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ALSA controls should return 1 if the value in the control changed but the
control put operation fsl_easrc_iec958_put_bits() unconditionally returns
0, causing ALSA to not generate any change events. This is detected by
mixer-test with large numbers of messages in the form:
No event generated for Context 3 IEC958 CS5
Context 3 IEC958 CS5.0 orig 5224 read 5225, is_volatile 0
Add a suitable check.
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://patch.msgid.link/20260205-asoc-fsl-easrc-fix-events-v1-1-39d4c766918b@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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In the TDM format the ADC and DAC serial ports will only operate as a
slave, the sysclk should not be less than 256FS and Quad-Speed Mode (100
to 200 kHz sample rates) is not supported by ADC. So add error checks
for these constraints.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/20260225100907.686470-1-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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'struct codec_connection' are not modified in this driver.
Constifying these structures moves some data to a read-only section, so
increases overall security.
On a x86_64, with allmodconfig:
Before:
======
text data bss dec hex filename
10034 3392 12 13438 347e sound/aoa/fabrics/layout.o
After:
=====
text data bss dec hex filename
10370 3040 12 13422 346e sound/aoa/fabrics/layout.o
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Link: https://patch.msgid.link/4009c337cc1a1a57795562279270c03687973b3b.1772138640.git.christophe.jaillet@wanadoo.fr
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The put() operation is expected to return:
1) 0 on success if no changes were made
2) 1 on success if changes were made
3) error code otherwise
Currently 2) is usually ignored when writing control-operations. While
forcing compliance is not an option right now, make it easier for
developers to adhere to the expectations and notice problems by logging
them when CONFIG_SND_CTL_DEBUG is enabled.
Due to large size of struct snd_ctl_elem_value, 'value_buf' is provided
as a reusable buffer for kctl->put() verification. This prevents
exhausting the stack when verifying the operation.
>From user perspective, patch introduces a new trace/events category
'snd_ctl' containing a single 'snd_ctl_put' event type. Log sample:
amixer-1086 [003] ..... 8.035939: snd_ctl_put: success: expected=0, actual=0 for ctl numid=1, iface=MIXER, name='Master Playback Volume', index=0, device=0, subdevice=0, card=0
amixer-1087 [003] ..... 8.938721: snd_ctl_put: success: expected=1, actual=1 for ctl numid=1, iface=MIXER, name='Master Playback Volume', index=0, device=0, subdevice=0, card=0
amixer-1088 [003] ..... 9.631470: snd_ctl_put: success: expected=1, actual=1 for ctl numid=1, iface=MIXER, name='Master Playback Volume', index=0, device=0, subdevice=0, card=0
amixer-1089 [000] ..... 9.636786: snd_ctl_put: fail: expected=1, actual=0 for ctl numid=5, iface=MIXER, name='Loopback Mute', index=0, device=0, subdevice=0, card=0
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Mark Brown <broonie@kernel.org>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260224205619.584795-1-cezary.rojewski@intel.com
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The use of snprintf() may cause a warning with W=1 due to the possibly
truncated string. As the truncation doesn't really matter (and won't
happen practically) in the case of dice driver, just shut it up by
replacing with scnprintf().
Link: https://patch.msgid.link/20260227155705.1557224-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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calibrated-impedance micro-speaker on TAS2781
On TAS2781, if the Speaker calibrated impedance is lower than default
value hard-coded inside the TAS2781, it will cuase vol lower than
normal. In order to fix this issue, the parameter of SineGainI need
updating.
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Tested-by: Matthew Schwartz <matthew.schwartz@linux.dev>
Link: https://patch.msgid.link/20260227144641.1243-1-shenghao-ding@ti.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The HP Pavilion 15-eh1xxx series uses the HP mainboard 88D1 with ALC245
and needs the ALC245_FIXUP_HP_MUTE_LED_V1_COEFBIT quirk to make the
mute led working.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=215978
Cc: <stable@vger.kernel.org>
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Link: https://patch.msgid.link/20260227121327.3751341-1-zhangheng@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Setting up the interface when suspended/resumeing fail on this card.
Adding a reset and delay quirk will eliminate this problem.
usb 1-1: New USB device found, idVendor=0624, idProduct=3d3f
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 1-1: Product: AB13X USB Audio
usb 1-1: Manufacturer: Generic
usb 1-1: SerialNumber: 20210726905926
Signed-off-by: Lianqin Hu <hulianqin@vivo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/TYUPR06MB621795D087BF2D594027C235D273A@TYUPR06MB6217.apcprd06.prod.outlook.com
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HP/Speaker auto-detect (VNID_HP_ASEL) has been off by default for every
CA0132 device since the driver was added in 2012. vnode_lswitch is
always initialized to 0 in ca0132_init_chip(), and no quirk or other
code path enables it. As a result, headphone jack detection works only
after the user manually turns on "HP/Speaker Auto Detect" in alsamixer,
which is not obvious on laptops with combo jacks (e.g. Google Link,
Alienware).
Change the default to follow the headphone pin config: if the pin verb
has presence detect enabled (no AC_DEFCFG_MISC_NO_PRESENCE) and the
codec supports it (AC_PINCAP_PRES_DETECT), enable HP_ASEL by default.
This lets firmware (coreboot, UEFI, etc.) express whether the headphone
jack supports insertion detection. Devices with combo jacks can default
to auto-detect; devices with fixed/no jack leave it off.
Signed-off-by: Matt DeVillier <matt.devillier@gmail.com>
Link: https://patch.msgid.link/20260226163055.825167-1-matt.devillier@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge series from Charles Keepax <ckeepax@opensource.cirrus.com>:
Another fairly mixed bag of small SDCA fixes/improvements. Fix one DisCo
property that was treated as mandatory but is actually not present in
the first version of the specification. Fix the counting of routes for
SU/GE DAPM widgets, this currently makes assumptions that are not
guaranteed to be true which can result in too many/few DAPM routes.
Then finally a couple improvements to the volume controls, simplify the
mapping between ALSA and SDCA volumes and pull the volume stuff back
into the SDCA code. It just wasn't sitting right with me that it was
being handled in the ASoC core given it is unlikely to ever see any
reuse outside of SDCA.
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It is cleaner to keep the SDCA code contained and not update the core
code for things that are unlikely to see reuse outside of SDCA. Move the
Q7.8 volume helpers back into the SDCA core code.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260225140118.402695-5-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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SDCA measures volumes in 256ths of a dB, whereas ALSA measures
volumes in 100ths of a dB. Currently the SDCA volume controls are
mapped to ALSA controls by mapping the step size and working out
the number of steps for this mapped step size. Due to quantization
of the step size this means the number of steps in the ALSA
control will rarely match the number of steps in the SDCA control,
leading to skipped values and multiple values that map to the
same volume. This is not a huge problem, the volume is still
increasing and the differences will be small but it is not really
desirable.
It is simpler and more accurate to count the number of steps based on
the SDCA volume levels. This gives a 1-to-1 mapping between control
values and register volumes. The TLV is based on a minimum and maximum
volume so still accurately specifies the volume range.
Tested-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260225140118.402695-4-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Device Layer Selector Unit's are controlled by a Group Entity control
rather than by the host directly. For the purposes of the ASoC class
driver the number of input routes to the SU is controlled by the number
of options within the Group Entity Selected Mode Control. ie. One valid
DAPM route for each valid route defined in the Group Entity.
Currently the code assumes that a Device Layer SU will have a number of
routes equal to the number of potential sources for the SU. ie. it
counts the routes using the SU, but then creates the routes using the
GE. However, this isn't actually true, it is perfectly allowed for the
GE to only define options for some of the potential sources of the SU.o
In such a case the number of routes return will not match those created,
leading to either an overflow of the routes array or undefined routes to
be past to the ASoC core, both of which generally lead to the sound card
failing to probe.
Update the handling for the counting of routes to count the connected
routes on the GE itself and then ignore the source routes on the SU.
This makes it match the logic generating the routes and ensuring that
both remain in sync.
Fixes: 2c8b3a8e6aa8 ("ASoC: SDCA: Create DAPM widgets and routes from DisCo")
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260225140118.402695-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add a default value for the function reset timeout since version 1.0
of the SDCA specification doesn't actually include this property, it
was added later.
Fixes: 7b6be935e7ef ("ASoC: SDCA: Parse Function Reset max delay")
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260225140118.402695-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The kernel-doc comments for sdca_irq_request() contained some typos
that lead to build warnings with W=1. Let's correct them.
Fixes: b126394d9ec6 ("ASoC: SDCA: Generic interrupt support")
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260226154753.1083320-1-tiwai@suse.de
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We don't process this driver code for kernel-doc, and the "/**" marker
leads to warnings with W=1 builds. Drop the superfluous markers, and
also fix the invalid mark up, too.
Link: https://patch.msgid.link/20260226155456.1092186-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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At fixing the memory leak of xfer buffer, we forgot to update the
corresponding comment, too. This resulted in a kernel-doc warning
with W=1. Let's correct it.
Fixes: 5c7ef5001292 ("ALSA: qc_audio_offload: avoid leaking xfer_buf allocation")
Link: https://patch.msgid.link/20260226154414.1081568-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We don't process USB-audio driver code for kernel-doc, and the "/**"
marker leads to warnings with W=1 builds. Drop the superfluous
markers.
Link: https://patch.msgid.link/20260226154414.1081568-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since commit af37511305c0 ("firmware: cs_dsp: Don't require client to
provide a struct cs_dsp_client_ops") the client doesn't have to provide
a struct cs_dsp_client_ops. So remove the dummy cs_dsp_client_ops.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260226124115.1811187-1-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v7.0
One quirk and a fix for handling of exotic peripherals on cs42l43.
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In cs35l56_hda_posture_put() assign ucontrol->value.integer.value[0] to
a long instead of an unsigned long. ucontrol->value.integer.value[0] is
a long.
This fixes the sparse warning:
sound/hda/codecs/side-codecs/cs35l56_hda.c:256:20: warning: unsigned value
that used to be signed checked against zero?
sound/hda/codecs/side-codecs/cs35l56_hda.c:252:29: signed value source
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 73cfbfa9caea8 ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Link: https://patch.msgid.link/20260226111728.1700431-1-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The entry of the validators table for UAC3 AC header descriptor is
defined with the wrong protocol version UAC_VERSION_2, while it should
have been UAC_VERSION_3. This results in the validator never matching
for actual UAC3 devices (protocol == UAC_VERSION_3), causing their
header descriptors to bypass validation entirely. A malicious USB
device presenting a truncated UAC3 header could exploit this to cause
out-of-bounds reads when the driver later accesses unvalidated
descriptor fields.
The bug was introduced in the same commit as the recently fixed UAC3
feature unit sub-type typo, and appears to be from the same copy-paste
error when the UAC3 section was created from the UAC2 section.
Fixes: 57f8770620e9 ("ALSA: usb-audio: More validations of descriptor units")
Cc: <stable@vger.kernel.org>
Signed-off-by: Jun Seo <jun.seo.93@proton.me>
Link: https://patch.msgid.link/20260226010820.36529-1-jun.seo.93@proton.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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fix mute/micmute LEDs and headset microphone for Acer Nitro ANV15-51.
[ The headset microphone issue is solved by Kailang]
Link: https://bugzilla.kernel.org/show_bug.cgi?id=220279
Cc: stable@vger.kernel.org
Signed-off-by: Zhang Heng <zhangheng@kylinos.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260209134149.3076957-1-zhangheng@kylinos.cn
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The default bdl_pos_adj of 32 for Nvidia HDA controllers is
insufficient on GA102 (and likely other recent Nvidia GPUs) after S3
suspend/resume. The controller's DMA timing degrades after resume,
causing premature IRQ detection in azx_position_ok() which results in
silent HDMI/DP audio output despite userspace reporting a valid
playback state and correct ELD data.
Increase bdl_pos_adj to 64 for AZX_DRIVER_NVIDIA, matching the value
already used by Intel Apollo Lake for the same class of timing issue.
Cc: stable@vger.kernel.org
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=221069
Suggested-by: Charalampos Mitrodimas <charmitro@posteo.net>
Signed-off-by: Panagiotis Foliadis <pfoliadis@posteo.net>
Link: https://patch.msgid.link/20260225-nvidia-audio-fix-v1-1-b1383c37ec49@posteo.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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commit d1965f008f22 ("ASoC: cs-amp-lib-test: Use faux bus instead of
creating a dummy platform device") replaced all use of platform device
with faux_bus but forgot to remove the header include.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260225112339.1179585-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
The CS47L47 is a SDCA smart codec with UAJ (headset, jack detect) and DMIC.
This series adds the initial support for the Cirrus Logic CS47L47 codec.
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Replace the remaining with inclusive terms; it's only this function
name we overlooked at the previous conversion.
Fixes: 53837b4ac2bd ("ALSA: usb-audio: Replace slave/master terms")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260225085233.316306-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Although DIYINHK USB Audio 2.0 (ID 20b1:2009) shows the implicit
feedback source for the capture stream, this would cause several
problems for the playback. Namely, the device can get wMaxPackSize
1024 for 24/32 bit format with 6 channels, and when a high sample rate
like 352.8kHz or 384kHz is played, the packet size overflows the max
limit. Also, the device has another two playback altsets, and those
aren't properly handled with the implicit feedback.
Since the device has been working well even before introducing the
implicit feedback, we can assume that it works fine in the async mode.
This patch adds the explicit skip of the implicit fb detection to make
the playback running in the async mode.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=221076
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260225085233.316306-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When the packet sizes are taken from the capture stream in the
implicit feedback mode, the sizes might be larger than the upper
boundary defined by the descriptor. As already done for other
transfer modes, we have to cap the sizes accordingly at sending,
otherwise this would lead to an error in USB core at submission of
URBs.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=221076
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260225085233.316306-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We calculate the possible packet sizes beforehand for adaptive and
synchronous endpoints, but we didn't take care of the max frame size
for those pre-calculated values. When a device or a bus limits the
packet size, a high sample rate or a high number of channels may lead
to the packet sizes that are larger than the given limit, which
results in an error from the USB core at submitting URBs.
As a simple workaround, just add the sanity checks of pre-calculated
packet sizes to have the upper boundary of ep->maxframesize.
Fixes: f0bd62b64016 ("ALSA: usb-audio: Improve frames size computation")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=221076
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20260225085233.316306-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add the SoundWire ID for CS47L47 to the class driver.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Link: https://patch.msgid.link/20260223150256.326143-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add a device info entry for the Cirrus Logic CS47L47.
CS47L47 has UAJ (headset speaker + mic + jack detect) and DMICs.
The audio ports are similar to the CS42L45 so can be based on the
CS42L45 code.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Link: https://patch.msgid.link/20260223150256.326143-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Merge series from Mingyou Chen <qby140326@gmail.com>:
Add DMIC support for the AMD RPL platforma.
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The audio enable GPIO is not toggled in any critical section where it
could not sleep, allow the audio enable GPIO to sleep. This allows the
driver to operate the audio enable GPIO connected to I2C GPIO expander.
Signed-off-by: Marek Vasut <marex@nabladev.com>
Link: https://patch.msgid.link/20260220202332.241035-1-marex@nabladev.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The ASUS ExpertBook BM1503CDA (Ryzen 5 7535U, Barcelo-R) has an
internal DMIC connected through the AMD ACP (Audio CoProcessor)
but is missing from the DMI quirk table, so the acp6x machine
driver probe returns -ENODEV and no DMIC capture device is created.
Add the DMI entry so the internal microphone works out of the box.
Signed-off-by: Azamat Almazbek uulu <almazbek1608@gmail.com>
Reviewed-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://patch.msgid.link/20260221114813.5610-1-almazbek1608@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Merge series from Yauhen Kharuzhy <jekhor@gmail.com>:
There are two Intel Cherry Trail-based devices using the RT5677 as a sound
codec: Lenovo Yoga Book YB1-X90 (Android tablet) and YB1-X91 (Windows
tablet).
They both have the same hardware configuration, but the X90 doesn't have
correct ACPI table definitions for many peripherals, whereas the X91 does.
Devices missing in the ACPI are defined in the board-specific driver
platform/x86/x86-android-tablets. In the X91 tablet, an ACPI _CRS method
for the RT5677 contains GPIO configuration entries which were not
supported by the codec driver before.
To support such device definitions, some modifications are added to the
RT5677 code: ACPI, SPI, and I2C matching ids have been introduced,
as well as some GPIO-related magic.
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Merge series from Luca Weiss <luca.weiss@fairphone.com>:
First a patch which fixes a logic error in the aw88261 driver. And then
we add "firmware-name" support for that driver to support loading
the device-specific firmware properly.
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For some exotic peripherals the type detect can return a reserved value
of 0x4. This will currently return an error and not report anything to
user-space, update this to report the insert normally.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260223093616.3800350-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add QUIRK_FLAG_SKIP_CLOCK_SELECTOR for Focusrite devices.
During interface parsing, snd_usb_clock_find_source() reads the clock
selector value then writes it back unchanged. On Focusrite devices
this redundant write results in a ~300ms delay per altsetting, adding
~1.8s to probe time on a typical device with 6 altsettings.
Enabling SKIP_CLOCK_SELECTOR skips the redundant write-back.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/00e53ae0a508b41516b41833daa17823381a649c.1771594828.git.g@b4.vu
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Add a quirk flag to skip the usb_set_interface(),
snd_usb_init_pitch(), and snd_usb_init_sample_rate() calls in
__snd_usb_parse_audio_interface(). These are redundant with
snd_usb_endpoint_prepare() at stream-open time.
Enable the quirk for Focusrite devices, as init_sample_rate(rate_max)
sets 192kHz during probing, which disables the internal mixer and Air
and Safe modes.
Fixes: 16f1f838442d ("Revert "ALSA: usb-audio: Drop superfluous interface setup at parsing"")
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/65a7909b15f9feb76c2a6f4f8814c240ddc50737.1771594828.git.g@b4.vu
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Remove QUIRK_FLAG_VALIDATE_RATES for Focusrite. With the previous
commit, focusrite_valid_sample_rate() produces correct rate tables
without USB probing.
QUIRK_FLAG_VALIDATE_RATES sends SET_CUR requests for each rate (~25ms
each) and leaves the device at 192kHz. This is a problem because that
rate: 1) disables the internal mixer, so outputs are silent until an
application opens the PCM and sets a lower rate, and 2) the Air and
Safe modes get disabled.
Fixes: 5963e5262180 ("ALSA: usb-audio: Enable rate validation for Scarlett devices")
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/09b9c012024c998c4ca14bd876ef0dce0d0b6101.1771594828.git.g@b4.vu
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Replace the bLength == 10 max_rate check in
focusrite_valid_sample_rate() with filtering that also examines the
bmControls VAL_ALT_SETTINGS bit.
When VAL_ALT_SETTINGS is readable, the device uses strict
per-altsetting rate filtering (only the highest rate pair for that
altsetting is valid). When it is not readable, all rates up to
max_rate are valid.
For devices without the bLength == 10 Format Type descriptor extension
but with VAL_ALT_SETTINGS readable and multiple altsettings (only seen
in Scarlett 18i8 3rd Gen playback), fall back to the Focusrite
convention: alt 1 = 48kHz, alt 2 = 96kHz, alt 3 = 192kHz.
This produces correct rate tables for all tested Focusrite devices
(all Scarlett 2nd, 3rd, and 4th Gen, Clarett+, and Vocaster) using
only USB descriptors, allowing QUIRK_FLAG_VALIDATE_RATES to be removed
for Focusrite in the next commit.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/7e18c1f393a6ecb6fc75dd867a2c4dbe135e3e22.1771594828.git.g@b4.vu
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Similar to other Samsung laptops, NT950QCT also requires the
ALC298_FIXUP_SAMSUNG_AMP quirk applied.
Cc: <stable@vger.kernel.org>
Signed-off-by: Juhyung Park <qkrwngud825@gmail.com>
Link: https://patch.msgid.link/20260222122609.281191-2-qkrwngud825@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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(NT950QCG-X716)
There's no product named "Samsung Galaxy Flex Book".
Use the correct "Samsung Galaxy Book Flex" name.
Link: https://www.samsung.com/sec/support/model/NT950QCG-X716
Link: https://www.samsung.com/us/computing/galaxy-books/galaxy-book-flex/galaxy-book-flex-15-6-qled-512gb-storage-s-pen-included-np950qcg-k01us
Cc: <stable@vger.kernel.org>
Signed-off-by: Juhyung Park <qkrwngud825@gmail.com>
Link: https://patch.msgid.link/20260222122609.281191-1-qkrwngud825@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Acer Aspire V3-572G has a combo jack (ALC283) but the BIOS
sets pin 0x19 to 0x411111f0 (not connected), so the headset mic
is not detected.
Add a quirk to override pin 0x19 as a headset mic and enable
headset mode.
Cc: stable@vger.kernel.org
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=221075
Suggested-by: Charalampos Mitrodimas <charmitro@posteo.net>
Signed-off-by: Panagiotis Foliadis <pfoliadis@posteo.net>
Reviewed-by: Charalampos Mitrodimas <charmitro@posteo.net>
Link: https://patch.msgid.link/20260221-fix-detect-mic-v1-1-b6e427b5275d@posteo.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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scarlett2_add_dsp_ctls() was incorrectly storing the precomp and PEQ
filter coefficient control pointers into the precomp_flt_switch_ctls
and peq_flt_switch_ctls arrays instead of the intended targets
precomp_flt_ctls and peq_flt_ctls. Pass NULL instead, as the filter
coefficient control pointers are not used, and remove the unused
precomp_flt_ctls and peq_flt_ctls arrays from struct scarlett2_data.
Additionally, scarlett2_update_filter_values() was reading
dsp_input_count * peq_flt_count values for
SCARLETT2_CONFIG_PEQ_FLT_SWITCH, but the peq_flt_switch array is
indexed only by dsp_input_count (one switch per DSP input, not per
filter). Fix the read count.
Fixes: b64678eb4e70 ("ALSA: scarlett2: Add DSP controls")
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://patch.msgid.link/86497b71db060677d97c38a6ce5f89bb3b25361b.1771581197.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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On Star Labs StarFighter (Realtek ALC233/235), the internal speakers can
emit an audible pop when entering or leaving runtime suspend.
Mute the speaker output paths via snd_hda_gen_shutup_speakers() in the
Realtek shutup callback before the codec is powered down.
This is enough to avoid the pop without special EAPD handling.
Test results:
- runtime PM pop fixed
- still reaches D3 (PCI 0000:00:1f.3 power_state=D3hot)
- does not address pops on cold boot (G3 exit) or around display manager
start/shutdown
journalctl -k (boot):
- snd_hda_codec_alc269 hdaudioC0D0: ALC233: picked fixup for PCI SSID
7017:2014
- snd_hda_codec_alc269 hdaudioC0D0: autoconfig for ALC233: line_outs=1
(0x1b/0x0/0x0/0x0/0x0) type:speaker
Suggested-by: Takashi Iwai <tiwai@suse.com>
Tested-by: Sean Rhodes <sean@starlabs.systems>
Signed-off-by: Sean Rhodes <sean@starlabs.systems>
Link: https://patch.msgid.link/4d5fb71b132bb283fd41c622b8413770b2065242.1771532060.git.sean@starlabs.systems
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When simple-audio-card programs sysclk for CPU and codec DAIs during
hw_params, the ordering of these calls may matter on some platforms.
Some CPU DAIs finalize or adjust the MCLK rate as part of their
set_sysclk() callback (for example by calling clk_set_rate()). If the
codec sysclk is configured before the CPU DAI applies the final MCLK
rate, the codec may configure its internal clocking based on a
non-final MCLK value.
Such situations can arise depending on the clock provider/consumer
relationship between the CPU DAI and the codec.
Introduce an explicit sysclk ordering enum in simple-card-utils and use
it to control the order of snd_soc_dai_set_sysclk() calls in the mclk-fs
handling path. The default behaviour remains unchanged (codec-first)
to avoid regressions.
Signed-off-by: Stefano Radaelli <stefano.r@variscite.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/20260213150355.442609-1-stefano.r@variscite.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The McASP driver supports runtime PM callbacks for register save/restore
during device idle, but doesn't provide system suspend/resume callbacks.
This causes audio to fail to resume after system suspend.
Since the driver already handles runtime suspend & resume, we can reuse
existing runtime PM logics.
Signed-off-by: Sen Wang <sen@ti.com>
Link: https://patch.msgid.link/20260211221001.155843-1-sen@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Current soc-compress.c clears symmetric_rate, but it clears rate only,
not clear other symmetric_channels/sample_bits.
static int soc_compr_clean(...)
{
...
if (!snd_soc_dai_active(cpu_dai))
=> cpu_dai->symmetric_rate = 0;
if (!snd_soc_dai_active(codec_dai))
=> codec_dai->symmetric_rate = 0;
...
};
This feature was added when v3.7 kernel [1], and there was only
symmetric_rate, no symmetric_channels/sample_bits in that timing.
symmetric_channels/sample_bits were added in v3.14 [2],
but I guess it didn't notice that soc-compress.c is updating symmetric_xxx.
We are clearing symmetry_xxx by soc_pcm_set_dai_params(), but is soc-pcm.c
local function. Makes it global function and clear symmetry_xxx by it.
[1] commit 1245b7005de02 ("ASoC: add compress stream support")
[2] commit 3635bf09a89cf ("ASoC: soc-pcm: add symmetry for channels and
sample bits")
Fixes: 3635bf09a89c ("ASoC: soc-pcm: add symmetry for channels and sample bits")
Cc: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87ms15e3kv.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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