- 05 Jan, 2022 1 commit
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David Rhodes authored
Add IDs for the CS35L51/53 variants, the functionality is shared with CS35L41. Signed-off-by: David Rhodes <david.rhodes@cirrus.com> Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20220105113026.18955-2-ckeepax@opensource.cirrus.comSigned-off-by: Mark Brown <broonie@kernel.org>
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- 04 Jan, 2022 1 commit
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Alyssa Ross authored
modprobe can't handle spaces in aliases. Fixes: 9e28f653 ("ASoC: fsl_mqs: Add MQS component driver") Signed-off-by: Alyssa Ross <hi@alyssa.is> Link: https://lore.kernel.org/r/20220104132218.1690103-1-hi@alyssa.isSigned-off-by: Mark Brown <broonie@kernel.org>
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- 31 Dec, 2021 11 commits
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Mark Brown authored
Merge series from Trevor Wu <trevor.wu@mediatek.com>: This series of patches repairs some problems for pcmif BE dai. The unexpected control flow is corrected, and the missing playback support of DPCM is added.
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Mark Brown authored
Merge series from Lucas Tanure <tanureal@opensource.cirrus.com>: Initial refactoring to support use of cs35l41 from the HDA driver stack.
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Mark Brown authored
So we can send to Linus.
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Lucas Tanure authored
ASoC and HDA will use the same registers to configure internal boost for the device Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com> Link: https://lore.kernel.org/r/20211217115708.882525-7-tanureal@opensource.cirrus.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Lucas Tanure authored
ASoC and HDA will use the same register to set channels for the device Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com> Link: https://lore.kernel.org/r/20211217115708.882525-6-tanureal@opensource.cirrus.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Lucas Tanure authored
ASoC and HDA systems require the same errata patches, so move it to the shared code using a function the correctly applies the patches by revision Also, move CS35L41_DSP1_CCM_CORE_CTRL write to errata patch function as is required to be written at boot, but not in regmap_register_patch sequence as will affect waking up from hibernation Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com> Link: https://lore.kernel.org/r/20211217115708.882525-5-tanureal@opensource.cirrus.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Lucas Tanure authored
ASoC and HDA systems for all revisions of CS35L41 will benefit from having this initialization, so add it to reg_sequence of each revision By moving to reg_sequence all gains are set to zero. And boost, monitoring parts, and class D amplifier are disabled. Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com> Link: https://lore.kernel.org/r/20211217115708.882525-4-tanureal@opensource.cirrus.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Lucas Tanure authored
ASoC and HDA will do the same cs35l41_otp_unpack, so move it to shared code Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com> Link: https://lore.kernel.org/r/20211217115708.882525-3-tanureal@opensource.cirrus.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Lucas Tanure authored
To support CS35L41 in HDA systems the HDA driver for CS35L41 would have to duplicate some functions that already exist on ASoC driver So instead of duplicate the code, use the new lib source as a shared resource for both ASoC and HDA Also, change the way CONFIG_SND_SOC_CS35L41 is selected, as reported by Intel Kernel test robot, it is possible to build SND_SOC_CS35L41_SPI/I2C without the main driver, which would lead to build failures. Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com> Reported-by: kernel test robot <lkp@intel.com> Link: https://lore.kernel.org/r/20211217115708.882525-2-tanureal@opensource.cirrus.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Trevor Wu authored
PCM1_BE should be a dai_link for both playback and capture. In the patch, the missing DPCM playback support is added. Signed-off-by: Trevor Wu <trevor.wu@mediatek.com> Link: https://lore.kernel.org/r/20211230084731.31372-3-trevor.wu@mediatek.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Trevor Wu authored
Originally, the conditions for preventing reentry are not correct. dai->component->active is not the state specifically for pcmif dai, so it is not a correct condition to indicate the status of pcmif dai. On the other hand, snd_soc_dai_stream_actvie() in prepare ops for both playback and capture possibly return true at the first entry when these two streams are opened at the same time. In the patch, I refer to the implementation in mt8192-dai-pcm.c. Clock and enabling bit for PCMIF are managed by DAPM, and the condition for prepare ops is replaced by the status of dai widget. Fixes: 1f95c019 ("ASoC: mediatek: mt8195: support pcm in platform driver") Signed-off-by: Trevor Wu <trevor.wu@mediatek.com> Link: https://lore.kernel.org/r/20211230084731.31372-2-trevor.wu@mediatek.comSigned-off-by: Mark Brown <broonie@kernel.org>
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- 29 Dec, 2021 5 commits
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Derek Fang authored
The mclk might not be registered as a fixed clk name "mclk" on some platforms. In those platforms, if the mclk needed to be controlled by codec driver and acquired by a fixed name, it would be a problem. This patch to fix the issue that wclk becomes an orphan due to the fixed mclk's name. Signed-off-by: Derek Fang <derek.fang@realtek.com> Link: https://lore.kernel.org/r/20211227055446.27563-1-derek.fang@realtek.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Trevor Wu authored
Playback pop is observed and the root cause is the reference clock provided by MT8195 is diabled before RT5682 finishes the control flow. To ensure the reference clock supplied to RT5682 is disabled after RT5682 finishes all register controls. We replace BCLK with MCLK for RT5682 reference clock, and makes use of set_bias_level_post to handle MCLK which guarantees MCLK is off after all RT5682 register access. Signed-off-by: Trevor Wu <trevor.wu@mediatek.com> Reviewed-by: Tzung-Bi Shih <tzungbi@google.com> Link: https://lore.kernel.org/r/20211228064821.27865-1-trevor.wu@mediatek.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Jiasheng Jiang authored
Because of the potential failure of the ioremap(), the buf->area could be NULL. Therefore, we need to check it and return -ENOMEM in order to transfer the error. Fixes: f09aecd5 ("ASoC: SAMSUNG: Add I2S0 internal dma driver") Signed-off-by: Jiasheng Jiang <jiasheng@iscas.ac.cn> Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@canonical.com> Link: https://lore.kernel.org/r/20211228034026.1659385-1-jiasheng@iscas.ac.cnSigned-off-by: Mark Brown <broonie@kernel.org>
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Tzung-Bi Shih authored
Uses of_device_get_match_data() helper to clean some boilerplate code. Signed-off-by: Tzung-Bi Shih <tzungbi@google.com> Link: https://lore.kernel.org/r/20211227062153.3887447-1-tzungbi@google.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Fabio Estevam authored
The Chip ID - Register 01h contains the following description as per the CS4265 datasheet: "Bits 7 through 4 are the part number ID, which is 1101b (0Dh)" The current error message is incorrect as it prints CS4265_CHIP_ID, which is the register number, instead of printing the expected part number ID value. To make it clearer, also do a shift by 4, so that the error message would become: [ 4.218083] cs4265 1-004f: CS4265 Part Number ID: 0x0 Expected: 0xd Signed-off-by: Fabio Estevam <festevam@denx.de> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20211222141920.1482451-1-festevam@gmail.comSigned-off-by: Mark Brown <broonie@kernel.org>
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- 24 Dec, 2021 15 commits
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Mark Brown authored
Merge series from Bard Liao <yung-chuan.liao@linux.intel.com>: This series contains three topics. 1. SoundWire: Intel: remove pdm support 2. ASoC/SoundWire: dai: expand 'stream' concept beyond SoundWire 3. ASoC/SOF/SoundWire: fix suspend-resume on pause with dynamic pipelines The topics are independent but the changes are dependent. So please allow me to send them in one series.
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V sujith kumar Reddy authored
Configure the speaker gpio pin based on power sequence of the DAPM speaker events. Enable speaker after widget power up and Disable before widget powerdown. Signed-off-by: V sujith kumar Reddy <vsujithkumar.reddy@amd.com> Link: https://lore.kernel.org/r/20211224150058.2444776-1-vsujithkumar.reddy@amd.comSigned-off-by: Mark Brown <broonie@kernel.org>
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chiminghao authored
return value form directly instead of taking this in another redundant variable. Reported-by: Zeal Robot <zealci@zte.com.cm> Signed-off-by: chiminghao <chi.minghao@zte.com.cn> Link: https://lore.kernel.org/r/20211209015707.409870-1-chi.minghao@zte.com.cnSigned-off-by: Mark Brown <broonie@kernel.org>
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Jiasheng Jiang authored
Yes, you are right and now the return code depending on the init_clks(). Fixes: 6078c651 ("soc: mediatek: Refine scpsys to support multiple platform") Signed-off-by: Jiasheng Jiang <jiasheng@iscas.ac.cn> Link: https://lore.kernel.org/r/20211222015157.1025853-1-jiasheng@iscas.ac.cnSigned-off-by: Mark Brown <broonie@kernel.org>
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Tzung-Bi Shih authored
Device nodes can be released after components have bound. Shortens the lifecycle of the device nodes. Releases the reference counts after snd_soc_register_card. Signed-off-by: Tzung-Bi Shih <tzungbi@google.com> Link: https://lore.kernel.org/r/20211224064719.2031210-5-tzungbi@google.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Tzung-Bi Shih authored
Eliminates error messages if snd_soc_register_card() failed. Kernel emits messages if device probe error anyway. Signed-off-by: Tzung-Bi Shih <tzungbi@google.com> Link: https://lore.kernel.org/r/20211224064719.2031210-4-tzungbi@google.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Tzung-Bi Shih authored
Fixes the device_node leak. Signed-off-by: Tzung-Bi Shih <tzungbi@google.com> Link: https://lore.kernel.org/r/20211224064719.2031210-3-tzungbi@google.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Tzung-Bi Shih authored
Fixes the device_node leak. Signed-off-by: Tzung-Bi Shih <tzungbi@google.com> Link: https://lore.kernel.org/r/20211224064719.2031210-2-tzungbi@google.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Pierre-Louis Bossart authored
While the hardware supports PDM streams, this capability has never been tested or enabled on any product, so this is dead-code. Let's remove all this. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Acked-By: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20211224021034.26635-8-yung-chuan.liao@linux.intel.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Pierre-Louis Bossart authored
cppcheck warning: drivers/soundwire/intel.c:1487:10: style: Variable 'ret' is assigned a value that is never used. [unreadVariable] int ret = 0; ^ Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Acked-By: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20211224021034.26635-7-yung-chuan.liao@linux.intel.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Pierre-Louis Bossart authored
Overloading the tx_mask with a linear value is asking for trouble and only works because the codec_dai hw_params() is called before the cpu_dai hw_params(). Move to the more generic set_stream() API to pass the hdac_stream information. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Link: https://lore.kernel.org/r/20211224021034.26635-6-yung-chuan.liao@linux.intel.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Pierre-Louis Bossart authored
The HDAudio ASoC support relies on the set_tdm_slots() helper to store the HDaudio stream tag in the tx_mask. This only works because of the pre-existing order in soc-pcm.c, where the hw_params() is handled for codec_dais *before* cpu_dais. When the order is reversed, the stream_tag is used as a mask in the codec fixup functions: /* fixup params based on TDM slot masks */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && codec_dai->tx_mask) soc_pcm_codec_params_fixup(&codec_params, codec_dai->tx_mask); As a result of this confusion, the codec_params_fixup() ends-up generating bad channel masks, depending on what stream_tag was allocated. We could add a flag to state that the tx_mask is really not a mask, but it would be quite ugly to persist in overloading concepts. Instead, this patch suggests a more generic get/set 'stream' API based on the existing model for SoundWire. We can expand the concept to store 'stream' opaque information that is specific to different DAI types. In the case of HDAudio DAIs, we only need to store a stream tag as an unsigned char pointer. The TDM rx_ and tx_masks should really only be used to store masks. Rename get_sdw_stream/set_sdw_stream callbacks and helpers as get_stream/set_stream. No functionality change beyond the rename. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Acked-By: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20211224021034.26635-5-yung-chuan.liao@linux.intel.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Ranjani Sridharan authored
This patch provides both a simplification of the suspend flows and a better balanced operation during suspend/resume transition, as part of the transition of Sound Open Firmware (SOF) to dynamic pipelines: the DSP resources are only enabled when required instead of enabled on startup. The exiting code relies on a convoluted way of dealing with suspend signals. Since there is no .suspend DAI callback, we used the component .suspend and marked all the component DAI dmas as 'suspended'. The information was used in the .prepare stage to differentiate resume operations from xrun handling, and only reinitialize SHIM registers and DMA in the former case. While this solution has been working reliably for about 2 years, there is a much better solution consisting in trapping the TRIGGER_SUSPEND in the .trigger DAI ops. The DMA is still marked in the same way for the .prepare op to run, but in addition the callbacks sent to DSP firmware are now balanced. Normal operation: hw_params -> intel_params_stream hw_free -> intel_free_stream suspend -> intel_free_stream prepare -> intel_params_stream This balanced operation was not required with existing SOF firmware relying on static pipelines instantiated at every boot. With the on-going transition to dynamic pipelines, it's however a requirement to keep the use count for the DAI widget balanced across all transitions. The component suspend is not removed but instead modified to deal with a corner case: when a substream is PAUSED, the ALSA core does not throw the TRIGGER_SUSPEND. This is problematic since the refcount for all pipelines and widgets is not balanced, leading to issues on resume. The trigger callback keeps track of the 'paused' state with a new flag, which is tested during the component suspend called later to release the remaining DSP resources. These resources will be re-enabled in the .prepare step. The IPC used in the TRIGGER_SUSPEND to release DSP resources is not a problem since the BE dailink is already marked as non-atomic. Co-developed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Acked-By: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20211224021034.26635-4-yung-chuan.liao@linux.intel.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Pierre-Louis Bossart authored
We don't really need to pass a substream to the callback, we only need the direction. No functionality change, only simplification to enable improve suspend with paused streams. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Acked-By: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20211224021034.26635-3-yung-chuan.liao@linux.intel.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Pierre-Louis Bossart authored
We have a helper, use it to simplify widget lookup Suggested-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Link: https://lore.kernel.org/r/20211224021034.26635-2-yung-chuan.liao@linux.intel.comSigned-off-by: Mark Brown <broonie@kernel.org>
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- 23 Dec, 2021 7 commits
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Mark Brown authored
Merge series from Stephan Gerhold <stephan@gerhold.net>: Some sound card setups might require extra pin switches to allow turning off certain audio components. simple-card supports this already using the "pin-switches" and "widgets" device tree property. This series makes it possible to use the same properties for the Qcom sound cards. To implement that, the function that parses the "pin-switches" property in simple-card-utils.c is first moved into the ASoC core. Then two simple function calls are added to the common Qcom sound card DT parser. Finally there is a small patch for the msm8916-wcd-analog codec to make it possible to model sound card setups used in some MSM8916 smartphones. (See PATCH 2/4 for an explanation of some real example use cases.) Using pin switches rather than patching codec drivers with switches was originally suggested by Mark Brown on a patch for the tfa989x codec: https://lore.kernel.org/alsa-devel/YXaMVHo9drCIuD3u@sirena.org.uk/
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Stephan Gerhold authored
The analog codec has separate output paths for the left headphone channel (HPH_L) and the right headphone channel (HPH_R). While they are usually used together for actual headphones output, some devices also have an analog speaker amplifier connected to one of the headphone channels. To allow modelling that properly (and to avoid powering on the unneeded output path), HPH_L and HPH_R should be represented by separate outputs rather than a shared HEADPHONE output that always activates both paths. Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Stephan Gerhold <stephan@gerhold.net> Link: https://lore.kernel.org/r/20211214142049.20422-5-stephan@gerhold.netSigned-off-by: Mark Brown <broonie@kernel.org>
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Stephan Gerhold authored
Use the DT helpers in the ASoC core to parse the "pin-switches" and "widgets" properties from the device tree. This allows adding extra mixers to disable e.g. an extra speaker amplifier that would be normally powered on automatically because it is connected to a shared output pin. Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Stephan Gerhold <stephan@gerhold.net> Link: https://lore.kernel.org/r/20211214142049.20422-4-stephan@gerhold.netSigned-off-by: Mark Brown <broonie@kernel.org>
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Stephan Gerhold authored
Some sound card setups might require extra pin switches to allow turning off certain audio components. There are two real examples for this in smartphones/tablets based on MSM8916: 1. Analog speaker amplifiers connected to headphone outputs. The MSM8916 analog codec does not have a separate "Line Out" port so some devices have an analog speaker amplifier connected to one of the headphone outputs. A pin switch is necessary to allow playback on headphones without also activating the speaker. 2. External speaker codec also used as earpiece. Some smartphones have two front-facing (stereo) speakers that can be also configured to act as an earpiece during voice calls. A pin switch is needed to allow disabling the second speaker during voice calls. There are existing bindings that allow setting up such pin switches in simple-card.yaml. Document the same for Qcom sound cards. One variant of example 1 above is added to the examples in the DT schema: There is an analog speaker amplifier connected to the HPH_R (right headphone channel) output. Adding a "Speaker" pin switch and widget allows turning off the speaker when audio should be only played via the connected headphones. Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Stephan Gerhold <stephan@gerhold.net> Acked-by: Rob Herring <robh@kernel.org> Link: https://lore.kernel.org/r/20211214142049.20422-3-stephan@gerhold.netSigned-off-by: Mark Brown <broonie@kernel.org>
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Stephan Gerhold authored
The ASoC core already has several helpers to parse card properties from the device tree. Move the parsing code for "pin-switches" from simple-card-utils to a shared snd_soc_of_parse_pin_switches() function so other drivers can also use it to set up pin switches configured in the device tree. Cc: Paul Cercueil <paul@crapouillou.net> Signed-off-by: Stephan Gerhold <stephan@gerhold.net> Link: https://lore.kernel.org/r/20211214142049.20422-2-stephan@gerhold.netSigned-off-by: Mark Brown <broonie@kernel.org>
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Tzung-Bi Shih authored
The of_parse_phandle() document: >>> Use of_node_put() on it when done. The driver didn't call of_node_put(). Fixes the leak. Signed-off-by: Tzung-Bi Shih <tzungbi@google.com> Link: https://lore.kernel.org/r/20211214040028.2992627-1-tzungbi@google.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Mark Brown authored
Merge series from Alexander Stein <alexander.stein@mailbox.org>: Following up [1] here are more fix for missing sound-name-prefix properties in the arch/arm64/boot/dts/amlogic/ subtree. [1] https://www.spinics.net/lists/devicetree/msg466125.html
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