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This has been quite a small release, there's a lot of driver specific
cleanups and minor enhancements but hardly anything on the core and only
one new driver. Highlights include:
- SoundWire support for AMD ACP 6.3 systems.
- Support for reporting version information for AVS firmware.
- Support DSPless mode for Intel Soundwire systems.
- Support for configuring CS35L56 amplifiers using EFI calibration
data.
- Log which component is being operated on as part of power management
trace events.
- Support for Microchip SAM9x7, NXP i.MX95 and Qualcomm WCD939x
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Merge tag 'asoc-v6.9' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v6.9
This has been quite a small release, there's a lot of driver specific
cleanups and minor enhancements but hardly anything on the core and only
one new driver. Highlights include:
- SoundWire support for AMD ACP 6.3 systems.
- Support for reporting version information for AVS firmware.
- Support DSPless mode for Intel Soundwire systems.
- Support for configuring CS35L56 amplifiers using EFI calibration
data.
- Log which component is being operated on as part of power management
trace events.
- Support for Microchip SAM9x7, NXP i.MX95 and Qualcomm WCD939x
The 4th Gen input preamp gain range is 0dB to +69dB, although the
control values range from 0 to 70. Replace SCARLETT2_MAX_GAIN with
SCARLETT2_MAX_GAIN_VALUE and SCARLETT2_MAX_GAIN_DB, and update the TLV
again.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: a45cf0a083 ("ALSA: scarlett2: Fix Scarlett 4th Gen input gain range")
Message-ID: <Ze7OMA8ntG7KteGa@m.b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input gain range TLV was declared as -70dB to 0dB, but the preamp
gain range is actually 0dB to +70dB. Rename SCARLETT2_GAIN_BIAS to
SCARLETT2_MAX_GAIN and update the TLV to fix.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: 0a995e38dc ("ALSA: scarlett2: Add support for software-controllable input gain")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <9168317b5ac5335943d3f14dbcd1cc2d9b2299d0.1710047969.git.g@b4.vu>
The meanings of the raw_auto_gain_status values were originally
guessed through experimentation, but the official names have now been
discovered. Update the autogain status control strings accordingly.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: 0a995e38dc ("ALSA: scarlett2: Add support for software-controllable input gain")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <8bd12a5e7dc714801dd9887c4bc5cb35c384e27c.1710047969.git.g@b4.vu>
The value currently being read to determine the low-voltage state is
actually the front panel state. Fix the code to use the correct offset
for the low-voltage state.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: d7cfa2fdfc ("ALSA: scarlett2: Add power status control")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <d97b7d87f43b0e54f37e1552394be2f3ae182704.1710047969.git.g@b4.vu>
After system_resume the amplifers will remain off, even if they were on
before system_suspend.
Use playback_started bool to save the playback state, and restore power
state based on it.
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <1742b61901781826f6e6212ffe1d21af542d134a.1709918447.git.soyer@irl.hu>
The runtime_resume function calls prmg_load and apply_calibration
functions, but system_resume also calls them, so calling
pm_runtime_force_resume before reset is unnecessary.
For consistency, do not call the pm_runtime_force_suspend in
system_suspend, as runtime_suspend does the same.
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <d0b4cc1248b9d375d59c009563da42d60d69eac3.1709918447.git.soyer@irl.hu>
The amplifier doesn't loose register state in software shutdown mode, so
there is no need to reset the cur_* values.
Without these resets, the amplifier can be turned on after
runtime_suspend without waiting for the program and
profile to be restored.
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <aa27ae084150988bf6a0ead7e3403bc485d790f8.1709918447.git.soyer@irl.hu>
The system_resume function uses dev_info for tracing, but the other pm
functions use dev_dbg.
Use dev_dbg as the other pm functions.
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <140f3c689c9eb5874e6eb48a570fcd8207f06a41.1709918447.git.soyer@irl.hu>
Realtek codec on HP Envy laptop series are heavily modified by vendor.
Therefore, need intervention to make it work properly. The patch fixes:
- B&O soundbar speakers (between lid and keyboard) activation
- Enable LED on mute button
- Add missing process coefficient which affects the output amplifier
- Volume control synchronization between B&O soundbar and side speakers
- Unmute headset output on several HP Envy models
- Auto-enable headset mic when plugged
This patch was tested on HP Envy x360 13-AR0107AU with Realtek ALC285
The only unsolved problem is output amplifier of all built-in speakers
is too weak, which causes volume of built-in speakers cannot be loud
as vendor's proprietary driver due to missing _DSD parameter in the
firmware. The solution is currently on research. Expected to has another
patch in the future.
Potential fix to related issues, need test before close those issues:
- https://bugzilla.kernel.org/show_bug.cgi?id=189331
- https://bugzilla.kernel.org/show_bug.cgi?id=216632
- https://bugzilla.kernel.org/show_bug.cgi?id=216311
- https://bugzilla.kernel.org/show_bug.cgi?id=213507
Signed-off-by: Athaariq Ardhiansyah <foss@athaariq.my.id>
Message-ID: <20240310140249.3695-1-foss@athaariq.my.id>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the ACPI HIDs and smi_node descriptions for the CS35L54 and CS35L57
Boosted Smart Amplifiers.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Acked-by: Ilpo Järvinen <ilpo.jarvinen@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240308135900.603192-4-rf@opensource.cirrus.com>
Add the HID for the CS35L54 and CS35L57 Boosted Smart Amplifiers. These
have the same control interface as the CS35L56 so are handled by the
cs35l56-hda driver.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240308135900.603192-3-rf@opensource.cirrus.com>
The CS35L54 and CS35L57 are Boosted Smart Amplifiers. The CS35L54 has
I2C/SPI control and I2S/TDM audio. The CS35L57 also has SoundWire
control and audio.
The hardware differences between L54, L56 and L57 do not affect the
driver control interface so they can all be handled by the same driver.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240308135900.603192-2-rf@opensource.cirrus.com>
Some more driver specific fixes for v6.8, plus one new x86 platform
quirk. All good fixes to have if you have systems that use the relevant
hardware.
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Merge tag 'asoc-fix-v6.8-rc7' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.8
Some more driver specific fixes for v6.8, plus one new x86 platform
quirk. All good fixes to have if you have systems that use the relevant
hardware.
PM constants for PCI devices are defined with bitwise annotation.
When used as is, sparse complains about that:
.../catpt/dsp.c:390:9: warning: restricted pci_power_t degrades to integer
.../catpt/dsp.c:414:9: warning: restricted pci_power_t degrades to integer
Force them to be u32 in the driver.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240307163734.3852754-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
hda_component.h uses hda_codec_dev from sound/hda_codec.h.
Include sound/hda_codec.h instead of assuming that it has already
been included by the parent .c file.
This isn't causing any problems with current code, so no need to
backport to older kernels.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Message-ID: <20240307111216.45053-2-rf@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the conventional include guards around the content of the
hda_component.h header file. This prevents double-declaration of
struct hda_component if the header gets included multiple times.
This isn't causing any problems with current code, so no need to
backport to older kernels.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Message-ID: <20240307111216.45053-1-rf@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Zhang Yi <zhangyi@everest-semi.com>:
We found that using 0x45 as the default value for interrupt-clk
would cause a headset detection error.So we took 0x00 as the default
value for interrupt-clk and passed the test.
We removed mic1-src and mic2-src, which were not used.
The kerneldoc for struct cs_dsp refers to a fw_file_name member but
there's no such member.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240307105516.40250-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Initialize debugfs_root to -ENODEV so that if the client never sets a
valid debugfs root the debugfs files will not be created.
A NULL pointer passed to any of the debugfs_create_*() functions means
"create in the root of debugfs". It doesn't mean "ignore".
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240307105353.40067-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We don't use mic1_src and mic2_src.so we delete these two members.
We changed the default value of interrupt-clk for headphone detection
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240307051222.24010-2-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Convert the RT1015 Mono Class D Audio Amplifier to DT schema.
Signed-off-by: Javier García <javier.gar.tab@gmail.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://msgid.link/r/20240304142315.14522-1-javier.gar.tab@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
intel-mid.h is providing some core parts of the South Complex PM,
which are usually are not used by individual drivers. In particular,
this driver doesn't use it, so simply remove the unused header.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://msgid.link/r/20240305160723.1363534-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The documented vendor prefix for LG Electronics is 'lg' not 'lge'. Just
change the example to 'lg' as there doesn't appear to be any dependency
on the existing compatible string.
Signed-off-by: Rob Herring <robh@kernel.org>
Link: https://msgid.link/r/20240305152131.3424326-1-robh@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The HP EliteBook using ALC236 codec which using 0x02 to
control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Signed-off-by: Andy Chi <andy.chi@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240304134033.773348-1-andy.chi@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC machine driver can use snd_soc_{of_}get_dlc() (A) to get DAI name
for dlc (snd_soc_dai_link_component). In this function call
dlc->dai_name is parsed via snd_soc_dai_name_get() (B).
(A) int snd_soc_get_dlc(...)
{
...
(B) dlc->dai_name = snd_soc_dai_name_get(dai);
...
}
(B) has a priority to return dai->name as dlc->dai_name. In most cases
card can probe successfully. However it has an issue that ASoC tries to
rebind card. Here is a simplified flow for example:
| a) Card probes successfully at first
| b) One of the component bound to this card is removed for some
| reason the component->dev is released
| c) That component is re-registered
v d) ASoC calls snd_soc_try_rebind_card()
a) points dlc->dai_name to dai->name. b) releases all resource of the
old DAI. c) creates new DAI structure. In result d) can not use
dlc->dai_name to add new created DAI.
So it's reasonable that prefer to return dai->driver->name in
snd_soc_dai_name_get() because dai->driver is a pre-defined global
variable. Also update snd_soc_is_matching_dai() for alignment.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240304072128.2845432-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a KUnit test for the cs-amp-lib library. This has test cases
for cs_amp_get_efi_calibration_data() and cs_amp_write_cal_coeffs().
A KUNIT_STATIC_STUB_REDIRECT() has been added to
cs_amp_get_efi_variable() and cs_amp_write_cal_coeff() so that the
KUnit test can redirect these to test harness functions.
Much of the testing involves invoking the same function with different
parameters, i.e. the number of amps and the amp index within the array.
This uses parameterization rather than looping. The idea is to avoid
looping over configurations within one test case as that has a higher
chance of having a bug that doesn't actually test all the expected cases.
Having the test run exactly one configuration, and then tear-down, is less
prone to accidentally skipped configurations.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240304143705.26362-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The HP Pavilion Aero Laptop 13-be2xxx(8BD6) requires a quirk entry for its internal microphone to function.
Signed-off-by: Al Raj Hassain <alrajhassain@gmail.com>
Reviewed-by: Mario Limonciello <mario.limonciello@amd.com>
Link: https://msgid.link/r/20240304103924.13673-1-alrajhassain@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Timing select registers for SRC and CMD are by default
referring to the corresponding SSI word select.
The calculation rule from HW spec skips SSI8, which has
no clock connection.
>From section 43.2.18 CMD Output Timing Select Register (CMDOUT_TIMSEL),
of R-Car Series, 3rd Generation Hardware User’s Manual Rev.2.20:
CMD0_OUT_DIVCLK_ Output Timing
SEL [4:0] Signal Select
B'0 0110: ssi_ws0
B'0 0111: ssi_ws1
B'0 1000: ssi_ws2
B'0 1001: ssi_ws3
B'0 1010: ssi_ws4
B'0 1011: ssi_ws5
B'0 1100: ssi_ws6
B'0 1101: ssi_ws7
<GAP>
B'0 1110: ssi_ws9
B'0 1111: Setting prohibited
Fix the erroneous prohibited setting of timsel value 1111 (0xf) for SSI9
by using timsel value 1110 (0xe) instead. This is possible because SSI8
is not connected as shown by <GAP> in the table above.
[21.695055] rcar_sound ec500000.sound: b adg[0]-CMDOUT_TIMSEL (32):00000f00/00000f1f
Correct the timsel assignment.
Fixes: 629509c5bc ("ASoC: rsnd: add Gen2 SRC and DMAEngine support")
Suggested-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Andreas Pape <Andreas.Pape4@bosch.com>
Signed-off-by: Yeswanth Rayapati <yeswanth.rayapati@in.bosch.com>
Tested-by: Yeswanth Rayapati <yeswanth.rayapati@in.bosch.com>
[erosca: massage commit description]
Signed-off-by: Eugeniu Rosca <eugeniu.rosca@bosch.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://msgid.link/r/20240301085003.3057-1-erosca@de.adit-jv.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There was one overlooked place to be replaced with
snd_ctl_find_id_mixer() for code simplification.
No functional change, only code refactoring.
Link: https://lore.kernel.org/r/20240304082158.8583-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDMI codecs which are present and functional from audio perspective lack
i915 support on drm side what results in -ENODEV during the probing
sequence. There is no reason to perform recovery procedure e.g.: reset
the HDAudio controller if this is the case.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240226124432.1203798-4-cezary.rojewski@intel.com
If i915 does not support given platform but the hardware i.e.: HDAudio
codec is still there, the codec-probing procedure will succeed for such
device but the follow up initialization will always end up with -ENODEV.
While bus could filter out address '2' which Intel's HDMI/DP codecs
always enumerate on, more robust approach is to check for i915 presence
before registering display codecs.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240226124432.1203798-3-cezary.rojewski@intel.com
Commit 78f613ba1e ("drm/i915: finish removal of CNL") and its friends
removed support for i915 for all CNL-based platforms. HDAudio library,
however, still treats such platforms as valid candidates for i915
binding. Update query mechanism to reflect changes made in drm tree.
At the same time, i915 support for LKF-based platforms has not been
provided so remove them from valid binding candidates.
Link: https://lore.kernel.org/all/20210728215946.1573015-1-lucas.demarchi@intel.com/
Reviewed-by: Rodrigo Vivi <rodrigo.vivi@intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240226124432.1203798-2-cezary.rojewski@intel.com
When building feature controls from a unit without a name, we try to
derive a name first from the feature unit's input, then fall back to the
output terminal.
If a feature unit connects directly to a "USB Streaming" input terminal
rather than a mixer or other virtual type, the control receives the
somewhat meaningless name "PCM", even if the output had a descriptive
type such as "Headset" or "Speaker".
Here is an example of such AudioControl descriptor from a USB headset
which ends up named "PCM Playback" and is therefore not recognized as
headphones by userspace:
AudioControl Interface Descriptor:
bLength 12
bDescriptorType 36
bDescriptorSubtype 2 (INPUT_TERMINAL)
bTerminalID 4
wTerminalType 0x0101 USB Streaming
bAssocTerminal 5
bNrChannels 2
wChannelConfig 0x0003
Left Front (L)
Right Front (R)
iChannelNames 0
iTerminal 0
AudioControl Interface Descriptor:
bLength 9
bDescriptorType 36
bDescriptorSubtype 3 (OUTPUT_TERMINAL)
bTerminalID 5
wTerminalType 0x0402 Headset
bAssocTerminal 4
bSourceID 6
iTerminal 0
AudioControl Interface Descriptor:
bLength 13
bDescriptorType 36
bDescriptorSubtype 6 (FEATURE_UNIT)
bUnitID 6
bSourceID 4
bControlSize 2
bmaControls(0) 0x0002
Volume Control
bmaControls(1) 0x0000
bmaControls(2) 0x0000
iFeature 0
Other headsets and DACs I tried that used their output terminal for
naming only did so due to their input being an unnamed sidetone mixer.
Instead of always starting with the input terminal, check the type of it
first. If it seems uninteresting, invert the order and use the output
terminal first for naming.
This makes userspace recognize headsets with simple controls as
headphones, and leads to more consistent naming of playback devices
based on their outputs irrespective of sidetone mixers.
Signed-off-by: Kenny Levinsen <kl@kl.wtf>
Link: https://lore.kernel.org/r/20240301231107.42679-1-kl@kl.wtf
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Whilst this laptop contains _DSD inside the BIOS, there is an error in
this configuration. Override the _DSD in the BIOS with the correct
configuration for this laptop.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240301160154.158398-4-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clang prior to 17.0.0 has a bug in its asm goto jump scope analysis to
determine that no variables with the cleanup attribute are skipped by an
indirect jump. Instead of only checking the scope of each label that is
a possible target of each asm goto statement, it checks the scope of
every label, which can cause an error when a variable with the cleanup
attribute is used between two asm goto statements with different scopes,
even if they have completely different label targets:
sound/core/hwdep.c:273:8: error: cannot jump from this asm goto statement to one of its possible targets
if (get_user(device, (int __user *)arg))
^
arch/powerpc/include/asm/uaccess.h:295:5: note: expanded from macro 'get_user'
__get_user(x, _gu_addr) : \
^
arch/powerpc/include/asm/uaccess.h:283:2: note: expanded from macro '__get_user'
__get_user_size_allowed(__gu_val, __gu_addr, __gu_size, __gu_err); \
^
arch/powerpc/include/asm/uaccess.h:199:3: note: expanded from macro '__get_user_size_allowed'
__get_user_size_goto(x, ptr, size, __gus_failed); \
^
arch/powerpc/include/asm/uaccess.h:187:10: note: expanded from macro '__get_user_size_goto'
case 1: __get_user_asm_goto(x, (u8 __user *)ptr, label, "lbz"); break; \
^
arch/powerpc/include/asm/uaccess.h:158:2: note: expanded from macro '__get_user_asm_goto'
asm_volatile_goto( \
^
include/linux/compiler_types.h:366:33: note: expanded from macro 'asm_volatile_goto'
#define asm_volatile_goto(x...) asm goto(x)
^
sound/core/hwdep.c:291:9: note: possible target of asm goto statement
if (put_user(device, (int __user *)arg))
^
arch/powerpc/include/asm/uaccess.h:66:5: note: expanded from macro 'put_user'
__put_user(x, _pu_addr) : -EFAULT; \
^
arch/powerpc/include/asm/uaccess.h:52:9: note: expanded from macro '__put_user'
\
^
sound/core/hwdep.c:276:4: note: jump bypasses initialization of variable with __attribute__((cleanup))
scoped_guard(mutex, ®ister_mutex) {
^
include/linux/cleanup.h:169:20: note: expanded from macro 'scoped_guard'
for (CLASS(_name, scope)(args), \
To avoid this issue, move the put_user() call out of the scoped_guard()
scope, which allows the asm goto scope analysis to see that the variable
with the cleanup attribute will never be skipped by the asm goto
statements.
There should be no functional change because prior to the refactoring,
put_user() was not called under register_mutex, so this call does not
even need to be in the scoped_guard() in the first place.
Fixes: e6684d08cc ("ALSA: hwdep: Use guard() for locking")
Closes: https://github.com/ClangBuiltLinux/linux/issues/2003
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://lore.kernel.org/r/20240301-fix-snd-hwdep-guard-v1-1-6aab033f3f83@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In azx_probe_codecs function, when bus->codec_mask is becomes to 0(no codecs),
execute azx_init_chip, bus->codec_mask will be initialized to a value again,
this causes snd_hda_codec_new function to run, the process is as follows:
-->snd_hda_codec_new
-->snd_hda_codec_device_init
-->snd_hdac_device_init---snd_hdac_read_parm(...AC_PAR_VENDOR_ID) 2s
---snd_hdac_read_parm(...AC_PAR_VENDOR_ID) 2s
---snd_hdac_read_parm(...AC_PAR_SUBSYSTEM_ID) 2s
---snd_hdac_read_parm(...AC_PAR_REV_ID) 2s
---snd_hdac_read_parm(...AC_PAR_NODE_COUNT) 2s
when no codecs, read communication is error, each command will be polled for
2 second, a total of 10s, it is easy to some problem.
like this:
2 [ 14.833404][ 6] [ T164] hda 0006:00: Codec #0 probe error; disabling it...
3 [ 14.844178][ 6] [ T164] hda 0006:00: codec_mask = 0x1
4 [ 14.880532][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0000
5 [ 15.891988][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0000
6 [ 16.978090][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0001
7 [ 18.140895][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0002
8 [ 19.135516][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0004
10 [ 19.900086][ 6] [ T164] hda 0006:00: no codecs initialized
11 [ 45.573398][ 2] [ C2] watchdog: BUG: soft lockup - CPU#2 stuck for 22s! [kworker/2:0:25]
Here, when bus->codec_mask is 0, use a direct break to avoid execute snd_hda_codec_new function.
Signed-off-by: songxiebing <songxiebing@kylinos.cn>
Link: https://lore.kernel.org/r/20240301011841.7247-1-soxiebing@163.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
