Nick Weihs [Thu, 25 Jul 2024 05:47:22 +0000 (22:47 -0700)]
ALSA: hda/realtek: Implement sound init sequence for Samsung Galaxy Book3 Pro 360
Samsung Galaxy Book3 Pro 360 sends a large amount of data to the codec
through hda processing coefficients. This data was captured using a
modified version of QEMU, but the actual content of the data remains
opaque to me. Elliding any part of the data seems to cause sound to
not work.
When a Function Block declares it being a legacy MIDI1 device, it has
to be only with a single UMP Group. Correct the attribute when a
device declares it wrongly.
When a device tries to update the FB name string even if its Endpoint
is declared as static, we should skip it, just already done for the FB
info update reply.
ALSA: usb-audio: Add a quirk for Sonix HD USB Camera
Sonix HD USB Camera does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x84".
This patch adds the USB ID to quirks.c and avoids those error messages.
(snip)
[1.789698] usb 3-3: new high-speed USB device number 2 using xhci_hcd
[1.984121] usb 3-3: New USB device found, idVendor=0c45, idProduct=6340, bcdDevice= 0.00
[1.984124] usb 3-3: New USB device strings: Mfr=2, Product=1, SerialNumber=0
[1.984127] usb 3-3: Product: USB 2.0 Camera
[1.984128] usb 3-3: Manufacturer: Sonix Technology Co., Ltd.
[5.440957] usb 3-3: 3:1: cannot get freq at ep 0x84
[12.130679] usb 3-3: 3:1: cannot get freq at ep 0x84
[12.175065] usb 3-3: 3:1: cannot get freq at ep 0x84
ALSA: hda: tas2781: mark const variables as __maybe_unused
An earlier patch changed the DECLARE_TLV_DB_SCALE declaration, but
now there are additional static const variables that cause
the same build warnings:
In file included from sound/pci/hda/tas2781_hda_i2c.c:23:
include/sound/tas2781-tlv.h:23:28: error: 'tas2563_dvc_table' defined but not used [-Werror=unused-const-variable=]
23 | static const unsigned char tas2563_dvc_table[][4] = {
| ^~~~~~~~~~~~~~~~~
In file included from include/sound/tlv.h:10,
from sound/pci/hda/tas2781_hda_i2c.c:22:
include/sound/tas2781-tlv.h:20:35: error: 'tas2563_dvc_tlv' defined but not used [-Werror=unused-const-variable=]
20 | static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1);
| ^~~~~~~~~~~~~~~
ALSA: usb-audio: Fix microphone sound on HD webcam.
I own an external usb Webcam, HD webcam, which had low mic volume and
inconsistent sound quality. Video works as expected.
(snip)
[ 95.473820][ 1] [ T73] usb 5-2.2: new high-speed USB device number 7 using xhci_hcd
[ 95.773974][ 1] [ T73] usb 5-2.2: New USB device found, idVendor=1bcf, idProduct=2281, bcdDevice= 0.05
[ 95.783445][ 1] [ T73] usb 5-2.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3
[ 95.791872][ 1] [ T73] usb 5-2.2: Product: HD webcam
[ 95.797001][ 1] [ T73] usb 5-2.2: Manufacturer: Sunplus IT Co
[ 95.802996][ 1] [ T73] usb 5-2.2: SerialNumber: 20200513
[ 96.092610][ 2] [ T3680] usb 5-2.2: Warning! Unlikely big volume range (=4096), cval->res is probably wrong.
[ 96.102436][ 2] [ T3680] usb 5-2.2: [5] FU [Mic Capture Volume] ch = 1, val = 0/4096/1
Set up quirk cval->res to 16 for 256 levels,
Set GET_SAMPLE_RATE quirk flag to stop trying to get the sample rate.
Confirmed that happened anyway later due to the backoff mechanism,
After 3 failures.
All audio stream on device interfaces share the same values,
apart from wMaxPacketSize and tSamFreq :
Mark Brown [Tue, 16 Jul 2024 14:47:59 +0000 (15:47 +0100)]
kselftest/alsa: Use card name rather than number in test names
Currently for the PCM and mixer tests we report test names which identify
the card being tested with the card number. This ensures we have unique
names but since card numbers are dynamically assigned at runtime the names
we end up with will often not be stable on systems with multiple cards
especially where those cards are provided by separate modules loeaded at
runtime. This makes it difficult for automated systems and UIs to relate
test results between runs on affected platforms.
Address this by replacing our use of card numbers with card names which are
more likely to be stable across runs. We use the card ID since it is
guaranteed to be unique by default, unlike the long name. There is still
some vulnerability to ordering issues if multiple cards with the same base
ID are present in the system but have separate dependencies but not all
drivers put distinguishing information in their long names.
Seunghun Han [Thu, 18 Jul 2024 08:09:08 +0000 (17:09 +0900)]
ALSA: hda/realtek: Fix the speaker output on Samsung Galaxy Book Pro 360
Samsung Galaxy Book Pro 360 (13" 2022 NT935QDB-KC71S) with codec SSID
144d:c1a4 requires the same workaround to enable the speaker amp
as other Samsung models with the ALC298 codec.
ALSA: seq: ump: Skip useless ports for static blocks
When the UMP Endpoint is configured with static blocks, the block
configuration will never change, hence the unused ports will be
unchanged as well. Creating sequencer ports for those unused ports
is simply useless, and it might be rather confusing for users.
The idea behind the inactive ports was for allowing connections
from/to ports that can become usable later, but this will never
happen for inactive groups in static blocks.
Let's change the sequencer UMP binding to skip those unused ports when
the UMP EP is with static blocks.
Shengjiu Wang [Wed, 17 Jul 2024 06:44:53 +0000 (14:44 +0800)]
ALSA: pcm_dmaengine: Don't synchronize DMA channel when DMA is paused
When suspended, the DMA channel may enter PAUSE state if dmaengine_pause()
is supported by DMA.
At this state, dmaengine_synchronize() should not be called, otherwise
the DMA channel can't be resumed successfully.
Merge tag 'asoc-v6.11' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for for v6.11
There are a lot of changes in here, though the big bulk of things is
cleanups and simplifications of various kinds which are internally
rather than externally visible. A good chunk of those are DT schema
conversions, but there's also a lot of changes in the code.
Highlights:
- Syncing of features between simple-audio-card and the two
audio-graph cards so there is no reason to stick with an older
driver.
- Support for specifying the order of operations for components within
cards to allow quirking for unusual systems.
- New support for Asahi Kasei AK4619, Cirrus Logic CS530x, Everest
Semiconductors ES8311, NXP i.MX95 and LPC32xx, Qualcomm LPASS v2.5
and WCD937x, Realtek RT1318 and RT1320 and Texas Instruments PCM5242.
Instead of the explicit "1 << x", use BIT() macro for one bit values.
This will improve the readability and also avoids the possible bad
value for 31bit shift.
ALSA: usb: Fix UBSAN warning in parse_audio_unit()
A malformed USB descriptor may pass the lengthy mixer description with
a lot of channels, and this may overflow the 32bit integer shift
size, as caught by syzbot UBSAN test. Although this won't cause any
real trouble, it's better to address.
This patch introduces a sanity check of the number of channels to bail
out the parsing when too many channels are found.
Merge tag 'asoc-fix-v6.10-rc7' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.10
A few fairly small fixes for ASoC, there's a relatively large set of
hardening changes for the cs_dsp firmware file parsing and a couple of
other small device specific fixes.
ASoC: dt-bindings: cirrus,cs4270: Convert to dtschema
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.
firmware: cs_dsp: Clarify wmfw format version log message
Change the log message of the wmfw format version to include
the file name, and change the message to say "format" instead
of "Firmware version". Merge this with the message that logs
the timestamp.
The wmfw format version is information that is useful to have
logged because the behaviour of firmware controls depends on
the wmfw format. So "unexpected" behaviour could be caused by
having expectations based on one format of wmfw when a
different format has been loaded.
But the original message was confusing. It reported the file
format version but didn't actually log the name of the file it
referred to. It also called it "Firmware version", which is
confusing when a later message also logs a firmware version
that is the version of the actual firmware within the wmfw.
The logging of the firmware timestamp has been merged into this.
That was originally a dbg-only message, but as we are already
logging a line of info, we might as well add a few extra
characters to log the timestamp. The timestamp is now logged
in hexadecimal - it's not particularly useful as a decimal
value.
firmware: cs_dsp: Don't allocate temporary buffer for info text
Don't allocate a temporary buffer to hold a NUL-terminated copy
of the NAME/INFO string from the wmfw/bin. It can be printed
directly to the log. Also limit the maximum number of characters
that will be logged from this string.
The NAME/INFO blocks in the firmware files are an array of
characters with a length, not a NUL-terminated C string. The
original code allocated a temporary buffer to make a
NUL-terminated copy of the string and then passed that to
dev_info(). There's no need for this: printf formatting can
use "%.*s" to print a character array of a given length.
ASoc: TAS2781: rename the tas2781_reset as tasdevice_reset
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.
ALSA: ppc: keywest: Drop explicit initialization of struct i2c_device_id::driver_data to 0
The driver doesn't use the driver_data member of struct i2c_device_id,
so don't explicitly initialize this member.
This prepares putting driver_data in an anonymous union which requires
either no initialization or named designators. But it's also a nice
cleanup on its own.
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.
ASoC: dt-bindings: update fsl-asoc-card bindings after imx-spdif merge
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.
ASoC: fsl-asoc-card: merge spdif support from imx-spdif.c
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.
ASoC: fsl-asoc-card: add compatibility to use 2 codecs in dai-links
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.
ASoC: fsl-asoc-card: add second dai link component for codecs
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.
firmware: cs_dsp: Use strnlen() on name fields in V1 wmfw files
Use strnlen() instead of strlen() on the algorithm and coefficient name
string arrays in V1 wmfw files.
In V1 wmfw files the name is a NUL-terminated string in a fixed-size
array. cs_dsp should protect against overrunning the array if the NUL
terminator is missing.
ASoC: dapm: Simplify snd_soc_dai_link_event_pre_pmu() with cleanup.h
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.
ALSA: seq: Add tempo base unit for MIDI2 Set Tempo messages
MIDI2 Set Tempo message defines the tempo in 10ns unit for finer
accuracy, while MIDI1 was defined in 1us unit. For adapting this
different unit, introduce "tempo_base" field to snd_seq_queue_tempo
struct so that user-space can pass the proper tempo base unit.
The accepted value is limited, it must be either 0, 10 or 1000.
The protocol version is bumped to 1.0.4 along with this.
The access with the older protocol version ignores the tempo-base
value in ioctls and always treats as 1000.
The internal mic boost on the VAIO models VJFE-CL and VJFE-IL is too high.
Fix this by applying the ALC269_FIXUP_LIMIT_INT_MIC_BOOST fixup to the machine
to limit the gain.
ASoc: pcm6240: Remove unnecessary name-prefix for all the controls
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.
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().
ASoc: tas2781: Set "Speaker Force Firmware Load" as the common kcontrol for 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.
Stefan Binding [Wed, 3 Jul 2024 14:07:28 +0000 (15:07 +0100)]
ALSA: hda: cs35l41: Fix missing Speaker ID GPIO description in _DSD
Laptop 10431A63 contains valid _DSD, but missing Speaker ID
description. Add this discription, but keep the rest of the _DSD to
ensure the correct firmware and tuning is loaded for this laptop.
ASoC: amd: Adjust error handling in case of absent codec device
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.
ASoC: codecs: wcd939x: Fix typec mux and switch leak during device removal
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.
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.
Peter Ujfalusi [Thu, 4 Jul 2024 08:59:44 +0000 (10:59 +0200)]
ASoC: SOF: ipc4-topology: Use single token list for the copiers
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.
ASoC: fsl: fsl_qmc_audio: Add support for non-interleaved mode.
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.
dt-bindings: sound: fsl,qmc-audio: Add support for multiple QMC channels per DAI
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.
soc: fsl: cpm1: qmc: Introduce functions to get a channel from a phandle list
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.
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.
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.
ASoC: fsl: fsl_qmc_audio: Identify the QMC channel involved in completion 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.
ASoC: fsl: fsl_qmc_audio: Split channel buffer and PCM pointer handling
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.
ASoC: fsl: fsl_qmc_audio: Fix issues detected by checkpatch
./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.
Kai Vehmanen [Thu, 4 Jul 2024 08:57:08 +0000 (10:57 +0200)]
ASoC: SOF: Intel: hda: fix null deref on system suspend entry
When system enters suspend with an active stream, SOF core
calls hw_params_upon_resume(). On Intel platforms with HDA DMA used
to manage the link DMA, this leads to call chain of
A bug is hit in hda_dai_suspend() as hda_link_dma_cleanup() is run first,
which clears hext_stream->link_substream, and then hda_ipc4_post_trigger()
is called with a NULL snd_pcm_substream pointer.
ASoC: dapm: Use unsigned for number of widgets in snd_soc_dapm_new_controls()
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.
ASoC: codecs: lpass-rx-macro: Keep static regmap_config as const
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.
Notice that we let the gpiolib handle line inversion for the
active low reset line (nreset !reset).
There are no upstream device trees using the tas5086 compatible
string, if there were, we would need to ascertain that they all
set the GPIO_ACTIVE_LOW flag on their GPIO lines.
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