December 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda/realtek: Use ALC285_FIXUP_HP_GPIO_LED on another HP laptop
by Kai-Heng Feng 25 Dec '21

25 Dec '21

The audio mute and mic mute LEDs don't work, so use the quirk to make them work. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 28255e752c4a1..344741869281f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8730,6 +8730,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x88d0, "HP Pavilion 15-eh1xxx (mainboard 88D0)", ALC287_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x89c3, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 2.33.1

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[PATCH] ALSA: hda/realtek: Add speaker fixup for some Yoga 15ITL5 devices
by arsgeiger＠gmail.com 25 Dec '21

25 Dec '21

From: Arie Geiger <arsgeiger(a)gmail.com> This patch adds another possible subsystem ID for the ALC287 used by the Lenovo Yoga 15ITL5. It uses the same initalization as the others. This patch has been tested and works for my device. Signed-off-by: Arie Geiger <arsgeiger(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3599f4c85ebf..188cbcde6e49 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8901,6 +8901,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x384a, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), -- 2.34.1

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[PATCH 1/2] ALSA: hda: Add AlderLake-N PCI ID
by Kai Vehmanen 25 Dec '21

25 Dec '21

Add HD Audio PCI ID for Intel AlderLake-N. Add rules to snd_intel_dsp_find_config() to choose DSP-based SOF driver for ADL-N systems with PCH-DMIC or Soundwire codecs, and plain HDA driver for the rest (DSP not used). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index b5bfa94bcb89..e035cecb37f7 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -374,6 +374,10 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x51cc, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x54c8, + }, #endif }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b98d7975adea..cdd4a58244d7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2496,6 +2496,9 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-M */ { PCI_DEVICE(0x8086, 0x51cc), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Alderlake-N */ + { PCI_DEVICE(0x8086, 0x54c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, base-commit: 74a50bd2899fdc9b6b2f2eb9624551e4aa795ad8 -- 2.34.1

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[PATCH] ALSA: hda/hdmi: Disable silent stream on GLK
by Ville Syrjala 25 Dec '21

25 Dec '21

From: Ville Syrjälä <ville.syrjala(a)linux.intel.com> The silent stream stuff recurses back into i915 audio component .get_power() from the .pin_eld_notify() hook. On GLK this will deadlock as i915 may already be holding the relevant modeset locks during .pin_eld_notify() and the GLK audio vs. CDCLK workaround will try to grab the same locks from .get_power(). Until someone comes up with a better fix just disable the silent stream support on GLK. Cc: stable(a)vger.kernel.org Cc: Harsha Priya <harshapriya.n(a)intel.com> Cc: Emmanuel Jillela <emmanuel.jillela(a)intel.com> Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: Takashi Iwai <tiwai(a)suse.de> Closes: https://gitlab.freedesktop.org/drm/intel/-/issues/2623 Fixes: 951894cf30f4 ("ALSA: hda/hdmi: Add Intel silent stream support") Signed-off-by: Ville Syrjälä <ville.syrjala(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 415701bd10ac..ffcde7409d2a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2947,7 +2947,8 @@ static int parse_intel_hdmi(struct hda_codec *codec) /* Intel Haswell and onwards; audio component with eld notifier */ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, - const int *port_map, int port_num, int dev_num) + const int *port_map, int port_num, int dev_num, + bool send_silent_stream) { struct hdmi_spec *spec; int err; @@ -2980,7 +2981,7 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, * Enable silent stream feature, if it is enabled via * module param or Kconfig option */ - if (enable_silent_stream) + if (send_silent_stream) spec->send_silent_stream = true; return parse_intel_hdmi(codec); @@ -2988,12 +2989,18 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, static int patch_i915_hsw_hdmi(struct hda_codec *codec) { - return intel_hsw_common_init(codec, 0x08, NULL, 0, 3); + return intel_hsw_common_init(codec, 0x08, NULL, 0, 3, + enable_silent_stream); } static int patch_i915_glk_hdmi(struct hda_codec *codec) { - return intel_hsw_common_init(codec, 0x0b, NULL, 0, 3); + /* + * Silent stream calls audio component .get_power() from + * .pin_eld_notify(). On GLK this will deadlock in i915 due + * to the audio vs. CDCLK workaround. + */ + return intel_hsw_common_init(codec, 0x0b, NULL, 0, 3, false); } static int patch_i915_icl_hdmi(struct hda_codec *codec) @@ -3004,7 +3011,8 @@ static int patch_i915_icl_hdmi(struct hda_codec *codec) */ static const int map[] = {0x0, 0x4, 0x6, 0x8, 0xa, 0xb}; - return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 3); + return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 3, + enable_silent_stream); } static int patch_i915_tgl_hdmi(struct hda_codec *codec) @@ -3016,7 +3024,8 @@ static int patch_i915_tgl_hdmi(struct hda_codec *codec) static const int map[] = {0x4, 0x6, 0x8, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf}; int ret; - ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 4); + ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 4, + enable_silent_stream); if (!ret) { struct hdmi_spec *spec = codec->spec; -- 2.32.0

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Re: [PATCH] ALSA: hda: use swap() to make code cleaner
by Takashi Iwai 25 Dec '21

25 Dec '21

On Sat, 18 Dec 2021 02:54:16 +0100, davidcomponentone(a)gmail.com wrote: > > From: Yang Guang <yang.guang5(a)zte.com.cn> > > Use the macro 'swap()' defined in 'include/linux/minmax.h' to avoid > opencoding it. > > Reported-by: Zeal Robot <zealci(a)zte.com.cn> > Signed-off-by: David Yang <davidcomponentone(a)gmail.com> > Signed-off-by: Yang Guang <yang.guang5(a)zte.com.cn> Applied now. Thanks. Takashi

1 0

[PATCH v1 0/2] kselftest: alsa: Small enhancements to mixer-test
by Mark Brown 25 Dec '21

25 Dec '21

These two patches improve the mixer test, checking that the default values for enums are valid and extending them to cover all the values for multi-value controls, not just the first one. It also factors out the validation that values are OK for controls for future reuse. Mark Brown (2): kselftest: alsa: Factor out check that values meet constraints kselftest: alsa: Validate values read from enumerations tools/testing/selftests/alsa/mixer-test.c | 158 ++++++++++++++-------- 1 file changed, 99 insertions(+), 59 deletions(-) base-commit: b73dad806533cad55df41a9c0349969b56d4ff7f -- 2.30.2

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[PATCH 0/2] ALSA/ASoC: hdac_ext cleanups
by Pierre-Louis Bossart 25 Dec '21

25 Dec '21

One function moved since it doesn't use the hdac_ext parts, and renaming of variables to identify when hdac_stream and hdac_ext_streams are used. No functionality change. Pierre-Louis Bossart (2): ALSA/ASoC: hda: move/rename snd_hdac_ext_stop_streams to hdac_stream.c ALSA: HDA: hdac_ext_stream: use consistent prefixes for variables include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 27 ++-- sound/hda/ext/hdac_ext_stream.c | 216 +++++++++++++++----------------- sound/hda/hdac_stream.c | 16 +++ sound/soc/intel/skylake/skl.c | 4 +- 5 files changed, 132 insertions(+), 132 deletions(-) -- 2.25.1

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[PATCH v2 0/7] ASoC/SoundWire: improve suspend flows and use set_stream() instead of set_tdm_slots() for HDAudio
by Bard Liao 24 Dec '21

24 Dec '21

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. The code is basically not changed, so I still take Vinod Koul's Ack. v2: - ASoC/SoundWire: dai: expand 'stream' concept beyond SoundWire: rename .set_sdw_stream to .set_stream and .get_sdw_stream to .get_stream in intel_pdm_dai_ops. Pierre-Louis Bossart (6): ASOC: SOF: Intel: use snd_soc_dai_get_widget() ASoC/soundwire: intel: simplify callbacks for params/hw_free ASoC/SoundWire: dai: expand 'stream' concept beyond SoundWire ASoC: Intel/SOF: use set_stream() instead of set_tdm_slots() for HDAudio soundwire: intel: remove unnecessary init soundwire: intel: remove PDM support Ranjani Sridharan (1): soundwire: intel: improve suspend flows drivers/soundwire/cadence_master.c | 36 +--- drivers/soundwire/cadence_master.h | 14 +- drivers/soundwire/intel.c | 253 ++++++++++++++-------------- drivers/soundwire/qcom.c | 8 +- drivers/soundwire/stream.c | 4 +- include/linux/soundwire/sdw_intel.h | 4 +- include/sound/soc-dai.h | 32 ++-- sound/soc/codecs/hdac_hda.c | 22 +-- sound/soc/codecs/max98373-sdw.c | 2 +- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt1316-sdw.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt700.c | 2 +- sound/soc/codecs/rt711-sdca.c | 2 +- sound/soc/codecs/rt711.c | 2 +- sound/soc/codecs/rt715-sdca.c | 2 +- sound/soc/codecs/rt715.c | 2 +- sound/soc/codecs/sdw-mockup.c | 2 +- sound/soc/codecs/wcd938x.c | 2 +- sound/soc/codecs/wsa881x.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 6 +- sound/soc/intel/skylake/skl-pcm.c | 7 +- sound/soc/qcom/sdm845.c | 4 +- sound/soc/qcom/sm8250.c | 4 +- sound/soc/sof/intel/hda-dai.c | 7 +- sound/soc/sof/intel/hda.c | 12 +- 26 files changed, 190 insertions(+), 247 deletions(-) -- 2.17.1

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[PATCH 0/4] ASoC: mediatek: fix device_node leak
by Tzung-Bi Shih 24 Dec '21

24 Dec '21

The series is a follow up series. The 1st and 2nd patch follow [1] to fix the device_node leak. The 3rd patch follows [2] to reduce the log verbosity. The 4th patch cleans the device_node related code on MT8195. In order to align to previous platforms. [1]: https://patchwork.kernel.org/project/alsa-devel/patch/20211214040028.299262… [2]: https://patchwork.kernel.org/project/alsa-devel/patch/20211220093408.207206… Tzung-Bi Shih (4): ASoC: mediatek: mt8173: fix device_node leak ASoC: mediatek: mt8183: fix device_node leak ASoC: mediatek: mt8173: reduce log verbosity in probe() ASoC: mediatek: mt8195: release device_node after snd_soc_register_card sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 5 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 5 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 6 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 7 +- .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 53 +++++---------- .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 64 ++++++------------- 8 files changed, 56 insertions(+), 95 deletions(-) -- 2.34.1.448.ga2b2bfdf31-goog

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[PATCH] ASoC: amd: acp: Power on/off the speaker enable gpio pin based on DAPM callback.
by V sujith kumar Reddy 24 Dec '21

24 Dec '21

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(a)amd.com> --- sound/soc/amd/acp/Kconfig | 6 +++--- sound/soc/amd/acp/acp-legacy-mach.c | 19 ++++++++++++++++--- sound/soc/amd/acp/acp-mach-common.c | 25 +++++++++++++++++++++++++ sound/soc/amd/acp/acp-mach.h | 10 +++++++++- sound/soc/amd/acp/acp-sof-mach.c | 21 ++++++++++++++++++--- 5 files changed, 71 insertions(+), 10 deletions(-) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 154be5e70821..d5838df3064b 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -32,7 +32,7 @@ config SND_AMD_ASOC_RENOIR config SND_SOC_AMD_MACH_COMMON tristate - depends on X86 && PCI && I2C + depends on X86 && PCI && I2C && GPIOLIB select CLK_FIXED_FCH select SND_SOC_RT5682_I2C select SND_SOC_DMIC @@ -44,14 +44,14 @@ config SND_SOC_AMD_MACH_COMMON config SND_SOC_AMD_LEGACY_MACH tristate "AMD Legacy Machine Driver Support" - depends on X86 && PCI && I2C + depends on X86 && PCI && I2C && GPIOLIB select SND_SOC_AMD_MACH_COMMON help This option enables legacy sound card support for ACP audio. config SND_SOC_AMD_SOF_MACH tristate "AMD SOF Machine Driver Support" - depends on X86 && PCI && I2C + depends on X86 && PCI && I2C && GPIOLIB select SND_SOC_AMD_MACH_COMMON help This option enables SOF sound card support for ACP audio. diff --git a/sound/soc/amd/acp/acp-legacy-mach.c b/sound/soc/amd/acp/acp-legacy-mach.c index de0f8024e2fb..0ad1cf41b308 100644 --- a/sound/soc/amd/acp/acp-legacy-mach.c +++ b/sound/soc/amd/acp/acp-legacy-mach.c @@ -27,6 +27,7 @@ static struct acp_card_drvdata rt5682_rt1019_data = { .hs_codec_id = RT5682, .amp_codec_id = RT1019, .dmic_codec_id = NONE, + .gpio_spkr_en = EN_SPKR_GPIO_GB, }; static const struct snd_kcontrol_new acp_controls[] = { @@ -41,15 +42,16 @@ static const struct snd_kcontrol_new acp_controls[] = { static const struct snd_soc_dapm_widget acp_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_SPK("Spk", event_spkr_handler), + SND_SOC_DAPM_SPK("Left Spk", event_spkr_handler), + SND_SOC_DAPM_SPK("Right Spk", event_spkr_handler), }; static int acp_asoc_probe(struct platform_device *pdev) { struct snd_soc_card *card = NULL; struct device *dev = &pdev->dev; + unsigned int spkr_gpio; int ret; if (!pdev->id_entry) @@ -67,9 +69,20 @@ static int acp_asoc_probe(struct platform_device *pdev) card->controls = acp_controls; card->num_controls = ARRAY_SIZE(acp_controls); card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data; + spkr_gpio = ((struct acp_card_drvdata *)(card->drvdata))->gpio_spkr_en; acp_legacy_dai_links_create(card); + if (gpio_is_valid(spkr_gpio)) { + ret = devm_gpio_request(dev, spkr_gpio, "spkren"); + if (ret) { + dev_err(dev, "(%s) gpio request failed: %d\n", + __func__, ret); + return ret; + } + gpio_direction_output(spkr_gpio, 0); + } + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 7785f12aa006..03d8d1af14b3 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -71,6 +71,31 @@ static const struct snd_soc_dapm_route rt5682_map[] = { { "IN1P", NULL, "Headset Mic" }, }; +int event_spkr_handler(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct acp_card_drvdata *drvdata = snd_soc_card_get_drvdata(card); + + if (!gpio_is_valid(drvdata->gpio_spkr_en)) + return 0; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + gpio_set_value(drvdata->gpio_spkr_en, 1); + break; + case SND_SOC_DAPM_PRE_PMD: + gpio_set_value(drvdata->gpio_spkr_en, 0); + break; + default: + dev_warn(card->dev, "%s invalid setting\n", __func__); + break; + } + return 0; +} +EXPORT_SYMBOL_NS_GPL(event_spkr_handler, SND_SOC_AMD_MACH); + /* Define card ops for RT5682 CODEC */ static int acp_card_rt5682_init(struct snd_soc_pcm_runtime *rtd) { diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index 5dc47cfbff10..fd6299844ebe 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -17,6 +17,12 @@ #include <linux/input.h> #include <linux/module.h> #include <sound/soc.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> + +#define EN_SPKR_GPIO_GB 0x11F +#define EN_SPKR_GPIO_NK 0x146 +#define EN_SPKR_GPIO_NONE -EINVAL enum be_id { HEADSET_BE_ID = 0, @@ -49,9 +55,11 @@ struct acp_card_drvdata { unsigned int dai_fmt; struct clk *wclk; struct clk *bclk; + unsigned int gpio_spkr_en; }; int acp_sofdsp_dai_links_create(struct snd_soc_card *card); int acp_legacy_dai_links_create(struct snd_soc_card *card); - +int event_spkr_handler(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event); #endif diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index 854eb7214cea..07de46142655 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -27,6 +27,7 @@ static struct acp_card_drvdata sof_rt5682_rt1019_data = { .hs_codec_id = RT5682, .amp_codec_id = RT1019, .dmic_codec_id = DMIC, + .gpio_spkr_en = EN_SPKR_GPIO_GB, }; static struct acp_card_drvdata sof_rt5682_max_data = { @@ -36,6 +37,7 @@ static struct acp_card_drvdata sof_rt5682_max_data = { .hs_codec_id = RT5682, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static struct acp_card_drvdata sof_rt5682s_max_data = { @@ -45,6 +47,7 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .hs_codec_id = RT5682S, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static const struct snd_kcontrol_new acp_controls[] = { @@ -58,15 +61,16 @@ static const struct snd_kcontrol_new acp_controls[] = { static const struct snd_soc_dapm_widget acp_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_SPK("Spk", event_spkr_handler), + SND_SOC_DAPM_SPK("Left Spk", event_spkr_handler), + SND_SOC_DAPM_SPK("Right Spk", event_spkr_handler), }; static int acp_sof_probe(struct platform_device *pdev) { struct snd_soc_card *card = NULL; struct device *dev = &pdev->dev; + unsigned int spkr_gpio; int ret; if (!pdev->id_entry) @@ -84,9 +88,20 @@ static int acp_sof_probe(struct platform_device *pdev) card->controls = acp_controls; card->num_controls = ARRAY_SIZE(acp_controls); card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data; + spkr_gpio = ((struct acp_card_drvdata *)(card->drvdata))->gpio_spkr_en; acp_sofdsp_dai_links_create(card); + if (gpio_is_valid(spkr_gpio)) { + ret = devm_gpio_request(dev, spkr_gpio, "spkren"); + if (ret) { + dev_err(dev, "(%s) gpio request failed: %d\n", + __func__, ret); + return ret; + } + gpio_direction_output(spkr_gpio, 0); + } + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, -- 2.25.1

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