A bunch more small fixes here, plus some more of the constant stream of
quirks. The most notable change here is Richard's change to the cs_dsp
code for the KUnit tests which is relatively large, mostly due to
boilerplate. The tests were triggering large numbers of error messages
as part of verifying that problems with input data are appropriately
detected which in turn caused runtime issues for the framework due to
the performance impact of pushing the logging out, while the logging is
valuable in normal operation it's basically useless while doing tests
designed to trigger it so rate limiting is an appropriate fix.
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Merge tag 'asoc-fix-v6.19-rc8' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.19
A bunch more small fixes here, plus some more of the constant stream of
quirks. The most notable change here is Richard's change to the cs_dsp
code for the KUnit tests which is relatively large, mostly due to
boilerplate. The tests were triggering large numbers of error messages
as part of verifying that problems with input data are appropriately
detected which in turn caused runtime issues for the framework due to
the performance impact of pushing the logging out, while the logging is
valuable in normal operation it's basically useless while doing tests
designed to trigger it so rate limiting is an appropriate fix.
fsl_xcvr_activate_ctl() has
lockdep_assert_held(&card->snd_card->controls_rwsem),
but fsl_xcvr_mode_put() calls it without acquiring this lock.
Other callers of fsl_xcvr_activate_ctl() in fsl_xcvr_startup() and
fsl_xcvr_shutdown() properly acquire the lock with down_read()/up_read().
Add the missing down_read()/up_read() calls around fsl_xcvr_activate_ctl()
in fsl_xcvr_mode_put() to fix the lockdep assertion and prevent potential
race conditions when multiple userspace threads access the control.
Signed-off-by: Ziyi Guo <n7l8m4@u.northwestern.edu>
Link: https://patch.msgid.link/20260202174112.2018402-1-n7l8m4@u.northwestern.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
The link_mask variable is not changed after setting to
hdev->info.link_mask until it is used for another purpose to get the
used SoundWire links and set to mach->mach_params.links. Besides, the
link_mask variable should be reset before any link id is added to the
link_mask. To fix the issue above and avoid confusing, use the
hdev->info.link_mask variable directly to check if the SoundWire link
is enabled.
Fixes: 5226d19d4c ("ASoC: SOF: Intel: use sof_sdw as default SDW machine driver")
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/20260203072405.3716307-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The load detection process for 3-pole jacks requires slightly
updated reference values to ensure an accurate result. Update
the code to apply different tunings for the 3-pole and 4-pole
cases. This also updates the thresholds overall so update the
relevant comments to match.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260130150927.2964664-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add DMI entry for Huawei Matebook D (BOD-WXX9) with HEADPHONE_GPIO
and DMIC quirks.
This device has ES8336 codec with:
- GPIO 16 (headphone-enable) for headphone amplifier control
- GPIO 17 (speakers-enable) for speaker amplifier control
- GPIO 269 for jack detection IRQ
- 2-channel DMIC
Hardware investigation shows that both GPIO 16 and 17 are required
for proper audio routing, as headphones and speakers share the same
physical output (HPOL/HPOR) and are separated only via amplifier
enable signals.
RFC: Seeking advice on GPIO control issue:
GPIO values change in driver (gpiod_get_value() shows logical value
changes) but not physically (debugfs gpio shows no change). The same
gpiod_set_value_cansleep() calls work correctly in probe context with
msleep(), but fail when called from DAPM event callbacks.
Context information from diagnostics:
- in_atomic=0, in_interrupt=0, irqs_disabled=0
- Process context: pipewire
- GPIO 17 (speakers): changes in driver, no physical change
- GPIO 16 (headphone): changes in driver, no physical change
In Windows, audio switching works without visible GPIO changes,
suggesting possible ACPI/firmware involvement.
Any suggestions on how to properly control these GPIOs from DAPM
events would be appreciated.
Signed-off-by: Tagir Garaev <tgaraev653@gmail.com>
Link: https://patch.msgid.link/20260201121728.16597-1-tgaraev653@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a quirk for a Lenovo laptop (SSID: 0x17aa3821) to allow using sidecar
CS35L57 amps with CS42L43 codec.
Signed-off-by: Maciej Strozek <mstrozek@opensource.cirrus.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20260128092410.1540583-1-mstrozek@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A couple of small fixes and a couple of quirks, nothing major
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Merge tag 'asoc-fix-v6.19-rc7' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.19
A couple of small fixes and a couple of quirks, nothing major
Add a quirk for a Lenovo laptop (SSID: 0x17aa3821) to allow using sidecar
CS35L57 amps with CS42L43 codec.
Signed-off-by: Maciej Strozek <mstrozek@opensource.cirrus.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20260128092410.1540583-1-mstrozek@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
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>
A couple of device IDs and a couple of small fixes, nothing hugely
remarkable.
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Merge tag 'asoc-fix-v6.19-rc6' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.19
A couple of device IDs and a couple of small fixes, nothing hugely
remarkable.
If it play a 5s above silence media stream, it will cause silence
detection trigger.
Speaker will make no sound when you use another app to play a stream.
Add this patch will solve this issue.
GPIO2: Mute Hotkey GPIO3: Mic Mute LED
Enable this will turn on hotkey and LED support.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/f4929e137a7949238cc043d861a4d9f8@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
Add ASUS ExpertBook PM1503CDA to the DMI quirks table to enable
internal DMIC support via the ACP6x machine driver.
Signed-off-by: Anatolii Shirykalov <pipocavsobake@gmail.com>
Link: https://patch.msgid.link/20260119145618.3171435-1-pipocavsobake@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
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>
ASP_TX5 was incorrectly mapped to a channel value of 3 corrects,
the channel value of 4.
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Ricardo Rivera-Matos <rriveram@opensource.cirrus.com>
Link: https://patch.msgid.link/20260115192523.1335742-2-rriveram@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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
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>
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>
A moderately large collection of fixes since I missed a week, plus a few
new device IDs and quirks. It's all fairly minor, including a bunch of
work on the device tree bindings fixes which have no runtime effect.
There's one SoundWire change here exporting a symbol which was required
for a fix to the ASoC SoundWire code.
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Merge tag 'asoc-fix-v6.19-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.19
A moderately large collection of fixes since I missed a week, plus a few
new device IDs and quirks. It's all fairly minor, including a bunch of
work on the device tree bindings fixes which have no runtime effect.
There's one SoundWire change here exporting a symbol which was required
for a fix to the ASoC SoundWire code.
Merge patches that Pengutronix have been carrying in their tree for a
while and were upstreamed by Sascha Hauer together with some new
features that are going into the next release.
Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
This series fixes a problem with soc_sdw_utils.c calling the wrong
codec init callbacks, because it assumed that the DAI name could be
used to uniquely identify the codec. This isn't the case, especially
on SDCA which is a generic driver for many parts.
The first patch is needed to add a missing export to SoundWire core.
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>
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>
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>
Looking at section 8.6.1.1.69 in datasheets for both 5140 and 6140 (3140
doesn't support DRE). REG ADCX140_DSP_CFG1 BIT 3 field "DRE_AGC_SEL" it
select either DRE or AGC.
It states:
* 0 = DRE
* 1 = AGC
The control is called "DRE_ENABLE" and for it to be true it has to be
active low.
This commit will invert the control so "DRE_ENABLE" is active low.
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-v4-1-8f7ecec525c8@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
asoc_sdw_rtd_init() needs to call the rtd_init() callbacks for each
codec in a dailink. It was finding the codecs by looking for the
matching DAI name in codec_info_list[] but this isn't correct, because
the DAI name isn't guaranteed to be unique. Parts using the same codec
driver (so the same DAI names) might require different machine driver
setup.
Instead, get the struct sdw_slave and extract the SoundWire part ID.
Use this to lookup the entry in codec_info_list[]. This is the same
identity info that was used to find the entry when the machine driver
created the dailink.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e377c94773 ("ASoC: intel/sdw_utils: move soundwire codec_info_list structure")
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260112140758.215799-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
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>
Just like GA403U, this GA403W needs to remap woofers to DAC1. Similarly to
other Asus devices, headphones/headset MIC is not working, however the pin
config alone is not enough to fix it.
From Windows dump of GA403W:
0x12, 0x90a60140 # Correctly set by codec out of the box
0x13, 0x90a60550
0x14, 0x90170510
0x17, 0x90170120 # Correctly set by codec out of the box
0x19, 0x03a11050 # Set by ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC
0x1a, 0x411115F0
0x1b, 0x03a11c30 # Set by ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC
0x1d, 0x40663A45 # Correctly set by codec out of the box
0x21, 0x03211430
Even with all the values set, MIC of the jack is not detected. Until a
complete solution is found, set ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC
for GA403W which fixes audio volume control for woofers. No need to
create new quirk with missing pin config just yet, since its not
making the situation better.
Signed-off-by: Aleksandrs Vinarskis <alex@vinarskis.com>
Link: https://patch.msgid.link/20260112-asus-rog-audio-v1-1-513957b4704e@vinarskis.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
