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

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

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[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

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[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

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[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

2 1

[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

4 5

[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

2 1

[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

5 23

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?

1 0

[PATCH] ALSA: aoa: Fix typos in PCM fix patch
by Takashi Iwai 18 Aug '23

18 Aug '23

There was typos in the previous fix for PCM to detach the struct device that caused build errors. Corrected here. Fixes: bc41a7228ced ("ALSA: pcm: Don't embed device") Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202308181347.q3XPr3Lm-lkp@intel.com/ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/aoa/soundbus/i2sbus/pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 3680eb6eabc9..07df5cc0f2d7 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -972,7 +972,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, goto out_put_ci_module; snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, &i2sbus_playback_ops); - dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK]->dev.parent = + dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].dev->parent = &dev->ofdev.dev; i2sdev->out.created = 1; } @@ -989,7 +989,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, goto out_put_ci_module; snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, &i2sbus_record_ops); - dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE]->dev.parent = + dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].dev->parent = &dev->ofdev.dev; i2sdev->in.created = 1; } -- 2.35.3

1 0

[tiwai-sound:topic/dev-split-kref 3/11] sound/aoa/soundbus/i2sbus/pcm.c:975:47: error: member reference type 'struct snd_pcm_str' is not a pointer; did you mean to use '.'?
by kernel test robot 18 Aug '23

18 Aug '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/dev-split-kref head: 6e1b44385310dd465cae48213eae65d0b8095d13 commit: 7a3dce3fb37c25de3f3c347f4e69ba358cb81a3d [3/11] ALSA: pcm: Don't embed device config: powerpc-g5_defconfig (https://download.01.org/0day-ci/archive/20230818/202308181347.q3XPr3Lm-lkp@…) compiler: clang version 15.0.7 (https://github.com/llvm/llvm-project.git 8dfdcc7b7bf66834a761bd8de445840ef68e4d1a) reproduce: (https://download.01.org/0day-ci/archive/20230818/202308181347.q3XPr3Lm-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/202308181347.q3XPr3Lm-lkp@intel.com/ All errors (new ones prefixed by >>): >> sound/aoa/soundbus/i2sbus/pcm.c:975:47: error: member reference type 'struct snd_pcm_str' is not a pointer; did you mean to use '.'? dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK]->dev.parent = ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~ . >> sound/aoa/soundbus/i2sbus/pcm.c:975:52: error: member reference type 'struct device *' is a pointer; did you mean to use '->'? dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK]->dev.parent = ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^ -> sound/aoa/soundbus/i2sbus/pcm.c:992:46: error: member reference type 'struct snd_pcm_str' is not a pointer; did you mean to use '.'? dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE]->dev.parent = ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~ . sound/aoa/soundbus/i2sbus/pcm.c:992:51: error: member reference type 'struct device *' is a pointer; did you mean to use '->'? dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE]->dev.parent = ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^ -> 4 errors generated. vim +975 sound/aoa/soundbus/i2sbus/pcm.c 866 867 int 868 i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, 869 struct codec_info *ci, void *data) 870 { 871 int err, in = 0, out = 0; 872 struct transfer_info *tmp; 873 struct i2sbus_dev *i2sdev = soundbus_dev_to_i2sbus_dev(dev); 874 struct codec_info_item *cii; 875 876 if (!dev->pcmname || dev->pcmid == -1) { 877 printk(KERN_ERR "i2sbus: pcm name and id must be set!\n"); 878 return -EINVAL; 879 } 880 881 list_for_each_entry(cii, &dev->codec_list, list) { 882 if (cii->codec_data == data) 883 return -EALREADY; 884 } 885 886 if (!ci->transfers || !ci->transfers->formats 887 || !ci->transfers->rates || !ci->usable) 888 return -EINVAL; 889 890 /* we currently code the i2s transfer on the clock, and support only 891 * 32 and 64 */ 892 if (ci->bus_factor != 32 && ci->bus_factor != 64) 893 return -EINVAL; 894 895 /* If you want to fix this, you need to keep track of what transport infos 896 * are to be used, which codecs they belong to, and then fix all the 897 * sysclock/busclock stuff above to depend on which is usable */ 898 list_for_each_entry(cii, &dev->codec_list, list) { 899 if (cii->codec->sysclock_factor != ci->sysclock_factor) { 900 printk(KERN_DEBUG 901 "cannot yet handle multiple different sysclocks!\n"); 902 return -EINVAL; 903 } 904 if (cii->codec->bus_factor != ci->bus_factor) { 905 printk(KERN_DEBUG 906 "cannot yet handle multiple different bus clocks!\n"); 907 return -EINVAL; 908 } 909 } 910 911 tmp = ci->transfers; 912 while (tmp->formats && tmp->rates) { 913 if (tmp->transfer_in) 914 in = 1; 915 else 916 out = 1; 917 tmp++; 918 } 919 920 cii = kzalloc(sizeof(struct codec_info_item), GFP_KERNEL); 921 if (!cii) 922 return -ENOMEM; 923 924 /* use the private data to point to the codec info */ 925 cii->sdev = soundbus_dev_get(dev); 926 cii->codec = ci; 927 cii->codec_data = data; 928 929 if (!cii->sdev) { 930 printk(KERN_DEBUG 931 "i2sbus: failed to get soundbus dev reference\n"); 932 err = -ENODEV; 933 goto out_free_cii; 934 } 935 936 if (!try_module_get(THIS_MODULE)) { 937 printk(KERN_DEBUG "i2sbus: failed to get module reference!\n"); 938 err = -EBUSY; 939 goto out_put_sdev; 940 } 941 942 if (!try_module_get(ci->owner)) { 943 printk(KERN_DEBUG 944 "i2sbus: failed to get module reference to codec owner!\n"); 945 err = -EBUSY; 946 goto out_put_this_module; 947 } 948 949 if (!dev->pcm) { 950 err = snd_pcm_new(card, dev->pcmname, dev->pcmid, 0, 0, 951 &dev->pcm); 952 if (err) { 953 printk(KERN_DEBUG "i2sbus: failed to create pcm\n"); 954 goto out_put_ci_module; 955 } 956 } 957 958 /* ALSA yet again sucks. 959 * If it is ever fixed, remove this line. See below. */ 960 out = in = 1; 961 962 if (!i2sdev->out.created && out) { 963 if (dev->pcm->card != card) { 964 /* eh? */ 965 printk(KERN_ERR 966 "Can't attach same bus to different cards!\n"); 967 err = -EINVAL; 968 goto out_put_ci_module; 969 } 970 err = snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, 1); 971 if (err) 972 goto out_put_ci_module; 973 snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, 974 &i2sbus_playback_ops); > 975 dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK]->dev.parent = 976 &dev->ofdev.dev; 977 i2sdev->out.created = 1; 978 } 979 980 if (!i2sdev->in.created && in) { 981 if (dev->pcm->card != card) { 982 printk(KERN_ERR 983 "Can't attach same bus to different cards!\n"); 984 err = -EINVAL; 985 goto out_put_ci_module; 986 } 987 err = snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, 1); 988 if (err) 989 goto out_put_ci_module; 990 snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, 991 &i2sbus_record_ops); 992 dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE]->dev.parent = 993 &dev->ofdev.dev; 994 i2sdev->in.created = 1; 995 } 996 997 /* so we have to register the pcm after adding any substream 998 * to it because alsa doesn't create the devices for the 999 * substreams when we add them later. 1000 * Therefore, force in and out on both busses (above) and 1001 * register the pcm now instead of just after creating it. 1002 */ 1003 err = snd_device_register(card, dev->pcm); 1004 if (err) { 1005 printk(KERN_ERR "i2sbus: error registering new pcm\n"); 1006 goto out_put_ci_module; 1007 } 1008 /* no errors any more, so let's add this to our list */ 1009 list_add(&cii->list, &dev->codec_list); 1010 1011 dev->pcm->private_data = i2sdev; 1012 dev->pcm->private_free = i2sbus_private_free; 1013 1014 /* well, we really should support scatter/gather DMA */ 1015 snd_pcm_set_managed_buffer_all( 1016 dev->pcm, SNDRV_DMA_TYPE_DEV, 1017 &macio_get_pci_dev(i2sdev->macio)->dev, 1018 64 * 1024, 64 * 1024); 1019 1020 return 0; 1021 out_put_ci_module: 1022 module_put(ci->owner); 1023 out_put_this_module: 1024 module_put(THIS_MODULE); 1025 out_put_sdev: 1026 soundbus_dev_put(dev); 1027 out_free_cii: 1028 kfree(cii); 1029 return err; 1030 } 1031 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH RFC 0/6] ALSA: Fix UAF with delayed kobj release
by Takashi Iwai 18 Aug '23

18 Aug '23

Hi, this is a test patch set for addressing the UAF problems with delayed kobj releases reported by Curtis: https://lore.kernel.org/r/20230801171928.1460120-1-cujomalainey@chromium.org The patch introduced a simple helper for allocating memory with a refcount, and converts the card object, control, PCM and compress objects with the new type. With the refcount, the actual memory release is delayed until all referrer are gone. It's just a RFC and only lightly tested. I myself am not sure whether this is the best way to go. It might be better to take Curtis' approach, just converting the device to its own allocation, too. (But I don't know whether Curtis' patch set covers all cases -- can still be a UAF of card_dev due to devres vs kobj release?) thanks, Takashi === Takashi Iwai (6): ALSA: core: Introduced referenced memory allocator ALSA: core: Fix potential UAF by delayed kobject release of card_dev ALSA: core: Associate memory reference with device initialization ALSA: pcm: Release memory with reference ALSA: control: Reference card by ctl_dev ALSA: compress: Reference card by the device include/sound/core.h | 7 ++- sound/core/compress_offload.c | 2 +- sound/core/control.c | 2 +- sound/core/hwdep.c | 2 +- sound/core/init.c | 105 +++++++++++++++++++++++++++------ sound/core/pcm.c | 6 +- sound/core/rawmidi.c | 2 +- sound/core/seq/seq_clientmgr.c | 2 +- sound/core/timer.c | 2 +- 9 files changed, 101 insertions(+), 29 deletions(-) -- 2.35.3

2 23

[PATCH v2 0/5] ASoC: Convert remaining Realtek codecs to GPIO descriptors
by Linus Walleij 18 Aug '23

18 Aug '23

After dropping unused headers a few Realtek devices actually using the GPIO descriptors remain. Converting them to use optional GPIO descriptors is pretty straight-forward. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Changes in v2: - Rebased on asoc-6.6 - Link to v1: https://lore.kernel.org/r/20230817-descriptors-asoc-rt-v1-0-434f5f177cee@li… --- Linus Walleij (5): ASoC: rt5640: Convert to just use GPIO descriptors ASoC: rt5665: Convert to use GPIO descriptors ASoC: rt5668: Convert to use GPIO descriptors ASoC: rt5682: Convert to use GPIO descriptors ASoC: rt5682s: Convert to use GPIO descriptors include/sound/rt5665.h | 2 -- include/sound/rt5668.h | 3 --- include/sound/rt5682.h | 3 --- include/sound/rt5682s.h | 3 --- sound/soc/codecs/rt5640.c | 55 ++++++++++++------------------------------- sound/soc/codecs/rt5640.h | 2 +- sound/soc/codecs/rt5665.c | 17 +++++++------ sound/soc/codecs/rt5668.c | 17 +++++++------ sound/soc/codecs/rt5682-i2c.c | 11 ++++----- sound/soc/codecs/rt5682-sdw.c | 5 ++++ sound/soc/codecs/rt5682.c | 20 ++++++++++++---- sound/soc/codecs/rt5682.h | 3 +++ sound/soc/codecs/rt5682s.c | 16 ++++++------- sound/soc/codecs/rt5682s.h | 2 ++ 14 files changed, 68 insertions(+), 91 deletions(-) --- base-commit: 17b9f4387ebabb19b871bbe2d06562e48e4e7130 change-id: 20230817-descriptors-asoc-rt-9c072934d376 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

2 6

[PATCH 0/5] ASoC: Convert remaining Realtek codecs to GPIO descriptors
by Linus Walleij 18 Aug '23

18 Aug '23

After dropping unused headers a few Realtek devices actually using the GPIO descriptors remain. Converting them to use optional GPIO descriptors is pretty straight-forward. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Linus Walleij (5): ASoC: rt5640: Convert to just use GPIO descriptors ASoC: rt5665: Convert to use GPIO descriptors ASoC: rt5668: Convert to use GPIO descriptors ASoC: rt5682: Convert to use GPIO descriptors ASoC: rt5682s: Convert to use GPIO descriptors include/sound/rt5665.h | 2 -- include/sound/rt5668.h | 3 --- include/sound/rt5682.h | 3 --- include/sound/rt5682s.h | 3 --- sound/soc/codecs/rt5640.c | 55 ++++++++++++------------------------------- sound/soc/codecs/rt5640.h | 2 +- sound/soc/codecs/rt5665.c | 17 +++++++------ sound/soc/codecs/rt5668.c | 17 +++++++------ sound/soc/codecs/rt5682-i2c.c | 11 ++++----- sound/soc/codecs/rt5682-sdw.c | 4 ++++ sound/soc/codecs/rt5682.c | 20 ++++++++++++---- sound/soc/codecs/rt5682.h | 3 +++ sound/soc/codecs/rt5682s.c | 16 ++++++------- sound/soc/codecs/rt5682s.h | 2 ++ 14 files changed, 67 insertions(+), 91 deletions(-) --- base-commit: c5b06b35a73b80f1af931c2371de084fa77b5c23 change-id: 20230817-descriptors-asoc-rt-9c072934d376 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

2 7

[PATCH v1] ASoC: tas2781: fixed register access error when switching to other chips
by Shenghao Ding 18 Aug '23

18 Aug '23

fixed register access error when switching to other tas2781 -- refresh the page inside regmap on the switched tas2781 Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Changes in v1: - fixed register access error when switching to other tas2781 --- sound/soc/codecs/tas2781-comlib.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index a88c6c28a394..ffb26e4a7e2f 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -57,16 +57,17 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, if (client->addr != tasdev->dev_addr) { client->addr = tasdev->dev_addr; - if (tasdev->cur_book == book) { - ret = regmap_write(map, - TASDEVICE_PAGE_SELECT, 0); - if (ret < 0) { - dev_err(tas_priv->dev, "%s, E=%d\n", - __func__, ret); - goto out; - } + /* All tas2781s share the same regmap, clear the page + * inside regmap once switching to another tas2781. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", + __func__, ret); + goto out; } - goto out; } if (tasdev->cur_book != book) { -- 2.34.1

2 1

[PATCH v2 0/2] ASoC: cs35l56: Update ACPI HID and property
by Richard Fitzgerald 18 Aug '23

18 Aug '23

These two patches add an ACPI HID and update the way the platform- specific firmware identifier is extracted from the ACPI. CHANGES SINCE V1: - Rebased to apply on v6.4 and v6.5. - Change kstrdup() to devm_kstrdup() To apply on v6.6 a 1-line change is needed: static int cs35l56_get_firmware_uid(struct cs35l56_private *cs35l56) { - struct device *dev = cs35l56->dev; + struct device *dev = cs35l56->base.dev; Maciej Strozek (1): ASoC: cs35l56: Read firmware uuid from a device property instead of _SUB Simon Trimmer (1): ASoC: cs35l56: Add an ACPI match table sound/soc/codecs/cs35l56-i2c.c | 9 +++++++++ sound/soc/codecs/cs35l56-spi.c | 9 +++++++++ sound/soc/codecs/cs35l56.c | 31 ++++++++++++------------------- 3 files changed, 30 insertions(+), 19 deletions(-) -- 2.30.2

2 3

[PATCH 0/2] ASoC: cs35l56: Update ACPI HID and property
by Richard Fitzgerald 18 Aug '23

18 Aug '23

These two patches add an ACPI HID and update the way the platform- specific firmware identifier is extracted from the ACPI. Maciej Strozek (1): ASoC: cs35l56: Read firmware uuid from a device property instead of _SUB Simon Trimmer (1): ASoC: cs35l56: Add an ACPI match table sound/soc/codecs/cs35l56-i2c.c | 9 +++++++++ sound/soc/codecs/cs35l56-spi.c | 9 +++++++++ sound/soc/codecs/cs35l56.c | 31 ++++++++++++++----------------- 3 files changed, 32 insertions(+), 17 deletions(-) -- 2.30.2

2 6

[PATCH v2 01/11] ASoC: dt-bindings: Add common sound card properties
by Krzysztof Kozlowski 18 Aug '23

18 Aug '23

Almost every board machine / sound cards has "audio-routing" and "model" properties, so move them to common schema to have only one definition of these properties. For amlogic,gx-sound-card, drop the minItems: 2 from the "audio-routing", because any limit here - lower or upper- is rather meaningless. This will also fix `dtbs_check` warnings like: meson-gxm-s912-libretech-pc.dtb: sound: audio-routing: ['AU2 INL', 'ACODEC LOLN', 'AU2 INR', 'ACODEC LORN', '7J4-14 LEFT', 'AU2 OUTL', '7J4-11 RIGHT', 'AU2 OUTR'] is too long Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v2: 1. Drop minItems/maxItems from amlogic,gx-sound-card --- .../sound/amlogic,axg-sound-card.yaml | 15 +++-------- .../bindings/sound/amlogic,gx-sound-card.yaml | 17 +++--------- .../devicetree/bindings/sound/fsl,rpmsg.yaml | 17 +++--------- .../bindings/sound/google,sc7180-trogdor.yaml | 17 +++--------- .../sound/google,sc7280-herobrine.yaml | 17 +++--------- .../bindings/sound/imx-audio-card.yaml | 18 +++---------- .../bindings/sound/sound-card-common.yaml | 27 +++++++++++++++++++ 7 files changed, 50 insertions(+), 78 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sound-card-common.yaml diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml index bf1234550343..5db718e4d0e7 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml @@ -9,6 +9,9 @@ title: Amlogic AXG sound card maintainers: - Jerome Brunet <jbrunet(a)baylibre.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: const: amlogic,axg-sound-card @@ -17,23 +20,12 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle-array description: list of auxiliary devices - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: - A list of the connections between audio components. Each entry is a - pair of strings, the first being the connection's sink, the second - being the connection's source. - audio-widgets: $ref: /schemas/types.yaml#/definitions/non-unique-string-array description: A list off component DAPM widget. Each entry is a pair of strings, the first being the widget type, the second being the widget name - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - patternProperties: "^dai-link-[0-9]+$": type: object @@ -108,7 +100,6 @@ patternProperties: - sound-dai required: - - model - dai-link-0 unevaluatedProperties: false diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index b358fd601ed3..d4277d342e69 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -9,6 +9,9 @@ title: Amlogic GX sound card maintainers: - Jerome Brunet <jbrunet(a)baylibre.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: items: @@ -18,14 +21,6 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle-array description: list of auxiliary devices - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - minItems: 2 - description: |- - A list of the connections between audio components. Each entry is a - pair of strings, the first being the connection's sink, the second - being the connection's source. - audio-widgets: $ref: /schemas/types.yaml#/definitions/non-unique-string-array minItems: 2 @@ -33,10 +28,6 @@ properties: A list off component DAPM widget. Each entry is a pair of strings, the first being the widget type, the second being the widget name - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - patternProperties: "^dai-link-[0-9]+$": type: object @@ -86,7 +77,7 @@ required: - model - dai-link-0 -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml index 6df0e03a1d4b..188f38baddec 100644 --- a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml @@ -17,6 +17,9 @@ description: | such as SAI, MICFIL, .etc through building rpmsg channels between Cortex-A and Cortex-M. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: @@ -27,10 +30,6 @@ properties: - fsl,imx8ulp-rpmsg-audio - fsl,imx93-rpmsg-audio - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - clocks: items: - description: Peripheral clock for register access @@ -66,13 +65,6 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle to a node of audio codec - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: | - A list of the connections between audio components. Each entry is a - pair of strings, the first being the connection's sink, the second - being the connection's source. - fsl,enable-lpa: $ref: /schemas/types.yaml#/definitions/flag description: enable low power audio path. @@ -101,9 +93,8 @@ properties: required: - compatible - - model -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml index ba5b7728cf33..bac940553965 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -13,23 +13,15 @@ maintainers: description: This binding describes the SC7180 sound card which uses LPASS for audio. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: - google,sc7180-trogdor - google,sc7180-coachz - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: - A list of the connections between audio components. Each entry is a - pair of strings, the first being the connection's sink, the second - being the connection's source. - - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - "#address-cells": const: 1 @@ -86,11 +78,10 @@ patternProperties: required: - compatible - - model - "#address-cells" - "#size-cells" -additionalProperties: false +unevaluatedProperties: false examples: diff --git a/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml index 0b1a01a4c14e..ec4b6e547ca6 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml @@ -13,22 +13,14 @@ maintainers: description: This binding describes the SC7280 sound card which uses LPASS for audio. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: - google,sc7280-herobrine - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: - A list of the connections between audio components. Each entry is a - pair of strings, the first being the connection's sink, the second - being the connection's source. - - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - "#address-cells": const: 1 @@ -97,11 +89,10 @@ patternProperties: required: - compatible - - model - "#address-cells" - "#size-cells" -additionalProperties: false +unevaluatedProperties: false examples: diff --git a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml index b6f5d486600e..f7ad5ea2491e 100644 --- a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml +++ b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml @@ -9,23 +9,14 @@ title: NXP i.MX audio sound card. maintainers: - Shengjiu Wang <shengjiu.wang(a)nxp.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: - fsl,imx-audio-card - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: - A list of the connections between audio components. Each entry is a - pair of strings, the first being the connection's sink, the second - being the connection's source. Valid names could be power supplies, - MicBias of codec and the jacks on the board. - patternProperties: ".*-dai-link$": description: @@ -84,9 +75,8 @@ patternProperties: required: - compatible - - model -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/sound-card-common.yaml b/Documentation/devicetree/bindings/sound/sound-card-common.yaml new file mode 100644 index 000000000000..3a941177f684 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sound-card-common.yaml @@ -0,0 +1,27 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sound-card-common.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Board Sound Card Common Properties + +maintainers: + - Mark Brown <broonie(a)kernel.org> + +properties: + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + +required: + - model + +additionalProperties: true -- 2.34.1

4 20

[PATCH 00/12] ASoC: dt-bindings: Add sound-card-common.yaml
by Krzysztof Kozlowski 18 Aug '23

18 Aug '23

Hi, Almost every board machine / sound cards has "audio-routing" and "model" properties, so we can make things simpler by introducing one common binding. It is also expected that given property has only one definition. If the patchset looks good, then the next steps will be: 1. Convert more vendor audio-routing properties to audio-routing, 2. Add dai-links children to common binding. Best regards, Krzysztof Krzysztof Kozlowski (12): ASoC: dt-bindings: amlogic,gx-sound-card: correct maxItems constraints ASoC: dt-bindings: Add common sound card properties ASoC: dt-bindings: mediatek,mt8188-mt6359: use common sound card ASoC: dt-bindings: samsung,aries-wm8994: use common sound card ASoC: dt-bindings: samsung,midas-audio: use common sound card ASoC: dt-bindings: samsung,odroid: use common sound card ASoC: dt-bindings: samsung,tm2: use common sound card ASoC: samsung: odroid: use of_property_present to check for property ASoC: samsung: aries_wm8994: parse audio-routing ASoC: samsung: midas_wm1811: parse audio-routing ASoC: samsung: odroid: parse audio-routing ASoC: samsung: tm2_wm5110: parse audio-routing .../sound/amlogic,axg-sound-card.yaml | 15 +++-------- .../bindings/sound/amlogic,gx-sound-card.yaml | 16 +++++------ .../devicetree/bindings/sound/fsl,rpmsg.yaml | 17 +++--------- .../bindings/sound/google,sc7180-trogdor.yaml | 17 +++--------- .../sound/google,sc7280-herobrine.yaml | 17 +++--------- .../bindings/sound/imx-audio-card.yaml | 18 +++---------- .../sound/mediatek,mt8188-mt6359.yaml | 17 +++++------- .../bindings/sound/samsung,aries-wm8994.yaml | 16 +++++------ .../bindings/sound/samsung,midas-audio.yaml | 16 +++++------ .../bindings/sound/samsung,odroid.yaml | 14 +++++----- .../bindings/sound/samsung,tm2.yaml | 16 +++++------ .../bindings/sound/sound-card-common.yaml | 27 +++++++++++++++++++ sound/soc/samsung/aries_wm8994.c | 10 ++++--- sound/soc/samsung/midas_wm1811.c | 10 ++++--- sound/soc/samsung/odroid.c | 15 ++++++----- sound/soc/samsung/tm2_wm5110.c | 10 ++++--- 16 files changed, 119 insertions(+), 132 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sound-card-common.yaml -- 2.34.1

5 30

[PATCH v1] ALSA: hda: cs35l41: Support systems with missing _DSD properties
by Stefan Binding 18 Aug '23

18 Aug '23

Some systems using CS35L41 with HDA were released without some required _DSD properties in ACPI. To support these special cases, add an api to configure the correct properties for systems with this issue. This initial commit moves the no _DSD support for Lenovo Legion Laptops (CLSA0100, CLSA0101) into a new framework which can be extended to support additional laptops in the future. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/Makefile | 2 +- sound/pci/hda/cs35l41_hda.c | 65 ++++++------------------- sound/pci/hda/cs35l41_hda.h | 1 + sound/pci/hda/cs35l41_hda_property.c | 73 ++++++++++++++++++++++++++++ sound/pci/hda/cs35l41_hda_property.h | 18 +++++++ 5 files changed, 108 insertions(+), 51 deletions(-) create mode 100644 sound/pci/hda/cs35l41_hda_property.c create mode 100644 sound/pci/hda/cs35l41_hda_property.h diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index c6e6509e7b8e..5506255be895 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -28,7 +28,7 @@ snd-hda-codec-via-objs := patch_via.o snd-hda-codec-hdmi-objs := patch_hdmi.o hda_eld.o # side codecs -snd-hda-scodec-cs35l41-objs := cs35l41_hda.o +snd-hda-scodec-cs35l41-objs := cs35l41_hda.o cs35l41_hda_property.o snd-hda-scodec-cs35l41-i2c-objs := cs35l41_hda_i2c.o snd-hda-scodec-cs35l41-spi-objs := cs35l41_hda_spi.o snd-hda-scodec-cs35l56-objs := cs35l56_hda.o diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 825e551be9bb..f9b77353c266 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -19,6 +19,7 @@ #include "hda_component.h" #include "cs35l41_hda.h" #include "hda_cs_dsp_ctl.h" +#include "cs35l41_hda_property.h" #define CS35L41_FIRMWARE_ROOT "cirrus/" #define CS35L41_PART "cs35l41" @@ -1315,8 +1316,7 @@ static int cs35l41_hda_apply_properties(struct cs35l41_hda *cs35l41) return cs35l41_hda_channel_map(cs35l41->dev, 0, NULL, 1, &hw_cfg->spk_pos); } -static int cs35l41_get_speaker_id(struct device *dev, int amp_index, - int num_amps, int fixed_gpio_id) +int cs35l41_get_speaker_id(struct device *dev, int amp_index, int num_amps, int fixed_gpio_id) { struct gpio_desc *speaker_id_desc; int speaker_id = -ENODEV; @@ -1370,49 +1370,6 @@ static int cs35l41_get_speaker_id(struct device *dev, int amp_index, return speaker_id; } -/* - * Device CLSA010(0/1) doesn't have _DSD so a gpiod_get by the label reset won't work. - * And devices created by serial-multi-instantiate don't have their device struct - * pointing to the correct fwnode, so acpi_dev must be used here. - * And devm functions expect that the device requesting the resource has the correct - * fwnode. - */ -static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physdev, int id, - const char *hid) -{ - struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; - - /* check I2C address to assign the index */ - cs35l41->index = id == 0x40 ? 0 : 1; - cs35l41->channel_index = 0; - cs35l41->reset_gpio = gpiod_get_index(physdev, NULL, 0, GPIOD_OUT_HIGH); - cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, 0, 0, 2); - hw_cfg->spk_pos = cs35l41->index; - hw_cfg->gpio2.func = CS35L41_INTERRUPT; - hw_cfg->gpio2.valid = true; - hw_cfg->valid = true; - - if (strncmp(hid, "CLSA0100", 8) == 0) { - hw_cfg->bst_type = CS35L41_EXT_BOOST_NO_VSPK_SWITCH; - } else if (strncmp(hid, "CLSA0101", 8) == 0) { - hw_cfg->bst_type = CS35L41_EXT_BOOST; - hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; - hw_cfg->gpio1.valid = true; - } else { - /* - * Note: CLSA010(0/1) are special cases which use a slightly different design. - * All other HIDs e.g. CSC3551 require valid ACPI _DSD properties to be supported. - */ - dev_err(cs35l41->dev, "Error: ACPI _DSD Properties are missing for HID %s.\n", hid); - hw_cfg->valid = false; - hw_cfg->gpio1.valid = false; - hw_cfg->gpio2.valid = false; - return -EINVAL; - } - - return 0; -} - static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, int id) { struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; @@ -1438,12 +1395,17 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i sub = NULL; cs35l41->acpi_subsystem_id = sub; + ret = cs35l41_add_dsd_properties(cs35l41, physdev, id, hid); + if (!ret) { + dev_info(cs35l41->dev, "Using extra _DSD properties, bypassing _DSD in ACPI\n"); + goto put_physdev; + } + property = "cirrus,dev-index"; ret = device_property_count_u32(physdev, property); - if (ret <= 0) { - ret = cs35l41_no_acpi_dsd(cs35l41, physdev, id, hid); - goto err_put_physdev; - } + if (ret <= 0) + goto err; + if (ret > ARRAY_SIZE(values)) { ret = -EINVAL; goto err; @@ -1533,7 +1495,10 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i err: dev_err(cs35l41->dev, "Failed property %s: %d\n", property, ret); -err_put_physdev: + hw_cfg->valid = false; + hw_cfg->gpio1.valid = false; + hw_cfg->gpio2.valid = false; +put_physdev: put_device(physdev); return ret; diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index bdb35f3be68a..b93bf762976e 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -83,5 +83,6 @@ extern const struct dev_pm_ops cs35l41_hda_pm_ops; int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int irq, struct regmap *regmap); void cs35l41_hda_remove(struct device *dev); +int cs35l41_get_speaker_id(struct device *dev, int amp_index, int num_amps, int fixed_gpio_id); #endif /*__CS35L41_HDA_H__*/ diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c new file mode 100644 index 000000000000..673f23257a09 --- /dev/null +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -0,0 +1,73 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// CS35L41 ALSA HDA Property driver +// +// Copyright 2023 Cirrus Logic, Inc. +// +// Author: Stefan Binding <sbinding(a)opensource.cirrus.com> + +#include <linux/gpio/consumer.h> +#include <linux/string.h> +#include "cs35l41_hda_property.h" + +/* + * Device CLSA010(0/1) doesn't have _DSD so a gpiod_get by the label reset won't work. + * And devices created by serial-multi-instantiate don't have their device struct + * pointing to the correct fwnode, so acpi_dev must be used here. + * And devm functions expect that the device requesting the resource has the correct + * fwnode. + */ +static int lenovo_legion_no_acpi(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid) +{ + struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; + + /* check I2C address to assign the index */ + cs35l41->index = id == 0x40 ? 0 : 1; + cs35l41->channel_index = 0; + cs35l41->reset_gpio = gpiod_get_index(physdev, NULL, 0, GPIOD_OUT_HIGH); + cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, 0, 0, 2); + hw_cfg->spk_pos = cs35l41->index; + hw_cfg->gpio2.func = CS35L41_INTERRUPT; + hw_cfg->gpio2.valid = true; + hw_cfg->valid = true; + + if (strcmp(hid, "CLSA0100") == 0) { + hw_cfg->bst_type = CS35L41_EXT_BOOST_NO_VSPK_SWITCH; + } else if (strcmp(hid, "CLSA0101") == 0) { + hw_cfg->bst_type = CS35L41_EXT_BOOST; + hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; + hw_cfg->gpio1.valid = true; + } + + return 0; +} + +struct cs35l41_prop_model { + const char *hid; + const char *ssid; + int (*add_prop)(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid); +}; + +const struct cs35l41_prop_model cs35l41_prop_model_table[] = { + { "CLSA0100", NULL, lenovo_legion_no_acpi }, + { "CLSA0101", NULL, lenovo_legion_no_acpi }, + {} +}; + +int cs35l41_add_dsd_properties(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid) +{ + const struct cs35l41_prop_model *model; + + for (model = cs35l41_prop_model_table; model->hid > 0; model++) { + if (!strcmp(model->hid, hid) && + (!model->ssid || + (cs35l41->acpi_subsystem_id && + !strcmp(model->ssid, cs35l41->acpi_subsystem_id)))) + return model->add_prop(cs35l41, physdev, id, hid); + } + + return -ENOENT; +} diff --git a/sound/pci/hda/cs35l41_hda_property.h b/sound/pci/hda/cs35l41_hda_property.h new file mode 100644 index 000000000000..fd834042e2fd --- /dev/null +++ b/sound/pci/hda/cs35l41_hda_property.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * CS35L41 ALSA HDA Property driver + * + * Copyright 2023 Cirrus Logic, Inc. + * + * Author: Stefan Binding <sbinding(a)opensource.cirrus.com> + */ + +#ifndef CS35L41_HDA_PROP_H +#define CS35L41_HDA_PROP_H + +#include <linux/device.h> +#include "cs35l41_hda.h" + +int cs35l41_add_dsd_properties(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid); +#endif /* CS35L41_HDA_PROP_H */ -- 2.34.1

3 2

[PATCH v4 0/6] ASoC: mediatek: Add support for MT7986 SoC
by Maso Huang 17 Aug '23

17 Aug '23

Changes in v4: - fix typo of authors and copyright - refine with bitfield helper for readability [2/6] - refine with human readable error message [3/6] - use "mt7986-wm8960-sound" as compatible string [4/6] [5/6] - refine properties based on reviewer's suggestions [5/6] - remove assigned-clocks and assigned-clocks-parents [6/6] - constrain clocks per variants [6/6] Changes in v3: - merge clk api to mt7986-afe-pcm.c, remove [2/7] in v2 - refine based on reviewer's suggestions [1/6] [2/6] - fix the comment format, move in clk api, and simplify based on reviewer's suggestions [3/6] - remove redundant prefix in dt-binding [6/6] Changes in v2: - v1 title: [PATCH 0/7] ASoC: mediatek: Add support for MT79xx SoC - add missing maintainers - rename mt79xx to mt7986 in all files - use clk bulk api in mt7986-afe-clk.c [2/7] - refine mt79xx-afe-pcm.c based on reviewer's suggestions [4/7] - refine dt-binding files based on reviewer's suggestions [6/7] [7/7] - transpose [3/7] and [4/7] in v1 to fix test build errors Maso Huang (6): ASoC: mediatek: mt7986: add common header ASoC: mediatek: mt7986: support etdm in platform driver ASoC: mediatek: mt7986: add platform driver ASoC: mediatek: mt7986: add machine driver with wm8960 ASoC: dt-bindings: mediatek,mt7986-wm8960: add mt7986-wm8960 document ASoC: dt-bindings: mediatek,mt7986-afe: add audio afe document .../bindings/sound/mediatek,mt7986-afe.yaml | 160 +++++ .../sound/mediatek,mt7986-wm8960.yaml | 67 ++ sound/soc/mediatek/Kconfig | 20 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt7986/Makefile | 9 + sound/soc/mediatek/mt7986/mt7986-afe-common.h | 49 ++ sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 622 ++++++++++++++++++ sound/soc/mediatek/mt7986/mt7986-dai-etdm.c | 411 ++++++++++++ sound/soc/mediatek/mt7986/mt7986-reg.h | 196 ++++++ sound/soc/mediatek/mt7986/mt7986-wm8960.c | 196 ++++++ 10 files changed, 1731 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt7986-afe.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml create mode 100644 sound/soc/mediatek/mt7986/Makefile create mode 100644 sound/soc/mediatek/mt7986/mt7986-afe-common.h create mode 100644 sound/soc/mediatek/mt7986/mt7986-afe-pcm.c create mode 100644 sound/soc/mediatek/mt7986/mt7986-dai-etdm.c create mode 100644 sound/soc/mediatek/mt7986/mt7986-reg.h create mode 100644 sound/soc/mediatek/mt7986/mt7986-wm8960.c -- 2.18.0

4 9

[PATCH v2] ASoC: tas2783: Add source files for tas2783 soundwire driver
by Baojun Xu 17 Aug '23

17 Aug '23

Add source file and header file for tas2783 driver. Update Kconfig and Makefile for tas2783 driver. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- Change in v2: - change all registers access to regmap. - change header file include order. - remove tas2783_sdw_read(). - remove regmap_bus and relative function. - change custom access to regmap. - return if get calibration data from EFI failed. - remove mutex protect for null function. - remove tasdevice_startup() as no actions needed. - remove sysclk function as no action needed. - remove tasdevice_mute(). - remove crc8 look up table generate. - remove first_hw_init flag. - change some define name to TAS2783_XXX. - remove SMS_HTONL(), use le_to_cpup(). - remove enum channel define. - change struct and element name for FW. - remove global address elements. - remove irq and gpio in structure. - remove tasdevice structure. --- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 872 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 113 +++++ 4 files changed, 999 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c2de4ee72183..a4d334faca5e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -240,6 +240,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1758,6 +1759,17 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + select REGMAP + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + Note the TAS2783 driver implements a flexible and configurable + algo coff setting, for one, two, even multiple TAS2783 chips. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b48a9a323b84..3298dfa3b9d5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -276,6 +276,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -648,6 +649,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..f916ac9c5520 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,871 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier. +// +// Copyright (C) 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2783 driver implements a flexible and configurable +// algo coefficient setting for single TAS2783 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#include <linux/crc32.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2783.h" + +static const unsigned int tas2783_calibration_reg[] = { + TAS2783_CALIBRATION_RE, + TAS2783_CALIBRATION_RE_LOW, + TAS2783_CALIBRATION_INV_RE, + TAS2783_CALIBRATION_POW, + TAS2783_CALIBRATION_TLIMIT, + 0, +}; + +static const struct reg_default tas2783_reg_defaults[] = { + // Default values for ROM mode. Activated. + { 0x8002, 0x1a}, // Put AMP in power down mode, B0P0R02. + { 0x800e, 0x44}, // VSense from slot 4, B0P0R0E. + { 0x800f, 0x40}, // ISense from slot 0, B0P0R0E. + { 0x8097, 0xc8}, // SARBurstMask = 0, B0P1R17. + { 0x80b5, 0x74}, + { 0x8099, 0x20}, + { 0xfe8d, 0x0d}, + { 0xfebe, 0x4a}, + { 0x8230, 0x00}, + { 0x8231, 0x00}, + { 0x8232, 0x00}, + { 0x8233, 0x01}, // YMEM[70] = 1. B0P4R30. + { 0x8418, 0x00}, // 0 dB in B0P8R18. + { 0x8419, 0x00}, + { 0x841a, 0x00}, + { 0x841b, 0x00}, + { 0x8428, 0x40}, // Unmute channel, B0P8R28. + { 0x8429, 0x00}, + { 0x842a, 0x00}, + { 0x842b, 0x00}, + { 0x8548, 0x00}, // 0dB in B0PAR48. + { 0x8549, 0x00}, + { 0x854a, 0x00}, + { 0x854b, 0x00}, + { 0x8558, 0x40}, // unmute channel, B0PAR58. + { 0x8559, 0x00}, + { 0x855a, 0x00}, + { 0x855b, 0x00}, + { 0x800a, 0x3a}, // Enable both channel in B0P0RA. + { 0x805c, 0xd9}, // Enable clock detected in B0P0R5C. + { 0x8002, 0x00}, // Put AMP in active mode, B0P0R02. + /* Below register was used to select function, entity, CS, CN. */ + { SDW_SDCA_CTL(1, 1, 2, 0), 0}, // 0x40400088. + { SDW_SDCA_CTL(1, 1, 1, 0), 0}, // 0x40400090. + { SDW_SDCA_CTL(1, 2, 1, 0), 0}, // 0x40400108. +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0xc000: // Page 0 ~ 127. + case 0xfe80 ... 0xfeff: // Page 253. + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_UDMPU21, + TAS2783_SDCA_CTL_UDMPU_CLUSTER, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, TAS2783_DEVICE_CHANNEL_LEFT): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, TAS2783_DEVICE_CHANNEL_RIGHT): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_PDE23, + TAS2783_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_PDE22, + TAS2783_SDCA_CTL_REQ_POWER_STATE, 0): + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8001: + // Only reset register was volatiled. + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x41008000 + TASDEVICE_REG(0xa1, 0x60, 0x7f), + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static int tasdevice_clamp(int val, int max, unsigned int invert) +{ + /* Keep in valid area, out of range value don't care. */ + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + return val; +} + +static int tas2783_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + int val = 0, ret; + + if (!map) { + ret = -EINVAL; + dev_err(tas_dev->dev, "%s, regmap doesn't exist.\n", + __func__); + goto out; + } + /* Read the primary device as the whole */ + ret = regmap_read(map, mc->reg, &val); + dev_dbg(tas_dev->dev, "%s, get digital vol %d from %x with %d\n", + __func__, val, mc->reg, ret); + if (ret) { + dev_err(tas_dev->dev, "%s, get digital vol error %x.\n", + __func__, ret); + goto out; + } + ucontrol->value.integer.value[0] = + tasdevice_clamp(val, mc->max, mc->invert); + +out: + return ret; +} + +static int tas2783_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + int val, ret; + + if (!map) { + dev_err(tas_dev->dev, "%s, regmap doesn't exist.\n", + __func__); + return -EINVAL; + } + val = tasdevice_clamp(ucontrol->value.integer.value[0], + mc->max, mc->invert); + + ret = regmap_write(map, mc->reg, val); + if (ret != 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + return ret; + } + dev_dbg(tas_dev->dev, "%s, Put digital vol %d into %x.\n", + __func__, val, mc->reg); + + return 1; +} + +static int tas2783_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + unsigned char mask; + int ret, val; + + if (!map) { + dev_err(tas_dev->dev, "%s, regmap doesn't exist.\n", + __func__); + return -EINVAL; + } + /* Read the primary device */ + ret = regmap_read(map, mc->reg, &val); + dev_dbg(tas_dev->dev, "%s, get AMP vol %d from %x with %d\n", + __func__, val, mc->reg, ret); + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + ucontrol->value.integer.value[0] = tasdevice_clamp(val, mc->max, + mc->invert); + + return ret; +} + +static int tas2783_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + unsigned char mask; + int val, ret; + + if (!map) { + dev_err(tas_dev->dev, "%s, regmap doesn't exist.\n", + __func__); + return -EINVAL; + } + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = tasdevice_clamp(ucontrol->value.integer.value[0], mc->max, + mc->invert); + ret = regmap_update_bits(map, mc->reg, mask, val << mc->shift); + if (ret != 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + return ret; + } + + dev_dbg(tas_dev->dev, "wr amp %x into 0x%08x by %x shift %d.\n", + val, mc->reg, mask, mc->shift); + + return ret; +} + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, 0, tas2783_amp_getvol, + tas2783_amp_putvol, amp_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, tas2783_digital_getvol, + tas2783_digital_putvol, dvc_tlv), +}; + +static void tas2783_apply_calib( + struct tasdevice_priv *tas_priv, unsigned int *cali_data) +{ + struct regmap *map = tas_priv->regmap; + int i = 0, ret; + u8 *reg_start; + + if (!map) { + dev_err(tas_priv->dev, "%s, regmap doesn't exist.\n", + __func__); + return; + } + if (!tas_priv->sdw_peripheral) { + dev_err(tas_priv->dev, "%s, slaver doesn't exist.\n", + __func__); + return; + } + if ((tas_priv->sdw_peripheral->id.unique_id < + TAS2783_DEVICE_ID_START) || + (tas_priv->sdw_peripheral->id.unique_id > + TAS2783_DEVICE_ID_END)) { + dev_err(tas_priv->dev, "%s, wrong unique id: %x.\n", + __func__, tas_priv->sdw_peripheral->id.unique_id); + return; + } + reg_start = + (u8 *)(cali_data+(tas_priv->sdw_peripheral->id.unique_id - + TAS2783_DEVICE_ID_START)*TAS2783_CALIBRATION_SIZE); + while (tas2783_calibration_reg[i] != 0) { + ret = regmap_bulk_write(map, tas2783_calibration_reg[i], + reg_start + i, 4); + if (ret != 0) { + dev_err(tas_priv->dev, "Cali failed %x:%d\n", + tas2783_calibration_reg[i], ret); + break; + } + i++; + } +} + +static int tas2783_calibration(struct tasdevice_priv *tas_priv) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = TAS2783_CALIDATA_NAME; + struct tm *tm = &tas_priv->tm; + unsigned int attr, crc; + unsigned int *tmp_val; + efi_status_t status; + + tas_priv->cali_data.total_sz = TAS2783_MAX_CALIDATA_SIZE; + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, tas_priv->cali_data.data); + dev_dbg(tas_priv->dev, "cali get %lx bytes with result : %ld\n", + tas_priv->cali_data.total_sz, status); + if (status == EFI_BUFFER_TOO_SMALL) { + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, + tas_priv->cali_data.data); + dev_dbg(tas_priv->dev, "cali get %lx bytes result:%ld\n", + tas_priv->cali_data.total_sz, status); + } + /* Failed got calibration data from EFI. */ + if (status != 0) { + dev_dbg(tas_priv->dev, "cali get %lx error with:%ld\n", + tas_priv->cali_data.total_sz, status); + return 0; + } + /* Print all content of calibration data for debug. */ + for (int i = 0; i < tas_priv->cali_data.total_sz; i += 4) { + dev_dbg(tas_priv->dev, "cali get %02x %02x %02x %02x", + tas_priv->cali_data.data[i], + tas_priv->cali_data.data[i+1], + tas_priv->cali_data.data[i+2], + tas_priv->cali_data.data[i+3]); + } + + tmp_val = (unsigned int *)tas_priv->cali_data.data; + + crc = crc32(~0, tas_priv->cali_data.data, 84) ^ ~0; + dev_dbg(tas_priv->dev, "cali crc 0x%08x PK tmp_val 0x%08x\n", + crc, tmp_val[21]); + + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + dev_dbg(tas_priv->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", + tm->tm_year, tm->tm_mon, tm->tm_mday, + tm->tm_hour, tm->tm_min, tm->tm_sec); + tas2783_apply_calib(tas_priv, tmp_val); + } else { + dev_dbg(tas_priv->dev, "CRC error!\n"); + tas_priv->cali_data.total_sz = 0; + } + + return 0; +} + +static void tasdevice_rca_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct regmap *map = tas_dev->regmap; + struct tas2783_firmware_node *p; + int offset = 0, num_nodes = 0, img_sz, ret; + unsigned char *buf; + + mutex_lock(&tas_dev->codec_lock); + + if (!fmw || !fmw->data) { + dev_err(tas_dev->dev, + "Failed to read %s, no side - effect on driver running\n", + tas_dev->rca_binaryname); + ret = -EINVAL; + goto out; + } + if (!map) { + dev_err(tas_dev->dev, "Failed to load regmap.\n"); + ret = -EINVAL; + goto out; + } + buf = (unsigned char *)fmw->data; + + img_sz = le32_to_cpup((__le32 *)&buf[offset]); + dev_dbg(tas_dev->dev, "Got %x:%lx.\n", img_sz, fmw->size); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_err(tas_dev->dev, + "Size not match, %d %u", (int)fmw->size, img_sz); + ret = -EINVAL; + goto out; + } + + while ((offset < img_sz) && (num_nodes < TAS2783_MAX_NODES)) { + /* Store firmware into context of driver. */ + p = (struct tas2783_firmware_node *)(buf + offset); + tas_dev->firmware_node[num_nodes].vendor_id = + p->vendor_id; + tas_dev->firmware_node[num_nodes].file_id = p->file_id; + tas_dev->firmware_node[num_nodes].version_id = + p->version_id; + tas_dev->firmware_node[num_nodes].length = p->length; + tas_dev->firmware_node[num_nodes].download_addr = + p->download_addr; + tas_dev->firmware_node[num_nodes].start_addr = + ((char *)p) + sizeof(unsigned int)*5; + + ret = regmap_bulk_write(map, p->download_addr, + p->start_addr, p->length); + dev_dbg(tas_dev->dev, "Wr %d :%x:%x:%x:%x:%x:%x %d.\n", + num_nodes, p->vendor_id, p->file_id, + p->version_id, p->length, p->download_addr, + p->start_addr[0], ret); + + offset += sizeof(unsigned int)*5 + p->length; + num_nodes++; + } + + tas2783_calibration(tas_dev); + +out: + mutex_unlock(&tas_dev->codec_lock); + if (fmw) + release_firmware(fmw); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"} +}; + +static int tasdevice_set_sdw_stream( + struct snd_soc_dai *dai, void *sdw_stream, int direction) +{ + struct sdw_stream_data *stream; + + if (!sdw_stream) + return 0; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = sdw_stream; + + /* Use tx_mask or rx_mask to set dma_data */ + snd_soc_dai_dma_data_set(dai, direction, stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_data *stream; + int retval; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!tasdevice->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + retval = sdw_stream_add_slave(tasdevice->sdw_peripheral, + &stream_config, &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tasdevice->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tasdevice->sdw_peripheral, stream->sdw_stream); + return 0; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static void tas2783_reset(struct tasdevice_priv *tas_dev) +{ + struct regmap *map = tas_dev->regmap; + unsigned char value_sdw; + int ret; + + if (!map) { + dev_err(tas_dev->dev, "Failed to load regmap.\n"); + return; + } + value_sdw = TAS2873_REG_SWRESET_RESET; + ret = regmap_write(map, TAS2873_REG_SWRESET, value_sdw); + dev_dbg(tas_dev->dev, "%s TAS2783 was reseted %d.\n", + __func__, ret); + usleep_range(1000, 1050); +} + +static int tasdevice_codec_probe(struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + int ret; + + dev_dbg(tas_dev->dev, "%s called for TAS2783 start.\n", + __func__); + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + + tas2783_reset(tas_dev); + + tas_dev->codec = codec; + + scnprintf(tas_dev->rca_binaryname, 64, "MY_SWFT_x%01x.bin", + tas_dev->sdw_peripheral->id.unique_id); + + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_dev->rca_binaryname, tas_dev->dev, GFP_KERNEL, + tas_dev, tasdevice_rca_ready); + dev_dbg(tas_dev->dev, + "%s: request_firmware %x open status: 0x%08x\n", + __func__, tas_dev->sdw_peripheral->id.unique_id, ret); + + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); + + dev_dbg(tas_dev->dev, "%s was called end.\n", __func__); + return ret; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_codec_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + mutex_init(&tas_dev->dev_lock); + + dev_set_drvdata(tas_dev->dev, tas_dev); + + mutex_init(&tas_dev->codec_lock); + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:0x%x\n", + __func__, ret); + } + dev_dbg(tas_dev->dev, "%s was called for TAS2783.\n", __func__); + + return ret; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + struct sdw_dpn_prop *dpn; + unsigned long addr; + int nval, i, j; + u32 bit; + + prop->scp_int1_mask = + SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x04; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + dev_dbg(&slave->dev, "%s was performed.\n", __func__); + + return 0; +} + +static int tasdevice_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct tasdevice_priv *tasdevice = dev_get_drvdata(dev); + + if (tasdevice->hw_init) + return 0; + + /* PM runtime is only enabled when + * a Slave reports as Attached + * set autosuspend parameters + */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + + pm_runtime_get_noresume(&slave->dev); + + /* Mark Slave initialization complete */ + tasdevice->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + return 0; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tasdevice = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tasdevice->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tasdevice->hw_init = false; + + /* Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tasdevice->hw_init || tasdevice->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static void tasdevice_remove(struct tasdevice_priv *tas_dev) +{ + snd_soc_unregister_component(tas_dev->dev); + + mutex_destroy(&tas_dev->dev_lock); + mutex_destroy(&tas_dev->codec_lock); +} + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + int ret; + + dev_dbg(dev, "%s was called.\n", __func__); + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) { + ret = -ENOMEM; + goto out; + } + tas_dev->dev = &peripheral->dev; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_peripheral = peripheral; + tas_dev->hw_init = false; + + dev_dbg(dev, "%d chip id %x for TAS2783.\n", + peripheral->id.unique_id, tas_dev->chip_id); + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, + &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) { + ret = PTR_ERR(tas_dev->regmap); + dev_err(dev, "Failed devm_regmap_init: %d\n", ret); + goto out; + } + ret = tasdevice_init(tas_dev); + +out: + if (ret < 0 && tas_dev != NULL) + tasdevice_remove(tas_dev); + + return ret; + +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = + dev_get_drvdata(&peripheral->dev); + + if (tas_dev) + tasdevice_remove(tas_dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..09575e741400 --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,113 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier. + * + * Copyright (C) 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2783 driver implements a flexible and configurable + * algo coff setting for single TAS2783 chips. + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 +\ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1A) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x0d, 0x3c) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +#define TAS2783_CALIBRATION_SIZE 5 // int data number. + +#define TAS2783_DEVICE_ID_START 0x08 // Unique id start +#define TAS2783_DEVICE_ID_END 0x0F // Unique id end + +/* TAS2783 SDCA Control - function number */ +#define FUNC_NUM_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_PDE23 0x0C +#define TAS2783_SDCA_ENT_PDE22 0x0B +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_UDMPU21 0x10 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 + +#define TAS2783_MAX_NODES 100 +#define TAS2783_MAX_CALIDATA_SIZE 252 +#define TAS2783_CALIDATA_NAME L"CALI_DATA" + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; + unsigned char *start_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct tas2783_firmware_node firmware_node[TAS2783_MAX_NODES]; + struct snd_soc_component *component; + struct calibration_data cali_data; + struct sdw_slave *sdw_peripheral; + struct sdw_bus_params params; + struct mutex codec_lock; + struct regmap *regmap; + struct mutex dev_lock; + struct device *dev; + struct tm tm; + + enum sdw_slave_status status; + + bool hw_init; + + unsigned int chip_id; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + + void *codec; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

3 3

[PATCH v3 1/2] ALSA: hda/tas2781: Add tas2781 HDA driver
by Shenghao Ding 17 Aug '23

17 Aug '23

Create tas2781 side codec HDA driver for Lenovo Laptops. The quantity of the speakers has been define in ACPI. All of the tas2781s in the laptop will be aggregated as one audio speaker. The code supports realtek codec as the primary codec. Code offers several controls for digtial/analog gain setting during playback, and other for eq params setting in case of different audio profiles, such as music, voice, movie, etc. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Changes in v3: - oprimize the indentation - fix the ambiguous error message - The check of the given value to kcontrol's put func is in the sound/tas2781-comlib.c - In tasdevice_info_programs, corrected max value - Seperate snd_kcontrol_new tas2781_dsp_controls into two standalone kcontrol. - All the controls set as const static - Add static for page_array & rgno_array - Remove uncessary blank lines - Add descriptions for tas2781_save_calibration - remove global addr handling in the code - checking subid in switch statement in function tas2781_hda_bind - add force firmware load Kcontrol - rename the kcontrol name to be more undertandable - remove Superfluous cast in tasdevice_fw_ready - correct weird line break in function tas2781_acpi_get_i2c_resource - correct Referencing adev after acpi_dev_put() in tas2781_hda_read_acpi - As to checking the given value in tasdevice_set_profile_id, it seems done by the tasdevice_info_profile - replace strcpy with strscpy in tas2781_hda_read_acpi - rewrite the subid judgement - Add tiwai(a)suse.de into Cc list - remove the cast in tas2781_acpi_get_i2c_resource - remove else in tas2781_acpi_get_i2c_resource - fix the return value in tasdevice_set_profile_id - remove unneeded NL in tasdevice_config_get - Unifiy the comment style - remove ret = 0 in tasdevice_fw_ready - remove ret in tas2781_save_calibration - remove unused ret in tas2781_hda_playback - add force firmware load Kcontrol --- sound/pci/hda/Kconfig | 15 + sound/pci/hda/Makefile | 2 + sound/pci/hda/tas2781_hda_i2c.c | 858 ++++++++++++++++++++++++++++++++ 3 files changed, 875 insertions(+) create mode 100644 sound/pci/hda/tas2781_hda_i2c.c diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 886255a03e8b..e66257277492 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -130,6 +130,21 @@ config SND_HDA_SCODEC_CS35L41_SPI comment "Set to Y if you want auto-loading the side codec driver" depends on SND_HDA=y && SND_HDA_SCODEC_CS35L41_SPI=m +config SND_HDA_SCODEC_TAS2781_I2C + tristate "Build TAS2781 HD-audio side codec support for I2C Bus" + depends on I2C + depends on ACPI + depends on SND_SOC + select SND_SOC_TAS2781_COMLIB + select SND_SOC_TAS2781_FMWLIB + select CRC32_SARWATE + help + Say Y or M here to include TAS2781 I2C HD-audio side codec support + in snd-hda-intel driver, such as ALC287. + +comment "Set to Y if you want auto-loading the side codec driver" + depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_I2C=m + config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 00d306104484..1c76609690d6 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -32,6 +32,7 @@ snd-hda-scodec-cs35l41-objs := cs35l41_hda.o snd-hda-scodec-cs35l41-i2c-objs := cs35l41_hda_i2c.o snd-hda-scodec-cs35l41-spi-objs := cs35l41_hda_spi.o snd-hda-cs-dsp-ctls-objs := hda_cs_dsp_ctl.o +snd-hda-scodec-tas2781-i2c-objs := tas2781_hda_i2c.o # common driver obj-$(CONFIG_SND_HDA) := snd-hda-codec.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_HDA_SCODEC_CS35L41) += snd-hda-scodec-cs35l41.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L41_I2C) += snd-hda-scodec-cs35l41-i2c.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L41_SPI) += snd-hda-scodec-cs35l41-spi.o obj-$(CONFIG_SND_HDA_CS_DSP_CONTROLS) += snd-hda-cs-dsp-ctls.o +obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_I2C) += snd-hda-scodec-tas2781-i2c.o # this must be the last entry after codec drivers; # otherwise the codec patches won't be hooked before the PCI probe diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c new file mode 100644 index 000000000000..4ccbe12ad451 --- /dev/null +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -0,0 +1,858 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA I2C driver +// +// Copyright 2023 Texas Instruments, Inc. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> + +#include <linux/acpi.h> +#include <linux/crc8.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/firmware.h> +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <sound/hda_codec.h> +#include <sound/soc.h> +#include <sound/tas2781.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "hda_local.h" +#include "hda_auto_parser.h" +#include "hda_component.h" +#include "hda_jack.h" +#include "hda_generic.h" + +#define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 + +/* No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD + * Define two controls, one is Volume control callbacks, the other is + * flag setting control callbacks. + */ + +/* Volume control callbacks for tas2781 */ +#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .invert = xinvert} } + +/* Flag control callbacks for tas2781 */ +#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = xdata } + +enum calib_data { + R0_VAL = 0, + INV_R0, + R0LOW, + POWER, + TLIM, + CALIB_MAX +}; + +static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) +{ + struct tasdevice_priv *tas_priv = data; + struct acpi_resource_i2c_serialbus *sb; + + if (i2c_acpi_get_i2c_resource(ares, &sb)) { + if (tas_priv->ndev < TASDEVICE_MAX_CHANNELS && + sb->slave_address != TAS2781_GLOBAL_ADDR) { + tas_priv->tasdevice[tas_priv->ndev].dev_addr = + (unsigned int)sb->slave_address; + tas_priv->ndev++; + } + } + return 1; +} + +static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) +{ + struct acpi_device *adev; + struct device *physdev; + LIST_HEAD(resources); + const char *sub; + int ret; + + adev = acpi_dev_get_first_match_dev(hid, NULL, -1); + if (!adev) { + dev_err(p->dev, + "Failed to find an ACPI device for %s\n", hid); + return -ENODEV; + } + + ret = acpi_dev_get_resources(adev, &resources, tas2781_get_i2c_res, p); + if (ret < 0) + goto err; + + acpi_dev_free_resource_list(&resources); + strscpy(p->dev_name, hid, sizeof(p->dev_name)); + physdev = get_device(acpi_get_first_physical_node(adev)); + acpi_dev_put(adev); + + /* No side-effect to the playback even if subsystem_id is NULL*/ + sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); + if (IS_ERR(sub)) + sub = NULL; + + p->acpi_subsystem_id = sub; + + put_device(physdev); + + return 0; + +err: + dev_err(p->dev, "read acpi error, ret: %d\n", ret); + put_device(physdev); + + return ret; +} + +static void tas2781_hda_playback_hook(struct device *dev, int action) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + dev_dbg(tas_priv->dev, "%s: action = %d\n", __func__, action); + switch (action) { + case HDA_GEN_PCM_ACT_OPEN: + pm_runtime_get_sync(dev); + mutex_lock(&tas_priv->codec_lock); + tasdevice_tuning_switch(tas_priv, 0); + mutex_unlock(&tas_priv->codec_lock); + break; + case HDA_GEN_PCM_ACT_CLOSE: + mutex_lock(&tas_priv->codec_lock); + tasdevice_tuning_switch(tas_priv, 1); + mutex_unlock(&tas_priv->codec_lock); + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + break; + default: + dev_dbg(tas_priv->dev, "Playback action not supported: %d\n", + action); + break; + } +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->rcabin.ncfgs - 1; + + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + + return 0; +} + +static int tasdevice_hda_clamp(int val, int max) +{ + if (val > max) + val = max; + + if (val < 0) + val = 0; + return val; +} + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int nr_profile = ucontrol->value.integer.value[0]; + int max = tas_priv->rcabin.ncfgs - 1; + int val, ret = 0; + + val = tasdevice_hda_clamp(nr_profile, max); + + if (tas_priv->rcabin.profile_cfg_id != nr_profile) { + tas_priv->rcabin.profile_cfg_id = nr_profile; + ret = 1; + } + + return ret; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_fw->nr_programs - 1; + + return 0; +} + +static int tasdevice_info_config(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_fw->nr_configurations - 1; + + return 0; +} + +static int tasdevice_program_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + int nr_program = ucontrol->value.integer.value[0]; + int max = tas_fw->nr_programs - 1; + int val, ret = 0; + + val = tasdevice_hda_clamp(nr_program, max); + + if (tas_priv->cur_prog != nr_program) { + tas_priv->cur_prog = nr_program; + ret = 1; + } + + return ret; +} + +static int tasdevice_config_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + + return 0; +} + +static int tasdevice_config_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + int nr_config = ucontrol->value.integer.value[0]; + int max = tas_fw->nr_configurations - 1; + int val, ret = 0; + + val = tasdevice_hda_clamp(nr_config, max); + + if (tas_priv->cur_conf != nr_config) { + tas_priv->cur_conf = nr_config; + ret = 1; + } + + return ret; +} + +/* + * tas2781_digital_getvol - get the volum control + * @kcontrol: control pointer + * @ucontrol: User data + * Customer Kcontrol for tas2781 is primarily for regmap booking, paging + * depends on internal regmap mechanism. + * tas2781 contains book and page two-level register map, especially + * book switching will set the register BXXP00R7F, after switching to the + * correct book, then leverage the mechanism for paging to access the + * register. + */ +static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_digital_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_amp_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_digital_putvol. */ + return tasdevice_digital_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_amp_putvol. */ + return tasdevice_amp_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = (int)tas_priv->force_fwload_status; + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + tas_priv->force_fwload_status ? "ON" : "OFF"); + + return 0; +} + +static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + bool change, val = (bool)ucontrol->value.integer.value[0]; + + if (tas_priv->force_fwload_status == val) + change = false; + else { + change = true; + tas_priv->force_fwload_status = val; + } + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + tas_priv->force_fwload_status ? "ON" : "OFF"); + + return change; +} + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static const struct snd_kcontrol_new tas2781_prof_ctrl = { + .name = "Speaker Profile Id", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, +}; + +static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl = { + .name = "Speaker Program Id", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, +}; + +static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl = { + .name = "Speaker Config Id", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_config, + .get = tasdevice_config_get, + .put = tasdevice_config_put, +}; + +static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) +{ + static const unsigned char page_array[CALIB_MAX] = { + 0x17, 0x18, 0x18, 0x0d, 0x18 + }; + static const unsigned char rgno_array[CALIB_MAX] = { + 0x74, 0x0c, 0x14, 0x3c, 0x7c + }; + unsigned char *data; + int i, j, rc; + + for (i = 0; i < tas_priv->ndev; i++) { + data = tas_priv->cali_data.data + + i * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + for (j = 0; j < CALIB_MAX; j++) { + rc = tasdevice_dev_bulk_write(tas_priv, i, + TASDEVICE_REG(0, page_array[j], rgno_array[j]), + &(data[4 * j]), 4); + if (rc < 0) + dev_err(tas_priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + i, j, rc); + } + } +} + +/* Update the calibrate data, including speaker impedance, f0, etc, into algo. + * Calibrate data is done by manufacturer in the factory. These data are used + * by Algo for calucating the speaker temperature, speaker membrance excursion + * and f0 in real time during playback. + */ +static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) +{ + efi_guid_t efi_guid = EFI_GUID(0x02f9af02, 0x7734, 0x4233, 0xb4, 0x3d, + 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct tm *tm = &tas_priv->tm; + unsigned int attr, crc; + unsigned int *tmp_val; + efi_status_t status; + + /* Lenovo devices */ + if (tas_priv->catlog_id == LENOVO) + efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, 0x09, + 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + + tas_priv->cali_data.total_sz = 0; + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, tas_priv->cali_data.data); + if (status == EFI_BUFFER_TOO_SMALL) { + /* Allocate data buffer of data_size bytes */ + tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + tas_priv->cali_data.total_sz, GFP_KERNEL); + if (!tas_priv->cali_data.data) + return -ENOMEM; + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, + tas_priv->cali_data.data); + if (status != EFI_SUCCESS) + return -EINVAL; + } + + tmp_val = (unsigned int *)tas_priv->cali_data.data; + + crc = crc32(~0, tas_priv->cali_data.data, 84) ^ ~0; + dev_dbg(tas_priv->dev, "cali crc 0x%08x PK tmp_val 0x%08x\n", + crc, tmp_val[21]); + + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + dev_dbg(tas_priv->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", + tm->tm_year, tm->tm_mon, tm->tm_mday, + tm->tm_hour, tm->tm_min, tm->tm_sec); + tas2781_apply_calib(tas_priv); + } else + tas_priv->cali_data.total_sz = 0; + + return 0; +} + +static void tasdev_fw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct hda_codec *codec = tas_priv->codec; + int i, ret; + + pm_runtime_get_sync(tas_priv->dev); + mutex_lock(&tas_priv->codec_lock); + + ret = tasdevice_rca_parser(tas_priv, fmw); + if (ret) + goto out; + + ret = snd_ctl_add(codec->card, + snd_ctl_new1(&tas2781_prof_ctrl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_prof_ctrl.name, ret); + goto out; + } + + for (i = 0; i < ARRAY_SIZE(tas2781_snd_controls); i++) { + ret = snd_ctl_add(codec->card, + snd_ctl_new1(&tas2781_snd_controls[i], tas_priv)); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_snd_controls[i].name, ret); + goto out; + } + } + + tasdevice_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%04X.bin", + codec->core.subsystem_id & 0xffff); + ret = tasdevice_dsp_parser(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dspfw load %s error\n", + tas_priv->coef_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + + ret = snd_ctl_add(codec->card, + snd_ctl_new1(&tas2781_dsp_prog_ctrl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_dsp_prog_ctrl.name, ret); + goto out; + } + + ret = snd_ctl_add(codec->card, + snd_ctl_new1(&tas2781_dsp_conf_ctrl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_dsp_conf_ctrl.name, ret); + goto out; + } + + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + tasdevice_prmg_load(tas_priv, 0); + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + tas2781_save_calibration(tas_priv); + +out: + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + /*If DSP FW fail, kcontrol won't be created */ + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + } + mutex_unlock(&tas_priv->codec_lock); + if (fmw) + release_firmware(fmw); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_put_autosuspend(tas_priv->dev); +} + +static int tas2781_hda_bind(struct device *dev, struct device *master, + void *master_data) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct hda_component *comps = master_data; + struct hda_codec *codec; + unsigned int subid; + int ret; + + if (!comps || tas_priv->index < 0 || + tas_priv->index >= HDA_MAX_COMPONENTS) + return -EINVAL; + + comps = &comps[tas_priv->index]; + if (comps->dev) + return -EBUSY; + + codec = comps->codec; + subid = codec->core.subsystem_id >> 16; + + switch (subid) { + case 0x17aa: + tas_priv->catlog_id = LENOVO; + break; + default: + tas_priv->catlog_id = OTHERS; + break; + } + + pm_runtime_get_sync(dev); + + comps->dev = dev; + + strscpy(comps->name, dev_name(dev), sizeof(comps->name)); + + ret = tascodec_init(tas_priv, codec, tasdev_fw_ready); + if (ret) + return ret; + + comps->playback_hook = tas2781_hda_playback_hook; + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + + return 0; +} + +static void tas2781_hda_unbind(struct device *dev, + struct device *master, void *master_data) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct hda_component *comps = master_data; + + if (comps[tas_priv->index].dev == dev) + memset(&comps[tas_priv->index], 0, sizeof(*comps)); + + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static const struct component_ops tas2781_hda_comp_ops = { + .bind = tas2781_hda_bind, + .unbind = tas2781_hda_unbind, +}; + +static void tas2781_hda_remove(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + pm_runtime_get_sync(tas_priv->dev); + pm_runtime_disable(tas_priv->dev); + + component_del(tas_priv->dev, &tas2781_hda_comp_ops); + + pm_runtime_put_noidle(tas_priv->dev); + + tasdevice_remove(tas_priv); +} + +static int tas2781_hda_i2c_probe(struct i2c_client *clt) +{ + struct tasdevice_priv *tas_priv; + const char *device_name; + int ret; + + if (strstr(dev_name(&clt->dev), "TIAS2781")) + device_name = "TIAS2781"; + else + return -ENODEV; + + 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); + if (ret) + return dev_err_probe(tas_priv->dev, ret, + "Platform not supported\n"); + + ret = tasdevice_init(tas_priv); + if (ret) + goto err; + + pm_runtime_set_autosuspend_delay(tas_priv->dev, 3000); + pm_runtime_use_autosuspend(tas_priv->dev); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_set_active(tas_priv->dev); + pm_runtime_get_noresume(tas_priv->dev); + pm_runtime_enable(tas_priv->dev); + + pm_runtime_put_autosuspend(tas_priv->dev); + + ret = component_add(tas_priv->dev, &tas2781_hda_comp_ops); + if (ret) { + dev_err(tas_priv->dev, "Register component failed: %d\n", ret); + pm_runtime_disable(tas_priv->dev); + goto err; + } + + tas2781_reset(tas_priv); +err: + if (ret) + tas2781_hda_remove(&clt->dev); + return ret; +} + +static void tas2781_hda_i2c_remove(struct i2c_client *clt) +{ + tas2781_hda_remove(&clt->dev); +} + +static int tas2781_runtime_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int i; + + dev_dbg(tas_priv->dev, "Runtime Suspend\n"); + + mutex_lock(&tas_priv->codec_lock); + + if (tas_priv->playback_started) { + tasdevice_tuning_switch(tas_priv, 1); + tas_priv->playback_started = false; + } + + for (i = 0; i < tas_priv->ndev; i++) { + tas_priv->tasdevice[i].cur_book = -1; + tas_priv->tasdevice[i].cur_prog = -1; + tas_priv->tasdevice[i].cur_conf = -1; + } + + regcache_cache_only(tas_priv->regmap, true); + regcache_mark_dirty(tas_priv->regmap); + + mutex_unlock(&tas_priv->codec_lock); + + return 0; +} + +static int tas2781_runtime_resume(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long calib_data_sz = + tas_priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + int ret; + + dev_dbg(tas_priv->dev, "Runtime Resume\n"); + + mutex_lock(&tas_priv->codec_lock); + + regcache_cache_only(tas_priv->regmap, false); + ret = regcache_sync(tas_priv->regmap); + if (ret) { + dev_err(tas_priv->dev, + "Failed to restore register cache: %d\n", ret); + goto out; + } + + tasdevice_prmg_load(tas_priv, tas_priv->cur_prog); + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + if (tas_priv->cali_data.total_sz > calib_data_sz) + tas2781_apply_calib(tas_priv); + +out: + mutex_unlock(&tas_priv->codec_lock); + + return ret; +} + +static int tas2781_system_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + dev_dbg(tas_priv->dev, "System Suspend\n"); + + ret = pm_runtime_force_suspend(dev); + if (ret) + return ret; + + /* Shutdown chip before system suspend */ + regcache_cache_only(tas_priv->regmap, false); + tasdevice_tuning_switch(tas_priv, 1); + regcache_cache_only(tas_priv->regmap, true); + regcache_mark_dirty(tas_priv->regmap); + + /* + * Reset GPIO may be shared, so cannot reset here. + * However beyond this point, amps may be powered down. + */ + return 0; +} + +static int tas2781_system_resume(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long calib_data_sz = + tas_priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + int i, ret; + + dev_dbg(tas_priv->dev, "System Resume\n"); + + ret = pm_runtime_force_resume(dev); + if (ret) + return ret; + + mutex_lock(&tas_priv->codec_lock); + + for (i = 0; i < tas_priv->ndev; i++) { + tas_priv->tasdevice[i].cur_book = -1; + tas_priv->tasdevice[i].cur_prog = -1; + tas_priv->tasdevice[i].cur_conf = -1; + } + tas2781_reset(tas_priv); + tasdevice_prmg_load(tas_priv, tas_priv->cur_prog); + + /* If calibrated data occurs error, dsp will still work with default + * calibrated data inside algo. + */ + if (tas_priv->cali_data.total_sz > calib_data_sz) + tas2781_apply_calib(tas_priv); + mutex_unlock(&tas_priv->codec_lock); + + return 0; +} + +static const struct dev_pm_ops tas2781_hda_pm_ops = { + RUNTIME_PM_OPS(tas2781_runtime_suspend, tas2781_runtime_resume, NULL) + SYSTEM_SLEEP_PM_OPS(tas2781_system_suspend, tas2781_system_resume) +}; + +static const struct i2c_device_id tas2781_hda_i2c_id[] = { + { "tas2781-hda", 0 }, + {} +}; + +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", + .acpi_match_table = tas2781_acpi_hda_match, + .pm = &tas2781_hda_pm_ops, + }, + .id_table = tas2781_hda_i2c_id, + .probe_new = tas2781_hda_i2c_probe, + .remove = tas2781_hda_i2c_remove, +}; +module_i2c_driver(tas2781_hda_i2c_driver); + +MODULE_DESCRIPTION("TAS2781 HDA Driver"); +MODULE_AUTHOR("Shenghao Ding, TI, <shenghao-ding(a)ti.com>"); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS(SND_SOC_TAS2781_FMWLIB); -- 2.34.1

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Fix 'errno undeclared' compile error with older gcc versions
by GitHub pull_request - opened 17 Aug '23

17 Aug '23

alsa-project/alsa-gobject pull request #83 was opened from michele-perrone: It looks like errno.h has to be included explicitly when compiling with older gcc versions. This issue was encountered on the following system: - Ubuntu 20.04 LTS - Linux 5.4.0-156 - gcc 10.5.0 - Meson 0.53.2 Request URL : https://github.com/alsa-project/alsa-gobject/pull/83 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/83.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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Re: [PATCH] Input: pwm-beeper - Support volume setting via sysfs
by Dmitry Torokhov 17 Aug '23

17 Aug '23

On Sat, May 13, 2023 at 11:02:30PM +0200, Marek Vasut wrote: > On 5/13/23 03:51, Marek Vasut wrote: > > On 5/13/23 03:12, Jeff LaBundy wrote: > > > Hi Marek, > > > > Hi, > > > > > On Fri, May 12, 2023 at 08:55:51PM +0200, Marek Vasut wrote: > > > > The PWM beeper volume can be controlled by adjusting the PWM duty cycle, > > > > expose volume setting via sysfs, so users can make the beeper quieter. > > > > This patch adds sysfs attribute 'volume' in range 0..50000, i.e. from 0 > > > > to 50% in 1/1000th of percent steps, this resolution should be > > > > sufficient. > > > > > > > > The reason for 50000 cap on volume or PWM duty cycle is because > > > > duty cycle > > > > above 50% again reduces the loudness, the PWM wave form is inverted wave > > > > form of the one for duty cycle below 50% and the beeper gets quieter the > > > > closer the setting is to 100% . Hence, 50% cap where the wave > > > > form yields > > > > the loudest result. > > > > > > > > Signed-off-by: Marek Vasut <marex(a)denx.de> > > > > --- > > > > An alternative option would be to extend the userspace input > > > > ABI, e.g. by > > > > using SND_TONE top 16bits to encode the duty cycle in 0..50000 > > > > range, and > > > > bottom 16bit to encode the existing frequency in Hz . Since frequency in > > > > Hz is likely to be below some 25 kHz for audible bell, this fits > > > > in 16bits > > > > just fine. Thoughts ? > > > > --- > > > > > > Thanks for the patch; this seems like a useful feature. > > > > > > My first thought is that 50000 seems like an oddly specific limit to > > > impose > > > upon user space. Ideally, user space need not even care that the > > > beeper is > > > implemented via PWM and why 50000 is significant. > > > > > > Instead, what about accepting 0..255 as the LED subsystem does for > > > brightness, > > > then map these values to 0..50000 internally? In fact, the leds-pwm > > > driver > > > does something similar. > > > > The pwm_set_relative_duty_cycle() function can map whatever range to > > whatever other range of the PWM already, so that's not an issues here. > > It seems to me the 0..127 or 0..255 range is a bit too limiting . I > > think even for the LEDs the reason for that limit is legacy design, but > > here I might be wrong. > > > > > I'm also curious as to whether this function should be a rogue sysfs > > > control > > > limited to this driver, or a generic operation in input. For > > > example, input > > > already allows user space to specify the magnitude of an FF effect; > > > perhaps > > > something similar is warranted here? > > > > See the "An alternative ..." part above, I was wondering about this too, > > whether this can be added into the input ABI, but I am somewhat > > reluctant to fiddle with the ABI. > > Thinking about this further, we could try and add some > > EV_SND SND_TONE_WITH_VOLUME > > to avoid overloading EV_SND SND_TONE , and at the same time allow the user > to set both frequency and volume for the tone without any race condition > between the two. > > The EV_SND SND_TONE_WITH_VOLUME would still take one 32bit parameter, except > this time the parameter 16 LSbits would be the frequency and 16 MSbits would > be the volume. > > But again, here I would like input from the maintainers. Beeper was supposed to be an extremely simple device with minimal controls. I wonder if there is need for volume controls, etc, etc are we not better moving it over to the sound subsystem. We already have: sound/drivers/pcsp/pcsp.c and sound/pci/hda/hda_beep.c there, can we have other "advanced" beepers there as well? Adding sound maintainers to CC... Thanks. -- Dmitry

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[PATCH -next v2] ALSA: Make SND_PCMTEST depend on DEBUG_FS
by Ruan Jinjie 17 Aug '23

17 Aug '23

Since pcmtest is a test module that manipulates or gets notification via debugfs, without DEBUG_FS it can not work fine. So make SND_PCMTEST depend on DEBUG_FS. Signed-off-by: Ruan Jinjie <ruanjinjie(a)huawei.com> --- v2: - make it depend on DEBUG_FS instead of dropping error check. --- sound/drivers/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 41c171468c1e..6debd8e95cb7 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -111,6 +111,7 @@ config SND_ALOOP config SND_PCMTEST tristate "Virtual PCM test driver" + depends on DEBUG_FS select SND_PCM help Say 'Y' or 'M' to include support for the Virtual PCM test driver. -- 2.34.1

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