We only unregister the platform device during the .remove operation,
but if the probe fails we will never reach this sequence.
Suggested-by: NBard Liao <yung-chuan.liao@linux.intel.com>
Fixes: dd96daca ("ASoC: SOF: Intel: Add APL/CNL HW DSP support")
Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: NRanjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: NGuennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20210302003410.1178535-1-pierre-louis.bossart@linux.intel.comSigned-off-by: NMark Brown <broonie@kernel.org>

Most steps in this table are steps of 3dB (300 centi-dB), so we can
simplify the table.

This not only reduces the amount of space it takes inside the kernel,
this also makes alsa-lib's mixer code actually accept the table, where
as before this change alsa-lib saw the "ADC PGA Gain" control as a
control without a dB scale.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210228160441.241110-1-hdegoede@redhat.comSigned-off-by: NMark Brown <broonie@kernel.org>

According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has
the following bit field definitions:

| BITS  | FIELD       | RW | RESET | DEFINITION                        |
| 15    | RSVD        | RO | 0x0   | Reserved                          |
| 14    | RSVD        | RW | 0x1   | Reserved                          |
| 13:12 | MAX_GAIN    | RW | 0x1   | Max Gain of AVC in expander mode  |
| 11:10 | RSVD        | RO | 0x0   | Reserved                          |
| 9:8   | LBI_RESP    | RW | 0x1   | Integrator Response               |
| 7:6   | RSVD        | RO | 0x0   | Reserved                          |
| 5     | HARD_LMT_EN | RW | 0x0   | Enable hard limiter mode          |
| 4:1   | RSVD        | RO | 0x0   | Reserved                          |
| 0     | EN          | RW | 0x0   | Enable/Disable AVC                |

The original default value written to the DAP_AVC_CTRL register during
sgtl5000_i2c_probe() was 0x0510.  This would incorrectly write values to
bits 4 and 10, which are defined as RESERVED.  It would also not set
bits 12 and 14 to their correct RESET values of 0x1, and instead set
them to 0x0.  While the DAP_AVC module is effectively disabled because
the EN bit is 0, this default value is still writing invalid values to
registers that are marked as read-only and RESERVED as well as not
setting bits 12 and 14 to their correct default values as defined by the
datasheet.

The correct value that should be written to the DAP_AVC_CTRL register is
0x5100, which configures the register bits to the default values defined
by the datasheet, and prevents any writes to bits defined as
'read-only'.  Generally speaking, it is best practice to NOT attempt to
write values to registers/bits defined as RESERVED, as it generally
produces unwanted/undefined behavior, or errors.

Also, all credit for this patch should go to my colleague Dan MacDonald
<dmacdonald@curbellmedical.com> for finding this error in the first
place.

[1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdfSigned-off-by: NBenjamin Rood <benjaminjrood@gmail.com>
Reviewed-by: NFabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-devSigned-off-by: NMark Brown <broonie@kernel.org>

The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.

E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.

Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.

Note this mirrors commit 3f31f7d9 ("ASoC: rt5670: Fix dac- and adc-
vol-tlv values being off by a factor of 10") which made the exact same
change to the rt5670 codec driver.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.comSigned-off-by: NMark Brown <broonie@kernel.org>

The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.

E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.

Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.

Note this mirrors commit 3f31f7d9 ("ASoC: rt5670: Fix dac- and adc-
vol-tlv values being off by a factor of 10") which made the exact same
change to the rt5670 codec driver.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.comSigned-off-by: NMark Brown <broonie@kernel.org>

When I added the quirk for the "HP Pavilion x2 10-p0XX" I copied the
byt_rt5640_quirk_table[] entry for the HP Pavilion x2 10-k0XX / 10-n0XX
models since these use almost the same settings.

While doing this I accidentally also copied and kept the non-standard
OVCD_TH_1500UA setting used on those models. This too low threshold is
causing headsets to often be seen as headphones (without a headset-mic)
and when correctly identified it is causing ghost play/pause
button-presses to get detected.

Correct the HP Pavilion x2 10-p0XX quirk to use the default OVCD_TH_2000UA
setting, fixing these problems.

Fixes: fbdae7d6 ("ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 Detachable quirks")
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Acked-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210224105052.42116-1-hdegoede@redhat.comSigned-off-by: NMark Brown <broonie@kernel.org>

Add missed MODULE_DEVICE_TABLE for the driver can be loaded
automatically at boot.

Fixes: 92088477 ("ASoC: ak5558: Add support for AK5558 ADC driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: NShengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1614149872-25510-2-git-send-email-shengjiu.wang@nxp.comSigned-off-by: NMark Brown <broonie@kernel.org>

Add missed MODULE_DEVICE_TABLE for the driver can be loaded
automatically at boot.

Fixes: 08660086 ("ASoC: ak4458: Add support for AK4458 DAC driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: NShengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1614149872-25510-1-git-send-email-shengjiu.wang@nxp.comSigned-off-by: NMark Brown <broonie@kernel.org>

For reliable output-mute LED control we need a "DAC1 Playback Switch"
control. The "DAC Playback volume" control is the only control in the
path from the DAC1 data input to the speaker output, so the UCM profile
for the speaker output will have its PlaybackMixerElem set to "DAC1".

But userspace (pulseaudio) will set the "DAC1 Playback Volume" control to
its softest setting (which is not fully muted) while still showing the
speaker as being enabled at a low volume in the UI.

If we were to set the SNDRV_CTL_ELEM_ACCESS_SPK_LED on the "DAC1 Playback
Volume" control, this would mean then what pressing KEY_VOLUMEDOWN the
speaker-mute LED (embedded in the volume-mute toggle key) would light
while the UI is still showing the speaker as being enabled at a low
volume, meaning that the UI and the LED are out of sync.

Only after an _extra_ KEY_VOLUMEDOWN press would the UI show the
speaker as being muted.

The path from DAC1 data input to the speaker output does have
a digital mixer with DAC1's data as one of its inputs direclty after
the "DAC1 Playback Volume" control.

This commit adds an emulated "DAC1 Playback Switch" control by:

1. Declaring the enable flag for that mixers DAC1 input as well as the
"DAC1 Playback Switch" control both as SND_SOC_NOPM controls.

2. Storing the settings of both controls as driver-private data

3. Only clearing the mute flag for the DAC1 input of that mixer if the
stored values indicate both controls are enabled.

This is a preparation patch for adding "audio-mute" LED trigger support.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-5-hdegoede@redhat.comSigned-off-by: NMark Brown <broonie@kernel.org>

The SND_SOC_DAPM_MIXER declaration for "Sto1 ADC MIXL" and "Sto1 ADC MIXR"
was using the mute bits from the RT5670_STO1_ADC_DIG_VOL control as mixer
master mute bits.

But these bits are already exposed to userspace as controls as part of the
"ADC Capture Volume" / "ADC Capture Switch" control pair:

        SOC_DOUBLE("ADC Capture Switch", RT5670_STO1_ADC_DIG_VOL,
                RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1),
        SOC_DOUBLE_TLV("ADC Capture Volume", RT5670_STO1_ADC_DIG_VOL,
                        RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
                        127, 0, adc_vol_tlv),

Both the fact that the mute bits belong to the same reg as the vol-ctrl
and the "Digital Mixer Path" diagram in the datasheet clearly shows that
these mute bits are not part of the mixer and having 2 separate controls
poking at the same bits is a bad idea.

Remove the master-mute bits settings from the  "Sto1 ADC MIXL" and
"Sto1 ADC MIXR" DAPM widget declarations, avoiding these bits getting
poked from 2 different places.

This should not cause any issues for userspace. AFAICT the rt567x codecs
are only used on x86/ACPI devices and the UCM profiles used there already
set the "ADC Capture Switch" as needed.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-4-hdegoede@redhat.comSigned-off-by: NMark Brown <broonie@kernel.org>

The RT5670_L_MUTE_SFT and RT5670_R_MUTE_SFT bits (bits 15 and 7) of the
RT5670_HP_VOL register are set / unset by the headphones deplop code
run by rt5670_hp_event() on SND_SOC_DAPM_POST_PMU / SND_SOC_DAPM_PRE_PMD.

So we should not also export a control to userspace which toggles these
same bits.

This should not cause any issues for userspace. AFAICT the rt567x codecs
are only used on x86/ACPI devices and the UCM profiles used there do not
use the "HP Playback Switch" control.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-3-hdegoede@redhat.comSigned-off-by: NMark Brown <broonie@kernel.org>

The "OUT Channel Switch" control is a left over from code copied from
thr rt5640 codec driver.

With the rt5640 codec driver the output volume controls have 2 pairs of
mute bits:
bit 7, 15: Mute Control for Spk/Headphone/Line Output Port
bit 6, 14: Mute Control for Spk/Headphone/Line Volume Channel

Bits 7 and 15 are normal mute bits on the rt5670/5672 which are
controlled by 2 dapm widgets:
	SND_SOC_DAPM_SWITCH("LOUT L Playback", SND_SOC_NOPM, 0, 0,
			    &lout_l_enable_control),
	SND_SOC_DAPM_SWITCH("LOUT R Playback", SND_SOC_NOPM, 0, 0,
			    &lout_r_enable_control),

But on the 5670/5672 bit 6 is always reserved, where as bit 14 is
"LOUT Differential Mode" on the 5670 and also reserved on the 5672.

So the "OUT Channel Switch" control which is controlling bits 6+14
of the "LINE Output Control" register is bogus -> remove it.

This should not cause any issues for userspace. AFAICT the rt567x codecs
are only used on x86/ACPI devices and the UCM profiles used there do not
use the "OUT Channel Switch" control.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-2-hdegoede@redhat.comSigned-off-by: NMark Brown <broonie@kernel.org>

When using the driver in I2S TDM mode, the _fsl_ssi_set_dai_fmt()
function rewrites the number of slots previously set by the
fsl_ssi_set_dai_tdm_slot() function to 2 by default.
To fix this, let's use the saved slot count value or, if TDM
is not used and the slot count is not set, proceed as before.

Fixes: 4f14f5c1 ("ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode")
Signed-off-by: NAlexander Shiyan <shc_work@mail.ru>
Acked-by: NNicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20210216114221.26635-1-shc_work@mail.ruSigned-off-by: NMark Brown <broonie@kernel.org>

There is potential read of the uninitialized variable ec_tx if the call
to snd_soc_component_read fails or returns an unrecognized w->name. To
avoid this corner case, initialize ec_tx to -1 so that it is caught
later when ec_tx is bounds checked.

Addresses-Coverity: ("Uninitialized scalar variable")
Fixes: 4f692926 ("ASoC: codecs: lpass-rx-macro: add dapm widgets and route")
Signed-off-by: NColin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210215163313.84026-1-colin.king@canonical.comSigned-off-by: NMark Brown <broonie@kernel.org>

Enable BCLK detection after calibration.
Signed-off-by: NJack Yu <jack.yu@realtek.com>
Link: https://lore.kernel.org/r/20210222090057.29532-2-jack.yu@realtek.comSigned-off-by: NMark Brown <broonie@kernel.org>

Remove 0x100 cache re-sync to solve i2c communication error.
Signed-off-by: NJack Yu <jack.yu@realtek.com>
Link: https://lore.kernel.org/r/20210222090057.29532-1-jack.yu@realtek.comSigned-off-by: NMark Brown <broonie@kernel.org>

There is another MSI board (1462:cc34) that has dual Realtek codecs,
and we need to apply the existing quirk for fixing the conflicts of
Master control.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211743
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210303142346.28182-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>

This adds a new SND_PCI_QUIRK(...) and applies it to the Intel NUC 10
devices. This fixes the issue of the devices not having audio input and
output on the headset jack because the kernel does not recognize when
something is plugged in.

The new quirk was inspired by the quirk for the Intel NUC 8 devices, but
it turned out that the NUC 10 uses another pin. This information was
acquired by black box testing likely pins.
Co-developed-by: NEckhart Mohr <e.mohr@tuxedocomputers.com>
Signed-off-by: NEckhart Mohr <e.mohr@tuxedocomputers.com>
Signed-off-by: NWerner Sembach <wse@tuxedocomputers.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210302180414.23194-1-wse@tuxedocomputers.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

If the platform set the dyn_pcm_assign to true, it will call
hdmi_find_pcm_slot() to find a pcm slot when hdmi/dp monitor is
connected and need to create a pcm.

So far only intel_hsw_common_init() and patch_nvhdmi() set the
dyn_pcm_assign to true, here we let tgl platforms assign the pcm slot
dynamically first, if the driver runs for a period of time and there
is no regression reported, we could set no_fixed_assgin to true in
the intel_hsw_common_init(), and then set it to true in the
patch_nvhdmi().

This change comes from the discussion between Takashi and
Kai Vehmanen. Please refer to:
https://github.com/alsa-project/alsa-lib/pull/118Suggested-and-reviewed-by: NTakashi Iwai <tiwai@suse.de>
Suggested-and-reviewed-by: NKai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: NHui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210301111202.2684-1-hui.wang@canonical.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

This applies a SND_PCI_QUIRK(...) to the Clevo NH55RZQ barebone. This
fixes the issue of the device not recognizing a pluged in microphone.

The device has both, a microphone only jack, and a speaker + microphone
combo jack. The combo jack already works. The microphone-only jack does
not recognize when a device is pluged in without this patch.
Signed-off-by: NEckhart Mohr <e.mohr@tuxedocomputers.com>
Co-developed-by: NWerner Sembach <wse@tuxedocomputers.com>
Signed-off-by: NWerner Sembach <wse@tuxedocomputers.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/0eee6545-5169-ef08-6cfa-5def8cd48c86@tuxedocomputers.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

We rely on implicit includes, list out explicitly what this code
relies on.
Suggested-by: NGuennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: NKai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: NGuennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: NBard Liao <bard.liao@intel.com>
Acked-by: NMark Brown <broonie@kernel.org>
Acked-by: NVinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20210302003125.1178419-8-pierre-louis.bossart@linux.intel.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

The ACPI scan capabilities is called from the intel-dspconfig as well
as the SOF/HDaudio drivers. This creates dependencies and randconfig issues
when HDaudio and SOF/SoundWire are not all configured as modules.

To simplify Kconfig dependencies between HDAudio, SoundWire, SOF and
intel-dspconfig, move the ACPI scan helpers to a dedicated
module. This follows the same idea as NHLT helpers which are already
handled as a dedicated module.

The only functional change is that the kernel parameter to filter
links is now handled by a different module, but that was only provided
for developers needing work-arounds for early BIOS releases.
Reported-by: NArnd Bergmann <arnd@arndb.de>
Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: NKai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: NGuennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: NBard Liao <bard.liao@intel.com>
Acked-by: NMark Brown <broonie@kernel.org>
Acked-by: NVinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20210302003125.1178419-7-pierre-louis.bossart@linux.intel.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

The Kconfig file is way too convoluted. Track platforms where
SoundWire is supported, and add simpler conditions to make sure there
is no module/built-in issue.

The use of 'depends on' is less intuitive if a required 'depend' is
missing, but that's a small price to pay for clarity and simplicity.
Suggested-by: NArnd Bergmann <arnd@arndb.de>
Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: NKai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: NGuennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: NBard Liao <bard.liao@intel.com>
Acked-by: NMark Brown <broonie@kernel.org>
Acked-by: NVinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20210302003125.1178419-6-pierre-louis.bossart@linux.intel.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

There is no reason why we should call the intel_dspcfg helpers from
common code, this should be moved in Intel-specific code and only
called from platforms where a conflict may occur with the HDaudio or
SST/Skylake driver.
Suggested-by: NArnd Bergmann <arnd@arndb.de>
Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: NKai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: NGuennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: NBard Liao <bard.liao@intel.com>
Acked-by: NMark Brown <broonie@kernel.org>
Acked-by: NVinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20210302003125.1178419-5-pierre-louis.bossart@linux.intel.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Move PCI IDs and device-specific definitions out of common code. No
functionality change for now, just code split and removal of
IF_ENABLED() which made the configurations too complicated in case of
reuse of IP across generations.

Additional changes to address the DSP_CONFIG case and SoundWire
depends/select confusions are provided in follow-up patches.
Suggested-by: NArnd Bergmann <arnd@arndb.de>
Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: NKai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: NGuennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: NBard Liao <bard.liao@intel.com>
Acked-by: NMark Brown <broonie@kernel.org>
Acked-by: NVinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20210302003125.1178419-4-pierre-louis.bossart@linux.intel.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

The SOF-ACPI driver is backwards from the normal Linux model, it has a
generic driver that knows about all the specific drivers, as opposed to
having hardware specific drivers that link against a common framework.

This requires ugly Kconfig magic and leads to missed dependencies as
seen in this link error:

arm-linux-gnueabi-ld: sound/soc/sof/sof-pci-dev.o: in function `sof_acpi_probe':
sof-pci-dev.c:(.text+0x1c): undefined reference to `snd_intel_dsp_driver_probe'

Change it to use the normal probe order of starting with a specific
device in a driver, turning the sof-acpi-dev.c driver into a
library (exported symbols are name-spaced to avoid symbol pollution).

For backwards-compatibility with previous Kconfigs, the default values
for platform drivers uses the top-level ACPI configurations. The
modules were also renamed to allow for gradual transitions in test
scripts.
Co-developed-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: NArnd Bergmann <arnd@arndb.de>
Reviewed-by: NKai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: NGuennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: NBard Liao <bard.liao@intel.com>
Acked-by: NMark Brown <broonie@kernel.org>
Acked-by: NVinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20210302003125.1178419-3-pierre-louis.bossart@linux.intel.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Multiple bug reports report issues with the SOF and SST drivers when
dealing with single microphone cases.

We currently read the DMIC array information unconditionally but we
don't check that the configuration type is actually a mic array.

When the DMIC link does not rely on a mic array configuration, the
recommendation is to check the format information to infer the maximum
number of channels, and map this to the number of microphones.

This leaves a potential for a mismatch between actual microphones
available in hardware and what the ACPI table contains, but we have no
other source of information.

Note that single microphone configurations can alternatively be
handled with a 'mic array' configuration along with a 'vendor-defined'
geometry.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201251
BugLink: https://github.com/thesofproject/linux/issues/2725
Fixes: 7a33ea70 ('ALSA: hda: intel-nhlt: handle NHLT VENDOR_DEFINED DMIC geometry')
Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: NGuennadi Liakhovetski <guennadi.liakhovetski@intel.com>
Reviewed-by: NRander Wang <rander.wang@intel.com>
Reviewed-by: NKai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210302000146.1177770-1-pierre-louis.bossart@linux.intel.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

v5.12-rc1 flags new warnings with make W=1, fix missing or broken
function descriptors.

sound/pci/hda/hda_codec.c:3492: warning: expecting prototype for
snd_hda_input_mux_info_info(). Prototype was for
snd_hda_input_mux_info() instead

sound/pci/hda/hda_codec.c:3521: warning: expecting prototype for
snd_hda_input_mux_info_put(). Prototype was for
snd_hda_input_mux_put() instead

sound/pci/hda/hda_codec.c:3958: warning: expecting prototype for
_snd_hda_pin_ctl(). Prototype was for _snd_hda_set_pin_ctl() instead

sound/pci/hda/hda_jack.c:223: warning: expecting prototype for
snd_hda_set_dirty_all(). Prototype was for
snd_hda_jack_set_dirty_all() instead

sound/pci/hda/hda_jack.c:309: warning: expecting prototype for
snd_hda_jack_detect_enable_mst(). Prototype was for
snd_hda_jack_detect_enable_callback_mst() instead

sound/pci/hda/hda_generic.c:3933: warning: expecting prototype for
snd_dha_gen_add_mute_led_cdev(). Prototype was for
snd_hda_gen_add_mute_led_cdev() instead

sound/pci/hda/hda_generic.c:4093: warning: expecting prototype for
snd_dha_gen_add_micmute_led_cdev(). Prototype was for
snd_hda_gen_add_micmute_led_cdev() instead

sound/pci/hda/patch_ca0132.c:2357: warning: expecting prototype for
Prepare and send the SCP message to DSP(). Prototype was for
dspio_scp() instead

sound/pci/hda/patch_ca0132.c:2883: warning: expecting prototype for
Allocate router ports(). Prototype was for dsp_allocate_router_ports()
instead

sound/pci/hda/patch_ca0132.c:3202: warning: expecting prototype for
Write a block of data into DSP code or data RAM using pre(). Prototype
was for dspxfr_one_seg() instead

sound/pci/hda/patch_ca0132.c:3397: warning: expecting prototype for
data overlay to DSP memories(). Prototype was for dspxfr_image()
instead

sound/hda/hdac_regmap.c:393: warning: expecting prototype for
snd_hdac_regmap_init(). Prototype was for snd_hdac_regmap_exit()
instead

sound/hda/ext/hdac_ext_controller.c:142: warning: expecting prototype
for snd_hdac_ext_bus_get_link_index(). Prototype was for
snd_hdac_ext_bus_get_link() instead

sound/hda/ext/hdac_ext_stream.c:140: warning: expecting prototype for
snd_hdac_ext_linkstream_start(). Prototype was for
snd_hdac_ext_link_stream_start() instead
Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210301174617.116960-1-pierre-louis.bossart@linux.intel.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

This commit only contains the fix about the `URB_CONTROL` request
direction to set the samplerate of Pioneer DJM devices (`URB_CONTROL out`).

Fixes: 3b85f5fc ("ALSA: usb-audio: Add DJM450 to Pioneer format quirk")
Signed-off-by: NNicolas MURE <nicolas.mure2019@gmail.com>
Link: https://lore.kernel.org/r/20210301142927.14552-1-nicolas.mure2019@gmail.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

The Corsair Virtuoso SE RGB Wireless is a USB headset with a mic and a
sidetone feature. Assign the Corsair Virtuoso name map to the SE product
ids as well, in order to label its mixer appropriately and allow
userspace to pick the correct volume controls.
Signed-off-by: NAndrea Fagiani <andfagiani@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/40bbdf55-f854-e2ee-87b4-183e6451352c@gmail.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

The Acer SWIFT Swift SF314-54/55 laptops with ALC256 cannot detect
both the headset mic and the internal mic. Introduce new fixup
to enable the jack sense and the headset mic. However, the internal
mic actually connects to Intel SST audio. It still needs Intel SST
support to make internal mic capture work.
Signed-off-by: NChris Chiu <chris.chiu@canonical.com>
Acked-by: NJian-Hong Pan <jhp@endlessos.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210226010440.8474-1-chris.chiu@canonical.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Currently the mask operation on variable conf is just 3 bits so
the switch statement case value of 8 is unreachable dead code.
The function daio_mgr_dao_init can be passed a 4 bit value,
function dao_rsc_init calls it with conf set to:

     conf = (desc->msr & 0x7) | (desc->passthru << 3);

so clearly when desc->passthru is set to 1 then conf can be
at least 8.

Fix this by changing the mask to 0xf.

Fixes: 8cc72361 ("ALSA: SB X-Fi driver merge")
Signed-off-by: NColin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210227001527.1077484-1-colin.king@canonical.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

The recent fix for the hw constraints for implicit feedback streams
via commit e4ea77f8 ("ALSA: usb-audio: Always apply the hw
constraints for implicit fb sync") added the check of the matching
endpoints and whether those EPs are already opened.  This is needed
and correct, per se, even for the normal streams without the implicit
feedback, as the endpoint setup is exclusive.

However, it's reported that there seem applications that behave in
unexpected ways to update the hw_params without clearing the previous
setup via hw_free, and those hit a problem now: then hw_params is
called with still the previous EP setup kept, hence it's restricted
with the previous own setup.  Although the obvious fix is to call
snd_pcm_hw_free() API in the application side, it's a kind of
unwelcome change.

This patch tries to ease the situation: in the endpoint check, we add
a couple of more conditions and now skip the endpoint that is being
used only by the stream in question itself.  That is, in addition to
the presence check of ep (ep->cur_audiofmt is non-NULL), when the
following conditions are met, we skip such an ep:
- ep->opened == 1, and
- ep->cur_audiofmt == subs->cur_audiofmt.

subs->cur_audiofmt is non-NULL only if it's a re-setup of hw_params,
and ep->cur_audiofmt points to the currently set up parameters.  So if
those match, it must be this stream itself.

Fixes: e4ea77f8 ("ALSA: usb-audio: Always apply the hw constraints for implicit fb sync")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211941
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210228080138.9936-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Some USB audio firmware seem to report broken dB values for the volume
controls, and this screws up applications like PulseAudio who blindly
trusts the given data.  For example, Edifier G2000 reports a PCM
volume from -128dB to -127dB, and this results in barely inaudible
sound.

This patch adds a sort of sanity check at parsing the dB values in
USB-audio driver and disables the dB reporting if the range looks
bogus.  Here, we assume -96dB as the bottom line of the max dB.

Note that, if one can figure out that proper dB range later, it can be
patched in the mixer maps.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211929
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210227105737.3656-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>

The commit 93db51d0 ("ALSA: usb-audio: Check valid altsetting at
parsing rates for UAC2/3") changed the behavior of the function
set_sample_rate_v2v3() slightly to treat the inconsistent sample rate
as an error.  It was done by assumption that the sample rate
validation should have been done at the parser phase as implemented in
that patch.  But the validation is later selectively enabled only for
certain devices as it causes a regression (the commit fe773b87
"ALSA: usb-audio: workaround for iface reset issue"), and now the
inconsistency surfaced as a fatal error while it worked in the past as
is, as reported for FiiO M3K DAC.

For recovering from the regression, change set_sample_rate_v2v3()
again to ignore the sample rate difference as non-error.

BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1182633
Fixes: 93db51d0 ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210227082002.21185-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>

In case of error, the function devm_platform_ioremap_resource()
returns ERR_PTR() and never returns NULL. The NULL test in the
return value check should be replaced with IS_ERR().

Fixes: 1448f8ac ("sound: Add n64 driver")
Reported-by: NHulk Robot <hulkci@huawei.com>
Signed-off-by: NWei Yongjun <weiyongjun1@huawei.com>
Reviewed-by: NLauri Kasanen <cand@gmx.com>
Link: https://lore.kernel.org/r/20210224013803.2146953-1-weiyongjun1@huawei.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Add rules to select SOF driver for Alder Lake systems if a digital
microphone or SoundWire codecs are present in the system. This is
following same rules as for older Tiger Lake systems.
Signed-off-by: NKai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: NXiuli Pan <xiuli.pan@intel.com>
Reviewed-by: NRanjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210210111310.2227417-1-kai.vehmanen@linux.intel.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

In case DPCM runtime has multiple CPU DAIs, dpcm_init_runtime_hw() is
called multiple times, once for each CPU DAI. This will lead to
ignoring hw limits of all but the last DAI.

Fix this by moving soc_pcm_hw_init() up by one level to
dpcm_init_runtime_hw().

Fixes: 140f553d ("ASoC: soc-pcm: fix hwparams min/max init for dpcm")
Suggested-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: NKai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210216172251.3023723-1-kai.vehmanen@linux.intel.comSigned-off-by: NMark Brown <broonie@kernel.org>

The Acer One S1002 tablet is using an analog mic on IN1 and has
its jack-detect connected to JD2_IN4N, instead of using the default
IN3 for its internal mic and JD1_IN4P for jack-detect.

Note it is also using AIF2 instead of AIF1 which is somewhat unusual,
this is correctly advertised in the ACPI CHAN package, so the speakers
do work without the quirk.

Add a quirk for the mic and jack-detect settings.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Acked-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210216213555.36555-5-hdegoede@redhat.comSigned-off-by: NMark Brown <broonie@kernel.org>

Add a DMI quirk for the Jumper EZpad 7 tablet, this tablet has
a jack-detect switch which reads 1/high when a jack is inserted,
rather then using the standard active-low setup which most
jack-detect switches use. All other settings are using the defaults.

Add a DMI-quirk setting the defaults + the BYT_RT5651_JD_NOT_INV
flags for this.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Acked-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210216213555.36555-4-hdegoede@redhat.comSigned-off-by: NMark Brown <broonie@kernel.org>