ALSA: aaci: Delete unused variable in aaci_do_suspend

The variable aaci is not used anymore and can be deleted.

Fixes: 792a6c51875c ("[ALSA] Fix PM support")
Signed-off-by: Thomas Weißschuh <thomas.weissschuh@linutronix.de>
Link: https://lore.kernel.org/r/20240312-aaci-unused-v1-1-09be643f67c2@linutronix.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge tag 'asoc-v6.9' of https://git./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

ALSA: scarlett2: Fix Scarlett 4th Gen input gain range again

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: a45cf0a08347 ("ALSA: scarlett2: Fix Scarlett 4th Gen input gain range")
Message-ID: <Ze7OMA8ntG7KteGa@m.b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: scarlett2: Fix Scarlett 4th Gen input gain range

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: 0a995e38dc44 ("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>

ALSA: scarlett2: Fix Scarlett 4th Gen autogain status values

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: 0a995e38dc44 ("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>

ALSA: scarlett2: Fix Scarlett 4th Gen 4i4 low-voltage detection

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: d7cfa2fdfc8a ("ALSA: scarlett2: Add power status control")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <d97b7d87f43b0e54f37e1552394be2f3ae182704.1710047969.git.g@b4.vu>

ALSA: hda/tas2781: restore power state after system_resume

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: 5be27f1e3ec9 ("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>

ALSA: hda/tas2781: do not call pm_runtime_force_* in system_resume/suspend

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: 5be27f1e3ec9 ("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>

ALSA: hda/tas2781: do not reset cur_* values in runtime_suspend

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: 5be27f1e3ec9 ("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>

ALSA: hda/tas2781: add lock to system_suspend

Add the missing lock around tasdevice_tuning_switch().

Fixes: 5be27f1e3ec9 ("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: <c666da13d4bc48cd1ab1357479e0c6096541372c.1709918447.git.soyer@irl.hu>

ALSA: hda/tas2781: use dev_dbg in system_resume

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: 5be27f1e3ec9 ("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>

ALSA: hda/realtek: fix ALC285 issues on HP Envy x360 laptops

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>

Merge branch 'for-next' into for-linus

Prep for 6.9 merge.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

platform/x86: serial-multi-instantiate: Add support for CS35L54 and CS35L57

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>

ALSA: hda: cs35l56: Add support for CS35L54 and CS35L57

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>

ASoC: cs35l56: Add support for CS35L54 and CS35L57

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>

Merge tag 'asoc-fix-v6.8-rc7' of https://git./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.

ASoC: Intel: catpt: Carefully use PCI bitwise constants

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>

ALSA: hda: hda_component: Include sound/hda_codec.h

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>

ALSA: hda: hda_component: Add missing #include guards

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>

ASoC: codecs: ES8326: change members of private

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.

firmware: cirrus: cs_dsp: Remove non-existent member from kerneldoc

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>

firmware: cirrus: cs_dsp: Initialize debugfs_root to invalid

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>

ASoC: codecs: ES8326: change support for ES8326

Removed mic1-src and mic2-src. and changed default value
of interrupt-clk

Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240307051222.24010-3-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: codecs: ES8326: Changing members of private structure

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>

ASoC: wm8962: Fix up incorrect error message in wm8962_set_fll

Use source instead of ret, which seems to be unrelated and will always
be zero.

Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Link: https://msgid.link/r/20240306161439.1385643-5-stuarth@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: wm8962: Enable both SPKOUTR_ENA and SPKOUTL_ENA in mono mode

Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Link: https://msgid.link/r/20240306161439.1385643-2-stuarth@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: wm8962: Enable oscillator if selecting WM8962_FLL_OSC

Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Link: https://msgid.link/r/20240306161439.1385643-1-stuarth@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: trace: add event to snd_soc_dapm trace events

Add the event value to the snd_soc_dapm_start and snd_soc_dapm_done trace
events to make them more informative.

Trace before:

           aplay-229   [000]   250.140309: snd_soc_dapm_start:   card=vscn-2046
           aplay-229   [000]   250.167531: snd_soc_dapm_done:    card=vscn-2046
           aplay-229   [000]   251.169588: snd_soc_dapm_start:   card=vscn-2046
           aplay-229   [000]   251.195245: snd_soc_dapm_done:    card=vscn-2046

Trace after:

           aplay-214   [000]   693.290612: snd_soc_dapm_start:   card=vscn-2046 event=1
           aplay-214   [000]   693.315508: snd_soc_dapm_done:    card=vscn-2046 event=1
           aplay-214   [000]   694.537349: snd_soc_dapm_start:   card=vscn-2046 event=2
           aplay-214   [000]   694.563241: snd_soc_dapm_done:    card=vscn-2046 event=2

Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://msgid.link/r/20240306-improve-asoc-trace-events-v1-2-edb252bbeb10@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: trace: add component to set_bias_level trace events

The snd_soc_bias_level_start and snd_soc_bias_level_done trace events
currently look like:

           aplay-229   [000]  1250.140778: snd_soc_bias_level_start: card=vscn-2046 val=1
           aplay-229   [000]  1250.140784: snd_soc_bias_level_done: card=vscn-2046 val=1
           aplay-229   [000]  1250.140786: snd_soc_bias_level_start: card=vscn-2046 val=2
           aplay-229   [000]  1250.140788: snd_soc_bias_level_done: card=vscn-2046 val=2
    kworker/u8:1-21    [000]  1250.140871: snd_soc_bias_level_start: card=vscn-2046 val=1
    kworker/u8:0-11    [000]  1250.140951: snd_soc_bias_level_start: card=vscn-2046 val=1
    kworker/u8:0-11    [000]  1250.140956: snd_soc_bias_level_done: card=vscn-2046 val=1
    kworker/u8:0-11    [000]  1250.140959: snd_soc_bias_level_start: card=vscn-2046 val=2
    kworker/u8:0-11    [000]  1250.140961: snd_soc_bias_level_done: card=vscn-2046 val=2
    kworker/u8:1-21    [000]  1250.167219: snd_soc_bias_level_done: card=vscn-2046 val=1
    kworker/u8:1-21    [000]  1250.167222: snd_soc_bias_level_start: card=vscn-2046 val=2
    kworker/u8:1-21    [000]  1250.167232: snd_soc_bias_level_done: card=vscn-2046 val=2
    kworker/u8:0-11    [000]  1250.167440: snd_soc_bias_level_start: card=vscn-2046 val=3
    kworker/u8:0-11    [000]  1250.167444: snd_soc_bias_level_done: card=vscn-2046 val=3
    kworker/u8:1-21    [000]  1250.167497: snd_soc_bias_level_start: card=vscn-2046 val=3
    kworker/u8:1-21    [000]  1250.167506: snd_soc_bias_level_done: card=vscn-2046 val=3

There are clearly multiple calls, one per component, but they cannot be
discriminated from each other.

Change the ftrace events to also print the component name, to make it clear
which part of the code is involved. This requires changing the passed value
from a struct snd_soc_card, where the DAPM context is not kwown, to a
struct snd_soc_dapm_context where it is obviously known but the a card
pointer is also available.

With this change, the resulting trace becomes:

           aplay-247   [000]  1436.357332: snd_soc_bias_level_start: card=vscn-2046 component=(none) val=1
           aplay-247   [000]  1436.357338: snd_soc_bias_level_done: card=vscn-2046 component=(none) val=1
           aplay-247   [000]  1436.357340: snd_soc_bias_level_start: card=vscn-2046 component=(none) val=2
           aplay-247   [000]  1436.357343: snd_soc_bias_level_done: card=vscn-2046 component=(none) val=2
    kworker/u8:4-215   [000]  1436.357437: snd_soc_bias_level_start: card=vscn-2046 component=ff560000.codec val=1
    kworker/u8:5-231   [000]  1436.357518: snd_soc_bias_level_start: card=vscn-2046 component=ff320000.i2s val=1
    kworker/u8:5-231   [000]  1436.357523: snd_soc_bias_level_done: card=vscn-2046 component=ff320000.i2s val=1
    kworker/u8:5-231   [000]  1436.357526: snd_soc_bias_level_start: card=vscn-2046 component=ff320000.i2s val=2
    kworker/u8:5-231   [000]  1436.357528: snd_soc_bias_level_done: card=vscn-2046 component=ff320000.i2s val=2
    kworker/u8:4-215   [000]  1436.383217: snd_soc_bias_level_done: card=vscn-2046 component=ff560000.codec val=1
    kworker/u8:4-215   [000]  1436.383221: snd_soc_bias_level_start: card=vscn-2046 component=ff560000.codec val=2
    kworker/u8:4-215   [000]  1436.383231: snd_soc_bias_level_done: card=vscn-2046 component=ff560000.codec val=2
    kworker/u8:5-231   [000]  1436.383468: snd_soc_bias_level_start: card=vscn-2046 component=ff320000.i2s val=3
    kworker/u8:5-231   [000]  1436.383472: snd_soc_bias_level_done: card=vscn-2046 component=ff320000.i2s val=3
    kworker/u8:4-215   [000]  1436.383503: snd_soc_bias_level_start: card=vscn-2046 component=ff560000.codec val=3
    kworker/u8:4-215   [000]  1436.383513: snd_soc_bias_level_done: card=vscn-2046 component=ff560000.codec val=3

Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://msgid.link/r/20240306-improve-asoc-trace-events-v1-1-edb252bbeb10@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: dt-bindings: rt1015: Convert to dtschema

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>

ASoC: Intel: atom: sst_ipc: Remove unused intel-mid.h

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>

ASoC: dt-bindings: nvidia: Fix 'lge' vendor prefix

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>

ALSA: hda/realtek: fix mute/micmute LEDs for HP EliteBook

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: soc-core.c: Prefer to return dai->driver->name in snd_soc_dai_name_get()

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>

ASoC: cs-amp-lib: Add KUnit test for calibration helpers

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>

ASoC: amd: yc: Add HP Pavilion Aero Laptop 13-be2xxx(8BD6) into DMI quirk table

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>

ASoC: rcar: adg: correct TIMSEL setting for SSI9

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: 629509c5bc478c ("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>

ALSA: ac97: More cleanup with snd_ctl_find_id_mixer()

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>

ALSA: hda: Reuse for_each_pcm_streams()

Use the macro to improve readability.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240226124432.1203798-6-cezary.rojewski@intel.com

ASoC: codecs: hda: Cleanup error messages

Be cohesive and use same pattern in each error message.

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/20240226124432.1203798-5-cezary.rojewski@intel.com

ASoC: Intel: avs: Ignore codecs with no suppoting driver

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

ASoC: codecs: hda: Skip HDMI/DP registration if i915 is missing

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

ALSA: hda: Skip i915 initialization on CNL/LKF-based platforms

Commit 78f613ba1efb ("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

ALSA: usb-audio: Name feature ctl using output if input is PCM

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>

ALSA: hda: cs35l41: Overwrite CS35L41 configuration for ASUS UM5302LA

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>

ALSA: hda/realtek: Add quirks for Lenovo Thinkbook 16P laptops

These models use 2 CS35L41 amps with HDA using I2C.
Both models have _DSD support inside cs35l41_hda_property.c.

Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218437
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240301160154.158398-3-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda: cs35l41: Support Lenovo Thinkbook 16P

Adds sound support for 2 Lenovo Thinkbook 16P laptops using CS35L41
HDA with External Boost.

SSIDs:
- 17AA38A9
- 17AA38AB

Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218437
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240301160154.158398-2-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek - Add Headset Mic supported Acer NB platform

It will be enable headset Mic for Acer NB platform.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/fe0eb6661ca240f3b7762b5b3257710d@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hwdep: Move put_user() call out of scoped_guard() in snd_hwdep_control_ioctl()

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, &register_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: e6684d08cc19 ("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>

ALSA: hda: optimize the probe codec process

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>

ALSA: hda/realtek - Fix headset Mic no show at resume back for Lenovo ALC897 platform

Headset Mic will no show at resume back.
This patch will fix this issue.

Fixes: d7f32791a9fc ("ALSA: hda/realtek - Add headset Mic support for Lenovo ALC897 platform")
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/4713d48a372e47f98bba0c6120fd8254@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: Intel: bytcr_rt5640: Add an extra entry for the Chuwi Vi8 tablet

The bios version can differ depending if it is a dual-boot variant of the tablet.
Therefore another DMI match is required.

Signed-off-by: Alban Boyé <alban.boye@protonmail.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240228192807.15130-1-alban.boye@protonmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: madera: Fix typo in madera_set_fll_clks shift value

Fix a typo in the shift value used in madera_set_fll_clks.

Fixes: 3863857dd5ca3 ("ASoC: madera: Enable clocks for input pins when used for the FLL")
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Link: https://msgid.link/r/20240229114637.352098-1-stuarth@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>

Merge tag 'asoc-fix-v6.8-rc5' of https://git./linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v6.8

A few small fixes, some driver specific and one slightly larger one
from Richard which adds a new core helper and updates a small clutch of
drivers to deal with the fact that they were using a helper which
requires that the lock for the list of controls without holding that
lock.  We also have some quirks for new AMD based Lenovo systems.

ALSA: control_led: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

A couple of functions that use snd_card_ref() and *_unref() are also
cleaned up with a defined class, too.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-25-tiwai@suse.de

ALSA: pcm: oss: Use guard() for setup

The setup_mutex in PCM oss code can be simplified with guard().
(params_lock is tough and not trivial to covert, though.)

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-24-tiwai@suse.de

ALSA: pcm: Use guard() for PCM stream locks

Define guard() usage for PCM stream locking and use it in appropriate
places.

The pair of snd_pcm_stream_lock() and snd_pcm_stream_unlock() can be
presented with guard(pcm_stream_lock) now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-23-tiwai@suse.de

ALSA: pcm: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-22-tiwai@suse.de

ALSA: seq: prioq: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-21-tiwai@suse.de

ALSA: seq: virmidi: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-20-tiwai@suse.de

ALSA: seq: ump: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-19-tiwai@suse.de

ALSA: seq: midi: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-18-tiwai@suse.de

ALSA: seq: timer: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-17-tiwai@suse.de

ALSA: seq: queue: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-16-tiwai@suse.de

ALSA: seq: ports: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-15-tiwai@suse.de

ALSA: seq: memory: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-14-tiwai@suse.de

ALSA: seq: fifo: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-13-tiwai@suse.de

ALSA: core: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-12-tiwai@suse.de

ALSA: jack: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-11-tiwai@suse.de

ALSA: rawmidi: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

There are a few remaining explicit mutex and spinlock calls, and those
are the places where the temporary unlock/relocking happens -- which
guard() doens't cover well yet.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-10-tiwai@suse.de

ALSA: control: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

The lops calls under multiple rwsems are factored out as a simple
macro, so that it can be called easily from snd_ctl_dev_register()
and snd_ctl_dev_disconnect().

There are a few remaining explicit rwsem and spinlock calls, and those
are the places where the lock downgrade happens or where the temporary
unlock/relocking happens -- which guard() doens't cover well yet.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-9-tiwai@suse.de

ALSA: mixer_oss: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-8-tiwai@suse.de

ALSA: info: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-7-tiwai@suse.de

ALSA: hwdep: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

There are still a few remaining explicit mutex_lock/unlock calls, and
those are for the places where we do temporary unlock/relock, which
doesn't fit well with the guard(), so far.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-6-tiwai@suse.de

ALSA: hrtimer: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-5-tiwai@suse.de

ALSA: timer: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

For making changes easier, some functions widen the application of
register_mutex, but those shouldn't influence on any actual
performance.

Also, one code block was factored out as a function so that guard()
can be applied cleanly without much indentation.

There are still a few remaining explicit spin_lock/unlock calls, and
those are for the places where we do temporary unlock/relock, which
doesn't fit well with the guard(), so far.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-4-tiwai@suse.de

ALSA: compress_offload: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

The explicit mutex_lock/unlock are still seen only in
snd_compress_wait_for_drain() which does temporary unlock/relocking.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-3-tiwai@suse.de

ALSA: ump: Use guard() for locking

We can simplify the code gracefully with new guard() macro and co for
automatic cleanup of locks.

Only the code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240227085306.9764-2-tiwai@suse.de

ALSA: kunit: Fix sparse warnings

There were a few sparse warnings about the cast of strong-typed
snd_pcm_format_t.  Fix them with cast with __force.

For spreading the ugly mess, put them in the definitions
WRONG_FORMAT_1 and WRONG_FORMAT_2 and use them in the callers.

Fixes: 3e39acf56ede ("ALSA: core: Add sound core KUnit test")
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/r/202402270303.PmvmQrJV-lkp@intel.com
Link: https://lore.kernel.org/r/20240227104912.18921-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: amd: yc: Fix non-functional mic on Lenovo 21J2

Like many other models, the Lenovo 21J2 (ThinkBook 16 G5+ APO)
needs a quirk entry for the internal microphone to function.

Signed-off-by: Jiawei Wang <me@jwang.link>
Link: https://msgid.link/r/20240228073914.232204-2-me@jwang.link
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: amd: yc: add new YC platform variant (0x63) support

The Lenovo 21J2 (ThinkBook 16 G5+ APO) has this new variant,
as detected with lspci:

64:00.5 Multimedia controller: Advanced Micro Devices, Inc. [AMD]
        ACP/ACP3X/ACP6x Audio Coprocessor (rev 63)

Signed-off-by: Jiawei Wang <me@jwang.link>
Link: https://msgid.link/r/20240228073914.232204-1-me@jwang.link
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: meson: axg-fifo: use FIELD helpers

Use FIELD_GET() and FIELD_PREP() helpers instead of doing it manually.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240227150826.573581-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: cs35l56: Prevent bad sign extension in cs35l56_read_silicon_uid()

Cast u8 values to u32 when using them to build a 32-bit unsigned value
that is then stored in a u64. This avoids the possibility of a bad sign
extension where the u8 is implicitly extended to an int, thus changing it
from an unsigned to a signed value.

Whether this is a real problem is debatable, but it does no harm to
ensure that the u8 are cast to a suitable type for shifting.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e1830f66f6c6 ("ASoC: cs35l56: Add helper functions for amp calibration")
Link: https://msgid.link/r/20240227100042.99-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: hda/realtek - ALC285 reduce pop noise from Headphone port

It had pop noise from Headphone port when system reboot state.
If NID 58h Index 0x0 to fill default value, it will reduce pop noise.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/7493e207919a4fb3a0599324fd010e3e@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: meson: axg fixes and clean-up

Merge series from Jerome Brunet <jbrunet@baylibre.com>:

This are various fixes and clean up gathered while working on Amlogic audio
support. These help better handle higher and unusual clock configuration
for TDM, SPDIF or PDM.

ASoC: codecs: Simplify mclk initialization

Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:

The patchset may not cover all codecs found in the codecs/ directory -
noticed a possible improvement and grepped for similar pattern across C
files found in the directory. Those addressed here seem pretty
straightforward.

Most of clk_xxx() functions do check if provided clk-pointer is
non-NULL. These do not check if the pointer is an error-pointer.
Providing such to a clk_xxx() results in a panic.

By utilizing _optional() variant of devm_clk_get() the driver code is
both simplified and more robust. There is no need to remember about
IS_ERR(clk) checks each time mclk is accessed.

ASoC: amd: yc: Add Lenovo ThinkBook 21J0 into DMI quirk table

This patch adds Lenovo 21J0 (ThinkBook 16 G5+ ARP) to the DMI quirks table
to enable internal microphone array.

Cc: linux-sound@vger.kernel.org
Signed-off-by: Johnny Hsieh <mnixry@outlook.com>
Link: https://msgid.link/r/TYSPR04MB8429D62DFDB6727866ECF1DEC55A2@TYSPR04MB8429.apcprd04.prod.outlook.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: meson: axg-fifo: take continuous rates

The rate of the stream does not matter for the fifos of the axg family.
Fifos will just push or pull data to/from the DDR according to consumption
or production of the downstream element, which is the DPCM backend.

Drop the rate list and allow continuous rates. The lower and upper rate are
set according what is known to work with the different backends

This allows the PDM input backend to also use continuous rates.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-6-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: meson: axg-spdifin: use max width for rate detection

Use maximum width between 2 edges to setup spdifin thresholds
and detect the input sample rate. This comes from Amlogic SDK and
seems to be marginally more reliable than minimum width.

This is done to align with a future eARC support.
No issue was reported with minimum width so far, this is considered
to be an update so no Fixes tag is set.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-5-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: meson: axg-tdm-interface: update error format error traces

ASoC stopped using CBS_CFS and CBM_CFM a few years ago but the traces in
the amlogic tdm interface driver did not follow.

Update this to match the new format names

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-4-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: meson: axg-tdm-interface: add frame rate constraint

According to Amlogic datasheets for the SoCs supported by this driver, the
maximum bit clock rate is 100MHz.

The tdm interface allows the rates listed by the DAI driver, regardless of
the number slots or their width. However, these will impact the bit clock
rate.

Hitting the 100MHz limit is very unlikely for most use cases but it is
possible.

For example with 32 slots / 32 bits wide, the maximum rate is no longer
384kHz but ~96kHz.

Add the constraint accordingly if the component is not already active.
If it is active, the rate is already constrained by the first stream rate.

Fixes: d60e4f1e4be5 ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: meson: axg-tdm-interface: fix mclk setup without mclk-fs

By default, when mclk-fs is not provided, the tdm-interface driver
requests an MCLK that is 4x the bit clock, SCLK.

However there is no justification for this:

* If the codec needs MCLK for its operation, mclk-fs is expected to be set
  according to the codec requirements.
* If the codec does not need MCLK the minimum is 2 * SCLK, because this is
  minimum the divider between SCLK and MCLK can do.

Multiplying by 4 may cause problems because the PLL limit may be reached
sooner than it should, so use 2x instead.

Fixes: d60e4f1e4be5 ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: codecs: tx-macro: correct TX SMIC MUXn widgets on SM8350+

Starting with Qualcomm SM8350 SoC, so Low Power Audio SubSystem (LPASS)
block version v9.2, the register responsible for TX SMIC MUXn muxes is
different.  In earlier LPASS versions this mux had bit fields for
analogue (ADCn) and digital (SWR_DMICn) MICs.  Choice of ADCn was
selecting the analogue path in CDC_TX_TOP_CSR_SWR_DMICn_CTL register.

With LPASS v9.2 and newer, the bit fields are integrated into just
SWR_MICn and there is no distinction for analogue or digital MIC in the
register.

Fix support for LPASS v9.2+:
1. Add new set of widgets and audio routes for LPASS v9.2.
2. Do not choose analogue or digital in CDC_TX_TOP_CSR_SWR_DMICn_CTL
   based on value of the mux.
3. Replace all the input widgets (TX SWR_ADCn, TX SWR_DMICn) with TX
   SWR_INPUTn ones.

The change is not backwards compatible with older DTBs and existing
mixer settings, therefore it does not change handling of older platforms
with working micrphones (SC8280xp) but only the ones with issues
(SM8450, SM8550) which need the fix.

Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240226115925.53953-3-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: codecs: tx-macro: split widgets per different LPASS versions

TX macro codec differs slightly between different Qualcomm Low Power
Audio SubSystem (LPASS) block versions.  In LPASS version 9.2 the
register responsible for TX SMIC MUXn muxes is different, thus to
properly support it, the driver needs to register different widgets per
different LPASS version.

Prepare for supporting this register difference by refactoring existing
code:
1. Move few widgets (TX SMIC MUXn, TX SWR_ADCn, TX SWR_DMICn) out of
   common 'tx_macro_dapm_widgets[]' array to a new per-variant specific
   array 'tx_macro_dapm_widgets_v9[]'.
2. Move also related audio routes into new array.
3. Store pointers to these variant-specific arrays in new variant-data
   structure 'tx_macro_data'.
4. Add variant-specific widgets and routes in component probe, instead
   of driver probe.

The change should have no real impact, except re-shuffling code and
registering some widgets and audio routes in component probe, instead of
driver probe.

Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240226115925.53953-2-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: hda/realtek: Add special fixup for Lenovo 14IRP8

Lenovo Slim/Yoga Pro 9 14IRP8 requires a special fixup because there is
a collision of its PCI SSID (17aa:3802) with Lenovo Yoga DuetITL 2021
codec SSID.

Fixes: 3babae915f4c ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=208555
Link: https://lore.kernel.org/all/d5b42e483566a3815d229270abd668131a0d9f3a.camel@irl.hu
Cc: stable@vger.kernel.org
Signed-off-by: Willian Wang <git@willian.wang>
Reviewed-by: Gergo Koteles <soyer@irl.hu>
Link: https://lore.kernel.org/r/170879111795.8.6687687359006700715.273812184@willian.wang
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: cs35l56: Apply calibration from EFI

Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:

Factory calibration of the speakers stores the calibration information
into an EFI variable.

This set of patches adds support for applying speaker calibration
data from that EFI variable.

The HDA patch (#5) depends on the ASoC patches #2 and #3

ASoC: dt-bindings: microchip: add sam9x7

Add sam9x7 compatible in the DT documentation.

Signed-off-by: Varshini Rajendran <varshini.rajendran@microchip.com>
Acked-by: Conor Dooley <conor.dooley@microchip.com>
Link: https://lore.kernel.org/r/20240223172657.672439-1-varshini.rajendran@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: dt-bindings: atmel-classd: add sam9x7 compatible

Add sam9x7 compatible to DT bindings documentation.

Signed-off-by: Varshini Rajendran <varshini.rajendran@microchip.com>
Link: https://lore.kernel.org/r/20240223172608.672190-1-varshini.rajendran@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: hda: cs35l56: Apply amp calibration from EFI data

If there are factory calibration settings in EFI, extract the
settings and write them to the firmware calibration controls.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240223153910.2063698-6-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>