August 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: adau1761: Reset codec at probe and shutdown
by Niklas Carlsson 08 Aug '22

08 Aug '22

From: Niklas Carlsson <niklasc(a)axis.com> ADAU1X61 doesn't have a reset pin, which leads to the codec retaining the register values after the device has been rebooted without a power cycle. This causes the driver to be out of sync with the codec. Simulating a hardware reset ensures that the codec is in a defined state. Link: https://lore.kernel.org/all/alpine.DEB.2.02.1603291544160.3594@lnxricardw1.… Link: https://lore.kernel.org/all/alpine.DEB.2.02.1604211547540.31333@lnxricardw1… Signed-off-by: Niklas Carlsson <niklasc(a)axis.com> --- sound/soc/codecs/adau1761-i2c.c | 6 +++++ sound/soc/codecs/adau1761-spi.c | 6 +++++ sound/soc/codecs/adau1761.c | 43 +++++++++++++++++++++++++++++++++ sound/soc/codecs/adau1761.h | 1 + 4 files changed, 56 insertions(+) diff --git a/sound/soc/codecs/adau1761-i2c.c b/sound/soc/codecs/adau1761-i2c.c index 0683caf86aea..23bffbd74469 100644 --- a/sound/soc/codecs/adau1761-i2c.c +++ b/sound/soc/codecs/adau1761-i2c.c @@ -36,6 +36,11 @@ static int adau1761_i2c_remove(struct i2c_client *client) return 0; } +static void adau1761_i2c_shutdown(struct i2c_client *client) +{ + adau1761_shutdown(&client->dev); +} + static const struct i2c_device_id adau1761_i2c_ids[] = { { "adau1361", ADAU1361 }, { "adau1461", ADAU1761 }, @@ -63,6 +68,7 @@ static struct i2c_driver adau1761_i2c_driver = { }, .probe_new = adau1761_i2c_probe, .remove = adau1761_i2c_remove, + .shutdown = adau1761_i2c_shutdown, .id_table = adau1761_i2c_ids, }; module_i2c_driver(adau1761_i2c_driver); diff --git a/sound/soc/codecs/adau1761-spi.c b/sound/soc/codecs/adau1761-spi.c index 7c9242c2ff94..5bbbc3d3b5be 100644 --- a/sound/soc/codecs/adau1761-spi.c +++ b/sound/soc/codecs/adau1761-spi.c @@ -50,6 +50,11 @@ static void adau1761_spi_remove(struct spi_device *spi) adau17x1_remove(&spi->dev); } +static void adau1761_spi_shutdown(struct spi_device *spi) +{ + adau1761_shutdown(&spi->dev); +} + static const struct spi_device_id adau1761_spi_id[] = { { "adau1361", ADAU1361 }, { "adau1461", ADAU1761 }, @@ -77,6 +82,7 @@ static struct spi_driver adau1761_spi_driver = { }, .probe = adau1761_spi_probe, .remove = adau1761_spi_remove, + .shutdown = adau1761_spi_shutdown, .id_table = adau1761_spi_id, }; module_spi_driver(adau1761_spi_driver); diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 8f887227981f..ac544e1e8dc7 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -974,6 +974,39 @@ static struct snd_soc_dai_driver adau1761_dai_driver = { .ops = &adau17x1_dai_ops, }; +static void adau1761_reset(struct regmap *regmap) +{ + size_t i; + + /* + * Handle the lack of a reset pin in ADAU1X61 by manually writing all + * registers to their default values. + * + * Steps for performing the reset: + * 1) Make sure that the cache is marked as dirty by writing all + * default values directly to the cache. + * + * 2) Enable the core clock which is needed for writing all registers + * except CLOCK_CONTROL. + * + * 3) Use regcache_sync() for synchronizing the dirty cache back to + * the hardware. + */ + + regcache_cache_only(regmap, true); + for (i = 0; i < ARRAY_SIZE(adau1761_reg_defaults); i++) + regmap_write(regmap, adau1761_reg_defaults[i].reg, + adau1761_reg_defaults[i].def); + regcache_cache_only(regmap, false); + + regmap_update_bits(regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, + ADAU17X1_CLOCK_CONTROL_SYSCLK_EN); + regcache_sync(regmap); + regmap_update_bits(regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, 0); +} + int adau1761_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev)) { @@ -997,6 +1030,8 @@ int adau1761_probe(struct device *dev, struct regmap *regmap, if (ret) return ret; + adau1761_reset(regmap); + /* Enable cache only mode as we could miss writes before bias level * reaches standby and the core clock is enabled */ regcache_cache_only(regmap, true); @@ -1006,6 +1041,14 @@ int adau1761_probe(struct device *dev, struct regmap *regmap, } EXPORT_SYMBOL_GPL(adau1761_probe); +void adau1761_shutdown(struct device *dev) +{ + struct adau *adau = dev_get_drvdata(dev); + + adau1761_reset(adau->regmap); +} +EXPORT_SYMBOL_GPL(adau1761_shutdown); + const struct regmap_config adau1761_regmap_config = { .val_bits = 8, .reg_bits = 16, diff --git a/sound/soc/codecs/adau1761.h b/sound/soc/codecs/adau1761.h index 7beabf448ad1..98d698ebde27 100644 --- a/sound/soc/codecs/adau1761.h +++ b/sound/soc/codecs/adau1761.h @@ -16,6 +16,7 @@ struct device; int adau1761_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev)); +void adau1761_shutdown(struct device *dev); extern const struct regmap_config adau1761_regmap_config; -- 2.30.2

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[PATCH] ALSA: hda/conexant: Add quirk for LENOVO 20149 Notebook model
by Meng Tang 08 Aug '22

08 Aug '22

There is another LENOVO 20149 (Type1Sku0) Notebook model with CX20590, the device PCI SSID is 17aa:3977, which headphones are not responding, that requires the quirk CXT_PINCFG_LENOVO_NOTEBOOK. Add the corresponding entry to the quirk table. Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/pci/hda/patch_conexant.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 83ae21a01bbf..7b1a30a551f6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -222,6 +222,7 @@ enum { CXT_PINCFG_LEMOTE_A1205, CXT_PINCFG_COMPAQ_CQ60, CXT_FIXUP_STEREO_DMIC, + CXT_PINCFG_LENOVO_NOTEBOOK, CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, @@ -772,6 +773,14 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_stereo_dmic, }, + [CXT_PINCFG_LENOVO_NOTEBOOK] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x05d71030 }, + { } + }, + .chain_id = CXT_FIXUP_STEREO_DMIC, + }, [CXT_FIXUP_INC_MIC_BOOST] = { .type = HDA_FIXUP_FUNC, .v.func = cxt5066_increase_mic_boost, @@ -971,7 +980,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), - SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_PINCFG_LENOVO_NOTEBOOK), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo G50-70", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), -- 2.20.1

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[PATCH] ASoC: Intel: kbl_rt5663_rt5514_max98927: Simplify clk_get() usage
by Christophe JAILLET 07 Aug '22

07 Aug '22

If clk_get() returns -ENOENT, there is no need to defer the driver, -ENOENT will be returned the same for each retries. So, return the error code directly instead of -EPROBE_DEFER. Remove this special case and use dev_err_probe() to simplify code. It will also be less verbose if the clk is really deferred. Fixes: 47cbea216281 ("ASoC: Intel: eve: Enable mclk and ssp sclk early") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- This is based on my understanding of clk_get(). Review with care. Not sure the Fixes tag is needed. The patch does not fix anything. If devm_clk_get() returns -ENOENT, it will just loop several time until the framework gives up. If it returns -EPROBE_DEFER, this case is already handled by the "return ret;" So this patch should be a no-op, just a clean-up. --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 31 ++++--------------- 1 file changed, 6 insertions(+), 25 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 2c79fca57b19..e23001ca2d04 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -798,7 +798,6 @@ static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; struct snd_soc_acpi_mach *mach; - int ret; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -815,32 +814,14 @@ static int kabylake_audio_probe(struct platform_device *pdev) &constraints_dmic_2ch : &constraints_dmic_channels; ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); - if (IS_ERR(ctx->mclk)) { - ret = PTR_ERR(ctx->mclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_mclk, defer probe\n"); - return -EPROBE_DEFER; - } - - dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", - ret); - return ret; - } + if (IS_ERR(ctx->mclk)) + return dev_err_probe(&pdev->dev, PTR_ERR(ctx->mclk), + "Failed to get ssp1_mclk\n"); ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); - if (IS_ERR(ctx->sclk)) { - ret = PTR_ERR(ctx->sclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } - - dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", - ret); - return ret; - } + if (IS_ERR(ctx->sclk)) + return dev_err_probe(&pdev->dev, PTR_ERR(ctx->sclk), + "Failed to get ssp1_sclk\n"); return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.34.1

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[PATCH] ASoC: Intel: fix sof_es8336 probe
by Pierre-Louis Bossart 05 Aug '22

05 Aug '22

Changes to add HDMI capture support broke the machine driver probe for all other platforms. The commit listed in the Fixes tag added a board_id descriptor but didn't add the default name for the sof_essx8336 machine driver. Add the missing entry and remove the now-useless platform driver alias. Reported-by: Eugene J Markow <ejmarkow(a)yahoo.com> BugLink: https://github.com/thesofproject/linux/issues/3336 Fixes: 86b1959a2ccb ("ASoC: Intel: sof_es8336: add support for HDMI_In capture") Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_es8336.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index c7f33c89588e..606cc3242a60 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -759,6 +759,9 @@ static int sof_es8336_remove(struct platform_device *pdev) } static const struct platform_device_id board_ids[] = { + { + .name = "sof-essx8336", /* default quirk == 0 */ + }, { .name = "adl_es83x6_c1_h02", .driver_data = (kernel_ulong_t)(SOF_ES8336_SSP_CODEC(1) | @@ -786,5 +789,4 @@ module_platform_driver(sof_es8336_driver); MODULE_DESCRIPTION("ASoC Intel(R) SOF + ES8336 Machine driver"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:sof-essx8336"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); -- 2.34.1

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[PATCH resent] ASoC: DPCM: Don't pick up BE without substream
by Takashi Iwai 05 Aug '22

05 Aug '22

When DPCM tries to add valid BE connections at dpcm_add_paths(), it doesn't check whether the picked BE actually supports for the given stream direction. Due to that, when an asymmetric BE stream is present, it picks up wrongly and this may result in a NULL dereference at a later point where the code assumes the existence of a corresponding BE substream. This patch adds the check for the presence of the substream for the target BE for avoiding the problem above. Note that we have already some fix for non-existing BE substream at commit 6246f283d5e0 ("ASoC: dpcm: skip missing substream while applying symmetry"). But the code path we've hit recently is rather happening before the previous fix. So this patch tries to fix at picking up a BE instead of parsing BE lists. Fixes: bbf7d3b1c4f4 ("ASoC: soc-pcm: align BE 'atomicity' with that of the FE") Reported-by: Alex Natalsson <harmoniesworlds(a)gmail.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/CADs9LoPZH_D+eJ9qjTxSLE5jGyhKsjMN7g2NighZ16biVxsy… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Seems that the previous submission was overlooked since I forgot to put Cc to Mark. Resubmitted... sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a827cc3c158a..0c1de5624842 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1318,6 +1318,9 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; + if (!snd_soc_dpcm_get_substream(be, stream)) + continue; + for_each_rtd_dais(be, i, dai) { w = snd_soc_dai_get_widget(dai, stream); -- 2.35.3

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[PATCH 0/3] ASoC: Fix theoretical buffer overflow by snprintf()
by Takashi Iwai 05 Aug '22

05 Aug '22

Hi, this is a patch series to paper over the theoretical buffer overflow that might be caused by snprintf(). snprintf() is notorious for its behavior and the usage of a safer version, scnprintf(), is recommended. Takashi === Takashi Iwai (3): ASoC: Intel: avs: Fix potential buffer overflow by snprintf() ASoC: SOF: debug: Fix potential buffer overflow by snprintf() ASoC: SOF: Intel: hda: Fix potential buffer overflow by snprintf() sound/soc/intel/avs/pcm.c | 4 ++-- sound/soc/sof/debug.c | 6 +++--- sound/soc/sof/intel/hda.c | 2 +- 3 files changed, 6 insertions(+), 6 deletions(-) -- 2.35.3

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[PATCH resend] ASoC: SOF: ipc3-topology: Fix clang -Wformat warning
by Justin Stitt 05 Aug '22

05 Aug '22

When building with Clang we encounter these warnings: | sound/soc/sof/ipc3-topology.c:2343:4: error: format specifies type | 'unsigned char' but the argument has type 'int' [-Werror,-Wformat] | SOF_ABI_MAJOR, SOF_ABI_MINOR, SOF_ABI_PATCH); | ^~~~~~~~~~~~~~~^~~~~~~~~~~~~~~^~~~~~~~~~~~~ Use correct format specifier `%d` since args are of type int. Link: https://github.com/ClangBuiltLinux/linux/issues/378 Reported-by: Nathan Chancellor <nathan(a)kernel.org> Suggested-by: Nathan Chancellor <nathan(a)kernel.org> Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- Note: resending with the subject corrected, this patch fixes a build warning that will break the Clang build when CONFIG_WERROR=y. Original patch: https://lore.kernel.org/all/20220721211218.4039288-1-justinstitt@google.com/ Reported by Nathan here: https://lore.kernel.org/all/YtmrCJjQrSbv8Aj1@dev-arch.thelio-3990X/ sound/soc/sof/ipc3-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index b2cc046b9f60..65923e7a5976 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -2338,7 +2338,7 @@ static int sof_ipc3_parse_manifest(struct snd_soc_component *scomp, int index, } dev_info(scomp->dev, - "Topology: ABI %d:%d:%d Kernel ABI %hhu:%hhu:%hhu\n", + "Topology: ABI %d:%d:%d Kernel ABI %d:%d:%d\n", man->priv.data[0], man->priv.data[1], man->priv.data[2], SOF_ABI_MAJOR, SOF_ABI_MINOR, SOF_ABI_PATCH); -- 2.37.1.359.gd136c6c3e2-goog

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[PATCH v2] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path
by Biju Das 05 Aug '22

05 Aug '22

We usually do cleanup in reverse order of init. Currently in case of error rz_ssi_release_dma_channels() done in the reverse order. This patch improves error handling in rz_ssi_probe() error path. While at it, use "goto cleanup" style to reduce code duplication. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- v1->v2: Replaced the label err_dma->err_reset. --- sound/soc/sh/rz-ssi.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 0d0594a0e4f6..7ace0c0db5b1 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1017,32 +1017,36 @@ static int rz_ssi_probe(struct platform_device *pdev) ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL); if (IS_ERR(ssi->rstc)) { - rz_ssi_release_dma_channels(ssi); - return PTR_ERR(ssi->rstc); + ret = PTR_ERR(ssi->rstc); + goto err_reset; } reset_control_deassert(ssi->rstc); pm_runtime_enable(&pdev->dev); ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); - return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n"); + dev_err(&pdev->dev, "pm_runtime_resume_and_get failed\n"); + goto err_pm; } ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component, rz_ssi_soc_dai, ARRAY_SIZE(rz_ssi_soc_dai)); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - - pm_runtime_put(ssi->dev); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); dev_err(&pdev->dev, "failed to register snd component\n"); + goto err_snd_soc; } + return 0; + +err_snd_soc: + pm_runtime_put(ssi->dev); +err_pm: + pm_runtime_disable(ssi->dev); + reset_control_assert(ssi->rstc); +err_reset: + rz_ssi_release_dma_channels(ssi); + return ret; } -- 2.25.1

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[PATCH] ALSA: ice1712: remove redundant assignment to new
by Colin Ian King 05 Aug '22

05 Aug '22

The variable new is initialized with a value but it is never read. It is being re-assigned a new value in every case path in the following switch statement. The assignment is redundant and can be removed. Cleans up clang scan build warning: sound/pci/ice1712/quartet.c:569:8: warning: Although the value stored to 'new' is used in the enclosing expression, the value is never actually read from 'new' [deadcode.DeadStores] Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/pci/ice1712/quartet.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index 0dfa093f7dca..20b3e8f94719 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -566,7 +566,7 @@ static int qtet_ain12_sw_put(struct snd_kcontrol *kcontrol, { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned int old, new, tmp, masked_old; - old = new = get_scr(ice); + old = get_scr(ice); masked_old = old & (SCR_AIN12_SEL1 | SCR_AIN12_SEL0); tmp = ucontrol->value.integer.value[0]; if (tmp == 2) -- 2.35.3

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[PATCH v2] ALSA: hda/realtek: Add quirk for another Asus K42JZ model
by Meng Tang 05 Aug '22

05 Aug '22

There is another Asus K42JZ model with the PCI SSID 1043:1313 that requires the quirk ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE. Add the corresponding entry to the quirk table. Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 383a814b8539..4461237a3474 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6842,6 +6842,7 @@ enum { ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ASUS_ZENBOOK, ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A, + ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE, ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, @@ -7427,6 +7428,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269VB_FIXUP_ASUS_ZENBOOK, }, + [ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a110f0 }, /* use as headset mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC + }, [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, @@ -9124,6 +9134,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), -- 2.20.1

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