- 08 Jul, 2024 5 commits
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Elinor Montmasson authored
The imx-spdif machine driver creates audio card to directly use an S/PDIF device. However, it doesn't support interacting with an ASRC. fsl-asoc-card already has the support to create audio card which can use the ASRC. Merge the S/PDIF support from imx-spdif into driver fsl-asoc-card to extend the support of S/PDIF audio card with the use of ASRC devices. fsl-asoc-card uses slightly different DT properties than imx-spdif: * the "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an SPDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". To keep backward compatibility, support for "spdif-controller", "spdif-in" and "spdif-out" is also added to fsl-asoc-card. However, it is recommended to use the new properties if possible. It is better to declare transmitter and/or receiver in DT than using the dummy codec. DTs using compatible "fsl,imx-audio-spdif" are still compatible, and fsl-asoc-card will behave the same as imx-spdif for these DTs. Signed-off-by:
Elinor Montmasson <elinor.montmasson@savoirfairelinux.com> Link: https://patch.msgid.link/20240627083104.123357-5-elinor.montmasson@savoirfairelinux.comSigned-off-by:
Mark Brown <broonie@kernel.org>
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Elinor Montmasson authored
Adapt the driver to work with configurations using two codecs or more. Modify fsl_asoc_card_probe() to handle use cases where 2 codecs are given in the device tree. This will be needed to add support for the SPDIF. Use cases using one codec will ignore any given codecs other than the first. Co-developed-by:
Philip-Dylan Gleonec <philip-dylan.gleonec@savoirfairelinux.com> Signed-off-by:
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Elinor Montmasson authored
Add a second dai link component for codecs that will be used for use cases with 2 codecs. It is needed for future integration of the SPDIF support, which will use spdif_receiver and spdif_transmitter drivers. To prevent deferring in use cases using only one codec, also set by default the number of codecs to 1 for the relevant dai links. Co-developed-by:
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Elinor Montmasson authored
Add support for dai links using multiple codecs for multi-codec use cases. Co-developed-by:
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Shenghao Ding authored
Use get_unaligned_be16 instead of be16_to_cpup and get_unaligned_be32 instead of be32_to_cpup for potentially broken alignment. Signed-off-by:
Shenghao Ding <shenghao-ding@ti.com> Link: https://patch.msgid.link/20240707083011.98-1-shenghao-ding@ti.comSigned-off-by:
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- 05 Jul, 2024 1 commit
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Shenghao Ding authored
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:
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- 04 Jul, 2024 17 commits
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Mark Brown authored
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(). -
Mark Brown authored
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.
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Shenghao Ding authored
ASoc: tas2781: Set "Speaker Force Firmware Load" as the common kcontrol for both tas27871 and tas2563 Set "Speaker Force Firmware Load" as the common kcontrol for both tas27871 and tas2563 and move it into newly-created tasdevice_snd_controls, and keep the digital gain and analog gain in tas2781_snd_controls. Signed-off-by:
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Aleksandr Mishin authored
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: 02527c3f ("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.ruSigned-off-by:
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Krzysztof Kozlowski authored
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: 10f514bd ("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.orgSigned-off-by:
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Shengjiu Wang authored
commit c721f189 ("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.comSigned-off-by:
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Peter Ujfalusi authored
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.comSigned-off-by:
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Herve Codina authored
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.comSigned-off-by:
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Herve Codina authored
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:
Rob Herring (Arm) <robh@kernel.org> Link: https://patch.msgid.link/20240701113038.55144-10-herve.codina@bootlin.comSigned-off-by:
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Herve Codina authored
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:
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Herve Codina authored
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:
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Herve Codina authored
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:
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Herve Codina authored
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 authored
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:
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Herve Codina authored
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:
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Herve Codina authored
./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 authored
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: 075c7125 ("ASoC: fsl: Add support for QMC audio") Cc: stable@vger.kernel.org Signed-off-by:
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- 03 Jul, 2024 17 commits
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Krzysztof Kozlowski authored
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:
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Krzysztof Kozlowski authored
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:
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Krzysztof Kozlowski authored
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:
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Krzysztof Kozlowski authored
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:
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Krzysztof Kozlowski authored
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:
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Krzysztof Kozlowski authored
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:
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Linus Walleij authored
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.orgSigned-off-by:
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Javier Carrasco authored
`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.comSigned-off-by:
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Javier Carrasco authored
`wsa883x_regmap_config` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by:
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Javier Carrasco authored
`wsa881x_regmap_config` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by:
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Javier Carrasco authored
`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:
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Javier Carrasco authored
`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:
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Javier Carrasco authored
`cs53l30_regmap` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by:
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Javier Carrasco authored
`cs35l36_regmap` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by:
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Javier Carrasco authored
`cs35l35_regmap` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by:
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Javier Carrasco authored
`cs35l34_regmap` is not modified and can be declared as const to move its data to a read-only section. Signed-off-by:
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Mark Brown authored
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.
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