| Age | Commit message (Collapse) | Author | Files | Lines |
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Remove the bclk int from the private structure and pass it into
the set_bclk function.
Signed-off-by: Paul Handrigan <[email protected]>
Link: https://lore.kernel.org/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Calculate the proper bclk rate using the number of tdm slots
and the width of the tdm data.
Fixes: 2884c29152c0 ("ASoC: cs530x: Support for cs530x ADCs")
Signed-off-by: Paul Handrigan <[email protected]>
Link: https://lore.kernel.org/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Convert the Cirrus Logic CS4270 audio CODEC bindings to DT schema. Add
missing va-supply, vd-supply and vlc-supply properties, because they
are already being used in the DTS and the driver for this device.
Cc: Daniel Baluta <[email protected]>
Signed-off-by: Animesh Agarwal <[email protected]>
Reviewed-by: Krzysztof Kozlowski <[email protected]>
Link: https://lore.kernel.org/r/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Allocate memory, which is being freed at end of the scope, with
scoped/cleanup.h to reduce number of error paths and make code a bit
simpler.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Allocate memory, which is being freed at end of the scope, with
scoped/cleanup.h to reduce number of error paths and make code a bit
simpler.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Replace a comma between expression statements by a semicolon.
Signed-off-by: Chen Ni <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Rename the tas2781_reset as tasdevice_reset in case of misunderstanding.
RESET register for both tas2563 and tas2781 is same and the use of reset
pin is also same.
Signed-off-by: Shenghao Ding <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Merge series from Elinor Montmasson <[email protected]>:
This is a series of patches aiming to make the machine driver
`fsl-asoc-card` compatible with S/PDIF controllers on imx boards. The
main goal is to allow the use of S/PDIF controllers with ASRC modules.
The `imx-spdif` machine driver already has specific support for S/PDIF
controllers but doesn't support using an ASRC with it. However, the
`fsl-asoc-card` machine driver has the necessary code to create a sound
card which can use an ASRC module.
It is then possible to extend the support for S/PDIF audio cards by
merging the `imx-spdif` driver into `fsl-asoc-card`.
The first three patches adapt the `fsl-asoc-card` driver to support
multiple codec use cases.
The driver can get 2 codec phandles from the device tree, and
codec-related variables are doubled.
`for_each_codecs` macros are also used when possible to ease adding
other multi-codec use cases in the future.
It makes possible to use the two S/PDIF dummy codec drivers
`spdif_receiver` and `spdif_transmitter` instead of `snd-soc-dummy`,
which was used in `imx-spdif`.
The fourth patch merges the S/PDIF support from `imx-spdif` to
`fsl-asoc-card`.
`fsl-asoc-card` offers the same functionalities as `imx-spdif` did, but
this merge also extends the S/PDIF support with the possibility of using
an ASRC.
Compatible "fsl,imx-audio-spdif" is kept, but `fsl-asoc-card` uses
different DT properties compared to `imx-spdif`:
* The "spdif-controller" property from `imx-spdif` is named "audio-cpu"
in `fsl-asoc-card`.
* `fsl-asoc-card` uses codecs explicitly declared in DT with
"audio-codec". With an S/PDIF, codec drivers `spdif_transmitter` and
`spdif_receiver` should be used. Driver `imx-spdif` used instead the
dummy codec and a pair of boolean properties, "spdif-in" and
"spdif-out".
Backward compatibility is therefore implemented in `fsl-asoc-card`.
However, it is recommended to use the new properties when needed.
Especially, declaring and using S/PDIF transmitter and/or receiver nodes
is better than using the dummy codec.
The last three patches update the device tree bindings of
`fsl-asoc-card` and update all in-tree device trees to use the
`fsl-asoc-card` properties.
Note that as the old properties are still supported:
* previous versions of in-tree device trees are still supported.
* out-of-tree device trees are still supported.
This series of patches was successfully built for arm64 and x86 on top
of the latest "for-next" branch of the ASoC git tree on the 26th of June
2024.
These modifications have also been tested on an i.MX8MN evaluation board
with a linux kernel RT v6.1.26-rt8.
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Merge series from Krzysztof Kozlowski <[email protected]>:
Allocate the memory with scoped/cleanup.h to reduce error handling
(simpler error paths) and make the code a bit smaller.
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The S/PDIF audio card support with compatible "fsl,imx-audio-spdif"
was merged from imx-spdif into the fsl-asoc-card driver.
It makes possible to use an S/PDIF with an ASRC.
This merge introduces new DT bindings to use with compatible
"fsl,imx-audio-spdif" to follow the way fsl-asoc-card works:
* the "spdif-controller" property from imx-spdif is named "audio-cpu"
in fsl-asoc-card.
* fsl-asoc-card uses codecs explicitly declared in DT
with "audio-codec".
With an SPDIF, codec drivers spdif_transmitter and
spdif_receiver should be used.
Driver imx-spdif used instead the dummy codec and a pair of
boolean properties, "spdif-in" and "spdif-out".
In an upcoming commit, in-tree DTs will be modified to follow these new
properties:
* Property "spdif-controller" will be renamed "audio-cpu".
* spdif_transmitter and spdif_receiver nodes will be declared
and linked to the fsl-asoc-card node with the property "audio-codec".
To keep backward compatibility with other DTs, support for
"spdif-controller", "spdif-in" and "spdif-out" properties is kept.
However, it is recommended to use the new properties if possible.
It is better to declare transmitter and/or receiver
in DT than using the dummy codec.
DTs using compatible "fsl,imx-audio-spdif" are still supported, and
fsl-asoc-card will behave the same as imx-spdif for these DTs.
Signed-off-by: Elinor Montmasson <[email protected]>
Reviewed-by: Rob Herring (Arm) <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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The imx-spdif machine driver creates audio card to directly use an
S/PDIF device. However, it doesn't support interacting with an ASRC.
fsl-asoc-card already has the support to create audio card which can
use the ASRC.
Merge the S/PDIF support from imx-spdif into driver fsl-asoc-card
to extend the support of S/PDIF audio card with the use of ASRC devices.
fsl-asoc-card uses slightly different DT properties than imx-spdif:
* the "spdif-controller" property from imx-spdif is named "audio-cpu" in
fsl-asoc-card.
* fsl-asoc-card uses codecs explicitly declared in DT
with "audio-codec".
With an SPDIF, codec drivers spdif_transmitter and
spdif_receiver should be used.
Driver imx-spdif used instead the dummy codec and a pair of
boolean properties, "spdif-in" and "spdif-out".
To keep backward compatibility, support for "spdif-controller",
"spdif-in" and "spdif-out" is also added to fsl-asoc-card.
However, it is recommended to use the new properties if possible.
It is better to declare transmitter and/or receiver in DT
than using the dummy codec.
DTs using compatible "fsl,imx-audio-spdif" are still compatible, and
fsl-asoc-card will behave the same as imx-spdif
for these DTs.
Signed-off-by: Elinor Montmasson <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Adapt the driver to work with configurations using two codecs or more.
Modify fsl_asoc_card_probe() to handle use cases where 2 codecs are
given in the device tree.
This will be needed to add support for the SPDIF.
Use cases using one codec will ignore any given codecs other than the
first.
Co-developed-by: Philip-Dylan Gleonec <[email protected]>
Signed-off-by: Philip-Dylan Gleonec <[email protected]>
Signed-off-by: Elinor Montmasson <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add a second dai link component for codecs that will be used for use
cases with 2 codecs.
It is needed for future integration of the SPDIF support, which will
use spdif_receiver and spdif_transmitter drivers.
To prevent deferring in use cases using only one codec, also set
by default the number of codecs to 1 for the relevant dai links.
Co-developed-by: Philip-Dylan Gleonec <[email protected]>
Signed-off-by: Philip-Dylan Gleonec <[email protected]>
Signed-off-by: Elinor Montmasson <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add support for dai links using multiple codecs for multi-codec
use cases.
Co-developed-by: Philip-Dylan Gleonec <[email protected]>
Signed-off-by: Philip-Dylan Gleonec <[email protected]>
Signed-off-by: Elinor Montmasson <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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broken alignment
Use get_unaligned_be16 instead of be16_to_cpup and get_unaligned_be32
instead of be32_to_cpup for potentially broken alignment.
Signed-off-by: Shenghao Ding <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Allocate the memory with scoped/cleanup.h in
snd_soc_dai_link_event_pre_pmu() to reduce error handling (less error
paths) and make the code a bit simpler.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Allocate the memory with scoped/cleanup.h in dapm_cnew_widget() to
reduce error handling (less error paths) and make the code a bit
simpler.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Allocate the memory with scoped/cleanup.h to reduce error handling (less
error paths) and make the code a bit simpler.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Allocate the memory with scoped/cleanup.h, instead of devm interface, to
make the code more obvious that memory is not used outside this scope.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Acked-by: Kuninori Morimoto <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Allocate the memory with scoped/cleanup.h, instead of devm interface, to
make the code more obvious that memory is not used outside this scope.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Acked-by: Kuninori Morimoto <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Allocate the memory with scoped/cleanup.h, instead of devm interface, to
make the code more obvious that memory is not used outside this scope.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Acked-by: Kuninori Morimoto <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Allocate the memory with scoped/cleanup.h to reduce error handling (less
error paths) and make the code a bit simpler.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Acked-by: Kuninori Morimoto <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Allocate the memory with scoped/cleanup.h to reduce error handling (less
error paths) and make the code a bit simpler.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Allocate the memory with scoped/cleanup.h to reduce error handling (less
error paths) and make the code a bit simpler.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Allocate the memory with scoped/cleanup.h in audio_iio_aux_probe() to
reduce error handling (less error paths) and make the code a bit
simpler.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Allocate the memory with scoped/cleanup.h in audio_iio_aux_add_dapms()
to reduce error handling (less error paths) and make the code a bit
simpler.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Adding name-prefix for each audio controls is a redundant, because
name-prefix will be automatically added behind the control name when
creating a new control.
Signed-off-by: Shenghao Ding <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Merge series from Herve Codina <[email protected]>:
The qmc_audio driver supports only audio in interleaved mode.
Non-interleaved mode can be easily supported using several QMC channel
per DAI. In that case, data related to ch0 are sent to (received from)
the first QMC channel, data related to ch1 use the next QMC channel and
so on up to the last channel.
In terms of constraints and settings, the interleaved and
non-interleaved modes are slightly different.
In interleaved mode:
- The sample size should fit in the number of time-slots available for
the QMC channel.
- The number of audio channels should fit in the number of time-slots
(taking into account the sample size) available for the QMC channel.
In non-interleaved mode:
- The number of audio channels is the number of available QMC
channels.
- Each QMC channel should have the same number of time-slots.
- The sample size equals the number of time-slots of one QMC channel.
This series add support for the non-interleaved mode in the qmc_audio
driver and is composed of the following parts:
- Patches 1 and 2: Fix some issues in the qmc_audio
- Patches 3 to 6: Prepare qmc_audio for the non-interleaved mode
- Patches 7 and 8: Extend the QMC driver API
- Patches 9 and 10: The support for non-interleaved mode itself
Compared to the previous iteration, this v2 series mainly improves
qmc_audio_access_is_interleaved().
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Merge series from Krzysztof Kozlowski <[email protected]>:
Improve a bit the Qualcomm LPASS RX macro driver and align similar parts
of code with LPASS WSA macro driver for consistency.
No external dependencies.
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both tas27871 and tas2563
Set "Speaker Force Firmware Load" as the common kcontrol
for both tas27871 and tas2563 and move it into newly-created
tasdevice_snd_controls, and keep the digital gain and analog
gain in tas2781_snd_controls.
Signed-off-by: Shenghao Ding <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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acpi_get_first_physical_node() can return NULL in several cases (no such
device, ACPI table error, reference count drop to 0, etc).
Existing check just emit error message, but doesn't perform return.
Then this NULL pointer is passed to devm_acpi_dev_add_driver_gpios()
where it is dereferenced.
Adjust this error handling by adding error code return.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Fixes: 02527c3f2300 ("ASoC: amd: add Machine driver for Jadeite platform")
Signed-off-by: Aleksandr Mishin <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Driver does not unregister typec structures (typec_mux_dev and
typec_switch_desc) during removal leading to leaks. Fix this by moving
typec registering parts to separate function and using devm interface to
release them. This also makes code a bit simpler:
- Smaller probe() function with less error paths and no #ifdefs,
- No need to store typec_mux_dev and typec_switch_desc in driver state
container structure.
Cc: [email protected]
Fixes: 10f514bd172a ("ASoC: codecs: Add WCD939x Codec driver")
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Reviewed-by: Neil Armstrong <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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commit c721f189e89c0 ("reset: Instantiate reset GPIO controller for
shared reset-gpios") check if there is no "resets" property
will fallback to "reset-gpios".
So don't need to handle "reset-gpios" separately in the driver,
the "reset-gpios" handler is duplicated with "resets" control handler,
remove it.
Signed-off-by: Shengjiu Wang <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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There is no need to keep separate token list for dai and 'common' copier
token list when the 'common' list is actually the aif list, the
SOF_COPIER_DEEP_BUFFER_TOKENS are not applicable for buffers.
We could have separate lists for all types but it is probably simpler to
just use a single list for all types of copiers. Function specific tokens
will be only parsed by function specific code anyways.
Reviewed-by: Bard Liao <[email protected]>
Reviewed-by: Ranjani Sridharan <[email protected]>
Signed-off-by: Peter Ujfalusi <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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The current fsl_qmc_audio works in interleaved mode. The audio samples
are interleaved and all data are sent to (received from) one QMC
channel.
Using several QMC channels, non interleaved mode can be easily
supported. In that case, data related to ch0 are sent to (received from)
the first QMC channel, data related to ch1 use the next QMC channel and
so on up to the last channel.
In terms of constraints and settings, the two modes are slightly
different:
- Interleaved mode:
- The sample size should fit in the number of time-slots available
for the QMC channel.
- The number of audio channels should fit in the number of
time-slots (taking into account the sample size) available for the
QMC channel.
- Non-interleaved mode:
- The number of audio channels is the number of available QMC
channels.
- Each QMC channel should have the same number of time-slots.
- The sample size equals the number of time-slots of one QMC
channel.
Add support for the non-interleaved mode allowing multiple QMC channel
per DAI. The DAI switches in non-interleaved mode when more that one QMC
channel is available.
Signed-off-by: Herve Codina <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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The QMC audio uses one QMC channel per DAI and uses this QMC channel to
transmit interleaved audio channel samples.
In order to work in non-interleave mode, a QMC audio DAI needs to use
multiple QMC channels. In that case, the DAI maps each QMC channel to
exactly one audio channel.
Allow QMC audio DAIs with multiple QMC channels attached.
Signed-off-by: Herve Codina <[email protected]>
Reviewed-by: Rob Herring (Arm) <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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No function in the QMC API is available to get the number of phandles
present in a phandle list.
Fill this lack introducing qmc_chan_count_phandles().
Signed-off-by: Herve Codina <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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qmc_chan_get_byphandle() and the resource managed version retrieve a
channel from a simple phandle.
Extend the API and introduce qmc_chan_get_byphandles_index() and the
resource managed version in order to retrieve a channel from a phandle
list using the provided index to identify the phandle in the list.
Also update qmc_chan_get_byphandle() and the resource managed version to
use qmc_chan_get_byphandles_index() and so avoid code duplication.
Signed-off-by: Herve Codina <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Constraints are set by qmc_dai_startup(). These constraints are specific
to the interleaved mode.
With the future introduction of support for non-interleaved mode, a new
set of constraints will be set. To make the code clear and keep
qmc_dai_startup() simple, extract the current interleaved mode
constraints settings to a specific function.
Signed-off-by: Herve Codina <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Submitting data to QMC channels is done in several places: transfer
completions and DAI start. The operation done is simple and consist in
one function call.
With the future introduction of support for non-interleaved mode,
submitting data will be more complex.
To avoid copy/paste of code in several places, introduce
qmc_audio_pcm_{read,write}_submit() whose goal is to handle this
data submission.
Signed-off-by: Herve Codina <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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routines
The current QMC audio driver uses only one QMC channel per DAI. The
context used by QMC channel transfer (read and write) completion
routines does not contains any QMC channel and the only one available
per DAI is used to schedule the next transfer.
This works pretty well with only one QMC channel per DAI.
The future support for non-inlerleave mode will use several QMC channel
per DAI. In that case, QMC channel transfer completion routines need to
identify the QMC channel related to the completion.
In order to fill this lack, even if identifying the current QMC channel
among several QMC channels is not needed for the current code, add one
indirection level and introduce the qmc_dai_chan data structrure.
This structure contains the QMC channel involved in the completion and
refererences to the runtime context (capture and playback) used by the
DAI.
Signed-off-by: Herve Codina <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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The driver mixes some internal values for channel DMA buffer handling
and PCM pointer handling. In the currently supported interleaved mode,
this mix does not lead to any issues but in order to prepare the
support for the non-interleaved mode, having them clearly separated will
ease the support and avoid additional computation to convert values used
in channel DMA buffer management in values usable for PCM pointer.
Use a specific set of variable for PCM pointer handling and an other set
for channel DMA buffer.
Signed-off-by: Herve Codina <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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./scripts/checkpatch.pl --strict --codespell detected several issues
when running on the fsl_qmc_audio.c file:
- CHECK: spaces preferred around that '*' (ctx:VxV)
- CHECK: Alignment should match open parenthesis
- CHECK: Comparison to NULL could be written "!prtd"
- CHECK: spaces preferred around that '/' (ctx:VxV)
- CHECK: Lines should not end with a '('
- CHECK: Please don't use multiple blank lines
Some of them are present several times.
Fix all of these issues without any functional changes.
Signed-off-by: Herve Codina <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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devm_kasprintf() can return a NULL pointer on failure but this returned
value is not checked.
Fix this lack and check the returned value.
Fixes: 075c7125b11c ("ASoC: fsl: Add support for QMC audio")
Cc: [email protected]
Signed-off-by: Herve Codina <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Driver's probe() has two allocations which are needed only within the
probe() itself - for devm_regmap_init_mmio().
Usage of devm interface is a bit misleading here, because these can be
freed right after each scope finishes.
This makes the code a bit more obvious and self documenting.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-6-6d98d4dd1ef5@linaro.org
Signed-off-by: Mark Brown <[email protected]>
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Driver uses ARRAY_SIZE() to get number of widgets later passed to
snd_soc_dapm_new_controls(), which is an 'unsigned int'.
Reviewed-by: Dmitry Baryshkov <[email protected]>
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-5-6d98d4dd1ef5@linaro.org
Signed-off-by: Mark Brown <[email protected]>
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Number of widgets in array passed to snd_soc_dapm_new_controls() cannot
be negative, so make it explicit by using 'unsigned int', just like
snd_soc_add_component_controls() is doing.
Reviewed-by: Dmitry Baryshkov <[email protected]>
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-4-6d98d4dd1ef5@linaro.org
Signed-off-by: Mark Brown <[email protected]>
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The driver has static 'struct regmap_config', which is then customized
depending on device version. This works fine, because there should not
be two devices in a system simultaneously and even less likely that such
two devices would have different versions, thus different regmap config.
However code is cleaner and more obvious when static data in the driver
is also const - it serves as a template.
Mark the 'struct regmap_config' as const and duplicate it in the probe()
with kmemdup to allow customizing per detected device variant.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-3-6d98d4dd1ef5@linaro.org
Signed-off-by: Mark Brown <[email protected]>
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Allocate the default register values array with scoped/cleanup.h to
reduce number of error paths and make code a bit simpler.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-2-6d98d4dd1ef5@linaro.org
Signed-off-by: Mark Brown <[email protected]>
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Eliminate PDS cleanup by using devm_add_action_or_reset() which results
in one less error path and smaller cleanup in remove().
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-1-6d98d4dd1ef5@linaro.org
Signed-off-by: Mark Brown <[email protected]>
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