| Age | Commit message (Collapse) | Author | Files | Lines |
|---|
|
[ Upstream commit 829ee558f3527fd602c6e2e9f270959d1de09fe0 ]
ASUS VivoBook X515UA with PCI SSID 1043:106f had a default quirk
pickup via pin table that applies ALC256_FIXUP_ASUS_MIC, but this adds
a bogus built-in mic pin 0x13 enabled. This was no big problem
because the pin 0x13 was assigned as the secondary mic, but the recent
fix made the entries sorted, hence this bogus pin appeared now as the
primary input and it broke.
For fixing the bug, put the right quirk entry for this device pointing
to ALC256_FIXUP_ASUS_MIC_NO_PRESENCE.
Fixes: 3b4309546b48 ("ALSA: hda: Fix headset detection failure due to unstable sort")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=219897
Link: https://patch.msgid.link/20250324153233.21195-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 3424c8f53bc63c87712a7fc22dc13d0cc85fb0d6 ]
The infamous mmap_lock taken in copy_from/to_user() can be often
problematic when it's called inside another mutex, as they might lead
to deadlocks.
In the case of ALSA timer code, the bad pattern is with
guard(mutex)(®ister_mutex) that covers copy_from/to_user() -- which
was mistakenly introduced at converting to guard(), and it had been
carefully worked around in the past.
This patch fixes those pieces simply by moving copy_from/to_user() out
of the register mutex lock again.
Fixes: 3923de04c817 ("ALSA: pcm: oss: Use guard() for setup")
Reported-by: syzbot+2b96f44164236dda0f3b@syzkaller.appspotmail.com
Closes: https://lore.kernel.org/67dd86c8.050a0220.25ae54.0059.GAE@google.com
Link: https://patch.msgid.link/20250321172653.14310-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 38e94cefbf45c1edc5b751ab90a3088f7c6fac1a ]
Mixer quicks for the Pioneer DJM-A9 mixer was added in 5289d00 with
additional capture level values added to the common DJM array of values.
This breaks the existing DJM mixers however as alsa-utils relies on
enumeration of the actual mixer options based on the value array which
results in error when storing state.
This commit just separates the A9 values into a separate array and
references them in the corresponding mixer control.
Fixes: 5289d0069639 ("ALSA: usb-audio: Add Pioneer DJ/AlphaTheta DJM-A9 Mixer")
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Link: https://patch.msgid.link/20250316153323.16381-1-livvy@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 45ff65e30deb919604e68faed156ad96ce7474d9 ]
For 'ti,j7200-cpb-audio' compatible, there is support for only one PLL for
48k. For 11025, 22050, 44100 and 88200 sampling rates, due to absence of
J721E_CLK_PARENT_44100, we get EINVAL while running any audio application.
Add support for these rates by using the 48k parent clock and adjusting
the clock for these rates later in j721e_configure_refclk.
Fixes: 6748d0559059 ("ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI)")
Signed-off-by: Jayesh Choudhary <j-choudhary@ti.com>
Link: https://patch.msgid.link/20250318113524.57100-1-j-choudhary@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit f1d742c35b659fb0122da0a8ff09ad9309cb29d8 ]
ADX startup() callback uses atomic poll timeout on ADX status register.
This is unnecessary because:
- The startup() callback itself is non-atomic.
- The subsequent timeout call in the same function already uses a
non-atomic version.
Using atomic version can hog CPU when it is not really needed,
so replace it with non-atomic version.
Fixes: a99ab6f395a9e ("ASoC: tegra: Add Tegra210 based ADX driver")
Signed-off-by: Ritu Chaudhary <rituc@nvidia.com>
Signed-off-by: Sheetal <sheetal@nvidia.com>
Link: https://patch.msgid.link/20250311062010.33412-1-sheetal@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 5a0c72c1da3cbc0cd4940a95d1be2830104c6edf ]
The workaround for Dell machines to skip the pin-shutup for mic pins
introduced alc_headset_mic_no_shutup() that is replaced from the
generic snd_hda_shutup_pins() for certain codecs. The problem is that
the call is done unconditionally even if spec->no_shutup_pins is set.
This seems causing problems on other platforms like Lenovo.
This patch corrects the behavior and the driver honors always
spec->no_shutup_pins flag and skips alc_headset_mic_no_shutup() if
it's set.
Fixes: dad3197da7a3 ("ALSA: hda/realtek - Fixup headphone noise via runtime suspend")
Reported-and-tested-by: Oleg Gorobets <oleg.goro@gmail.com>
Link: https://patch.msgid.link/20250315143020.27184-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit de74ec718e0788e1998eb7289ad07970e27cae27 ]
commit 419d1918105e ("ASoC: simple-card-utils: use __free(device_node) for
device node") uses __free(device_node) for dlc->of_node, but we need to
keep it while driver is in use.
Don't use __free(device_node) in graph_util_parse_dai().
Fixes: 419d1918105e ("ASoC: simple-card-utils: use __free(device_node) for device node")
Reported-by: Thuan Nguyen <thuan.nguyen-hong@banvien.com.vn>
Reported-by: Detlev Casanova <detlev.casanova@collabora.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Thuan Nguyen <thuan.nguyen-hong@banvien.com.vn>
Tested-by: Detlev Casanova <detlev.casanova@collabora.com>
Link: https://patch.msgid.link/87eczisyhh.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 02e1cf7a352a3ba5f768849f2b4fcaaaa19f89e3 ]
Add condition check to register ACP PDM sound card by reading
_WOV acpi entry.
Fixes: 09068d624c49 ("ASoC: amd: acp: fix for acp platform device creation failure")
Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu@amd.com>
Link: https://patch.msgid.link/20250310183201.11979-15-venkataprasad.potturu@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit ad5a0970f86d82e39ebd06d45a1f7aa48a1316f8 ]
Currently the return value from spi_setup() is not checked for a failure.
It is unlikely it will ever fail in this particular case but it is still
better to add this check for the sake of completeness and correctness. This
is cheap since it is performed once when the device is being probed.
Handle spi_setup() return value.
Found by Linux Verification Center (linuxtesting.org) with Svace.
Fixes: 872fc0b6bde8 ("ASoC: cs35l41: Set the max SPI speed for the whole device")
Signed-off-by: Vitaliy Shevtsov <v.shevtsov@mt-integration.ru>
Link: https://patch.msgid.link/20250304115643.2748-1-v.shevtsov@mt-integration.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
commit be8cd366beb80c709adbc7688ee72750f5aee3ff upstream.
This patch applies the ALC294 bass speaker fixup (ALC294_FIXUP_BASS_SPEAKER_15),
previously introduced in commit a7df7f909cec ("ALSA: hda: improve bass
speaker support for ASUS Zenbook UM5606WA"), to the ASUS Zenbook UM5606KA.
This hardware configuration matches ASUS Zenbook UM5606WA, where DAC NID
0x06 was removed from the bass speaker (NID 0x15), routing both speaker
pins to DAC NID 0x03.
This resolves the bass speaker routing issue, ensuring correct audio
output on ASUS UM5606KA.
Signed-off-by: Andres Traumann <andres.traumann.01@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250325102535.8172-1-andres.traumann.01@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit 35ef1c79d2e09e9e5a66e28a66fe0df4368b0f3d upstream.
The mute LED on this HP laptop uses ALC236 and requires a quirk to function.
This patch enables the existing quirk for the device.
Tested on my laptop and the LED behaviour works as intended.
Cc: stable@vger.kernel.org
Signed-off-by: Dhruv Deshpande <dhrv.d@proton.me>
Link: https://patch.msgid.link/20250317085621.45056-1-dhrv.d@proton.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit 486f6205c233da1baa309bde5f634eb1f8319a33 upstream.
Many Poly/Plantronics headset families name the feature, input,
and/or output units in a such a way to produce control names
that are not recognized by user space. As such, the volume and
mute events do not get routed to the headset's audio controls.
As an example from a product family:
The microphone mute control is named
Headset Microphone Capture Switch
and the headset volume control is named
Headset Earphone Playback Volume
The quirk fixes these to become
Headset Capture Switch
Headset Playback Volume
Signed-off-by: Terry Junge <linuxhid@cosmicgizmosystems.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Cc: stable@vger.kernel.org
Signed-off-by: Jiri Kosina <jkosina@suse.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.14
The bulk of this is driver specific fixes, mostly unremarkable. There's
also one core fix from Charles, fixing up confusion around the limiting
of maximum control values.
|
|
The custom suspend function causes a build warning when CONFIG_PM_SLEEP
is disabled:
sound/soc/codecs/cs42l43.c:2405:12: error: unused function 'cs42l43_codec_runtime_force_suspend' [-Werror,-Wunused-function]
Change SET_SYSTEM_SLEEP_PM_OPS() to the newer SYSTEM_SLEEP_PM_OPS(),
to avoid this.
Fixes: 164b7dd4546b ("ASoC: cs42l43: Add jack delay debounce after suspend")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Maciej Strozek <mstrozek@opensource.cirrus.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20250305172738.3437513-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Add a fixup to enable the mute LED on HP Pavilion x360 Convertible
14-dy1xxx with ALC295 codec. The appropriate coefficient index and bits
were identified through a brute-force method, as detailed in
https://bbs.archlinux.org/viewtopic.php?pid=2079504#p2079504.
Signed-off-by: Navon John Lukose <navonjohnlukose@gmail.com>
Link: https://patch.msgid.link/20250307213319.35507-1-navonjohnlukose@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Free some resources in the error handling path of the probe, as already
done in the remove function.
Fixes: e3523e01869d ("ASoC: wm0010: Add initial wm0010 DSP driver")
Fixes: fd8b96574456 ("ASoC: wm0010: Clear IRQ as wake source and include missing header")
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/5139ba1ab8c4c157ce04e56096a0f54a1683195c.1741549792.git.christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT722_SDCA_ENT_FU15,
RT722_SDCA_CTL_FU_CH_GAIN, CH_01) ... SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY,
RT722_SDCA_ENT_FU15, RT722_SDCA_CTL_FU_CH_GAIN, CH_04) are used by the
"FU15 Boost Volume" control, but not marked as readable.
And the mbq size are 2 for those registers.
Fixes: 7f5d6036ca005 ("ASoC: rt722-sdca: Add RT722 SDCA driver")
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Shuming Fan <shumingf@realtek.com>
Link: https://patch.msgid.link/20250310080440.58797-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The internal microphone on the Lenovo ThinkPad E16 model requires a
quirk entry to work properly. This was fixed in a previous patch (linked
below), but depending on the specific variant of the model, the product
name may be "21M5" or "21M6".
The following patch fixed this issue for the 21M5 variant:
https://lore.kernel.org/all/20240725065442.9293-1-tiwai@suse.de/
This patch adds support for the microphone on the 21M6 variant.
Link: https://github.com/ramaureirac/thinkpad-e14-linux/issues/31
Cc: stable@vger.kernel.org
Signed-off-by: Thomas Mizrahi <thomasmizra@gmail.com>
Link: https://patch.msgid.link/20250308041303.198765-1-thomasmizra@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The range of ADC volume is -1 -> 3 (-6 to 18dB) so the number of levels
should actually be 4.
Fixes: fc918cbe874e ("ASoC: cs42l43: Add support for the cs42l43")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20250306133254.1861046-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
CS35L41 HDA
Laptop uses 2 CS35L41 Amps with HDA, using External boost with I2C
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250305170714.755794-8-sbinding@opensource.cirrus.com
|
|
CS35L41 HDA
Add support for ASUS B5605CCA and B5405CCA.
Laptops use 2 CS35L41 Amps with HDA, using Internal boost, with SPI
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250305170714.755794-7-sbinding@opensource.cirrus.com
|
|
CS35L41 HDA
Add support for ASUS B3405CCA / P3405CCA, B3605CCA / P3605CCA,
B3405CCA, B3605CCA.
Laptops use 2 CS35L41 Amps with HDA, using Internal boost, with SPI
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250305170714.755794-6-sbinding@opensource.cirrus.com
|
|
Add support for ASUS B3405CVA, B5405CVA, B5605CVA, B3605CVA.
Laptops use 2 CS35L41 Amps with HDA, using Internal boost, with SPI
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250305170714.755794-5-sbinding@opensource.cirrus.com
|
|
Add support for ASUS G614PH/PM/PP and G614FH/FM/FP.
Laptops use 2 CS35L41 Amps with HDA, using Internal boost, with I2C
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250305170714.755794-4-sbinding@opensource.cirrus.com
|
|
CS35L41 HDA
Add support for ASUS GA603KP, GA603KM and GA603KH.
Laptops use 2 CS35L41 Amps with HDA, using Internal boost, with I2C
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250305170714.755794-3-sbinding@opensource.cirrus.com
|
|
Add support for ASUS G814PH/PM/PP and G814FH/FM/FP.
Laptops use 2 CS35L41 Amps with HDA, using Internal boost, with I2C.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250305170714.755794-2-sbinding@opensource.cirrus.com
|
|
This reverts commit 9bdd10d57a88 ("ASoC: ops: Shift tested values in
snd_soc_put_volsw() by +min"), and makes some additional related
updates.
There are two ways the platform_max could be interpreted; the maximum
register value, or the maximum value the control can be set to. The
patch moved from treating the value as a control value to a register
one. When the patch was applied it was technically correct as
snd_soc_limit_volume() also used the register interpretation. However,
even then most of the other usages treated platform_max as a
control value, and snd_soc_limit_volume() has since been updated to
also do so in commit fb9ad24485087 ("ASoC: ops: add correct range
check for limiting volume"). That patch however, missed updating
snd_soc_put_volsw() back to the control interpretation, and fixing
snd_soc_info_volsw_range(). The control interpretation makes more
sense as limiting is typically done from the machine driver, so it is
appropriate to use the customer facing representation rather than the
internal codec representation. Update all the code to consistently use
this interpretation of platform_max.
Finally, also add some comments to the soc_mixer_control struct to
hopefully avoid further patches switching between the two approaches.
Fixes: fb9ad24485087 ("ASoC: ops: add correct range check for limiting volume")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20250228151456.3703342-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
"generic_new_peripheral_assigned: invalid dev_num 1, wake supported 1"
is reported by our internal CI test.
Rt1320's wake feature is not used in Linux and that's why it is not in
the wake_capable_list[] list in intel_auxdevice.c.
However, BIOS may set it as wake-capable. Overwrite wake_capable to 0
in the codec driver to align with wake_capable_list[].
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Acked-by: Shuming Fan <shumingf@realtek.com>
Link: https://patch.msgid.link/20250305134113.201326-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Dell XPS 13 7390 with the Realtek ALC3271 codec experiences
persistent humming noise when the power_save mode is enabled.
This issue occurs when the codec enters power saving mode,
leading to unwanted noise from the speakers.
This patch adds the affected model (PCI ID 0x1028:0x0962) to the
power_save denylist to ensure power_save is disabled by default,
preventing power-off related noise issues.
Steps to Reproduce
1. Boot the system with `snd_hda_intel` loaded.
2. Verify that `power_save` mode is enabled:
```sh
cat /sys/module/snd_hda_intel/parameters/power_save
````
output: 10 (default power save timeout)
3. Wait for the power save timeout
4. Observe a persistent humming noise from the speakers
5. Disable `power_save` manually:
```sh
echo 0 | sudo tee /sys/module/snd_hda_intel/parameters/power_save
````
6. Confirm that the noise disappears immediately.
This issue has been observed on my system, and this patch
successfully eliminates the unwanted noise. If other users
experience similar issues, additional reports would be helpful.
Signed-off-by: Hoku Ishibe <me@hokuishi.be>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250224020517.51035-1-me@hokuishi.be
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Add ALC222 its own depop functions for alc_init and alc_shutup.
[note: this fixes pop noise issues on the models with two headphone
jacks -- tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The alternative path leads to a build error after a recent change:
sound/pci/hda/patch_realtek.c: In function 'alc233_fixup_lenovo_low_en_micmute_led':
include/linux/stddef.h:9:14: error: called object is not a function or function pointer
9 | #define NULL ((void *)0)
| ^
sound/pci/hda/patch_realtek.c:5041:49: note: in expansion of macro 'NULL'
5041 | #define alc233_fixup_lenovo_line2_mic_hotkey NULL
| ^~~~
sound/pci/hda/patch_realtek.c:5063:9: note: in expansion of macro 'alc233_fixup_lenovo_line2_mic_hotkey'
5063 | alc233_fixup_lenovo_line2_mic_hotkey(codec, fix, action);
Using IS_REACHABLE() is somewhat questionable here anyway since it
leads to the input code not working when the HDA driver is builtin
but input is in a loadable module. Replace this with a hard compile-time
dependency on CONFIG_INPUT. In practice this won't chance much
other than solve the compiler error because it is rare to require
sound output but no input support.
Fixes: f603b159231b ("ALSA: hda/realtek - add supported Mic Mute LED for Lenovo platform")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://patch.msgid.link/20250304142620.582191-1-arnd@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Hardware reports jack absent after reset/suspension regardless of jack
state, so introduce an additional delay only in suspension case to allow
proper detection to take place after a short delay.
Signed-off-by: Maciej Strozek <mstrozek@opensource.cirrus.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20250304140504.139245-1-mstrozek@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Commit 4204eccc7b2a ("ASoC: tegra: Add support for S24_LE audio format")
added support for the S24_LE audio format, but duplicated S16_LE in
OUT_DAI() for ADX instead.
Fix this by adding support for the S24_LE audio format.
Compile-tested only.
Cc: stable@vger.kernel.org
Fixes: 4204eccc7b2a ("ASoC: tegra: Add support for S24_LE audio format")
Signed-off-by: Thorsten Blum <thorsten.blum@linux.dev>
Link: https://patch.msgid.link/20250222225700.539673-2-thorsten.blum@linux.dev
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Temperatures are reported in units of Celsius however hwmon expects
values to be in millidegree of Celsius. Userspace tools observe values
close to zero and report it as "Not available" or incorrect values like
0C or 1C. Add a simple conversion to fix that.
Before the change:
wsa884x-virtual-0
Adapter: Virtual device
temp1: +0.0°C
--
wsa884x-virtual-0
Adapter: Virtual device
temp1: +0.0°C
Also reported as N/A before first amplifier power on.
After this change and initial wsa884x power on:
wsa884x-virtual-0
Adapter: Virtual device
temp1: +39.0°C
--
wsa884x-virtual-0
Adapter: Virtual device
temp1: +37.0°C
Tested on sm8550 only.
Cc: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Alexey Klimov <alexey.klimov@linaro.org>
Link: https://patch.msgid.link/20250221044024.1207921-1-alexey.klimov@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
create_sdw_dailinks()
Initialize current_be_id to 0 to handle the unlikely case when there are
no devices connected to a DAI.
In this case create_sdw_dailink() would return without touching the passed
pointer to current_be_id.
Found by gcc -fanalyzer
Fixes: 59bf457d8055 ("ASoC: intel: sof_sdw: Factor out SoundWire DAI creation")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Cc: stable@vger.kernel.org
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20250303065552.78328-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The module parameter defines number of iso packets per one URB. User is
allowed to set any value to the parameter of type int, which can lead to
various kinds of weird and incorrect behavior like integer overflows,
truncations, etc. Number of packets should be a small non-negative number.
Since this parameter is read-only, its value can be validated on driver
probe.
Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2")
Signed-off-by: Murad Masimov <m.masimov@mt-integration.ru>
Link: https://patch.msgid.link/20250303100413.835-1-m.masimov@mt-integration.ru
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Support Mic Mute LED for ThinkCentre M series.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/c211a2702f1f411e86bd7420d7eebc03@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
snd_seq_client_use_ptr() is supposed to return the snd_seq_client
object for the given client ID, and it tries to handle the module
auto-loading when no matching object is found. Although the module
handling is performed only conditionally with "!in_interrupt()", this
condition may be fragile, e.g. when the code is called from the ALSA
timer callback where the spinlock is temporarily disabled while the
irq is disabled. Then his doesn't fit well and spews the error about
sleep from invalid context, as complained recently by syzbot.
Also, in general, handling the module-loading at each time if no
matching object is found is really an overkill. It can be still
useful when performed at the top-level ioctl or proc reads, but it
shouldn't be done at event delivery at all.
For addressing the issues above, this patch disables the module
handling in snd_seq_client_use_ptr() in normal cases like event
deliveries, but allow only in limited and safe situations.
A new function client_load_and_use_ptr() is used for the cases where
the module loading can be done safely, instead.
Reported-by: syzbot+4cb9fad083898f54c517@syzkaller.appspotmail.com
Closes: https://lore.kernel.org/67c272e5.050a0220.dc10f.0159.GAE@google.com
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250301114530.8975-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
After some digging around I have found that this laptop has Cirrus's smart
aplifiers connected to SPI bus (spi1-CSC3551:00-cs35l41-hda).
To get them correctly detected and working I had to modify patch_realtek.c
with ASUS EXPERTBOOK P5405CSA 1.0 SystemID (0x1043, 0x1f63) and add
corresponding hda_quirk (ALC245_FIXUP_CS35L41_SPI_2).
Signed-off-by: Daniel Bárta <daniel.barta@trustlab.cz>
Link: https://patch.msgid.link/20250227161256.18061-2-daniel.barta@trustlab.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Use the basic quirk for this type of amplifier. Sound works in speakers,
headphones, and microphone. Whereas none worked before.
Tested-by: Kyle Gospodnetich <me@kylegospodneti.ch>
Signed-off-by: Antheas Kapenekakis <lkml@antheas.dev>
Link: https://patch.msgid.link/20250227175107.33432-3-lkml@antheas.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
In commit 1e9c708dc3ae ("ALSA: hda/tas2781: Add new quirk for Lenovo,
ASUS, Dell projects") Baojun adds a bunch of projects to the file,
including for the Ally X. Turns out the initial Ally X was not sorted
properly, so the kernel had 2 quirks for it.
The previous quirk overrode the new one due to being earlier and they
are different. When AB testing, the normal pin fixup seems to work ok
but causes a bit of a minor popping. Given the other config is more
complicated and may cause undefined behavior, revert it.
Fixes: 1e9c708dc3ae ("ALSA: hda/tas2781: Add new quirk for Lenovo, ASUS, Dell projects")
Signed-off-by: Antheas Kapenekakis <lkml@antheas.dev>
Link: https://patch.msgid.link/20250227175107.33432-2-lkml@antheas.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.14
More driver specific fixes, the firmware change is part of fixing the
race conditions in the Cirrus driver.
|
|
This fixes a regression introduced a few weeks ago in stable kernels
6.12.14 and 6.13.3. The internal microphone on ASUS Vivobook N705UD /
X705UD laptops is broken: the microphone appears in userspace (e.g.
Gnome settings) but no sound is detected.
I bisected it to commit 3b4309546b48 ("ALSA: hda: Fix headset detection
failure due to unstable sort").
I figured out the cause:
1. The initial pins enabled for the ALC256 driver are:
cfg->inputs == {
{ pin=0x19, type=AUTO_PIN_MIC,
is_headset_mic=1, is_headphone_mic=0, has_boost_on_pin=1 },
{ pin=0x1a, type=AUTO_PIN_MIC,
is_headset_mic=0, is_headphone_mic=0, has_boost_on_pin=1 } }
2. Since 2017 and commits c1732ede5e8 ("ALSA: hda/realtek - Fix headset
and mic on several ASUS laptops with ALC256") and 28e8af8a163 ("ALSA:
hda/realtek: Fix mic and headset jack sense on ASUS X705UD"), the
quirk ALC256_FIXUP_ASUS_MIC is also applied to ASUS X705UD / N705UD
laptops.
This added another internal microphone on pin 0x13:
cfg->inputs == {
{ pin=0x13, type=AUTO_PIN_MIC,
is_headset_mic=0, is_headphone_mic=0, has_boost_on_pin=1 },
{ pin=0x19, type=AUTO_PIN_MIC,
is_headset_mic=1, is_headphone_mic=0, has_boost_on_pin=1 },
{ pin=0x1a, type=AUTO_PIN_MIC,
is_headset_mic=0, is_headphone_mic=0, has_boost_on_pin=1 } }
I don't know what this pin 0x13 corresponds to. To the best of my
knowledge, these laptops have only one internal microphone.
3. Before 2025 and commit 3b4309546b48 ("ALSA: hda: Fix headset
detection failure due to unstable sort"), the sort function would let
the microphone of pin 0x1a (the working one) *before* the microphone
of pin 0x13 (the phantom one).
4. After this commit 3b4309546b48, the fixed sort function puts the
working microphone (pin 0x1a) *after* the phantom one (pin 0x13). As
a result, no sound is detected anymore.
It looks like the quirk ALC256_FIXUP_ASUS_MIC is not needed anymore for
ASUS Vivobook X705UD / N705UD laptops. Without it, everything works
fine:
- the internal microphone is detected and records actual sound,
- plugging in a jack headset is detected and can record actual sound
with it,
- unplugging the jack headset makes the system go back to internal
microphone and can record actual sound.
Cc: stable@vger.kernel.org
Cc: Kuan-Wei Chiu <visitorckw@gmail.com>
Cc: Chris Chiu <chris.chiu@canonical.com>
Fixes: 3b4309546b48 ("ALSA: hda: Fix headset detection failure due to unstable sort")
Tested-by: Adrien Vergé <adrienverge@gmail.com>
Signed-off-by: Adrien Vergé <adrienverge@gmail.com>
Link: https://patch.msgid.link/20250226135515.24219-1-adrienverge@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Merge series from Bard Liao <yung-chuan.liao@linux.intel.com>:
Currently, we assume that the PCH DMIC pins are pin-muxed with SoundWire
links. However, we do see a HW design that use PCH DMIC along with 3
SoundWire links. Remove the check and add warning to let users know that
SoundWire MIC and PCH DMIC are both present and they could overwrite it
with kernel params.
|
|
ASUS VivoBook 15 with SSID 1043:1460 took an incorrect quirk via the
pin pattern matching for ASUS (ALC256_FIXUP_ASUS_MIC), resulting in
the two built-in mic pins (0x13 and 0x1b). This had worked without
problems casually in the past because the right pin (0x1b) was picked
up as the primary device. But since we fixed the pin enumeration for
other bugs, the bogus one (0x13) is picked up as the primary device,
hence the bug surfaced now.
For addressing the regression, this patch explicitly specifies the
quirk entry with ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, which sets up only
the headset mic pin.
Fixes: 3b4309546b48 ("ALSA: hda: Fix headset detection failure due to unstable sort")
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=219807
Link: https://patch.msgid.link/20250225154540.13543-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
When SPI is used for control, the driver must hold the SPI bus lock
while issuing the sequence of writes to perform a soft reset.
>From the time the driver writes the SYSTEM_RESET command until the
driver does a write to terminate the reset, there must not be any
activity on the SPI bus lines. If there is any SPI activity during the
soft-reset, another soft-reset will be triggered. The state of the SPI
chip select is irrelevant.
A repeated soft-reset does not in itself cause any problems, and it is
not an infinite loop. The problem is a race between these resets and
the driver polling for boot completion. There is a time window between
soft resets where the driver could read HALO_STATE as 2 (fully booted)
while the chip is actually soft-resetting. Although this window is
small, it is long enough that it is possible to hit it in normal
operation.
To prevent this race and ensure the chip really is fully booted, the
driver calls spi_bus_lock() to prevent other activity while resetting.
It then issues the SYSTEM_RESET mailbox command. After allowing
sufficient time for reset to take effect, the driver issues a PING
mailbox command, which will force completion of the full soft-reset
sequence. The SPI bus lock can then be released. The mailbox is
checked for any boot or wakeup response from the firmware, before the
value in HALO_STATE will be trusted.
This does not affect SoundWire or I2C control.
Fixes: 8a731fd37f8b ("ASoC: cs35l56: Move utility functions to shared file")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250225131843.113752-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Typically, SoundWire MIC and PCH DMIC will not coexist. However, we may
want to use both of them in some special cases. Add a warning to let
users know that SoundWire MIC and PCH DMIC are both present and they
could overwrite it with kernel params.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://patch.msgid.link/20250225093716.67240-3-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Currently, we assume that the PCH DMIC pins are pin-muxed with SoundWire
links. However, we do see a HW design that use PCH DMIC along with 3
SoundWire links. Remove the check now.
With this change the PCM DMIC will be presented if it is reported by the
BIOS irrespective of whether there are SDW links present or not.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://patch.msgid.link/20250225093716.67240-2-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
If stream names of DAI driver are duplicated there'll be warnings when
machine driver tries to add widgets on a route:
[ 8.831335] fsl-asoc-card sound-wm8960: ASoC: sink widget CPU-Playback overwritten
[ 8.839917] fsl-asoc-card sound-wm8960: ASoC: source widget CPU-Capture overwritten
Use different stream names to avoid such warnings.
DAI names in AUDMIX are also updated accordingly.
Fixes: 15c958390460 ("ASoC: fsl_sai: Add separate DAI for transmitter and receiver")
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20250217010437.258621-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The ES8328 codec driver, which is also used for the ES8388 chip that
appears to have an identical register map, claims that the output can
either take the route from DAC->Mixer->Output or through DAC->Output
directly. To the best of what I could find, this is not true, and
creates problems.
Without DACCONTROL17 bit index 7 set for the left channel, as well as
DACCONTROL20 bit index 7 set for the right channel, I cannot get any
analog audio out on Left Out 2 and Right Out 2 respectively, despite the
DAPM routes claiming that this should be possible. Furthermore, the same
is the case for Left Out 1 and Right Out 1, showing that those two don't
have a direct route from DAC to output bypassing the mixer either.
Those control bits toggle whether the DACs are fed (stale bread?) into
their respective mixers. If one "unmutes" the mixer controls in
alsamixer, then sure, the audio output works, but if it doesn't work
without the mixer being fed the DAC input then evidently it's not a
direct output from the DAC.
ES8328/ES8388 are seemingly not alone in this. ES8323, which uses a
separate driver for what appears to be a very similar register map,
simply flips those two bits on in its probe function, and then pretends
there is no power management whatsoever for the individual controls.
Fair enough.
My theory as to why nobody has noticed this up to this point is that
everyone just assumes it's their fault when they had to unmute an
additional control in ALSA.
Fix this in the es8328 driver by removing the erroneous direct route,
then get rid of the playback switch controls and have those bits tied to
the mixer's widget instead, which until now had no register to play
with.
Fixes: 567e4f98922c ("ASoC: add es8328 codec driver")
Signed-off-by: Nicolas Frattaroli <nicolas.frattaroli@collabora.com>
Link: https://patch.msgid.link/20250222-es8328-route-bludgeoning-v1-1-99bfb7fb22d9@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
