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.comSigned-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.comSigned-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.comSigned-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.comSigned-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.comSigned-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.comSigned-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.comSigned-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.comSigned-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.pdfSigned-off-by: Benjamin Rood <benjaminjrood@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-devSigned-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.comSigned-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.comSigned-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.comSigned-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.comSigned-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.comSigned-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.comSigned-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.comSigned-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.comSigned-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.comSigned-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.ruSigned-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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

Merge series "ASoC: Intel: bytcr_rt5640: Add quirks for 4 more tablet / 2-in-1 models" from Hans de Goede <hdegoede@redhat.com>:

Hi All,

Here is a patch series adding quirks with device-specific settings for
4 more tablet / 2-in-1 models.

Regards,

Hans

Hans de Goede (4):
  ASoC: Intel: bytcr_rt5640: Add quirk for the Estar Beauty HD MID 7316R tablet
  ASoC: Intel: bytcr_rt5640: Add quirk for the Voyo Winpad A15 tablet
  ASoC: Intel: bytcr_rt5651: Add quirk for the Jumper EZpad 7 tablet
  ASoC: Intel: bytcr_rt5640: Add quirk for the Acer One S1002 tablet

 sound/soc/intel/boards/bytcr_rt5640.c | 37 +++++++++++++++++++++++++++
 sound/soc/intel/boards/bytcr_rt5651.c | 13 ++++++++++
 2 files changed, 50 insertions(+)

--
2.30.1

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: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210216172251.3023723-1-kai.vehmanen@linux.intel.comSigned-off-by: Mark 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: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210216213555.36555-5-hdegoede@redhat.comSigned-off-by: Mark 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: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210216213555.36555-4-hdegoede@redhat.comSigned-off-by: Mark Brown <broonie@kernel.org>

The Voyo Winpad A15 tablet uses a Bay Trail (non CR) SoC, so it is using
SSP2 (AIF1) and it mostly works with the defaults. But instead of using
DMIC1 it is using an analog mic on IN1, add a quirk for this.
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/20210216213555.36555-3-hdegoede@redhat.comSigned-off-by: Mark Brown <broonie@kernel.org>

The Estar Beauty HD MID 7316R tablet almost fully works with out default
settings. The only problem is that it has only 1 speaker so any sounds
only playing on the right channel get lost.

Add a quirk for this model using the default settings + MONO_SPEAKER.
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/20210216213555.36555-2-hdegoede@redhat.comSigned-off-by: Mark Brown <broonie@kernel.org>

When runtime is initialized with dpcm_init_runtime_hw(), some of the
min/max calculations assume that defaults are set. For example
calculation of channel min/max values may be done using zero-initialized
data and soc_pcm_hw_update_chan() will always return max-channels of 0
in this case. This will result in failure to open the PCM at all.

Fix the issue by calling soc_pcm_hw_init() before calling any
soc_pcm_hw_update_*() functions.

Remove the conditional code on runtime->hw.formats as this field
is anyways set in soc_pcm_hw_init().

Fixes: 6cb56a45 ("ASoC: soc-pcm: add soc_pcm_hw_update_chan()")
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210214220414.2876690-1-kai.vehmanen@linux.intel.comSigned-off-by: Mark Brown <broonie@kernel.org>

We have soc_pcm_hw_update_xxx() now.
This patch creates same function for format.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/87pn1g90oa.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: Mark Brown <broonie@kernel.org>

We have soc_pcm_hw_update_rate() now.
This patch creates same function for chan.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/87r1lw90oo.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: Mark Brown <broonie@kernel.org>

To update hw, we need to follow setting order

	1) set hw->rates
	2) call snd_pcm_limit_hw_rates()
	3) update hw->rate_min/max

To avoid random settings, this patch adds new soc_pcm_hw_update_rate()
and share updating code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/87sg6c90qv.wl-kuninori.morimoto.gx@renesas.comSigned-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.comSigned-off-by: Mark Brown <broonie@kernel.org>

With recent SOF 1.7 pre-releases, kernel has been emitting following
warnings at probe:

[10006.645216] sof-audio-pci 0000:00:1f.3: warn: FW ABI is more recent than kernel
[10006.652137] sof-audio-pci 0000:00:1f.3: warn: topology ABI is more recent than kernel

The warnings are emitted due to increase of the patch-level in firmware
mainline (to 3.17.1). But the patch level should not be considered even
in the strict ABI check, so modify the kernel side logic that makes the
check and only consider the major.minor components.

BugLink: https://github.com/thesofproject/linux/issues/2647Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20210211172440.2371447-1-kai.vehmanen@linux.intel.comSigned-off-by: Mark Brown <broonie@kernel.org>

It is recommended to not specifiy clocks property in an endpoint subnode.
This series moves clocks to device node.

However after moving the clocks to device node, the audio playback or
capture fails. The specified clock is not actually getting enabled and
hence the failure is seen. There seems to be a bug in simple-card-utils.c
where clock handle is not assigned when parsing clocks from device node.

Fix the same and revert original change which actually added clocks
property in endpoint subnode. Also update Jetson AGX Xavier DT where the
usage is found.

Sameer Pujar (3):
  ASoC: simple-card-utils: Fix device module clock
  Revert "ASoC: audio-graph-card: Add clocks property to endpoint node"
  arm64: tegra: Move clocks from RT5658 endpoint to device node

 .../devicetree/bindings/sound/audio-graph-port.yaml         |  3 ---
 arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts          |  2 +-
 sound/soc/generic/simple-card-utils.c                       | 13 ++++++-------
 3 files changed, 7 insertions(+), 11 deletions(-)

--
2.7.4

This patch adds dapm widgets and routes on this codec
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210211122735.5691-8-srinivas.kandagatla@linaro.orgSigned-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.orgSigned-off-by: Mark Brown <broonie@kernel.org>

This binding is for LPASS has internal codec TX macro which is
for connecting with Soundwire TX codecs like WCD938x.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20210211122735.5691-6-srinivas.kandagatla@linaro.orgSigned-off-by: Mark Brown <broonie@kernel.org>