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 <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240703-asoc-cleanup-h-v1-4-71219dfd0aef@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240703-asoc-cleanup-h-v1-3-71219dfd0aef@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240703-asoc-cleanup-h-v1-2-71219dfd0aef@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240703-asoc-cleanup-h-v1-1-71219dfd0aef@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://patch.msgid.link/20240705160344.6481-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20240705141012.5368-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 <shenghao-ding@ti.com>
Link: https://patch.msgid.link/20240705064846.1723-1-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Merge v6.10-rc6 into drm-next
The exynos-next pull is based on a newer -rc than drm-next. hence
backmerge first to make sure the unrelated conflicts we accumulated
don't end up randomly in the exynos merge pull, but are separated out.
Conflicts are all benign: Adjacent changes in amdgpu and fbdev-dma
code, and cherry-pick conflict in xe.
Signed-off-by: Daniel Vetter <daniel.vetter@ffwll.ch>
Merge series from Herve Codina <herve.codina@bootlin.com>:
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().
Merge series from Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>:
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.
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 <shenghao-ding@ti.com>
Link: https://patch.msgid.link/20240704094939.1824-1-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://patch.msgid.link/20240703140802.27688-1-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 <amishin@t-argos.ru>
Link: https://patch.msgid.link/20240703191007.8524-1-amishin@t-argos.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
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: stable@vger.kernel.org
Fixes: 10f514bd172a ("ASoC: codecs: Add WCD939x Codec driver")
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Neil Armstrong <neil.armstrong@linaro.org>
Link: https://patch.msgid.link/20240701122616.414158-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1720009575-11677-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://patch.msgid.link/20240704085944.371450-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-11-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-7-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-6-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-5-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-4-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
./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 <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-3-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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: stable@vger.kernel.org
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-2-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The HDaudio specification Section 3.6.2 limits the number of BDL entries to 256.
Make sure we don't allow more periods than this normative value.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://patch.msgid.link/20240704090106.371497-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Those registers will be used when JD source is RT711_JD2_1P8V_1PORT.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Shuming Fan <shumingf@realtek.com>
Link: https://patch.msgid.link/20240704092327.652609-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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
hda_dsp_set_hw_params_upon_resume()
-> hda_dsp_dais_suspend()
-> hda_dai_suspend()
-> hda_ipc4_post_trigger()
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.
Fixes: 2b009fa0823c ("ASoC: SOF: Intel: hda: Unify DAI drv ops for IPC3 and IPC4")
Link: https://github.com/thesofproject/linux/issues/5080
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://patch.msgid.link/20240704085708.371414-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-6-6d98d4dd1ef5@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <dmitry.baryshkov@linaro.org>
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-5-6d98d4dd1ef5@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <dmitry.baryshkov@linaro.org>
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-4-6d98d4dd1ef5@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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 <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-3-6d98d4dd1ef5@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Switch the driver to use GPIO descriptors.
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.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://patch.msgid.link/20240701-asoc-tas-gpios-v1-1-d69ec5d79939@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
These two commits set the upper limit of the Speaker Volume control
to +12dB instead of +100dB.
This should have been a simple 1-line change to the #define in the
header file, but only the HDA cs35l56 driver is using this define.
The ASoC cs35l56 driver was using hardcoded numbers instead of the
header defines.
So the first commit changes the ASoC driver to use the #defined
constants. The second commit corrects the value of the constant.
'devmodel' hasn't actually been used since:
'commit 3275158fa52a ("parport: remove use of devmodel")'
and everyone now has it set to true and has been fixed up; remove
the flag.
(There are still comments all over about it)
Signed-off-by: Dr. David Alan Gilbert <linux@treblig.org>
Acked-by: Sudip Mukherjee <sudipm.mukherjee@gmail.com>
Link: https://lore.kernel.org/r/20240502154823.67235-4-linux@treblig.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
In the current flow all interrupts are disabled in runtime suspend
phase. However interrupts enablement only exists in fsl_xcvr_prepare().
After resume fsl_xcvr_prepare() may not be called so it will cause all
interrupts still disabled even if resume from suspend. Interrupts
should be explictily enabled after resume.
Also, DPATH reset setting only exists in fsl_xcvr_prepare(). After
resume from suspend DPATH should be reset otherwise there'll be channel
swap issue.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20240628094354.780720-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the match() callback, the struct device_driver * should not be
changed, so change the function callback to be a const *. This is one
step of many towards making the driver core safe to have struct
device_driver in read-only memory.
Because the match() callback is in all busses, all busses are modified
to handle this properly. This does entail switching some container_of()
calls to container_of_const() to properly handle the constant *.
For some busses, like PCI and USB and HV, the const * is cast away in
the match callback as those busses do want to modify those structures at
this point in time (they have a local lock in the driver structure.)
That will have to be changed in the future if they wish to have their
struct device * in read-only-memory.
Cc: Rafael J. Wysocki <rafael@kernel.org>
Reviewed-by: Alex Elder <elder@kernel.org>
Acked-by: Sumit Garg <sumit.garg@linaro.org>
Link: https://lore.kernel.org/r/2024070136-wrongdoer-busily-01e8@gregkh
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The "Speaker Volume" control was being defined using four hardcoded magic
numbers. There are #defines in the cs35l56.h header for these numbers, so
change the code to use the defined constants.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20240703095517.208077-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
