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ucm2 profile for Universal Audio Volt 2
by GitHub pull_request - edited 19 Aug '23

19 Aug '23

alsa-project/alsa-ucm-conf pull request #340 was edited from z411: This adds a profile for the (relatively recent) Universal Audio Volt 2 USB audio interface. It has a Stereo output and 2 mono inputs that are (like most budget interfaces) combined into a stereo input. This profile effectively splits these inputs into two Line inputs. These inputs can be used for XLR microphones, Line level signals or Hi-Z instruments. The only weird thing is that in my machine the Mono Input 2 appears first in order but pretty sure I defined the prorities in the configuration file just fine. Everything works well on my machine. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12385352… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ucm2 profile for Universal Audio Volt 2
by GitHub pull_request - edited 19 Aug '23

19 Aug '23

alsa-project/alsa-ucm-conf pull request #340 was edited from z411: This adds a profile for the (relatively recent) Universal Audio Volt 2 USB audio interface. It has a Stereo output and 2 mono inputs that are (like most budget interfaces) combined into a stereo input. This profile effectively splits these inputs into two Line inputs. These inputs can be used for XLR microphones, Line level inputs or Hi-Z inputs. The only weird thing is that in my machine the Mono Input 2 appears first in order but pretty sure I defined the prorities in the configuration file just fine. Everything works well on my machine. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12385352… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Universal Audio Volt 2
by GitHub pull_request - opened 19 Aug '23

19 Aug '23

alsa-project/alsa-ucm-conf pull request #340 was opened from z411: This adds a profile for the (relatively recent) Universal Audio Vol 2 USB aduio interface. It has a Stereo output and 2 mono inputs that are (like most budget interfaces) combined into a stereo input. This profile effectively splits these inputs into two Line inputs. These inputs can be used for XLR microphones, Line level inputs or Hi-Z inputs. The only weird thing is that in my machine the Mono Input 2 appears first in order but pretty sure I defined the prorities in the configuration file just fine. Everything works well on my machine. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12385352… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v4 1/2] ALSA: hda/tas2781: Add tas2781 HDA driver
by Shenghao Ding 18 Aug '23

18 Aug '23

Integrate tas2781 configs for Lenovo Laptops. All of the tas2781s in the laptop will be aggregated as one audio device. The code support realtek as the primary codec. Rename "struct cs35l41_dev_name" to "struct scodec_dev_name" for all other side codecs instead of the certain one. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Changes in v4: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://lore.kernel.org/oe-kbuild-all/202308172137.SthCPFbA-lkp@intel. | com/ - remove workaround code for 0x17aa38be, laptop vendor will fix it in acpi. - rename comp_match_tas2781_dev_name to avoid indentation - simplify the check of vendor id with Lenovo - ThinkPad is one of Lenovo's brands, I was suggested me to use ALC269_FIXUP_THINKPAD_ACPI. - Add comments on ACARD_SINGLE_RANGE_EXT_TLV - Add the range check for tas_priv->tasdevice[] in tas2781_acpi_get_i2c_res. - remove acpi_subsystem_id - Issue in Laptop 0x17aa38be ACPI talbe caused codec->bus->pci->subsystem_device is not equal to (codec->core.subsystem_id & 0xffff) in snd_hda_pick_fixup. The former is 0x3802 and the latter is 0x38be leads to getting the wrong fixup_id and enter into the wrong entry. Although, this issue has been raised to the laptop manufacturer, but the ACPI table is locked, cannot be changed any more. Correct the wrong entry in the code. - Rename "struct cs35l41_dev_name" to "struct scodec_dev_name" for all other side codecs instead of one certain codec. - Ignore the checkpatch complaints in alc269_fixup_tbl - Drop the hunk which is irrelevant with my code in alc_fixup_headset_mode_alc255_no_hp_mic - Add tiwai(a)suse.de into Cc list - remove useless index - combine ALC287_FIXUP_TAS2781_I2C_2 and ALC287_FIXUP_TAS2781_I2C_4 together as ALC287_FIXUP_TAS2781_I2C, The code view all the tas2781s in the laptop as one instance. - delete the white space at the end of the line in alc_fixup_headset_mode_alc255_no_hp_mic --- sound/pci/hda/patch_realtek.c | 88 +++++++++++++++++++++++++++++++++-- 1 file changed, 85 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 44fccfb93cff..ba1b02ed184a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6721,7 +6721,7 @@ static void comp_generic_playback_hook(struct hda_pcm_stream *hinfo, struct hda_ } } -struct cs35l41_dev_name { +struct scodec_dev_name { const char *bus; const char *hid; int index; @@ -6730,7 +6730,7 @@ struct cs35l41_dev_name { /* match the device name in a slightly relaxed manner */ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) { - struct cs35l41_dev_name *p = data; + struct scodec_dev_name *p = data; const char *d = dev_name(dev); int n = strlen(p->bus); char tmp[32]; @@ -6746,12 +6746,32 @@ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) return !strcmp(d + n, tmp); } +static int comp_match_tas2781_dev_name(struct device *dev, + void *data) +{ + struct scodec_dev_name *p = data; + const char *d = dev_name(dev); + int n = strlen(p->bus); + char tmp[32]; + + /* check the bus name */ + if (strncmp(d, p->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); +} + static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char *bus, const char *hid, int count) { struct device *dev = hda_codec_dev(cdc); struct alc_spec *spec = cdc->spec; - struct cs35l41_dev_name *rec; + struct scodec_dev_name *rec; int ret, i; switch (action) { @@ -6779,6 +6799,41 @@ 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) +{ + 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); + ret = component_master_add_with_match(dev, &comp_master_ops, + spec->match); + if (ret) + codec_err(cdc, + "Fail to register component aggregator %d\n", + ret); + else + spec->gen.pcm_playback_hook = + comp_generic_playback_hook; + break; + case HDA_FIXUP_ACT_FREE: + component_master_del(dev, &comp_master_ops); + break; + } +} + static void cs35l41_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 2); @@ -6806,6 +6861,12 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st cs35l41_generic_fixup(cdc, action, "i2c", "CLSA0101", 2); } +static void tas2781_fixup_i2c(struct hda_codec *cdc, + const struct hda_fixup *fix, int action) +{ + tas2781_generic_fixup(cdc, action, "i2c", "TIAS2781"); +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -7231,6 +7292,7 @@ enum { ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, + ALC287_FIXUP_TAS2781_I2C, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9309,6 +9371,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, + [ALC287_FIXUP_TAS2781_I2C] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9884,6 +9952,20 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual powe mode2 YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bf, "Yoga S980-14.5 proX LX Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38c3, "Y980 DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.34.1

3 5

[PATCH] ASoC: amd: acp: Add kcontrols and widgets per-codec in common code
by Alper Nebi Yasak 18 Aug '23

18 Aug '23

Commit 7ac3404c2e21 ("ASoC: amd: acp: Map missing jack kcontrols") adds card kcontrols and DAPM widgets corresponding to jacks for Headphone and Headset Mic. But these were already being added in acp-legacy-mach.c and acp-sof-mach.c, so this causes a probe error: sof_mach rt5682s-hs-rt1019: control 2:0:0:Headphone Jack Switch:0 is already present sof_mach rt5682s-hs-rt1019: ASoC: Failed to add Headphone Jack Switch: -16 sof_mach rt5682s-hs-rt1019: devm_snd_soc_register_card(sof-rt5682s-hs-rt1019) failed: -16 sof_mach: probe of rt5682s-hs-rt1019 failed with error -16 Removing the new duplicates from the common code might be enough to fix the issue. But some of the codecs use different kcontrols and widgets. NAU8821 alone has an "Int Mic". MAX98360A has a single "Spk", while RT1019 has "Left Spk" and "Right Spk" (and two codec instances with name prefixes). Since there are per-codec differences, add the kcontrols and widgets in the common code as each dai-link is being initialized, instead of in the two machine drivers' probe(). Also, MAX98388 has Left and Right instances, but uses a single "SPK" widget. That seems weird, so normalize it to use two widgets (one per each instance like RT1019) while we're adding controls. Fixes: 7ac3404c2e21 ("ASoC: amd: acp: Map missing jack kcontrols") Signed-off-by: Alper Nebi Yasak <alpernebiyasak(a)gmail.com> --- Applies onto linux-next. sound/soc/amd/acp/acp-legacy-mach.c | 22 +--------- sound/soc/amd/acp/acp-mach-common.c | 68 +++++++++++++++++++++++++++-- sound/soc/amd/acp/acp-sof-mach.c | 21 +-------- 3 files changed, 67 insertions(+), 44 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-mach.c b/sound/soc/amd/acp/acp-legacy-mach.c index 676ad50638d0..6d57d17ddfd7 100644 --- a/sound/soc/amd/acp/acp-legacy-mach.c +++ b/sound/soc/amd/acp/acp-legacy-mach.c @@ -75,23 +75,6 @@ static struct acp_card_drvdata rt5682s_rt1019_rmb_data = { .tdm_mode = false, }; -static const struct snd_kcontrol_new acp_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Spk"), - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), - -}; - -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), -}; - static int acp_asoc_probe(struct platform_device *pdev) { struct snd_soc_card *card = NULL; @@ -110,11 +93,8 @@ static int acp_asoc_probe(struct platform_device *pdev) card->dev = dev; card->owner = THIS_MODULE; card->name = pdev->id_entry->name; - card->dapm_widgets = acp_widgets; - card->num_dapm_widgets = ARRAY_SIZE(acp_widgets); - card->controls = acp_controls; - card->num_controls = ARRAY_SIZE(acp_controls); card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data; + /* Widgets and controls added per-codec in acp-mach-common.c */ acp_card_drvdata = card->drvdata; dmi_id = dmi_first_match(acp_quirk_table); diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index f3abaa182fbb..a06af82b8056 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -513,6 +513,16 @@ SND_SOC_DAILINK_DEF(rt1019, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1019:00", "rt1019-aif"), COMP_CODEC("i2c-10EC1019:01", "rt1019-aif"))); +static const struct snd_kcontrol_new rt1019_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_soc_dapm_widget rt1019_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + static const struct snd_soc_dapm_route rt1019_map_lr[] = { { "Left Spk", NULL, "Left SPO" }, { "Right Spk", NULL, "Right SPO" }, @@ -533,10 +543,25 @@ static int acp_card_rt1019_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct acp_card_drvdata *drvdata = card->drvdata; + int ret; if (drvdata->amp_codec_id != RT1019) return -EINVAL; + ret = snd_soc_dapm_new_controls(&card->dapm, rt1019_widgets, + ARRAY_SIZE(rt1019_widgets)); + if (ret) { + dev_err(rtd->dev, "unable to add widget dapm controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_add_card_controls(card, rt1019_controls, + ARRAY_SIZE(rt1019_controls)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + return snd_soc_dapm_add_routes(&rtd->card->dapm, rt1019_map_lr, ARRAY_SIZE(rt1019_map_lr)); } @@ -667,6 +692,14 @@ static const struct snd_soc_ops acp_card_rt1019_ops = { SND_SOC_DAILINK_DEF(max98360a, DAILINK_COMP_ARRAY(COMP_CODEC("MX98360A:00", "HiFi"))); +static const struct snd_kcontrol_new max98360a_controls[] = { + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget max98360a_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + static const struct snd_soc_dapm_route max98360a_map[] = { {"Spk", NULL, "Speaker"}, }; @@ -675,10 +708,25 @@ static int acp_card_maxim_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct acp_card_drvdata *drvdata = card->drvdata; + int ret; if (drvdata->amp_codec_id != MAX98360A) return -EINVAL; + ret = snd_soc_dapm_new_controls(&card->dapm, max98360a_widgets, + ARRAY_SIZE(max98360a_widgets)); + if (ret) { + dev_err(rtd->dev, "unable to add widget dapm controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_add_card_controls(card, max98360a_controls, + ARRAY_SIZE(max98360a_controls)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + return snd_soc_dapm_add_routes(&rtd->card->dapm, max98360a_map, ARRAY_SIZE(max98360a_map)); } @@ -744,13 +792,19 @@ SND_SOC_DAILINK_DEF(max98388, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-ADS8388:00", "max98388-aif1"), COMP_CODEC("i2c-ADS8388:01", "max98388-aif1"))); +static const struct snd_kcontrol_new max98388_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + static const struct snd_soc_dapm_widget max98388_widgets[] = { - SND_SOC_DAPM_SPK("SPK", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), }; static const struct snd_soc_dapm_route max98388_map[] = { - { "SPK", NULL, "Left BE_OUT" }, - { "SPK", NULL, "Right BE_OUT" }, + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, }; static struct snd_soc_codec_conf max98388_conf[] = { @@ -803,6 +857,14 @@ static int acp_card_max98388_init(struct snd_soc_pcm_runtime *rtd) /* Don't need to add routes if widget addition failed */ return ret; } + + ret = snd_soc_add_card_controls(card, max98388_controls, + ARRAY_SIZE(max98388_controls)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + return snd_soc_dapm_add_routes(&rtd->card->dapm, max98388_map, ARRAY_SIZE(max98388_map)); } diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index a1c893f33f74..5223033a122f 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -94,22 +94,6 @@ static struct acp_card_drvdata sof_nau8821_max98388_data = { .tdm_mode = false, }; -static const struct snd_kcontrol_new acp_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Spk"), - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), -}; - -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), -}; - static int acp_sof_probe(struct platform_device *pdev) { struct snd_soc_card *card = NULL; @@ -128,11 +112,8 @@ static int acp_sof_probe(struct platform_device *pdev) card->dev = dev; card->owner = THIS_MODULE; card->name = pdev->id_entry->name; - card->dapm_widgets = acp_widgets; - card->num_dapm_widgets = ARRAY_SIZE(acp_widgets); - card->controls = acp_controls; - card->num_controls = ARRAY_SIZE(acp_controls); card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data; + /* Widgets and controls added per-codec in acp-mach-common.c */ acp_card_drvdata = card->drvdata; dmi_id = dmi_first_match(acp_quirk_table); base-commit: 47762f08697484cf0c2f2904b8c52375ed26c8cb -- 2.40.1

2 1

[PATCH] ASoC: rt5645: improve the depop sequences of CBJ detection
by shumingf＠realtek.com 18 Aug '23

18 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> This patch will improve the depop function of CBJ detection before the headphone playback when the system resumes. We also want to complete the CBJ detection in the system-level resume function. It could avoid triggering the detection at the playing state of headphones. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5645.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e49aa28280e0..038d93e20883 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -80,6 +80,7 @@ static const struct reg_sequence init_list[] = { static const struct reg_sequence rt5650_init_list[] = { {0xf6, 0x0100}, + {RT5645_PWR_ANLG1, 0x02}, }; static const struct reg_default rt5645_reg[] = { @@ -1696,6 +1697,9 @@ static void hp_amp_power(struct snd_soc_component *component, int on) regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_component_write(component, RT5645_DEPOP_M2, 0x1140); + snd_soc_component_update_bits(component, RT5645_PWR_ANLG1, + RT5645_PWR_HP_L | RT5645_PWR_HP_R, + RT5645_PWR_HP_L | RT5645_PWR_HP_R); msleep(90); } else { /* depop parameters */ @@ -1743,7 +1747,8 @@ static void hp_amp_power(struct snd_soc_component *component, int on) snd_soc_component_write(component, RT5645_DEPOP_M2, 0x1140); msleep(100); snd_soc_component_write(component, RT5645_DEPOP_M1, 0x0001); - + snd_soc_component_update_bits(component, RT5645_PWR_ANLG1, + RT5645_PWR_HP_L | RT5645_PWR_HP_R, 0); } else { snd_soc_component_update_bits(component, RT5645_DEPOP_M1, RT5645_HP_SG_MASK | @@ -3150,7 +3155,7 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse unsigned int val; if (jack_insert) { - regmap_write(rt5645->regmap, RT5645_CHARGE_PUMP, 0x0e06); + regmap_write(rt5645->regmap, RT5645_CHARGE_PUMP, 0x0206); /* for jack type detect */ snd_soc_dapm_force_enable_pin(dapm, "LDO2"); @@ -3195,6 +3200,8 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse if (rt5645->pdata.level_trigger_irq) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR); + + regmap_write(rt5645->regmap, RT5645_CHARGE_PUMP, 0x0e06); } else { /* jack out */ rt5645->jack_type = 0; @@ -4003,13 +4010,13 @@ static int rt5645_i2c_probe(struct i2c_client *i2c) regmap_write(rt5645->regmap, RT5645_AD_DA_MIXER, 0x8080); - ret = regmap_register_patch(rt5645->regmap, init_list, + ret = regmap_multi_reg_write(rt5645->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); if (rt5645->codec_type == CODEC_TYPE_RT5650) { - ret = regmap_register_patch(rt5645->regmap, rt5650_init_list, + ret = regmap_multi_reg_write(rt5645->regmap, rt5650_init_list, ARRAY_SIZE(rt5650_init_list)); if (ret != 0) dev_warn(&i2c->dev, "Apply rt5650 patch failed: %d\n", @@ -4220,8 +4227,7 @@ static int __maybe_unused rt5645_sys_resume(struct device *dev) if (rt5645->hp_jack) { rt5645->jack_type = 0; - queue_delayed_work(system_power_efficient_wq, - &rt5645->jack_detect_work, msecs_to_jiffies(0)); + rt5645_jack_detect_work(&rt5645->jack_detect_work.work); } return 0; } -- 2.34.1

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[PATCH] ASoC: SOF: intel: hda: Clean up link DMA for IPC3 during stop
by Peter Ujfalusi 18 Aug '23

18 Aug '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> With IPC3, we reset hw_params during the stop trigger, so we should also clean up the link DMA during the stop trigger. Fixes: 1bf83fa6654c ("ASoC: SOF: Intel: hda-dai: Do not perform DMA cleanup during stop") Closes: https://github.com/thesofproject/linux/issues/4455 Closes: https://github.com/thesofproject/linux/issues/4482 Closes: https://bugzilla.kernel.org/show_bug.cgi?id=217673 Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/hda-dai-ops.c | 11 ++++++++++- sound/soc/sof/intel/hda-dai.c | 5 ++--- sound/soc/sof/intel/hda.h | 2 ++ 3 files changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index f3513796c189..f33051eac1c0 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -372,6 +372,7 @@ static const struct hda_dai_widget_dma_ops hda_ipc4_chain_dma_ops = { static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, struct snd_pcm_substream *substream, int cmd) { + struct hdac_ext_stream *hext_stream = hda_get_hext_stream(sdev, cpu_dai, substream); struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); switch (cmd) { @@ -379,9 +380,17 @@ static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *c case SNDRV_PCM_TRIGGER_STOP: { struct snd_sof_dai_config_data data = { 0 }; + int ret; data.dai_data = DMA_CHAN_INVALID; - return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_FREE, &data); + ret = hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_FREE, &data); + if (ret < 0) + return ret; + + if (cmd == SNDRV_PCM_TRIGGER_STOP) + return hda_link_dma_cleanup(substream, hext_stream, cpu_dai); + + break; } case SNDRV_PCM_TRIGGER_PAUSE_PUSH: return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_PAUSE, NULL); diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 3297dea493aa..863865f3d77e 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -107,9 +107,8 @@ hda_dai_get_ops(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai return sdai->platform_private; } -static int hda_link_dma_cleanup(struct snd_pcm_substream *substream, - struct hdac_ext_stream *hext_stream, - struct snd_soc_dai *cpu_dai) +int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream, + struct snd_soc_dai *cpu_dai) { const struct hda_dai_widget_dma_ops *ops = hda_dai_get_ops(substream, cpu_dai); struct sof_intel_hda_stream *hda_stream; diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 3f7c6fb05e5d..5b9e4ebcc18b 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -963,5 +963,7 @@ const struct hda_dai_widget_dma_ops * hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget); int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags, struct snd_sof_dai_config_data *data); +int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream, + struct snd_soc_dai *cpu_dai); #endif -- 2.41.0

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[PATCH -next] ALSA: asihpi: Remove unused declarations
by Yue Haibing 18 Aug '23

18 Aug '23

These are not implemented, so can remove them. Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- sound/pci/asihpi/hpi.h | 16 ---------------- sound/pci/asihpi/hpi_internal.h | 5 ----- 2 files changed, 21 deletions(-) diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 3aebec763fb8..04a5cf6572cd 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -1191,19 +1191,6 @@ u16 hpi_adapter_set_mode_ex(u16 adapter_index, u32 adapter_mode, u16 hpi_adapter_get_mode(u16 adapter_index, u32 *padapter_mode); -u16 hpi_adapter_get_assert2(u16 adapter_index, u16 *p_assert_count, - char *psz_assert, u32 *p_param1, u32 *p_param2, - u32 *p_dsp_string_addr, u16 *p_processor_id); - -u16 hpi_adapter_test_assert(u16 adapter_index, u16 assert_id); - -u16 hpi_adapter_enable_capability(u16 adapter_index, u16 capability, u32 key); - -u16 hpi_adapter_self_test(u16 adapter_index); - -u16 hpi_adapter_debug_read(u16 adapter_index, u32 dsp_address, char *p_bytes, - int *count_bytes); - u16 hpi_adapter_set_property(u16 adapter_index, u16 property, u16 paramter1, u16 paramter2); @@ -1488,9 +1475,6 @@ u16 hpi_pad_get_program_type(u32 h_control, u32 *ppTY); u16 hpi_pad_get_rdsPI(u32 h_control, u32 *ppI); -u16 hpi_pad_get_program_type_string(u32 h_control, const u32 data_type, - const u32 pTY, char *psz_string, const u32 string_length); - /****************************/ /* AES/EBU Receiver control */ /****************************/ diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index 6859d51389f5..e569e3b33b8e 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -1394,17 +1394,12 @@ u32 hpi_indexes_to_handle(const char c_object, const u16 adapter_index, void hpi_send_recv(struct hpi_message *phm, struct hpi_response *phr); /* used in PnP OS/driver */ -u16 hpi_subsys_create_adapter(const struct hpi_resource *p_resource, - u16 *pw_adapter_index); - u16 hpi_outstream_host_buffer_get_info(u32 h_outstream, u8 **pp_buffer, struct hpi_hostbuffer_status **pp_status); u16 hpi_instream_host_buffer_get_info(u32 h_instream, u8 **pp_buffer, struct hpi_hostbuffer_status **pp_status); -u16 hpi_adapter_restart(u16 adapter_index); - /* The following 3 functions were last declared in header files for driver 3.10. HPI_ControlQuery() used to be the recommended way -- 2.34.1

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[PATCH v3 0/9] sound: Use -EPROBE_DEFER instead of i915 module loading.
by Maarten Lankhorst 18 Aug '23

18 Aug '23

From: Maarten Lankhorst <dev(a)lankhorst.se> Explicitly loading i915 becomes a problem when upstreaming the new intel driver for Tiger Lake and higher graphics (xe). By loading i915, it doesn't wait for driver load of xe, and will fail completely before it loads. -EPROBE_DEFER has to be returned before any device is created in probe(), otherwise the removal of the device will cause EPROBE_DEFER to try again in an infinite loop. The conversion is done in gradual steps. First I add an argument to snd_hdac_i915_init to allow for -EPROBE_DEFER so I can convert each driver separately. Then I convert each driver to move snd_hdac_i915_init out of the workqueue. Finally I drop the ability to choose modprobe behavior after the last user is converted. I suspect the avs and skylake drivers used snd_hdac_i915_init purely for the modprobe, but I don't have the hardware to test if it can be safely removed. It can still be done easily in a followup patch to simplify probing. --- Changes since previous version: - Keep the workqueue for soc/sof/intel, instead only move out snd_hdac_i915_init. (Patch 8/9) Rest of patches unchanged. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Daniel Baluta <daniel.baluta(a)nxp.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Cc: sound-open-firmware(a)alsa-project.org Maarten Lankhorst (9): ALSA: hda/intel: Fix error handling in azx_probe() ALSA: hda/i915: Allow override of gpu binding. ALSA: hda/i915: Add an allow_modprobe argument to snd_hdac_i915_init ALSA: hda/i915: Allow xe as match for i915_component_master_match ASoC: Intel: avs: Move snd_hdac_i915_init to before probe_work. ASoC: Intel: Skylake: Move snd_hdac_i915_init to before probe_work. ALSA: hda/intel: Move snd_hdac_i915_init to before probe_work. ASoC: SOF: Intel: Move binding to display driver outside of deferred probe ALSA: hda/i915: Remove extra argument from snd_hdac_i915_init sound/hda/hdac_i915.c | 25 ++++++++------- sound/pci/hda/hda_intel.c | 60 ++++++++++++++++++----------------- sound/soc/intel/avs/core.c | 13 +++++--- sound/soc/intel/skylake/skl.c | 31 ++++++------------ sound/soc/sof/core.c | 19 ++++------- 5 files changed, 70 insertions(+), 78 deletions(-) -- 2.39.2

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Re: [PATCH 3/4] ASoC: amd: acp: Add machine driver that enables sound for systems with a ES8336 codec
by Zhu Ning 18 Aug '23

18 Aug '23

> > This is saying that the machine driver should disable jack detection > > over suspend and restart it during resume. The machine driver should > > suspend before the rest of the card which should mean that the CODEC > > gets powered off then. The core can't tell if jack detection is > > supposed to work over suspend, it is a standard wake event on systems > > like phones, but it sounds like on this system the power gets removed > > from the device so that can't work. > > > > Sorry, I don't understand what you are trying to say here. My intention > is to find a way to have sound working when suspending/resuming > with jack inserted by not fudging the jack in the machine driver but > fixing it in the CODEC. > > > I don't understand why that would be expected to help? The main effect > > of keeping the bias on all the time would be to consume more power. > > I don't fully understand the whole bias thing (I did try hard to > understand it from the code), but in this specific instance it helps for > suspending the CODEC. If idle_bias_on is true then idle_bias_off will be > false, that would mean that during suspend (with jack inserted) > in snd_soc_suspend() in switch (snd_soc_dapm_get_bias_level(dapm)) > the flow goes in case SND_SOC_BIAS_STANDBY and since dapm->idle_bias_off > is false it doesn't break from the case but falls through to > SND_SOC_BIAS_OFF case which in turn calls the CODEC component's suspend > callback. > > The reason the sound stops working with jack inserted is that the CODEC > suspend callback never gets called. It only gets called when the jack > is not inserted. Hi Postevca, I just reproduced the bug on my side. You are right that when jack inserted, the codec driver will never enter SND_SOC_BIAS_OFF or suspend or resume. This makes sense because the codec needs micbias power to detect button. Problem is that ALSA or dapm thought the codec power is on during suspend, but the system power is turned off and the registers are reset. The system power is controlled by EC and neither dapm or codec driver can do anything about it. The system power is just turned off. Disabling the jack detection before suspend is ok because the system stops working during suspend anyway. Maybe a detailed comment is needed to explain why?

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