The input MCLK is 12.288MHz, the desired output sysclk is 11.2896MHz
and sample rate is 44100Hz, with the configuration pllprescale=2,
postscale=sysclkdiv=1, some chip may have wrong bclk
and lrclk output with pll enabled in master mode, but with the
configuration pllprescale=1, postscale=2, the output clock is correct.

>From Datasheet, the PLL performs best when f2 is between
90MHz and 100MHz when the desired sysclk output is 11.2896MHz
or 12.288MHz, so sysclkdiv = 2 (f2/8) is the best choice.

So search available sysclk_divs from 2 to 1 other than from 1 to 2.

Fixes: 84fdc00d

 ("ASoC: codec: wm9860: Refactor PLL out freq search")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1616150926-22892-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>

Merge series "Do not handle MCLK device clock in simple-card-utils" from Sameer Pujar <spujar@nvidia.com>:

With commit 1e30f642 ("ASoC: simple-card-utils: Fix device module clock")
simple-card-utils can control MCLK clock for rate updates or enable/disable.
But this is breaking some platforms where it is expected that codec drivers
would actually handle the MCLK clock. One such example is following platform.
  - "arch/arm64/boot/dts/freescale/fsl-ls1028a-kontron-sl28-var3-ads2.dts"

In above case codec, wm8904, is using internal PLL and configures sysclk
based on fixed MCLK input. In such cases it is expected that, required PLL
output or sysclk, is just passed via set_sysclk() callback and card driver
need not actually update MCLK rate. Instead, codec can take ownership of
this clock and do the necessary configuration.

So the original commit is reverted and codec driver for rt5659 is updated
to fix my board which has this codec.

Sameer Pujar (2):
  ASoC: simple-card-utils: Do not handle device clock...

Add compatible string for new added platforms which support spdif module.
They are i.MX8QXP, i.MX8MM, i.MX8MN, i.MX8MQ.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1615884053-4264-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>

We do some IO operations in the snd_soc_component_set_jack callback
function and snd_soc_component_set_jack() will be called when soc
component is removed. However, we should not access SoundWire registers
when the bus is suspended.
So set regcache_cache_only(regmap, true) to avoid accessing in the
soc component removal process.

Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20210316005254.29699-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

Simple-card/audio-graph-card drivers do not handle MCLK clock when it
is specified in the codec device node. The expectation here is that,
the codec should actually own up the MCLK clock and do necessary setup
in the driver.

Suggested-by: Mark Brown <broonie@kernel.org>
Suggested-by: Michael Walle <michael@walle.cc>
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>

This reverts commit 1e30f642

 ("ASoC: simple-card-utils: Fix device
module clock"). The original patch ended up breaking following platform,
which depends on set_sysclk() to configure internal PLL on wm8904 codec
and expects simple-card-utils to not update the MCLK rate.
 - "arch/arm64/boot/dts/freescale/fsl-ls1028a-kontron-sl28-var3-ads2.dts"

It would be best if codec takes care of setting MCLK clock via DAI
set_sysclk() callback.

Reported-by: Michael Walle <michael@walle.cc>
Suggested-by: Mark Brown <broonie@kernel.org>
Suggested-by: Michael Walle <michael@walle.cc>
Fixes: 1e30f642

 ("ASoC: simple-card-utils: Fix device module clock")
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Tested-by: Michael Walle <michael@walle.cc>
Link: https://lore.kernel.org/r/1615829492-8972-2-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>

This patch correct tdm out bck inverse register to AUDIO_TOP_CON3[3].

Signed-off-by: Jiaxin Yu <jiaxin.yu@mediatek.com>
Link: https://lore.kernel.org/r/1615516005-781-1-git-send-email-jiaxin.yu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>

The max boundary check while parsing dai ids makes
sound card registration fail after common up dai ids.

Fixes: cd3484f7

 ("ASoC: qcom: Fix broken support to MI2S TERTIARY and QUATERNARY")

Signed-off-by: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Link: https://lore.kernel.org/r/20210311154557.24978-1-srivasam@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>

When initialize cadence qspi controller, it is need to set cqspi
to the driver_data field of struct device, because it will be
used in function cqspi_remove/suspend/resume(). Otherwise, there
will be a crash trace as below when invoking these finctions.

Fixes: 31fb632b

 ("spi: Move cadence-quadspi driver to drivers/spi/")
Cc: stable@vger.kernel.org
Signed-off-by: Meng Li <Meng.Li@windriver.com>
Link: https://lore.kernel.org/r/20210311091220.3615-1-Meng.Li@windriver.com
Signed-off-by: Mark Brown <broonie@kernel.org>

Merge series "ASoC: sdm845: array out of bound issues" from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>:

During testing John Stultz and Amit reported few array our bound issues
after enabling bound sanitizer

This patch series attempts to fix those!

changes since v1:
	- make sure the wcd is not de-referenced without intialization

Srinivas Kandagatla (3):
  ASoC: qcom: sdm845: Fix array out of bounds access
  ASoC: qcom: sdm845: Fix array out of range on rx slim channels
  ASoC: codecs: wcd934x: add a sanity check in set channel map

 sound/soc/codecs/wcd934x.c | 6 ++++++
 sound/soc/qcom/sdm845.c    | 6 +++---
 2 files changed, 9 insertions(+), 3 deletions(-)

--
2.21.0

The ADSPCS_SPA is Set Power Active bit. To check if DSP is powered
down, we need to check ADSPCS_CPA, the Current Power Active bit.

Fixes: 747503b1

 ("ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations")
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210309004127.4940-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

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: a61f3b4f

 ("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>

WCD934x has only 13 RX SLIM ports however we are setting it as 16
in set_channel_map, this will lead to array out of bounds error!

Orignally caught by enabling USBAN array out of bounds check:

Fixes: 5caf64c6

 ("ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210309142129.14182-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>

Static analysis Coverity had detected a potential array out-of-bounds
write issue due to the fact that MAX AFE port Id was set to 16 instead
of using AFE_PORT_MAX macro.

Fix this by properly using AFE_PORT_MAX macro.

Fixes: 1b93a884

 ("ASoC: qcom: sdm845: handle soundwire stream")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210309142129.14182-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>

Merge series "Report jack and button detection + Capture Support" from Lucas Tanure <tanureal@opensource.cirrus.com>:

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 61fbeb5d 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: 61fbeb5d

 (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: 2c4066e5

 ("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: 908e6b1d

 ("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>

dev_pm_ops already enable/disable the codec if not in use

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-5-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>

The minimum value is 0x3f (-63dB), which also is mute

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-4-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>

Remove the hard coded 32 bits width and replace with the correct width
calculated by params_width.

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210305173442.195740-3-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>

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>

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: dd96daca

 ("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>

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 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: 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 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: 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>

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: 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>

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: 92088477

 ("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: 08660086

 ("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>

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: 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>

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: 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>