| Commit message (Collapse) | Author | Age | Files | Lines |
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The driver was setting bit clock polarity opposite to intended polarity.
Also simplify the code by grouping ADC and DAC clock configurations into
a single field.
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-2-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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In 61fbeb5 the sirf prima/atlas drivers were removed. This cleans
up a stray header and some Kconfig entries for the codec that
were missed in the process.
Fixes: 61fbeb5dcb3d (ASoC: remove sirf prima/atlas drivers)
Signed-off-by: Peter Robinson <pbrobinson@gmail.com>
Cc: Arnd Bergmann <arnd@arndb.de>
Cc: Mark Brown <broonie@kernel.org>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20210307162338.1160604-1-pbrobinson@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Attempting to use the RX MIX path at 48kHz plays at 96kHz, because these
controls are incorrectly toggling the first bit of the register, which
is part of the FS_RATE field.
Fix the problem by using the same method used by the "WSA RX_MIX EC0_MUX"
control, which is to use SND_SOC_NOPM as the register and use an enum in
the shift field instead.
Fixes: 2c4066e5d428 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route")
Signed-off-by: Jonathan Marek <jonathan@marek.ca>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210305005049.24726-1-jonathan@marek.ca
Signed-off-by: Mark Brown <broonie@kernel.org>
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An interface can have multiple decimators enabled, so loop over all active
decimators. Otherwise only one channel will be unmuted, and other channels
will be zero. This fixes recording from dual DMIC as a single two channel
stream.
Also remove the now unused "active_decimator" field.
Fixes: 908e6b1df26e ("ASoC: codecs: lpass-va-macro: Add support to VA Macro")
Signed-off-by: Jonathan Marek <jonathan@marek.ca>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210304215646.17956-1-jonathan@marek.ca
Signed-off-by: Mark Brown <broonie@kernel.org>
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Many systems do not use ACPI and hence do not provide a DMI table. On
non-ACPI systems a warning, such as the following, is printed on boot.
WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name!
The variable 'dmi_available' is not exported and so currently cannot be
used by kernel modules without adding an accessor. However, it is
possible to use the function is_acpi_device_node() to determine if the
sound card is an ACPI device and hence indicate if we expect a DMI table
to be present. Therefore, call is_acpi_device_node() to see if we are
using ACPI and only parse the DMI table if we are booting with ACPI.
Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/20210303115526.419458-1-jonathanh@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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We only unregister the platform device during the .remove operation,
but if the probe fails we will never reach this sequence.
Suggested-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Fixes: dd96daca6c83e ("ASoC: SOF: Intel: Add APL/CNL HW DSP support")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20210302003410.1178535-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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<hdegoede@redhat.com>:
Hi All,
Here is a series of rt5640/rt5651 volume-control fixes which I wrote
while working on a bytcr-rt5640 UCM profile patch-series adding
hardware-volume control to devices using this UCM profile.
The UCM series will also work on older kernels, but it works best on
kernels with this series applied, giving e.g. finer grained volume
control and support for hardware muting the outputs.
Regards,
Hans
Hans de Goede (5):
ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor
of 10
ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor
of 10
ASoC: rt5640: Add emulated 'DAC1 Playback Switch' control
ASoC: rt5640: Rename 'Mono DAC Playback Volume' to 'DAC2 Playback
Volume'
ASoC: Intel: bytcr_rt5640: Add used AIF to the components string
sound/soc/codecs/rt5640.c | 106 +++++++++++++++++++++++---
sound/soc/codecs/rt5640.h | 4 +
sound/soc/codecs/rt5651.c | 4 +-
sound/soc/intel/boards/bytcr_rt5640.c | 11 ++-
4 files changed, 111 insertions(+), 14 deletions(-)
--
2.30.1
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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 3f31f7d9b540 ("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: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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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 3f31f7d9b540 ("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: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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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: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210228160441.241110-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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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.pdf
Signed-off-by: Benjamin Rood <benjaminjrood@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev
Signed-off-by: Mark Brown <broonie@kernel.org>
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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: fbdae7d6d04d ("ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 Detachable quirks")
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210224105052.42116-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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<hdegoede@redhat.com>:
Hi All,
While working on adding hardware-volume control support to the UCM
profile for the rt5672 and on adding LED trigger support to the
rt5670 codec driver. I hit / noticed a couple of issues this series
fixes these issues.
Regards,
Hans
Hans de Goede (4):
ASoC: rt5670: Remove 'OUT Channel Switch' control
ASoC: rt5670: Remove 'HP Playback Switch' control
ASoC: rt5670: Remove ADC vol-ctrl mute bits poking from Sto1 ADC mixer
settings
ASoC: rt5670: Add emulated 'DAC1 Playback Switch' control
sound/soc/codecs/rt5670.c | 110 +++++++++++++++++++++++++++++++++-----
sound/soc/codecs/rt5670.h | 9 ++--
2 files changed, 101 insertions(+), 18 deletions(-)
--
2.30.1
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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: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-5-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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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: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-4-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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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: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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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: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210215142118.308516-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add missed MODULE_DEVICE_TABLE for the driver can be loaded
automatically at boot.
Fixes: 920884777480 ("ASoC: ak5558: Add support for AK5558 ADC driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1614149872-25510-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add missed MODULE_DEVICE_TABLE for the driver can be loaded
automatically at boot.
Fixes: 08660086eff9 ("ASoC: ak4458: Add support for AK4458 DAC driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1614149872-25510-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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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: 4f14f5c11db1 ("ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode")
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20210216114221.26635-1-shc_work@mail.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
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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: 4f692926f562 ("ASoC: codecs: lpass-rx-macro: add dapm widgets and route")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210215163313.84026-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Enable BCLK detection after calibration.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://lore.kernel.org/r/20210222090057.29532-2-jack.yu@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Remove 0x100 cache re-sync to solve i2c communication error.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://lore.kernel.org/r/20210222090057.29532-1-jack.yu@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The HP EliteBook 850 G8 Notebook PC is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210316094236.89028-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The HP EliteBook 840 G8 Notebook PC is using ALC236 codec which is
using 0x02 to control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210316074626.79895-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The HP EliteBook 840 G8 Notebook PC is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210316065452.75659-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Built-in microphone and combojack on Xiaomi Notebook Pro (1d72:1701) needs
to be fixed, the existing quirk for Dell works well on that machine.
Signed-off-by: Xiaoliang Yu <yxl_22@outlook.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/OS0P286MB02749B9E13920E6899902CD8EE6C9@OS0P286MB0274.JPNP286.PROD.OUTLOOK.COM
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There is another fix for headset-mic problem on Redmibook (1d72:1602),
it also works on Redmibook Air (1d72:1947), which has the same issue.
Signed-off-by: Xiaoliang Yu <yxl_22@outlook.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/TYBP286MB02856DC016849DEA0F9B6A37EE6F9@TYBP286MB0285.JPNP286.PROD.OUTLOOK.COM
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When node is removed from IEEE 1394 bus, any transaction fails to the node.
In the case, ALSA dice driver doesn't stop isochronous contexts even if
they are running. As a result, null pointer dereference occurs in callback
from the running context.
This commit fixes the bug to release isochronous contexts always.
Cc: <stable@vger.kernel.org> # v5.4 or later
Fixes: e9f21129b8d8 ("ALSA: dice: support AMDTP domain")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210312093407.23437-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Recently we found the micmute led init state is not correct after
freshly installing the ubuntu linux on a Lenovo AIO machine. The
internal mic is not muted, but the micmute led is on and led mode is
'follow mute'. If we mute internal mic, the led is keeping on, then
unmute the internal mic, the led is off. And from then on, the
micmute led will work correctly.
So the micmute led init state is not correct. The led is controlled
by codec gpio (ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), in the
patch_realtek, the gpio data is set to 0x4 initially and the led is
on with this data. In the hda_generic, the led_value is set to
0 initially, suppose users set the 'capture switch' to on from
user space and the micmute led should change to be off with this
operation, but the check "if (val == spec->micmute_led.led_value)" in
the call_micmute_led_update() will skip the led setting.
To guarantee the led state will be set by the 1st time of changing
"Capture Switch", set -1 to the init led_value.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210312041408.3776-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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MODULE_SUPPORTED_DEVICE was added in pre-git era and never was
implemented. We can safely remove it, because the kernel has grown
to have many more reliable mechanisms to determine if device is
supported or not.
Signed-off-by: Leon Romanovsky <leonro@nvidia.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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The per_pin->work might be still floating at the suspend, and this may
hit the access to the hardware at an unexpected timing. Cancel the
work properly at the suspend callback for avoiding the buggy access.
Note that the bug doesn't trigger easily in the recent kernels since
the work is queued only when the repoll count is set, and usually it's
only at the resume callback, but it's still possible to hit in
theory.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182377
Reported-and-tested-by: Abhishek Sahu <abhsahu@nvidia.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210310112809.9215-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When HD-audio bus receives unsolicited events during its system
suspend/resume (S3 and S4) phase, the controller driver may still try
to process events although the codec chips are already (or yet)
powered down. This might screw up the codec communication, resulting
in CORB/RIRB errors. Such events should be rather skipped, as the
codec chip status such as the jack status will be fully refreshed at
the system resume time.
Since we're tracking the system suspend/resume state in codec
power.power_state field, let's add the check in the common unsol event
handler entry point to filter out such events.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182377
Tested-by: Abhishek Sahu <abhsahu@nvidia.com>
Cc: <stable@vger.kernel.org> # 183ab39eb0ea: ALSA: hda: Initialize power_state
Link: https://lore.kernel.org/r/20210310112809.9215-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The HD-audio controller driver processes the unsolicited events via
its work asynchronously, and this might be pending when the system
goes to suspend. When a lengthy event handling like ELD byte reads is
running, this might trigger unexpected accesses among suspend/resume
procedure, typically seen with Nvidia driver that still requires the
handling via unsolicited event verbs for ELD updates.
This patch adds the flush of unsol_work to assure that pending events
are processed before going into suspend.
Buglink: https://bugzilla.suse.com/show_bug.cgi?id=1182377
Reported-and-tested-by: Abhishek Sahu <abhsahu@nvidia.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210310112809.9215-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The problem was in wrong "if" placement. chip->quirk_type is freed
in snd_card_free_when_closed(), but inside if statement it's accesed.
Fixes: 9799110825db ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()")
Signed-off-by: Pavel Skripkin <paskripkin@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/16da19126ff461e5e64a9aec648cce28fb8ed73e.1615242183.git.paskripkin@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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syzbot reported null pointer dereference in usb_audio_probe.
The problem was in case, when quirk == NULL. It's not an
error condition, so quirk must be checked before dereferencing.
Call Trace:
usb_probe_interface+0x315/0x7f0 drivers/usb/core/driver.c:396
really_probe+0x291/0xe60 drivers/base/dd.c:554
driver_probe_device+0x26b/0x3d0 drivers/base/dd.c:740
__device_attach_driver+0x1d1/0x290 drivers/base/dd.c:846
bus_for_each_drv+0x15f/0x1e0 drivers/base/bus.c:431
__device_attach+0x228/0x4a0 drivers/base/dd.c:914
bus_probe_device+0x1e4/0x290 drivers/base/bus.c:491
device_add+0xbdb/0x1db0 drivers/base/core.c:3242
usb_set_configuration+0x113f/0x1910 drivers/usb/core/message.c:2164
usb_generic_driver_probe+0xba/0x100 drivers/usb/core/generic.c:238
usb_probe_device+0xd9/0x2c0 drivers/usb/core/driver.c:293
really_probe+0x291/0xe60 drivers/base/dd.c:554
driver_probe_device+0x26b/0x3d0 drivers/base/dd.c:740
__device_attach_driver+0x1d1/0x290 drivers/base/dd.c:846
bus_for_each_drv+0x15f/0x1e0 drivers/base/bus.c:431
__device_attach+0x228/0x4a0 drivers/base/dd.c:914
bus_probe_device+0x1e4/0x290 drivers/base/bus.c:491
device_add+0xbdb/0x1db0 drivers/base/core.c:3242
usb_new_device.cold+0x721/0x1058 drivers/usb/core/hub.c:2555
hub_port_connect drivers/usb/core/hub.c:5223 [inline]
hub_port_connect_change drivers/usb/core/hub.c:5363 [inline]
port_event drivers/usb/core/hub.c:5509 [inline]
hub_event+0x2357/0x4320 drivers/usb/core/hub.c:5591
process_one_work+0x98d/0x1600 kernel/workqueue.c:2275
worker_thread+0x64c/0x1120 kernel/workqueue.c:2421
kthread+0x3b1/0x4a0 kernel/kthread.c:292
ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:294
Reported-by: syzbot+719da9b149a931f5143f@syzkaller.appspotmail.com
Fixes: 9799110825db ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()")
Signed-off-by: Pavel Skripkin <paskripkin@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/f1ebad6e721412843bd1b12584444c0a63c6b2fb.1615242183.git.paskripkin@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The new AE-5 Plus model has a different Subsystem ID compared to the
non-plus model. Adding the new id to the list of quirks.
Signed-off-by: Simeon Simeonoff <sim.simeonoff@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/998cafbe10b648f724ee33570553f2d780a38963.camel@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The mute and mic-mute LEDs on HP ZBook Studio G5 are controlled via
GPIO bits 0x10 and 0x20, respectively, and we need the extra setup for
those.
As the similar code is already present for other HP models but with
different GPIO pins, this patch factors out the common helper code and
applies those GPIO values for each model.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211893
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210306095018.11746-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Other Plantronics headset models seem requiring the same workaround as
C320-M to add the 20ms delay for the control messages, too. Apply the
workaround generically for devices with the vendor ID 0x047f.
Note that the problem didn't surface before 5.11 just with luck.
Since 5.11 got a big code rewrite about the stream handling, the
parameter setup procedure has changed, and this seemed triggering the
problem more often.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182552
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304085009.4770-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Dell AE515 sound bar (413c:a506) spews the error messages when the
driver tries to read the current sample frequency, hence it needs to
be on the list in snd_usb_get_sample_rate_quirk().
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211551
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304083021.2152-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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On some Lenovo systems if the microphone is disabled in the BIOS
only the NHLT table header is created, with no data. This means
the endpoints field is not correctly set to zero - leading to an
unintialised variable and hence invalid descriptors are parsed
leading to page faults.
The Lenovo firmware team is addressing this, but adding a check
preventing invalid tables being parsed is worthwhile.
Tested on a Lenovo T14.
Tested-by: Philipp Leskovitz <philipp.leskovitz@secunet.com>
Reported-by: Philipp Leskovitz <philipp.leskovitz@secunet.com>
Signed-off-by: Mark Pearson <markpearson@lenovo.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210302141003.7342-1-markpearson@lenovo.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Rear audio on Lenovo ThinkStation P620 stops working after commit
1965c4364bdd ("ALSA: usb-audio: Disable autosuspend for Lenovo
ThinkStation P620"):
[ 6.013526] usbcore: registered new interface driver snd-usb-audio
[ 6.023064] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.023083] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.023090] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.023098] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.023103] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.023110] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.045846] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.045866] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.045877] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.045886] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.045894] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.045908] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
I overlooked the issue because when I was working on the said commit,
only the front audio is tested. Apology for that.
Changing supports_autosuspend in driver is too late for disabling
autosuspend, because it was already used by USB probe routine, so it can
break the balance on the following code that depends on
supports_autosuspend.
Fix it by using usb_disable_autosuspend() helper, and balance the
suspend count in disconnect callback.
Fixes: 1965c4364bdd ("ALSA: usb-audio: Disable autosuspend for Lenovo ThinkStation P620")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304043419.287191-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The microphone in the Plantronics C320-M headset will randomly
fail to initialize properly, at least when using Microsoft Teams.
Introducing a 20ms delay on the control messages appears to
resolve the issue.
Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1065
Tested-by: Andreas Kempe <kempe@lysator.liu.se>
Signed-off-by: John Ernberg <john.ernberg@actia.se>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210303181405.39835-1-john.ernberg@actia.se
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The commit c02f77d32d2c ("ALSA: hda - Workaround for crackled sound on
AMD controller (1022:1457)") introduced a few workarounds for the
recent AMD HD-audio controller, and one of them is the forced BATCH
PCM mode so that PulseAudio avoids the timer-based scheduling. This
was thought to cover for some badly working applications, but this
actually worsens for more others. In total, this wasn't a good idea
to enforce it.
This is a partial revert of the commit above for dropping the PCM
BATCH enforcement part to recover from the regression again.
Fixes: c02f77d32d2c ("ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210308160726.22930-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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: Eckhart Mohr <e.mohr@tuxedocomputers.com>
Signed-off-by: Eckhart Mohr <e.mohr@tuxedocomputers.com>
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210302180414.23194-1-wse@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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/118
Suggested-and-reviewed-by: Takashi Iwai <tiwai@suse.de>
Suggested-and-reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210301111202.2684-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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: Eckhart Mohr <e.mohr@tuxedocomputers.com>
Co-developed-by: Werner Sembach <wse@tuxedocomputers.com>
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/0eee6545-5169-ef08-6cfa-5def8cd48c86@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ALSA/ASoC/SOF/SoundWire: fix Kconfig issues
In January, Intel kbuild bot and Arnd Bergmann reported multiple
issues with randconfig. This patchset builds on Arnd's suggestions to
a) expose ACPI and PCI devices in separate modules, while sof-acpi-dev
and sof-pci-dev become helpers. This will result in minor changes
required for developers/testers, i.e. modprobe snd-sof-pci will no
longer result in a probe. The SOF CI was already updated to deal with
this module dependency change and introduction of new modules.
b) Fix SOF/SoundWire/DSP_config dependencies by moving the code
required to detect SoundWire presence in ACPI tables to sound/hda.
Link: https://lore.kernel.org/r/20210302003125.1178419-1-pierre-louis.bossart@linux.intel.com
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