Mirror the driver as prepared for mainline (work/linux commit
917ea639925b, "ASoC: codecs: wcd9378: add TX/capture codec driver"),
per the carry-mirrors-upstream rule. On top of the previous carry
state this brings the maintainer-eye pre-review fixes:
- latch the per-ADC sys-usage bit and target SDCA function at
PRE_PMU; POST_PMD previously recomputed them from the live input
mux and could tear down the wrong function after a mux change
- clear the requested sys_usage_mask bit when no profile matches
- drop the unreachable -EACCES carve-out on the TX runtime-PM hold
(caused a usage-count underflow in unbind)
- mark the write-1-clear FUNC_STAT registers volatile
- drop the unused is_dapm parameter, the stale sys_usage write-skip
cache, the SDW_SCP_INT1_IMPL_DEF unmask and the PS0 re-request
hack; clamp DT channel-map reads
- name the SWRS_SCP_SDCA_INTRTYPE registers; trim the PS0-failure
debug dump; Kconfig imply + help text
One deviation from the mainline patch: v7.1.2 predates
sdw_slave_wait_for_init(), so the open-coded
wait_for_completion_timeout() stays here.
Assisted-by: Claude:claude-fable-5
Signed-off-by: Jorijn van der Graaf <jorijnvdgraaf@catcrafts.net>
Measured acoustically on the FP6 with a fixed tone played through the
speakers: each TX gain code adds 1.5 dB (+6 dB per 4 codes, +30 dB over
the 0..20 range), not the 0.25 dB the TLV inherited from the wcd937x
driver family. With the correct scale, userspace volume mapping (e.g.
PulseAudio) can use the real analog range.
Assisted-by: Claude:claude-fable-5
Signed-off-by: Jorijn van der Graaf <jorijnvdgraaf@catcrafts.net>
The SDCA function engine (the SmartMIC/SmartJACK/SmartAMP sequencer
machinery activated by the FUNC_ACT class-load) dies when the TX
SoundWire bus enters clock-stop. All its registers keep their values,
so a regcache sync on resume restores nothing visible - the PDE simply
never services power-state requests again: PDE11_ACT_PS stays in PS3,
SEQ_TXn_STAT stays at pwr_dn_rdy, and even the TXn_VALID_CFG_OVR /
TXn_SEQ_TRIGGER_OVR sequencer overrides and a TX0_SEQ_SOFT_RST pulse
are ignored. Re-toggling FUNC_ACT (a real 0->1 edge on the bus) does
not revive it either; only a full codec reset does. The result was
capture recording pure digital silence: the whole DPCM -> CDC-DMA ->
TX macro -> SoundWire transport ran, but the ADC never powered.
Hold a runtime PM reference on the TX slave for as long as the codec
is bound, so the bus never clock-stops. This matches the downstream
stack, which marks the TX SoundWire master 'qcom,is-always-on' - with
full documentation available, Qualcomm made the same trade-off.
Also perform the class-load activation with plain writes instead of
update_bits so the 0->1 activation edge always reaches the hardware
regardless of regcache state.
Verified on the FP6: from a fresh boot, repeated captures across what
were previously bus suspend/resume cycles now power the sequencer every
time (PDE11 reaches PS0) and record live mic signal instead of zeros.
Assisted-by: Claude:claude-fable-5
Signed-off-by: Jorijn van der Graaf <jorijnvdgraaf@catcrafts.net>
The DAPM power-up sequence runs during snd_pcm prepare, but the BE
port graph is only started at trigger time. A codec that powers up
synchronously from a DAPM widget event and needs a running bit clock
at that point - such as aw88261 since commit caea99ac809d ("ASoC:
codecs: aw88261: remove async start") - can therefore never see a
live clock: its power-up check runs before the trigger and fails on
every stream start.
Start the graph at the end of prepare instead, mirroring what
q6afe_dai_prepare() does on the legacy stack, so the interface
clocks already run when DAPM powers up the codec. The FE side
already starts its own graph at prepare in q6apm_dai_prepare();
only the BE waited for trigger. The trigger-time start is kept as
a fallback, guarded by is_port_started.
Tested on the Fairphone (Gen. 6) - 2x aw88261 on Senary MI2S:
without this the amplifiers fail to power up with SYSST reporting
"no clock" on every stream start; with it they start synchronously,
including for the first short stream of the boot.
Assisted-by: Claude:claude-fable-5
Signed-off-by: Jorijn van der Graaf <jorijnvdgraaf@catcrafts.net>
The v7.1.2-milos base predates two aw88261 changes this branch needs:
the hw_params format-negotiation series (Val Packett's work, in broonie
for-7.2) and our power-up SYSST fix. The Senary MI2S carry drives the
two AW88261 amps in S16_LE end to end, which relies on the amplifier
negotiating its format; without it the machine driver has to force
32-bit slots. The SYSST fix is also required: aw88261_dev_start()
otherwise fails "check sysst fail" on this firmware profile because it
demands SWS/BSTS while the amp is still muted.
Backport aw88261.c/.h wholesale to the mainline state (matching
work/linux tag audio-mainline-aw88261) so this branch tracks mainline
rather than the older milos base driver. Local carry only; milos
absorbs it on the next rebase past those commits.
Assisted-by: Claude:claude-opus-4-8
Signed-off-by: Jorijn van der Graaf <jorijnvdgraaf@catcrafts.net>
Extend the clock-provider DAI fmt setup to Senary MI2S; without it
q6i2s_set_fmt() is never called, ws_src remains external and the DSP
does not drive the I2S clocks.
On the Fairphone (Gen. 6) the speaker amplifiers sit on this
interface; the board DTS enabling it is headed upstream separately
via linux-arm-msm.
Assisted-by: Claude:claude-fable-5
Signed-off-by: Jorijn van der Graaf <jorijnvdgraaf@catcrafts.net>
Replace the transport-test skeleton with a functional driver modeled on
wcd937x: platform parent device (qcom,wcd9378-codec) as component master
over the two SoundWire slaves, owning reset GPIO, supplies and micbias
config; regmap (MAPLE cache, 32-bit paged SDCA addresses) on the TX
slave; capture DAI (index 1) with sdw stream plumbing; DAPM TX path
AMICn -> ADCn MUX -> TXn SEQUENCER -> ADCn_OUTPUT with the SDCA
SmartMIC power sequence (ITxx_USAGE mode, PDE11 PS0 request, HPF init
hold) and IT11_MICB-based refcounted micbias control; sys-usage profile
auto-selection; SCP bus-clock indication (base clk, busclock scale,
host-clk-div2) per the downstream capture-start sequence.
Verified on FP6: probes and binds without any manual per-boot hacks
(gpio162 reset, runtime PM force, l8b always-on all obsolete), sound
card registers, full DPCM/SoundWire/CDC-DMA transport carries data.
KNOWN ISSUE: the SmartMIC sequencer never leaves PWR_DN (PDE11_ACT_PS
stays PS3, SEQ_TX0_STAT=PWR_DN_RDY) although every register the
downstream driver writes has been replicated and verified on hardware
by bypassed readback - capture records digital silence. Investigation
notes in journal/mic.md.
Assisted-by: Claude:claude-fable-5
Signed-off-by: Jorijn van der Graaf <jorijnvdgraaf@catcrafts.net>
Bring-up skeleton for the Qualcomm WCD9378 codec (SoundWire dev id
0x0217:0x0110, one slave per RX/TX bus). Probes both slaves, maps the
SDCA control space (32-bit paged addresses) through regmap-sdw with
prop.paging_support set, and dumps the device identity registers on
ATTACHED as a transport self-test:
DEV_MANU_ID_0/1 = 0x17/0x02 (Qualcomm 0x0217)
DEV_PART_ID_0/1 = 0x10/0x01 (WCD9378 0x0110)
ANA_TX_CH1 0x20, ANA_MICB1 0x10 (downstream reset defaults)
The analog core is WCD937x register-compatible; full codec function
(TX/ADC path first) to be built on top of this. Not for upstream in
this form.
Assisted-by: Claude:claude-fable-5
Signed-off-by: Jorijn van der Graaf <jorijnvdgraaf@catcrafts.net>
A few more fixes for this release, some smaller driver specific ones
plus a final quirk.
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Merge tag 'asoc-fix-v7.1-rc7' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v7.1
A few more fixes for this release, some smaller driver specific ones
plus a final quirk.
sdca_dev_unregister_functions() iterates over all SDCA function
descriptors and calls sdca_dev_unregister() on each func_dev without
checking for NULL. When a function registration has failed partway
through, or the device cleanup races with probe deferral, func_dev
entries may be NULL, leading to a kernel oops:
BUG: kernel NULL pointer dereference, address: 0000000000000040
RIP: 0010:device_del+0x1e/0x3e0
Call Trace:
sdca_dev_unregister_functions+0x37/0x60 [snd_soc_sdca]
release_nodes+0x35/0xb0
devres_release_all+0x90/0x100
device_unbind_cleanup+0xe/0x80
device_release_driver_internal+0x1c1/0x200
bus_remove_device+0xc6/0x130
device_del+0x161/0x3e0
device_unregister+0x17/0x60
sdw_delete_slave+0xb6/0xd0 [soundwire_bus]
sdw_bus_master_delete+0x1e/0x50 [soundwire_bus]
...
sof_probe_work+0x19/0x30 [snd_sof]
This was observed on a Lenovo ThinkPad X1 Carbon G14 (Panther Lake)
with the SOF audio driver probe failing due to missing Panther Lake
firmware, causing the subsequent cleanup of SoundWire devices to
trigger the crash.
Fix this with three changes:
1) Add a NULL guard in sdca_dev_unregister() so that callers do not
need to pre-validate the pointer (defense in depth).
2) In sdca_dev_unregister_functions(), skip NULL func_dev entries
and clear func_dev to NULL after unregistration, making the
function idempotent and safe against double-invocation.
3) In sdca_dev_register_functions(), roll back all previously
registered functions when a later one fails, so the function
array is never left in a partially-populated state.
Fixes: 4496d1c65b ("ASoC: SDCA: add function devices")
Signed-off-by: Kean Ren <rh_king@163.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260611023757.1553960-1-rh_king@163.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The "invalid position" error occurred when the DMA position descriptor
returned an invalid address value (e.g., pos = -1048838144). This happened
because the `bytes_to_frames()` function returns a signed value, but when
`addr < runtime->dma_addr`, the subtraction produces a negative result that
gets interpreted as a large unsigned integer in comparisons.
when the addr is abnormal, for example,the DMA controller is abnormal in
hardware,x=0 should not be a point(x == runtime->buffer_size),but a range,
which includes the addr address being less than runtime ->dma1-adr, and
the addr exceeding the DMA address range.the value of pos should not better
a negative，return 0, maybe better.
[ 32.834431][ 2] soc-audio soc-audio: invalid position: , pos = -1048838144
[ 32.845019][ 2] soc-audio soc-audio: invalid position: , pos = -1048838144
[ 32.855588][ 2] soc-audio soc-audio: invalid position: , pos = -1048838144
[ 32.866145][ 2] soc-audio soc-audio: invalid position: , pos = -1048838144
[ 32.995394][ 2] soc-audio soc-audio: invalid position: , pos = -1048838144
[ 33.006025][ 2] soc-audio soc-audio: invalid position: , pos = -1048838144
[ 33.016748][ 2] soc-audio soc-audio: invalid position: , pos = -1048838144
Signed-off-by: Li Jun <lijun01@kylinos.cn>
[Remove XRUN reporting I'd mistakenly avised adding on prior review -- broonie]
Link: https://patch.msgid.link/20260611010045.3668574-1-lijun01@kylinos.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Firmware will set dsp_ack to 1 when firmware sends response for the IPC
command issued by host. Similarly dsp_msg flag will be updated to 1.
During ACP D0 entry, the value read from the sof_dsp_ack_write scratch
flag can be uninitialized. A non-zero garbage value is treated as a
pending DSP IPC ack before SOF_FW_BOOT_COMPLETE, causing a spurious
"IPC reply before FW_BOOT_COMPLETE" log.
Fix the condition checks for ipc flags.
Fixes: 738a2b5e2c ("ASoC: SOF: amd: Add IPC support for ACP IP block")
Link: https://github.com/thesofproject/linux/pull/5642
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Tested-by: Umang Jain <uajain@igalia.com>
Link: https://patch.msgid.link/20260609160938.3717513-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In wm_adsp_control_remove() check that the priv pointer is not NULL
before attempting to cleanup what it points to.
When cs_dsp creates a control it calls wm_adsp_control_add_cb() so that
wm_adsp can create its own private control data. There are two cases
where private data is not created:
1. The control is a SYSTEM control, so an ALSA control is not created.
2. The codec driver has registered a control_add() callback that
hides the control, so wm_adsp_control_add() is not called.
When cs_dsp_remove destroys its control list it calls
wm_adsp_control_remove() for each control. But wm_adsp_control_remove()
was attempting to cleanup the private data pointed to by cs_ctl->priv
without checking the pointer for NULL.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 0700bc2fb9 ("ASoC: wm_adsp: Separate generic cs_dsp_coeff_ctl handling")
Link: https://patch.msgid.link/20260604101244.1402862-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
At releasing a timer object, e.g. when a userspace timer
(CONFIG_SND_UTIMER) gets closed and snd_timer_free() is called, it
tries to detach the timer instances and release the resources.
However, it's still possible that other in-flight tasks are holding
the timer instance where the to-be-deleted timer object is associated,
and this may lead to racy accesses.
Fortunately, most of ioctls dealing with the timer instance list
already have the protection with register_mutex, and this also avoids
such races. But, SNDRV_TIMER_IOCTL_PARAMS isn't protected, hence the
concurrent ioctl may lead to use-after-free.
This patch just adds the guard with register_mutex to protect
snd_timer_user_params() for covering the code path as a quick
workaround. It's no hot-path but rather a rarely issued ioctl, so the
performance penalty doesn't matter.
Reported-by: Kyle Zeng <kylebot@openai.com>
Tested-by: Kyle Zeng <kylebot@openai.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20260606161145.1933447-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When snd_timer object is freed via snd_timer_free() and still pending
snd_timer_instance objects are assigned to the timer object, it tries
to unlink all instances and just set NULL to each ti->timer, then
releases the resources immediately. The problem is, however, when
there are slave timer instances that are associated with a master
instance linked to this timer: namely, those slave instances still
point to the freed timer object although the master instance is
unlinked, which may lead to user-after-free. The bug can be easily
triggered particularly when a new userspace-driven timers
(CONFIG_SND_UTIMER) is involved, since it can create and delete the
timer object via a simple file open/close, while the other
applications may keep accessing to that timer.
This patch is an attempt to paper over the problem above: now instead
of just unlinking, call snd_timer_close[_locked]() forcibly for each
pending timer instance, so that all assigned slave timer instances are
properly detached, too. Since snd_timer_close() might be called later
by the driver that created that instance, the check of
SNDRV_TIMER_IFLG_DEAD is added at the beginning, too.
Reported-by: Kyle Zeng <kylebot@openai.com>
Tested-by: Kyle Zeng <kylebot@openai.com>
Fixes: 37745918e0 ("ALSA: timer: Introduce virtual userspace-driven timers")
Cc: stable@vger.kernel.org
Link: https://patch.msgid.link/20260606161145.1933447-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dummy sequencer port forwards events by copying an incoming
struct snd_seq_event into a stack temporary, rewriting source and
destination, and dispatching the temporary to subscribers. That legacy
event storage is smaller than struct snd_seq_ump_event.
When a UMP event reaches the dummy client, the copy leaves the UMP flag
set but only provides legacy-sized stack storage. The subscriber
delivery path then uses snd_seq_event_packet_size() and copies a
UMP-sized packet from that stack object, reading past the end of the
temporary.
Use the existing union __snd_seq_event storage and copy the packet size
reported for the incoming event before rewriting the common routing
fields. This preserves the full UMP packet for UMP events while keeping
legacy event handling unchanged.
Fixes: 32cb23a0f9 ("ALSA: seq: dummy: Allow UMP conversion")
Signed-off-by: Kyle Zeng <kylebot@openai.com>
Link: https://patch.msgid.link/20260605080204.32045-1-kylebot@openai.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Setting up the interface when suspended/resumeing fail on this card.
Adding a reset and delay quirk will eliminate this problem.
usb 1-1: new full-speed USB device number 2 using xhci-hcd
usb 1-1: New USB device found, idVendor=3c20, idProduct=3d21
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 1-1: Product: AB13X USB Audio
usb 1-1: Manufacturer: Generic
usb 1-1: SerialNumber: 20210726905926
Signed-off-by: Lianqin Hu <hulianqin@vivo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/TYUPR06MB62174610061C213260E1A992D2102@TYUPR06MB6217.apcprd06.prod.outlook.com
snd_pcm_drain() uses init_waitqueue_entry which does not clear
entry.prev/next, and add_wait_queue with a conditional
remove_wait_queue that is skipped when to_check is no longer
in the group after concurrent UNLINK. The orphaned wait entry
remains on the unlinked substream sleep queue. On the next
drain iteration, add_wait_queue adds the entry to a new queue
while still linked on the old one, corrupting both lists. A
subsequent wake_up dereferences NULL at the func pointer
(mapped from the spinlock at offset 0 of the misinterpreted
wait_queue_head_t), causing a kernel panic.
Replace init_waitqueue_entry/add_wait_queue/conditional
remove_wait_queue with init_wait_entry/prepare_to_wait/
finish_wait. init_wait_entry clears prev/next via
INIT_LIST_HEAD on each iteration and sets
autoremove_wake_function which auto-removes the entry on
wake-up. finish_wait safely handles both the already-removed
and still-queued cases.
Fixes: 9b1dbd69ba ("ALSA: pcm: fix use-after-free on linked stream runtime in snd_pcm_drain")
Signed-off-by: Ji'an Zhou <eilaimemedsnaimel@gmail.com>
Link: https://patch.msgid.link/20260604142559.3840881-1-eilaimemedsnaimel@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's only one actual fix here, for the TDM configuration on the
Freescale SAI controller, everytihng else is DMI quirks for AMD systems.
One of those is relatively large as it adds a bunch of different structs
but it's all data.
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Merge tag 'asoc-fix-v7.1-rc6' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v7.1
There's only one actual fix here, for the TDM configuration on the
Freescale SAI controller, everytihng else is DMI quirks for AMD systems.
One of those is relatively large as it adds a bunch of different structs
but it's all data.
The ASUS Vivobook 18 M1807GA (AMD ACP7.x, PCI 1022:15e2, subsystem
1043:3531) exposes a single Realtek RT721 SDCA codec on SoundWire link 1.
The BIOS reports the ACP audio config flag as 0 (SoundWire mode), so
snd_pci_ps claims the device, brings up the SoundWire managers and
enumerates the RT721 peripheral (sdw:0:1:025d:0721:01); the rt721-sdca
codec driver binds successfully.
No sound card is created, however: acp63_sdw_machine_select() walks
snd_soc_acpi_amd_acp70_sdw_machines[] and finds no entry whose declared
SoundWire peripherals are all present on the bus. The only existing RT721
entry, acp70_rt721_l1u0_tas2783x2_l1u8b, additionally requires two
TAS2783 amplifiers and deliberately exposes the RT721 as jack + DMIC
only. This M1807GA variant has no external amplifiers - the RT721's
internal AIF2 amplifier path drives the speakers - so that entry never
matches and no machine device is registered.
Add a standalone RT721 machine entry for link 1 exposing all three RT721
endpoints (jack/AIF1, speaker amplifier/AIF2, DMIC/AIF3), mirroring the
standalone RT722 configuration. Place it after the TAS2783 combo entry so
platforms that do have the external amplifiers continue to match the more
specific entry first.
ACPI _ADR of the codec: 0x000130025D072101
(link_id=1 version=3 mfg_id=0x025d Realtek part_id=0x0721 class=0x01).
Verified on the hardware: with the entry present the amd_sdw machine
binds, an "amd-soundwire" card is registered exposing the rt721-sdca
AIF1 (SimpleJack) and AIF2 (SmartAmp) PCM devices, and audio plays out
of the built-in speakers.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=221282
Signed-off-by: Aleksander Pshenitsyn <brains.fatman@gmail.com>
Link: https://patch.msgid.link/20260531101159.14241-1-brains.fatman@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The MSI Raider A18 HX A9WJG has an internal digital microphone
connected through AMD ACP6x, but this machine does not expose the
AcpDmicConnected ACPI property, so acp_yc_mach does not bind.
Add a DMI quirk for this model.
This was tested on an MSI Raider A18 HX A9WJG with board MS-182L,
BIOS E182LAMS.31A, AMD ACP6x rev 0x62, and Realtek ALC274. After
applying the quirk, the internal microphone appears as an acp6x DMIC
capture device and records correctly.
Signed-off-by: David Glushkov <david.glushkov@sntiq.com>
Link: https://patch.msgid.link/20260531214512.170716-1-david.glushkov@sntiq.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When configuring 32 slots TDM (channels == slots == 32), the xMR
(Mask Register) write used:
~0UL - ((1 << min(channels, slots)) - 1)
The literal "1" is a signed 32-bit int. Shifting it by 32 positions is
undefined behaviour which may set this register to 0xFFFFFFFF, masking
all 32 slots.
Use GENMASK_U32() macro instead. For 32 slots this produces a zero mask:
~GENMASK_U32(31, 0) = ~0xFFFFFFFF = 0x00000000
Behaviour for fewer than 32 slots is unchanged.
Fixes: 770f58d7d2 ("ASoC: fsl_sai: Support multiple data channel enable bits")
Cc: stable@vger.kernel.org
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20260601083327.1535185-1-chancel.liu@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The MSI Bravo 17 C7VF routes its internal digital microphone through
the ACP6x. The machine driver only enables the DMIC for boards present
in the DMI quirk table, so on this model the internal mic is never
detected and no capture device is created.
Add a quirk entry matching the board's DMI identifiers so the DMIC is
enabled and the internal microphone works.
Signed-off-by: João Miguel <jmiguel.ghp@gmail.com>
Link: https://patch.msgid.link/20260523213548.5219-1-jmiguel.ghp@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Lenovo Yoga Pro 7 15ASH11 with AMD RYZEN AI MAX+ 388 (Strix Halo, ACP
7.0) uses Realtek ALC287 series codec and no any DMIC connected by ACP.
All DMICs directly connet with ALC codec.
Without this quirk, Input Device of Gnome Sound settings shows Internal
Stereo Microphone and Digital Microphone by default. In fact, Digital
Microphone of ACP doesn't work due to no connecting with ALC287 codec,
the Internal Stereo Microphone as analog device based on snd_hda_intel
driver can work well.
Add a DMI quirk to override the flag to 0, consistent with the existing
entry for the HN7306EA.
Signed-off-by: Jackie Dong <xy-jackie@139.com>
Link: https://patch.msgid.link/20260527102005.58528-1-xy-jackie@139.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It makes no sense to restore the saved value for a sticky mixer, since
setting any value is a no-op.
However, in some rare cases, SET_CUR is effective despite GET_CUR always
returns a constant value. These mixers are not sticky, but there's no
way to distinguish them. Without any additional information, the best
thing we can do is to set the mixer value to the maximum before bailing
out, so that a soft mixer can still reach the maximum hardware volume if
the mixer turns out to be non-sticky. Meanwhile, all channels must be
synchronized to prevent imbalance volume.
Fixes: 86aa1ea1f1 ("ALSA: usb-audio: Do not expose sticky mixers")
Signed-off-by: Rong Zhang <i@rong.moe>
Link: https://patch.msgid.link/20260531-uac-sticky-error-path-v1-1-12c2329d17ef@rong.moe
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This round of fixes is mostly Sirini's Qualcomm cleanups that have been
in review for a while, we also have a couple of small fixes from Cássio.
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Merge tag 'asoc-fix-v7.1-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v7.1
This round of fixes is mostly Sirini's Qualcomm cleanups that have been
in review for a while, we also have a couple of small fixes from Cássio.
simple_mux_control_put() rejects values greater than e->items, but
enum control values are zero based. For the two-entry mux used by this
driver, valid values are 0 and 1, so value 2 must be rejected as well.
Accepting e->items can store an invalid mux state, pass it to the GPIO
setter, and pass it on to the DAPM mux update path where it is used as
an index into the enum text array.
Use the same >= e->items check used by the ASoC enum helpers.
Fixes: 342fbb7578 ("ASoC: add simple-mux")
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260527-asoc-simple-mux-enum-bounds-v1-1-3f805b9fc671@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Setting up the interface when suspended/resumeing fail on this card.
Adding a reset and delay quirk will eliminate this problem.
usb 1-1: new full-speed USB device number 2 using xhci-hcd
usb 1-1: New USB device found, idVendor=25aa, idProduct=600b
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 1-1: Product: TAE1159
usb 1-1: Manufacturer: Generic
usb 1-1: SerialNumber: 20210726905926
Signed-off-by: Lianqin Hu <hulianqin@vivo.com>
Link: https://patch.msgid.link/TYUPR06MB621736D7C85D43200E54E740D2082@TYUPR06MB6217.apcprd06.prod.outlook.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 21.5" Retina 4K iMac (Late 2015, DMI product name "iMac16,1") ships
with a Cirrus Logic CS4208 codec wired to an external speaker amplifier
enabled through codec GPIO0 -- the same arrangement as the late-2013
MacBookPro 11,x. Without a matching entry in cs4208_mac_fixup_tbl[] the
fixup picker logs:
snd_hda_codec_cs420x hdaudioC1D0: CS4208: picked fixup for codec SSID 106b:0000
i.e. an empty fixup name, GPIO0 stays low, the external amp is never
powered up, and the internal speakers are silent on a stock kernel.
The codec SSID reported by hardware is 0x106b:0x7f00. Reusing CS4208_MBP11
(GPIO0 + SPDIF switch fixup) makes the internal speakers and S/PDIF
output work out of the box, removing the need for users to set
`options snd_hda_intel model=mbp11` via /etc/modprobe.d/.
Tested on iMac16,1 (kernel 6.17.0): four internal drivers
(Left tweeter, Left woofer, Right tweeter, Right woofer, exposed as the
4 channels of the analog-surround-40 ALSA profile) produce audio after
the fixup is applied.
Signed-off-by: Jakub Pisarczyk <pisarz77@gmail.com>
Link: https://patch.msgid.link/20260526201830.34097-1-pisarz77@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> says:
Here is the set of patches, that fixes one of the isssue reported by
Richard Acayan, while doing fix for the reported issue, found various
other issues in the existing code.
This set contains some of those cleanups along with few trivial coding
style patches which looked uncomfortable to read.
Patch 1 should be enough to fix the issue reported.
Tested this is on UNO-Q.
Link: https://patch.msgid.link/20260518092347.3446946-1-srinivas.kandagatla@oss.qualcomm.com
The ASM_CLIENT_EVENT_DATA_WRITE_DONE case does not declare any local
variables or require a separate scope, so drop the unnecessary braces.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260518092347.3446946-5-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix error handling in q6asm_dai_compr_set_params() and q6asm_dai_prepare()
for both CMD_CLOSE and q6asm_unmap_memory_regions().
In both the functions, we are doing q6asm_audio_client_free in failure
cases, which means if prepare or set_params fail, we can never recover.
Now open and close are done in respective dai_open/close functions.
Fixes: 2a9e92d371 ("ASoC: qdsp6: q6asm: Add q6asm dai driver")
Cc: Stable@vger.kernel.org
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260518092347.3446946-4-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
q6asm_dai_close() and q6asm_dai_compr_free() currently issue CMD_CLOSE
whenever prtd->state is non-zero.
After prepare() closes an existing stream, the state is updated to
Q6ASM_STREAM_STOPPED. Since this state is also non-zero, the close and
free paths can send CMD_CLOSE again for a stream that has already been
closed.
Restrict CMD_CLOSE to the Q6ASM_STREAM_RUNNING state so the command is
sent only when the ASM stream is still active.
Fixes: 2a9e92d371 ("ASoC: qdsp6: q6asm: Add q6asm dai driver")
Cc: Stable@vger.kernel.org
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260518092347.3446946-3-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The q6asm-dai stream state is used by prepare() to decide whether an
existing stream setup needs to be closed before opening/configuring a new
one. Updating the state from trigger or asynchronous DSP callbacks can make
that state stale or incorrect relative to the actual setup lifetime.
In particular, setting Q6ASM_STREAM_STOPPED on STOP or EOS completion can
make prepare() believe there is no active setup to close, which can result
in opening/configuring the same stream more than once.
Keep stream state updates tied to prepare(), where the stream is actually
closed and reopened, and stop changing it from trigger and EOS callbacks.
Fixes: bfbb12dfa1 ("ASoC: qcom: q6asm-dai: perform correct state check before closing")
Cc: Stable@vger.kernel.org
Closes: https://lore.kernel.org/all/afS7rTHdc9TyIeLx@rdacayan/
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260518092347.3446946-2-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
byt_cht_es8316_init() enables MCLK before configuring the codec sysclk
and creating the headset jack. If either of those later steps fails, the
function returns without disabling MCLK, leaving the clock enabled after
card registration fails.
Track whether this driver enabled MCLK and disable it on the init error
paths. Add the matching DAI link exit callback so the same clock enable
is also balanced when ASoC cleans up a successfully initialized link.
Fixes: a03bdaa565 ("ASoC: Intel: add machine driver for BYT/CHT + ES8316")
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260519-asoc-bytcht-es8316-mclk-leak-v1-1-b4a11cdc2afd@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The internal mic boost on the Positivo DN140 is too high.
Fix this by applying the ALC269_FIXUP_LIMIT_INT_MIC_BOOST fixup to the machine
to limit the gain.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Link: https://patch.msgid.link/20260524185324.28959-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Firmware 2417 for the Scarlett 4th Gen 2i2 moved the direct monitor
gain parameter by 4 bytes, from offset 0x2a0 to 0x2a4, breaking the
"Direct Monitor X Mix Y" controls.
Special-case the offset in the get/set config helpers when the
running firmware is 2417 or later.
Fixes: 4e809a2996 ("ALSA: scarlett2: Add support for Solo, 2i2, and 4i4 Gen 4")
Cc: <stable@vger.kernel.org>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://patch.msgid.link/ahIWTueUlWA5xiV+@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
cs35l56_hda_read_acpi() gets an allocated ACPI _SUB string from
acpi_get_subsystem_id(). On success, that string is used to create the
firmware system name.
Several error paths after the _SUB lookup can return without releasing
the allocated string. This includes speaker ID lookup errors other than
-ENOENT, and errors after a firmware system name has been allocated.
Use scoped cleanup for the temporary _SUB string and make
cs35l56->system_name device-managed. This releases the temporary _SUB
string on every error path and lets devres release the firmware system
name on probe failure and device removal.
Fixes: 6f03b446cb ("ALSA: hda: cs35l56: Add support for speaker id")
Fixes: 40b1c2f9b2 ("ALSA: hda/cs35l56: Workaround bad dev-index on Lenovo Yoga Book 9i GenX")
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260522-alsa-cs35l56-system-name-leak-v4-1-a6154dd09cd9@gmail.com
The BIOS on the Lenovo Yoga Slim 7 14AGP11 (AMD Ryzen AI / Kraken
Point chassis; board LNVNB161216, product 83QS) programs the PCI
subsystem ID of the HDA function as 17aa:0000. As a result no entry
in alc269_fixup_tbl[] matches via SND_PCI_QUIRK, the fixup falls back
to the generic auto-routing path, and the bass speaker pin is left
mis-routed. Laptop speakers sound noticeably thin.
The codec's own internal subsystem ID register reports 0x17aa394c
correctly, so an HDA_CODEC_QUIRK entry (which matches on the codec
SSID rather than on the PCI SSID) binds the chassis to the existing
ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN fixup. This mirrors the same
workaround already in place for the closely-related Yoga 7 2-in-1
14AKP10 and 16AKP10 entries earlier in the table.
With this change the kernel log goes from
ALC287: picked fixup for PCI SSID 17aa:0000
to
ALC287: picked fixup alc287-yoga9-bass-spk-pin
and speaker routing matches what the firmware intended. Verified by
the reporter against the equivalent modprobe override
(model=,alc287-yoga9-bass-spk-pin).
Link: https://bugzilla.kernel.org/show_bug.cgi?id=221438
Signed-off-by: Kris Kater <kris@kater.nu>
Link: https://patch.msgid.link/20260522060902.9423-1-kris@kater.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The comment for the pin configuration 0x21 in the fixup
ALC299_FIXUP_PREDATOR_SPK states "use as headset mic, without its own
jack detect", but the fixup name and the actual usage indicate that the
pin is meant to be used as internal speaker. Correct the comment to
avoid confusion.
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260522060742.1384390-1-zhangheng@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
