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[PATCH v2 1/1] ASoC: codecs: max98090: Allow dsp_a mode
by Maxim Kochetkov 23 Jun '23

23 Jun '23

TDM mode for max98090 is dsp_a compatible with such limitations: 1) Up to four timeslots supported. 2) Only 16 bits timeslots supported. 3) Only 2 active timeslots (L/R) supported. We want to setup TDM mode only when dsp_a mode is selected. So move M98090_REG_TDM_FORMAT/M98090_REG_TDM_CONTROL registers setup from max98090_set_tdm_slot() to the max98090_dai_set_fmt(). Also extend max98090_set_tdm_slot() with all TDM limitations check. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/codecs/max98090.c | 53 ++++++++++++++++++++----------------- sound/soc/codecs/max98090.h | 3 ++- 2 files changed, 30 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 7bc463910d4f..2adf744c6526 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1581,7 +1581,7 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_component *component = codec_dai->component; struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); struct max98090_cdata *cdata; - u8 regval; + u8 regval, tdm_regval; max98090->dai_fmt = fmt; cdata = &max98090->dai[0]; @@ -1590,6 +1590,7 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, cdata->fmt = fmt; regval = 0; + tdm_regval = 0; switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { case SND_SOC_DAIFMT_CBC_CFC: /* Set to consumer mode PLL - MAS mode off */ @@ -1635,7 +1636,8 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, regval |= M98090_RJ_MASK; break; case SND_SOC_DAIFMT_DSP_A: - /* Not supported mode */ + tdm_regval |= M98090_TDM_MASK; + break; default: dev_err(component->dev, "DAI format unsupported"); return -EINVAL; @@ -1664,11 +1666,20 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, * seen for the case of TDM mode. The remaining cases have * normal logic. */ - if (max98090->tdm_slots > 1) + if (tdm_regval) regval ^= M98090_BCI_MASK; snd_soc_component_write(component, M98090_REG_INTERFACE_FORMAT, regval); + + regval = 0; + if (tdm_regval) + regval = max98090->tdm_lslot << M98090_TDM_SLOTL_SHIFT | + max98090->tdm_rslot << M98090_TDM_SLOTR_SHIFT | + 0 << M98090_TDM_SLOTDLY_SHIFT; + + snd_soc_component_write(component, M98090_REG_TDM_FORMAT, regval); + snd_soc_component_write(component, M98090_REG_TDM_CONTROL, tdm_regval); } return 0; @@ -1679,33 +1690,22 @@ static int max98090_set_tdm_slot(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); - struct max98090_cdata *cdata; - cdata = &max98090->dai[0]; if (slots < 0 || slots > 4) return -EINVAL; - max98090->tdm_slots = slots; - max98090->tdm_width = slot_width; - - if (max98090->tdm_slots > 1) { - /* SLOTL SLOTR SLOTDLY */ - snd_soc_component_write(component, M98090_REG_TDM_FORMAT, - 0 << M98090_TDM_SLOTL_SHIFT | - 1 << M98090_TDM_SLOTR_SHIFT | - 0 << M98090_TDM_SLOTDLY_SHIFT); - - /* FSW TDM */ - snd_soc_component_update_bits(component, M98090_REG_TDM_CONTROL, - M98090_TDM_MASK, - M98090_TDM_MASK); - } + if (slot_width != 16) + return -EINVAL; - /* - * Normally advisable to set TDM first, but this permits either order - */ - cdata->fmt = 0; - max98090_dai_set_fmt(codec_dai, max98090->dai_fmt); + if (rx_mask != tx_mask) + return -EINVAL; + + if (!rx_mask) + return -EINVAL; + + max98090->tdm_slots = slots; + max98090->tdm_lslot = ffs(rx_mask) - 1; + max98090->tdm_rslot = fls(rx_mask) - 1; return 0; } @@ -2408,6 +2408,9 @@ static int max98090_probe(struct snd_soc_component *component) max98090->pa1en = 0; max98090->pa2en = 0; + max98090->tdm_lslot = 0; + max98090->tdm_rslot = 1; + ret = snd_soc_component_read(component, M98090_REG_REVISION_ID); if (ret < 0) { dev_err(component->dev, "Failed to read device revision: %d\n", diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index a197114b0dad..6ce8dd176e48 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1533,7 +1533,8 @@ struct max98090_priv { struct snd_soc_jack *jack; unsigned int dai_fmt; int tdm_slots; - int tdm_width; + int tdm_lslot; + int tdm_rslot; u8 lin_state; unsigned int pa1en; unsigned int pa2en; -- 2.40.1

2 1

[PATCH] ASoC: tas2781: fix Kconfig dependencies
by Arnd Bergmann 23 Jun '23

23 Jun '23

From: Arnd Bergmann <arnd(a)arndb.de> The new driver has two modules that both get enabled for build testing when all codecs are selected. The comlib part has an i2c dependency, so this remains disabled on builds without i2c, but then the other one fails to link: ERROR: modpost: "tasdevice_dev_bulk_write" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined! ERROR: modpost: "tasdevice_dev_update_bits" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined! ERROR: modpost: "tasdevice_dev_bulk_read" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined! ERROR: modpost: "tasdevice_dev_read" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined! ERROR: modpost: "tasdevice_dev_write" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined! There are many ways to address this, adding an explicit dependency seems to be the clearest method that keeps the structure of the driver otherwise unchanged. Fixes: ef3bcde75d06d ("ASoC: tas2781: Add tas2781 driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f481f02a35710..2a62dbd5339e4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1741,6 +1741,7 @@ config SND_SOC_TAS2781_COMLIB tristate config SND_SOC_TAS2781_FMWLIB + depends on SND_SOC_TAS2781_COMLIB tristate default n -- 2.39.2

2 1

Re: [PATCH 6/7] soundwire: intel: remove unnecessary init
by K Smart 23 Jun '23

23 Jun '23

REMOVE NOW

1 0

[PATCH] ASoC: qcom: common: add default jack dapm pins
by Srinivas Kandagatla 23 Jun '23

23 Jun '23

If the soundcard does not specify the dapm pins, let the common code add these pins for jack. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/common.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index c1f24af17506..555feb845c41 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -8,6 +8,11 @@ #include "qdsp6/q6afe.h" #include "common.h" +static const struct snd_soc_dapm_widget qcom_jack_snd_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + int qcom_snd_parse_of(struct snd_soc_card *card) { struct device_node *np; @@ -169,6 +174,11 @@ int qcom_snd_parse_of(struct snd_soc_card *card) of_node_put(platform); } + if (!card->dapm_widgets) { + card->dapm_widgets = qcom_jack_snd_widgets; + card->num_dapm_widgets = ARRAY_SIZE(qcom_jack_snd_widgets); + } + return 0; err: of_node_put(cpu); -- 2.21.0

3 2

[PATCH] ASoC: dt-bindings: microchip,sama7g5-pdmc: Simplify "microchip,mic-pos" constraints
by Rob Herring 23 Jun '23

23 Jun '23

"enum" values should be integers or strings, not arrays (though json-schema does allow arrays, we do not). In this case, all possible combinations are allowed anyways, so there's little point in expressing as an array. Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml index 9b40268537cb..9aa65c975c4e 100644 --- a/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml @@ -56,13 +56,9 @@ properties: items: items: - description: value for DS line + enum: [0, 1] - description: value for sampling edge - anyOf: - - enum: - - [0, 0] - - [0, 1] - - [1, 0] - - [1, 1] + enum: [0, 1] minItems: 1 maxItems: 4 uniqueItems: true -- 2.40.1

4 3

[PATCH 1/2] ASoC: fsl-asoc-card: add nau8822 support
by Hui Wang 22 Jun '23

22 Jun '23

This is for an imx6sx EVB which has a nau8822 codec connects to the SSI2 interface, so add the nau8822 support in this machine driver. Because the codec driver nau8822.c doesn't handle mclk enabling, here adding a codec_priv->mclk for nau8822 and similar codecs which need to enable the mclk in the machine driver, and enable the mclk in the card_late_probe() conditionally. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/fsl/fsl-asoc-card.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index d9a0d4768c4d..89504865cb1d 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -27,6 +27,7 @@ #include "../codecs/wm8960.h" #include "../codecs/wm8994.h" #include "../codecs/tlv320aic31xx.h" +#include "../codecs/nau8822.h" #define CS427x_SYSCLK_MCLK 0 @@ -45,6 +46,7 @@ * @pll_id: PLL id for set_pll() */ struct codec_priv { + struct clk *mclk; unsigned long mclk_freq; unsigned long free_freq; u32 mclk_id; @@ -522,6 +524,9 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) return ret; } + if (!IS_ERR_OR_NULL(codec_priv->mclk)) + clk_prepare_enable(codec_priv->mclk); + return 0; } @@ -682,6 +687,14 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.free_freq = priv->codec_priv.mclk_freq; priv->card.dapm_routes = NULL; priv->card.num_dapm_routes = 0; + } else if (of_device_is_compatible(np, "fsl,imx-audio-nau8822")) { + codec_dai_name = "nau8822-hifi"; + priv->codec_priv.mclk_id = NAU8822_CLK_MCLK; + priv->codec_priv.fll_id = NAU8822_CLK_PLL; + priv->codec_priv.pll_id = NAU8822_CLK_PLL; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + if (codec_dev) + priv->codec_priv.mclk = devm_clk_get(codec_dev, NULL); } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; @@ -907,6 +920,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-wm8524", }, { .compatible = "fsl,imx-audio-si476x", }, { .compatible = "fsl,imx-audio-wm8958", }, + { .compatible = "fsl,imx-audio-nau8822", }, {} }; MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); -- 2.25.1

2 2

[PATCH] ASoC: max98388: set variable soc_codec_dev_max98388 storage-class-specifier to static
by Tom Rix 22 Jun '23

22 Jun '23

smatch reports sound/soc/codecs/max98388.c:890:39: warning: symbol 'soc_codec_dev_max98388' was not declared. Should it be static? This variable is only used in its defining file, so it should be static. Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/soc/codecs/max98388.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98388.c b/sound/soc/codecs/max98388.c index 8062a7115007..b7d1d5c7b72f 100644 --- a/sound/soc/codecs/max98388.c +++ b/sound/soc/codecs/max98388.c @@ -887,7 +887,7 @@ static const struct regmap_config max98388_regmap = { .cache_type = REGCACHE_RBTREE, }; -const struct snd_soc_component_driver soc_codec_dev_max98388 = { +static const struct snd_soc_component_driver soc_codec_dev_max98388 = { .probe = max98388_probe, .controls = max98388_snd_controls, .num_controls = ARRAY_SIZE(max98388_snd_controls), -- 2.27.0

2 1

Intel/sof-essx8336: Fix HiFi.conf
by GitHub pull_request - opened 22 Jun '23

22 Jun '23

alsa-project/alsa-ucm-conf pull request #328 was opened from xry111: 1. Enable Left/Right Headphone Mixer Left/Right DAC Switch in BootSequence. Despite they seem enabled in sof-essx8336.conf, if I don't add this I'll need to enable them with alsamixer. I guess they are killed by disdevall. 2. Switch Headphone on/off in Headphones enable/disable sequence. This should be obvious. 3. "Headset Switch" is not recognized and triggers errors running "alsaucm -c hw:0 set _verb Hifi". Change it to "Headset Mic Switch". And in the disable sequence we should turn it off, not on. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/328 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/328.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v5 00/13] Add support for IIO devices in ASoC
by Herve Codina 22 Jun '23

22 Jun '23

Several weeks ago, I sent a series [1] for adding a potentiometer as an auxiliary device in ASoC. The feedback was that the potentiometer should be directly handled in IIO (as other potentiometers) and something more generic should be present in ASoC in order to have a binding to import some IIO devices into sound cards. The series related to the IIO potentiometer device is already applied. This series introduces audio-iio-aux. Its goal is to offer the binding between IIO and ASoC. It exposes attached IIO devices as ASoC auxiliary devices and allows to control them through mixer controls. On my system, the IIO device is a potentiometer and it is present in an amplifier design present in the audio path. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20230614074904.29085-1-herve.codina@bo… This v5 series mainly: - Fixes {min,max}_array macros Best regards, Hervé [1] https://lore.kernel.org/linux-kernel/20230203111422.142479-1-herve.codina@b… [2] https://lore.kernel.org/linux-kernel/20230421085245.302169-1-herve.codina@b… Changes v4 -> v5 - Patches 1, 2, 3, 4, 5, 9, 10, 11, 12, 13 No changes. - Patch 6 Fix commit log. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>' - Patch 7 Fix the macros to be able to use them with: - an array defined as int *buff; - an array defined as int buff[N]; - Rework the way to "unconstify" the temporary variable to avoid issues due to integer promotion. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>' - Patch 8 Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>' Changes v3 -> v4 - Patches 1, 2 No changes. - Patches 3, 4, 5 Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 6 (new in v4) Fix headers inclusion order. - Patch 7 (patch 6 in v3) Add a comment related to __must_be_array() Use __array[0] of *__array - Patch 8 (patch 7 in v3) Fix minmax.h inclusion order. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 9 (patch 8 in v3) Add 'Suggested-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>'. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 10 (patch 9 in v3) Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 11 (patch 10 in v3) Fix a typo. Add 'Suggested-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 12 (patch 11 in v3) Fix typos in the commit log. Fix headers inclusion order. Removed unneeded variable initialization. Replace {0} by {}. Use struct device *dev in probe(). Check an error on the snd-control-invert-range property read. - Patch 13 (patch12 in v3) No changes. Changes v2 -> v3 - Patches 1, 2 No changes. - Patch 3, 4 Add 'Acked-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>'. - Patch 5 (new in v3) Removed the 'unused' variable and check the null pointer when used. - Patch 6 (new in v3) Introduce {min,max}_array(). - Patch 7 (new in v3) Use max_array() in iio_channel_read_max(). - Patch 8 (new in v3) Replace a FIXME comment by a TODO one. - Patch 9 (patch 5 in v2) Removed the 'unused' variable and check the null pointer when used. Use min_array(). Remplace a FIXME comment by a TODO one. - Patch 10 (patch 6 in v2) Convert existing macros to return a compound litteral instead of adding a new helper. - Patch 11 (patch 7 in v2) Remove the file name from the C file header. Use directly converted DAPM macros. Replace <linux/module.h> by <linux/mod_devicetable.h>. Add <linux/platform_device.h>. Be sure that min <= max. Swap values if it is not the case. Move the bool structure member after the int ones. Remove unneeded assignements. Use dev_err_probe() when relevant. Use str_on_off(). Use static_assert() instead of BUILD_BUG_ON(). Remove unneeded comma and blank line. Use device_property_*() instead of the OF API. - patch 8 available in v2 removed as already applied - Patch 12 (patch 9 in v2) Use devm_add_action_or_reset(). Call simple_populate_aux() from simple_parse_of(). Changes v1 -> v2 - Patch 1 Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the /schemas/iio/iio-consumer.yaml reference Remove the unneeded '|' after description - Patch 2 (new in v2) Introduce the simple-audio-card additional-devs subnode - Patch 3 (new in v2) Check err before switch() in iio_channel_read_max() - Patch 4 (new in v2) Fix raw reads and raw writes documentation - Patch 5 (patch 2 in v1) Check err before switch() in iio_channel_read_min() Fix documentation - Patch 6 (path 3 in v1) No changes - Patch 7 (patch 4 in v1) Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the mask usage from audio_iio_aux_{get,put}_volsw helpers Use directly PTR_ERR() in dev_err_probe() parameter Remove the '!!' construction Remove of_match_ptr() - Patch 8 (new in v2) Add a missing of_node_put() in the simple-card driver - Patch 9 (new in v2) Handle additional-devs in the simple-card driver Herve Codina (13): ASoC: dt-bindings: Add audio-iio-aux ASoC: dt-bindings: simple-card: Add additional-devs subnode iio: inkern: Check error explicitly in iio_channel_read_max() iio: consumer.h: Fix raw values documentation notes iio: inkern: Remove the 'unused' variable usage in iio_channel_read_max() iio: inkern: Fix headers inclusion order minmax: Introduce {min,max}_array() iio: inkern: Use max_array() to get the maximum value from an array iio: inkern: Replace a FIXME comment by a TODO one iio: inkern: Add a helper to query an available minimum raw value ASoC: soc-dapm.h: Convert macros to return a compound literal ASoC: codecs: Add support for the generic IIO auxiliary devices ASoC: simple-card: Handle additional devices .../bindings/sound/audio-iio-aux.yaml | 64 ++++ .../bindings/sound/simple-card.yaml | 53 +++ drivers/iio/inkern.c | 86 ++++- include/linux/iio/consumer.h | 37 +- include/linux/minmax.h | 64 ++++ include/sound/soc-dapm.h | 138 ++++--- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/audio-iio-aux.c | 338 ++++++++++++++++++ sound/soc/generic/simple-card.c | 46 ++- 10 files changed, 769 insertions(+), 71 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-iio-aux.yaml create mode 100644 sound/soc/codecs/audio-iio-aux.c -- 2.40.1

6 29

[PATCH] ASoC: dt-bindings: nau8821: Convert to dtschema
by Seven Lee 22 Jun '23

22 Jun '23

Convert the NAU8821 audio CODEC bindings to DT schema. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8821.txt | 55 -------- .../bindings/sound/nuvoton,nau8821.yaml | 120 ++++++++++++++++++ 2 files changed, 120 insertions(+), 55 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8821.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8821.txt b/Documentation/devicetree/bindings/sound/nau8821.txt deleted file mode 100644 index 7c84e7c7327a..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8821.txt +++ /dev/null @@ -1,55 +0,0 @@ -Nuvoton NAU88L21 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8821" - - - reg : the I2C address of the device. This is either 0x1B (CSB=0) or 0x54 (CSB=1). - -Optional properties: - - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. - - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, - otherwise pin in high impedance state. - - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. - - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. - - - nuvoton,vref-impedance: VREF Impedance selection - 0 - Open - 1 - 25 kOhm - 2 - 125 kOhm - 3 - 2.5 kOhm - - - nuvoton,micbias-voltage: Micbias voltage level. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - - nuvoton,dmic-clk-threshold: the ADC threshold of DMIC clock. - - nuvoton,key_enable: Headset button detection switch. - -Example: - - headset: nau8821@1b { - compatible = "nuvoton,nau8821"; - reg = <0x1b>; - interrupt-parent = <&gpio>; - interrupts = <23 IRQ_TYPE_LEVEL_LOW>; - nuvoton,jkdet-enable; - nuvoton,jkdet-pull-enable; - nuvoton,jkdet-pull-up; - nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - nuvoton,jack-insert-debounce = <7>; - nuvoton,jack-eject-debounce = <7>; - nuvoton,dmic-clk-threshold = 3072000; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml new file mode 100644 index 000000000000..728223d20876 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -0,0 +1,120 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8821.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU88L21 audio codec + +maintainers: + - Seven Lee <wtli(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8821 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-enable: + description: Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: Pull-up JKDET pin. If set then JKDET pin is pull up, + otherwise pull down. + type: boolean + + nuvoton,jkdet-polarity: + description: JKDET pin polarity. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,micbias-voltage: + description: MICBIAS output level select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # VDDA + - 1 # VDDA * 1 + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,vref-impedance: + description: VMID Tie-off impedance select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # open(default) + - 1 # 25KOhms + - 2 # 125KOhms + - 3 # 2.5KOhms + default: 2 + + nuvoton,jack-insert-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 0 + + nuvoton,dmic-clk-threshold: + description: DMIC clock speed expected value. Unit is Hz. + $ref: /schemas/types.yaml#/definitions/uint32 + default: 3072000 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1b { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8821"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <23 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,micbias-voltage = <6>; + nuvoton,vref-impedance = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <0>; + nuvoton,dmic-clk-threshold = <3072000>; + }; + }; -- 2.25.1

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