the UM3405GA subsystem ID and reuse the same ASUS I2C headset-mic
fixup used by the closely related UM3406HA. That fixup configures the
headset microphone pin and chains to CS35L41 I2C speaker-amp binding.
Takashi Iwai [Sun, 14 Jun 2026 09:07:11 +0000 (11:07 +0200)]
ALSA: timer: Fix racy timeri->timer changes with rwlock
Although we've covered the races around the timer object assignment
and release for timer instances, there are still races at starting or
stopping the timer instance. They refer to timeri->timer without
lock, hence they can still trigger UAFs.
For addressing it, this patch changes the existing slave_active_lock
spinlock to timeri_lock rwlock. It's a global rwlock applied as
read-lock when snd_timer_start() & co are called as well as
snd_timeri_timer_get() is called. In turn, the places where
timeri->timer is assigned or released are covered by the write-lock.
The patch replaces spinlock_irqsave with spinlock in a couple of
spaces because they are now already protected by timeri_lock, too.
Takashi Iwai [Sun, 14 Jun 2026 09:05:05 +0000 (11:05 +0200)]
ALSA: core: Fix unintuitive behavior of snd_power_ref_and_wait()
snd_power_ref_and_wait() takes the power refcount and doesn't leave it
no matter whether it returns an error or not. However, the majority
of callers don't expect but just returns without unreferencing in the
caller side upon errors.
For addressing the potential refcount unbalance, rather correct the
behavior of snd_power_ref_wait() to unreference upon returning an
error.
Note that the problem above is likely negligible; the function returns
an error only when the sound card is being shutdown, hence it doesn't
matter about the power refcount any longer at such a state.
Cen Zhang [Sun, 14 Jun 2026 00:48:01 +0000 (08:48 +0800)]
ALSA: seq: avoid stale FIFO cells during resize
snd_seq_fifo_resize() still needs to publish the replacement pool
before it waits for FIFO users. A blocking snd_seq_read() holds
f->use_lock while it sleeps, so concurrent senders must be able to
queue to the new pool and wake that reader instead of failing against a
closing old pool.
However, snd_seq_fifo_event_in() duplicates an event before it takes
f->lock, and snd_seq_read() can dequeue a cell and later call
snd_seq_fifo_cell_putback() if copy_to_user() or
snd_seq_expand_var_event() fails. If resize swaps f->pool and detaches
oldhead in between, either path can relink an old-pool cell after the
snapshot. That stale cell sits outside the drained oldhead list, keeps
oldpool->counter elevated, and can leave snd_seq_pool_delete() waiting
for the retired pool to drain.
Keep the existing swap-before-wait ordering in snd_seq_fifo_resize(),
but reject stale cells before any FIFO relink. Revalidate event-in cells
under f->lock and retry them against the published replacement pool, and
free stale putback cells instead of linking them back into the FIFO.
The buggy scenario involves two paths, with each column showing the
order within that path:
resize path: relink path:
1. Allocate newpool. 1. Take f->use_lock.
2. Swap f->pool to newpool and 2. Duplicate or dequeue an old-pool
detach oldhead. cell before oldpool closes.
3. Mark oldpool closing and 3. Reach a later relink point after
wait for FIFO users. resize published newpool.
4. Free oldhead and delete 4. Relink the old-pool cell after
oldpool. resize detached oldhead.
5. Drop f->use_lock.
The reproducer reports a resize ioctl blocked in the expected pool
teardown path:
Cen Zhang [Sun, 14 Jun 2026 00:48:00 +0000 (08:48 +0800)]
ALSA: seq: oss: Serialize readq reset state with q->lock
snd_seq_oss_readq_clear() resets qlen, head, and tail without
q->lock even though the normal reader and producer paths serialize the
same ring state under that spinlock. A reset can therefore race
snd_seq_oss_readq_free() or snd_seq_oss_readq_put_event() and leave
stale records in the queue, drop freshly queued ones, or report the
wrong readiness after wakeup. KCSAN reports a data race between
snd_seq_oss_readq_clear() and snd_seq_oss_readq_free().
Take q->lock while clearing the ring and resetting input_time. Factor
the enqueue logic into a caller-locked helper so
snd_seq_oss_readq_put_timestamp() updates its suppression state under
the same lock instead of racing the reset path.
The buggy scenario involves two paths, with each column showing the
order within that path:
reset path: locked readq updater:
1. snd_seq_oss_reset() or 1. A reader or callback producer
release reaches takes q->lock on the same queue.
snd_seq_oss_readq_clear().
2. snd_seq_oss_readq_clear() 2. The updater tests or modifies
resets qlen, head, tail, qlen, head, and tail.
and input_time.
3. snd_seq_oss_readq_clear() 3. The updater completes its
wakes sleepers on read-modify-write sequence.
q->midi_sleep.
4. Without q->lock, the reset 4. The resulting ring state drives
can overlap the locked later reads and readiness.
update.
KCSAN reports:
BUG: KCSAN: data-race in snd_seq_oss_readq_clear /
snd_seq_oss_readq_free
write to 0xffff8881069fe608 of 4 bytes by task 120516 on cpu 0:
snd_seq_oss_readq_free+0x6c/0x80
snd_seq_oss_read+0xcb/0x250
odev_read+0x38/0x60
vfs_read+0xff/0x600
ksys_read+0xb4/0x140
__x64_sys_read+0x46/0x60
do_syscall_64+0xbb/0x2f0
entry_SYSCALL_64_after_hwframe+0x77/0x7f
read to 0xffff8881069fe608 of 4 bytes by task 120517 on cpu 1:
snd_seq_oss_readq_clear+0x1f/0x90
snd_seq_oss_reset+0xa7/0xf0
snd_seq_oss_ioctl+0x6f6/0x7e0
odev_ioctl+0x56/0xc0
__x64_sys_ioctl+0xd1/0x120
do_syscall_64+0xbb/0x2f0
entry_SYSCALL_64_after_hwframe+0x77/0x7f
ASoC: dt-bindings: Fix RT5677 "realtek,gpio-config" type
"realtek,gpio-config" is described as six 8-bit GPIO configuration
values, and the RT5677 driver stores and reads those values as bytes.
The binding incorrectly documented the property as a uint32 array.
Document "realtek,gpio-config" as a uint8-array so the generated
schema matches the hardware definition and the existing driver helper.
Assisted-by: Codex:gpt-5-5 Signed-off-by: Rob Herring (Arm) <robh@kernel.org> Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@oss.qualcomm.com> Link: https://patch.msgid.link/20260612214911.1883234-1-robh@kernel.org Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC supports snd_soc_daifmt_parse_format() which can specify DAI
format by "dai-format" property from DT.
But strictly speaking, it is SW settings, so doesn't match to DT's policy.
Current ASoC is supporting auto format select via
snd_soc_dai_ops :: .auto_selectable_formats.
But the user is very few today.
DT doesn't need to specify the DAI format via "dai-format", if both CPU
and Codec drivers were supporting .auto_selectable_formats. It will be
automatically selected from .auto_selectable_formats.
But, I noticed that current auto format select method can't handle all cases.
For example, current .auto_selectable_formats is like below
It try to find DAI format from (A) first, and next it will use (A | B).
But it can't handle the format if some format were independent.
For example, DSP_x (y) can't use with xB_IF (x), etc.
So, I would like to update the method. New method doesn't use OR.
It try to find DAI format from (a), next it will use (b).
Switch old method to new method, Current auto select user need to update
.auto_selectable_formats. Fortunately, current few users doesn't have
above limitation. update (A)(B) to (a)(b) style is possible.
a = A
b = A | B
I would like to update method, and add .auto_selectable_formats
support on all drivers.
One note is that auto select might not find best format on some CPU/Codec
combination. So "dai-format" is necessary anyway.
And, there haven't been any big problems on .auto_selectable_formats,
because there were few users.
But if all drivers try to use this, it cannot be denied that they may
encounter unknown problems... In such case, "dai-format" can help, though.
ASoC: audio-graph-card2: recommend to use auto select DAI format
"Simple Audio Card", "Audio Graph Card", "Audio Graph Card2" are
possible to set DAI format via DT.
OTOH, ASoC is supporting .auto_selectable_formats to select DAI
format automatically. Let's recommend to use it on "Audio Graph Card2".
One note is that it keeps supporting DAI format setting via DT.
Current auto select format start with the highest priority format and
gradually add lower priority formats one by one, and search matched
format. Like A+X -> A+B+X -> A+B+C+X+Y... (a)
But in this method, we can't handle format if HW has some kind of
patterns, like A+X or B+Y etc (b).
Current drivers are using (a) style, this patch switch to use (b) style.
This is needed before update auto select format method.
Current auto select format start with the highest priority format and
gradually add lower priority formats one by one, and search matched
format. Like A+X -> A+B+X -> A+B+C+X+Y... (a)
But in this method, we can't handle format if HW has some kind of
patterns, like A+X or B+Y etc (b).
Current drivers are using (a) style, this patch switch to use (b) style.
This is needed before update auto select format method.
Current auto select format start with the highest priority format and
gradually add lower priority formats one by one, and search matched
format. Like A+X -> A+B+X -> A+B+C+X+Y... (a)
But in this method, we can't handle format if HW has some kind of
patterns, like A+X or B+Y etc (b).
Current drivers are using (a) style, this patch switch to use (b) style.
This is needed before update auto select format method.
Guangshuo Li [Thu, 4 Jun 2026 12:55:20 +0000 (20:55 +0800)]
ASoC: adau1372: Clear PLL_EN on failed PLL lock without reset GPIO
The PLL lock failure path in adau1372_set_power() unwinds by putting
the regmap back in cache-only mode, asserting the optional power-down
GPIO and disabling mclk.
adau1372_enable_pll() enables CLK_CTRL.PLL_EN before polling the PLL
lock bit. If the lock fails on a board without a power-down GPIO, the
error path disables mclk and returns an error, but leaves PLL_EN set in
the hardware register. The normal power-off path already handles the
no-GPIO case by explicitly clearing PLL_EN.
Mirror that cleanup in the PLL lock failure path and clear PLL_EN while
the regmap is still live, before switching it back to cache-only mode.
Takashi Iwai [Fri, 12 Jun 2026 11:33:45 +0000 (13:33 +0200)]
ALSA: seq: Don't re-bounce the error event
The error bouncing may fail again, and we have no check for
re-bouncing. For avoiding the loop, add the event type check at
bouncing, and stop re-bouncing if it's already a bounce error.
Mark Brown [Fri, 12 Jun 2026 15:12:13 +0000 (16:12 +0100)]
ASoC: use scoped OF node handling in manual cleanup paths
Cássio Gabriel <cassiogabrielcontato@gmail.com> says:
Some ASoC drivers still manually release child OF nodes
when leaving child-node iteration loops early.
Convert these focused cases to scoped OF node cleanup
so early returns and normal loop exits keep the same node
lifetime handling without explicit of_node_put() calls.
- Patch 1 updates qcom_snd_parse_of() to use
for_each_available_child_of_node_scoped() for link nodes and
__free(device_node) for temporary cpu/platform/codec child nodes.
- Patch 2 updates fsl_qmc_audio to use
for_each_available_child_of_node_scoped() for DAI child-node parsing.
- Patch 3 updates cygnus-ssp to use
for_each_available_child_of_node_scoped() for SSP child-node parsing.
Cássio Gabriel [Mon, 8 Jun 2026 13:39:11 +0000 (10:39 -0300)]
ASoC: qcom: common: use scoped OF node handling
qcom_snd_parse_of() manually drops the link child node and the
cpu/platform/codec child nodes on error paths and at the end of each
iteration.
Use for_each_available_child_of_node_scoped() for the link node and
__free(device_node) for the named child nodes. This keeps the existing
ownership rules for DAI component phandle references, while removing the
manual cleanup labels from a path that has previously needed OF refcount
fixes.
ASoC: sdw_utils: fix missing component_name for cs42l43 part_id 0x2A3B
commit 87a3f5c8ac20 ("ASoC: sdw_utils: cs42l43: allow spk component names
to be combined") moved spk:cs42l43-spk generation from rtd_init() into
the asoc_sdw_rtd_init() generic path by adding component_name to
codec_info_list entries. However, only the 0x4243 cs42l43 entry was
updated; the 0x2A3B entry (vendor_id 0x01fa, Cirrus Logic cs42l43 with
sidecar bridge) was missed.
Without component_name on the 0x2A3B dp6 DAI, asoc_sdw_rtd_init() never
accumulates spk_components and never appends 'spk:cs42l43-spk' (or its
sidecar alias 'spk:cs35l56-bridge') to card->components. The sof-soundwire
UCM regex ' spk:([a-z0-9]+...)' then fails to match, causing WirePlumber
to mark all HiFi profiles as unavailable=no and fall back to the Off
profile — resulting in Dummy Output in GNOME.
The existing sidecar redirect in asoc_sdw_rtd_init() already handles the
SOC_SDW_SIDECAR_AMPS case: when component_name is 'cs42l43-spk' and
sidecar amps are active, it substitutes 'cs35l56-bridge' into
card->components, which matches the existing cs35l56-bridge.conf UCM file.
These are tidyup details of mediatek drivers. Basically there is no
functional change. This is prepare for later Card capsuling.
This makes code review easy when Card capsuling happen.
Cameron Graham [Fri, 12 Jun 2026 09:46:01 +0000 (21:46 +1200)]
ALSA: hda/hdmi: Add force-connect quirk for HP EliteDesk 800 G5 Mini
The HP EliteDesk 800 G5 Mini (PCI subsystem 103c:8595) uses
Cannon Lake PCH cAVS HDA with DisplayPort audio pins 0x05 and
0x06 set to AC_JACK_PORT_NONE (N/A) in BIOS defaults, causing
hdmi_add_pin() to skip them and the DP audio device to not
appear in ALSA.
Add the board to the existing force_connect_list alongside the
similar HP EliteDesk 800 G4 entries.
HanQuan [Fri, 12 Jun 2026 10:32:22 +0000 (10:32 +0000)]
ALSA: seq: Fix kernel heap address leak in bounce_error_event()
The comment above bounce_error_event() documents that user clients
should receive SNDRV_SEQ_EVENT_BOUNCE with the original event embedded
as variable-length data, while kernel clients should receive
SNDRV_SEQ_EVENT_KERNEL_ERROR with a quoted kernel pointer.
However, the implementation unconditionally uses
SNDRV_SEQ_EVENT_KERNEL_ERROR with data.quote.event set to the raw
struct snd_seq_event pointer for all clients. When a bounce error
event is delivered to a USER_CLIENT via snd_seq_read(), the kernel
heap address in data.quote.event is exposed to userspace through
copy_to_user() in the fixed-length branch.
This is a distinct leak path from the one addressed by commit 705dd6dcbc0e ("ALSA: seq: Clear variable event pointer on read"),
which sanitizes data.ext.ptr in the variable-length branch of
snd_seq_read(). The bounce_error_event() leak uses fixed-length
events that take the else branch where no sanitization occurs.
Differentiate the bounce event by client type. For USER_CLIENT,
send SNDRV_SEQ_EVENT_BOUNCE with SNDRV_SEQ_EVENT_LENGTH_VARIABLE
and data.ext pointing to the original event. The variable-length
path in snd_seq_event_dup() copies the event data into chained
cells, and snd_seq_expand_var_event() copies only the content --
never the pointer -- to userspace. For KERNEL_CLIENT, keep the
existing SNDRV_SEQ_EVENT_KERNEL_ERROR behavior with the quoted
pointer.
Cássio Gabriel [Fri, 12 Jun 2026 03:43:55 +0000 (00:43 -0300)]
ALSA: pcxhr: Share PLL frequency register calculation
The PCXHR and HR222 clock paths duplicate the PLL divider calculation and
register encoding. The HR222 variant extends the same format with an
additional range for rates above those supported by the older boards.
Move the complete encoding into pcxhr_pll_freq_register() and pass each
hardware path its existing maximum frequency. The additional encoding
branch is unreachable with the older 110 kHz limit, so this preserves both
paths' accepted ranges and generated register values while removing the
duplicate implementation and its long-standing TODO.
qmi_stop_session() conditionally looks up the cached data and sync
endpoints, but removes each endpoint unconditionally.
The data endpoint is always present for an active offload stream, while
the sync endpoint is optional. When no sync endpoint exists, ep still
refers to the data endpoint and the code attempts to remove that endpoint
a second time. The current sideband implementation rejects the duplicate
removal, but the teardown path should not pass an unrelated endpoint for
an absent sync endpoint.
Only look up and remove an endpoint when its cached pipe exists, check the
lookup result, and clear the cached pipe after handling it. This matches
the normal stream-disable path.
Zhao Dongdong [Wed, 10 Jun 2026 07:20:43 +0000 (15:20 +0800)]
ASoC: SOF: topology: fix memory leak in snd_sof_load_topology
When the topology filename contains "dummy" and tplg_cnt is 0, the
function returns -EINVAL directly without freeing the tplg_files
allocated by kcalloc() at line 2497. This leaks memory on every
such topology load attempt.
Fix this by setting ret = -EINVAL and jumping to the out: label,
which already handles the kfree(tplg_files) cleanup.
Fixes: 99c159279c6d ("ASoC: SOF: don't check the existence of dummy topology") Cc: stable@vger.kernel.org Signed-off-by: Zhao Dongdong <zhaodongdong@kylinos.cn> Acked-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Link: https://patch.msgid.link/tencent_3EED6D778DC52C3703A2D1EE8119372E8E08@qq.com Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Thu, 11 Jun 2026 19:59:21 +0000 (20:59 +0100)]
ASoC: mediatek: Use guard() for mutex & spin locks
bui duc phuc <phucduc.bui@gmail.com> says:
This series converts mutex and spinlock handling in Mediatek ASoC drivers
to use guard() helpers.
Most patches are straightforward conversions to guard() helpers with no
functional change intended.
One exception is mt8192-afe-gpio, where the mutex release point moves from
immediately before dev_warn() to scope exit. However, the affected path
only emits a warning and immediately returns -EINVAL, without any further
processing.
ASoC: mediatek: mt8192: mt8192-afe-gpio: Use guard() for mutex locks
Convert the explicit mutex_lock()/mutex_unlock() pair to guard(mutex)
to simplify the locking logic and automatically release the mutex on
all exit paths.
This changes the mutex release point from immediately before dev_warn()
to automatic cleanup at scope exit. However, the affected path only emits
a warning and immediately returns -EINVAL, without any further processing.
Mark Brown [Thu, 11 Jun 2026 19:50:31 +0000 (20:50 +0100)]
ASoC: rockchip: Use guard() for spin locks
bui duc phuc <phucduc.bui@gmail.com> says:
This series converts spinlock handling in the Rockchip sound drivers
to use guard() helpers.
The changes are code cleanup only and should have no functional impact.
Mark Brown [Thu, 11 Jun 2026 19:49:43 +0000 (20:49 +0100)]
ASoC: rockchip: Reorder clock enable sequence
bui duc phuc <phucduc.bui@gmail.com> says:
This series reorders the runtime resume clock enable sequence in the
Rockchip SPDIF and PDM drivers to enable the bus clock before the
functional controller clock.
It also updates the SPDIF DT binding clock descriptions to match the
actual clock usage in the driver.
Additionally, this v2 adds two new patches addressing issues reported
by the Sashiko AI Review tool regarding regcache sync failure handling
and runtime PM resume status validation.
Testing:
- Patch 1: Verified (dt_binding_check passed).
- Patches 2 to 5: Compile tested only. Please help test if you have
the relevant Rockchip hardware.
ASoC: rockchip: rockchip_pdm: Handle runtime PM resume failures in set_fmt
rockchip_pdm_set_fmt() calls pm_runtime_get_sync() before accessing
hardware registers, but ignores its return value.
If the runtime resume fails, the function continues to perform register
accesses while the device state is undefined.
Replace pm_runtime_get_sync() with pm_runtime_resume_and_get() and
return early on failure to avoid unpowered register accesses.
Reported-by: Sashiko AI Review <sashiko-bot@kernel.org> Closes: https://lore.kernel.org/all/20260522110302.349421F000E9@smtp.kernel.org/ Signed-off-by: bui duc phuc <phucduc.bui@gmail.com> Link: https://patch.msgid.link/20260602101608.45137-6-phucduc.bui@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: rockchip: spdif: Restore regcache cache-only mode on sync failure
If regcache_sync() fails during runtime resume, the driver disables the
clocks and returns an error. However, the regmap cache-only mode is left
disabled.
Restore cache-only mode in the error path so subsequent register accesses
continue to use the cache while the device is inactive.
Reported-by: Sashiko AI Review <sashiko-bot@kernel.org> Closes: https://lore.kernel.org/all/20260522103713.6C09D1F000E9@smtp.kernel.org/ Signed-off-by: bui duc phuc <phucduc.bui@gmail.com> Link: https://patch.msgid.link/20260602101608.45137-5-phucduc.bui@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Enable the 'hclk' bus clock before the 'clk' controller clock during
runtime resume.
The bus clock provides the register access interface, so enable it before
the controller clock. This also makes the resume sequence the reverse of
the suspend sequence, which keeps the clock ordering consistent.
Enable the 'hclk' bus clock before the 'mclk' controller clock during
runtime resume.
The bus clock provides the register access interface, so enable it before
the controller clock. This also makes the resume sequence the reverse of
the suspend sequence, which keeps the clock ordering consistent.
Sen Wang [Wed, 3 Jun 2026 21:18:30 +0000 (16:18 -0500)]
ASoC: ti: davinci-mcasp: Add audio-graph-card2 and DPCM support
Extend the McASP driver to support audio-graph-card2 of-graph topology,
while maintaining backwards compatibility for existing simple-audio-card
phandles and machine drivers, which now uses the default MCASP_GRAPH_NONE
code path.
Cássio Gabriel [Thu, 4 Jun 2026 03:10:58 +0000 (00:10 -0300)]
ASoC: topology: Check PCM and DAI name strings before use
Topology objects store several PCM and DAI names in fixed-size UAPI
arrays. Other topology parser paths validate these fields with bounded
strnlen() checks before using them as C strings, but the PCM and DAI
paths still pass some fixed-size arrays directly to strlen(),
devm_kstrdup(), DAI lookup, and diagnostic prints.
A malformed topology blob with a non-NUL-terminated PCM, DAI, or stream
capability name can therefore make the parser read past the end of the
fixed-size field.
Reject unterminated PCM and DAI name fields before consuming them as C
strings.
Fixes: 64527e8a3529 ("ASoC: topology: Add FE DAIs dynamically") Fixes: acfc7d46cddc ("ASoC: topology: Add FE DAI links dynamically") Fixes: 0038be9a84dc ("ASoC: topology: Add support for configuring existing BE DAIs") Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com> Link: https://patch.msgid.link/20260604-asoc-topology-check-pcm-dai-names-v1-1-e1b0f6f7c2ce@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Mon, 8 Jun 2026 10:27:14 +0000 (11:27 +0100)]
ASoC: cs35l56: Remove unnecessary conditionals waiting for enumeration
Commit [1] updated the core to use complete_all() which means that
the wait_for_completion() will now simply return if the device
is already attached, so skipping the completion isn't required
anymore. Update the code to simply call sdw_slave_wait_for_init()
unconditionally.
Charles Keepax [Mon, 8 Jun 2026 10:27:13 +0000 (11:27 +0100)]
ASoC: SDCA: Use new SoundWire enumeration helper
Now the new wait for SoundWire enumeration helper no longer depends on
unattach_request it is safe to use from probe time. Update the driver
to use the new core helper.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> Tested-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> Link: https://patch.msgid.link/20260608102714.2503120-10-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Mon, 8 Jun 2026 10:27:12 +0000 (11:27 +0100)]
ASoC: wcd939x: Use new SoundWire enumeration helper
Now the new wait for SoundWire enumeration helper no longer depends on
unattach_request it is safe to use from probe time. Update the driver
to use the new core helper.
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> Tested-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://patch.msgid.link/20260608102714.2503120-9-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Mon, 8 Jun 2026 10:27:11 +0000 (11:27 +0100)]
ASoC: wcd938x: Use new SoundWire enumeration helper
Now the new wait for SoundWire enumeration helper no longer depends on
unattach_request it is safe to use from probe time. Update the driver
to use the new core helper.
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> Tested-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://patch.msgid.link/20260608102714.2503120-8-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Mon, 8 Jun 2026 10:27:10 +0000 (11:27 +0100)]
ASoC: wcd937x: Use new SoundWire enumeration helper
Now the new wait for SoundWire enumeration helper no longer depends on
unattach_request it is safe to use from probe time. Update the driver
to use the new core helper.
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> Tested-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://patch.msgid.link/20260608102714.2503120-7-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Mon, 8 Jun 2026 10:27:09 +0000 (11:27 +0100)]
ASoC: pm4125: Use new SoundWire enumeration helper
Now the new wait for SoundWire enumeration helper no longer depends on
unattach_request it is safe to use from probe time. Update the driver
to use the new core helper.
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> Tested-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://patch.msgid.link/20260608102714.2503120-6-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Mon, 8 Jun 2026 10:27:08 +0000 (11:27 +0100)]
ASoC: rt5682: Use new SoundWire enumeration helper
Now the new wait for SoundWire enumeration helper no longer depends on
unattach_request it is safe to use from probe time. Update the driver
to use the new core helper.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> Tested-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> Link: https://patch.msgid.link/20260608102714.2503120-5-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Mon, 8 Jun 2026 10:27:06 +0000 (11:27 +0100)]
ASoC: wsa881x: Use new SoundWire enumeration helper
Now the new wait for SoundWire enumeration helper no longer depends on
unattach_request it can be used for code that also doesn't check this
flag. Update the driver to use the new core helper.
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> Tested-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://patch.msgid.link/20260608102714.2503120-3-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Mon, 8 Jun 2026 10:27:05 +0000 (11:27 +0100)]
soundwire: Always wait for initialisation of unattached devices
Currently in sdw_slave_wait_for_init() the waiting can be skipped
if unattach_request is not set. Doing so was added in [1] likely
because the core used to do a complete() on the completion so
waiting in the case an unattach hadn't actually happened would
block for the full timeout. However patch [2] updated the core to
use complete_all() which means that the wait_for_completion() will
now simply return if the device is already attached skipping the
completion doesn't add much.
Additionally, unattach_request is only set if the host initiates
a bus reset. However, the host doing a bus reset is not the only
reason a device may be unattached from the bus. Other options
could include the driver probing before the device enumerates, a
sync-loss, or the device itself powering down.
Removing the skip using unattached_request, doesn't cost much in
terms of efficiency and allows the sdw_slave_wait_for_init() helper
to be used outside of runtime resume.
[1] b2bd75f806c4 ("soundwire: sdw_slave: track unattach_request to handle all init sequences")
[2] c40d6b3249b1 ("soundwire: fix enumeration completion")
Mark Brown [Thu, 11 Jun 2026 19:44:46 +0000 (20:44 +0100)]
ASoC: Validate written enum values in custom controls
HyeongJun An <sammiee5311@gmail.com> says:
Some custom ASoC kcontrol put() handlers use the written enum value
(ucontrol->value.enumerated.item[0]) to index a table or compute a bit
shift before validating that the value is within the control's enum range.
An out-of-range value written from userspace is therefore consumed before
it is rejected.
This is the same class addressed for the Meson codecs in commit 1e001206804b ("ASoC: meson: g12a-tohdmitx: Validate written enum values")
and commit 3150b70e944e ("ASoC: meson: g12a-toacodec: Validate written
enum values").
Fix four more instances:
- hdac_hdmi reads e->texts[item] before validation.
- aiu converts the item before validating it.
- fsl_audmix converts the item and uses the result before validation.
- tegra210_ahub reads e->values[item] before validation.
HyeongJun An [Tue, 9 Jun 2026 12:43:16 +0000 (21:43 +0900)]
ASoC: tegra: tegra210_ahub: Validate written enum value
tegra_ahub_put_value_enum() reads e->values[item[0]] before
checking whether item[0] is within the enum item range. The existing
check therefore happens too late to prevent an out-of-range read of the
values array.
Move the check before the array access.
Fixes: 16e1bcc2caf4 ("ASoC: tegra: Add Tegra210 based AHUB driver") Assisted-by: Claude:claude-opus-4-8 Signed-off-by: HyeongJun An <sammiee5311@gmail.com> Link: https://patch.msgid.link/20260609124317.38046-5-sammiee5311@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
HyeongJun An [Tue, 9 Jun 2026 12:43:15 +0000 (21:43 +0900)]
ASoC: fsl: fsl_audmix: Validate written enum values
fsl_audmix_put_mix_clk_src() and fsl_audmix_put_out_src()
convert the user-provided enum item with snd_soc_enum_item_to_val()
before checking whether the item is within the enum's item count.
The generic snd_soc_put_enum_double() helper performs that
validation, but these callbacks use the converted value first: the
clock-source path tests it with BIT(), and the output-source path
indexes the prms transition table with it.
Reject out-of-range enum items before converting them.
Fixes: be1df61cf06e ("ASoC: fsl: Add Audio Mixer CPU DAI driver") Assisted-by: Claude:claude-opus-4-8 Signed-off-by: HyeongJun An <sammiee5311@gmail.com> Link: https://patch.msgid.link/20260609124317.38046-4-sammiee5311@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
HyeongJun An [Tue, 9 Jun 2026 12:43:14 +0000 (21:43 +0900)]
ASoC: meson: aiu: Validate written enum values
The AIU HDMI and internal codec mux put callbacks use the written enum
value with snd_soc_enum_item_to_val() before checking whether the value is
valid for the enumeration.
Reject out-of-range values before converting the enum item, matching the
validation already done by the G12A HDMI and internal codec mux controls.
Fixes: b82b734c0e9a ("ASoC: meson: aiu: add hdmi codec control support") Fixes: 65816025d461 ("ASoC: meson: aiu: add internal dac codec control support") Assisted-by: Claude:claude-opus-4-8 Signed-off-by: HyeongJun An <sammiee5311@gmail.com> Link: https://patch.msgid.link/20260609124317.38046-3-sammiee5311@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
HyeongJun An [Tue, 9 Jun 2026 12:43:13 +0000 (21:43 +0900)]
ASoC: codecs: hdac_hdmi: Validate written enum value
hdac_hdmi_set_pin_port_mux() uses the written enum value to index the
texts array before calling snd_soc_dapm_put_enum_double(), which validates
that the value is within the enum item range.
An out-of-range value can therefore make the driver read past the texts
array before the helper rejects the write. Move the lookup after the helper
has accepted the value.
Mark Brown [Thu, 11 Jun 2026 19:43:33 +0000 (20:43 +0100)]
ASoC: img: Use guard() for spin locks
bui duc phuc <phucduc.bui@gmail.com> says:
This series converts spinlock handling in several IMG ASoC drivers
to use guard() helpers.
All patches are straightforward cleanups with no functional change
intended.
When SND_SOC_SOF_INTEL_LNL is set, SND_SOF_SOF_HDA_SDW_BPT must also
be enabled, in order to let the soundwire support call into it.
However, there are configurations with SND_SOF_SOF_HDA_SDW_BPT=m
and SND_SOF_SOF_HDA_SDW_BPT=m but SOUNDWIRE_INTEL=y, which still
lead to a link failure:
aarch64-linux-ld: drivers/soundwire/intel_ace2x.o: in function `intel_ace2x_bpt_wait':
intel_ace2x.c:(.text+0xfc8): undefined reference to `hda_sdw_bpt_wait'
aarch64-linux-ld: drivers/soundwire/intel_ace2x.o: in function `intel_ace2x_bpt_send_async':
intel_ace2x.c:(.text+0x1ff8): undefined reference to `hda_sdw_bpt_get_buf_size_alignment'
Address this by moving the 'select SND_SOF_SOF_HDA_SDW_BPT' into
SND_SOC_SOF_HDA_GENERIC.
Fixes: 614d416dd8ae ("ASoC: SOF: Intel: hda-sdw-bpt: fix SND_SOF_SOF_HDA_SDW_BPT dependencies") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Link: https://patch.msgid.link/20260611132310.137688-2-arnd@kernel.org Signed-off-by: Mark Brown <broonie@kernel.org>
Arnd Bergmann [Thu, 11 Jun 2026 13:23:05 +0000 (15:23 +0200)]
ASoC: SOF: Intel: select SND_SOC_SDW_UTILS=y from SND_SOC_SOF_HDA_GENERIC=y
When SND_SOC_SOF_HDA_GENERIC=y but SND_SOC_SOF_INTEL_SOUNDWIRE=m, the
SND_SOC_SDW_UTILS is also set to =m even though there is a direct link
dependency from the hda.c:
aarch64-linux-ld: sound/soc/sof/intel/hda.o: in function `hda_machine_select':
hda.c:(.text+0x21ac): undefined reference to `codec_info_list'
hda.c:(.text+0x241c): undefined reference to `asoc_sdw_get_dai_type'
hda.c:(.text+0x25b4): undefined reference to `asoc_sdw_get_codec_info_list_count'
hda.c:(.text+0x25d8): undefined reference to `asoc_sdw_get_codec_info_list_count'
Change this the same way as the other related 'select' statements
to allow linking against it.
Fixes: 2b4d53eb5cf3 ("ASoC: SOF: Intel: select SND_SOC_SDW_UTILS in SND_SOC_SOF_HDA_GENERIC") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Tested-by: Julian Braha <julianbraha@gmail.com> Link: https://patch.msgid.link/20260611132310.137688-1-arnd@kernel.org Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: cs35l56: Fix wrong error test on simple_write_to_buffer()
In cs35l56_cal_data_debugfs_write() fix the if statement that checks for
error return to only check for negative values.
Reported by Sashiko:
simple_write_to_buffer() returns the positive number of bytes copied
on success. Since the condition returns immediately on any non-zero
value, is it possible that the written calibration data is discarded
and cs35l56_stash_calibration() is never called?
ASoC: soc-core: Create device_link to ensure correct suspend order
In snd_soc_bind_card() create a device_link from card to all components
to ensure correct order of system_suspend. The card is the consumer and
the components are the supplier, so that the card will system_suspend
before any of the components.
The PM core will normally system_suspend drivers in the opposite order
that they registered. This ensures children are suspended before their
parents, for example users of a bus driver should suspend before the bus
driver suspends.
For ASoC, snd_soc_suspend() shuts down any active audio, which requires
that the components are still able to communicate with their hardware.
Previously there was nothing to ensure this ordering, because there is
(usually) no relationship between a machine driver and component drivers.
If the machine driver registered before the codec drivers, the codec
drivers would be suspended before the machine driver snd_soc_suspend()
runs, so that ASoC is attempting to stop audio on a driver that has
already suspended.
Creating a device_link is safe if there is already a device_link between
those devices because of multiple components sharing the same dev.
device_link_add() kernel doc says:
"if a device link between the given @consumer and @supplier pair
exists already when this function is called for them, the existing link
will be returned regardless of its current type and status ...
The caller of this function is then expected to treat
the link as though it has just been created, so (in particular) if
DL_FLAG_STATELESS was passed in @flags, the link needs to be released
explicitly when not needed any more"
For the same reason it is safe if the codec driver or machine driver
later call device_link_add() to create a link between the same two
devices.
(I have tested creating multiple links between the card->dev and a
component->dev and did not encounter any problems with suspend/resume or
module unloading.)
The DL_FLAG_AUTOREMOVE_* flags assume that they are being called from
the probe() function of that device. This isn't guaranteed in ASoC card
binding because of deferred binding. The exact behavior and consequences
of the DL_FLAG_AUTOREMOVE_* are also unclear from the documentation.
So DL_FLAG_STATELESS is used for safety, and the links are removed
explicitly when the card unbinds or if the bind fails.
projects / linux / log