diff options
Diffstat (limited to 'sound')
30 files changed, 355 insertions, 148 deletions
diff --git a/sound/hda/codecs/hdmi/nvhdmi-mcp.c b/sound/hda/codecs/hdmi/nvhdmi-mcp.c index 8fd8d76fa72f..1c5fdfe872f2 100644 --- a/sound/hda/codecs/hdmi/nvhdmi-mcp.c +++ b/sound/hda/codecs/hdmi/nvhdmi-mcp.c @@ -350,8 +350,8 @@ static int nvhdmi_mcp_probe(struct hda_codec *codec, static const struct hda_codec_ops nvhdmi_mcp_codec_ops = { .probe = nvhdmi_mcp_probe, .remove = snd_hda_hdmi_simple_remove, - .build_controls = nvhdmi_mcp_build_pcms, - .build_pcms = nvhdmi_mcp_build_controls, + .build_pcms = nvhdmi_mcp_build_pcms, + .build_controls = nvhdmi_mcp_build_controls, .init = nvhdmi_mcp_init, .unsol_event = snd_hda_hdmi_simple_unsol_event, }; diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 8ad5febd822a..269b6c1e3b6d 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -3736,6 +3736,7 @@ enum { ALC285_FIXUP_ASUS_GA605K_I2C_SPEAKER2_TO_DAC1, ALC269_FIXUP_POSITIVO_P15X_HEADSET_MIC, ALC289_FIXUP_ASUS_ZEPHYRUS_DUAL_SPK, + ALC256_FIXUP_VAIO_RPL_MIC_NO_PRESENCE, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -6172,6 +6173,16 @@ static const struct hda_fixup alc269_fixups[] = { { 0x1e, 0x90170150 }, /* Internal Speaker */ { } }, + }, + [ALC256_FIXUP_VAIO_RPL_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a1113c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x22a190a0 }, /* dock mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST } }; @@ -6578,6 +6589,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8c16, "HP Spectre x360 2-in-1 Laptop 16-aa0xxx", ALC245_FIXUP_HP_SPECTRE_X360_16_AA0XXX), SND_PCI_QUIRK(0x103c, 0x8c17, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c21, "HP Pavilion Plus Laptop 14-ey0XXX", ALC245_FIXUP_HP_X360_MUTE_LEDS), + SND_PCI_QUIRK(0x103c, 0x8c2d, "HP Victus 15-fa1xxx (MB 8C2D)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8c30, "HP Victus 15-fb1xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8c46, "HP EliteBook 830 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c47, "HP EliteBook 840 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), @@ -6682,6 +6694,15 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8e60, "HP Trekker ", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e61, "HP Trekker ", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e62, "HP Trekker ", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8ed5, "HP Merino13X", ALC245_FIXUP_TAS2781_SPI_2), + SND_PCI_QUIRK(0x103c, 0x8ed6, "HP Merino13", ALC245_FIXUP_TAS2781_SPI_2), + SND_PCI_QUIRK(0x103c, 0x8ed7, "HP Merino14", ALC245_FIXUP_TAS2781_SPI_2), + SND_PCI_QUIRK(0x103c, 0x8ed8, "HP Merino16", ALC245_FIXUP_TAS2781_SPI_2), + SND_PCI_QUIRK(0x103c, 0x8ed9, "HP Merino14W", ALC245_FIXUP_TAS2781_SPI_2), + SND_PCI_QUIRK(0x103c, 0x8eda, "HP Merino16W", ALC245_FIXUP_TAS2781_SPI_2), + SND_PCI_QUIRK(0x103c, 0x8f40, "HP Lampas14", ALC287_FIXUP_TXNW2781_I2C), + SND_PCI_QUIRK(0x103c, 0x8f41, "HP Lampas16", ALC287_FIXUP_TXNW2781_I2C), + SND_PCI_QUIRK(0x103c, 0x8f42, "HP LampasW14", ALC287_FIXUP_TXNW2781_I2C), SND_PCI_QUIRK(0x1043, 0x1032, "ASUS VivoBook X513EA", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1034, "ASUS GU605C", ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), @@ -6959,6 +6980,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL5[03]RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa554, "VAIO VJFH52", ALC269_FIXUP_VAIO_VJFH52_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xa559, "VAIO RPL", ALC256_FIXUP_VAIO_RPL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL50NU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa650, "Clevo NP[567]0SN[CD]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -7080,6 +7102,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38a9, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38ab, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38b4, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), + HDA_CODEC_QUIRK(0x17aa, 0x391c, "Lenovo Yoga 7 2-in-1 14AKP10", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x38b5, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b6, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b7, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), diff --git a/sound/soc/amd/acp/amd-acp70-acpi-match.c b/sound/soc/amd/acp/amd-acp70-acpi-match.c index dcecac792e6d..871b4f054a84 100644 --- a/sound/soc/amd/acp/amd-acp70-acpi-match.c +++ b/sound/soc/amd/acp/amd-acp70-acpi-match.c @@ -30,6 +30,20 @@ static const struct snd_soc_acpi_endpoint spk_r_endpoint = { .group_id = 1 }; +static const struct snd_soc_acpi_endpoint spk_2_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 2, + .group_id = 1 +}; + +static const struct snd_soc_acpi_endpoint spk_3_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 3, + .group_id = 1 +}; + static const struct snd_soc_acpi_adr_device rt711_rt1316_group_adr[] = { { .adr = 0x000030025D071101ull, @@ -112,6 +126,134 @@ static const struct snd_soc_acpi_adr_device rt1320_1_single_adr[] = { } }; +static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = { + { /* Jack Playback Endpoint */ + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* DMIC Capture Endpoint */ + .num = 1, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* Jack Capture Endpoint */ + .num = 2, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* Speaker Playback Endpoint */ + .num = 3, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + +static const struct snd_soc_acpi_adr_device cs42l43_0_adr[] = { + { + .adr = 0x00003001FA424301ull, + .num_endpoints = ARRAY_SIZE(cs42l43_endpoints), + .endpoints = cs42l43_endpoints, + .name_prefix = "cs42l43" + } +}; + +static const struct snd_soc_acpi_adr_device cs42l43_1_cs35l56x4_1_adr[] = { + { + .adr = 0x00013001FA424301ull, + .num_endpoints = ARRAY_SIZE(cs42l43_endpoints), + .endpoints = cs42l43_endpoints, + .name_prefix = "cs42l43" + }, + { + .adr = 0x00013001FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00013101FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP2" + }, + { + .adr = 0x00013201FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_2_endpoint, + .name_prefix = "AMP3" + }, + { + .adr = 0x00013301FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_3_endpoint, + .name_prefix = "AMP4" + }, +}; + +static const struct snd_soc_acpi_adr_device cs35l56x4_1_adr[] = { + { + .adr = 0x00013301FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00013201FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP2" + }, + { + .adr = 0x00013101FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_2_endpoint, + .name_prefix = "AMP3" + }, + { + .adr = 0x00013001FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_3_endpoint, + .name_prefix = "AMP4" + }, +}; + +static const struct snd_soc_acpi_link_adr acp70_cs42l43_l1_cs35l56x4_l1[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(cs42l43_1_cs35l56x4_1_adr), + .adr_d = cs42l43_1_cs35l56x4_1_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr acp70_cs42l43_l0_cs35l56x4_l1[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_0_adr), + .adr_d = cs42l43_0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(cs35l56x4_1_adr), + .adr_d = cs35l56x4_1_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr acp70_cs35l56x4_l1[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(cs35l56x4_1_adr), + .adr_d = cs35l56x4_1_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr acp70_rt722_only[] = { { .mask = BIT(0), @@ -151,6 +293,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_sdw_machines[] = { .links = acp70_4_in_1_sdca, .drv_name = "amd_sdw", }, + { + .link_mask = BIT(0) | BIT(1), + .links = acp70_cs42l43_l0_cs35l56x4_l1, + .drv_name = "amd_sdw", + }, + { + .link_mask = BIT(1), + .links = acp70_cs42l43_l1_cs35l56x4_l1, + .drv_name = "amd_sdw", + }, + { + .link_mask = BIT(1), + .links = acp70_cs35l56x4_l1, + .drv_name = "amd_sdw", + }, {}, }; EXPORT_SYMBOL(snd_soc_acpi_amd_acp70_sdw_machines); diff --git a/sound/soc/codecs/cs-amp-lib-test.c b/sound/soc/codecs/cs-amp-lib-test.c index 2fde84309338..3406887cdfa2 100644 --- a/sound/soc/codecs/cs-amp-lib-test.c +++ b/sound/soc/codecs/cs-amp-lib-test.c @@ -7,6 +7,7 @@ #include <kunit/resource.h> #include <kunit/test.h> +#include <kunit/test-bug.h> #include <kunit/static_stub.h> #include <linux/device/faux.h> #include <linux/firmware/cirrus/cs_dsp.h> diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 6a3cca3d26c7..ead447a5da7f 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -581,17 +581,17 @@ static int cs4271_component_probe(struct snd_soc_component *component) ret = regcache_sync(cs4271->regmap); if (ret < 0) - return ret; + goto err_disable_regulator; ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN | CS4271_MODE2_CPEN, CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret < 0) - return ret; + goto err_disable_regulator; ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN, 0); if (ret < 0) - return ret; + goto err_disable_regulator; /* Power-up sequence requires 85 uS */ udelay(85); @@ -601,6 +601,10 @@ static int cs4271_component_probe(struct snd_soc_component *component) CS4271_MODE2_MUTECAEQUB); return 0; + +err_disable_regulator: + regulator_bulk_disable(ARRAY_SIZE(cs4271->supplies), cs4271->supplies); + return ret; } static void cs4271_component_remove(struct snd_soc_component *component) diff --git a/sound/soc/codecs/cs530x.c b/sound/soc/codecs/cs530x.c index b9eff240b929..535387cd7aa3 100644 --- a/sound/soc/codecs/cs530x.c +++ b/sound/soc/codecs/cs530x.c @@ -793,7 +793,7 @@ static int cs530x_set_sysclk(struct snd_soc_component *component, int clk_id, case CS530X_SYSCLK_SRC_PLL: break; default: - dev_err(component->dev, "Invalid clock id %d\n", clk_id); + dev_err(component->dev, "Invalid sysclk source: %d\n", source); return -EINVAL; } diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index ae89260ca215..3420011da444 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -2124,11 +2124,50 @@ static int da7213_probe(struct snd_soc_component *component) return 0; } +static int da7213_runtime_suspend(struct device *dev) +{ + struct da7213_priv *da7213 = dev_get_drvdata(dev); + + regcache_cache_only(da7213->regmap, true); + regcache_mark_dirty(da7213->regmap); + regulator_bulk_disable(DA7213_NUM_SUPPLIES, da7213->supplies); + + return 0; +} + +static int da7213_runtime_resume(struct device *dev) +{ + struct da7213_priv *da7213 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(DA7213_NUM_SUPPLIES, da7213->supplies); + if (ret < 0) + return ret; + regcache_cache_only(da7213->regmap, false); + return regcache_sync(da7213->regmap); +} + +static int da7213_suspend(struct snd_soc_component *component) +{ + struct da7213_priv *da7213 = snd_soc_component_get_drvdata(component); + + return da7213_runtime_suspend(da7213->dev); +} + +static int da7213_resume(struct snd_soc_component *component) +{ + struct da7213_priv *da7213 = snd_soc_component_get_drvdata(component); + + return da7213_runtime_resume(da7213->dev); +} + static const struct snd_soc_component_driver soc_component_dev_da7213 = { .probe = da7213_probe, .set_bias_level = da7213_set_bias_level, .controls = da7213_snd_controls, .num_controls = ARRAY_SIZE(da7213_snd_controls), + .suspend = da7213_suspend, + .resume = da7213_resume, .dapm_widgets = da7213_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(da7213_dapm_widgets), .dapm_routes = da7213_audio_map, @@ -2175,6 +2214,8 @@ static int da7213_i2c_probe(struct i2c_client *i2c) if (!da7213->fin_min_rate) return -EINVAL; + da7213->dev = &i2c->dev; + i2c_set_clientdata(i2c, da7213); /* Get required supplies */ @@ -2224,31 +2265,9 @@ static void da7213_i2c_remove(struct i2c_client *i2c) pm_runtime_disable(&i2c->dev); } -static int da7213_runtime_suspend(struct device *dev) -{ - struct da7213_priv *da7213 = dev_get_drvdata(dev); - - regcache_cache_only(da7213->regmap, true); - regcache_mark_dirty(da7213->regmap); - regulator_bulk_disable(DA7213_NUM_SUPPLIES, da7213->supplies); - - return 0; -} - -static int da7213_runtime_resume(struct device *dev) -{ - struct da7213_priv *da7213 = dev_get_drvdata(dev); - int ret; - - ret = regulator_bulk_enable(DA7213_NUM_SUPPLIES, da7213->supplies); - if (ret < 0) - return ret; - regcache_cache_only(da7213->regmap, false); - return regcache_sync(da7213->regmap); -} - -static DEFINE_RUNTIME_DEV_PM_OPS(da7213_pm, da7213_runtime_suspend, - da7213_runtime_resume, NULL); +static const struct dev_pm_ops da7213_pm = { + RUNTIME_PM_OPS(da7213_runtime_suspend, da7213_runtime_resume, NULL) +}; static const struct i2c_device_id da7213_i2c_id[] = { { "da7213" }, diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index b9ab791d6b88..29cbf0eb6124 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -595,6 +595,7 @@ enum da7213_supplies { /* Codec private data */ struct da7213_priv { struct regmap *regmap; + struct device *dev; struct mutex ctrl_lock; struct regulator_bulk_data supplies[DA7213_NUM_SUPPLIES]; struct clk *mclk; diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 2e1b77973a3e..92c177b82a02 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1638,7 +1638,7 @@ static int va_macro_probe(struct platform_device *pdev) if (ret) goto err_clkout; - va->fsgen = clk_hw_get_clk(&va->hw, "fsgen"); + va->fsgen = devm_clk_hw_get_clk(dev, &va->hw, "fsgen"); if (IS_ERR(va->fsgen)) { ret = PTR_ERR(va->fsgen); goto err_clkout; diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index cb1508fc99f8..5aff5a459a43 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1239,6 +1239,8 @@ static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("DMIC4_ENA", M98090_REG_DIGITAL_MIC_ENABLE, M98090_DIGMIC4_SHIFT, 0, max98090_shdn_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("DMIC34_HPF", M98090_REG_FILTER_CONFIG, + M98090_FLT_DMIC34HPF_SHIFT, 0, NULL, 0), }; static const struct snd_soc_dapm_route max98090_dapm_routes[] = { @@ -1427,8 +1429,8 @@ static const struct snd_soc_dapm_route max98091_dapm_routes[] = { /* DMIC inputs */ {"DMIC3", NULL, "DMIC3_ENA"}, {"DMIC4", NULL, "DMIC4_ENA"}, - {"DMIC3", NULL, "AHPF"}, - {"DMIC4", NULL, "AHPF"}, + {"DMIC3", NULL, "DMIC34_HPF"}, + {"DMIC4", NULL, "DMIC34_HPF"}, }; static int max98090_add_widgets(struct snd_soc_component *component) diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c index a4bd29d7220b..5f7b505d5414 100644 --- a/sound/soc/codecs/rt721-sdca.c +++ b/sound/soc/codecs/rt721-sdca.c @@ -281,6 +281,10 @@ static void rt721_sdca_jack_preset(struct rt721_sdca_priv *rt721) rt_sdca_index_write(rt721->mbq_regmap, RT721_BOOST_CTRL, RT721_BST_4CH_TOP_GATING_CTRL1, 0x002a); regmap_write(rt721->regmap, 0x2f58, 0x07); + + regmap_write(rt721->regmap, 0x2f51, 0x00); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_MISC_CTL, 0x0004); } static void rt721_sdca_jack_init(struct rt721_sdca_priv *rt721) diff --git a/sound/soc/codecs/rt721-sdca.h b/sound/soc/codecs/rt721-sdca.h index 71fac9cd8739..24ce188562ba 100644 --- a/sound/soc/codecs/rt721-sdca.h +++ b/sound/soc/codecs/rt721-sdca.h @@ -137,6 +137,7 @@ struct rt721_sdca_dmic_kctrl_priv { #define RT721_HDA_LEGACY_UAJ_CTL 0x02 #define RT721_HDA_LEGACY_CTL1 0x05 #define RT721_HDA_LEGACY_RESET_CTL 0x06 +#define RT721_MISC_CTL 0x07 #define RT721_XU_REL_CTRL 0x0c #define RT721_GE_REL_CTRL1 0x0d #define RT721_HDA_LEGACY_GPIO_WAKE_EN_CTL 0x0e diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index ba880b5de7e8..8f37aa00e62e 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -1957,7 +1957,8 @@ static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) { struct i2c_client *client = (struct i2c_client *)tas_priv->client; unsigned int dev_addrs[TASDEVICE_MAX_CHANNELS]; - int i, ndev = 0; + int ndev = 0; + int i, rc; if (tas_priv->isacpi) { ndev = device_property_read_u32_array(&client->dev, @@ -1968,8 +1969,12 @@ static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) } else { ndev = (ndev < ARRAY_SIZE(dev_addrs)) ? ndev : ARRAY_SIZE(dev_addrs); - ndev = device_property_read_u32_array(&client->dev, + rc = device_property_read_u32_array(&client->dev, "ti,audio-slots", dev_addrs, ndev); + if (rc != 0) { + ndev = 1; + dev_addrs[0] = client->addr; + } } tas_priv->irq = diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c index 1fb4227b711e..e273b80d033e 100644 --- a/sound/soc/codecs/tas2783-sdw.c +++ b/sound/soc/codecs/tas2783-sdw.c @@ -762,10 +762,17 @@ static void tas2783_fw_ready(const struct firmware *fmw, void *context) goto out; } - mutex_lock(&tas_dev->pde_lock); img_sz = fmw->size; buf = fmw->data; offset += FW_DL_OFFSET; + if (offset >= (img_sz - FW_FL_HDR)) { + dev_err(tas_dev->dev, + "firmware is too small"); + ret = -EINVAL; + goto out; + } + + mutex_lock(&tas_dev->pde_lock); while (offset < (img_sz - FW_FL_HDR)) { memset(&hdr, 0, sizeof(hdr)); offset += read_header(&buf[offset], &hdr); @@ -776,6 +783,14 @@ static void tas2783_fw_ready(const struct firmware *fmw, void *context) /* size also includes the header */ file_blk_size = hdr.length - FW_FL_HDR; + /* make sure that enough data is there */ + if (offset + file_blk_size > img_sz) { + ret = -EINVAL; + dev_err(tas_dev->dev, + "corrupt firmware file"); + break; + } + switch (hdr.file_id) { case 0: ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, @@ -808,7 +823,8 @@ static void tas2783_fw_ready(const struct firmware *fmw, void *context) break; } mutex_unlock(&tas_dev->pde_lock); - tas2783_update_calibdata(tas_dev); + if (!ret) + tas2783_update_calibdata(tas_dev); out: if (!ret) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index aabd90a8b3ec..cac26ba0aa4b 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -131,7 +131,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx943 = { .fifos = 8, .fifo_depth = 32, .dataline = 0xf, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_DSD_U32_BE, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_DSD_U32_LE, .use_edma = true, .use_verid = true, .volume_sx = false, @@ -823,7 +823,7 @@ static int fsl_micfil_hw_params(struct snd_pcm_substream *substream, break; } - if (format == SNDRV_PCM_FORMAT_DSD_U32_BE) { + if (format == SNDRV_PCM_FORMAT_DSD_U32_LE) { micfil->dec_bypass = true; /* * According to equation 29 in RM: diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 757e7868e322..72bfc91e21b9 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -353,7 +353,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, break; case SND_SOC_DAIFMT_PDM: val_cr2 |= FSL_SAI_CR2_BCP; - val_cr4 &= ~FSL_SAI_CR4_MF; sai->is_pdm_mode = true; break; case SND_SOC_DAIFMT_RIGHT_J: @@ -638,7 +637,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(slot_width); val_cr5 |= FSL_SAI_CR5_W0W(slot_width); - if (sai->is_lsb_first || sai->is_pdm_mode) + if (sai->is_lsb_first) val_cr5 |= FSL_SAI_CR5_FBT(0); else val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); @@ -653,12 +652,12 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_CHMOD; /* - * For SAI provider mode, when Tx(Rx) sync with Rx(Tx) clock, Rx(Tx) will - * generate bclk and frame clock for Tx(Rx), we should set RCR4(TCR4), - * RCR5(TCR5) for playback(capture), or there will be sync error. + * When Tx(Rx) sync with Rx(Tx) clock, Rx(Tx) will provide bclk and + * frame clock for Tx(Rx). We should set RCR4(TCR4), RCR5(TCR5) + * for playback(capture), or there will be sync error. */ - if (!sai->is_consumer_mode[tx] && fsl_sai_dir_is_synced(sai, adir)) { + if (fsl_sai_dir_is_synced(sai, adir)) { regmap_update_bits(sai->regmap, FSL_SAI_xCR4(!tx, ofs), FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | FSL_SAI_CR4_CHMOD_MASK, diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index d31058e2de5b..80c001120cdd 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -651,6 +651,7 @@ static void avs_dai_fe_shutdown(struct snd_pcm_substream *substream, struct snd_ data = snd_soc_dai_get_dma_data(dai, substream); + disable_work_sync(&data->period_elapsed_work); snd_hdac_ext_stream_release(data->host_stream, HDAC_EXT_STREAM_TYPE_HOST); avs_dai_shutdown(substream, dai); } @@ -754,6 +755,8 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so data = snd_soc_dai_get_dma_data(dai, substream); host_stream = data->host_stream; + if (runtime->state == SNDRV_PCM_STATE_XRUN) + hdac_stream(host_stream)->prepared = false; if (hdac_stream(host_stream)->prepared) return 0; diff --git a/sound/soc/intel/avs/probes.c b/sound/soc/intel/avs/probes.c index 693ecfe68fd0..74096236984a 100644 --- a/sound/soc/intel/avs/probes.c +++ b/sound/soc/intel/avs/probes.c @@ -14,8 +14,8 @@ #include "debug.h" #include "messages.h" -static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id node_id, - size_t buffer_size) +static int avs_dsp_init_probe(struct avs_dev *adev, struct snd_compr_params *params, int bps, + union avs_connector_node_id node_id, size_t buffer_size) { struct avs_probe_cfg cfg = {{0}}; struct avs_module_entry mentry; @@ -27,12 +27,16 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id return ret; /* - * Probe module uses no cycles, audio data format and input and output - * frame sizes are unused. It is also not owned by any pipeline. + * Probe module uses no cycles, input and output frame sizes are unused. + * It is also not owned by any pipeline. */ cfg.base.ibs = 1; /* BSS module descriptor is always segment of index=2. */ cfg.base.is_pages = mentry.segments[2].flags.length; + cfg.base.audio_fmt.sampling_freq = params->codec.sample_rate; + cfg.base.audio_fmt.bit_depth = bps; + cfg.base.audio_fmt.num_channels = params->codec.ch_out; + cfg.base.audio_fmt.valid_bit_depth = bps; cfg.gtw_cfg.node_id = node_id; cfg.gtw_cfg.dma_buffer_size = buffer_size; @@ -128,8 +132,6 @@ static int avs_probe_compr_set_params(struct snd_compr_stream *cstream, struct hdac_ext_stream *host_stream = avs_compr_get_host_stream(cstream); struct snd_compr_runtime *rtd = cstream->runtime; struct avs_dev *adev = to_avs_dev(dai->dev); - /* compr params do not store bit depth, default to S32_LE. */ - snd_pcm_format_t format = SNDRV_PCM_FORMAT_S32_LE; unsigned int format_val; int bps, ret; @@ -142,7 +144,7 @@ static int avs_probe_compr_set_params(struct snd_compr_stream *cstream, ret = snd_compr_malloc_pages(cstream, rtd->buffer_size); if (ret < 0) return ret; - bps = snd_pcm_format_physical_width(format); + bps = snd_pcm_format_physical_width(params->codec.format); if (bps < 0) return bps; format_val = snd_hdac_stream_format(params->codec.ch_out, bps, params->codec.sample_rate); @@ -166,7 +168,7 @@ static int avs_probe_compr_set_params(struct snd_compr_stream *cstream, node_id.vindex = hdac_stream(host_stream)->stream_tag - 1; node_id.dma_type = AVS_DMA_HDA_HOST_INPUT; - ret = avs_dsp_init_probe(adev, node_id, rtd->dma_bytes); + ret = avs_dsp_init_probe(adev, params, bps, node_id, rtd->dma_bytes); if (ret < 0) { dev_err(dai->dev, "probe init failed: %d\n", ret); avs_dsp_enable_d0ix(adev); diff --git a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c index 3c8b10e21ceb..4853f4f31786 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c @@ -227,33 +227,6 @@ static const struct snd_soc_acpi_endpoint cs42l43_amp_spkagg_endpoints[] = { }, }; -static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = { - { /* Jack Playback Endpoint */ - .num = 0, - .aggregated = 0, - .group_position = 0, - .group_id = 0, - }, - { /* DMIC Capture Endpoint */ - .num = 1, - .aggregated = 0, - .group_position = 0, - .group_id = 0, - }, - { /* Jack Capture Endpoint */ - .num = 2, - .aggregated = 0, - .group_position = 0, - .group_id = 0, - }, - { /* Speaker Playback Endpoint */ - .num = 3, - .aggregated = 0, - .group_position = 0, - .group_id = 0, - }, -}; - static const struct snd_soc_acpi_adr_device cs42l43_2_adr[] = { { .adr = 0x00023001fa424301ull, @@ -305,15 +278,6 @@ static const struct snd_soc_acpi_adr_device cs35l56_3_3amp_adr[] = { } }; -static const struct snd_soc_acpi_adr_device cs42l43_3_adr[] = { - { - .adr = 0x00033001FA424301ull, - .num_endpoints = ARRAY_SIZE(cs42l43_endpoints), - .endpoints = cs42l43_endpoints, - .name_prefix = "cs42l43" - } -}; - static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { { .adr = 0x000030025D071101ull, @@ -486,15 +450,6 @@ static const struct snd_soc_acpi_link_adr ptl_cs42l43_l2_cs35l56x6_l13[] = { {} }; -static const struct snd_soc_acpi_link_adr ptl_cs42l43_l3[] = { - { - .mask = BIT(3), - .num_adr = ARRAY_SIZE(cs42l43_3_adr), - .adr_d = cs42l43_3_adr, - }, - {} -}; - static const struct snd_soc_acpi_link_adr ptl_rt721_l0[] = { { .mask = BIT(0), @@ -714,13 +669,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_sdw_machines[] = { }, { .link_mask = BIT(3), - .links = ptl_cs42l43_l3, - .drv_name = "sof_sdw", - .sof_tplg_filename = "sof-ptl-cs42l43-l3.tplg", - .get_function_tplg_files = sof_sdw_get_tplg_files, - }, - { - .link_mask = BIT(3), .links = ptl_sdw_rt712_vb_l3_rt1320_l3, .drv_name = "sof_sdw", .machine_check = snd_soc_acpi_intel_sdca_is_device_rt712_vb, diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c index 5d025ad72263..c63b3444bc17 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c @@ -3176,7 +3176,6 @@ err_pm_put: static void mt8195_afe_pcm_dev_remove(struct platform_device *pdev) { - pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) mt8195_afe_runtime_suspend(&pdev->dev); } diff --git a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c index 10793bbe9275..d48252cd96ac 100644 --- a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c +++ b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c @@ -2238,7 +2238,6 @@ static void mt8365_afe_pcm_dev_remove(struct platform_device *pdev) mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_AFE); - pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) mt8365_afe_runtime_suspend(&pdev->dev); } diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 06a802f9dba5..67e9ca18883c 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -377,9 +377,9 @@ static void q6asm_audio_client_free_buf(struct audio_client *ac, spin_lock_irqsave(&ac->lock, flags); port->num_periods = 0; + spin_unlock_irqrestore(&ac->lock, flags); kfree(port->buf); port->buf = NULL; - spin_unlock_irqrestore(&ac->lock, flags); } /** diff --git a/sound/soc/renesas/rcar/ssiu.c b/sound/soc/renesas/rcar/ssiu.c index faf351126d57..244fb833292a 100644 --- a/sound/soc/renesas/rcar/ssiu.c +++ b/sound/soc/renesas/rcar/ssiu.c @@ -509,7 +509,7 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, int rsnd_ssiu_probe(struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); - struct device_node *node; + struct device_node *node __free(device_node) = rsnd_ssiu_of_node(priv); struct rsnd_ssiu *ssiu; struct rsnd_mod_ops *ops; const int *list = NULL; @@ -522,7 +522,6 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) * see * rsnd_ssiu_bufsif_to_id() */ - node = rsnd_ssiu_of_node(priv); if (node) nr = rsnd_node_count(priv, node, SSIU_NAME); else diff --git a/sound/soc/renesas/rz-ssi.c b/sound/soc/renesas/rz-ssi.c index e00940814157..81b883e8ac92 100644 --- a/sound/soc/renesas/rz-ssi.c +++ b/sound/soc/renesas/rz-ssi.c @@ -85,6 +85,7 @@ struct rz_ssi_stream { struct snd_pcm_substream *substream; int fifo_sample_size; /* sample capacity of SSI FIFO */ int dma_buffer_pos; /* The address for the next DMA descriptor */ + int completed_dma_buf_pos; /* The address of the last completed DMA descriptor. */ int period_counter; /* for keeping track of periods transferred */ int sample_width; int buffer_pos; /* current frame position in the buffer */ @@ -215,6 +216,7 @@ static void rz_ssi_stream_init(struct rz_ssi_stream *strm, rz_ssi_set_substream(strm, substream); strm->sample_width = samples_to_bytes(runtime, 1); strm->dma_buffer_pos = 0; + strm->completed_dma_buf_pos = 0; strm->period_counter = 0; strm->buffer_pos = 0; @@ -437,6 +439,10 @@ static void rz_ssi_pointer_update(struct rz_ssi_stream *strm, int frames) snd_pcm_period_elapsed(strm->substream); strm->period_counter = current_period; } + + strm->completed_dma_buf_pos += runtime->period_size; + if (strm->completed_dma_buf_pos >= runtime->buffer_size) + strm->completed_dma_buf_pos = 0; } static int rz_ssi_pio_recv(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm) @@ -778,10 +784,14 @@ no_dma: return -ENODEV; } -static int rz_ssi_trigger_resume(struct rz_ssi_priv *ssi) +static int rz_ssi_trigger_resume(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm) { + struct snd_pcm_substream *substream = strm->substream; + struct snd_pcm_runtime *runtime = substream->runtime; int ret; + strm->dma_buffer_pos = strm->completed_dma_buf_pos + runtime->period_size; + if (rz_ssi_is_stream_running(&ssi->playback) || rz_ssi_is_stream_running(&ssi->capture)) return 0; @@ -794,16 +804,6 @@ static int rz_ssi_trigger_resume(struct rz_ssi_priv *ssi) ssi->hw_params_cache.channels); } -static void rz_ssi_streams_suspend(struct rz_ssi_priv *ssi) -{ - if (rz_ssi_is_stream_running(&ssi->playback) || - rz_ssi_is_stream_running(&ssi->capture)) - return; - - ssi->playback.dma_buffer_pos = 0; - ssi->capture.dma_buffer_pos = 0; -} - static int rz_ssi_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -813,7 +813,7 @@ static int rz_ssi_dai_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - ret = rz_ssi_trigger_resume(ssi); + ret = rz_ssi_trigger_resume(ssi, strm); if (ret) return ret; @@ -852,7 +852,6 @@ static int rz_ssi_dai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_SUSPEND: rz_ssi_stop(ssi, strm); - rz_ssi_streams_suspend(ssi); break; case SNDRV_PCM_TRIGGER_STOP: diff --git a/sound/soc/sdca/sdca_functions.c b/sound/soc/sdca/sdca_functions.c index 13f68f7b6dd6..0ccb6775f4de 100644 --- a/sound/soc/sdca/sdca_functions.c +++ b/sound/soc/sdca/sdca_functions.c @@ -894,7 +894,8 @@ static int find_sdca_entity_control(struct device *dev, struct sdca_entity *enti return ret; } - control->values = devm_kzalloc(dev, hweight64(control->cn_list), GFP_KERNEL); + control->values = devm_kcalloc(dev, hweight64(control->cn_list), + sizeof(int), GFP_KERNEL); if (!control->values) return -ENOMEM; diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index 270c66b90228..3848c7df1916 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -638,7 +638,6 @@ struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, false}, .dai_name = "cs42l43-dp6", - .component_name = "cs42l43", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, .init = asoc_sdw_cs42l43_spk_init, @@ -1278,7 +1277,7 @@ static int is_sdca_endpoint_present(struct device *dev, struct sdw_slave *slave; struct device *sdw_dev; const char *sdw_codec_name; - int i; + int ret, i; dlc = kzalloc(sizeof(*dlc), GFP_KERNEL); if (!dlc) @@ -1308,13 +1307,16 @@ static int is_sdca_endpoint_present(struct device *dev, } slave = dev_to_sdw_dev(sdw_dev); - if (!slave) - return -EINVAL; + if (!slave) { + ret = -EINVAL; + goto put_device; + } /* Make sure BIOS provides SDCA properties */ if (!slave->sdca_data.interface_revision) { dev_warn(&slave->dev, "SDCA properties not found in the BIOS\n"); - return 1; + ret = 1; + goto put_device; } for (i = 0; i < slave->sdca_data.num_functions; i++) { @@ -1323,7 +1325,8 @@ static int is_sdca_endpoint_present(struct device *dev, if (dai_type == dai_info->dai_type) { dev_dbg(&slave->dev, "DAI type %d sdca function %s found\n", dai_type, slave->sdca_data.function[i].name); - return 1; + ret = 1; + goto put_device; } } @@ -1331,7 +1334,11 @@ static int is_sdca_endpoint_present(struct device *dev, "SDCA device function for DAI type %d not supported, skip endpoint\n", dai_info->dai_type); - return 0; + ret = 0; + +put_device: + put_device(sdw_dev); + return ret; } int asoc_sdw_parse_sdw_endpoints(struct snd_soc_card *card, diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 880f5afcce60..cc15624ecaff 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1362,6 +1362,11 @@ int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, ep->sample_rem = ep->cur_rate % ep->pps; ep->packsize[0] = ep->cur_rate / ep->pps; ep->packsize[1] = (ep->cur_rate + (ep->pps - 1)) / ep->pps; + if (ep->packsize[1] > ep->maxpacksize) { + usb_audio_dbg(chip, "Too small maxpacksize %u for rate %u / pps %u\n", + ep->maxpacksize, ep->cur_rate, ep->pps); + return -EINVAL; + } /* calculate the frequency in 16.16 format */ ep->freqm = ep->freqn; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6f00e0d52382..72b900505d2c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3086,6 +3086,8 @@ static int snd_usb_mixer_controls_badd(struct usb_mixer_interface *mixer, int i; assoc = usb_ifnum_to_if(dev, ctrlif)->intf_assoc; + if (!assoc) + return -EINVAL; /* Detect BADD capture/playback channels from AS EP descriptors */ for (i = 0; i < assoc->bInterfaceCount; i++) { diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c index 15960d25e748..6e09e074c0e7 100644 --- a/sound/usb/mixer_s1810c.c +++ b/sound/usb/mixer_s1810c.c @@ -178,7 +178,7 @@ snd_sc1810c_get_status_field(struct usb_device *dev, pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1; pkt_out.fields[SC1810C_STATE_F2_IDX] = SC1810C_SET_STATE_F2; - ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + ret = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SC1810C_SET_STATE_REQ, SC1810C_SET_STATE_REQTYPE, (*seqnum), 0, &pkt_out, sizeof(pkt_out)); @@ -597,15 +597,6 @@ int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer) if (!list_empty(&chip->mixer_list)) return 0; - dev_info(&dev->dev, - "Presonus Studio 1810c, device_setup: %u\n", chip->setup); - if (chip->setup == 1) - dev_info(&dev->dev, "(8out/18in @ 48kHz)\n"); - else if (chip->setup == 2) - dev_info(&dev->dev, "(6out/8in @ 192kHz)\n"); - else - dev_info(&dev->dev, "(8out/14in @ 96kHz)\n"); - ret = snd_s1810c_init_mixer_maps(chip); if (ret < 0) return ret; @@ -634,16 +625,28 @@ int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer) if (ret < 0) return ret; - // The 1824c has a Mono Main switch instead of a - // A/B select switch. - if (mixer->chip->usb_id == USB_ID(0x194f, 0x010d)) { - ret = snd_s1810c_switch_init(mixer, &snd_s1824c_mono_sw); + switch (chip->usb_id) { + case USB_ID(0x194f, 0x010c): /* Presonus Studio 1810c */ + dev_info(&dev->dev, + "Presonus Studio 1810c, device_setup: %u\n", chip->setup); + if (chip->setup == 1) + dev_info(&dev->dev, "(8out/18in @ 48kHz)\n"); + else if (chip->setup == 2) + dev_info(&dev->dev, "(6out/8in @ 192kHz)\n"); + else + dev_info(&dev->dev, "(8out/14in @ 96kHz)\n"); + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw); if (ret < 0) return ret; - } else if (mixer->chip->usb_id == USB_ID(0x194f, 0x010c)) { - ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw); + + break; + case USB_ID(0x194f, 0x010d): /* Presonus Studio 1824c */ + ret = snd_s1810c_switch_init(mixer, &snd_s1824c_mono_sw); if (ret < 0) return ret; + + break; } return ret; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 71638e6dfb20..5e30bff69f82 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2022,6 +2022,8 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x16d0, 0x09d8): /* NuPrime IDA-8 */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ + case USB_ID(0x16d0, 0x0ab1): /* PureAudio APA DAC */ + case USB_ID(0x16d0, 0xeca1): /* PureAudio Lotus DAC5, DAC5 SE, DAC5 Pro */ case USB_ID(0x1db5, 0x0003): /* Bryston BDA3 */ case USB_ID(0x20a0, 0x4143): /* WaveIO USB Audio 2.0 */ case USB_ID(0x22e1, 0xca01): /* HDTA Serenade DSD */ @@ -2267,6 +2269,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_FIXED_RATE), DEVICE_FLG(0x0fd9, 0x0008, /* Hauppauge HVR-950Q */ QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), + DEVICE_FLG(0x1038, 0x1294, /* SteelSeries Arctis Pro Wireless */ + QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE), DEVICE_FLG(0x1101, 0x0003, /* Audioengine D1 */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x12d1, 0x3a07, /* Huawei Technologies Co., Ltd. */ @@ -2297,6 +2301,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CLOCK_SOURCE), DEVICE_FLG(0x1686, 0x00dd, /* Zoom R16/24 */ QUIRK_FLAG_TX_LENGTH | QUIRK_FLAG_CTL_MSG_DELAY_1M), + DEVICE_FLG(0x16d0, 0x0ab1, /* PureAudio APA DAC */ + QUIRK_FLAG_DSD_RAW), + DEVICE_FLG(0x16d0, 0xeca1, /* PureAudio Lotus DAC5, DAC5 SE and DAC5 Pro */ + QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x17aa, 0x1046, /* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */ QUIRK_FLAG_DISABLE_AUTOSUSPEND), DEVICE_FLG(0x17aa, 0x104d, /* Lenovo ThinkStation P620 Internal Speaker + Front Headset */ |