[ Upstream commit ed92bc5264 ]
Free some resources in the error handling path of the probe, as already
done in the remove function.
Fixes: e3523e0186 ("ASoC: wm0010: Add initial wm0010 DSP driver")
Fixes: fd8b965744 ("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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 247fba1341 ]
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: 7f5d6036ca ("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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e26f1cfeac ]
The range of ADC volume is -1 -> 3 (-6 to 18dB) so the number of levels
should actually be 4.
Fixes: fc918cbe87 ("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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0eba2a7e85 ]
This reverts commit 9bdd10d57a ("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 fb9ad24485 ("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: fb9ad24485 ("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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f5468beeab ]
TX launch polarity needs to be the opposite of RX capture polarity, to
generate the right bit slot alignment.
Reviewed-by: Neal Gompa <neal@gompa.dev>
Signed-off-by: Hector Martin <marcan@marcan.st>
Signed-off-by: James Calligeros <jcalligeros99@gmail.com>
Link: https://patch.msgid.link/20250218-apple-codec-changes-v2-28-932760fd7e07@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 579cd64b9d ]
The scale starts at -100dB, not -128dB.
Signed-off-by: Hector Martin <marcan@marcan.st>
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://patch.msgid.link/20250208-asoc-tas2770-v1-1-cf50ff1d59a3@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ac84ca815a ]
In some cases, e.g. during resuming from suspend, there is a possibility
that some IPC reply messages get received by the host while the DSP
firmware has not yet reached the complete boot state.
Detect when this happens and do not attempt to process the unexpected
replies from DSP. Instead, provide proper debugging support.
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://patch.msgid.link/20250207-sof-vangogh-fixes-v1-3-67824c1e4c9a@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 33b7dc7843 ]
In enviornment without KMOD requesting module may fail to load
snd-hda-codec-hdmi, resulting in HDMI audio not usable.
Add softdep to loading HDMI codec module first to ensure we can load it
correctly.
Signed-off-by: Terry Cheong <htcheong@chromium.org>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Johny Lin <lpg76627@gmail.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://patch.msgid.link/20250206094723.18013-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 679074942c ]
Using `fsleep` instead of `msleep` resolves some customer complaints
regarding the precision of up/down DAPM event timing. `fsleep()`
automatically selects the appropriate sleep function, making the delay
time more predictable.
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Link: https://patch.msgid.link/20250205160849.500306-1-vitalyr@opensource.cirrus.com
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 89f9cf1858 ]
Current rsnd driver supports Synchronous SRC Mode, but HW allow to update
rate only within 1% from current rate. Adjust to it.
Becially, this feature is used to fine-tune subtle difference that occur
during sampling rate conversion in SRC. So, it should be called within 1%
margin of rate difference.
If there was difference over 1%, it will apply with 1% increments by using
loop without indicating error message.
Cc: Yoshihiro Shimoda <yoshihiro.shimoda.uh@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Yoshihiro Shimoda <yoshihiro.shimoda.uh@renesas.com>
Tested-by: Yoshihiro Shimoda <yoshihiro.shimoda.uh@renesas.com>
Link: https://patch.msgid.link/871pwd2qe8.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c3fc002b20 ]
rsnd_kctrl_accept_runtime() (1) is used for runtime convert rate
(= Synchronous SRC Mode). Now, rsnd driver has 2 kctrls for it
(A): "SRC Out Rate Switch"
(B): "SRC Out Rate" // it calls (1)
(A): can be called anytime
(B): can be called only runtime, and will indicate warning if it was used
at non-runtime.
To use runtime convert rate (= Synchronous SRC Mode), user might uses
command in below order.
(X): > amixer set "SRC Out Rate" on
> aplay xxx.wav &
(Y): > amixer set "SRC Out Rate" 48010 // convert rate to 48010Hz
(Y): calls B
(X): calls both A and B.
In this case, when user calls (X), it calls both (A) and (B), but it is not
yet start running. So, (B) will indicate warning.
This warning was added by commit b5c0886898 ("ASoC: rsnd: add warning
message to rsnd_kctrl_accept_runtime()"), but the message sounds like the
operation was not correct. Let's update warning message.
The message is very SRC specific, implement it in src.c
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Yoshihiro Shimoda <yoshihiro.shimoda.uh@renesas.com>
Link: https://patch.msgid.link/8734gt2qed.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 796106e29e ]
It will indicate "unsupported clock rate" when setup clock failed.
But it is unclear what kind of rate was failed. Indicate it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Yoshihiro Shimoda <yoshihiro.shimoda.uh@renesas.com>
Link: https://patch.msgid.link/874j192qej.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 76b0a22d4c ]
The internal mic boost on the Positivo ARN50 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/20250201143930.25089-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit dabbd325b2 ]
commit 90de551c1b ("ASoC: simple-card-utils.c: enable multi Component
support") added muiti Component support, but was missing to add
dlc->of_node. Because of it, Sound device list will indicates strange
name if it was DPCM connection and driver supports dai->driver->dai_args,
like below
> aplay -l
card X: sndulcbmix [xxxx], device 0: fe.(null).rsnd-dai.0 (*) []
... ^^^^^^
It will be fixed by this patch
> aplay -l
card X: sndulcbmix [xxxx], device 0: fe.sound@ec500000.rsnd-dai.0 (*) []
... ^^^^^^^^^^^^^^
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://patch.msgid.link/87ikpp2rtb.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit d0bbe33266 upstream.
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: f603b15923 ("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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 172a0f5097 ]
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: 1da177e4c3 ("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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit ca0dedaff9 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f603b15923 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1ee5aa765c upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c9ce148ea7 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c6557ccf80 upstream.
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 3b4309546b ("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 c1732ede5e ("ALSA: hda/realtek - Fix headset
and mic on several ASUS laptops with ALC256") and 28e8af8a16 ("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 3b4309546b ("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 3b4309546b, 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: 3b4309546b ("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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9af3b4f2d8 upstream.
Re-add the sample-rate quirk for the Pioneer DJM-900NXS2. This
device does not work without setting sample-rate.
Signed-off-by: Dmitry Panchenko <dmitry@d-systems.ee>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250220161540.3624660-1-dmitry@d-systems.ee
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 9e7c6779e3 ]
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: 3b4309546b ("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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a40ce9f4bd ]
These models use 2xCS35L41amps with HDA using SPI and I2C.
All models use Internal Boost.
Some models also use Realtek Speakers in conjunction with
CS35L41.
All models require DSD support to be added inside
cs35l41_hda_property.c
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20231218151221.388745-4-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 9e7c6779e3 ("ALSA: hda/realtek: Fix wrong mic setup for ASUS VivoBook 15")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5b0c02f9b8 ]
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: 567e4f9892 ("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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a3bdd8f5c2 ]
We fixed the UAF issue in USB MIDI code by canceling the pending work
at closing each MIDI output device in the commit below. However, this
assumed that it's the only one that is tied with the endpoint, and it
resulted in unexpected data truncations when multiple devices are
assigned to a single endpoint and opened simultaneously.
For addressing the unexpected MIDI message drops, simply replace
cancel_work_sync() with flush_work(). The drain callback should have
been already invoked before the close callback, hence the port->active
flag must be already cleared. So this just assures that the pending
work is finished before freeing the resources.
Fixes: 0125de3812 ("ALSA: usb-audio: Cancel pending work at closing a MIDI substream")
Reported-and-tested-by: John Keeping <jkeeping@inmusicbrands.com>
Closes: https://lore.kernel.org/20250217111647.3368132-1-jkeeping@inmusicbrands.com
Link: https://patch.msgid.link/20250218114024.23125-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 46c7b901e2 upstream.
The spcm->stream[substream->stream].substream is set during open and was
left untouched. After the first PCM stream it will never be NULL and we
have code which checks for substream NULLity as indication if the stream is
active or not.
For the compressed cstream pointer the same has been done, this change will
correct the handling of PCM streams.
Fixes: 090349a9fe ("ASoC: SOF: Add support for compress API for stream data/offset")
Cc: stable@vger.kernel.org
Reported-by: Curtis Malainey <cujomalainey@chromium.org>
Closes: https://github.com/thesofproject/linux/pull/5214
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://patch.msgid.link/20250205135232.19762-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6d1f86610f upstream.
Allows the LED on the dedicated mute button on the HP ProBook 450 G4
laptop to change colour correctly.
Signed-off-by: John Veness <john-linux@pelago.org.uk>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/2fb55d48-6991-4a42-b591-4c78f2fad8d7@pelago.org.uk
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 822b7ec657 upstream.
Check the return value of snd_ctl_rename_id() in
snd_hda_create_dig_out_ctls(). Ensure that failures
are properly handled.
[ Note: the error cannot happen practically because the only error
condition in snd_ctl_rename_id() is the missing ID, but this is a
rename, hence it must be present. But for the code consistency,
it's safer to have always the proper return check -- tiwai ]
Fixes: 5c219a3408 ("ALSA: hda: Fix kctl->id initialization")
Cc: stable@vger.kernel.org # 6.4+
Signed-off-by: Wentao Liang <vulab@iscas.ac.cn>
Link: https://patch.msgid.link/20250213074543.1620-1-vulab@iscas.ac.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a8c9a45338 upstream.
If 'micfil->quality' received from micfil_quality_set() somehow ends
up with an unpredictable value, switch() operator will fail to
initialize local variable qsel before regmap_update_bits() tries
to utilize it.
While it is unlikely, play it safe and enable a default case that
returns -EINVAL error.
Found by Linux Verification Center (linuxtesting.org) with static
analysis tool SVACE.
Fixes: bea1d61d58 ("ASoC: fsl_micfil: rework quality setting")
Cc: stable@vger.kernel.org
Signed-off-by: Nikita Zhandarovich <n.zhandarovich@fintech.ru>
Link: https://patch.msgid.link/20250116142436.22389-1-n.zhandarovich@fintech.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d8d99c3b5c upstream.
The nullity of sps->cstream should be checked similarly as it is done in
sof_set_stream_data_offset() function.
Assuming that it is not NULL if sps->stream is NULL is incorrect and can
lead to NULL pointer dereference.
Fixes: 090349a9fe ("ASoC: SOF: Add support for compress API for stream data/offset")
Cc: stable@vger.kernel.org
Reported-by: Curtis Malainey <cujomalainey@chromium.org>
Closes: https://github.com/thesofproject/linux/pull/5214
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://patch.msgid.link/20250205135232.19762-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit e77aa4b2ea ]
When a destination client is a user client in the legacy MIDI mode and
it sets the no-UMP-conversion flag, currently the all UMP events are
still passed as-is. But this may confuse the user-space, because the
event packet size is different from the legacy mode.
Since we cannot handle UMP events in user clients unless it's running
in the UMP client mode, we should filter out those events instead of
accepting blindly. This patch addresses it by slightly adjusting the
conditions for UMP event handling at the event delivery time.
Fixes: 329ffe11a0 ("ALSA: seq: Allow suppressing UMP conversions")
Link: https://lore.kernel.org/b77a2cd6-7b59-4eb0-a8db-22d507d3af5f@gmail.com
Link: https://patch.msgid.link/20250217170034.21930-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 08b613b9e2 ]
This patch corrects the full-scale volume setting logic. On certain
platforms, the full-scale volume bit is required. The current logic
mistakenly sets this bit and incorrectly clears reserved bit 0, causing
the headphone output to be muted.
Fixes: 342b6b610a ("ALSA: hda/cs8409: Fix Full Scale Volume setting for all variants")
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Link: https://patch.msgid.link/20250214210736.30814-1-vitalyr@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6b24e67b40 ]
Commit 2f45a4e289 ("ASoC: rockchip: i2s_tdm: Fixup config for
SND_SOC_DAIFMT_DSP_A/B") applied a partial change to fix the
configuration for DSP A and DSP B formats.
The shift control also needs updating to set the correct offset for
frame data compared to LRCK. Set the correct values.
Fixes: 081068fd64 ("ASoC: rockchip: add support for i2s-tdm controller")
Signed-off-by: John Keeping <jkeeping@inmusicbrands.com>
Link: https://patch.msgid.link/20250204161311.2117240-1-jkeeping@inmusicbrands.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 82a0a3e6f8 ]
My static checker rule complains about this code. The concern is that
if "sample_space" is negative then the "sample_space >= runtime->channels"
condition will not work as intended because it will be type promoted to a
high unsigned int value.
strm->fifo_sample_size is SSI_FIFO_DEPTH (32). The SSIFSR_TDC_MASK is
0x3f. Without any further context it does seem like a reasonable warning
and it can't hurt to add a check for negatives.
Cc: stable@vger.kernel.org
Fixes: 03e786bd43 ("ASoC: sh: Add RZ/G2L SSIF-2 driver")
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Reviewed-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://patch.msgid.link/e07c3dc5-d885-4b04-a742-71f42243f4fd@stanley.mountain
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6917192378 ]
The Vexia EDU ATLA 10 tablet comes in 2 different versions with
significantly different mainboards. The only outward difference is that
the charging barrel on one is marked 5V and the other is marked 9V.
The 5V version mostly works with the BYTCR defaults, except that it is
missing a CHAN package in its ACPI tables and the default of using
SSP0-AIF2 is wrong, instead SSP0-AIF1 must be used. That and its jack
detect signal is not inverted as it usually is.
Add a DMI quirk for the 5V version to fix sound not working.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://patch.msgid.link/20250123132507.18434-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 3b4309546b upstream.
The auto_parser assumed sort() was stable, but the kernel's sort() uses
heapsort, which has never been stable. After commit 0e02ca29a5
("lib/sort: optimize heapsort with double-pop variation"), the order of
equal elements changed, causing the headset to fail to work.
Fix the issue by recording the original order of elements before
sorting and using it as a tiebreaker for equal elements in the
comparison function.
Fixes: b9030a005d ("ALSA: hda - Use standard sort function in hda_auto_parser.c")
Reported-by: Austrum <austrum.lab@gmail.com>
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=219158
Tested-by: Austrum <austrum.lab@gmail.com>
Cc: stable@vger.kernel.org
Signed-off-by: Kuan-Wei Chiu <visitorckw@gmail.com>
Link: https://patch.msgid.link/20250128165415.643223-1-visitorckw@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1aec3ed2e3 upstream.
Positivo C6400 is equipped with ALC269VB, and it needs
ALC269VB_FIXUP_ASUS_ZENBOOK quirk to make its headset mic work.
Also must to limits the microphone boost.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250114170619.11510-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 7e24ec93ae ]
As reported by the kernel test robot, the following error occurs:
sound/soc/amd/yc/acp6x-mach.c: In function 'acp6x_probe':
>> sound/soc/amd/yc/acp6x-mach.c:573:15: error: implicit declaration of function 'acpi_evaluate_integer'; did you mean 'acpi_evaluate_object'? [-Werror=implicit-function-declaration]
573 | ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status);
| ^~~~~~~~~~~~~~~~~~~~~
| acpi_evaluate_object
cc1: some warnings being treated as errors
The function 'acpi_evaluate_integer' and its prototype in 'acpi_bus.h'
are only available when 'CONFIG_ACPI' is enabled. Add a 'depends on ACPI'
directive in Kconfig to ensure proper compilation.
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202501090345.pBIDRTym-lkp@intel.com/
Signed-off-by: Yu-Chun Lin <eleanor15x@gmail.com>
Link: https://patch.msgid.link/20250109171547.362412-1-eleanor15x@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 301c26a018 ]
commit 1f56643514 ("ASoC: lower "no backend DAIs enabled for ... Port"
log severity") ignores -EINVAL error message on common soc_pcm_ret().
It is used from many functions, ignoring -EINVAL is over-kill.
The reason why -EINVAL was ignored was it really should only be used
upon invalid parameters coming from userspace and in that case we don't
want to log an error since we do not want to give userspace a way to do
a denial-of-service attack on the syslog / diskspace.
So don't use soc_pcm_ret() on .prepare callback is better idea.
Link: https://lore.kernel.org/r/87v7vptzap.wl-kuninori.morimoto.gx@renesas.com
Cc: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87bjxg8jju.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit d85fc52cbb upstream.
Audio control requests that sets sampling frequency sometimes fail on
this card. Adding delay between control messages eliminates that problem.
usb 1-1: New USB device found, idVendor=2fc6, idProduct=f0b7
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 1-1: Product: iBasso DC07 Pro
usb 1-1: Manufacturer: iBasso
usb 1-1: SerialNumber: CTUA171130B
Signed-off-by: Lianqin Hu <hulianqin@vivo.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/TYUPR06MB62174A48D04E09A37996DF84D2ED2@TYUPR06MB6217.apcprd06.prod.outlook.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 5323186e2e ]
In commit
9e2ab4b18e ("ASoC: rockchip: i2s-tdm: Fix inaccurate sampling rates"),
the set_sysclk callback was removed as considered unused as the mclk rate
can be set in the hw_params callback.
The difference between hw_params and set_sysclk is that the former is
called with the audio sampling rate set in the params (e.g.: 48000 Hz)
while the latter is called with a clock rate already computed with
sampling_rate * mclk-fs (e.g.: 48000 * 256)
For HDMI audio using the Rockchip I2S TDM driver, the mclk-fs value must
be set to 128 instead of the default 256, and that value is set in the
device tree at the machine driver level (like a simple-audio-card
compatible node).
Therefore, the i2s_tdm driver has no idea that another mclk-fs value can
be configured and simply computes the mclk rate in the hw_params callback
with DEFAULT_MCLK_FS * params_rate(params), which is wrong for HDMI
audio.
Re-add the set_sysclk callback so that the mclk rate is computed by the
machine driver which has the correct mclk-fs value set in its device tree
node.
Fixes: 9e2ab4b18e ("ASoC: rockchip: i2s-tdm: Fix inaccurate sampling rates")
Signed-off-by: Detlev Casanova <detlev.casanova@collabora.com>
Link: https://patch.msgid.link/20250117163102.65807-1-detlev.casanova@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit cf4d74256f ]
While 'stack_dump_size' is a u32 bitfield of 16 bits, u32 has a bigger
upper bound than the type u16 of loop counter 'offset' what in theory
may lead to infinite loop condition.
Found out by Coverity static analyzer.
Fixes: c8c960c109 ("ASoC: Intel: avs: APL-based platforms support")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20250109122216.3667847-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
