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: 02527c3f23 ("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: 10f514bd17 ("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 c721f189e8 ("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>
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: 2b009fa082 ("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 3275158fa5 ("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>
Replace the open-coded parsing of "reg" with of_property_read_reg().
The #ifdef is also easily replaced with IS_ENABLED().
Signed-off-by: Rob Herring (Arm) <robh@kernel.org>
Link: https://patch.msgid.link/20240702215402.839673-1-robh@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace the open-coded parsing of "reg" with of_property_read_reg().
The #ifdef is also easily replaced with IS_ENABLED().
Signed-off-by: Rob Herring (Arm) <robh@kernel.org>
Link: https://patch.msgid.link/20240702215349.839350-1-robh@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>:
Code used to create standalone and widget controls is mostly same, with
with exception that in standalone case dynamic object needs to be
registered and control created directly.
Following patches clean up and unify kcontrol creation code in topology
code.
Code used to create standalone and widget enum control is same, with
exception that in standalone case dynamic object needs to be registered
and control created directly.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://patch.msgid.link/20240627101850.2191513-14-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Code used to create standalone and widget mixer control is same, with
exception that in standalone case dynamic object needs to be registered
and control created directly.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://patch.msgid.link/20240627101850.2191513-13-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Code used to create standalone and widget bytes control is same, with
exception that in standalone case dynamic object needs to be registered
and control created directly.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://patch.msgid.link/20240627101850.2191513-12-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_tplg_dbytes_create() missed se->dobj.index initialization, so add it
there. Additionally separate dynamic object initialization into separate
logical block code.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://patch.msgid.link/20240627101850.2191513-9-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Widget kcontrols do not need to be managed as topology dynamic objects
with an index and a linked list. As they are always associated with a
widget which is already a topology dynamic object, thus all
addition/removals of a widget will by design manage the kcontrol.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://patch.msgid.link/20240627101850.2191513-3-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of using goto, when there is no controls, just do a loop when
there are. Overall the check seems to be a bit redundant as
num_kcontrols will only be above 0 if kcontrols are set anyway, but
let's keep it, while simplifying code.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://patch.msgid.link/20240627101850.2191513-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
simple-audio-mux is designed to be used generally, thus "Input 1" or
"Input 2" are used to selecting MUX input. This numbered inputs would work,
but might be not user friendly in some case, for example in case of system
hardware design has some clear labels.
Adds new "state-labels" property and enable to select MUX by own state names.
Original
> amixer set "MUX" "Input 1"
> amixer set "MUX" "Input 2"
Use mux-names
sound_mux: mux {
compatible = "simple-audio-mux";
mux-gpios = <...>;
=> state-labels = "Label_A", "Label_B";
};
> amixer set "MUX" "Label_A"
> amixer set "MUX" "Label_B"
Merge series from srinivas.kandagatla@linaro.org:
Existing way of allocating soundwire master ports on Qualcommm platforms is
dynamic, and in linear order starting from 1 to MAX_PORTS.
This will work as long as soundwire device ports are 1:1 mapped
linearly. However on most Qcom SoCs like SM8550, SM8650, x1e80100, these
are NOT mapped in that order.
The result of this is that only one speaker among the pair of speakers
is always silent, With recent changes for WSA codec to support codec
versions and along with these patches we are able to get all speakers
working on these SoCs.
Optimize the memory usage in struct snd_pcm_runtime - use boolean
value for the standard sync ID scheme.
Introduce snd_pcm_set_sync_per_card function to build synchronization
IDs.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20240625172836.589380-3-perex@perex.cz
Until the commit e11f0f90a6 ("ALSA: pcm: remove SNDRV_PCM_IOCTL1_INFO
internal command"), there was a possibility to pass information
about the synchronized streams to the user space. The mentioned
commit removed blindly the appropriate code with an irrelevant comment.
The revert may be appropriate, but since this API was lost for several
years without any complains, it's time to improve it. The hardware
parameters may change the used stream clock source (e.g. USB hardware)
so move this synchronization ID to hw_params as read-only field.
It seems that pipewire can benefit from this API (disable adaptive
resampling for perfectly synchronized PCM streams) now.
Note that the contents of ID is not supposed to be used for direct
comparison with a specific byte sequence. The "empty" case is when
all bytes are zero (driver does not offer this information)
and all other cases must be only used for equal comparison among
PCM streams (including different sound cards) if they are using
identical hardware clock.
Cc: Takashi Sakamoto <takaswie@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20240625172836.589380-2-perex@perex.cz
The 'wsa->dev' is assigned closer to the end of the probe() function, so
the dev_err() must not use it - it is still NULL at this point. Instead
there is already a local 'dev' variable.
Fixes: 727de4fbc5 ("ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version")
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Link: https://patch.msgid.link/20240628095831.207942-2-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The 'rx->dev' is assigned closer to the end of the probe() function, so
the dev_err() must not use it - it is still NULL at this point. Instead
there is already a local 'dev' variable.
Fixes: dbacef0589 ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions")
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Link: https://patch.msgid.link/20240628095831.207942-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
bitfield.h is not explicitly included but it is required for FIELD_PREP
to be expanded by the preprocessor. If it is not implicitly included,
there will be a compiler error (as seen with ARCH=hexagon allmodconfig):
sound/soc/fsl/lpc3xxx-i2s.c:169:10: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
169 | tmp |= LPC3XXX_I2S_WW8 | LPC3XXX_I2S_WS_HP(LPC3XXX_I2S_WW8_HP);
| ^
sound/soc/fsl/lpc3xxx-i2s.h:42:30: note: expanded from macro 'LPC3XXX_I2S_WW8'
42 | #define LPC3XXX_I2S_WW8 FIELD_PREP(0x3, 0) /* Word width is 8bit */
| ^
sound/soc/fsl/lpc3xxx-i2s.c:205:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
205 | LPC3XXX_I2S_DMA1_TX_EN | LPC3XXX_I2S_DMA0_TX_DEPTH(4));
| ^
sound/soc/fsl/lpc3xxx-i2s.h:65:38: note: expanded from macro 'LPC3XXX_I2S_DMA0_TX_DEPTH'
65 | #define LPC3XXX_I2S_DMA0_TX_DEPTH(s) FIELD_PREP(0xF0000, s) /* Set the DMA1 TX Request level */
| ^
sound/soc/fsl/lpc3xxx-i2s.c:210:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
210 | LPC3XXX_I2S_DMA0_RX_EN | LPC3XXX_I2S_DMA1_RX_DEPTH(4));
| ^
sound/soc/fsl/lpc3xxx-i2s.h:70:38: note: expanded from macro 'LPC3XXX_I2S_DMA1_RX_DEPTH'
70 | #define LPC3XXX_I2S_DMA1_RX_DEPTH(s) FIELD_PREP(0x700, s) /* Set the DMA1 RX Request level */
| ^
Include bitfield.h explicitly, so that FIELD_PREP is always expanded,
clearing up the compiler error.
Fixes: 0959de657a ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs")
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://patch.msgid.link/20240701-lpc32xx-asoc-fix-include-for-field_prep-v1-1-0c5d7f71921b@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
clang points out that ret may be used uninitialized in
lpc32xx_i2s_probe() in an error pointer path (which becomes fatal with
CONFIG_WERROR):
sound/soc/fsl/lpc3xxx-i2s.c:326:47: error: variable 'ret' is uninitialized when used here [-Werror,-Wuninitialized]
326 | "failed to init register map: %d\n", ret);
| ^~~
sound/soc/fsl/lpc3xxx-i2s.c:310:9: note: initialize the variable 'ret' to silence this warning
310 | int ret;
| ^
| = 0
1 error generated.
One solution would be a small refactoring of the second parameter in
dev_err_probe(), PTR_ERR(i2s_info_p->regs), to be the value of ret in
the if statement. However, a nicer solution for debugging purposes,
which is the point of this statement, would be to use the '%pe'
specifier to symbolically print the error pointer value. Do so, which
eliminates the uninitialized use of ret, clearing up the warning.
Fixes: 0959de657a ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs")
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://patch.msgid.link/20240701-lpc32xx-asoc-fix-uninitialized-ret-v1-1-985d86189739@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add name_prefix as the prefix name of DSP firmwares
and calibrated data files which stored speaker
calibrated impedance.
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://patch.msgid.link/20240629101112.628-1-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support to parse static master port map information from device tree.
This is required for correct port mapping between soundwire device and
master ports.
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Neil Armstrong <neil.armstrong@linaro.org>
Tested-by: Neil Armstrong <neil.armstrong@linaro.org> # on SM8650-HDK
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Link: https://patch.msgid.link/20240626-port-map-v2-4-6cc1c5608cdd@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support to parse static master port map information from device tree.
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Tested-by: Neil Armstrong <neil.armstrong@linaro.org> # on SM8650-HDK
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Link: https://patch.msgid.link/20240626-port-map-v2-2-6cc1c5608cdd@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-audio-mux is designed to be used generally, thus "Input 1" or
"Input 2" are used to selecting MUX input. This numbered inputs would
work, but might be not user friendly in some case, for example in case
of system hardware design has some clear labels.
Adds new "state-labels" property and enable to select MUX by own state
names.
Original
> amixer set "MUX" "Input 1"
Use mux-names
sound_mux: mux {
compatible = "simple-audio-mux";
mux-gpios = <...>;
state-labels = "Label_A", "Label_B";
};
> amixer set "MUX" "Label_A"
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87msn27xpg.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add compatible string and specific soc data to support rpmsg sound card
on i.MX95 platform.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20240626071202.7149-2-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
struct sdw_slave_prop is zero-initialized by the SoundWire core so there
is no need to clear clk_stop_mode1 to false. Removing this also avoids
having an unnecessary build dependency on a struct member.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20240701104444.172556-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
No product was ever released with A1 silicon so there is no
need for the driver to include support for it.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20240701104444.172556-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch reverts a series of commits that allowed for the ASP
registers to be owned by either the driver or the firmware. Nothing
currently depends on the functionality that is being reverted, so
it is safe to remove.
The commits being reverted are (last 3 are bugfixes to the first 2):
commit 72a77d7631
("ASoC: cs35l56: Fix to ensure ASP1 registers match cache")
commit 07f7d6e7a1
("ASoC: cs35l56: Fix for initializing ASP1 mixer registers")
commit 4703b014f2
("ASoC: cs35l56: fix reversed if statement in cs35l56_dspwait_asp1tx_put()")
commit c14f09f010
("ASoC: cs35l56: Fix deadlock in ASP1 mixer register initialization")
commit dfd2ffb373
("ASoC: cs35l56: Prevent overwriting firmware ASP config")
These reverts have been squashed into a single commit because there
would be no reason to revert only some of them (which would just
reintroduce bugs).
The changes introduced by the commits were well-intentioned but
somewhat misguided. ACPI does not provide any information about how
audio hardware is linked together, so that information has to be
hardcoded into drivers. On Windows the firmware is customized to
statically setup appropriate configuration of the audio links,
and the intent of the commits was to re-use this information if the
Linux host drivers aren't taking control of the ASP. This would
avoid having to hardcode the ASP config into the machine driver on
some systems.
However, this added complexity and race conditions into the driver.
It also complicates implementation of new code.
The only case where the ASP is used but the host is not taking
ownership is when CS35L56 is used in SoundWire mode with the ASP
as a reference audio interconnect. But even in that case it's not
necessarily required even if the firmware initialized it. Typically
it is used to avoid the host SDCA drivers having to be capable of
aggregating capture paths from multiple SoundWire peripherals. But
the SOF SoundWire support is capable of doing that aggregation.
Reverting all these commits significantly simplifies the driver.
Let's just use the normal Linux mechanisms of the machine driver and
ALSA controls to set things up instead of trying to use the firmware
to do use-case setup.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20240701104444.172556-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Piotr Wojtaszczyk <piotr.wojtaszczyk@timesys.com>:
This pach set is to bring back audio to machines with a LPC32XX CPU.
The legacy LPC32XX SoC used to have audio spport in linux 2.6.27.
The support was dropped due to lack of interest from mainaeners.
Introduce support for Cirrus Logic Device CS40L50: a
haptic driver with waveform memory, integrated DSP,
and closed-loop algorithms.
The ASoC driver enables I2S streaming to the device.
Reviewed-by: David Rhodes <drhodes@opensource.cirrus.com>
Signed-off-by: James Ogletree <jogletre@opensource.cirrus.com>
Reviewed-by: Jeff LaBundy <jeff@labundy.com>
Reviewed-by: Ricardo Rivera-Matos <rriveram@opensource.cirrus.com>
Reviewed-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20240620161745.2312359-6-jogletre@opensource.cirrus.com
Signed-off-by: Lee Jones <lee@kernel.org>
clk_prepare_enable() may fail, so we should better check its return
value and propagate it in the case of error.
Fixes: 62a7fc32a6 ("ASoC: max98088: Add master clock handling")
Signed-off-by: Chen Ni <nichen@iscas.ac.cn>
Link: https://patch.msgid.link/20240628080534.843815-1-nichen@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
The dummy DAI should allow any (reasonable) rates possible.
Make the rate continuous for dummy and set range from 5512Hz to 768kHz
The change is mostly cosmetic as dummy is skipped when setting
the hwparams.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://patch.msgid.link/20240628120130.2015665-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Qualcomm LPASS WSA macro codec driver uses now parts of common
module, so it has to select SND_SOC_LPASS_MACRO_COMMON.
sound/soc/codecs/lpass-wsa-macro.o: in function `wsa_macro_probe':
sound/soc/codecs/lpass-wsa-macro.c:2767:(.text+0x1c9c): undefined reference to `lpass_macro_get_codec_version'
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202406272231.th1LtuLk-lkp@intel.com/
Fixes: 5dcf442bbb ("ASoC: codecs: lpass-wsa-macro: Prepare to accommodate new codec versions")
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240627125203.171048-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Shengjiu Wang <shengjiu.wang@nxp.com>:
The transmitter and receiver part of the SAI interface need to be
configured with different master/slave mode, especially to work
with the audiomix module.
The SAI1 TX is in master mode, but SAI1 RX is in slave mode.
So add another two DAIs for TX and RX separately in fsl_sai driver.
There will be three devices for audiomix sound card, hw:x,0 is
the playback device for one SAI, hw:x,1 is the playback device
for another SAI, hw:x,2 is the capture device for audmix
output.
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
Introduce the ability for sound cards to manually order the startup of
the various components in the card.
A relatively large batch of updates, largely due to the long interval
since I last sent fixes due to various travel and holidays. There's a
lot of driver specific fixes and quirks in here, none of them too major,
and also some fixes for recently introduced memory safety issues in the
topology code.
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Merge tag 'asoc-fix-v6.10-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.10
A relatively large batch of updates, largely due to the long interval
since I last sent fixes due to various travel and holidays. There's a
lot of driver specific fixes and quirks in here, none of them too major,
and also some fixes for recently introduced memory safety issues in the
topology code.
The version B will support the multi-lane function and integrate the DMIC function
in one SoundWire interface.
Due to some registers having different default values between version A and B,
this patch also removes the redundant default registers to avoid confusion.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://patch.msgid.link/20240620103237.2124196-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If IORESOURCE_MEM "lpass-rxtx-cdc-dma-lpm" or "lpass-va-cdc-dma-lpm"
resources is not provided in Device Tree due to any error,
platform_get_resource_byname() will return NULL which is later
dereferenced. According to sound/qcom,lpass-cpu.yaml, these resources
are provided, but DT can be broken due to any error. In such cases driver
must be able to protect itself, since the DT is external data for the
driver.
Adjust this issues by adding NULL return check.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Fixes: b138706225 ("ASoC: qcom: Add regmap config support for codec dma driver")
Signed-off-by: Aleksandr Mishin <amishin@t-argos.ru>
Link: https://patch.msgid.link/20240605104953.12072-1-amishin@t-argos.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
There will be three devices for this sound card, hw:x,0 is
the playback device for one SAI, hw:x,1 is the playback device
for another SAI, hw:x,2 is the capture device for audmix
output. then capture device and playback device can be configured
with different master/slave mode.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1718174452-17596-4-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As audmix requires playback and capture stream in different
master/slave mode, so separate playback and capture stream to
different DAI. There are three DAIs required, two DAIs for playback
one DAI for capture.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1718174452-17596-3-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The transmitter and receiver part of the SAI interface need to be
configured with different master/slave mode, especially to work
with the audiomix module.
+-------+ +-----------+
| SAI1 | --TX--> | |
| | <--RX-- | |
+-------+ | |
| AUDIOMIX |
+-------+ | |
| SAI2 | --TX--> | |
+-------+ +-----------+
The SAI1 TX is in master mode, but SAI1 RX is in slave mode.
So add another two DAIs for TX and RX separately. but only
defined fsl_sai_set_dai_fmt_tx() and fsl_sai_set_dai_fmt_rx()
ops function for current case, in the future, the other ops
function for TX and RX can be defined if required.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1718174452-17596-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some Sound Card might need special trigger ordering which is based on
CPU/Codec connection. It is already supported on ASoC, but Audio Graph
Card2 still not yet support it. Let's support it.
Cc: Maxim Kochetkov <fido_max@inbox.ru>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87msnqzoj8.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some Sound Card might need special trigger ordering which is based on
CPU/Codec connection. It is already supported on ASoC, but Audio Graph
Card still not yet support it. Let's support it.
Cc: Maxim Kochetkov <fido_max@inbox.ru>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87o786zojd.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some Sound Card might need special trigger ordering which is based on
CPU/Codec connection. It is already supported on ASoC, but Simple Audio
Card still not yet support it. Let's support it.
Cc: Maxim Kochetkov <fido_max@inbox.ru>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87plsmzojk.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some Sound Card might need special trigger ordering which is based on
CPU/Codec connection. It is already supported on ASoC, but Simple Audio
Card / Audio Graph Card still not support it. Let's support it.
Cc: Maxim Kochetkov <fido_max@inbox.ru>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87r0d2zojq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace 'int' with proper 'enum lpass_codec_version' in every place
which handles the parsed codec version (not raw register values!) to be
explicit about contents of the variable. This makes code easier to read
and compilers could check missing switch cases.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Link: https://patch.msgid.link/20240625165736.722106-2-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Qualcomm LPASS macro codec driver parses registers in order to
detect version of the codec. It recognizes codecs v2.0 - v2.8, however
we know that there are earlier versions and 'enum lpass_codec_version'
has also v1.0, v1.1 and v1.2. If by any chance we run on unrecognized
version, driver will use random value from the stack as the codec
version.
Fix it by mapping such cases to an enum of value 0:
LPASS_CODEC_VERSION_UNKNOWN.
Fixes: 378918d591 ("ASoC: codecs: lpass-macro: add helpers to get codec version")
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Link: https://patch.msgid.link/20240625165736.722106-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Modify register setting sequence of enabling inline command
to fix issue of random interrupt from push-button.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://patch.msgid.link/9a7a3a66cbcb426487ca6f558f45e922@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The conversion of SPP to MIDI2 UMP called a wrong function, and the
secondary argument wasn't taken. As a result, MSB of SPP was always
zero. Fix to call the right function.
Fixes: e9e02819a9 ("ALSA: seq: Automatic conversion of UMP events")
Link: https://patch.msgid.link/20240626145141.16648-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Starting with v2.5 of Qualcomm LPASS Codec, few registers in the WSA
macro block change. Bring proper support for this v2.5 and newer
versions, to fix second speaker playback (speaker was silent).
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Acked-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Link: https://patch.msgid.link/20240625-qcom-audio-wsa-second-speaker-v1-3-f65ffdfc368c@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver for Qualcomm LPASS WSA macro codec was developed and tested
on codec v2.1, however v2.5 has significant changes in the registers.
The driver correctly works for v2.1 codec, but has issues when running
on SoC with v2.5 codec (so starting with SM8450, even though playback
works properly on that SoC).
Prepare the driver for handling differences in register layouts of newer
version. This does not have functional impact on older codec versions,
but just:
1. Renames few soc_enums and widgets as v2.1,
2. For registers being different in v2.5, moves the defaults and regmap
configuration to new structures,
3. Adds new 'struct wsa_reg_layout' with offsets and masks for few
registers, so most of the code can stay unchaged on v2.5,
4. Chooses proper widgets, regmap config and register layout based on
version of the codec.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Acked-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Link: https://patch.msgid.link/20240625-qcom-audio-wsa-second-speaker-v1-2-f65ffdfc368c@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The Vivobook S 16X IPS needs a quirks-table entry for the internal microphone to function properly.
Signed-off-by: Vyacheslav Frantsishko <itmymaill@gmail.com>
Reviewed-by: Mario Limonciello <mario.limonciello@amd.com>
Link: https://patch.msgid.link/20240626070334.45633-1-itmymaill@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The recent fix for Lenovo IdeaPad 330-17IKB replaced the quirk entry,
and this eventually breaks the existing quirk for Lenovo Yoga Duet 7
13ITL6 equipped with the same PCI SSID 17aa:3820.
For applying a proper quirk for each model, check the codec SSID
additionally. Fortunately Yoga Duet has a different codec SSID,
0x17aa3802.
(Interestingly, 17aa:3802 has another conflict of SSID between another
Yoga model vs 14IRP8 which we had to work around similarly.)
Fixes: b1fd0d1285 ("ALSA: hda/realtek: Enable headset mic on IdeaPad 330-17IKB 81DM")
Link: https://patch.msgid.link/20240625155217.18767-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SDCA spec defines a 'selected_mode' control which can override
the 'detected_mode' reported by hardware.
This is useful for platform integration as well as in cases
where the hardware(e.g. 3.5mm jack cable) is not able to accurately detect the jack type.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Tested-by: yung-chuan.liao@linux.intel.com
Link: https://patch.msgid.link/20240625084303.2273911-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We have also v2.1 version of the codec (see 'enum lpass_codec_version'),
so handle it as well in all switch cases.
Fixes: dbacef0589 ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions")
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://patch.msgid.link/20240625160614.450506-1-krzysztof.kozlowski@linaro.org
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
These drivers don'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.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@baylibre.com>
Link: https://patch.msgid.link/20240624131728.1244053-2-u.kleine-koenig@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Zhang Yi <zhangyi@everest-semi.com>:
We propose four patches to solve headphone detection and suspend issues.
And there are several registers that should be read-only registers. So
we create es8326_writeable_register, and set these registers to false.
Both snd_dmaengine_pcm_close() and
snd_dmaengine_pcm_close_release_chan() are almost identical except for
a call of dma_release_channel() in the latter. For code
simplification, unify them internally.
Just a code refactoring, and no functional changes.
Link: https://patch.msgid.link/20240625113356.2123-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When dmaengine supports pause function, in suspend state,
dmaengine_pause() is called instead of dmaengine_terminate_async(),
In end of playback stream, the runtime->state will go to
SNDRV_PCM_STATE_DRAINING, if system suspend & resume happen
at this time, application will not resume playback stream, the
stream will be closed directly, the dmaengine_terminate_async()
will not be called before the dmaengine_synchronize(), which
violates the call sequence for dmaengine_synchronize().
This behavior also happens for capture streams, but there is no
SNDRV_PCM_STATE_DRAINING state for capture. So use
dmaengine_tx_status() to check the DMA status if the status is
DMA_PAUSED, then call dmaengine_terminate_async() to terminate
dmaengine before dmaengine_synchronize().
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1718851218-27803-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This HP Laptop uses ALC236 codec with COEF 0x07 controlling
the mute LED. Enable existing quirk for this device.
Signed-off-by: Aivaz Latypov <reichaivaz@gmail.com>
Link: https://patch.msgid.link/20240625081217.1049-1-reichaivaz@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The conversion from the legacy event to MIDI2 UMP for RPN and NRPN
missed the setup of the channel number, resulting in always the
channel 0. Fix it.
Fixes: e9e02819a9 ("ALSA: seq: Automatic conversion of UMP events")
Link: https://patch.msgid.link/20240625095200.25745-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a helper function max_98373_get_tx_mask() to get tx mask from
max98373 ACPI device properties at runtime.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://patch.msgid.link/20240624121119.91552-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Experimental tests show that JD2_100K is required, otherwise the jack
is detected always even with nothing plugged-in.
To avoid matching with other known quirks the SKU information is used.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add name_prefix as the prefix name of firmwares and
kcontrol to support corresponding TAS2563/TAS2781s.
name_prefix is not mandatory.
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://patch.msgid.link/20240621132309.564-1-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It is incorrect to request the input pin format of the destination widget
using the output pin index of the source module as the indexes are not
necessarily matching.
moduleA.out_pin1 can be connected to moduleB.in_pin0 for example.
Use the dst_queue_id to request the input format of the destination module.
This bug remained unnoticed likely because in nocodec topologies we don't
have process modules after a module copier, thus the pin/queue index is
ignored.
For the process module case, the code was likely have been tested in a
controlled way where all the pin/queue/format properties were present to
work.
Update the debug prints to have better information.
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@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>
Cc: stable@vger.kernel.org # v6.8+
Link: https://patch.msgid.link/20240624121519.91703-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There are two hardware connection methods for DMICs on the MT6358. In cases
where more than two DMICs are used, we need to time-multiplex these DMICs.
Therefore, we need to dynamically switch the modes of these DMICs based on
the actual usage scenarios.
---- DMIC1
AU_VIN0 ---
---- DMIC2
AU_VIN2 --- ----DMIC3
When we want to use DMIC1/2, configure it to one-wire mode. When we want to
use DMIC1/3, configure it to two-wire mode.
Signed-off-by: Jiaxin Yu <jiaxin.yu@mediatek.com>
Link: https://patch.msgid.link/20240613020725.27874-1-jiaxin.yu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We modified the regmap_config members to fix cach sync error.
There are several registers that should be read-only registers.
If these registers are written while synchronizing the register values,
the codec will enter an error state.So we create es8326_writeable_register,
and set these registers to false
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://patch.msgid.link/20240624030607.4307-5-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Executing regcache_sync at initialization, we can hear a gentle pop
noise. So we created es8326_init for initialization instead of
executing es8326_resume
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://patch.msgid.link/20240624030607.4307-4-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We modified the headphone detection setting to avoid an error button state
after codec resume from suspend state
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://patch.msgid.link/20240624030607.4307-2-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit e70b8dd267 ("ASoC: mediatek: mt8195: Remove afe-dai component
and rework codec link") removed the codec entry for the ETDM1_OUT_BE
dai link entirely instead of replacing it with COMP_EMPTY(). This worked
by accident as the remaining COMP_EMPTY() platform entry became the codec
entry, and the platform entry became completely empty, effectively the
same as COMP_DUMMY() since snd_soc_fill_dummy_dai() doesn't do anything
for platform entries.
This causes a KASAN out-of-bounds warning in mtk_soundcard_common_probe()
in sound/soc/mediatek/common/mtk-soundcard-driver.c:
for_each_card_prelinks(card, i, dai_link) {
if (adsp_node && !strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF")))
dai_link->platforms->of_node = adsp_node;
else if (!dai_link->platforms->name && !dai_link->platforms->of_node)
dai_link->platforms->of_node = platform_node;
}
where the code expects the platforms array to have space for at least one entry.
Add an COMP_EMPTY() entry so that dai_link->platforms has space.
Fixes: e70b8dd267 ("ASoC: mediatek: mt8195: Remove afe-dai component and rework codec link")
Signed-off-by: Chen-Yu Tsai <wenst@chromium.org>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://patch.msgid.link/20240624061257.3115467-1-wenst@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Andrejs Cainikovs <andrejs.cainikovs@gmail.com>:
This change adds local MCLK handling, which would cover a case when a
reference audio clock is present in a system, but is not allowed to be
changed, see [1].
[1]: https://lore.kernel.org/all/ZfBdxrzX3EnPuGOn@ediswmail9.ad.cirrus.com/
Merge series from Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>:
Make few pointers in ASoC functions as pointers to const, so the code is
clearer to read, a bit safer and allows further constifications (e.g.
placing some data as rodata).
Merge series from Christian Hewitt <christianshewitt@gmail.com>:
Update bindings and add a driver compatible for the pcm5242
chip used on the Odroid HiFi-Shield2 i2c mezzanine board.
Merge series from Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>:
Set of simple cleanups from similar issues in all Qualcomm WCD93xx
codecs.
The first patch "ASoC: codecs: wcd-mbhc: Constify passed MBHC reg
fields" is a requirement for few others, but except this they are
independent.
Merge series from Ricard Wanderlof <ricard.wanderlof@axis.com>:
In some cases, depending on system design, the MICBIAS pins on the
chip are not needed as such, but a couple of extra GPIO pins would be
useful. This patch allows the MICBIAS pins to be configured in the
device tree as general purpose output pins, controlled via the GPIO
framework.
Owing to their originally intended purpose there are some limitations:
when the MICBIAS pins are deactivated, they will float, so will likely
need a pulldown in many applications. When activated, they will
assume the voltage specified by the micbias1-vg and micbias2-vg
properties, respectively, meaning that the resulting output voltage
will be 2.0 V, 2.5 V or AVDD .
Merge series from Paul Handrigan <paulha@opensource.cirrus.com>:
This patchset provides ASoC support for the latest family
of Cirrus Logic multichannel, high performance audio ADCs.
The devices that are supported are CS5302 (2 channel ADC),
CS5304 (4 channel ADC), and CS5308 (8 channel ADC).
The value “-ENOMEM” was assigned to the local variable “ret”
in one if branch after a devm_kzalloc() call failed at the beginning.
This error code will trigger then a pcmdevice_remove() call with a passed
null pointer so that an undesirable dereference will be performed.
Thus return the appropriate error code directly.
Fixes: 1324eafd37 ("ASoc: PCM6240: Create PCM6240 Family driver code")
Signed-off-by: Hao Ge <gehao@kylinos.cn>
Link: https://patch.msgid.link/20240617020954.17252-1-hao.ge@linux.dev
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP common dma driver has a buffer size and period size restriction
which should be 64 byte aligned. Add pcm constraints for the same.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://patch.msgid.link/20240617072844.871468-8-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
I2S clock generation registers should be programmed before starting the I2S
dma when I2S controller is programmed as clock master. Move i2s clock
generation register programming sequence prior to i2s dma start.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://patch.msgid.link/20240617072844.871468-7-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP provides different IO configurations(ACP PDM, I2S and SoundWire).
I2S mclk should be programmed only when I2S configuration is selected and
I2S controller is programmed as clock master.
Modify the conditional check for programming i2s mclk.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://patch.msgid.link/20240617072844.871468-6-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In acp_i2s_probe(), acp_base null check is verified.
As already acp_base null check will be verified in acp platform
driver probe sequence, additional NULL check in acp_i2s_probe() is not
needed. Remove acp_i2s_probe() function.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://patch.msgid.link/20240617072844.871468-4-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
priv->pdev pointer was set after being used in
fsl_asoc_card_audmux_init().
Move this assignment at the start of the probe function, so
sub-functions can correctly use pdev through priv.
fsl_asoc_card_audmux_init() dereferences priv->pdev to get access to the
dev struct, used with dev_err macros.
As priv is zero-initialised, there would be a NULL pointer dereference.
Note that if priv->dev is dereferenced before assignment but never used,
for example if there is no error to be printed, the driver won't crash
probably due to compiler optimisations.
Fixes: 708b4351f0 ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support")
Signed-off-by: Elinor Montmasson <elinor.montmasson@savoirfairelinux.com>
Link: https://patch.msgid.link/20240620132511.4291-2-elinor.montmasson@savoirfairelinux.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The ACPI IDs used in the CS35L56 HDA drivers are all handled by the
serial multi-instantiate driver which starts multiple Linux device
instances from a single ACPI Device() node.
As serial multi-instantiate is not an optional part of the system add it
as a dependency in Kconfig so that it is not overlooked.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Link: https://lore.kernel.org/20240619161602.117452-1-simont@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
chip->flag variable assignment will be skipped when acp platform device
creation is skipped. In this case chip>flag value will not be set.
chip->flag variable should be assigned along with other structure
variables for 'chip' structure. Move chip->flag variable assignment
prior to acp platform device creation.
Fixes: 3a94c8ad0a ("ASoC: amd: acp: add code for scanning acp pdm controller")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240617072844.871468-3-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP supports different pin configurations for I2S IO. Checking ACP pin
configuration value against specific value breaks the functionality for
other I2S pin configurations. This check is no longer required in i2s dai
driver probe call as i2s configuration check will be verified during acp
platform device creation sequence.
Remove i2s_mode check in acp_i2s_probe() function.
Fixes: b24484c18b ("ASoC: amd: acp: ACP code generic to support newer platforms")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240617072844.871468-2-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When acp platform device creation is skipped, chip->chip_pdev value will
remain NULL. Add NULL check for chip->chip_pdev structure in
snd_acp_resume() function to avoid null pointer dereference.
Fixes: 088a40980e ("ASoC: amd: acp: add pm ops support for acp pci driver")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240617072844.871468-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for the Asahi Kasei AK4619 audio codec.
[Kuninori cleanuped and adjusted to upstream code]
Signed-off-by: Khanh Le <khanh.le.xr@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://msgid.link/r/87iky5wxvr.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
IRQ lookup functions such as those in ACPI can return error values when
an IRQ is not defined. The i2c core driver converts the error codes to a
value of 0 and the SPI bus driver passes them unaltered to client device
drivers.
The cs35l56 driver should only accept positive non-zero values as IRQ
numbers.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Fixes: 8a731fd37f ("ASoC: cs35l56: Move utility functions to shared file")
Link: https://msgid.link/r/20240617135338.82006-1-simont@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Dell SKU 0C64 has a single rt1318 amplifier.
The prefix name of control still needs to be set rt1318-1 corresponding to UCM config.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240612075740.1678082-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver does not modify static wcd939x_sdw_ch_info array, so it can be
made const for code safety.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Neil Armstrong <neil.armstrong@linaro.org>
Link: https://msgid.link/r/20240612-asoc-wcd9xxx-wide-cleanups-v1-21-0d15885b2a06@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver does not modify few static data (MBHC reg fields, IRQ chip), so
make them const for code safety.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Neil Armstrong <neil.armstrong@linaro.org>
Link: https://msgid.link/r/20240612-asoc-wcd9xxx-wide-cleanups-v1-20-0d15885b2a06@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
wcd934x_codec_parse_data() function is called only from probe(), so
printing errors on resource acquisition is discouraged, because it can
pollute dmesg in case of probe deferral. The actual deferral is here
unlikely, but still the code is a bit simpler with dev_err_probe().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240612-asoc-wcd9xxx-wide-cleanups-v1-10-0d15885b2a06@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver stores the voltage of mic bias in fields in state container
structure, but actually never reads them - except for the mic2 bias
(micb2_mv field). Drop the fields from the structure so the code will
be a bit simpler.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240612-asoc-wcd9xxx-wide-cleanups-v1-9-0d15885b2a06@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver does not modify few static data (arrays with sample rates, MBHC
reg fields, regmap config), so make them const for code safety.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240612-asoc-wcd9xxx-wide-cleanups-v1-8-0d15885b2a06@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver was prepared to adjust DMIC microphone parameters according to
DMIC clock rate which is read from platform data or Devicetree. The
latter part never happened, so the code is always called with
dmic_clk_rate=0, prints error and uses default/fallback values. All
this part can be simplified by dropping dead parts of code.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240612-asoc-wcd9xxx-wide-cleanups-v1-6-0d15885b2a06@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Error-level should not be used as debugging. The code (function
wcd9335_get_dmic_clk_val()) will always be called with same parameters,
so this is not really useful debug anyway, so drop it.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240612-asoc-wcd9xxx-wide-cleanups-v1-5-0d15885b2a06@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
wcd9335_parse_dt() function is called only from probe(), so printing
errors on resource acquisition is discouraged, because it can pollute
dmesg. Use dev_err_probe() to fix this and also make the code a bit
simpler.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240612-asoc-wcd9xxx-wide-cleanups-v1-4-0d15885b2a06@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver does not modify few static data (arrays with sample rates,
interrupt description, regmap config), so make them const for code
safety.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240612-asoc-wcd9xxx-wide-cleanups-v1-3-0d15885b2a06@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The wcd-mbhc-v2 helper code/module does not modify passed array of
registry fields, thus it can be made const for code safety. It will
also allow individual drivers to allocate the array in rodata.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240612-asoc-wcd9xxx-wide-cleanups-v1-1-0d15885b2a06@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
In only loading RCA (Reconfigurable Architecture) binary case, no DSP
program will be working inside tas2563/tas2781, that is dsp-bypass mode,
do not support speaker protection, or audio acoustic algorithms in this
mode.
Fixes: ef3bcde75d ("ASoC: tas2781: Add tas2781 driver")
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240614133646.910-1-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This change adds MCLK clock handling directly within driver.
When used in combination with simple-audio-card, and mclk-fs is set,
simple-audio-card will change MCLK frequency before configuring PLL.
In some cases, however, MCLK reference clock should be static (see [1]),
which means it needs to be moved away from simple-audio-card.
[1]: https://lore.kernel.org/all/ZfBdxrzX3EnPuGOn@ediswmail9.ad.cirrus.com/
Signed-off-by: Andrejs Cainikovs <andrejs.cainikovs@toradex.com>
Link: https://msgid.link/r/20240613084652.13113-4-andrejs.cainikovs@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Following bias state machine logic this bit is set twice before playback.
This change makes sure this bit set is set only once.
Signed-off-by: Andrejs Cainikovs <andrejs.cainikovs@toradex.com>
Suggested-by: Emanuele Ghidoli <emanuele.ghidoli@toradex.com>
Link: https://msgid.link/r/20240613084652.13113-3-andrejs.cainikovs@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Next commit in series makes a change which calls nau8822_set_pll() from
nau8822_set_dai_sysclk(). Moving latter after the former would avoid a
forward declaration, and this is exactly what this change does.
Signed-off-by: Andrejs Cainikovs <andrejs.cainikovs@toradex.com>
Link: https://msgid.link/r/20240613084652.13113-2-andrejs.cainikovs@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It is possible that during system boot when there multiple devices
attempting simultaneous initialization on a slow control bus the
download of firmware and tuning data may take a user perceivable amount
of time (a slow I2C bus with 4 amps this work could take over 2
seconds).
Adopt a pattern used in the ASoC driver and perform this activity in a
background thread so that interactive performance is not impaired. The
system_long_wq is a parallel workqueue and driver instances will perform
their firmware downloads in parallel to make best use of available bus
bandwidth.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Link: https://lore.kernel.org/20240618130011.62860-1-simont@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add boolean ti,micbias1-gpo and ti,micbias2-gpo devicetree properties.
When set, the respective MICBIAS pins can be used as general purpose
outputs controlled via the GPIO framework, in addition to the two
configurable GPIO pins.
This is useful in applications where the MICBIAS functionality is
not required, but it is useful to have a couple of extra GPIO pins.
The voltage on the respective MICBIAS pin in the active state is
governed by the ti,micbias1-vg and ti,micbias2-vg properties,
respectively (same properties as when the pins are used as
MICBIAS pins).
Signed-off-by: Ricard Wanderlof <ricard.wanderlof@axis.com>
Link: https://msgid.link/r/20240607-tlv320adc3xxx-micbias-gpo-v3-2-59dbec8b98f4@axis.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for handling jack events of USB (DisplayPort).
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Tested-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org> # X13s
Link: https://msgid.link/r/20240606104922.114229-5-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for Display Port Jack events.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Tested-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org> # X13s
Link: https://msgid.link/r/20240606104922.114229-4-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a common function to add Display port jack.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Tested-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org> # X13s
Link: https://msgid.link/r/20240606104922.114229-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Returned 'struct snd_soc_pcm_stream' by snd_soc_dai_get_pcm_stream() is
not modified by the users, so it can be changed as pointer to const.
This is a necessary step towards making the 'dai->driver' a pointer to
const.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240617-n-asoc-const-auto-selectable-formats-v1-5-8004f346ee38@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
get_channel_map() is supposed to obtain map of channels without
modifying the state of the given DAI, so make the pointer to 'struct
snd_soc_dai' as pointing to const.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240617-n-asoc-const-auto-selectable-formats-v1-4-8004f346ee38@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Several ASoC functions receive pointers to data which is not modified,
e.g. pointers to 'snd_soc_dai', 'snd_soc_pcm_runtime',
'snd_pcm_hw_params' and 'snd_soc_dai_link'.
All these pointers can be made as a pointer to const. This makes code
safer, serves as clear annotation of function's intentions (no ownership
passed to the function, no modifications) and allows putting pointed
structures in rodata (if ever applicable).
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240617-n-asoc-const-auto-selectable-formats-v1-3-8004f346ee38@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
There are lots of code checking NULL for kcontrol passed to
snd_ctl_remove() in the caller side. Let's make snd_ctl_remove()
accepting the NULL kcontrol instead a la free(), so that we can clean
up the caller side.
Link: https://lore.kernel.org/20240617100529.6667-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a simple sanity check to HD-audio HDMI Channel Map controls.
Although the value might not be accepted for the actual connection, we
can filter out some bogus values beforehand, and that should be enough
for making kselftest happier.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/20240616073454.16512-7-tiwai@suse.de
The values returned from Playback Channel Map and Capture Channel Map
controls may vary dynamically depending on the corresponding PCM
stream. Mark those as volatile to indicate the values are unstable
and not suitable for testing.
Note that we may change the driver to return -EINVAL, but this would
bring other side effects, such as "alsactl restore" would start
receiving unexpected errors. So we still keep returning 0 for those
invalid inputs.
Reported-by: Paul Menzel <pmenzel@molgen.mpg.de>
Closes: https://lore.kernel.org/r/1d44be36-9bb9-4d82-8953-5ae2a4f09405@molgen.mpg.de
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/20240616073454.16512-6-tiwai@suse.de
Although we have already a mechanism for sanity checks of input values
for control writes, it's not applied unless the kconfig
CONFIG_SND_CTL_INPUT_VALIDATION is set due to the performance reason.
Nevertheless, it still makes sense to apply the same check for user
elements despite of its cost, as that's the only way to filter out the
invalid values; the user controls are handled solely in ALSA core
code, and there is no corresponding driver, after all.
This patch adds the same input value validation for user control
elements at its put callback. The kselftest will be happier with this
change, as the incorrect values will be bailed out now with errors.
For other normal controls, the check is applied still only when
CONFIG_SND_CTL_INPUT_VALIDATION is set.
Reported-by: Paul Menzel <pmenzel@molgen.mpg.de>
Closes: https://lore.kernel.org/r/1d44be36-9bb9-4d82-8953-5ae2a4f09405@molgen.mpg.de
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Mark Brown <broonie@kernel.org>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/20240616073454.16512-4-tiwai@suse.de
So far the HD-audio driver has been tolerant about the input values
and accepts any values by correcting the amp volume and switch values
internally. But now our own selftest starts complaining about this
behavior, so let's be picky and change the behavior to return -EINVAL
for invalid input values instead.
Reported-by: Paul Menzel <pmenzel@molgen.mpg.de>
Closes: https://lore.kernel.org/r/1d44be36-9bb9-4d82-8953-5ae2a4f09405@molgen.mpg.de
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/20240616073454.16512-3-tiwai@suse.de
So far the vmaster code has been tolerant about the input values and
accepts any values by correcting internally. But now our own selftest
starts complaining about this behavior, so let's be picky and change
the behavior to return -EINVAL for invalid input values instead.
Reported-by: Paul Menzel <pmenzel@molgen.mpg.de>
Closes: https://lore.kernel.org/r/1d44be36-9bb9-4d82-8953-5ae2a4f09405@molgen.mpg.de
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/20240616073454.16512-2-tiwai@suse.de
Add initial Senarytech codec support for SN6186. Note that this hda
patch relies on the configuration default registers to be set correctly
(normally by BIOS/firmware) in order for it to set up pin widgets
properly.
Signed-off-by: bo liu <bo.liu@senarytech.com>
Link: https://lore.kernel.org/20240618010121.67335-1-bo.liu@senarytech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hda_component contains information shared from the amp drivers to
the codec that can be altered (for example as the driver unloads). Guard
the update and use of these to prevent use of stale data.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/20240617154105.108635-5-simont@opensource.cirrus.com
There is one codec shared across all of the bound HDA components and a
copy is usually stashed in the amp driver so it doesn't need to be in
every hda_component.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/20240617154105.108635-4-simont@opensource.cirrus.com
Change the hda_component binding APIs to pass the hds_component_parent
as the parameter so the array of components can be abstracted.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/20240617154105.108635-3-simont@opensource.cirrus.com
In preparation for moving duplicated members from the hda_component
structure introduce a parent structure that wraps the array of
components. This also allows us to confine the knowledge of the maximum
number of entries to the hda_component files and eliminate passing that
redundant information around and making direct accesses to the array.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/20240617154105.108635-2-simont@opensource.cirrus.com
With ARCH=m68k, make allmodconfig && make W=1 C=1 reports:
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/oss/dmasound/dmasound_core.o
Add the missing invocation of the MODULE_DESCRIPTION() macro.
Signed-off-by: Jeff Johnson <quic_jjohnson@quicinc.com>
Reviewed-by: Geert Uytterhoeven <geert@linux-m68k.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/20240617-md-m68k-sound-oss-dmasound-v1-1-5c19306be930@quicinc.com
The static arrays passed as 'auto_selectable_formats' are not modified
by the drivers nor by the core code, so make it const for code safety.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Acked-by: Herve Codina <herve.codina@bootlin.com>
Link: https://msgid.link/r/20240617125735.582963-2-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add device tree property to define auxiliary devices to be added to
Audio Graph Card which is already supported on Simle Card.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://msgid.link/r/878qz4ry81.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
'struct mtk_adsp_ipc_ops' is not modified in these drivers.
Constifying this structure moves some data to a read-only section, so
increase overall security.
In order to do it, "struct mtk_adsp_ipc" also needs to be adjusted to this
new const qualifier.
On a x86_64, with allmodconfig:
Before:
======
text data bss dec hex filename
15533 2383 0 17916 45fc sound/soc/sof/mediatek/mt8195/mt8195.o
After:
=====
text data bss dec hex filename
15557 2367 0 17924 4604 sound/soc/sof/mediatek/mt8195/mt8195.o
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Link: https://msgid.link/r/a45d6b2b5ec040ea0fc78fca662c2dca3f13a49f.1718312321.git.christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
With ARCH=arm, make allmodconfig && make W=1 C=1 reports:
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/fsl/imx-pcm-fiq.o
Add the missing invocation of the MODULE_DESCRIPTION() macro.
Signed-off-by: Jeff Johnson <quic_jjohnson@quicinc.com>
Link: https://msgid.link/r/20240616-md-arm-sound-soc-fsl-v2-1-228772e81a54@quicinc.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Similarly to other Lenovo laptops these also have a dual speaker
setup with a shared amplifier.
The model also seems to have a conflicting PCI SSID with the codec SSID for
the Legion Y9000X 2022 IAH7. Only tested on the Yoga Pro 7, as I don't have
access to the other laptop.
Signed-off-by: Gergely Meszaros <meszaros.gergely97@gmail.com>
Link: https://lore.kernel.org/r/20240616085233.16922-1-meszaros.gergely97@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>:
This patchset adds support to reading codec version and also adds
support for v2.5 codec version in rx macro.
LPASS 2.5 and up versions have changes in some of the rx blocks which
are required to get headset functional correctly.
Tested this on SM8450, X13s and x1e80100 crd.
This changes also fixes issue with sm8450, sm8550, sm8660 and x1e80100.
LPASS Codec v2.5 has significant changes in the rx register strides.
Due to this headset playback on SM8550, SM8650, x1e80100 and all SoCs
after SM8450 have only Left working.
This patch adjusts the registers to accomdate 2.5 changes. With this
fixed now L and R are functional on Headset playback.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Tested-by: Neil Armstrong <neil.armstrong@linaro.org> # on SM8650-HDK
Link: https://lore.kernel.org/r/20240612-lpass-codec-v25-v4-3-f63d3676dbc4@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
LPASS Codec v2.5 has significant changes in the rx block register strides.
This is a preparatory patch to do the required changes in the existing driver
to be able to accomdate these changes.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Tested-by: Neil Armstrong <neil.armstrong@linaro.org> # on SM8650-HDK
Link: https://lore.kernel.org/r/20240612-lpass-codec-v25-v4-2-f63d3676dbc4@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
LPASS Digital codec have changes in register layout across multiple
versions. Add a proper way read the codec version allowint all the lpass
macro drivers (tx, rx, wsa, va) to configure the registers correctly.
LPASS VA macro has the required registers to read the codec version.
Read the the version and make it available to other lpass codec macros
using the common helper functions.
Existing method of using LPASS IP version is not accurate as the same
the codec versioning is totally independent of LPASS IP block versions.
These helper functions should be able to provide a convient way to get
the codec version, and will help scale the drivers in right direction.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Tested-by: Neil Armstrong <neil.armstrong@linaro.org> # on SM8650-HDK
Link: https://lore.kernel.org/r/20240612-lpass-codec-v25-v4-1-f63d3676dbc4@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
There is an issue around with error handling and graph management with
the exising code, none of the error paths close the graph, which result in
leaving the loaded graph in dsp, however the driver thinks otherwise.
This can have a nasty side effect specially when we try to load the same
graph to dsp, dsp returns error which leaves the board with no sound and
requires restart.
Fix this by properly closing the graph when we hit errors between
open and close.
Fixes: 30ad723b93 ("ASoC: qdsp6: audioreach: add q6apm lpass dai support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Tested-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org> # X13s
Link: https://lore.kernel.org/r/20240613-q6apm-fixes-v1-1-d88953675ab3@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
If the ASP1 DAI is hooked up by the machine driver the ASP TX mixer
sources should be initialized to disconnected. There aren't currently
any available products using the ASP so this doesn't affect any
existing systems.
The cs35l56 does not have any fixed default for the mixer source
registers. When the cs35l56 boots, its firmware patches these registers
to setup a system-specific routing; this is so that Windows can use
generic SDCA drivers instead of needing knowledge of chip-specific
registers. The setup varies between end-products, which each have
customized firmware, and so the default register state varies between
end-products. It can also change if the firmware on an end-product is
upgraded - for example if a change was needed to the routing for Windows
use-cases. It must be emphasized that the settings applied by the
firmware are not internal magic tuning; they are statically implementing
use-case setup that on Linux would be done via ALSA controls.
The driver is currently syncing the mixer controls with whatever
initial state the firmware wrote to the registers, so that they report
the actual audio routing. But if the ASP DAI is hooked up this can create
a powered-up DAPM graph without anything intentionally setting up a path.
This can lead to parts of the audio system powering up unexpectedly.
For example when cs35l56 is connected to cs42l43 using a codec-codec link,
this can create a complete DAPM graph which then powers-up cs42l43. But
the cs42l43 can only be clocked from its SoundWire bus so this causes a
bunch of errors in the kernel log where cs42l43 is unexpectedly powered-up
without a clock.
If the host is taking ownership of the ASP (either directly or as a
codec-to-codec link) there is no need to keep the mixer settings that the
firmware wrote. The driver has ALSA controls for setting these using
standard Linux mechanisms. So if the machine driver hooks up the ASP the
ASP mixers are initialized to "None" (no input). This prevents unintended
DAPM-graph power-ups, and means the initial state of the mixers is
always going to be None.
Since the initial state of the mixers can vary from system to system and
potentially between firmware upgrades, no use-case manager can currently
assume that cs35l56 has a known initial state. The firmware could just as
easily default them to "None" as to any input source. So defaulting them
to "None" in the driver is not increasing the entropy of the system.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240613132527.46537-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The interface associated with the hda_component should be deactivated
before the driver is deconstructed during removal.
Fixes: 4e7914eb1d ("ALSA: hda/tas2781: remove sound controls in unbind")
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240613133713.75550-4-simont@opensource.cirrus.com
The interface associated with the hda_component should be deactivated
before the driver is deconstructed during removal.
Fixes: 7b2f3eb492 ("ALSA: hda: cs35l41: Add support for CS35L41 in HDA systems")
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240613133713.75550-3-simont@opensource.cirrus.com
The interface associated with the hda_component should be deactivated
before the driver is deconstructed during removal.
Fixes: 73cfbfa9ca ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240613133713.75550-2-simont@opensource.cirrus.com
In the quest to make struct device constant, start by making
to_auxiliary_drv() return a constant pointer so that drivers that call
this can be fixed up before the driver core changes.
As the return type previously was not constant, also fix up all callers
that were assuming that the pointer was not going to be a constant one
in order to not break the build.
Cc: Dave Ertman <david.m.ertman@intel.com>
Cc: Ira Weiny <ira.weiny@intel.com>
Cc: Rafael J. Wysocki <rafael@kernel.org>
Cc: Bingbu Cao <bingbu.cao@intel.com>
Cc: Tianshu Qiu <tian.shu.qiu@intel.com>
Cc: Mauro Carvalho Chehab <mchehab@kernel.org>
Cc: Michael Chan <michael.chan@broadcom.com>
Cc: David S. Miller <davem@davemloft.net>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Jakub Kicinski <kuba@kernel.org>
Cc: Paolo Abeni <pabeni@redhat.com>
Cc: Jesse Brandeburg <jesse.brandeburg@intel.com>
Cc: Tony Nguyen <anthony.l.nguyen@intel.com>
Cc: Saeed Mahameed <saeedm@nvidia.com>
Cc: Leon Romanovsky <leon@kernel.org>
Cc: Tariq Toukan <tariqt@nvidia.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Cc: Bard Liao <yung-chuan.liao@linux.intel.com>
Cc: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Cc: Daniel Baluta <daniel.baluta@nxp.com>
Cc: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Richard Cochran <richardcochran@gmail.com>
Cc: linux-media@vger.kernel.org
Cc: netdev@vger.kernel.org
Cc: intel-wired-lan@lists.osuosl.org
Cc: linux-rdma@vger.kernel.org
Cc: sound-open-firmware@alsa-project.org
Cc: linux-sound@vger.kernel.org
Acked-by: Sakari Ailus <sakari.ailus@linux.intel.com> # drivers/media/pci/intel/ipu6
Acked-by: Mark Brown <broonie@kernel.org>
Reviewed-by: Martin Habets <habetsm.xilinx@gmail.com>
Link: https://lore.kernel.org/r/20240611130103.3262749-7-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
'alsa_sndif_hw_param' has been unused since the original
commit 1cee559351 ("ALSA: xen-front: Implement ALSA virtual sound
driver").
Remove it.
Signed-off-by: Dr. David Alan Gilbert <linux@treblig.org>
Link: https://lore.kernel.org/r/20240601232604.198662-1-linux@treblig.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
dsp_driver=4 will force the AVS driver stack to be used, it is better to
docuement this.
Fixes: 1affc44ea5 ("ASoC: Intel: avs: PCI driver implementation")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240607060021.11503-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for this laptop, which uses CS35L41 HDA amps.
The laptop does not contain valid _DSD for these amps, so requires
entries into the CS35L41 configuration table to function correctly.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240606130351.333495-5-sbinding@opensource.cirrus.com
Add support for this laptop, which uses CS35L41 HDA amps.
The laptop does not contain valid _DSD for these amps, so requires
entries into the CS35L41 configuration table to function correctly.
[ fixed to lower hex numbers in quirk entries -- tiwai ]
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240606130351.333495-4-sbinding@opensource.cirrus.com
The Framework Laptop 16 does not have a combination headphone/headset
3.5mm jack; however, applying the pincfg from the Laptop 13 (nid=0x19)
erroneously informs hda that the node is present.
Fixes: 8804fa04a4 ("ALSA: hda/realtek: Add Framework laptop 16 to quirks")
Signed-off-by: Dustin L. Howett <dustin@howett.net>
Reviewed-by: Mario Limonciello <mario.limonciello@amd.com>
Link: https://lore.kernel.org/r/20240605-alsa-hda-realtek-remove-framework-laptop-16-from-quirks-v1-1-11d47fe8ec4d@howett.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The internal mic boost on the N14AP7 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://lore.kernel.org/r/20240605153923.2837-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP ProBook 445/465 G11 needs ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF quirk to
make mic-mute/audio-mute working.
Signed-off-by: Andy Chi <andy.chi@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240605092243.41963-1-andy.chi@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With ARCH=x86, make allmodconfig && make W=1 C=1 reports:
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/renoir/snd-acp3x-rn.o
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/yc/snd-soc-acp6x-mach.o
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/acp/snd-acp-i2s.o
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/acp/snd-acp-pdm.o
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/acp/snd-acp-legacy-common.o
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/acp/snd-acp-pci.o
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/ps/snd-soc-ps-mach.o
Add the missing invocations of the MODULE_DESCRIPTION() macro.
Signed-off-by: Jeff Johnson <quic_jjohnson@quicinc.com>
Link: https://lore.kernel.org/r/20240612-md-sound-soc-amd-v1-1-ad1de0409c11@quicinc.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If the ipc_prepare() callback fails for a module instance, on error rewind
we must skip the ipc_unprepare() call for ones that has positive use count.
The positive use count means that the module instance is in active use, it
cannot be unprepared.
The issue affects capture direction paths with branches (single dai with
multiple PCMs), the affected widgets are in the shared part of the paths.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20240612121203.15468-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Jai Luthra <j-luthra@ti.com>:
This series fixes two patches:
1. Fix the dmaengine API usage by calling dmaengine_synchronize() after
dmaengine_terminate_async() when xrun events occur in application
2. Use the McASP AFIFO property from DT to refine the period size,
instead of hardcoding minimum to 64 samples
This accidentally returns success instead of -EINVAL.
Fixes: c91d0c2e19 ("ASoC: samsung: midas_wm1811: Add GPIO-based headset jack detection")
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Acked-by: Artur Weber <aweber.kernel@gmail.com>
Link: https://lore.kernel.org/r/01590109-cf27-404b-88ff-b42bb73ca1c6@moroto.mountain
Signed-off-by: Mark Brown <broonie@kernel.org>
When cs35l56 is connected via cs42l43 there isn't an ACPI node for the
cs35l56 so all properties are under the cs42l43 ACPI node. We're adding
a property as a way for the cs42l43 driver to pass this info in via a
software node.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240611132556.1557075-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Set driver name to "HDMI". This simplifies the code and gets rid of
the following error messages:
ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e'
Signed-off-by: Primoz Fiser <primoz.fiser@norik.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://lore.kernel.org/r/20240610125847.773394-1-primoz.fiser@norik.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The minimum period size was enforced to 64 as older devices integrating
McASP with EDMA used an internal FIFO of 64 samples.
With UDMA based platforms this internal McASP FIFO is optional, as the
DMA engine internally does some buffering which is already accounted for
when registering the platform. So we should read the actual FIFO
configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to
64.
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Signed-off-by: Jai Luthra <j-luthra@ti.com>
Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Sometimes the stream may be stopped due to XRUN events, in which case
the userspace can call snd_pcm_drop() and snd_pcm_prepare() to stop and
start the stream again.
In these cases, we must wait for the DMA channel to synchronize before
marking the stream as prepared for playback, as the DMA channel gets
stopped by drop() without any synchronization. Make sure the ALSA core
synchronizes the DMA channel by adding a sync_stop() hook.
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Signed-off-by: Jai Luthra <j-luthra@ti.com>
Link: https://lore.kernel.org/r/20240611-asoc_next-v3-1-fcfd84b12164@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Andy Shevchenko <andriy.shevchenko@linux.intel.com>:
Replace or drop the legacy header that is subject to remove.
Not all of them were compile-tested, the series might have
hidden compilation errors.
