September 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/6] ASoC: rt5640: Fix various IRQ handling issues
by Hans de Goede 13 Sep '23

13 Sep '23

Hi All, The recent(ish) rt5640 changes to add HDA header jack-detect support and the related suspend/resume handling fixes have introduced several issues with IRQ handling on boards not using the HDA header jack-detect support. This series fixes these issues, see the individual commit messages for details. Regards, Hans Hans de Goede (6): ASoC: rt5640: Revert "Fix sleep in atomic context" ASoC: rt5640: Fix sleep in atomic context ASoC: rt5640: Do not disable/enable IRQ twice on suspend/resume ASoC: rt5640: Enable the IRQ on resume after configuring jack-detect ASoC: rt5640: Fix IRQ not being free-ed for HDA jack detect mode ASoC: rt5640: Only cancel jack-detect work on suspend if active sound/soc/codecs/rt5640.c | 29 ++++++++++++----------------- 1 file changed, 12 insertions(+), 17 deletions(-) -- 2.41.0

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AIF2 ADC Stereo Capture Route is unitialised in alsa-ucm-conf
by GitHub issues - edited 12 Sep '23

12 Sep '23

alsa-project/alsa-ucm-conf issue #351 was edited from broukema: DESCRIPTION: `AIF2 ADC Stereo Capture Route` is not initialised in `alsa-ucm-conf`. This implies that if its setting is significant, any accidental change by the user (using any of the various `alsa` tools) to a wrong setting will not be corrected by any of the three `PinePhone/*.conf` files, leading, in particular, to audio errors during a phone call. This was discovered by testing variations on the Mobian version of the PinePhone alsa-ucm-conf files. CODING ARGUMENT: All parameters that can significantly affect the audio system should be either initialised or overridden in configuration files that the user expects to necessarily work, such as those for phone calls. In `alsa-ucm-conf` commit b68aa52 (one beyond v1.2.10): ```` $ grep -ri "AIF2 ADC Stereo Capture" . $ ```` shows that `AIF2 ADC Stereo Capture Route` is not set for any device at all (PinePhone or other). HARDWARE ARGUMENT: For the PinePhone (PP), Megi recommends in https://xnux.eu/devices/feature/audio-pp.html that `AIF2 ADC Stereo Capture Route` is set to `Mix Mono`. TESTING ARGUMENT: Experimentation with different stored alsa settings (from ```` sudo alsactl store -f - > YYYYMMDD_example_alsa_settings ```` ) showed that settings files whose only substantial difference was in the setting of `AIF2 ADC Stereo Capture Route` had the effect that audio from the PP was heard on the remote phone when the value was `Mix Mono` and was not heard with the value `Stereo`. Testing was done with the equivalent of https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… with the settings file https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… that has the `Mix Mono` value selected. RECOMMENDATIONS: * `AIF2 ADC Stereo Capture Route` should be initialised or set to *something* in one of the .conf files (in the PP case, most likely in `VoiceCall.conf`) to prevent users who experiment and have difficulty remembering which of the 50 PP alsa parameters should be set in which way from accidentally sabotaging their otherwise working audio calls; * Based on Megi's 2020 page and my testing, it seems that the correct value should be `Mix Mono`. COMMENT: While setting all 50 of the alsa settings ```` $ sudo alsactl store -f - |grep "name " |wc 50 258 1788 ```` in the .conf files is most likely unnecessary, there could be others that are necessary too. Ideally, someone who really understands how the PP audio system works (including Megi's two diagrams at https://xnux.eu/devices/feature/audio-pp.html) should be able to make these recommendations. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/351 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Bug: AIF2 ADC Stereo Capture Route is unitialised in alsa-ucm-conf
by GitHub issues - opened 12 Sep '23

12 Sep '23

alsa-project/alsa-ucm-conf issue #351 was opened from broukema: DESCRIPTION: `AIF2 ADC Stereo Capture Route` is not initialised in `alsa-ucm-conf`. This implies that if its setting is significant, any accidental change by the user (using any of the various `alsa` tools) to a wrong setting will not be corrected by any of the three `PinePhone/*.conf` files, leading, in particular, to audio errors during a phone call. This was discovered by testing variations on the Mobian version of the PinePhone alsa-ucm-conf files. CODING ARGUMENT: All parameters that can significantly affect the audio system should be either initialised or overridden in configuration files that the user expects to necessarily work, such as those for phone calls. In `alsa-ucm-conf` commit b68aa52 (one beyond v1.2.10): ```` $ grep -ri "AIF2 ADC Stereo Capture" . $ ```` shows that `AIF2 ADC Stereo Capture Route` is not set for any device at all (PinePhone or other). HARDWARE ARGUMENT: For the PinePhone (PP), Megi recommends in https://xnux.eu/devices/feature/audio-pp.html that `AIF2 ADC Stereo Capture Route` is set to `Mix Mono`. TESTING ARGUMENT: Experimentation with different stored alsa settings (from ```` sudo alsactl store -f - > YYYYMMDD_example_alsa_settings ```` ) showed that settings files whose only substantial difference was in the setting of `AIF2 ADC Stereo Capture Route` had the effect that audio from the PP was heard on the remote phone when the value was `Mix Mono` and was not heard with the value `Stereo`. Testing was done with the equivalent of https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… with the settings file https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… that has the `Mix Mono` value selected. RECOMMENDATIONS: * `AIF2 ADC Stereo Capture Route` should be initialised or set to *something* in one of the .conf files (in the PP case, most likely in `VoiceCall.conf`) to prevent users who experiment and have difficulty remembering which of the 50 PP alsa parameters should be set in which way from accidentally sabotaging their otherwise working audio calls; * Based on Megi's 2020 page and my testing, it seems that the correct value should be `Mix Mono`. COMMENT: While setting all 50 of the alsa settings ```` $ sudo alsactl store -f - |grep "name " |wc 50 258 1788 ```` in the .conf files is most likely unnecessary, there could be others that are necessary too. Ideally, someone who really understands how the PP audio system works (including Megi's two diagrams at https://xnux.eu/devices/feature/audio-pp.html) should be able to make these recommendations. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/351 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Remove optional dependency on GTK 2
by GitHub issues - opened 12 Sep '23

12 Sep '23

alsa-project/alsa-tools issue #18 was opened from ChaseKnowlden: GTK 2 is not maintained since December 2020. Issue URL : https://github.com/alsa-project/alsa-tools/issues/18 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH] ASoC: rsnd: remove unneeded of_node_put()
by Kuninori Morimoto 12 Sep '23

12 Sep '23

The loop is not using "node", of_node_put(node) is not needed. Cc: Julia Lawall <julia.lawall(a)inria.fr> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e29c2fee95219..7552fa0a2578d 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1302,7 +1302,6 @@ static int rsnd_dai_of_node(struct rsnd_priv *priv, int *is_graph) i++; if (i >= RSND_MAX_COMPONENT) { dev_info(dev, "reach to max component\n"); - of_node_put(node); break; } } -- 2.25.1

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[PATCH 0/2] SM6115 TX Macro
by Konrad Dybcio 12 Sep '23

12 Sep '23

Like most Qualcomm SoCs, SM6115 has a TX Macro. Only some minor changes were required. Signed-off-by: Konrad Dybcio <konrad.dybcio(a)linaro.org> --- Konrad Dybcio (2): ASoC: dt-bindings: qcom,lpass-tx-macro: Add SM6115 ASoC: codecs: lpass-tx-macro: Add SM6115 support .../bindings/sound/qcom,lpass-tx-macro.yaml | 18 ++++++++++++++++++ sound/soc/codecs/lpass-macro-common.h | 2 ++ sound/soc/codecs/lpass-tx-macro.c | 22 +++++++++++++++------- 3 files changed, 35 insertions(+), 7 deletions(-) --- base-commit: 6269320850097903b30be8f07a5c61d9f7592393 change-id: 20230825-topic-6115tx-b523a0209639 Best regards, -- Konrad Dybcio <konrad.dybcio(a)linaro.org>

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[tiwai-sound:test/format-truncation-fixes 20/23] sound/pci/cmipci.c:3106:35: warning: '%d' directive output may be truncated writing between 1 and 11 bytes into a region of size 8
by kernel test robot 12 Sep '23

12 Sep '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/format-truncation-fixes head: 5eb64dfc2dc9c05e50f2f122d77b28cede293b6b commit: 1c87c035c832a55414b42288df76bb10a5b11b34 [20/23] ALSA: cmipci: Fix -Wformat-truncation warning config: alpha-allyesconfig (https://download.01.org/0day-ci/archive/20230913/202309130112.VUtCZ4wu-lkp@…) compiler: alpha-linux-gcc (GCC) 13.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20230913/202309130112.VUtCZ4wu-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202309130112.VUtCZ4wu-lkp@intel.com/ All warnings (new ones prefixed by >>): sound/pci/cmipci.c: In function 'snd_cmipci_create': >> sound/pci/cmipci.c:3106:35: warning: '%d' directive output may be truncated writing between 1 and 11 bytes into a region of size 8 [-Wformat-truncation=] 3106 | " (model %d)", cm->chip_version); | ^~ sound/pci/cmipci.c:3106:26: note: directive argument in the range [-2147483648, 67] 3106 | " (model %d)", cm->chip_version); | ^~~~~~~~~~~~~ sound/pci/cmipci.c:3105:17: note: 'snprintf' output between 11 and 21 bytes into a destination of size 16 3105 | snprintf(modelstr, sizeof(modelstr), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 3106 | " (model %d)", cm->chip_version); | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vim +3106 sound/pci/cmipci.c 3041 3042 /* initialize codec registers */ 3043 snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_RESET); 3044 snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_RESET); 3045 snd_cmipci_write(cm, CM_REG_INT_HLDCLR, 0); /* disable ints */ 3046 snd_cmipci_ch_reset(cm, CM_CH_PLAY); 3047 snd_cmipci_ch_reset(cm, CM_CH_CAPT); 3048 snd_cmipci_write(cm, CM_REG_FUNCTRL0, 0); /* disable channels */ 3049 snd_cmipci_write(cm, CM_REG_FUNCTRL1, 0); 3050 3051 snd_cmipci_write(cm, CM_REG_CHFORMAT, 0); 3052 snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_ENDBDAC|CM_N4SPK3D); 3053 #if CM_CH_PLAY == 1 3054 snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC); 3055 #else 3056 snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC); 3057 #endif 3058 if (cm->chip_version) { 3059 snd_cmipci_write_b(cm, CM_REG_EXT_MISC, 0x20); /* magic */ 3060 snd_cmipci_write_b(cm, CM_REG_EXT_MISC + 1, 0x09); /* more magic */ 3061 } 3062 /* Set Bus Master Request */ 3063 snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_BREQ); 3064 3065 /* Assume TX and compatible chip set (Autodetection required for VX chip sets) */ 3066 switch (pci->device) { 3067 case PCI_DEVICE_ID_CMEDIA_CM8738: 3068 case PCI_DEVICE_ID_CMEDIA_CM8738B: 3069 if (!pci_dev_present(intel_82437vx)) 3070 snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_TXVX); 3071 break; 3072 default: 3073 break; 3074 } 3075 3076 if (cm->chip_version < 68) { 3077 val = pci->device < 0x110 ? 8338 : 8738; 3078 } else { 3079 switch (snd_cmipci_read_b(cm, CM_REG_INT_HLDCLR + 3) & 0x03) { 3080 case 0: 3081 val = 8769; 3082 break; 3083 case 2: 3084 val = 8762; 3085 break; 3086 default: 3087 switch ((pci->subsystem_vendor << 16) | 3088 pci->subsystem_device) { 3089 case 0x13f69761: 3090 case 0x584d3741: 3091 case 0x584d3751: 3092 case 0x584d3761: 3093 case 0x584d3771: 3094 case 0x72848384: 3095 val = 8770; 3096 break; 3097 default: 3098 val = 8768; 3099 break; 3100 } 3101 } 3102 } 3103 sprintf(card->shortname, "C-Media CMI%d", val); 3104 if (cm->chip_version < 68) 3105 snprintf(modelstr, sizeof(modelstr), > 3106 " (model %d)", cm->chip_version); 3107 else 3108 modelstr[0] = '\0'; 3109 scnprintf(card->longname, sizeof(card->longname), 3110 "%s%s at %#lx, irq %i", 3111 card->shortname, modelstr, cm->iobase, cm->irq); 3112 3113 if (cm->chip_version >= 39) { 3114 val = snd_cmipci_read_b(cm, CM_REG_MPU_PCI + 1); 3115 if (val != 0x00 && val != 0xff) { 3116 if (mpu_port[dev]) 3117 iomidi = cm->iobase + CM_REG_MPU_PCI; 3118 integrated_midi = 1; 3119 } 3120 } 3121 if (!integrated_midi) { 3122 val = 0; 3123 iomidi = mpu_port[dev]; 3124 switch (iomidi) { 3125 case 0x320: val = CM_VMPU_320; break; 3126 case 0x310: val = CM_VMPU_310; break; 3127 case 0x300: val = CM_VMPU_300; break; 3128 case 0x330: val = CM_VMPU_330; break; 3129 default: 3130 iomidi = 0; break; 3131 } 3132 if (iomidi > 0) { 3133 snd_cmipci_write(cm, CM_REG_LEGACY_CTRL, val); 3134 /* enable UART */ 3135 snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_UART_EN); 3136 if (inb(iomidi + 1) == 0xff) { 3137 dev_err(cm->card->dev, 3138 "cannot enable MPU-401 port at %#lx\n", 3139 iomidi); 3140 snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, 3141 CM_UART_EN); 3142 iomidi = 0; 3143 } 3144 } 3145 } 3146 3147 if (cm->chip_version < 68) { 3148 err = snd_cmipci_create_fm(cm, fm_port[dev]); 3149 if (err < 0) 3150 return err; 3151 } 3152 3153 /* reset mixer */ 3154 snd_cmipci_mixer_write(cm, 0, 0); 3155 3156 snd_cmipci_proc_init(cm); 3157 3158 /* create pcm devices */ 3159 pcm_index = pcm_spdif_index = 0; 3160 err = snd_cmipci_pcm_new(cm, pcm_index); 3161 if (err < 0) 3162 return err; 3163 pcm_index++; 3164 err = snd_cmipci_pcm2_new(cm, pcm_index); 3165 if (err < 0) 3166 return err; 3167 pcm_index++; 3168 if (cm->can_ac3_hw || cm->can_ac3_sw) { 3169 pcm_spdif_index = pcm_index; 3170 err = snd_cmipci_pcm_spdif_new(cm, pcm_index); 3171 if (err < 0) 3172 return err; 3173 } 3174 3175 /* create mixer interface & switches */ 3176 err = snd_cmipci_mixer_new(cm, pcm_spdif_index); 3177 if (err < 0) 3178 return err; 3179 3180 if (iomidi > 0) { 3181 err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI, 3182 iomidi, 3183 (integrated_midi ? 3184 MPU401_INFO_INTEGRATED : 0) | 3185 MPU401_INFO_IRQ_HOOK, 3186 -1, &cm->rmidi); 3187 if (err < 0) 3188 dev_err(cm->card->dev, 3189 "no UART401 device at 0x%lx\n", iomidi); 3190 } 3191 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

topology: pre-processor: Add support for enum controls
by GitHub pull_request - opened 12 Sep '23

12 Sep '23

alsa-project/alsa-utils pull request #236 was opened from ranj063: Add support for adding enum controls in the topology pre-processor. Request URL : https://github.com/alsa-project/alsa-utils/pull/236 Patch URL : https://github.com/alsa-project/alsa-utils/pull/236.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v1] ALSA: hda/tas2781: Support ACPI_ID both TXNW2781 and TIAS2781
by Shenghao Ding 12 Sep '23

12 Sep '23

Support ACPI_ID both TXNW2781 and TIAS2781, TXNW2781 is the only one legal ACPI ID, TIAS2781 is the widely-used ACPI ID named by our customers, so far it is not registered. We have discussed this with them, they requested TIAS2781 must be supported for the laptops already released to market, their new laptops will switch to TXNW2781. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Changes in v1: - Add TXNW2781 into tas2781_acpi_hda_match and move it to the top - Redefine tas2781_generic_fixup, remove hid param - TIAS2781 has been used by our customers, see following dstd.dsl. We have discussed this with them, they requested TIAS2781 must be supported for the laptops already released to market, their new laptops will switch to TXNW2781 Name (_HID, "TIAS2781") // _HID: Hardware ID Name (_UID, Zero) // _UID: Unique ID Method (_SUB, 0, NotSerialized) // _SUB: Subsystem ID { If ((SPID == Zero)) { Return ("17AA3886") } If ((SPID == One)) { Return ("17AA3884") } } - Add TXNW2781 support in comp_match_tas2781_dev_name --- sound/pci/hda/patch_realtek.c | 36 ++++++++++++++++++--------------- sound/pci/hda/tas2781_hda_i2c.c | 33 ++++++++++++++++++------------ 2 files changed, 40 insertions(+), 29 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b7e78bfcff..6dae58a8ef 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6770,24 +6770,35 @@ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) return !strcmp(d + n, tmp); } +/* TIAS2781 is the unofficial ACPI id, but widely used in current devices. + * TXNW2781 is the official ACPI id, and will be used in the new devices. + * Check TIAS2781 or TXNW2781 + */ static int comp_match_tas2781_dev_name(struct device *dev, void *data) { - struct scodec_dev_name *p = data; + const char c[][10] = { "TXNW2781", "TIAS2781" }; const char *d = dev_name(dev); - int n = strlen(p->bus); + const char *bus = data; + int n = strlen(bus), i; char tmp[32]; /* check the bus name */ - if (strncmp(d, p->bus, n)) + if (strncmp(d, bus, n)) return 0; /* skip the bus number */ if (isdigit(d[n])) n++; - /* the rest must be exact matching */ - snprintf(tmp, sizeof(tmp), "-%s:00", p->hid); - return !strcmp(d + n, tmp); + for (i = 0; i < ARRAY_SIZE(c); i++) { + /* the rest must be exact matching */ + snprintf(tmp, sizeof(tmp), "-%s:00", c[i]); + + if (!strcmp(d + n, tmp)) + return 1; + } + + return 0; } static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char *bus, @@ -6824,24 +6835,17 @@ static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char } static void tas2781_generic_fixup(struct hda_codec *cdc, int action, - const char *bus, const char *hid) + const char *bus) { struct device *dev = hda_codec_dev(cdc); struct alc_spec *spec = cdc->spec; - struct scodec_dev_name *rec; int ret; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: - rec = devm_kmalloc(dev, sizeof(*rec), GFP_KERNEL); - if (!rec) - return; - rec->bus = bus; - rec->hid = hid; - rec->index = 0; spec->comps[0].codec = cdc; component_match_add(dev, &spec->match, - comp_match_tas2781_dev_name, rec); + comp_match_tas2781_dev_name, (void *)bus); ret = component_master_add_with_match(dev, &comp_master_ops, spec->match); if (ret) @@ -6888,7 +6892,7 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st static void tas2781_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { - tas2781_generic_fixup(cdc, action, "i2c", "TIAS2781"); + tas2781_generic_fixup(cdc, action, "i2c"); } /* for alc295_fixup_hp_top_speakers */ diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index fb80280293..8493952305 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -65,6 +65,16 @@ enum calib_data { CALIB_MAX }; +/* TIAS2781 is the unofficial ACPI id, but widely used in current devices. + * TXNW2781 is the official ACPI id, and will be used in the new devices. + */ +static const struct acpi_device_id tas2781_acpi_hda_match[] = { + {"TIAS2781", 0 }, + {"TXNW2781", 1 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); + static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) { struct tasdevice_priv *tas_priv = data; @@ -644,20 +654,23 @@ static void tas2781_hda_remove(struct device *dev) static int tas2781_hda_i2c_probe(struct i2c_client *clt) { struct tasdevice_priv *tas_priv; - const char *device_name; - int ret; + int ret, i; - if (strstr(dev_name(&clt->dev), "TIAS2781")) - device_name = "TIAS2781"; - else - return -ENODEV; + /* Check TIAS2781 or TXNW2781 */ + for (i = 0; i < ARRAY_SIZE(tas2781_acpi_hda_match); i++) + if (strstr(dev_name(&clt->dev), tas2781_acpi_hda_match[i].id)) + break; + + if (i == ARRAY_SIZE(tas2781_acpi_hda_match)) + return dev_err_probe(tas_priv->dev, -ENODEV, + "Device not available\n"); tas_priv = tasdevice_kzalloc(clt); if (!tas_priv) return -ENOMEM; tas_priv->irq_info.irq = clt->irq; - ret = tas2781_read_acpi(tas_priv, device_name); + ret = tas2781_read_acpi(tas_priv, tas2781_acpi_hda_match[i].id); if (ret) return dev_err_probe(tas_priv->dev, ret, "Platform not supported\n"); @@ -822,12 +835,6 @@ static const struct i2c_device_id tas2781_hda_i2c_id[] = { {} }; -static const struct acpi_device_id tas2781_acpi_hda_match[] = { - {"TIAS2781", 0 }, - {} -}; -MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); - static struct i2c_driver tas2781_hda_i2c_driver = { .driver = { .name = "tas2781-hda", -- 2.34.1

4 3

[PATCH] ASoC: amd: yc: Fix non-functional mic on Lenovo 82QF and 82UG
by August Wikerfors 12 Sep '23

12 Sep '23

Like the Lenovo 82TL and 82V2, the Lenovo 82QF (Yoga 7 14ARB7) and 82UG (Legion S7 16ARHA7) both need a quirk entry for the internal microphone to function. Commit c008323fe361 ("ASoC: amd: yc: Fix a non-functional mic on Lenovo 82SJ") restricted the quirk that previously matched "82" to "82V2", breaking microphone functionality on these devices. Fix this by adding specific quirks for these models, as was done for the Lenovo 82TL. Fixes: c008323fe361 ("ASoC: amd: yc: Fix a non-functional mic on Lenovo 82SJ") Closes: https://github.com/tomsom/yoga-linux/issues/51 Link: https://bugzilla.kernel.org/show_bug.cgi?id=208555#c780 Cc: stable(a)vger.kernel.org Signed-off-by: August Wikerfors <git(a)augustwikerfors.se> --- sound/soc/amd/yc/acp6x-mach.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 59aa2e9d3a79..94e9eb8e73f2 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -213,6 +213,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21J6"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "82QF"), + } + }, { .driver_data = &acp6x_card, .matches = { @@ -220,6 +227,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "82TL"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "82UG"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.42.0

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