commit 70b4db7d25 upstream.
The recent fix commit for addressing the OOB access of PCM URB data
buffer caused a regression on Behringer UMC2020HD device, resulting in
choppy sound. The fix used ep->max_urb_frames for the upper limit
check, and this is no right value to be referred.
Use the actual buffer size (ctx->buffer_size) as the upper limit
instead, which also avoids the regression on the device above.
Fixes: ef5749ef8b ("ALSA: usb-audio: Prevent excessive number of frames")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=220997
Link: https://patch.msgid.link/20260121082025.718748-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1aaedafb21 upstream.
Add a PCI quirk to enable microphone detection on the headphone jack of
TongFang X6AR55xU devices.
The former quirk entry did not acomplish this and is removed.
Fixes: b48fe9af1e ("ALSA: hda/realtek: Fix headset mic for TongFang X6AR55xU")
Signed-off-by: Tim Guttzeit <t.guttzeit@tuxedocomputers.com>
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com>
Link: https://patch.msgid.link/20260123221233.28273-1-wse@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 124bdc6ecc ]
When the support for the Sound Blaster X-Fi Surround 5.1 Pro was added,
the existing logic for the X-Fi Surround 5.1 in snd_audigy2nx_led_put()
was broken due to missing *else* before the added *if*: snd_usb_ctl_msg()
became incorrectly called twice and an error from first snd_usb_ctl_msg()
call ignored. As the added snd_usb_ctl_msg() call was totally identical
to the existing one for the "plain" X-Fi Surround 5.1, just merge those
two *if* statements while fixing the broken logic...
Found by Linux Verification Center (linuxtesting.org) with the Svace static
analysis tool.
Fixes: 7cdd8d7313 ("ALSA: usb-audio - Add support for USB X-Fi S51 Pro")
Signed-off-by: Sergey Shtylyov <s.shtylyov@auroraos.dev>
Link: https://patch.msgid.link/20260203161558.18680-1-s.shtylyov@auroraos.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 64e0924ed3 ]
The amp/speakers on the Lenovo Yoga Book 9i 13IRU8 laptop aren't
fully powered up, resulting in horrible tinny sound by default.
The kernel has an existing quirk for PCI SSID 0x17aa3843 which
matches this machine and several others. The quirk applies the
ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP fixup, however the fixup does not
work on this machine.
This patch modifies the existing quirk by adding a check for the
subsystem ID 0x17aa3881. If present, ALC287_FIXUP_TAS2781_I2C will
be applied instead of ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP. With this
change the TAS2781 amp is powered up, firmware is downloaded and
recognised by HDA/SOF - i.e. all is good, and we can boogie.
Code is re-used from alc298_fixup_lenovo_c940_duet7(), which fixes a
similar problem with two other Lenovo laptops.
Cross checked against ALSA cardinfo database for potential clashes.
Tested against 6.18.5 kernel built with Arch Linux default options.
Tested in HDA mode and SOF mode.
Note: Possible further work required to address quality of life issues
caused by the firmware's agressive power saving, and to improve ALSA
control mappings.
Signed-off-by: Martin Hamilton <m@martinh.net>
Link: https://patch.msgid.link/20260122-alc269-yogabook9i-fixup-v1-1-a6883429400f@martinh.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit b48fe9af1e ]
Add a PCI quirk to enable microphone detection on the headphone jack of
TongFang X6AR55xU devices.
Signed-off-by: Tim Guttzeit <t.guttzeit@tuxedocomputers.com>
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com>
Link: https://patch.msgid.link/20260119151626.35481-1-wse@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 49985bc466 ]
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=f06b
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 1-1: Product: MOONDROP Moonriver2 Ti
usb 1-1: Manufacturer: MOONDROP
usb 1-1: SerialNumber: MOONDROP Moonriver2 Ti
Signed-off-by: Lianqin Hu <hulianqin@vivo.com>
Reviewed-by: Cryolitia PukNgae <cryolitia@uniontech.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/TYUPR06MB6217911EFC7E9224935FA507D28DA@TYUPR06MB6217.apcprd06.prod.outlook.com
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d89aad92cf ]
When scanning for the reset pin, we could get an -EPROBE_DEFER.
The driver would assume that no reset pin had been defined,
which would mean that the chip would never be powered.
Now we both respect any error we get from devm_gpiod_get_optional.
We also now properly report the missing GPIO definition when
'gpio_reset' is NULL.
Signed-off-by: Dimitrios Katsaros <patcherwork@gmail.com>
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Link: https://patch.msgid.link/20260113-sound-soc-codecs-tvl320adcx140-v4-3-8f7ecec525c8@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 8e29db1b08 ]
Add DMI match for ASUSTeK COMPUTER INC. M6500RE to enable the
internal microphone.
Signed-off-by: Radhi Bajahaw <bajahawradhi@gmail.com>
Link: https://patch.msgid.link/20260112203814.155-1-bajahawradhi@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ef5749ef8b ]
In this case, the user constructed the parameters with maxpacksize 40
for rate 22050 / pps 1000, and packsize[0] 22 packsize[1] 23. The buffer
size for each data URB is maxpacksize * packets, which in this example
is 40 * 6 = 240; When the user performs a write operation to send audio
data into the ALSA PCM playback stream, the calculated number of frames
is packsize[0] * packets = 264, which exceeds the allocated URB buffer
size, triggering the out-of-bounds (OOB) issue reported by syzbot [1].
Added a check for the number of single data URB frames when calculating
the number of frames to prevent [1].
[1]
BUG: KASAN: slab-out-of-bounds in copy_to_urb+0x261/0x460 sound/usb/pcm.c:1487
Write of size 264 at addr ffff88804337e800 by task syz.0.17/5506
Call Trace:
copy_to_urb+0x261/0x460 sound/usb/pcm.c:1487
prepare_playback_urb+0x953/0x13d0 sound/usb/pcm.c:1611
prepare_outbound_urb+0x377/0xc50 sound/usb/endpoint.c:333
Reported-by: syzbot+6db0415d6d5c635f72cb@syzkaller.appspotmail.com
Closes: https://syzkaller.appspot.com/bug?extid=6db0415d6d5c635f72cb
Tested-by: syzbot+6db0415d6d5c635f72cb@syzkaller.appspotmail.com
Signed-off-by: Edward Adam Davis <eadavis@qq.com>
Link: https://patch.msgid.link/tencent_9AECE6CD2C7A826D902D696C289724E8120A@qq.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 22a507d768 ]
Even the device node don't exist, the graph_util_parse_link_direction()
will overwrite the playback_only and capture_only to be zero. Which
cause the playback_only and capture_only are not correct, so check device
node exist or not before update the value.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/20251229090432.3964848-1-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5b577d214f ]
The davinci_evm_probe() function calls of_parse_phandle() to acquire
device nodes for "ti,audio-codec" and "ti,mcasp-controller". These
functions return device nodes with incremented reference counts.
However, in several error paths (e.g., when the second of_parse_phandle(),
snd_soc_of_parse_card_name(), or devm_snd_soc_register_card() fails),
the function returns directly without releasing the acquired nodes,
leading to reference leaks.
This patch adds an error handling path 'err_put' to properly release
the device nodes using of_node_put() and clean up the pointers when
an error occurs.
Signed-off-by: Kery Qi <qikeyu2017@gmail.com>
Link: https://patch.msgid.link/20260107154836.1521-2-qikeyu2017@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 12cacdfb02 ]
Add missing quirks for some new Dell laptops using cs42l43's speaker
outputs.
Signed-off-by: Deep Harsora <Deep_Harsora@dell.com>
Signed-off-by: Maciej Strozek <mstrozek@opensource.cirrus.com>
Link: https://patch.msgid.link/20260102152132.3053106-1-mstrozek@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 9ed7a28225 ]
HP Laptop 15s-eq1xxx with ALC236 codec does not enable the
mute LED automatically. This patch adds a quirk entry for
subsystem ID 0x8706 using the ALC236_FIXUP_HP_MUTE_LED_COEFBIT2
fixup, enabling correct mute LED behavior.
Signed-off-by: Ruslan Krupitsa <krupitsarus@outlook.com>
Link: https://patch.msgid.link/AS8P194MB112895B8EC2D87D53A876085BBBAA@AS8P194MB1128.EURP194.PROD.OUTLOOK.COM
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 826af7fa62 upstream.
The PCM trigger callback of aloop driver tries to check the PCM state
and stop the stream of the tied substream in the corresponding cable.
Since both check and stop operations are performed outside the cable
lock, this may result in UAF when a program attempts to trigger
frequently while opening/closing the tied stream, as spotted by
fuzzers.
For addressing the UAF, this patch changes two things:
- It covers the most of code in loopback_check_format() with
cable->lock spinlock, and add the proper NULL checks. This avoids
already some racy accesses.
- In addition, now we try to check the state of the capture PCM stream
that may be stopped in this function, which was the major pain point
leading to UAF.
Reported-by: syzbot+5f8f3acdee1ec7a7ef7b@syzkaller.appspotmail.com
Closes: https://lore.kernel.org/69783ba1.050a0220.c9109.0011.GAE@google.com
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20260203141003.116584-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9210f5ff63 upstream.
imx-card currently sets the slot width to the physical sample width
for I2S links. This breaks controllers that use fixed-width slots
(e.g. 32-bit FIFO words), causing the unused bits in the slot to
contain undefined data when playing 16-bit streams.
Do not override the slot width in the machine driver and let the CPU
DAI select an appropriate default instead. This matches the behavior
of simple-audio-card and avoids embedding controller-specific policy
in the machine driver.
On an i.MX8MP-based board using SAI as the I2S master with 32-bit slots,
playing 16-bit audio resulted in spurious frequencies and an incorrect
SAI data waveform, as the slot width was forced to 16 bits. After this
change, audio artifacts are eliminated and the 16-bit samples correctly
occupy the first half of the 32-bit slot, with the remaining bits padded
with zeroes.
Cc: stable@vger.kernel.org
Fixes: aa736700f4 ("ASoC: imx-card: Add imx-card machine driver")
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20260118205030.1532696-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 213c4e5126 ]
The headphone GPIO should be set to the inverse of speaker_en.
When speakers are enabled, headphones should be disabled and vice versa.
Currently both GPIOs are set to the same value (speaker_en), causing
audio to play through both speakers and headphones simultaneously
when headphones are plugged in.
Tested on Huawei Matebook (BOD-WXX9) with ES8336 codec.
Fixes: 6e1ff1459e ("ASoC: Intel: sof_es8336: support a separate gpio to control headphone")
Signed-off-by: Tagir Garaev <tgaraev653@gmail.com>
Link: https://patch.msgid.link/20260121152435.101698-1-tgaraev653@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 403a0591be ]
rt1320_2_group2_adr works with rt1320_1_group2_adr and the name_prefix
should be rt1320-2.
Fixes: ffe450cb6b ("ASoC: Intel: soc-acpi-intel-ptl-match: add rt713_vb_l3_rt1320_l12 support")
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://patch.msgid.link/20260123113809.2238766-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 930e69757b upstream.
When snd_usb_create_mixer() fails, snd_usb_mixer_free() frees
mixer->id_elems but the controls already added to the card still
reference the freed memory. Later when snd_card_register() runs,
the OSS mixer layer calls their callbacks and hits a use-after-free read.
Call trace:
get_ctl_value+0x63f/0x820 sound/usb/mixer.c:411
get_min_max_with_quirks.isra.0+0x240/0x1f40 sound/usb/mixer.c:1241
mixer_ctl_feature_info+0x26b/0x490 sound/usb/mixer.c:1381
snd_mixer_oss_build_test+0x174/0x3a0 sound/core/oss/mixer_oss.c:887
...
snd_card_register+0x4ed/0x6d0 sound/core/init.c:923
usb_audio_probe+0x5ef/0x2a90 sound/usb/card.c:1025
Fix by calling snd_ctl_remove() for all mixer controls before freeing
id_elems. We save the next pointer first because snd_ctl_remove()
frees the current element.
Fixes: 6639b6c236 ("[ALSA] usb-audio - add mixer control notifications")
Cc: stable@vger.kernel.org
Cc: Andrey Konovalov <andreyknvl@gmail.com>
Signed-off-by: Berk Cem Goksel <berkcgoksel@gmail.com>
Link: https://patch.msgid.link/20260120102855.7300-1-berkcgoksel@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6f5c69f72e upstream.
The scarlett2_usb_get_config() function has a logic error in the
endianness conversion code that can cause buffer overflows when
count > 1.
The code checks `if (size == 2)` where `size` is the total buffer size in
bytes, then loops `count` times treating each element as u16 (2 bytes).
This causes the loop to access `count * 2` bytes when the buffer only
has `size` bytes allocated.
Fix by checking the element size (config_item->size) instead of the
total buffer size. This ensures the endianness conversion matches the
actual element type.
Fixes: ac34df733d ("ALSA: usb-audio: scarlett2: Update get_config to do endian conversion")
Cc: stable@vger.kernel.org
Signed-off-by: Samasth Norway Ananda <samasth.norway.ananda@oracle.com>
Link: https://patch.msgid.link/20260117012706.1715574-1-samasth.norway.ananda@oracle.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 61006c540c upstream.
In the audio mixer handling code of ctxfi driver, the conf field is
used as a kind of loop index, and it's referred in the index callbacks
(amixer_index() and sum_index()).
As spotted recently by fuzzers, the current code causes OOB access at
those functions.
| UBSAN: array-index-out-of-bounds in /build/reproducible-path/linux-6.17.8/sound/pci/ctxfi/ctamixer.c:347:48
| index 8 is out of range for type 'unsigned char [8]'
After the analysis, the cause was found to be the lack of the proper
(re-)initialization of conj field.
This patch addresses those OOB accesses by adding the proper
initializations of the loop indices.
Reported-by: Salvatore Bonaccorso <carnil@debian.org>
Tested-by: Karsten Hohmeier <linux@hohmatik.de>
Closes: https://bugs.debian.org/1121535
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/all/aSk8KJI35H7gFru6@eldamar.lan/
Link: https://patch.msgid.link/20260119133212.189129-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 6b971191fc ]
On at least the HyperX Cloud III, the range is 18944 (-18944 -> 0 in
steps of 1), so the original check for 255 steps is definitely obsolete.
Let's give ourselves a little more headroom before we emit a warning.
Fixes: 80acefff3b ("ALSA: usb-audio - Add volume range check and warn if it too big")
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: linux-sound@vger.kernel.org
Signed-off-by: Arun Raghavan <arunr@valvesoftware.com>
Link: https://patch.msgid.link/20260116225804.3845935-1-arunr@valvesoftware.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 29d71b8a5a upstream.
The soundwire update_status() callback may be called multiple times with
the same ATTACHED status but initialisation should only be done when
transitioning from UNATTACHED to ATTACHED.
Fixes: a0aab9e140 ("ASoC: codecs: add wsa881x amplifier support")
Cc: stable@vger.kernel.org # 5.6
Cc: Srinivas Kandagatla <srini@kernel.org>
Signed-off-by: Johan Hovold <johan@kernel.org>
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@oss.qualcomm.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260102111413.9605-3-johan@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 46378ab9fc ]
The word length is the physical width of the channel slots. So the
hw_params would misconfigure when format width and physical width
doesn't match. Like S24_LE which has data width of 24 bits but physical
width of 32 bits. So if using asymmetric formats you will get a lot of
noise.
Fixes: 689c7655b5 ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family")
Signed-off-by: Emil Svendsen <emas@bang-olufsen.dk>
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Link: https://patch.msgid.link/20260113-sound-soc-codecs-tvl320adcx140-v4-4-8f7ecec525c8@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit be7664c81d ]
The "snd_soc_component" in "adcx140_priv" was only used once but never
set. It was only used for reaching "dev" which is already present in
"adcx140_priv".
Fixes: 4e82971f7b ("ASoC: tlv320adcx140: Add a new kcontrol")
Signed-off-by: Emil Svendsen <emas@bang-olufsen.dk>
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Link: https://patch.msgid.link/20260113-sound-soc-codecs-tvl320adcx140-v4-2-8f7ecec525c8@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 390caeed08 ]
The CS42L43 codec's load detection can return different impedance values
that map to either HEADPHONE or LINEOUT jack types. However, the
soc_jack_pins array only maps SND_JACK_HEADPHONE to the "Headphone" DAPM
pin, not SND_JACK_LINEOUT.
When headphones are detected with an impedance that maps to LINEOUT
(such as impedance value 0x2), the driver reports SND_JACK_LINEOUT.
Since this doesn't match the jack pin mask, the "Headphone" DAPM pin
is not activated, and no audio is routed to the headphone outputs.
Fix by adding SND_JACK_LINEOUT to the Headphone pin mask, so that both
headphone and line-out detection properly enable the headphone output
path.
This fixes no audio output on devices like the Lenovo ThinkPad P16 Gen 3
where headphones are detected with LINEOUT impedance.
Fixes: d74bad3b74 ("ASoC: intel: sof_sdw_cs42l43: Create separate jacks for hp and mic")
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Cole Leavitt <cole@unwrap.rs>
Link: https://patch.msgid.link/20260114025518.28519-1-cole@unwrap.rs
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6a0243c402 ]
Change the test suite name string to "snd-hda-cirrus-scodec-test".
It was incorrectly named "snd-hda-scodec-cs35l56-test", a leftover
from when the code under test was actually in the cs35l56 driver.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2144833e7b ("ALSA: hda: cirrus_scodec: Add KUnit test")
Link: https://patch.msgid.link/20260113134056.619051-1-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c5e96e54ec ]
Set gpiochip parent to the struct device of the dummy GPIO driver
so that the software node will be associated with the GPIO chip.
The recent commit e5d527be7e ("gpio: swnode: don't use the
swnode's name as the key for GPIO lookup") broke cirrus_scodec_test,
because the software node no longer gets associated with the GPIO
driver by name.
Instead, setting struct gpio_chip.parent to the owning struct device
will find the node using a normal fwnode lookup.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2144833e7b ("ALSA: hda: cirrus_scodec: Add KUnit test")
Link: https://patch.msgid.link/20260113130954.574670-1-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 49aadf830e upstream.
The soundwire update_status() callback may be called multiple times with
the same ATTACHED status but initialisation should only be done when
transitioning from UNATTACHED to ATTACHED.
This avoids repeated initialisation of the codecs during boot of
machines like the Lenovo ThinkPad X13s:
[ 11.614523] wsa883x-codec sdw:1:0:0217:0202:00:1: WSA883X Version 1_1, Variant: WSA8835_V2
[ 11.618022] wsa883x-codec sdw:1:0:0217:0202:00:1: WSA883X Version 1_1, Variant: WSA8835_V2
[ 11.621377] wsa883x-codec sdw:1:0:0217:0202:00:1: WSA883X Version 1_1, Variant: WSA8835_V2
[ 11.624065] wsa883x-codec sdw:1:0:0217:0202:00:1: WSA883X Version 1_1, Variant: WSA8835_V2
[ 11.631382] wsa883x-codec sdw:1:0:0217:0202:00:2: WSA883X Version 1_1, Variant: WSA8835_V2
[ 11.634424] wsa883x-codec sdw:1:0:0217:0202:00:2: WSA883X Version 1_1, Variant: WSA8835_V2
Fixes: 43b8c7dc85 ("ASoC: codecs: add wsa883x amplifier support")
Cc: stable@vger.kernel.org # 6.0
Cc: Srinivas Kandagatla <srini@kernel.org>
Signed-off-by: Johan Hovold <johan@kernel.org>
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@oss.qualcomm.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260102111413.9605-2-johan@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 120f3e6ff7 upstream.
The soundwire update_status() callback may be called multiple times with
the same ATTACHED status but initialisation should only be done when
transitioning from UNATTACHED to ATTACHED.
Fix the inverted hw_init flag which was set to false instead of true
after initialisation which defeats its purpose and may result in
repeated unnecessary initialisation.
Similarly, the initial state of the flag was also inverted so that the
codec would only be initialised and brought out of regmap cache only
mode if its status first transitions to UNATTACHED.
Fixes: aa21a7d4f6 ("ASoC: codecs: wsa884x: Add WSA884x family of speakers")
Cc: stable@vger.kernel.org # 6.5
Cc: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Johan Hovold <johan@kernel.org>
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@oss.qualcomm.com>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@oss.qualcomm.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260102111413.9605-4-johan@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 90ed688792 ]
Drivers does cache sync during runtime resume, setting all writable
registers. Not all writable registers are set in cache default, resulting
in the erorr message:
fsl-sai 30c30000.sai: using zero-initialized flat cache, this may cause
unexpected behavior
Fix this by adding missing writable register defaults.
Signed-off-by: Alexander Stein <alexander.stein@ew.tq-group.com>
Link: https://patch.msgid.link/20251216102246.676181-1-alexander.stein@ew.tq-group.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e2cb8ef037 ]
Add a DMI quirk for the Honor MagicBook X16 2025 laptop
fixing the issue where the internal microphone was
not detected.
Signed-off-by: Andrew Elantsev <elantsew.andrew@gmail.com>
Link: https://patch.msgid.link/20251210203800.142822-1-elantsew.andrew@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 57d508b5f7 ]
'version' is an enum, thus cast of pointer on 64-bit compile test with
clang W=1 causes:
rockchip_pdm.c:583:17: error: cast to smaller integer type 'enum rk_pdm_version' from 'const void *' [-Werror,-Wvoid-pointer-to-enum-cast]
This was already fixed in commit 49a4a8d126 ("ASoC: rockchip: Fix
Wvoid-pointer-to-enum-cast warning") but then got bad in
commit 9958d85968 ("ASoC: Use device_get_match_data()").
Discussion on LKML also pointed out that 'uintptr_t' is not the correct
type and either 'kernel_ulong_t' or 'unsigned long' should be used,
with several arguments towards the latter [1].
Link: https://lore.kernel.org/r/CAMuHMdX7t=mabqFE5O-Cii3REMuyaePHmqX+j_mqyrn6XXzsoA@mail.gmail.com/ [1]
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@oss.qualcomm.com>
Link: https://patch.msgid.link/20251203141644.106459-2-krzysztof.kozlowski@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 161a0c617a upstream.
When config table entries don't match with the device to be probed,
currently we fall back to SND_INTEL_DSP_DRIVER_ANY, which means to
allow any drivers to bind with it.
This was set so with the assumption (or hope) that all controller
drivers should cover the devices generally, but in practice, this
caused a problem as reported recently. Namely, when a specific
kconfig for SOF isn't set for the modern Intel chips like Alderlake,
a wrong driver (AVS) got probed and failed. This is because we have
entries like:
#if IS_ENABLED(CONFIG_SND_SOC_SOF_ALDERLAKE)
/* Alder Lake / Raptor Lake */
{
.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
.device = PCI_DEVICE_ID_INTEL_HDA_ADL_S,
},
....
#endif
so this entry is effective only when CONFIG_SND_SOC_SOF_ALDERLAKE is
set. If not set, there is no matching entry, hence it returns
SND_INTEL_DSP_DRIVER_ANY as fallback. OTOH, if the kconfig is set, it
explicitly falls back to SND_INTEL_DSP_DRIVER_LEGACY when no DMIC or
SoundWire is found -- that was the working scenario. That being said,
the current setup may be broken for modern Intel chips that are
supposed to work with either SOF or legacy driver when the
corresponding kconfig were missing.
For addressing the problem above, this patch changes the fallback
driver to the legacy driver, i.e. return SND_INTEL_DSP_DRIVER_LEGACY
type as much as possible. When CONFIG_SND_HDA_INTEL is also disabled,
the fallback is set to SND_INTEL_DSP_DRIVER_ANY type, just to be sure.
Reported-by: Askar Safin <safinaskar@gmail.com>
Closes: https://lore.kernel.org/all/20251014034156.4480-1-safinaskar@gmail.com/
Tested-by: Askar Safin <safinaskar@gmail.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20251210131553.184404-1-tiwai@suse.de
Cc: Askar Safin <safinaskar@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e340663bbf upstream.
After speaker id retrieval was refactored to happen in tas2781_read_acpi,
devices that do not use a speaker id need a negative speaker_id value
instead of NULL, but no initialization was added to the TAS2563 code path.
This causes the driver to attempt to load a non-existent firmware file name
with a speaker id of 0 ("TAS2XXX38700.bin") instead of the correct file
name without a speaker id ("TAS2XXX3870.bin"), resulting in low volume and
these dmesg errors:
tas2781-hda i2c-INT8866:00: Direct firmware load for TAS2XXX38700.bin failed with error -2
tas2781-hda i2c-INT8866:00: tasdevice_dsp_parser: load TAS2XXX38700.bin error
tas2781-hda i2c-INT8866:00: dspfw load TAS2XXX38700.bin error
[...]
tas2781-hda i2c-INT8866:00: tasdevice_prmg_load: Firmware is NULL
Fix this by setting speaker_id to -1 as is done for other models.
Fixes: 945865a0dd ("ALSA: hda/tas2781: fix speaker id retrieval for multiple probes")
Cc: stable@vger.kernel.org
Signed-off-by: August Wikerfors <git@augustwikerfors.se>
Link: https://patch.msgid.link/20251222194704.87232-1-git@augustwikerfors.se
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 830988b6cf upstream.
If ac97_add_adapter() fails, put_device() is the correct way to drop
the device reference. kfree() is not required.
Add kfree() if idr_alloc() fails and in ac97_adapter_release() to do
the cleanup.
Found by code review.
Fixes: 74426fbff6 ("ALSA: ac97: add an ac97 bus")
Cc: stable@vger.kernel.org
Signed-off-by: Haoxiang Li <lihaoxiang@isrc.iscas.ac.cn>
Link: https://patch.msgid.link/20251219162845.657525-1-lihaoxiang@isrc.iscas.ac.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 81c53b52de upstream.
DSP expects the periods to be aligned to fragment sizes, currently
setting up to hw constriants on periods bytes is not going to work
correctly as we can endup with periods sizes aligned to 32 bytes however
not aligned to fragment size.
Update the constriants to use fragment size, and also set at step of
10ms for period size to accommodate DSP requirements of 10ms latency.
Fixes: 2a9e92d371 ("ASoC: qdsp6: q6asm: Add q6asm dai driver")
Cc: Stable@vger.kernel.org
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Tested-by: Alexey Klimov <alexey.klimov@linaro.org> # RB5, RB3
Link: https://patch.msgid.link/20251023102444.88158-4-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 74cc4f3ea4 upstream.
A matching Common object post processing instance is normally resused
across multiple streams. However currently we close this on DSP
even though there is a refcount on this copp object, this can result in
below error.
q6routing ab00000.remoteproc:glink-edge:apr:service@8:routing: Found Matching Copp 0x0
qcom-q6adm aprsvc:service:4:8: cmd = 0x10325 return error = 0x2
q6routing ab00000.remoteproc:glink-edge:apr:service@8:routing: DSP returned error[2]
q6routing ab00000.remoteproc:glink-edge:apr:service@8:routing: Found Matching Copp 0x0
qcom-q6adm aprsvc:service:4:8: cmd = 0x10325 return error = 0x2
q6routing ab00000.remoteproc:glink-edge:apr:service@8:routing: DSP returned error[2]
qcom-q6adm aprsvc:service:4:8: cmd = 0x10327 return error = 0x2
qcom-q6adm aprsvc:service:4:8: DSP returned error[2]
qcom-q6adm aprsvc:service:4:8: Failed to close copp -22
qcom-q6adm aprsvc:service:4:8: cmd = 0x10327 return error = 0x2
qcom-q6adm aprsvc:service:4:8: DSP returned error[2]
qcom-q6adm aprsvc:service:4:8: Failed to close copp -22
Fix this by addressing moving the adm_close to copp_kref destructor
callback.
Fixes: 7b20b2be51 ("ASoC: qdsp6: q6adm: Add q6adm driver")
Cc: Stable@vger.kernel.org
Reported-by: Martino Facchin <m.facchin@arduino.cc>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Tested-by: Alexey Klimov <alexey.klimov@linaro.org> # RB5, RB3
Link: https://patch.msgid.link/20251023102444.88158-3-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
