It lost its headset microphone functionality, and this patch will restore it.
Fixes: 97272a5704 ("ALSA: hda/realtek - Fixed Headphone noise issue for Dell QCM1255")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/3c251e40a2944ae4ae479e4c2e4e63a4@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This workaround was originally added for Tegra194, but no longer gets
applied for it. Commit 615d435400 ("ALSA: hda/tegra: fix tegra-hda on
tegra30 soc") changed the guard to tegra30-hda, which worked because all
affected archs had the fallback compatible. However, commit 7f0ea5acfc
("arm64: tegra: Use correct compatible string for Tegra194 HDA") removed
the fallback compatible for Tegra194, causing this to break.
Fixes: 7f0ea5acfc ("arm64: tegra: Use correct compatible string for Tegra194 HDA")
Cc: stable@vger.kernel.org
Assisted-by: LLM
Signed-off-by: Aaron Kling <webgeek1234@gmail.com>
Link: https://patch.msgid.link/20260926-tegra194-hda-sdo-v1-1-f7a636a07a58@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
setup_card() registers the input device, whose name and phys point into
struct snd_usb_caiaqdev, and then goes on to snd_card_register() and
snd_usb_caiaq_control_init(). If either fails, snd_probe() calls
snd_card_free(), which frees the device state but leaves the input
device registered: card_free() only clears the pointer, and the input
device is unregistered from snd_disconnect() alone. Reading its "uevent"
attribute afterwards dereferences freed memory:
BUG: KASAN: slab-use-after-free in string+0x4a9/0x4f0
Read of size 1 at addr ffff8880208f5043 by task caiaq/4967
add_uevent_var+0x183/0x3a0
input_dev_uevent+0x162/0x900
dev_uevent+0x2f1/0x870
uevent_show+0x1ca/0x3a0
...
Unregister the input device on the probe error path, as
snd_disconnect() does. snd_usb_caiaq_input_disconnect() is a no-op when
no input device was registered.
Reproduced with a raw-gadget Audio Kontrol 1 and a temporary hack that
makes snd_usb_caiaq_control_init() fail.
Fixes: 28abd224db ("ALSA: caiaq: Handle probe errors properly")
Cc: stable@vger.kernel.org
Reported-by: syzbot+2a123f6269da57ffefaa@syzkaller.appspotmail.com
Closes: https://syzkaller.appspot.com/bug?extid=2a123f6269da57ffefaa
Signed-off-by: Nguyen Ngoc Thang <ngocthang2710.1999@gmail.com>
Link: https://patch.msgid.link/20260925170717.22462-1-ngocthang2710.1999@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB audio device 0db0:cc78 (Realtek ALC4080) on the MSI MAG B850M
MORTAR WIFI motherboard exposes all playback controls as "PCM", which
makes it hard to tell the outputs apart.
Its mixer layout matches the one of the MSI MPG X570S Carbon Max Wifi
(units 29, 30 and 32), so reuse msi_mpg_x570s_carbon_max_wifi_alc4080_map
for this device too. With the mapping applied, the controls show up as
"Speaker", "Front Headphone" and "IEC958".
Tested with a patched snd-usb-audio module on a 7.2.7 kernel.
Signed-off-by: Tomasz Bojanowski <tomasz@bojanowski.me>
Link: https://patch.msgid.link/20260923184957.6687-1-tomasz@bojanowski.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Higole F9B is an Alder Lake-N mini PC with a 7" touchscreen and an
ALC269VC codec (subsystem ID 0x10ec111e). Its DMI strings are all
"Default string", so the SSID is the only way to identify it.
The internal speaker is silent out of the box. The generic parser
assigns DAC 0x03 to the speaker pin 0x14 and DAC 0x02 to the headphone
pin 0x15, but the speaker only reproduces what DAC 0x02 plays, whatever
its connection selection says. Checked one path at a time, only the
gain of DAC 0x02 and the mute of pin 0x14 change what comes out of the
speaker. User space only knows the path it was told the speaker uses,
so selecting the speaker turns the path that actually reaches it all
the way down.
Route both 0x14 and 0x15 through mixer 0x0c (DAC 0x02), which is what
ALC290_FIXUP_MONO_SPEAKERS already does. Tested on the device: the
speaker plays.
Cc: <stable@vger.kernel.org>
Assisted-by: LLM
Signed-off-by: Yu-Hsiang Tseng <asas1asas200@gmail.com>
Link: https://patch.msgid.link/20260923141344.1543158-1-asas1asas200@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS ROG Strix G16 G615LR (PCI SSID 1043:3f20) outputs a loud
full-scale square wave from internal speakers since the TAS2781
amplifier driver started loading. The quirk was using
ALC287_FIXUP_TXNW2781_I2C which chains to the ThinkPad headset
jack fixup, but this is an ASUS device that should use
ALC287_FIXUP_TXNW2781_I2C_ASUS which chains to ALC294_FIXUP_ASUS_SPK
for proper EAPD and pin configuration.
Fixes: f7cede182c ("ALSA: hda/realtek: Add Asus quirk for TAS amplifiers")
Cc: stable@vger.kernel.org
Cc: Baojun Xu <baojun.xu@ti.com>
Cc: Antheas Kapenekakis <lkml@antheas.dev>
Reported-by: Aleksei Arsenev <alesharik4@gmail.com>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=222045
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260923121640.272509-1-zhangheng@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP ENVY Laptop 13-aq1xxx (PCI SSID 103c:86ad) uses an ALC285 codec
with two speaker pairs: front (pin 0x14) and bottom (pin 0x17).
The BIOS leaves pin 0x14 as [N/A] ("not connected"), so the generic
parser never enables the front pair and only the bottom speakers are
audible. The vendor driver (RTKVHD64.sys) programs that pin with the
value 0x90170150 and runs the HP amplifier initialization, which is
already implemented in the driver as ALC285_FIXUP_HP_GPIO_AMP_INIT.
Add a board fixup that applies the pin configuration for 0x14 and chains
to the existing amplifier init fixup.
The pin value was recovered from the vendor driver's own configuration
data. The patch and this changelog were prepared with an AI coding
assistant (see the Assisted-by tag below); the change was reviewed and
tested by the submitter on the affected machine.
Tested on an HP ENVY 13-aq1xxx by rebuilding snd-hda-codec-alc269 from
the Fedora 7.1.10 source and loading it: the codec now reports
"line_outs=2 (0x14/0x17)", pin 0x14 comes up as an output (Pin-ctls
0x40) with its amplifier unmuted, and both speaker pairs play.
Assisted-by: OpenCode:deepseek-v4.1-flash checkpatch
Signed-off-by: Tarun Noonemunthala <ntarun2000@gmail.com>
Link: https://patch.msgid.link/20260923094014.58867-1-ntarun2000@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASUS SupremeFX Hi-Fi (0b05:1826, 0b05:1827) does not survive USB
autosuspend. After a suspend/resume cycle the firmware degrades: HID
probes begin to fail and the descriptors it returns become corrupted,
until the device stops responding entirely and needs a physical power
cycle to recover.
Add QUIRK_FLAG_DISABLE_AUTOSUSPEND for both product IDs.
Verified on 7.2.6 through the quirk_flags module parameter, with no
code change, and with the local udev rules that had been forcing
power/control commented out, so the parameter was the only mechanism
in play. Both IDs were passed as
0b05:1827:disable_autosuspend;0b05:1826:disable_autosuspend
The device then stayed awake across 2.5 hours (runtime_suspended_time
remained 0), survived a system suspend/resume cycle, and continued to
work afterwards with no failed probes and no power cycle needed.
Signed-off-by: Roman Bolotov <hadros@ikeepitoblique.com>
Link: https://patch.msgid.link/20260923064527.434441-1-hadros@ikeepitoblique.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ThinkBook 16p Gen 3 ARH (SSID 17aa:3871) has a TI TAS2563
amplifier device on I2C, exposed by ACPI as INT8866, the same
arrangement as the Yoga 7 14ARB7 (SSID 17aa:3870). Without a quirk,
the codec falls back to the vendor-wide Lenovo quirk
(ALC269_FIXUP_LENOVO_XPAD_ACPI), the amplifier is never bound, and the
internal speakers stay silent while the headphone output works.
Reuse the Yoga 7 14ARB7 fixup, which binds the INT8866 device as a side
codec. ALC287_FIXUP_TAS2781_I2C does not work on this machine because it
looks for a TIAS2781 device.
Assisted-by: LLM
Signed-off-by: KIM MinWoo <phrimm136@gmail.com>
Link: https://patch.msgid.link/20260923013756.106033-1-phrimm136@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mute LED on the HP Pavilion Gaming Laptop 15-ec1xxx (103c:87b2)
does not work automatically. Add the ALC285_FIXUP_HP_MUTE_LED quirk
to fix this behavior.
Signed-off-by: Luis Banha <luisbanha26@gmail.com>
Link: https://patch.msgid.link/20260921230105.16844-1-luisbanha26@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP Pavilion 15 with the ALC3227 codec (SSID 103c:2164) exposes the full
four-step internal mic boost on pin 0x12: nsteps=3 with stepsize=0x2f, that is
0, 12, 24 and 36 dB. Combined with the ADC capture gain of +30 dB the analogue
cascade reaches 66 dB, and the internal microphone clips on the ambient noise of
a quiet room; the upper two steps are unusable at any useful input level. This
is the defect class ALC269_FIXUP_LIMIT_INT_MIC_BOOST exists for.
The machine matches no SSID quirk and falls through to the pin table, which
selects ALC269_FIXUP_HP_MUTE_LED_MIC1 through a pin signature shared with
several other HP models. Add an SSID entry instead, using
ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED: the existing fixup that applies
alc269_fixup_limit_int_mic_boost() and chains to the same mute LED fixup, as
already done for 103c:218b. The boost becomes 0 or 12 dB and the LED handling is
unchanged.
Tested on the affected machine by selecting the same fixup through the model
string (snd-hda-intel.model), which bypasses the SSID and pin tables, with the
driver confirming the selection:
snd_hda_codec_alc269 hdaudioC1D0: ALC3227: picked fixup limit-mic-boost (model specified)
With that, Internal Mic Boost reports Limits 0 - 1 (0 or 12 dB) instead of 0 - 3
(0 to 36 dB), so alc269_fixup_limit_int_mic_boost() clamps this codec's internal
mic pin as intended. Note that /proc/asound/card1/codec#0 keeps printing the
unclamped hardware caps (nsteps=0x03): that file reads codec parameters uncached
by design, while the fixup overrides the driver's cached parameter that the
control is built from.
This does not by itself make the default capture volume sane -- the remaining
12 dB on top of the +30 dB ADC gain is still hotter than this machine's usable
operating point. It removes the two unusable steps from the hardware range, which
is what a quirk of this class can do.
Signed-off-by: Hernan Ganzo <gp@sneemgp.com>
Link: https://patch.msgid.link/20260921205538.41192-1-gp@sneemgp.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HiBy FC4 with USB ID 32bb:0004 advertises its native DSD
stream as UAC2 RAW_DATA with a 4-byte, 32-bit subslot. Without a
quirk snd-usb-audio leaves this alternate setting as
SNDRV_PCM_FORMAT_SPECIAL, so userspace cannot select a native DSD
format.
Mark the device with QUIRK_FLAG_DSD_RAW so the existing generic raw
DSD handling exposes the stream as DSD_U32_BE.
Tested on an FC4 at DSD256. With the quirk the playback alternate
setting changes from SPECIAL to DSD_U32_BE and reports DOP=0,
bitrev=0; native DSD256 playback works correctly.
Signed-off-by: Achilles Zhang <bnqzzdf@gmail.com>
Link: https://patch.msgid.link/20260921024344.391912-1-bnqzzdf@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI SSID quirk added for the Lenovo Yoga Pro 9 16IAH10 (PCI SSID
17aa:3846, codec SSID 17aa:3920) also matches every other machine that
shares PCI SSID 17aa:3846. snd_hda_pick_fixup() tries all PCI SSID entries
before it falls back to the codec SSID, so those machines now get
ALC287_FIXUP_YOGA9_SPEAKER2_TO_DAC1 and never reach their own
codec-SSID quirk.
The Lenovo Legion Pro 7 16IRX8H (product 82WQ) is one such machine: PCI
SSID 17aa:3846, codec SSID 17aa:3884. Before this quirk it picked
ALC287_FIXUP_TAS2781_I2C via the 17aa:3884 codec SSID fallback and bound
its TAS2781 amplifier. With it, the codec never binds the amplifier and
the internal speakers are silent:
snd_hda_codec_alc269 hdaudioC0D0: ALC287: picked fixup for PCI SSID 17aa:3846
versus, on a kernel without the quirk:
snd_hda_codec_alc269 hdaudioC0D0: ALC287: picked fixup for codec SSID 17aa:3884
snd_hda_codec_alc269 hdaudioC0D0: bound i2c-TIAS2781:00 (ops tas2781_hda_comp_ops ...)
Drop the PCI SSID entry. The Yoga Pro 9 16IAH10 has the same amplifier as
the Yoga S990-16 (codec SSID 17aa:3920), so it still gets a fixup through
the codec SSID fallback, using the existing 17aa:3920 entry. Switch that
entry from ALC287_FIXUP_TXNW2781_I2C to ALC287_FIXUP_YOGA9_SPEAKER2_TO_DAC1,
which applies the DAC routing fix and then chains to
ALC287_FIXUP_TXNW2781_I2C, so the amplifier setup is unchanged.
Fixes: 41d60cbfde ("ALSA: hda/realtek: Fix bass speaker DAC routing for Lenovo Yoga Pro 9 16IAH10")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=220540
Suggested-by: Zhang Heng <zhangheng@kylinos.cn>
Signed-off-by: Matt Barr <matthewjaybarr@gmail.com>
Link: https://patch.msgid.link/20260921214812.22139-1-matthewjaybarr@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Each playback URB is handed a slice of out.buffer that is exactly
LINE6_ISO_PACKETS * max_packet_size_out bytes, sized from the OUT
endpoint, but the number of bytes written into that slice is never
compared against it.
submit_audio_out_urb() takes the frame count from prev_fsize, which
audio_in_callback() derived from the *IN* endpoint's received packet
length, and rescales it with the playback frame size; when prev_fsize is
still zero it synthesizes a length from the sample rate instead. On a
device declaring a large IN wMaxPacketSize and a small OUT
wMaxPacketSize, the resulting memcpy(), or the memset() when the
playback stream is idle, runs past its slice and, as the reproducer
below shows, beyond the allocation.
usb_submit_urb() is not a backstop. max_packet_size_out comes from
usb_maxpacket(), which returns only the low 11 bits of wMaxPacketSize,
while USB core validates a high-speed isochronous length against that
base scaled by usb_endpoint_maxp_mult(). A length of up to three times
the allocated slice is therefore accepted, and even a rejected URB is
only rejected after the write.
Reject a packet that does not fit the slice the driver allocated for it.
Clamping it instead would silently shorten the capture-derived rate
feedback and desynchronize the two streams.
BUG: KASAN: slab-out-of-bounds in submit_audio_out_urb (sound/usb/line6/playback.c:229)
Write of size 1024 at addr ffff888100bbd400 by task kworker/1:2/5002
Workqueue: events line6_startup_work
Call Trace:
__asan_memcpy (mm/kasan/shadow.c:106)
submit_audio_out_urb (sound/usb/line6/playback.c:229)
line6_submit_audio_out_all_urbs (sound/usb/line6/playback.c:291)
line6_stream_start (sound/usb/line6/pcm.c:194)
line6_pcm_acquire (sound/usb/line6/pcm.c:337)
line6_startup_work (sound/usb/line6/driver.c:728)
process_one_work (kernel/workqueue.c:3396)
worker_thread (kernel/workqueue.c:3479 kernel/workqueue.c:3560)
kthread (kernel/kthread.c:436)
ret_from_fork (arch/x86/kernel/process.c:158)
ret_from_fork_asm (arch/x86/entry/entry_64.S:245)
The buggy address belongs to the object at ffff888100bbd400
which belongs to the cache kmalloc-256 of size 256
The buggy address is located 0 bytes inside of
allocated 256-byte region [ffff888100bbd400, ffff888100bbd500)
Cc: stable@vger.kernel.org
Fixes: 7a0f55aeeb ("ALSA: line6: Support assymetrical in/out configurations")
Reported-by: <co+be8fa7ea6f77fdce@bugs.sh>
Assisted-by: LLM
Signed-off-by: Xiang Mei <xmei5@asu.edu>
Link: https://patch.msgid.link/20260918234014.1318325-1-xmei5@asu.edu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
detect_usb_format() cross-checks bSubframeSize, bBitResolution and
tSamFreq between the capture and playback interfaces, but never relates
the two endpoints' wMaxPacketSize and bNrChannels. Each playback URB
gets a buffer of ua->playback.max_packet_bytes, while the number of
bytes written into it is derived from the capture stream:
capture_urb_complete() computes frames from the received capture packet
and capture.frame_bytes, and start_usb_playback() and playback_work()
multiply that by playback.frame_bytes.
A device declaring a large capture wMaxPacketSize with few capture
channels and a small playback wMaxPacketSize with many playback channels
therefore memset()s and memcpy()s past the end of the playback buffer,
in open() of the PCM node the driver registers during probe.
usb_submit_urb() rejects the over-long iso_frame_desc[0].length with
-EMSGSIZE, but only after the write.
Reject such descriptors at probe time. Genuine UA-101/UA-1000 hardware
declares proportional packet sizes and is unaffected.
BUG: KASAN: slab-out-of-bounds in start_usb_playback (sound/usb/misc/ua101.c:586)
Write of size 2048 at addr ffff8881098f3c00 by task exploit/5021
Call Trace:
__asan_memset (mm/kasan/shadow.c:84)
start_usb_playback (sound/usb/misc/ua101.c:586)
playback_pcm_open (sound/usb/misc/ua101.c:679)
snd_pcm_open_substream (sound/core/pcm_native.c:2829)
snd_pcm_open (sound/core/pcm_native.c:2865 sound/core/pcm_native.c:2932)
snd_pcm_playback_open (sound/core/pcm_native.c:2891)
snd_open (sound/core/sound.c:166)
chrdev_open (fs/char_dev.c:411)
do_dentry_open (fs/open.c:996)
vfs_open (fs/open.c:1101)
path_openat (fs/namei.c:4837 fs/namei.c:5000)
do_file_open (fs/namei.c:5029)
do_sys_openat2 (fs/open.c:1417)
__x64_sys_openat (fs/open.c:1423 fs/open.c:1439 fs/open.c:1434)
do_syscall_64 (arch/x86/entry/syscall_64.c:61 arch/x86/entry/syscall_64.c:84)
entry_SYSCALL_64_after_hwframe (arch/x86/entry/entry_64.S:121)
The buggy address belongs to the object at ffff8881098f3c00
which belongs to the cache kmalloc-192 of size 192
The buggy address is located 0 bytes inside of
allocated 168-byte region [ffff8881098f3c00, ffff8881098f3ca8)
Cc: stable@vger.kernel.org
Fixes: 63978ab3e3 ("sound: add Edirol UA-101 support")
Reported-by: <co+24304d323d28f156@bugs.sh>
Assisted-by: LLM
Signed-off-by: Xiang Mei <xmei5@asu.edu>
Link: https://patch.msgid.link/20260918225753.1278505-1-xmei5@asu.edu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
From 2adfedabfdd9f5ec800462b1f1c0859a198e63aa Mon Sep 17 00:00:00 2001
From: Jianyu Zhou <997536907@qq.com>
Date: Thu, 17 Sep 2026 14:34:46 +0800
Subject: [PATCH v2] ALSA: hda/realtek: Add quirk for IPASON SmartBook S1
The internal speaker of the IPASON SmartBook S1 (an Emdoor ODM laptop)
stays silent under Linux, while the Bluetooth audio and the headset
output work fine. The same machine plays sound under Windows with the
vendor driver, so the hardware and the amplifier are known good.
The BIOS reports the pin default of the actual internal speaker pin
(NID 0x1b) as 0x411111f0, i.e. the default association 0xf, which means
"unconnected". The pin parser therefore drops that node already at the
pin-configuration time, hence no mixer control is ever created for it
and the pin is never driven, no matter how the output pin 0x14 is set
up. Enabling 0x1b makes the speaker work immediately.
The existing ALC256_FIXUP_HONOR_MRB_XXX_M1020_AUDIO quirk makes the
speaker work, too, but it additionally rewrites the pins 0x14, 0x19 and
0x1a with Honor-specific values, which creates a bogus "Dock Mic" device
that does not exist on this hardware. Add a minimal fixup that only
enables 0x1b as the internal speaker. Touching 0x14 is not needed:
with the pin configuration overridden at runtime, both keeping its BIOS
value (0x90170110) and disabling it (0x411111f0) leave the speaker
working, so 0x1b alone decides the output.
Codec: Realtek ALC256 (0x10ec0256), SSID 0x2782:0x0206
BIOS pincfg: 0x14 0x90170110 0x18 0x411111f0 0x19 0x02a19040
0x1a 0x411111f0 0x1b 0x411111f0 0x1d 0x4067b945
0x1e 0x411111f0 0x21 0x02214020
Tested on 7.2.0-1-MANJARO and 6.18.45-1-MANJARO, with both the SOF and
the legacy HDA driver paths.
Signed-off-by: Jianyu Zhou <997536907@qq.com>
Link: https://patch.msgid.link/tencent_3AB9214FB134949B31E9C471BC3669117F05@qq.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The native sequencer ioctl path serializes handler calls with
client->ioctl_mutex, but the translated port-info compat path invokes the
same handlers through snd_seq_kernel_client_ctl() without taking that
mutex.
This lets concurrent compat CREATE_PORT requests pass the port-count
check before any request reaches the serialized insertion. The computed
integer port index can then exceed the address field range and wrap to an
existing index. Subsequent subscriber teardown can resolve the duplicate
address to the wrong port and access a freed subscriber.
Take ioctl_mutex while dispatching converted port-info requests, matching
the native ioctl path. All translated port-info commands share this
helper, so their accesses to the client port state are serialized as
well.
Fixes: b3defb791b ("ALSA: seq: Make ioctls race-free")
Cc: stable@vger.kernel.org
Assisted-by: Codex:gpt-5.6-sol gpt-6-astra
Signed-off-by: Kyle Zeng <kylebot@openai.com>
Signed-off-by: Bruno Produit <bruno.produit@trailofbits.com>
Link: https://patch.msgid.link/20260917105643.90102-1-bruno.produit@trailofbits.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A relatively large pile of fixes here, a lot of driver specific stuff
that's broadly unremarkable plus a few core fixes from Richard that fix
issues where SoundWire systems with multiple CODECs on the same link
would configure the CODECs to use the same bus slots leading to broken
audio.
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Merge tag 'asoc-fix-v7.3-rc3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v7.3
A relatively large pile of fixes here, a lot of driver specific stuff
that's broadly unremarkable plus a few core fixes from Richard that fix
issues where SoundWire systems with multiple CODECs on the same link
would configure the CODECs to use the same bus slots leading to broken
audio.
Like for SPI, the I2C driver needs an OF match table for the kernel to
be able to automatically load the driver when built as a module. Add
one.
Signed-off-by: Alvin Šipraga <alvin.sipraga@analog.com>
Reviewed-by: Nuno Sá <nuno.sa@analog.com>
Link: https://patch.msgid.link/20260914-asoc-adau1977-fixes-v1-3-aa2f0cabd728@analog.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Since commit 5ab23c7923 ("modpost: Create modalias for builtin
modules") MODULE_DEVICE_TABLE() is enough to reference a match table and
the data isn't discarded by the linker even when the driver is built-in
and CONFIG_OF is disabled. Drop the of_match_ptr() wrapping so that OF
matching keeps working regardless of CONFIG_OF. This also means we can
drop __maybe_unused since it's always used.
The entries in adau1977_spi_of_match were also erroneously indented with
spaces - replace the indentation with tabs to conform with coding style.
Signed-off-by: Alvin Šipraga <alvin.sipraga@analog.com>
Reviewed-by: Nuno Sá <nuno.sa@analog.com>
Link: https://patch.msgid.link/20260914-asoc-adau1977-fixes-v1-2-aa2f0cabd728@analog.com
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_ADAU1977_{SPI,I2C} are missing Kconfig text, so they don't show
up in menuconfig and can't be selected by a user - only by another
symbol such as a machine driver. Add the text to make these symbols
selectable and usable with generic machine drivers like the simple audio
card.
Signed-off-by: Alvin Šipraga <alvin.sipraga@analog.com>
Reviewed-by: Nuno Sá <nuno.sa@analog.com>
Link: https://patch.msgid.link/20260914-asoc-adau1977-fixes-v1-1-aa2f0cabd728@analog.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Vijendar Mukunda <Vijendar.Mukunda@amd.com> says:
This series fixes four defects in the AMD ACP SoundWire machine drivers
(acp-sdw-legacy-mach.c and acp-sdw-sof-mach.c).
A bounds check is added to validate the SoundWire link ID before it is
used as an array index in create_sdw_dailink(), preventing out-of-bounds
access when an unexpected link_mask value is encountered. The codec
config count in the SOF machine driver is refactored to use a dedicated
variable rather than reusing the endpoint-count variable for two
purposes, making the intent clearer and avoiding a stale value being
passed to the codec config array. An operator-precedence bug in the
ffs(link_mask - 1) expression is corrected to ffs(link_mask) - 1,
ensuring the link ID is derived from the correct bit position. Finally,
the SOF machine driver card name is shortened to fit within the 16-byte
snd_card driver[] field and eliminate a compile-time warning.
Link: https://patch.msgid.link/20260910161728.1452808-1-Vijendar.Mukunda@amd.com
The ALSA snd_card driver[] field is 16 bytes (including the NUL
terminator), leaving 15 usable characters. The SOF framework
prepends a "sof-" prefix when registering the card, so
card->name = "amd-soundwire" becomes driver name "sof-amd-soundwire"
which is 17 characters and overflows the driver[16] buffer, triggering
a kernel warning.
Fix by shortening the card name to "amd-sdw"; the resulting driver
name "sof-amd-sdw" fits within the 15-character limit.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Mario Limonciello (AMD) <superm1@kernel.org>
Link: https://patch.msgid.link/20260910161728.1452808-5-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ffs(link_mask - 1) computes ffs on (link_mask - 1) instead of
subtracting 1 from the result of ffs(link_mask). For a typical
power-of-2 link_mask this returns the wrong link ID, causing cpu_pin_id
lookup to select the incorrect SoundWire manager.
Fix the operator precedence to ffs(link_mask) - 1 in both
acp-sdw-sof-mach.c and acp-sdw-legacy-mach.c.
Fixes: 6d8348ddc5 ("ASoC: amd: acp: refactor SoundWire machine driver code")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Mario Limonciello (AMD) <superm1@kernel.org>
Link: https://patch.msgid.link/20260910161728.1452808-4-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
num_devs was used both as the endpoint count and as the output for
asoc_sdw_parse_sdw_endpoints(), which overwrites it with the codec
configuration count. Introduce a separate num_confs variable to hold
the codec conf count so the two values remain distinct across
codec_conf allocation and card->num_configs assignment.
Fixes: 6d8348ddc5 ("ASoC: amd: acp: refactor SoundWire machine driver code")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Mario Limonciello (AMD) <superm1@kernel.org>
Link: https://patch.msgid.link/20260910161728.1452808-3-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a bounds check in create_sdw_dailink() to validate that the
SoundWire link ID derived from link_mask does not exceed the maximum
supported by the platform. If the link ID is out of range or link_mask
is zero, log an error and return -EINVAL to prevent accessing invalid
CPU pin ID tables.
Applied to both acp-sdw-sof-mach.c and acp-sdw-legacy-mach.c.
Fixes: 6d8348ddc5 ("ASoC: amd: acp: refactor SoundWire machine driver code")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Mario Limonciello (AMD) <superm1@kernel.org>
Link: https://patch.msgid.link/20260910161728.1452808-2-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In cs_amp_alloc_get_efi_variable() the first call to
cs_amp_get_efi_variable() might return EFI_SUCCESS if the variable
exists with zero length. Trap this and return -ENOENT to prevent
returning an unexpected NULL pointer.
The first cs_amp_get_efi_variable() call was assumed to return
EFI_BUFFER_TOO_SMALL if the variable existed, but if instead it
returned EFI_SUCCESS this would be converted to 0 by
cs_amp_convert_efi_status() and then be returned as a NULL pointer.
Fixes: 00fd40bc7a ("ASoC: cs-amp-lib: Support Dell SSIDExV2 UEFI variable")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260914122611.2783563-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
stream_config is not initialized before being passed to
sdw_stream_add_slave(). The type field may contain garbage and is
later copied to stream->type by sdw_config_stream().
Zero-initialize stream_config so type defaults to SDW_STREAM_PCM.
While at it, use snd_sdw_params_to_config() helper instead of
open-coding the same logic.
Fixes: 63a511284c ("ASoC: rt712-sdca: Add RT712 SDCA driver for Mic topology")
Cc: stable@vger.kernel.org
Signed-off-by: Jiangshan Yi <yijiangshan@kylinos.cn>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Link: https://patch.msgid.link/20260914104712.379574-1-yijiangshan@kylinos.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
The put() callback of "IEC958 Playback Default" stores all 24 channel
status bytes and then returns 0. The core notifies userspace only on a
positive return, so a write that changes what the get() callback hands
back is never announced, and a mixer holding the control open keeps
showing the old value.
Compare the stored bytes and return 1 when they move, the way
snd_hda_spdif_default_put() does.
The same shape is in img-spdif-out and uniperif_player.
No board with this codec was to hand. The change is a comparison of
driver state with no hardware behaviour in it, and mixer-test counts the
missing notification as event_missing.
Fixes: 7a8e1d4421 ("ASoC: hdmi-codec: Add iec958 controls")
Signed-off-by: HyeongJun An <sammiee5311@gmail.com>
Assisted-by: Claude:claude-opus-5
Link: https://patch.msgid.link/20260915092515.2638542-1-sammiee5311@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
arm allmodconfig fails to build with gcc:
In file included from sound/soc/ux500/ux500_msp_i2s.c:20:
sound/soc/ux500/ux500_msp_i2s.h:151:38: error: suggest parentheses
around arithmetic in operand of '^' [-Werror=parentheses]
sound/soc/ux500/ux500_msp_i2s.c:204:21: note: in expansion of macro
'MSP_TX_CLKPOL_BIT'
cc1: all warnings being treated as errors
The macros never parenthesized their argument:
#define MSP_TX_CLKPOL_BIT(n) ((n & TCKPOL_MASK) << TCKPOL_SHIFT)
That went unnoticed while every caller passed a plain variable, but
configure_protocol() now passes an XOR expression, which binds as
"a ^ (b & MASK)" rather than "(a ^ b) & MASK", and gcc rightly
complains.
No functional change: tx_clk_pol and rx_clk_pol only ever hold
MSP_FALLING_EDGE (0) or MSP_RISING_EDGE (1), and bclk_inverted is a
bool, so masking before or after the XOR gives the same 0/1 result.
Parenthesize the argument anyway - it fixes the build and stops the
macros from silently mis-evaluating a future composite argument.
Fixes: 9ccbacf5a0 ("ASoC: ux500: Validate MSP DAI configuration")
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202609051547.G9SJp8UQ-lkp@intel.com/
Assisted-by: LLM
Signed-off-by: Sasha Levin <sashal@kernel.org>
Reviewed-by: Linus Walleij <linusw@kernel.org>
Link: https://patch.msgid.link/20260913173132.1172003-1-sashal@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Link: https://patch.msgid.link/20260916101803.2301508-1-oder_chiou@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
stop_urbs() clears ep->ready_playback_urbs with a bare INIT_LIST_HEAD()
instead of unlinking each queued snd_urb_ctx. If a URB survives past
wait_clear_urbs()'s forced STOPPING->STOPPED timeout, its ctx is left
looking "linked" (stale next/prev) even though the list head has
forgotten it. When the endpoint later restarts and re-queues that same
ctx onto the (now real) ready list, and the old URB's completion
handler then calls push_back_to_ready_list() for it a second time, the
ctx is still the list's own tail and list_add's double-add check trips:
kernel BUG at lib/list_debug.c:35 (list_add double add)
Guard push_back_to_ready_list() with a list_empty() check so a
still-linked ctx isn't re-added, and make stop_urbs() actually unlink
each ctx via list_del_init() instead of only resetting the head, so a
dropped ctx doesn't keep looking linked to that guard.
Reported-by: syzbot+9fe3b8d9f5c64ff410a7@syzkaller.appspotmail.com
Closes: https://syzkaller.appspot.com/bug?extid=9fe3b8d9f5c64ff410a7
Signed-off-by: Nguyen Ngoc Thang <ngocthang2710.1999@gmail.com>
Link: https://patch.msgid.link/20260915163110.58124-1-ngocthang2710.1999@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stream assignment can fail when hardware streams are exhausted.
Move the tracepoint after the NULL check because its payload accesses
the assigned stream tag.
Detected by static analysis and reviewed with AI-assisted source auditing.
Fixes: 184865085b ("ALSA: hda - rename hda_intel_trace.h to hda_controller_trace.h")
Assisted-by: LLM
Signed-off-by: Slavin Liu <bolin.liu@seu.edu.cn>
Link: https://patch.msgid.link/20260913125154.109944-1-bolin.liu@seu.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Skip the nonfunctional master mute control on the AVerMedia Live Gamer
ULTRA S GC553Pro (07ca:1553).
USB tracing shows that GET_CUR returns zero bytes instead of the required
one-byte value, both through usbfs and during ALSA initialization.
SET_CUR succeeds, but switching capture off does not mute HDMI audio.
Before the change, the driver exposed a misleading PCM Capture Switch
and logged: 3:2: failed to get current value for ch 0 (-22)
With the patch applied, the switch and warning are absent. A ten-second
sound recording through PipeWire confirmed that stereo 48 kHz, 16-bit
capture still works.
Tested on NixOS with the patched 7.3.0-rc3 kernel. The USB audio driver
object builds with Clang and W=1; sparse and strict checkpatch pass.
Signed-off-by: Asai Neko <sugar@sne.moe>
Link: https://patch.msgid.link/20260914-avermedia-gc553pro-alsa-v1-1-4c694e8b0cd5@sne.moe
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP OmniBook 7 Laptop 17-dc0xxx (SSID 103c:8d9c) has mute and
mic-mute LEDs on its F6 and F9 keys, but neither lights up. Its quirk
entry only sets up the two CS35L41 amplifiers, so no LED control is
registered for either key.
Writing to the ALC245 by hand with hda-verb, the mute LED responds to
COEF 0x0b bits 2-3 and the mic-mute LED to GPIO 0x04, lit when the pin
is low. That appears to match what ALC245_FIXUP_HP_X360_MUTE_LEDS
already does, so add a fixup that chains the two-amp I2C setup to it
and use it for this model.
Tested on 7.2.4 with the patched module: hda::mute and hda::micmute
are registered, and both LEDs follow the speaker and microphone mute
state, including from the F6 and F9 keys.
Signed-off-by: Jaeho Cho <jaeho2025@gmail.com>
Link: https://patch.msgid.link/20260914142445.3476212-1-jaeho2025@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb6fire_pcm_in_urb_handler() sizes each outgoing isochronous packet as
(actual_length - 4) / (in_n_analog << 2) * (out_n_analog << 2) + 4, where
actual_length is the unsigned length the device reported for the matching
IN packet. A packet completed with status 0 and actual_length < 4 wraps
the subtraction to 0x7fffffec; a zero-length isochronous packet is legal
on the bus, and the preceding loop rejects only non-zero status. The sum
reaches memset() on out_urb->buffer, a 4832-byte object from
kcalloc(PCM_MAX_PACKET_SIZE, PCM_N_PACKETS_PER_URB).
Even without the wrap the result is out of bounds: at 88.2/96 kHz the
4-in/6-out scaling turns a full 420-byte IN packet into 628, so eight
packets span 5024 bytes of that buffer. usb_submit_urb() rejects an
over-long descriptor only after the memset() and the
usb6fire_pcm_playback() copy of user PCM data have run.
Guard the subtraction as the sibling usb6fire_pcm_capture() already does,
and limit the frame count to what fits in rt->out_packet_size, the OUT
endpoint's wMaxPacketSize. This bounds total_length by the buffer size
while keeping each packet length aligned to a whole output frame.
BUG: KASAN: out-of-bounds in usb6fire_pcm_in_urb_handler (sound/usb/6fire/pcm.c:338)
Write of size 18446744073709551456 at addr ffff88802a3d0000 by task vhci_rx/5018
Call Trace:
dump_stack_lvl (lib/dump_stack.c:94 lib/dump_stack.c:120)
print_report (mm/kasan/report.c:378 mm/kasan/report.c:482)
kasan_report (mm/kasan/report.c:595)
kasan_check_range (mm/kasan/generic.c:186 mm/kasan/generic.c:200)
__asan_memset (mm/kasan/shadow.c:84)
usb6fire_pcm_in_urb_handler (sound/usb/6fire/pcm.c:338)
__usb_hcd_giveback_urb (drivers/usb/core/hcd.c:1657)
usb_hcd_giveback_urb (drivers/usb/core/hcd.c:1741)
vhci_rx_loop (drivers/usb/usbip/vhci_rx.c:107 drivers/usb/usbip/vhci_rx.c:242)
kthread (kernel/kthread.c:436)
ret_from_fork (arch/x86/kernel/process.c:158)
ret_from_fork_asm (arch/x86/entry/entry_64.S:245)
Allocated by task 10:
__kmalloc_cache_noprof (mm/slub.c:5563)
usb6fire_pcm_init (sound/usb/6fire/pcm.c:560 sound/usb/6fire/pcm.c:595)
usb6fire_chip_probe (sound/usb/6fire/chip.c:133)
usb_probe_interface (drivers/usb/core/driver.c:399)
The buggy address belongs to the object at ffff88802a3d0000
which belongs to the cache kmalloc-8k of size 8192
The buggy address is located 0 bytes inside of
4832-byte region [ffff88802a3d0000, ffff88802a3d12e0)
Kernel panic - not syncing: Fatal exception in interrupt
Fixes: c6d43ba816 ("ALSA: usb/6fire - Driver for TerraTec DMX 6Fire USB")
Reported-by: co+855929c2df672879@bugs.sh
Closes: https://lore.kernel.org/all/gisnub8aWGLbyZLcDCSc7zWsHonMWGcyRgt5%40bugs.sh/
Assisted-by: LLM
Signed-off-by: Xiang Mei <xmei5@asu.edu>
Link: https://patch.msgid.link/20260914074324.3590843-1-xmei5@asu.edu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Behringer FCA1616 (1397:0004) returns silent capture samples unless its
playback endpoint is active.
Use the existing fixed implicit-feedback mechanism to keep playback endpoint
0x01 on interface 1 active during capture.
Tested with 16-channel S32_LE capture at 44.1 and 48 kHz.
Signed-off-by: Kitty Makin <autumnull@posteo.net>
Link: https://patch.msgid.link/20260914002334.12691-1-autumnull@posteo.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the HP OMEN 15 ax-202nf, the keyboard mute LED is exposed through
NID 0x1b rather than 0x18. This reuses the existing quirk
(ALC269_FIXUP_HP_MUTE_LED_MIC3) to allow the keyboard LED to reflect
the built-in speaker mute state.
Tested on HP OMEN 15 ax-202nf with Realtek ALC295:
- hda::mute/brightness properly follows mute state
- mute/unmute via keyboard shortcut or via GUI volume control
- state is kept on suspend & resume, and reboot
- plugging a 3.5mm jack headset reflects the headset mute status
- unplugging reverts the LED to the speaker mute status
- USB/Bluetooth headsets are not covered
Signed-off-by: Xavier Goffin <xaviergoffin42@gmail.com>
Link: https://patch.msgid.link/20260913211155.20305-1-xaviergoffin42@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add DMI entry for Huawei Matebook B3-420 (BDZ-WXX9) with HEADPHONE_GPIO
and HEADSET_MIC1 quirks.
Similar to Huawei Matebook D (BOD-WXX9).
On the same machine,audio routing between speakers and headphones works
correctly when running Windows with the Huawei audio driver.
However, after reinstalling Linux, both the speakers and headphones output
sound simultaneously,indicating that the amplifier enable GPIOs are not
being toggled correctly to separate the two outputs.
Signed-off-by: Ai Chao <aichao@kylinos.cn>
Link: https://patch.msgid.link/20260911081932.2605407-1-aichao@kylinos.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Richard Fitzgerald <rf@opensource.cirrus.com> says:
Struct snd_soc_dai_link_ch_map had a single mask member to set the
CPU channel masks. But no fixup was done to the codec end of the link.
For example if a 4-channel CPU capture DAI was made from two codecs both
supplying 2 channels, the hw_params() of the codec would be passed a
channel count of 4.
On SoundWire this could cause multiple codecs to send data in the same
bits of a frame because the unused channels were not disabled.
The changes in this series are:
- Separate channel masks for CPU and codec in struct
snd_soc_dai_link_ch_map .
- Apply the codec channel mask as a channel count fixup if the machine
drive has not set a TDM mask.
- Set the codec channel mask in the SoundWire machine driver.
- Remove the workaround from the cs_amp machine driver.
Link: https://patch.msgid.link/20260910114500.1586637-1-rf@opensource.cirrus.com
Delete the asoc_sdw_cs_spk_feedback_rtd_init(). This is not needed now
that the ASoC bug it was working around has been fixed. And it was broken
anyway because it didn't match the way the core SoundWire code mapped
codec channels to frame bitslots.
This code was added to avoid a problem where multiple codec DP outputs
were mapped to the same SoundWire frame bit slot. This would allow a
user to break the SoundWire bus just by enabling mixer outputs using
ALSA controls.
As no production system has used the capture stream, this workaround
was of little consequence and the problem of conflicting DP mappings
was not investigated.
The ASoC bug that enabled too many channels on each codec has now been
fixed. So this workaround can be completely deleted.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260910114500.1586637-6-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In asoc_sdw_hw_params() set the codec_ch_mask member of struct
snd_soc_dai_link_ch_map for capture streams. ASoC will then pass the
correct number of channels to each codec hw_params(). This prevents
trying to enable more channels on the codec DP than have been allocated
bitslots in the SoundWire frame, which would cause bus clash errors.
In theory codec_ch_mask could also be set for playback streams, but for
those the CPU is the only sender so there is no risk of bus clash.
For playback streams codec_ch_mask is set to 0 to preserve the existing
behavior and avoid introducing bugs.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260910114500.1586637-5-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In __soc_pcm_hw_params() if there is a snd_soc_dai_link_ch_map with
non-zero codec_ch_mask, use that channel mask to restrict which channels
are enabled on the codec. But only if there isn't a TDM mask.
It is possible that a snd_soc_dai_link_ch_map could include the same codec
multiple times on different CPUs so the for_each_rtd_ch_maps() loop
accumulates the channel masks for all entries of that codec.
If a TDM mask was also set, it takes priority and is used instead of any
possible snd_soc_dai_link_ch_map entries. (They cannot be ANDed together
because the bit positions are indicating different things: TDM is a bit
for each TDM slot, codec_ch_mask is a bit for each codec channel.)
This fixes a problem of incorrect TX channels enabled on the codec when
multiple codecs are aggregated on a single capture link. For example:
- Two CPUs with six 4-channel codecs.
- The machine driver chooses to assign one channel from each codec to
one channel on the CPU
- But the codec hw_params() would be passed a channel count of 6, which
(a) is more channels than the codec has and (b) allows enabling channels
that should not be driving the audio bus.
Fixes: ac950278b0 ("ASoC: add N cpus to M codecs dai link support")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260910114500.1586637-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a codec_ch_mask member to snd_soc_dai_link_ch_map.
The CPU and codec channel masks are not necessarily the same, and are
quite likely different. SoundWire and I2S/TDM both support assigning
different sample slots to each codec, so for example channel 0 on each
codec could map to different channels at the CPU.
It is also possible for one TX channel to map to multiple RX channels.
So it isn't _always_ safe to assume that the total number of set bits
in the CPU ch_mask is the same as the total number of enabled channels
on the codec.
For example consider this mapping on a capture stream:
CPU0 CODEC0 cpu_ch_mask = 0x03
CPU1 CODEC0 cpu_ch_mask = 0x03
This could be either four TX channels on the codec split across two
receiving CPUs, or two TX channels on the codec duplicated to two CPUs.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260910114500.1586637-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Rename the ch_mask member of snd_soc_dai_link_ch_map to cpu_ch_mask,
as that is what it is used for.
The CPU and codec channel masks are not necessarily the same, and are
quite likely different. SoundWire and I2S/TDM both support assigning
different sample slots to each codec, so for example channel 0 on each
codec could map to different channels at the CPU. So it's quite normal
that the channel mask at the CPU end is different for each codec, but
the codec channel masks are the same for each codec.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260910114500.1586637-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add pll2 reconfiguration sequence in order to fix calibration
time-out issue and to support 24.576MHz MCLK on specific platforms.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://patch.msgid.link/20260909085449.862350-1-jack.yu@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In wm_adsp_request_firmware_file() only log the "Failed to request
FILENAME" message when there is a real error (not when the file is
missing). Add a new debug message to log the sequence of filenames tried
during the file search.
People have enabled debug messages, seen the "Failed to request" messages
that are only logging the normal file search sequence, and reported them
as errors.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260910121047.1592541-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
