set channel map can be passed with a channel maps, however if
the number of channels that are passed are more than the actual
supported channels then we would be accessing array out of bounds.
So add a sanity check to validate these numbers!
Fixes: a61f3b4f47 ("ASoC: wcd934x: add support to wcd9340/wcd9341 codec")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210309142129.14182-4-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Hi All,
Here is a patch series for reporting to user space jack and button events and
add the support for Capture. With some cleanups and fixes along the way.
Regards,
Lucas Tanure
Lucas Tanure (12):
ASoC: cs42l42: Fix Bitclock polarity inversion
ASoC: cs42l42: Fix channel width support
ASoC: cs42l42: Fix mixer volume control
ASoC: cs42l42: Don't enable/disable regulator at Bias Level
ASoC: cs42l42: Always wait at least 3ms after reset
ASoC: cs42l42: Remove power if the driver is being removed
ASoC: cs42l42: Disable regulators if probe fails
ASoC: cs42l42: Provide finer control on playback path
ASoC: cs42l42: Set clock source for both ways of stream
ASoC: cs42l42: Add Capture Support
ASoC: cs42l42: Report jack and button detection
ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called
Richard Fitzgerald (3):
ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT
ASoC: cs42l42: Only start PLL if it is needed
ASoC: cs42l42: Wait for PLL to lock before switching to it
sound/soc/codecs/cs42l42.c | 437 +++++++++++++++++++++----------------
sound/soc/codecs/cs42l42.h | 41 +++-
2 files changed, 282 insertions(+), 196 deletions(-)
--
2.30.1
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: 61fbeb5dcb (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>
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: 2c4066e5d4 ("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>
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: 908e6b1df2 ("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>
This delay is part of the power-up sequence defined in the datasheet.
A runtime_resume is a power-up so must also include the delay.
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-6-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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
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>
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>
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 3f31f7d9b5 ("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>
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 3f31f7d9b5 ("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>
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
Add missed MODULE_DEVICE_TABLE for the driver can be loaded
automatically at boot.
Fixes: 9208847774 ("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>
Add missed MODULE_DEVICE_TABLE for the driver can be loaded
automatically at boot.
Fixes: 08660086ef ("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>
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>
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>
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>
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>
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: 4f692926f5 ("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>
This callback structure has never been used and it is not clear why it
was added in the first place. Remove it to clear up the code a little.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210211172106.16258-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Qualcomm LPASS (Low Power Audio SubSystem) has internal codec
TX macro block which is used for connecting with external
Soundwire TX Codecs like WCD938x.
This patch adds support to the codec part of the TX Macro block
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210211122735.5691-7-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
LPASS RX Codec Macro is available in Qualcomm LPASS (Low Power Audio SubSystem).
This is used for connecting with SoundWire devices like WCD938x Codecs to provide
headphone/ear/lineout functionality.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210211122735.5691-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for the internal codec found in the JZ4760 SoC from Ingenic.
Signed-off-by: Christophe Branchereau <cbranchereau@gmail.com>
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Link: https://lore.kernel.org/r/20210123140958.12895-3-paul@crapouillou.net
Signed-off-by: Mark Brown <broonie@kernel.org>
No need to show the options to build Ingenic-specific drivers on all
MIPS kernel configurations if Ingenic SoCs support is not enabled.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Link: https://lore.kernel.org/r/20210123140958.12895-2-paul@crapouillou.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Existing hdmi-codec driver only support standard pcm format.
Support of IEC958 encoded format pass from ALSA IEC958 plugin is needed
so that the IEC958 encoded data can be streamed to the HDMI chip.
Signed-off-by: Sia Jee Heng <jee.heng.sia@intel.com>
Link: https://lore.kernel.org/r/20210204014258.10197-2-jee.heng.sia@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
regcache sync will be done in sdw device suspend/resume functions.
And we have different jack detection mechanism for SoundWire.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210204201739.25206-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Follow pattern from other drivers and use cancel_work_sync() for both
.remove() and .suspend().
Fixes: 03f6fc6de9 ('ASoC: rt5682: Add the soundwire support')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Link: https://lore.kernel.org/r/20210204201739.25206-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Make sure the workqueues are not running after the .remove() callback,
which can lead to timeout errors.
A previous fix to use cancel_work_sync was applied for the suspend
case but the remove case is missing
Fixes: 501ef01339 ('ASoC: rt711: wait for the delayed work to finish when the system suspends')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Link: https://lore.kernel.org/r/20210204201739.25206-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Make sure the workqueues are not running after the .remove() callback,
which can lead to timeout errors.
A previous fix to use cancel_work_sync was applied for the suspend
case but the remove case is missing
Fixes: 5f2df2a458 ('ASoC: rt700: wait for the delayed work to finish when the system suspends')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Link: https://lore.kernel.org/r/20210204201739.25206-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit a06cd8cf97 ("ASoC: da7218: skip of_device_id table
when !CONFIG_OF") because we want to make of_match_device() stop using
of_match_ptr() internally, confusing compilers and causing ifdef
pollution.
Reported-by: kernel test robot <lkp@intel.com>
Cc: Geert Uytterhoeven <geert+renesas@glider.be>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Cc: Mark Brown <broonie@kernel.org>
Signed-off-by: Stephen Boyd <swboyd@chromium.org>
Link: https://lore.kernel.org/r/20210202192016.49028-1-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
DT properties "dmic-mode" and "mic-type-X" are optional. Reduces the
log verbosity and changes the message a bit to avoid misleading.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20210202033557.1621029-1-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The correct mask is 0x1f8 (Bit 3-8), but due to missing BIT() 0xf (Bit
0-3) was set instead. This means setting of CPCAP_BIT_MIC1_RX_TIMESLOT0
(Bit 3) still worked (part of both masks). On the other hand the code
does not properly clear the other MIC timeslot bits. I think this
is not a problem, since they are probably initialized to 0 and not
touched by the driver anywhere else. But the mask also contains some
wrong bits, that will be cleared. Bit 0 (CPCAP_BIT_SMB_CDC) should be
safe, since the driver enforces it to be 0 anyways.
Bit 1-2 are CPCAP_BIT_FS_INV and CPCAP_BIT_CLK_INV. This means enabling
audio recording forces the codec into SND_SOC_DAIFMT_NB_NF mode, which
is obviously bad.
The bug probably remained undetected, because there are not many use
cases for routing microphone to the CPU on platforms using cpcap and
user base is small. I do remember having some issues with bad sound
quality when testing voice recording back when I wrote the driver.
It probably was this bug.
Fixes: f6cdf2d344 ("ASoC: cpcap: new codec")
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Reviewed-by: Tony Lindgren <tony@atomide.com>
Link: https://lore.kernel.org/r/20210123172945.3958622-1-sre@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_soc_put_volsw in max98373_feedback_get is a typo, change it
to snd_soc_get_volsw.
Fixes: 349dd23931 ("ASoC: max98373: don't access volatile registers in bias level off")
Signed-off-by: Judy Hsiao <judyhsiao@google.com>
Link: https://lore.kernel.org/r/20210127135620.1143942-1-judyhsiao@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Reset (aka power off) happens when the reset gpio is made active.
Change function name to ak4458_reset to match devicetree property "reset-gpios"
Signed-off-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Reviewed-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://lore.kernel.org/r/ce650f47-4ff6-e486-7846-cc3d033f3601@blennerhassett.gen.nz
Signed-off-by: Mark Brown <broonie@kernel.org>
