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2024-07-08ASoC: codecs: audio-iio-aux: Simplify audio_iio_aux_probe() with cleanup.hKrzysztof Kozlowski
Allocate the memory with scoped/cleanup.h in audio_iio_aux_probe() to reduce error handling (less error paths) and make the code a bit simpler. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> Link: https://patch.msgid.link/20240703-asoc-cleanup-h-v1-2-71219dfd0aef@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-08ASoC: codecs: audio-iio-aux: Simplify audio_iio_aux_add_dapms() with cleanup.hKrzysztof Kozlowski
Allocate the memory with scoped/cleanup.h in audio_iio_aux_add_dapms() to reduce error handling (less error paths) and make the code a bit simpler. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> Link: https://patch.msgid.link/20240703-asoc-cleanup-h-v1-1-71219dfd0aef@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-06ALSA: seq: Add tempo base unit for MIDI2 Set Tempo messagesTakashi Iwai
MIDI2 Set Tempo message defines the tempo in 10ns unit for finer accuracy, while MIDI1 was defined in 1us unit. For adapting this different unit, introduce "tempo_base" field to snd_seq_queue_tempo struct so that user-space can pass the proper tempo base unit. The accepted value is limited, it must be either 0, 10 or 1000. The protocol version is bumped to 1.0.4 along with this. The access with the older protocol version ignores the tempo-base value in ioctls and always treats as 1000. Reviewed-by: Jaroslav Kysela <perex@perex.cz> Link: https://patch.msgid.link/20240705160344.6481-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2024-07-06ALSA: hda/realtek: Limit mic boost on VAIO PRO PXEdson Juliano Drosdeck
The internal mic boost on the VAIO models VJFE-CL and VJFE-IL is too high. Fix this by applying the ALC269_FIXUP_LIMIT_INT_MIC_BOOST fixup to the machine to limit the gain. Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com> Cc: <stable@vger.kernel.org> Link: https://patch.msgid.link/20240705141012.5368-1-edson.drosdeck@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2024-07-05ASoc: pcm6240: Remove unnecessary name-prefix for all the controlsShenghao Ding
Adding name-prefix for each audio controls is a redundant, because name-prefix will be automatically added behind the control name when creating a new control. Signed-off-by: Shenghao Ding <shenghao-ding@ti.com> Link: https://patch.msgid.link/20240705064846.1723-1-shenghao-ding@ti.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-05ALSA: pcm: Fix id copying in snd_pcm_set_sync_per_card()Jaroslav Kysela
Avoid to use strncpy and do proper length limiting (12 bytes) to avoid out of array access. Fixes: d712c58c55d9 ("ALSA: pcm: optimize and clarify stream synchronization ID API") Reported-by: kernel test robot <lkp@intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202407050501.o5Z3bibi-lkp@intel.com/ Signed-off-by: Jaroslav Kysela <perex@perex.cz> Link: https://patch.msgid.link/20240705075828.19746-1-perex@perex.cz Signed-off-by: Takashi Iwai <tiwai@suse.de>
2024-07-05ALSA: hda: cs35l41: Fix swapped l/r audio channels for Lenovo ThinBook 13x Gen4Dmitry Savin
Fixes audio channel assignment in configuration table for ThinkBook 13x Gen4. Fixes: b32f92d1af37 ("ALSA: hda: cs35l41: Support Lenovo Thinkbook 13x Gen 4") Signed-off-by: Dmitry Savin <envelsavinds@gmail.com> Reviewed-by: Stefan Binding <sbinding@opensource.cirrus.com> Link: https://patch.msgid.link/20240704211402.87776-1-envelsavinds@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2024-07-05ALSA: pcm: Fix missing export of snd_pcm_set_sync_per-card()Takashi Iwai
The recent change for the PCM sync API forgot to export a new API function that is currently used in emu10k1 driver. This patch adds the missing export. Fixes: d712c58c55d9 ("ALSA: pcm: optimize and clarify stream synchronization ID API") Reported-by: kernel test robot <lkp@intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202407051343.1Sz8AKFt-lkp@intel.com/ Reviewed-by: Jaroslav Kysela <perex@perex.cz> Link: https://patch.msgid.link/20240705063555.17220-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2024-07-04Add support for non-interleaved mode in qmc_audioMark Brown
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().
2024-07-04ASoC: codecs: lpass-rx-macro: Few code cleanupsMark Brown
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.
2024-07-04ASoc: tas2781: Set "Speaker Force Firmware Load" as the common kcontrol for ↵Shenghao Ding
both tas27871 and tas2563 Set "Speaker Force Firmware Load" as the common kcontrol for both tas27871 and tas2563 and move it into newly-created tasdevice_snd_controls, and keep the digital gain and analog gain in tas2781_snd_controls. Signed-off-by: Shenghao Ding <shenghao-ding@ti.com> Link: https://patch.msgid.link/20240704094939.1824-1-shenghao-ding@ti.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-04ALSA: hda: cs35l41: Fix missing Speaker ID GPIO description in _DSDStefan Binding
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>
2024-07-04ASoC: amd: Adjust error handling in case of absent codec deviceAleksandr Mishin
acpi_get_first_physical_node() can return NULL in several cases (no such device, ACPI table error, reference count drop to 0, etc). Existing check just emit error message, but doesn't perform return. Then this NULL pointer is passed to devm_acpi_dev_add_driver_gpios() where it is dereferenced. Adjust this error handling by adding error code return. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 02527c3f2300 ("ASoC: amd: add Machine driver for Jadeite platform") Signed-off-by: Aleksandr Mishin <amishin@t-argos.ru> Link: https://patch.msgid.link/20240703191007.8524-1-amishin@t-argos.ru Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-04ASoC: codecs: wcd939x: Fix typec mux and switch leak during device removalKrzysztof Kozlowski
Driver does not unregister typec structures (typec_mux_dev and typec_switch_desc) during removal leading to leaks. Fix this by moving typec registering parts to separate function and using devm interface to release them. This also makes code a bit simpler: - Smaller probe() function with less error paths and no #ifdefs, - No need to store typec_mux_dev and typec_switch_desc in driver state container structure. Cc: stable@vger.kernel.org Fixes: 10f514bd172a ("ASoC: codecs: Add WCD939x Codec driver") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> Reviewed-by: Neil Armstrong <neil.armstrong@linaro.org> Link: https://patch.msgid.link/20240701122616.414158-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-04ASoC: ak4458: remove "reset-gpios" property handlerShengjiu Wang
commit c721f189e89c0 ("reset: Instantiate reset GPIO controller for shared reset-gpios") check if there is no "resets" property will fallback to "reset-gpios". So don't need to handle "reset-gpios" separately in the driver, the "reset-gpios" handler is duplicated with "resets" control handler, remove it. Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com> Link: https://patch.msgid.link/1720009575-11677-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-04ASoC: SOF: ipc4-topology: Use single token list for the copiersPeter Ujfalusi
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>
2024-07-04ASoC: fsl: fsl_qmc_audio: Add support for non-interleaved mode.Herve Codina
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>
2024-07-04dt-bindings: sound: fsl,qmc-audio: Add support for multiple QMC channels per DAIHerve Codina
The QMC audio uses one QMC channel per DAI and uses this QMC channel to transmit interleaved audio channel samples. In order to work in non-interleave mode, a QMC audio DAI needs to use multiple QMC channels. In that case, the DAI maps each QMC channel to exactly one audio channel. Allow QMC audio DAIs with multiple QMC channels attached. Signed-off-by: Herve Codina <herve.codina@bootlin.com> Reviewed-by: Rob Herring (Arm) <robh@kernel.org> Link: https://patch.msgid.link/20240701113038.55144-10-herve.codina@bootlin.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-04soc: fsl: cpm1: qmc: Introduce qmc_chan_count_phandles()Herve Codina
No function in the QMC API is available to get the number of phandles present in a phandle list. Fill this lack introducing qmc_chan_count_phandles(). Signed-off-by: Herve Codina <herve.codina@bootlin.com> Link: https://patch.msgid.link/20240701113038.55144-9-herve.codina@bootlin.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-04soc: fsl: cpm1: qmc: Introduce functions to get a channel from a phandle listHerve Codina
qmc_chan_get_byphandle() and the resource managed version retrieve a channel from a simple phandle. Extend the API and introduce qmc_chan_get_byphandles_index() and the resource managed version in order to retrieve a channel from a phandle list using the provided index to identify the phandle in the list. Also update qmc_chan_get_byphandle() and the resource managed version to use qmc_chan_get_byphandles_index() and so avoid code duplication. Signed-off-by: Herve Codina <herve.codina@bootlin.com> Link: https://patch.msgid.link/20240701113038.55144-8-herve.codina@bootlin.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-04ASoC: fsl: fsl_qmc_audio: Introduce qmc_dai_constraints_interleaved()Herve Codina
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>
2024-07-04ASoC: fsl: fsl_qmc_audio: Introduce qmc_audio_pcm_{read,write}_submit()Herve Codina
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>
2024-07-04ASoC: fsl: fsl_qmc_audio: Identify the QMC channel involved in completion ↵Herve Codina
routines The current QMC audio driver uses only one QMC channel per DAI. The context used by QMC channel transfer (read and write) completion routines does not contains any QMC channel and the only one available per DAI is used to schedule the next transfer. This works pretty well with only one QMC channel per DAI. The future support for non-inlerleave mode will use several QMC channel per DAI. In that case, QMC channel transfer completion routines need to identify the QMC channel related to the completion. In order to fill this lack, even if identifying the current QMC channel among several QMC channels is not needed for the current code, add one indirection level and introduce the qmc_dai_chan data structrure. This structure contains the QMC channel involved in the completion and refererences to the runtime context (capture and playback) used by the DAI. Signed-off-by: Herve Codina <herve.codina@bootlin.com> Link: https://patch.msgid.link/20240701113038.55144-5-herve.codina@bootlin.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-04ASoC: fsl: fsl_qmc_audio: Split channel buffer and PCM pointer handlingHerve Codina
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>
2024-07-04ASoC: fsl: fsl_qmc_audio: Fix issues detected by checkpatchHerve Codina
./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>
2024-07-04ASoC: fsl: fsl_qmc_audio: Check devm_kasprintf() returned valueHerve Codina
devm_kasprintf() can return a NULL pointer on failure but this returned value is not checked. Fix this lack and check the returned value. Fixes: 075c7125b11c ("ASoC: fsl: Add support for QMC audio") Cc: stable@vger.kernel.org Signed-off-by: Herve Codina <herve.codina@bootlin.com> Link: https://patch.msgid.link/20240701113038.55144-2-herve.codina@bootlin.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-04ASoC: SOF: Intel: hda-pcm: Limit the maximum number of periods by ↵Peter Ujfalusi
MAX_BDL_ENTRIES 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>
2024-07-04ASoC: rt711-sdw: add missing readable registersBard Liao
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>
2024-07-04ASoC: SOF: Intel: hda: fix null deref on system suspend entryKai Vehmanen
When system enters suspend with an active stream, SOF core calls hw_params_upon_resume(). On Intel platforms with HDA DMA used to manage the link DMA, this leads to call chain of hda_dsp_set_hw_params_upon_resume() -> hda_dsp_dais_suspend() -> hda_dai_suspend() -> hda_ipc4_post_trigger() A bug is hit in hda_dai_suspend() as hda_link_dma_cleanup() is run first, which clears hext_stream->link_substream, and then hda_ipc4_post_trigger() is called with a NULL snd_pcm_substream pointer. Fixes: 2b009fa0823c ("ASoC: SOF: Intel: hda: Unify DAI drv ops for IPC3 and IPC4") Link: https://github.com/thesofproject/linux/issues/5080 Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://patch.msgid.link/20240704085708.371414-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-03ASoC: codecs: lpass-wsa-macro: Simplify with cleanup.hKrzysztof Kozlowski
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>
2024-07-03ASoC: codecs: lpass-rx-macro: Use unsigned for number of widgetsKrzysztof Kozlowski
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>
2024-07-03ASoC: dapm: Use unsigned for number of widgets in snd_soc_dapm_new_controls()Krzysztof Kozlowski
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>
2024-07-03ASoC: codecs: lpass-rx-macro: Keep static regmap_config as constKrzysztof Kozlowski
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>
2024-07-03ASoC: codecs: lpass-rx-macro: Simplify with cleanup.hKrzysztof Kozlowski
Allocate the default register values array with scoped/cleanup.h to reduce number of error paths and make code a bit simpler. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-2-6d98d4dd1ef5@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-03ASoC: codecs: lpass-rx-macro: Simplify PDS cleanup with devmKrzysztof Kozlowski
Eliminate PDS cleanup by using devm_add_action_or_reset() which results in one less error path and smaller cleanup in remove(). Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-1-6d98d4dd1ef5@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-03ASoC: tas5086: Convert to GPIO descriptorsLinus Walleij
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>
2024-07-03ASoC: wsa884x: Constify struct regmap_configJavier Carrasco
`wsa884x_regmap_config` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by: Javier Carrasco <javier.carrasco.cruz@gmail.com> Link: https://patch.msgid.link/20240703-sound-const-regmap_config-v1-9-2e379b0446a7@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-03ASoC: wsa883x: Constify struct regmap_configJavier Carrasco
`wsa883x_regmap_config` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by: Javier Carrasco <javier.carrasco.cruz@gmail.com> Link: https://patch.msgid.link/20240703-sound-const-regmap_config-v1-8-2e379b0446a7@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-03ASoC: wsa881x: Constify struct regmap_configJavier Carrasco
`wsa881x_regmap_config` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by: Javier Carrasco <javier.carrasco.cruz@gmail.com> Link: https://patch.msgid.link/20240703-sound-const-regmap_config-v1-7-2e379b0446a7@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-03ASoC: jz4770: Constify struct regmap_configJavier Carrasco
`jz4760_codec_regmap_config` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by: Javier Carrasco <javier.carrasco.cruz@gmail.com> Link: https://patch.msgid.link/20240703-sound-const-regmap_config-v1-6-2e379b0446a7@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-03ASoC: jz4760: Constify struct regmap_configJavier Carrasco
`jz4760_codec_regmap_config` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by: Javier Carrasco <javier.carrasco.cruz@gmail.com> Link: https://patch.msgid.link/20240703-sound-const-regmap_config-v1-5-2e379b0446a7@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-03ASoC: cs53l30: Constify struct regmap_configJavier Carrasco
`cs53l30_regmap` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by: Javier Carrasco <javier.carrasco.cruz@gmail.com> Link: https://patch.msgid.link/20240703-sound-const-regmap_config-v1-4-2e379b0446a7@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-03ASoC: cs35l36: Constify struct regmap_configJavier Carrasco
`cs35l36_regmap` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by: Javier Carrasco <javier.carrasco.cruz@gmail.com> Link: https://patch.msgid.link/20240703-sound-const-regmap_config-v1-3-2e379b0446a7@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-03ASoC: cs35l35: Constify struct regmap_configJavier Carrasco
`cs35l35_regmap` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by: Javier Carrasco <javier.carrasco.cruz@gmail.com> Link: https://patch.msgid.link/20240703-sound-const-regmap_config-v1-2-2e379b0446a7@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-03ASoC: cs35l34: Constify struct regmap_configJavier Carrasco
`cs35l34_regmap` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by: Javier Carrasco <javier.carrasco.cruz@gmail.com> Link: https://patch.msgid.link/20240703-sound-const-regmap_config-v1-1-2e379b0446a7@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-03ASoC: cs35l56: Set correct upper volume limitMark Brown
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.
2024-07-03ASoC: fsl_xcvr: Improve suspend/resume flow in fsl_xcvr_trigger()Chancel Liu
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>
2024-07-03ASoC: cs35l56: Limit Speaker Volume to +12dB maximumRichard Fitzgerald
Change CS35L56_MAIN_RENDER_USER_VOLUME_MAX to 48, to limit the maximum value of the Speaker Volume control to +12dB. The minimum value is unchanged so that the default 0dB has the same integer control value. The original maximum of 400 (+100dB) was the largest value that can be mathematically handled by the DSP. The actual maximum amplification is +12dB. Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Link: https://patch.msgid.link/20240703095517.208077-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org>
2024-07-03ASoC: cs35l56: Use header defines for Speaker Volume control definitionRichard Fitzgerald
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>
2024-07-02ASoC: tas2781: Use of_property_read_reg()Rob Herring (Arm)
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>