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

[PATCH] ASoC: intel: sof_sdw: add RT711 SDCA card for MTL platform
by Pierre-Louis Bossart 14 Jun '22

14 Jun '22

From: Yong Zhi <yong.zhi(a)intel.com> Enable on-board rt711 based sound card for MTL RVP. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Uday M Bhat <uday.m.bhat(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_sdw.c | 8 +++++ .../intel/common/soc-acpi-intel-mtl-match.c | 31 +++++++++++++++++++ 2 files changed, 39 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 77af3a7655c44..d23846572543b 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -315,6 +315,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { RT711_JD2 | SOF_SDW_FOUR_SPK), }, + /* MeteorLake devices */ + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Intel_mtlrvp"), + }, + .driver_data = (void *)(RT711_JD1 | SOF_SDW_TGL_HDMI), + }, {} }; diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index cc594b27e03be..74d3b82f8d35c 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -15,6 +15,31 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_machines); +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, +}; + +static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { + { + .adr = 0x000030025D071101ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt711" + } +}; + +static const struct snd_soc_acpi_link_adr mtl_rvp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_sdca_0_adr), + .adr_d = rt711_sdca_0_adr, + }, + {} +}; + /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { /* mockup tests need to be first */ @@ -36,6 +61,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-rt715-rt711-rt1308-mono.tplg", }, + { + .link_mask = BIT(0), + .links = mtl_rvp, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt711.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_sdw_machines); -- 2.34.1

2 1

[PATCH 1/2] ASoC: dt-bindings: fsl, mqs: Add compatible string for i.MX93 platform
by Shengjiu Wang 14 Jun '22

14 Jun '22

Add compatible string "fsl,imx93-mqs" for i.MX93 platform Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,mqs.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.txt b/Documentation/devicetree/bindings/sound/fsl,mqs.txt index 40353fc30255..d66284b8bef2 100644 --- a/Documentation/devicetree/bindings/sound/fsl,mqs.txt +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.txt @@ -2,7 +2,7 @@ fsl,mqs audio CODEC Required properties: - compatible : Must contain one of "fsl,imx6sx-mqs", "fsl,codec-mqs" - "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs". + "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs", "fsl,imx93-mqs". - clocks : A list of phandles + clock-specifiers, one for each entry in clock-names - clock-names : "mclk" - must required. -- 2.17.1

2 2

[PATCH 0/2] soundwire: Intel: add trigger callback
by Bard Liao 14 Jun '22

14 Jun '22

The .trigger callback will be used in ASoC. Hi Vinod, Could you provied a tag so that Mark can take the tag to his tree? Bard Liao (1): soundwire: Intel: add trigger callback Pierre-Louis Bossart (1): soundwire: intel: uniquify debug message drivers/soundwire/intel.c | 36 ++++++++++++++++++----------- include/linux/soundwire/sdw_intel.h | 1 + 2 files changed, 23 insertions(+), 14 deletions(-) -- 2.17.1

4 11

[PATCH] ALSA: hda/realtek: Apply fixup for Lenovo Yoga Duet 7 properly
by Takashi Iwai 14 Jun '22

14 Jun '22

It turned out that Lenovo shipped two completely different products with the very same PCI SSID, where both require different quirks; namely, Lenovo C940 has already the fixup for its speaker (ALC298_FIXUP_LENOVO_SPK_VOLUME) with the PCI SSID 17aa:3818, while Yoga Duet 7 has also the very same PCI SSID but requires a different quirk, ALC287_FIXUP_YOGA7_14TIL_SPEAKERS. Fortunately, both are with different codecs (C940 with ALC298 and Duet 7 with ALC287), hence we can apply different fixes by checking the codec ID. This patch implements that special fixup function. For easier handling, the internal function for applying a specific fixup entry is exported as __snd_hda_apply_fixup(), so that it can be called from the codec driver. The rest is simply calling it with a different fixup ID depending on the codec ID. Reported-by: Hans de Goede <hdegoede(a)redhat.com> Tested-by: nikitashvets(a)flyium.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/5ca147d1-3a2d-60c6-c491-8aa844183222@redhat.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_auto_parser.c | 7 ++++--- sound/pci/hda/hda_local.h | 1 + sound/pci/hda/patch_realtek.c | 24 +++++++++++++++++++++++- 3 files changed, 28 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index cd1db943b7e0..7c6b1fe8dfcc 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -819,7 +819,7 @@ static void set_pin_targets(struct hda_codec *codec, snd_hda_set_pin_ctl_cache(codec, cfg->nid, cfg->val); } -static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) +void __snd_hda_apply_fixup(struct hda_codec *codec, int id, int action, int depth) { const char *modelname = codec->fixup_name; @@ -829,7 +829,7 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) if (++depth > 10) break; if (fix->chained_before) - apply_fixup(codec, fix->chain_id, action, depth + 1); + __snd_hda_apply_fixup(codec, fix->chain_id, action, depth + 1); switch (fix->type) { case HDA_FIXUP_PINS: @@ -870,6 +870,7 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) id = fix->chain_id; } } +EXPORT_SYMBOL_GPL(__snd_hda_apply_fixup); /** * snd_hda_apply_fixup - Apply the fixup chain with the given action @@ -879,7 +880,7 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) void snd_hda_apply_fixup(struct hda_codec *codec, int action) { if (codec->fixup_list) - apply_fixup(codec, codec->fixup_id, action, 0); + __snd_hda_apply_fixup(codec, codec->fixup_id, action, 0); } EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index aca592651870..682dca2057db 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -348,6 +348,7 @@ void snd_hda_apply_verbs(struct hda_codec *codec); void snd_hda_apply_pincfgs(struct hda_codec *codec, const struct hda_pintbl *cfg); void snd_hda_apply_fixup(struct hda_codec *codec, int action); +void __snd_hda_apply_fixup(struct hda_codec *codec, int id, int action, int depth); void snd_hda_pick_fixup(struct hda_codec *codec, const struct hda_model_fixup *models, const struct snd_pci_quirk *quirk, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9b6ee775ee3f..b937f63d0d09 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7004,6 +7004,7 @@ enum { ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, + ALC298_FIXUP_LENOVO_C940_DUET7, ALC287_FIXUP_13S_GEN2_SPEAKERS, ALC256_FIXUP_SET_COEF_DEFAULTS, ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, @@ -7022,6 +7023,23 @@ enum { ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE, }; +/* A special fixup for Lenovo C940 and Yoga Duet 7; + * both have the very same PCI SSID, and we need to apply different fixups + * depending on the codec ID + */ +static void alc298_fixup_lenovo_c940_duet7(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + int id; + + if (codec->core.vendor_id == 0x10ec0298) + id = ALC298_FIXUP_LENOVO_SPK_VOLUME; /* C940 */ + else + id = ALC287_FIXUP_YOGA7_14ITL_SPEAKERS; /* Duet 7 */ + __snd_hda_apply_fixup(codec, id, action, 0); +} + static const struct hda_fixup alc269_fixups[] = { [ALC269_FIXUP_GPIO2] = { .type = HDA_FIXUP_FUNC, @@ -8721,6 +8739,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE, }, + [ALC298_FIXUP_LENOVO_C940_DUET7] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_lenovo_c940_duet7, + }, [ALC287_FIXUP_13S_GEN2_SPEAKERS] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -9274,7 +9296,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), - SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), + SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7), SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3820, "Yoga Duet 7 13ITL6", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), -- 2.35.3

1 0

How to add 2 different SND_PCI_QUIRKs for 2 models with the same PCI subsys vend + prod ids?
by Hans de Goede 14 Jun '22

14 Jun '22

Hi All, There is a problem with the sound on the Lenovo Yoga Duet 7 13IML05, which requires a SND_PCI_QUIRK to fix. But there already is an existing quirk for another Lenovo laptop using the same PCI subsys vend + prod ids. For more details see: https://wiki.archlinux.org/title/Lenovo_Yoga_Duet_7_13IML05#Audio So I guess this means that we need to add a way to also include a DMI match for SND_PCI_QUIRKs ? Maybe add a: const struct dmi_systemid *dmi_ids; member to struct snd_pci_quirk and a new SND_PCI_QUIRK_DMI macro to set this? Regards, Hans

2 5

[PATCH AUTOSEL 4.9 01/12] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0d9c4a57301b..f733f6b42b53 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -141,7 +141,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -368,7 +368,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 4

[PATCH AUTOSEL 4.14 01/14] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0d9c4a57301b..f733f6b42b53 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -141,7 +141,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -368,7 +368,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 4

[PATCH AUTOSEL 4.19 09/18] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 02c557e1f779..c5b0b56d9c94 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -697,7 +697,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 08/18] ASoC: es8328: Fix event generation for deemphasis control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ] Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 3afa163f7652..dcb01889e177 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -165,13 +165,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 07/18] ASoC: wm8962: Fix suspend while playing music
by Sasha Levin 14 Jun '22

14 Jun '22

From: Adam Ford <aford173(a)gmail.com> [ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ] If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index dde015fd70a4..3f75cb3209ff 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3861,6 +3861,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.35.1

1 0