My laptop, HP ProBook 450 G8 (32M40EA), has Realtek ALC236 codec on its
integrated sound card, and uses GPIO pins 0x2 and 0x1 for speaker mute
and mic mute LEDs correspondingly, as found out by me through hda-verb
invocations. This matches the GPIO masks used by the
alc236_fixup_hp_gpio_led() function.
PCI subsystem vendor and device IDs happen to be 0x103c and 0x8a75,
which has not been covered in the ALC2xx driver code yet.
Signed-off-by: Ilyas Gasanov <public@gsnoff.com>
Link: https://patch.msgid.link/20251125235441.53629-1-public@gsnoff.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since 8b3a087f7f ("ALSA: usb-audio: Unify virtual type units type to
UAC3 values") usb-audio is using UAC3_CLOCK_SOURCE instead of
bDescriptorSubtype, later refactored with e0ccdef926 ("ALSA: usb-audio:
Clean up check_input_term()") into parse_term_uac2_clock_source().
This breaks the clock source selection for at least my
1397:0003 BEHRINGER International GmbH FCA610 Pro.
Fix by using UAC2_CLOCK_SOURCE in parse_term_uac2_clock_source().
Fixes: 8b3a087f7f ("ALSA: usb-audio: Unify virtual type units type to UAC3 values")
Signed-off-by: René Rebe <rene@exactco.de>
Link: https://patch.msgid.link/20251125.154149.1121389544970412061.rene@exactco.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Integrated amplifier LEAK Stereo 230 by IAG Limited has built-in
ESS9038Q2M DAC served by XMOS controller. It supports both DSD Native
and DSD-over-PCM (DoP) operational modes. But it doesn't work properly
by default and tries DSD-to-PCM conversion. USB quirks below allow it
to operate as designed.
Add DSD_RAW quirk flag for IAG Limited devices (vendor ID 0x2622)
Add DSD format quirk for LEAK Stereo 230 (USB ID 0x2622:0x0061)
Signed-off-by: Ivan Zhaldak <i.v.zhaldak@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20251117125848.30769-1-i.v.zhaldak@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
__snd_vortex_probe() uses pci_read_config_word() that returns PCIBIOS_*
codes (positive values on error). However, the function checks 'err < 0'
which can never be true for PCIBIOS_* codes, causing errors to be silently
ignored.
Check for non-zero return value and convert PCIBIOS_* codes using
pcibios_err_to_errno() into normal errno before returning them.
Signed-off-by: Haotian Zhang <vulab@iscas.ac.cn>
Reviewed-by: Philipp Stanner <phasta@kernel.org>
Link: https://patch.msgid.link/20251117065559.1138-1-vulab@iscas.ac.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PureAudio APA DAC and Lotus DAC5 series are USB Audio
2.0 Class devices that support native Direct Stream Digital (DSD)
playback via specific vendor protocols.
Without these quirks, the devices may only function in standard
PCM mode, or fail to correctly report their DSD format capabilities
to the ALSA framework, preventing native DSD playback under Linux.
This commit adds new quirk entries for the mentioned DAC models
based on their respective Vendor/Product IDs (VID:PID), for example:
0x16d0:0x0ab1 (APA DAC), 0x16d0:0xeca1 (DAC5 series), etc.
The quirk ensures correct DSD format handling by setting the required
SNDRV_PCM_FMTBIT_DSD_U32_BE format bit and defining the DSD-specific
Audio Class 2.0 (AC2.0) endpoint configurations. This allows the ALSA
DSD API to correctly address the device for high-bitrate DSD streams,
bypassing the need for DoP (DSD over PCM).
Test on APA DAC and Lotus DAC5 SE under Arch Linux.
Tested-by: Lushih Hsieh <bruce@mail.kh.edu.tw>
Signed-off-by: Lushih Hsieh <bruce@mail.kh.edu.tw>
Link: https://patch.msgid.link/20251114052053.54989-1-bruce@mail.kh.edu.tw
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: Fixes for v6.18
A small collection of fixes, all driver specific and none especially
remarkable unless you have the hardware (many not even then).
rsnd_ssiu_probe() leaks an OF node reference obtained by
rsnd_ssiu_of_node(). The node reference is acquired but
never released across all return paths.
Fix it by declaring the device node with the __free(device_node)
cleanup construct to ensure automatic release when the variable goes
out of scope.
Fixes: 4e7788fb80 ("ASoC: rsnd: add SSIU BUSIF support")
Signed-off-by: Haotian Zhang <vulab@iscas.ac.cn>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/20251112065709.1522-1-vulab@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
In snd_usb_create_streams(), for UAC version 3 devices, the Interface
Association Descriptor (IAD) is retrieved via usb_ifnum_to_if(). If this
call fails, a fallback routine attempts to obtain the IAD from the next
interface and sets a BADD profile. However, snd_usb_mixer_controls_badd()
assumes that the IAD retrieved from usb_ifnum_to_if() is always valid,
without performing a NULL check. This can lead to a NULL pointer
dereference when usb_ifnum_to_if() fails to find the interface descriptor.
This patch adds a NULL pointer check after calling usb_ifnum_to_if() in
snd_usb_mixer_controls_badd() to prevent the dereference.
This issue was discovered by syzkaller, which triggered the bug by sending
a crafted USB device descriptor.
Fixes: 17156f23e9 ("ALSA: usb: add UAC3 BADD profiles support")
Signed-off-by: Haein Lee <lhi0729@kaist.ac.kr>
Link: https://patch.msgid.link/vwhzmoba9j2f.vwhzmob9u9e2.g6@dooray.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM stream data in USB-audio driver is transferred over USB URB
packet buffers, and each packet size is determined dynamically. The
packet sizes are limited by some factors such as wMaxPacketSize USB
descriptor. OTOH, in the current code, the actually used packet sizes
are determined only by the rate and the PPS, which may be bigger than
the size limit above. This results in a buffer overflow, as reported
by syzbot.
Basically when the limit is smaller than the calculated packet size,
it implies that something is wrong, most likely a weird USB
descriptor. So the best option would be just to return an error at
the parameter setup time before doing any further operations.
This patch introduces such a sanity check, and returns -EINVAL when
the packet size is greater than maxpacksize. The comparison with
ep->packsize[1] alone should suffice since it's always equal or
greater than ep->packsize[0].
Reported-by: syzbot+bfd77469c8966de076f7@syzkaller.appspotmail.com
Closes: https://syzkaller.appspot.com/bug?extid=bfd77469c8966de076f7
Link: https://lore.kernel.org/690b6b46.050a0220.3d0d33.0054.GAE@google.com
Cc: Lizhi Xu <lizhi.xu@windriver.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20251109091211.12739-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the commit referenced by the Fixes tag, clk_hw_get_clk()
was added in va_macro_probe() to get the fsgen clock,
but forgot to add the corresponding clk_put() in va_macro_remove().
This leads to a clock reference leak when the driver is unloaded.
Switch to devm_clk_hw_get_clk() to automatically manage the
clock resource.
Fixes: 30097967e0 ("ASoC: codecs: va-macro: use fsgen as clock")
Suggested-by: Konrad Dybcio <konrad.dybcio@oss.qualcomm.com>
Signed-off-by: Haotian Zhang <vulab@iscas.ac.cn>
Reviewed-by: Konrad Dybcio <konrad.dybcio@oss.qualcomm.com>
Link: https://patch.msgid.link/20251106143114.729-1-vulab@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
The bus_find_device_by_name() function returns a device pointer with an
incremented reference count, but the original code was missing put_device()
calls in some return paths, leading to reference count leaks.
Fix this by ensuring put_device() is called before function exit after
bus_find_device_by_name() succeeds
This follows the same pattern used elsewhere in the kernel where
bus_find_device_by_name() is properly paired with put_device().
Found via static analysis and code review.
Fixes: 4f8ef33dd4 ("ASoC: soc_sdw_utils: skip the endpoint that doesn't present")
Cc: stable@vger.kernel.org
Signed-off-by: Miaoqian Lin <linmq006@gmail.com>
Link: https://patch.msgid.link/20251029071804.8425-1-linmq006@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The probe function enables regulators at the beginning
but fails to disable them in its error handling path.
If any operation after enabling the regulators fails,
the probe will exit with an error, leaving the regulators
permanently enabled, which could lead to a resource leak.
Add a proper error handling path to call regulator_bulk_disable()
before returning an error.
Fixes: 9a397f4736 ("ASoC: cs4271: add regulator consumer support")
Signed-off-by: Haotian Zhang <vulab@iscas.ac.cn>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20251105062246.1955-1-vulab@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
After restructuring and splitting the HDMI codec driver code, each
HDMI codec driver contains the own build_controls and build_pcms ops.
A copy-n-paste error put the wrong entries for nvhdmi-mcp driver; both
build_controls and build_pcms are swapped. Unfortunately both
callbacks have the very same form, and the compiler didn't complain
it, either. This resulted in a NULL dereference because the PCM
instance hasn't been initialized at calling the build_controls
callback.
Fix it by passing the proper entries.
Fixes: ad781b550f ("ALSA: hda/hdmi: Rewrite to new probe method")
Cc: <stable@vger.kernel.org>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=220743
Link: https://patch.msgid.link/20251106104647.25805-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since snd_soc_suspend() is invoked through snd_soc_pm_ops->suspend(),
and snd_soc_pm_ops is associated with the soc_driver (defined in
sound/soc/soc-core.c), and there is no parent-child relationship between
the soc_driver and the DA7213 codec driver, the power management subsystem
does not enforce a specific suspend/resume order between the DA7213 driver
and the soc_driver.
Because of this, the different codec component functionalities, called from
snd_soc_resume() to reconfigure various functions, can race with the
DA7213 struct dev_pm_ops::resume function, leading to misapplied
configuration. This occasionally results in clipped sound.
Fix this by dropping the struct dev_pm_ops::{suspend, resume} and use
instead struct snd_soc_component_driver::{suspend, resume}. This ensures
the proper configuration sequence is handled by the ASoC subsystem.
Cc: stable@vger.kernel.org
Fixes: 431e040065 ("ASoC: da7213: Add suspend to RAM support")
Signed-off-by: Claudiu Beznea <claudiu.beznea.uj@bp.renesas.com>
Link: https://patch.msgid.link/20251104114914.2060603-1-claudiu.beznea.uj@bp.renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Fixes for v6.18
A bigger batch of fixes than I'd like, things built up due to holidays
and some last minute issues which caused me to hold off on sending a pul
request. None of these are super remarkable, and there's a few new
device IDs in here too including a relatively big block of AMD devices.
The Cirrus Logic CS530x support subject line is actually a fix that was
on the start of that series and got pulled in here, I forgot to fix the
subject up when merging.
When the driver supports DMA, it enqueues four DMA descriptors per
substream before the substream is started. New descriptors are enqueued in
the DMA completion callback, and each time a new descriptor is queued, the
dma_buffer_pos is incremented.
During suspend, the DMA transactions are terminated. There might be cases
where the four extra enqueued DMA descriptors are not completed and are
instead canceled on suspend. However, the cancel operation does not take
into account that the dma_buffer_pos was already incremented.
Previously, the suspend code reinitialized dma_buffer_pos to zero, but this
is not always correct.
To avoid losing any audio periods during suspend/resume and to prevent
clip sound, save the completed DMA buffer position in the DMA callback and
reinitialize dma_buffer_pos on resume.
Cc: stable@vger.kernel.org
Fixes: 1fc778f7c8 ("ASoC: renesas: rz-ssi: Add suspend to RAM support")
Signed-off-by: Claudiu Beznea <claudiu.beznea.uj@bp.renesas.com>
Link: https://patch.msgid.link/20251029141134.2556926-3-claudiu.beznea.uj@bp.renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When configured for default synchronisation (Rx syncs to Tx) and the
SAI operates in consumer mode (clocks provided externally to Tx), a
synchronisation error occurs on Tx on the first attempt after device
initialisation when the playback stream is started while a capture
stream is already active. This results in channel shift/swap on the
playback stream.
Subsequent streams (ie after that first failing one) always work
correctly, no matter the order, with or without the other stream active.
This issue was observed (and fix tested) on an i.MX6UL board connected
to an ADAU1761 codec, where the codec provides both frame and bit clock
(connected to TX pins).
To fix this, always initialize the 'other' xCR4 and xCR5 registers when
we're starting a stream which is synced to the opposite one, irregardless
of the producer/consumer status.
Fixes: 51659ca069 ("ASoC: fsl-sai: set xCR4/xCR5/xMR for SAI master mode")
Signed-off-by: Maarten Zanders <maarten@zanders.be>
Reviewed-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20251024135716.584265-1-maarten@zanders.be
Signed-off-by: Mark Brown <broonie@kernel.org>
Revert 3293d3d7b0 ("ASoC: sdw_utils: add name_prefix for rt1321 part
id") due to dependencies on -next which for some reason don't show up
in my builds.
Signed-off-by: Mark Brown <broonie@kernel.org>
The log messages for the PreSonus STUDIO 1810c about
device_setup are not applicable to the 1824c, and should
not be logged when 1824c initializes.
Refactor from if statement to switch statement as there
might be more STUDIO series devices added later.
Fixes: 080564558e ("ALSA: usb-audio: enable support for Presonus Studio 1824c within 1810c file")
Signed-off-by: Roy Vegard Ovesen <roy.vegard.ovesen@gmail.com>
Link: https://patch.msgid.link/aPaYTP7ceuABf8c7@ark
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Shengjiu Wang <shengjiu.wang@nxp.com>:
The DSD little endian format requires the msb first, the previous
understanding is not correct. The issue is found by testing with
pipewire.
Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
Small set of patches two of which fix problems observed during shutdown
and XRUN scenarios for PCM streaming. These ensure HDAudio HOST stream
is reset and re-setup during XRUNs and synchronize avs_dai_fe_shutdown()
and period-elapsed work (thread) so that no slab-use-after-free panics
occur.
The last change makes the 'format' parameter provided by a
compress-application taken into account - currently its ignored. Such
approach helps us increase the coverage of data probing (debug)
functionality.
Both mt8195-afe-pcm and mt8365-afe-pcm drivers use devm_pm_runtime_enable()
in probe function, which automatically calls pm_runtime_disable() on device
removal via devres mechanism. However, the remove callbacks explicitly call
pm_runtime_disable() again, resulting in double pm_runtime_disable() calls.
Fix by removing the redundant pm_runtime_disable() calls from remove
functions, letting the devres framework handle it automatically.
Fixes: 2ca0ec01d4 ("ASoC: mediatek: mt8195-afe-pcm: Simplify runtime PM during probe")
Fixes: e1991d102b ("ASoC: mediatek: mt8365: Add the AFE driver support")
Signed-off-by: Haotian Zhang <vulab@iscas.ac.cn>
Link: https://patch.msgid.link/20251020170440.585-1-vulab@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
The data probing is a debug feature. Currently parameters channels and
rate specified by the application are read while the format is ignored.
More robust approach is to read all of them.
Audio format, while not used by the Probe module for PCM streaming,
takes part in the gateway initialization on the DSP side. With full
parametrization we gain better coverage with the data probing feature.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20251023092348.3119313-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
avs_dai_fe_shutdown() handles the shutdown procedure for HOST HDAudio
stream while period-elapsed work services its IRQs. As the former
frees the DAI's private context, these two operations shall be
synchronized to avoid slab-use-after-free or worse errors.
Fixes: 0dbb186c35 ("ASoC: Intel: avs: Update stream status in a separate thread")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20251023092348.3119313-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The pcm->prepare() function may be called multiple times in a row by the
userspace, as mentioned in the documentation. The driver shall take that
into account and prevent redundancy. However, the exact same function is
called during XRUNs and in such case, the particular stream shall be
reset and setup anew.
Fixes: 9114700b49 ("ASoC: Intel: avs: Generic PCM FE operations")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20251023092348.3119313-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Vitaly Rodionov <vitalyr@opensource.cirrus.com>:
This patch series introduces DAC, CODEC, and SPI control bus support
for Cirrus Logic CS530x variants, along with general code cleanup
and resolution of checkpatch.pl warnings.
Changes since v1,v2,v3:
- Signed off all patches
- Splitted "tidy up" pach in 3 separate simple patches
- Fixed commit subject to much preferred subject prefix for binding patches.
- Moved dt-bindings related patch down the chain
- Added all relevant maintainers to CC list
Simon Trimmer (4):
ASoC: cs530x: Correct log message with expected variable
ASoC: cs530x: Add CODEC and DAC support
ASoC: cs530x: Check the DEVID matches the devtype
ASoC: cs530x: Rename i2c related structures
Vitaly Rodionov (7):
ASoC: cs530x: Update the copyright headers
ASoC: cs530x: Sort #include directives and tidy up whitespaces
ASoC: cs530x: Remove unused struct members and constants
ASoC: cs530x: Correct constant naming
ASoC: dt-bindings: sound: cirrus: cs530x: Add cs530x variants
ASoC: cs530x: Correct MCLK reference frequency values
ASoC: cs530x: Add SPI bus support for cs530x parts
.../bindings/sound/cirrus,cs530x.yaml | 4 +
sound/soc/codecs/Kconfig | 10 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/cs530x-i2c.c | 24 +-
sound/soc/codecs/cs530x-spi.c | 92 ++++
sound/soc/codecs/cs530x.c | 516 +++++++++++++++---
sound/soc/codecs/cs530x.h | 120 ++--
7 files changed, 634 insertions(+), 134 deletions(-)
create mode 100644 sound/soc/codecs/cs530x-spi.c
--
2.43.0
Removing this match entry ensures that a PTL system comprising of a
cs42l43 codec on link3 will use function topologies.
Previously the behaviour would be use the monolithic topology associated
with this codec match table entry in preference to function topologies
and if the system had a number of smart amplifiers then they would not
be instantiated.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Link: https://patch.msgid.link/20251016112601.187020-1-simont@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The filter configuration register(0x26) has AHPF(bit6) for primary record
path, which is common in max98090 and max98091 and has been defined as
DAPM suppy widget as "AHPF" in
"struct snd_soc_dapm_widget max98090_dapm_widget[]".
It is the DC-Blocking filter for the primary record path.
But the same functionality for secondary record path in the configuration
register(0x26) is DMIC2_HPF(bit2). It is not present as a DAPM supply
widget in the current code. With this patch adding it as a DAPM supply
widget. In the current code, the mics on secondary record path in
code are named as "DMIC3" and "DMIC4", so accordingly naming
DMIC2_HPF(bit2) as "DMIC34_HPF", and declaring it as a DAPM supply
widget in "struct snd_soc_dapm_widget max98091_dapm_widget[]". Also it
is specific to max98091, and should be visible or working only when
max98091 codec chip is used. Therefore, written in
"max98091_dapm_widget[]". As "AHPF" is not part
of secondary record path, replacing it with "DMIC34_HPF" in the
ALSA routes to "DMIC3" and "DMIC4" in "max98091_dapm_routes[]".
Signed-off-by: Sharique Mohammad <sharq0406@gmail.com>
Link: https://patch.msgid.link/20251016151152.1107083-1-sharq0406@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds some match entries for a few system configurations:
cs42l43 link 0 UID 0
cs35l56 link 1 UID 0
cs35l56 link 1 UID 1
cs35l56 link 1 UID 2
cs35l56 link 1 UID 3
cs42l43 link 1 UID 0
cs35l56 link 1 UID 0
cs35l56 link 1 UID 1
cs35l56 link 1 UID 2
cs35l56 link 1 UID 3
cs35l56 link 1 UID 0
cs35l56 link 1 UID 1
cs35l56 link 1 UID 2
cs35l56 link 1 UID 3
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Link: https://patch.msgid.link/20251016150837.320886-1-simont@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This laptop requires the same quirk as Lenovo Yoga9 14IAP7 for
fixing the bass speaker problems.
Use HDA_CODEC_QUIRK to match on the codec SSID to avoid conflict with
the Lenovo Legion Slim 7 16IRH8, which has the same PCI SSID.
Signed-off-by: J-Donald Tournier <jdtournier@gmail.com>
Link: https://patch.msgid.link/20251018145322.39119-1-jdournier@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
