diff options
Diffstat (limited to 'sound')
71 files changed, 828 insertions, 335 deletions
diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index b8ff5cccd0c8..5431d2c49421 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -158,7 +158,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct device_node *child, *sound = NULL; struct resource *r; int i, layout = 0, rlen, ok = force; - char node_name[6]; + char node_name[8]; static const char *rnames[] = { "i2sbus: %pOFn (control)", "i2sbus: %pOFn (tx)", "i2sbus: %pOFn (rx)" }; diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 696a958d93e9..088cff799e0b 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -343,3 +343,29 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, return NULL; } EXPORT_SYMBOL(intel_nhlt_get_endpoint_blob); + +int intel_nhlt_ssp_device_type(struct device *dev, struct nhlt_acpi_table *nhlt, + u8 virtual_bus_id) +{ + struct nhlt_endpoint *epnt; + int i; + + if (!nhlt) + return -EINVAL; + + epnt = (struct nhlt_endpoint *)nhlt->desc; + for (i = 0; i < nhlt->endpoint_count; i++) { + /* for SSP link the virtual bus id is the SSP port number */ + if (epnt->linktype == NHLT_LINK_SSP && + epnt->virtual_bus_id == virtual_bus_id) { + dev_dbg(dev, "SSP%d: dev_type=%d\n", virtual_bus_id, + epnt->device_type); + return epnt->device_type; + } + + epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); + } + + return -EINVAL; +} +EXPORT_SYMBOL(intel_nhlt_ssp_device_type); diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c index 0ba8f0c4cd99..3a593da09280 100644 --- a/sound/oss/dmasound/dmasound_paula.c +++ b/sound/oss/dmasound/dmasound_paula.c @@ -725,7 +725,13 @@ static void __exit amiga_audio_remove(struct platform_device *pdev) dmasound_deinit(); } -static struct platform_driver amiga_audio_driver = { +/* + * amiga_audio_remove() lives in .exit.text. For drivers registered via + * module_platform_driver_probe() this is ok because they cannot get unbound at + * runtime. So mark the driver struct with __refdata to prevent modpost + * triggering a section mismatch warning. + */ +static struct platform_driver amiga_audio_driver __refdata = { .remove_new = __exit_p(amiga_audio_remove), .driver = { .name = "amiga-audio", diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index d36234b88fb4..941bfbf812ed 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -255,7 +255,7 @@ lookup_voices(struct snd_emux *emu, struct snd_emu10k1 *hw, /* check if sample is finished playing (non-looping only) */ if (bp != best + V_OFF && bp != best + V_FREE && (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_SINGLESHOT)) { - val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch) - 64; + val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch); if (val >= vp->reg.loopstart) bp = best + V_OFF; } @@ -362,7 +362,7 @@ start_voice(struct snd_emux_voice *vp) map = (hw->silent_page.addr << hw->address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); - addr = vp->reg.start + 64; + addr = vp->reg.start; temp = vp->reg.parm.filterQ; ccca = (temp << 28) | addr; if (vp->apitch < 0xe400) @@ -430,9 +430,6 @@ start_voice(struct snd_emux_voice *vp) /* Q & current address (Q 4bit value, MSB) */ CCCA, ccca, - /* cache */ - CCR, REG_VAL_PUT(CCR_CACHEINVALIDSIZE, 64), - /* reset volume */ VTFT, vtarget | vp->ftarget, CVCF, vtarget | CVCF_CURRENTFILTER_MASK, diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 72ec872afb8d..8fb688e41414 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -108,7 +108,10 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431F12", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, + { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "17AA38A9", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA38AB", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA38B4", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, @@ -496,7 +499,10 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431F12", generic_dsd_config }, { "CSC3551", "10431F1F", generic_dsd_config }, { "CSC3551", "10431F62", generic_dsd_config }, + { "CSC3551", "10433A60", generic_dsd_config }, { "CSC3551", "17AA386F", generic_dsd_config }, + { "CSC3551", "17AA3877", generic_dsd_config }, + { "CSC3551", "17AA3878", generic_dsd_config }, { "CSC3551", "17AA38A9", generic_dsd_config }, { "CSC3551", "17AA38AB", generic_dsd_config }, { "CSC3551", "17AA38B4", generic_dsd_config }, diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 41974b3897a7..1a3f84599cb5 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -1024,8 +1024,8 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) goto err; } - dev_dbg(cs35l56->base.dev, "DSP system name: '%s', amp name: '%s'\n", - cs35l56->system_name, cs35l56->amp_name); + dev_info(cs35l56->base.dev, "DSP system name: '%s', amp name: '%s'\n", + cs35l56->system_name, cs35l56->amp_name); regmap_multi_reg_write(cs35l56->base.regmap, cs35l56_hda_dai_config, ARRAY_SIZE(cs35l56_hda_dai_config)); @@ -1045,14 +1045,14 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) pm_runtime_mark_last_busy(cs35l56->base.dev); pm_runtime_enable(cs35l56->base.dev); + cs35l56->base.init_done = true; + ret = component_add(cs35l56->base.dev, &cs35l56_hda_comp_ops); if (ret) { dev_err(cs35l56->base.dev, "Register component failed: %d\n", ret); goto pm_err; } - cs35l56->base.init_done = true; - return 0; pm_err: diff --git a/sound/pci/hda/cs35l56_hda_i2c.c b/sound/pci/hda/cs35l56_hda_i2c.c index 13beee807308..40f2f97944d5 100644 --- a/sound/pci/hda/cs35l56_hda_i2c.c +++ b/sound/pci/hda/cs35l56_hda_i2c.c @@ -56,10 +56,19 @@ static const struct i2c_device_id cs35l56_hda_i2c_id[] = { {} }; +static const struct acpi_device_id cs35l56_acpi_hda_match[] = { + { "CSC3554", 0 }, + { "CSC3556", 0 }, + { "CSC3557", 0 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, cs35l56_acpi_hda_match); + static struct i2c_driver cs35l56_hda_i2c_driver = { .driver = { - .name = "cs35l56-hda", - .pm = &cs35l56_hda_pm_ops, + .name = "cs35l56-hda", + .acpi_match_table = cs35l56_acpi_hda_match, + .pm = &cs35l56_hda_pm_ops, }, .id_table = cs35l56_hda_i2c_id, .probe = cs35l56_hda_i2c_probe, diff --git a/sound/pci/hda/cs35l56_hda_spi.c b/sound/pci/hda/cs35l56_hda_spi.c index a3b2fa76663d..7f02155fe61e 100644 --- a/sound/pci/hda/cs35l56_hda_spi.c +++ b/sound/pci/hda/cs35l56_hda_spi.c @@ -56,10 +56,19 @@ static const struct spi_device_id cs35l56_hda_spi_id[] = { {} }; +static const struct acpi_device_id cs35l56_acpi_hda_match[] = { + { "CSC3554", 0 }, + { "CSC3556", 0 }, + { "CSC3557", 0 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, cs35l56_acpi_hda_match); + static struct spi_driver cs35l56_hda_spi_driver = { .driver = { - .name = "cs35l56-hda", - .pm = &cs35l56_hda_pm_ops, + .name = "cs35l56-hda", + .acpi_match_table = cs35l56_acpi_hda_match, + .pm = &cs35l56_hda_pm_ops, }, .id_table = cs35l56_hda_spi_id, .probe = cs35l56_hda_spi_probe, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a17c36a36aa5..cdcb28aa9d7b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6875,11 +6875,38 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st comp_generic_fixup(cdc, action, "i2c", "CLSA0101", "-%s:00-cs35l41-hda.%d", 2); } +static void cs35l56_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "i2c", "CSC3556", "-%s:00-cs35l56-hda.%d", 2); +} + +static void cs35l56_fixup_i2c_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "i2c", "CSC3556", "-%s:00-cs35l56-hda.%d", 4); +} + +static void cs35l56_fixup_spi_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 2); +} + static void cs35l56_fixup_spi_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 4); } +static void alc285_fixup_asus_ga403u(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + /* + * The same SSID has been re-used in different hardware, they have + * different codecs and the newer GA403U has a ALC285. + */ + if (cdc->core.vendor_id == 0x10ec0285) + cs35l56_fixup_i2c_two(cdc, fix, action); + else + alc_fixup_inv_dmic(cdc, fix, action); +} + static void tas2781_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { @@ -7436,6 +7463,10 @@ enum { ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, ALC256_FIXUP_HEADPHONE_AMP_VOL, ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX, + ALC285_FIXUP_CS35L56_SPI_2, + ALC285_FIXUP_CS35L56_I2C_2, + ALC285_FIXUP_CS35L56_I2C_4, + ALC285_FIXUP_ASUS_GA403U, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9643,6 +9674,22 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc245_fixup_hp_spectre_x360_eu0xxx, }, + [ALC285_FIXUP_CS35L56_SPI_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_spi_two, + }, + [ALC285_FIXUP_CS35L56_I2C_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_i2c_two, + }, + [ALC285_FIXUP_CS35L56_I2C_4] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_i2c_four, + }, + [ALC285_FIXUP_ASUS_GA403U] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_asus_ga403u, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10096,7 +10143,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1a83, "ASUS UM5302LA", ALC294_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), - SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1043, 0x1b13, "ASUS U41SV/GA403U", ALC285_FIXUP_ASUS_GA403U), SND_PCI_QUIRK(0x1043, 0x1b93, "ASUS G614JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC287_FIXUP_CS35L41_I2C_2), @@ -10104,6 +10151,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c33, "ASUS UX5304MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c43, "ASUS UX8406MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1c63, "ASUS GU605M", ALC285_FIXUP_CS35L56_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), SND_PCI_QUIRK(0x1043, 0x1c9f, "ASUS G614JU/JV/JI", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JY/JZ/JI/JG", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), @@ -10115,11 +10163,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1da2, "ASUS UP6502ZA/ZD", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x1df3, "ASUS UM5606", ALC285_FIXUP_CS35L56_I2C_4), SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402ZA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), + SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALC285_FIXUP_CS35L56_I2C_2), + SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_CS35L56_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), @@ -10133,7 +10184,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x3a30, "ASUS G814JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x3a50, "ASUS G834JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), @@ -10159,7 +10210,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x12cc, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x12f6, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), - SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), + SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_AMP), @@ -10333,6 +10384,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x386f, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C), + SND_PCI_QUIRK(0x17aa, 0x3877, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x3878, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), 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 power mode2 YC", ALC287_FIXUP_TAS2781_I2C), @@ -10403,6 +10456,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d05, 0x1147, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d05, 0x115c, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1387, "TongFang GMxIXxx", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 4475cea8e9f7..48dae3339305 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -89,7 +89,7 @@ struct tas2781_hda { struct snd_kcontrol *dsp_prog_ctl; struct snd_kcontrol *dsp_conf_ctl; struct snd_kcontrol *prof_ctl; - struct snd_kcontrol *snd_ctls[3]; + struct snd_kcontrol *snd_ctls[2]; }; static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) @@ -161,8 +161,6 @@ static void tas2781_hda_playback_hook(struct device *dev, int action) pm_runtime_put_autosuspend(dev); break; default: - dev_dbg(tas_hda->dev, "Playback action not supported: %d\n", - action); break; } } @@ -185,8 +183,15 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, { struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + mutex_lock(&tas_priv->codec_lock); + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d\n", + __func__, kcontrol->id.name, tas_priv->rcabin.profile_cfg_id); + + mutex_unlock(&tas_priv->codec_lock); + return 0; } @@ -200,11 +205,19 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, val = clamp(nr_profile, 0, max); + mutex_lock(&tas_priv->codec_lock); + + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d -> %d\n", + __func__, kcontrol->id.name, + tas_priv->rcabin.profile_cfg_id, val); + if (tas_priv->rcabin.profile_cfg_id != val) { tas_priv->rcabin.profile_cfg_id = val; ret = 1; } + mutex_unlock(&tas_priv->codec_lock); + return ret; } @@ -241,8 +254,15 @@ static int tasdevice_program_get(struct snd_kcontrol *kcontrol, { struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + mutex_lock(&tas_priv->codec_lock); + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d\n", + __func__, kcontrol->id.name, tas_priv->cur_prog); + + mutex_unlock(&tas_priv->codec_lock); + return 0; } @@ -257,11 +277,18 @@ static int tasdevice_program_put(struct snd_kcontrol *kcontrol, val = clamp(nr_program, 0, max); + mutex_lock(&tas_priv->codec_lock); + + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d -> %d\n", + __func__, kcontrol->id.name, tas_priv->cur_prog, val); + if (tas_priv->cur_prog != val) { tas_priv->cur_prog = val; ret = 1; } + mutex_unlock(&tas_priv->codec_lock); + return ret; } @@ -270,8 +297,15 @@ static int tasdevice_config_get(struct snd_kcontrol *kcontrol, { struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + mutex_lock(&tas_priv->codec_lock); + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d\n", + __func__, kcontrol->id.name, tas_priv->cur_conf); + + mutex_unlock(&tas_priv->codec_lock); + return 0; } @@ -286,54 +320,39 @@ static int tasdevice_config_put(struct snd_kcontrol *kcontrol, val = clamp(nr_config, 0, max); + mutex_lock(&tas_priv->codec_lock); + + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d -> %d\n", + __func__, kcontrol->id.name, tas_priv->cur_conf, val); + if (tas_priv->cur_conf != val) { tas_priv->cur_conf = val; ret = 1; } + mutex_unlock(&tas_priv->codec_lock); + 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, +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; + int ret; - return tasdevice_digital_getvol(tas_priv, ucontrol, mc); -} + mutex_lock(&tas_priv->codec_lock); -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; + ret = tasdevice_amp_getvol(tas_priv, ucontrol, mc); - return tasdevice_amp_getvol(tas_priv, ucontrol, mc); -} + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %ld\n", + __func__, kcontrol->id.name, ucontrol->value.integer.value[0]); -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; + mutex_unlock(&tas_priv->codec_lock); - /* The check of the given value is in tasdevice_digital_putvol. */ - return tasdevice_digital_putvol(tas_priv, ucontrol, mc); + return ret; } static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, @@ -342,9 +361,19 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int ret; + + mutex_lock(&tas_priv->codec_lock); + + dev_dbg(tas_priv->dev, "%s: kcontrol %s: -> %ld\n", + __func__, kcontrol->id.name, ucontrol->value.integer.value[0]); /* The check of the given value is in tasdevice_amp_putvol. */ - return tasdevice_amp_putvol(tas_priv, ucontrol, mc); + ret = tasdevice_amp_putvol(tas_priv, ucontrol, mc); + + mutex_unlock(&tas_priv->codec_lock); + + return ret; } static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, @@ -352,9 +381,13 @@ static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, { struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + mutex_lock(&tas_priv->codec_lock); + 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"); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d\n", + __func__, kcontrol->id.name, tas_priv->force_fwload_status); + + mutex_unlock(&tas_priv->codec_lock); return 0; } @@ -365,14 +398,20 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); bool change, val = (bool)ucontrol->value.integer.value[0]; + mutex_lock(&tas_priv->codec_lock); + + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d -> %d\n", + __func__, kcontrol->id.name, + tas_priv->force_fwload_status, val); + 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"); + + mutex_unlock(&tas_priv->codec_lock); return change; } @@ -381,9 +420,6 @@ 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), }; diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 320ac792c7fe..3182c634464d 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -278,7 +278,8 @@ static void run_spu_dma(struct work_struct *work) dreamcastcard->clicks++; if (unlikely(dreamcastcard->clicks >= AICA_PERIOD_NUMBER)) dreamcastcard->clicks %= AICA_PERIOD_NUMBER; - mod_timer(&dreamcastcard->timer, jiffies + 1); + if (snd_pcm_running(dreamcastcard->substream)) + mod_timer(&dreamcastcard->timer, jiffies + 1); } } @@ -290,6 +291,8 @@ static void aica_period_elapsed(struct timer_list *t) /*timer function - so cannot sleep */ int play_period; struct snd_pcm_runtime *runtime; + if (!snd_pcm_running(substream)) + return; runtime = substream->runtime; dreamcastcard = substream->pcm->private_data; /* Have we played out an additional period? */ @@ -350,12 +353,19 @@ static int snd_aicapcm_pcm_open(struct snd_pcm_substream return 0; } +static int snd_aicapcm_pcm_sync_stop(struct snd_pcm_substream *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + + del_timer_sync(&dreamcastcard->timer); + cancel_work_sync(&dreamcastcard->spu_dma_work); + return 0; +} + static int snd_aicapcm_pcm_close(struct snd_pcm_substream *substream) { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; - flush_work(&(dreamcastcard->spu_dma_work)); - del_timer(&dreamcastcard->timer); dreamcastcard->substream = NULL; kfree(dreamcastcard->channel); spu_disable(); @@ -401,6 +411,7 @@ static const struct snd_pcm_ops snd_aicapcm_playback_ops = { .prepare = snd_aicapcm_pcm_prepare, .trigger = snd_aicapcm_pcm_trigger, .pointer = snd_aicapcm_pcm_pointer, + .sync_stop = snd_aicapcm_pcm_sync_stop, }; /* TO DO: set up to handle more than one pcm instance */ diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 8c8b1dcac628..5f35b90eab8d 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -115,7 +115,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id goto unregister_dmic_dev; } - acp_init(chip); + ret = acp_init(chip); + if (ret) + goto unregister_dmic_dev; + res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL); if (!res) { ret = -ENOMEM; @@ -133,11 +136,9 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id } } - if (flag == FLAG_AMD_LEGACY_ONLY_DMIC) { - ret = check_acp_pdm(pci, chip); - if (ret < 0) - goto skip_pdev_creation; - } + ret = check_acp_pdm(pci, chip); + if (ret < 0) + goto skip_pdev_creation; chip->flag = flag; memset(&pdevinfo, 0, sizeof(pdevinfo)); diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c index 01ef4db5407d..287ac01a3873 100644 --- a/sound/soc/codecs/cs-amp-lib.c +++ b/sound/soc/codecs/cs-amp-lib.c @@ -56,6 +56,11 @@ static int _cs_amp_write_cal_coeffs(struct cs_dsp *dsp, dev_dbg(dsp->dev, "Calibration: Ambient=%#x, Status=%#x, CalR=%d\n", data->calAmbient, data->calStatus, data->calR); + if (list_empty(&dsp->ctl_list)) { + dev_info(dsp->dev, "Calibration disabled due to missing firmware controls\n"); + return -ENOENT; + } + ret = cs_amp_write_cal_coeff(dsp, controls, controls->ambient, data->calAmbient); if (ret) return ret; diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 860d5cda67bf..94685449f0f4 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2364,7 +2364,8 @@ static int cs42l43_codec_runtime_resume(struct device *dev) static int cs42l43_codec_suspend(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; disable_irq(cs42l43->irq); @@ -2373,7 +2374,8 @@ static int cs42l43_codec_suspend(struct device *dev) static int cs42l43_codec_suspend_noirq(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; enable_irq(cs42l43->irq); @@ -2382,7 +2384,8 @@ static int cs42l43_codec_suspend_noirq(struct device *dev) static int cs42l43_codec_resume(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; enable_irq(cs42l43->irq); @@ -2391,7 +2394,8 @@ static int cs42l43_codec_resume(struct device *dev) static int cs42l43_codec_resume_noirq(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; disable_irq(cs42l43->irq); diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 15289dadafea..17bd6b516077 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -412,9 +412,9 @@ static const struct _coeff_div coeff_div_v3[] = { {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x8A, 0x0A, 0x27, 0x27}, {128, 8000, 1024000, 0x60, 0x00, 0x05, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, - {128, 16000, 2048000, 0x20, 0x00, 0x31, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, - {128, 44100, 5644800, 0xE0, 0x00, 0x01, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, - {128, 48000, 6144000, 0xE0, 0x00, 0x01, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {128, 16000, 2048000, 0x20, 0x00, 0x31, 0x35, 0x08, 0x19, 0x1F, 0x3F}, + {128, 44100, 5644800, 0xE0, 0x00, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, + {128, 48000, 6144000, 0xE0, 0x00, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x8A, 0x1B, 0x23, 0x47}, {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x8A, 0x1B, 0x23, 0x47}, {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, @@ -423,10 +423,10 @@ static const struct _coeff_div coeff_div_v3[] = { {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0xCA, 0x0A, 0x27, 0x27}, - {256, 8000, 2048000, 0x60, 0x00, 0x31, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, - {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, - {256, 44100, 11289600, 0xE0, 0x00, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, - {256, 48000, 12288000, 0xE0, 0x00, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {256, 8000, 2048000, 0x60, 0x00, 0x31, 0x35, 0x08, 0x19, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x08, 0x19, 0x1F, 0x3F}, + {256, 44100, 11289600, 0xE0, 0x01, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, + {256, 48000, 12288000, 0xE0, 0x01, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x23, 0x47}, {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, @@ -435,10 +435,10 @@ static const struct _coeff_div coeff_div_v3[] = { {400, 48000, 19200000, 0xE4, 0x04, 0x35, 0x6d, 0xCA, 0x0A, 0x1F, 0x1F}, {500, 48000, 24000000, 0xF8, 0x04, 0x3F, 0x6D, 0xCA, 0x0A, 0x1F, 0x1F}, - {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, - {512, 16000, 8192000, 0x20, 0x00, 0x30, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, - {512, 44100, 22579200, 0xE0, 0x00, 0x00, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, - {512, 48000, 24576000, 0xE0, 0x00, 0x00, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x08, 0x19, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x30, 0x35, 0x08, 0x19, 0x1F, 0x3F}, + {512, 44100, 22579200, 0xE0, 0x00, 0x00, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, + {512, 48000, 24576000, 0xE0, 0x00, 0x00, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, {768, 32000, 24576000, 0xE0, 0x02, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, @@ -835,7 +835,6 @@ static void es8326_jack_detect_handler(struct work_struct *work) dev_dbg(comp->dev, "Report hp remove event\n"); snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); /* mute adc when mic path switch */ - regmap_write(es8326->regmap, ES8326_ADC_SCALE, 0x33); regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326->hp = 0; @@ -843,6 +842,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x0a); regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x03); + regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); /* * Inverted HPJACK_POL bit to trigger one IRQ to double check HP Removal event */ @@ -865,6 +865,8 @@ static void es8326_jack_detect_handler(struct work_struct *work) * set auto-check mode, then restart jack_detect_work after 400ms. * Don't report jack status. */ + regmap_write(es8326->regmap, ES8326_INT_SOURCE, + (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); es8326_enable_micbias(es8326->component); usleep_range(50000, 70000); @@ -891,7 +893,6 @@ static void es8326_jack_detect_handler(struct work_struct *work) snd_soc_jack_report(es8326->jack, SND_JACK_HEADSET, SND_JACK_HEADSET); - regmap_write(es8326->regmap, ES8326_ADC_SCALE, 0x33); regmap_update_bits(es8326->regmap, ES8326_PGA_PDN, 0x08, 0x08); regmap_update_bits(es8326->regmap, ES8326_PGAGAIN, @@ -987,7 +988,7 @@ static int es8326_resume(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x0E); regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0); usleep_range(10000, 15000); - regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xe9); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xd9); regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xcb); /* set headphone default type and detect pin */ regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x83); @@ -1038,8 +1039,7 @@ static int es8326_resume(struct snd_soc_component *component) es8326_enable_micbias(es8326->component); usleep_range(50000, 70000); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00); - regmap_write(es8326->regmap, ES8326_INT_SOURCE, - (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); + regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); regmap_write(es8326->regmap, ES8326_INTOUT_IO, es8326->interrupt_clk); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, @@ -1060,6 +1060,8 @@ static int es8326_resume(struct snd_soc_component *component) es8326->hp = 0; es8326->hpl_vol = 0x03; es8326->hpr_vol = 0x03; + + es8326_irq(es8326->irq, es8326); return 0; } @@ -1070,6 +1072,9 @@ static int es8326_suspend(struct snd_soc_component *component) cancel_delayed_work_sync(&es8326->jack_detect_work); es8326_disable_micbias(component); es8326->calibrated = false; + regmap_write(es8326->regmap, ES8326_CLK_MUX, 0x2d); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0x00); + regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF); regcache_cache_only(es8326->regmap, true); regcache_mark_dirty(es8326->regmap); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h index ee12caef8105..c3e52e7bdef5 100644 --- a/sound/soc/codecs/es8326.h +++ b/sound/soc/codecs/es8326.h @@ -104,7 +104,7 @@ #define ES8326_MUTE (3 << 0) /* ES8326_CLK_CTL */ -#define ES8326_CLK_ON (0x7e << 0) +#define ES8326_CLK_ON (0x7f << 0) #define ES8326_CLK_OFF (0 << 0) /* ES8326_CLK_INV */ diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index 47511f70119a..0b3bf920bcab 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -537,7 +537,7 @@ static int rt1316_sdw_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt1316->sdw_slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -577,12 +577,12 @@ static int rt1316_sdw_parse_dt(struct rt1316_sdw_priv *rt1316, struct device *de if (rt1316->bq_params_cnt) { rt1316->bq_params = devm_kzalloc(dev, rt1316->bq_params_cnt, GFP_KERNEL); if (!rt1316->bq_params) { - dev_err(dev, "Could not allocate bq_params memory\n"); + dev_err(dev, "%s: Could not allocate bq_params memory\n", __func__); ret = -ENOMEM; } else { ret = device_property_read_u8_array(dev, "realtek,bq-params", rt1316->bq_params, rt1316->bq_params_cnt); if (ret < 0) - dev_err(dev, "Could not read list of realtek,bq-params\n"); + dev_err(dev, "%s: Could not read list of realtek,bq-params\n", __func__); } } @@ -759,7 +759,7 @@ static int __maybe_unused rt1316_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT1316_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt1318-sdw.c b/sound/soc/codecs/rt1318-sdw.c index ff364bde4a08..462c9a4b1be5 100644 --- a/sound/soc/codecs/rt1318-sdw.c +++ b/sound/soc/codecs/rt1318-sdw.c @@ -606,7 +606,7 @@ static int rt1318_sdw_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt1318->sdw_slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -631,8 +631,8 @@ static int rt1318_sdw_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT1318_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } @@ -835,7 +835,7 @@ static int __maybe_unused rt1318_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT1318_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); return -ETIMEDOUT; } diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index e67c2e19cb1a..f9ee42c13dba 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -132,7 +132,7 @@ static int rt5682_sdw_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt5682->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -315,8 +315,8 @@ static int rt5682_sdw_init(struct device *dev, struct regmap *regmap, &rt5682_sdw_indirect_regmap); if (IS_ERR(rt5682->regmap)) { ret = PTR_ERR(rt5682->regmap); - dev_err(dev, "Failed to allocate register map: %d\n", - ret); + dev_err(dev, "%s: Failed to allocate register map: %d\n", + __func__, ret); return ret; } @@ -400,7 +400,7 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave) } if (val != DEVICE_ID) { - dev_err(dev, "Device with ID register %x is not rt5682\n", val); + dev_err(dev, "%s: Device with ID register %x is not rt5682\n", __func__, val); ret = -ENODEV; goto err_nodev; } @@ -648,7 +648,7 @@ static int rt5682_bus_config(struct sdw_slave *slave, ret = rt5682_clock_config(&slave->dev); if (ret < 0) - dev_err(&slave->dev, "Invalid clk config"); + dev_err(&slave->dev, "%s: Invalid clk config", __func__); return ret; } @@ -763,19 +763,19 @@ static int __maybe_unused rt5682_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt5682->disable_irq_lock); if (rt5682->disable_irq == true) { - mutex_lock(&rt5682->disable_irq_lock); sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); rt5682->disable_irq = false; - mutex_unlock(&rt5682->disable_irq_lock); } + mutex_unlock(&rt5682->disable_irq_lock); goto regmap_sync; } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index 0ebf344a1b60..434b926f96c8 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -37,8 +37,8 @@ static int rt700_index_write(struct regmap *regmap, ret = regmap_write(regmap, addr, value); if (ret < 0) - pr_err("Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -52,8 +52,8 @@ static int rt700_index_read(struct regmap *regmap, *value = 0; ret = regmap_read(regmap, addr, value); if (ret < 0) - pr_err("Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -930,14 +930,14 @@ static int rt700_pcm_hw_params(struct snd_pcm_substream *substream, port_config.num += 2; break; default: - dev_err(component->dev, "Invalid DAI id %d\n", dai->id); + dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id); return -EINVAL; } retval = sdw_stream_add_slave(rt700->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -945,8 +945,8 @@ static int rt700_pcm_hw_params(struct snd_pcm_substream *substream, /* bit 3:0 Number of Channel */ val |= (params_channels(params) - 1); } else { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index 935e597022d3..2636c2eea4bc 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -438,20 +438,20 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt711->disable_irq_lock); if (rt711->disable_irq == true) { - mutex_lock(&rt711->disable_irq_lock); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); rt711->disable_irq = false; - mutex_unlock(&rt711->disable_irq_lock); } + mutex_unlock(&rt711->disable_irq_lock); goto regmap_sync; } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 447154cb6010..1e8dbfc3ecd9 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -36,8 +36,8 @@ static int rt711_sdca_index_write(struct rt711_sdca_priv *rt711, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt711->slave->dev, - "Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + "%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -52,8 +52,8 @@ static int rt711_sdca_index_read(struct rt711_sdca_priv *rt711, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt711->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -1293,13 +1293,13 @@ static int rt711_sdca_pcm_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt711->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } if (params_channels(params) > 16) { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } @@ -1318,8 +1318,8 @@ static int rt711_sdca_pcm_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT711_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 3f5773310ae8..0d3b43dd22e6 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -408,7 +408,7 @@ static int rt711_bus_config(struct sdw_slave *slave, ret = rt711_clock_config(&slave->dev); if (ret < 0) - dev_err(&slave->dev, "Invalid clk config"); + dev_err(&slave->dev, "%s: Invalid clk config", __func__); return ret; } @@ -536,19 +536,19 @@ static int __maybe_unused rt711_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt711->disable_irq_lock); if (rt711->disable_irq == true) { - mutex_lock(&rt711->disable_irq_lock); sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); rt711->disable_irq = false; - mutex_unlock(&rt711->disable_irq_lock); } + mutex_unlock(&rt711->disable_irq_lock); goto regmap_sync; } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); return -ETIMEDOUT; } diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 66eaed13b0d6..5446f9506a16 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -37,8 +37,8 @@ static int rt711_index_write(struct regmap *regmap, ret = regmap_write(regmap, addr, value); if (ret < 0) - pr_err("Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -52,8 +52,8 @@ static int rt711_index_read(struct regmap *regmap, *value = 0; ret = regmap_read(regmap, addr, value); if (ret < 0) - pr_err("Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -428,7 +428,7 @@ static void rt711_jack_init(struct rt711_priv *rt711) RT711_HP_JD_FINAL_RESULT_CTL_JD12); break; default: - dev_warn(rt711->component->dev, "Wrong JD source\n"); + dev_warn(rt711->component->dev, "%s: Wrong JD source\n", __func__); break; } @@ -1020,7 +1020,7 @@ static int rt711_pcm_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt711->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -1028,8 +1028,8 @@ static int rt711_pcm_hw_params(struct snd_pcm_substream *substream, /* bit 3:0 Number of Channel */ val |= (params_channels(params) - 1); } else { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt712-sdca-dmic.c b/sound/soc/codecs/rt712-sdca-dmic.c index 0926b26619bd..012b79e72cf6 100644 --- a/sound/soc/codecs/rt712-sdca-dmic.c +++ b/sound/soc/codecs/rt712-sdca-dmic.c @@ -139,8 +139,8 @@ static int rt712_sdca_dmic_index_write(struct rt712_sdca_dmic_priv *rt712, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt712->slave->dev, - "Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + "%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -155,8 +155,8 @@ static int rt712_sdca_dmic_index_read(struct rt712_sdca_dmic_priv *rt712, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt712->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -317,7 +317,8 @@ static int rt712_sdca_dmic_set_gain_put(struct snd_kcontrol *kcontrol, for (i = 0; i < p->count; i++) { err = regmap_write(rt712->mbq_regmap, p->reg_base + i, gain_val[i]); if (err < 0) - dev_err(&rt712->slave->dev, "0x%08x can't be set\n", p->reg_base + i); + dev_err(&rt712->slave->dev, "%s: 0x%08x can't be set\n", + __func__, p->reg_base + i); } return changed; @@ -667,13 +668,13 @@ static int rt712_sdca_dmic_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt712->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } if (params_channels(params) > 4) { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } @@ -698,8 +699,8 @@ static int rt712_sdca_dmic_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT712_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } @@ -923,7 +924,8 @@ static int __maybe_unused rt712_sdca_dmic_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT712_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", + __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c index 01ac555cd79b..4e9ab3ef135b 100644 --- a/sound/soc/codecs/rt712-sdca-sdw.c +++ b/sound/soc/codecs/rt712-sdca-sdw.c @@ -438,20 +438,21 @@ static int __maybe_unused rt712_sdca_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt712->disable_irq_lock); if (rt712->disable_irq == true) { - mutex_lock(&rt712->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); rt712->disable_irq = false; - mutex_unlock(&rt712->disable_irq_lock); } + mutex_unlock(&rt712->disable_irq_lock); goto regmap_sync; } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT712_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt712-sdca.c b/sound/soc/codecs/rt712-sdca.c index 6954fbe7ec5f..b503de9fda80 100644 --- a/sound/soc/codecs/rt712-sdca.c +++ b/sound/soc/codecs/rt712-sdca.c @@ -34,8 +34,8 @@ static int rt712_sdca_index_write(struct rt712_sdca_priv *rt712, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt712->slave->dev, - "Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + "%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -50,8 +50,8 @@ static int rt712_sdca_index_read(struct rt712_sdca_priv *rt712, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt712->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -1060,13 +1060,13 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt712->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } if (params_channels(params) > 16) { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } @@ -1085,8 +1085,8 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT712_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } @@ -1106,7 +1106,7 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream, sampling_rate); break; default: - dev_err(component->dev, "Wrong DAI id\n"); + dev_err(component->dev, "%s: Wrong DAI id\n", __func__); return -EINVAL; } diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index ab54a67a27eb..ee450126106f 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -237,7 +237,7 @@ static int __maybe_unused rt715_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->enumeration_complete, msecs_to_jiffies(RT715_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Enumeration not complete, timed out\n"); + dev_err(&slave->dev, "%s: Enumeration not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 4533eedd7e18..3fb7b9adb61d 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -41,8 +41,8 @@ static int rt715_sdca_index_write(struct rt715_sdca_priv *rt715, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt715->slave->dev, - "Failed to set private value: %08x <= %04x %d\n", - addr, value, ret); + "%s: Failed to set private value: %08x <= %04x %d\n", + __func__, addr, value, ret); return ret; } @@ -59,8 +59,8 @@ static int rt715_sdca_index_read(struct rt715_sdca_priv *rt715, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt715->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -152,8 +152,8 @@ static int rt715_sdca_set_amp_gain_put(struct snd_kcontrol *kcontrol, mc->shift); ret = regmap_write(rt715->mbq_regmap, mc->reg + i, gain_val); if (ret != 0) { - dev_err(component->dev, "Failed to write 0x%x=0x%x\n", - mc->reg + i, gain_val); + dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n", + __func__, mc->reg + i, gain_val); return ret; } } @@ -188,8 +188,8 @@ static int rt715_sdca_set_amp_gain_4ch_put(struct snd_kcontrol *kcontrol, ret = regmap_write(rt715->mbq_regmap, reg_base + i, gain_val); if (ret != 0) { - dev_err(component->dev, "Failed to write 0x%x=0x%x\n", - reg_base + i, gain_val); + dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n", + __func__, reg_base + i, gain_val); return ret; } } @@ -224,8 +224,8 @@ static int rt715_sdca_set_amp_gain_8ch_put(struct snd_kcontrol *kcontrol, reg = i < 7 ? reg_base + i : (reg_base - 1) | BIT(15); ret = regmap_write(rt715->mbq_regmap, reg, gain_val); if (ret != 0) { - dev_err(component->dev, "Failed to write 0x%x=0x%x\n", - reg, gain_val); + dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n", + __func__, reg, gain_val); return ret; } } @@ -246,8 +246,8 @@ static int rt715_sdca_set_amp_gain_get(struct snd_kcontrol *kcontrol, for (i = 0; i < 2; i++) { ret = regmap_read(rt715->mbq_regmap, mc->reg + i, &val); if (ret < 0) { - dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", - mc->reg + i, ret); + dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n", + __func__, mc->reg + i, ret); return ret; } ucontrol->value.integer.value[i] = rt715_sdca_get_gain(val, mc->shift); @@ -271,8 +271,8 @@ static int rt715_sdca_set_amp_gain_4ch_get(struct snd_kcontrol *kcontrol, for (i = 0; i < 4; i++) { ret = regmap_read(rt715->mbq_regmap, reg_base + i, &val); if (ret < 0) { - dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", - reg_base + i, ret); + dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n", + __func__, reg_base + i, ret); return ret; } ucontrol->value.integer.value[i] = rt715_sdca_get_gain(val, gain_sft); @@ -297,8 +297,8 @@ static int rt715_sdca_set_amp_gain_8ch_get(struct snd_kcontrol *kcontrol, for (i = 0; i < 8; i += 2) { ret = regmap_read(rt715->mbq_regmap, reg_base + i, &val_l); if (ret < 0) { - dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", - reg_base + i, ret); + dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n", + __func__, reg_base + i, ret); return ret; } ucontrol->value.integer.value[i] = (val_l >> gain_sft) / 10; @@ -306,8 +306,8 @@ static int rt715_sdca_set_amp_gain_8ch_get(struct snd_kcontrol *kcontrol, reg = (i == 6) ? (reg_base - 1) | BIT(15) : reg_base + 1 + i; ret = regmap_read(rt715->mbq_regmap, reg, &val_r); if (ret < 0) { - dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", - reg, ret); + dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n", + __func__, reg, ret); return ret; } ucontrol->value.integer.value[i + 1] = (val_r >> gain_sft) / 10; @@ -834,15 +834,15 @@ static int rt715_sdca_pcm_hw_params(struct snd_pcm_substream *substream, 0xaf00); break; default: - dev_err(component->dev, "Invalid DAI id %d\n", dai->id); + dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id); return -EINVAL; } retval = sdw_stream_add_slave(rt715->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(component->dev, "Unable to configure port, retval:%d\n", - retval); + dev_err(component->dev, "%s: Unable to configure port, retval:%d\n", + __func__, retval); return retval; } @@ -893,8 +893,8 @@ static int rt715_sdca_pcm_hw_params(struct snd_pcm_substream *substream, val = 0xf; break; default: - dev_err(component->dev, "Unsupported sample rate %d\n", - params_rate(params)); + dev_err(component->dev, "%s: Unsupported sample rate %d\n", + __func__, params_rate(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index 21f37babd148..7e13868ff99f 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -482,7 +482,7 @@ static int rt715_bus_config(struct sdw_slave *slave, ret = rt715_clock_config(&slave->dev); if (ret < 0) - dev_err(&slave->dev, "Invalid clk config"); + dev_err(&slave->dev, "%s: Invalid clk config", __func__); return 0; } @@ -554,7 +554,7 @@ static int __maybe_unused rt715_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT715_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 9f732a5abd53..299c9b12377c 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -40,8 +40,8 @@ static int rt715_index_write(struct regmap *regmap, unsigned int reg, ret = regmap_write(regmap, addr, value); if (ret < 0) { - pr_err("Failed to set private value: %08x <= %04x %d\n", - addr, value, ret); + pr_err("%s: Failed to set private value: %08x <= %04x %d\n", + __func__, addr, value, ret); } return ret; @@ -55,8 +55,8 @@ static int rt715_index_write_nid(struct regmap *regmap, ret = regmap_write(regmap, addr, value); if (ret < 0) - pr_err("Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -70,8 +70,8 @@ static int rt715_index_read_nid(struct regmap *regmap, *value = 0; ret = regmap_read(regmap, addr, value); if (ret < 0) - pr_err("Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -862,14 +862,14 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream, rt715_index_write(rt715->regmap, RT715_SDW_INPUT_SEL, 0xa000); break; default: - dev_err(component->dev, "Invalid DAI id %d\n", dai->id); + dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id); return -EINVAL; } retval = sdw_stream_add_slave(rt715->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -883,8 +883,8 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream, val |= 0x0 << 8; break; default: - dev_err(component->dev, "Unsupported sample rate %d\n", - params_rate(params)); + dev_err(component->dev, "%s: Unsupported sample rate %d\n", + __func__, params_rate(params)); return -EINVAL; } @@ -892,8 +892,8 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream, /* bit 3:0 Number of Channel */ val |= (params_channels(params) - 1); } else { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index eb76f4c675b6..65d584c1886e 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -467,13 +467,13 @@ static int __maybe_unused rt722_sdca_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt722->disable_irq_lock); if (rt722->disable_irq == true) { - mutex_lock(&rt722->disable_irq_lock); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); rt722->disable_irq = false; - mutex_unlock(&rt722->disable_irq_lock); } + mutex_unlock(&rt722->disable_irq_lock); goto regmap_sync; } diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c index 0e1c65a20392..e0ea3a23f7cc 100644 --- a/sound/soc/codecs/rt722-sdca.c +++ b/sound/soc/codecs/rt722-sdca.c @@ -35,8 +35,8 @@ int rt722_sdca_index_write(struct rt722_sdca_priv *rt722, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt722->slave->dev, - "Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + "%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -51,8 +51,8 @@ int rt722_sdca_index_read(struct rt722_sdca_priv *rt722, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt722->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -663,7 +663,8 @@ static int rt722_sdca_dmic_set_gain_put(struct snd_kcontrol *kcontrol, for (i = 0; i < p->count; i++) { err = regmap_write(rt722->mbq_regmap, p->reg_base + i, gain_val[i]); if (err < 0) - dev_err(&rt722->slave->dev, "%#08x can't be set\n", p->reg_base + i); + dev_err(&rt722->slave->dev, "%s: %#08x can't be set\n", + __func__, p->reg_base + i); } return changed; @@ -1211,13 +1212,13 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt722->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } if (params_channels(params) > 16) { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } @@ -1236,8 +1237,8 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT722_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index e451c009f2d9..7d5c096e06cd 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -683,11 +683,12 @@ static void wm_adsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - struct cs_dsp_coeff_ctl *cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); + struct cs_dsp_coeff_ctl *cs_ctl; struct wm_coeff_ctl *ctl; int ret; mutex_lock(&dsp->cs_dsp.pwr_lock); + cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len); mutex_unlock(&dsp->cs_dsp.pwr_lock); diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index c018f84fe025..fc072dc58968 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -296,5 +296,6 @@ static struct platform_driver avs_da7219_driver = { module_platform_driver(avs_da7219_driver); +MODULE_DESCRIPTION("Intel da7219 machine driver"); MODULE_AUTHOR("Cezary Rojewski <cezary.rojewski@intel.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/dmic.c b/sound/soc/intel/avs/boards/dmic.c index ba2bc7f689eb..d9e5e85f5233 100644 --- a/sound/soc/intel/avs/boards/dmic.c +++ b/sound/soc/intel/avs/boards/dmic.c @@ -96,4 +96,5 @@ static struct platform_driver avs_dmic_driver = { module_platform_driver(avs_dmic_driver); +MODULE_DESCRIPTION("Intel DMIC machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index 1090082e7d5b..5c90a6007577 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -326,4 +326,5 @@ static struct platform_driver avs_es8336_driver = { module_platform_driver(avs_es8336_driver); +MODULE_DESCRIPTION("Intel es8336 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c index 28f254eb0d03..027373d6a16d 100644 --- a/sound/soc/intel/avs/boards/i2s_test.c +++ b/sound/soc/intel/avs/boards/i2s_test.c @@ -204,4 +204,5 @@ static struct platform_driver avs_i2s_test_driver = { module_platform_driver(avs_i2s_test_driver); +MODULE_DESCRIPTION("Intel i2s test machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c index a83b95f25129..1ff85e4d8e16 100644 --- a/sound/soc/intel/avs/boards/max98357a.c +++ b/sound/soc/intel/avs/boards/max98357a.c @@ -154,4 +154,5 @@ static struct platform_driver avs_max98357a_driver = { module_platform_driver(avs_max98357a_driver) +MODULE_DESCRIPTION("Intel max98357a machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/max98373.c b/sound/soc/intel/avs/boards/max98373.c index 3b980a025e6f..8d31586b73ea 100644 --- a/sound/soc/intel/avs/boards/max98373.c +++ b/sound/soc/intel/avs/boards/max98373.c @@ -211,4 +211,5 @@ static struct platform_driver avs_max98373_driver = { module_platform_driver(avs_max98373_driver) +MODULE_DESCRIPTION("Intel max98373 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/max98927.c b/sound/soc/intel/avs/boards/max98927.c index 86dd2b228df3..572ec58073d0 100644 --- a/sound/soc/intel/avs/boards/max98927.c +++ b/sound/soc/intel/avs/boards/max98927.c @@ -208,4 +208,5 @@ static struct platform_driver avs_max98927_driver = { module_platform_driver(avs_max98927_driver) +MODULE_DESCRIPTION("Intel max98927 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index 1c1e2083f474..55db75efae41 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -313,4 +313,5 @@ static struct platform_driver avs_nau8825_driver = { module_platform_driver(avs_nau8825_driver) +MODULE_DESCRIPTION("Intel nau8825 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/probe.c b/sound/soc/intel/avs/boards/probe.c index a9469b5ecb40..8be6887bbc6e 100644 --- a/sound/soc/intel/avs/boards/probe.c +++ b/sound/soc/intel/avs/boards/probe.c @@ -69,4 +69,5 @@ static struct platform_driver avs_probe_mb_driver = { module_platform_driver(avs_probe_mb_driver); +MODULE_DESCRIPTION("Intel probe machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c index bfcb8845fd15..1cf524216087 100644 --- a/sound/soc/intel/avs/boards/rt274.c +++ b/sound/soc/intel/avs/boards/rt274.c @@ -276,4 +276,5 @@ static struct platform_driver avs_rt274_driver = { module_platform_driver(avs_rt274_driver); +MODULE_DESCRIPTION("Intel rt274 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c index 28d7d86b1cc9..4740bba10570 100644 --- a/sound/soc/intel/avs/boards/rt286.c +++ b/sound/soc/intel/avs/boards/rt286.c @@ -247,4 +247,5 @@ static struct platform_driver avs_rt286_driver = { module_platform_driver(avs_rt286_driver); +MODULE_DESCRIPTION("Intel rt286 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c index 80f490b9e118..6e409e29f697 100644 --- a/sound/soc/intel/avs/boards/rt298.c +++ b/sound/soc/intel/avs/boards/rt298.c @@ -266,4 +266,5 @@ static struct platform_driver avs_rt298_driver = { module_platform_driver(avs_rt298_driver); +MODULE_DESCRIPTION("Intel rt298 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt5514.c b/sound/soc/intel/avs/boards/rt5514.c index 60105f453ae2..097ae5f73241 100644 --- a/sound/soc/intel/avs/boards/rt5514.c +++ b/sound/soc/intel/avs/boards/rt5514.c @@ -192,4 +192,5 @@ static struct platform_driver avs_rt5514_driver = { module_platform_driver(avs_rt5514_driver); +MODULE_DESCRIPTION("Intel rt5514 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt5663.c b/sound/soc/intel/avs/boards/rt5663.c index b4762c2a7bf2..1880c315cc4d 100644 --- a/sound/soc/intel/avs/boards/rt5663.c +++ b/sound/soc/intel/avs/boards/rt5663.c @@ -265,4 +265,5 @@ static struct platform_driver avs_rt5663_driver = { module_platform_driver(avs_rt5663_driver); +MODULE_DESCRIPTION("Intel rt5663 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c index 243f979fda98..594a971ded9e 100644 --- a/sound/soc/intel/avs/boards/rt5682.c +++ b/sound/soc/intel/avs/boards/rt5682.c @@ -341,5 +341,6 @@ static struct platform_driver avs_rt5682_driver = { module_platform_driver(avs_rt5682_driver) +MODULE_DESCRIPTION("Intel rt5682 machine driver"); MODULE_AUTHOR("Cezary Rojewski <cezary.rojewski@intel.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c index 4a0e136835ff..d6f7f046c24e 100644 --- a/sound/soc/intel/avs/boards/ssm4567.c +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -200,4 +200,5 @@ static struct platform_driver avs_ssm4567_driver = { module_platform_driver(avs_ssm4567_driver) +MODULE_DESCRIPTION("Intel ssm4567 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 2d25748ca706..b27e89ff6a16 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -263,7 +263,7 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; int min = mc->min; int sign_bit = mc->sign_bit; - unsigned int mask = (1 << fls(max)) - 1; + unsigned int mask = (1ULL << fls(max)) - 1; unsigned int invert = mc->invert; int val; int ret; diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index be7dc1e02284..c12c7f820529 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -704,6 +704,10 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) goto unregister_dev; } + ret = acp_init(sdev); + if (ret < 0) + goto free_smn_dev; + sdev->ipc_irq = pci->irq; ret = request_threaded_irq(sdev->ipc_irq, acp_irq_handler, acp_irq_thread, IRQF_SHARED, "AudioDSP", sdev); @@ -713,10 +717,6 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) goto free_smn_dev; } - ret = acp_init(sdev); - if (ret < 0) - goto free_ipc_irq; - /* scan SoundWire capabilities exposed by DSDT */ ret = acp_sof_scan_sdw_devices(sdev, chip->sdw_acpi_dev_addr); if (ret < 0) { diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 9b00ede2a486..cc84d4c81be9 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -339,8 +339,7 @@ static int sof_init_environment(struct snd_sof_dev *sdev) ret = snd_sof_probe(sdev); if (ret < 0) { dev_err(sdev->dev, "failed to probe DSP %d\n", ret); - sof_ops_free(sdev); - return ret; + goto err_sof_probe; } /* check machine info */ @@ -358,15 +357,18 @@ static int sof_init_environment(struct snd_sof_dev *sdev) ret = validate_sof_ops(sdev); if (ret < 0) { snd_sof_remove(sdev); + snd_sof_remove_late(sdev); return ret; } } + return 0; + err_machine_check: - if (ret) { - snd_sof_remove(sdev); - sof_ops_free(sdev); - } + snd_sof_remove(sdev); +err_sof_probe: + snd_sof_remove_late(sdev); + sof_ops_free(sdev); return ret; } diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 2b385cddc385..d71bb66b9991 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -57,6 +57,9 @@ struct snd_sof_dsp_ops sof_hda_common_ops = { .pcm_pointer = hda_dsp_pcm_pointer, .pcm_ack = hda_dsp_pcm_ack, + .get_dai_frame_counter = hda_dsp_get_stream_llp, + .get_host_byte_counter = hda_dsp_get_stream_ldp, + /* firmware loading */ .load_firmware = snd_sof_load_firmware_raw, diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index c50ca9e72d37..b073720b4cf4 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -7,6 +7,7 @@ #include <sound/pcm_params.h> #include <sound/hdaudio_ext.h> +#include <sound/hda_register.h> #include <sound/hda-mlink.h> #include <sound/sof/ipc4/header.h> #include <uapi/sound/sof/header.h> @@ -362,6 +363,16 @@ static int hda_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: snd_hdac_ext_stream_clear(hext_stream); + + /* + * Save the LLP registers in case the stream is + * restarting due PAUSE_RELEASE, or START without a pcm + * close/open since in this case the LLP register is not reset + * to 0 and the delay calculation will return with invalid + * results. + */ + hext_stream->pplcllpl = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL); + hext_stream->pplcllpu = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU); break; default: dev_err(sdev->dev, "unknown trigger command %d\n", cmd); diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 31ffa1a8f2ac..ef5c915db8ff 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -681,17 +681,27 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; const struct sof_intel_dsp_desc *chip = hda->desc; struct hdac_bus *bus = sof_to_bus(sdev); + bool imr_lost = false; int ret, j; /* - * The memory used for IMR boot loses its content in deeper than S3 state - * We must not try IMR boot on next power up (as it will fail). - * + * The memory used for IMR boot loses its content in deeper than S3 + * state on CAVS platforms. + * On ACE platforms due to the system architecture the IMR content is + * lost at S3 state already, they are tailored for s2idle use. + * We must not try IMR boot on next power up in these cases as it will + * fail. + */ + if (sdev->system_suspend_target > SOF_SUSPEND_S3 || + (chip->hw_ip_version >= SOF_INTEL_ACE_1_0 && + sdev->system_suspend_target == SOF_SUSPEND_S3)) + imr_lost = true; + + /* * In case of firmware crash or boot failure set the skip_imr_boot to true * as well in order to try to re-load the firmware to do a 'cold' boot. */ - if (sdev->system_suspend_target > SOF_SUSPEND_S3 || - sdev->fw_state == SOF_FW_CRASHED || + if (imr_lost || sdev->fw_state == SOF_FW_CRASHED || sdev->fw_state == SOF_FW_BOOT_FAILED) hda->skip_imr_boot = true; diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index 18f07364d219..d7b446f3f973 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -259,8 +259,37 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev, snd_pcm_hw_constraint_mask64(substream->runtime, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S32); + /* + * The dsp_max_burst_size_in_ms is the length of the maximum burst size + * of the host DMA in the ALSA buffer. + * + * On playback start the DMA will transfer dsp_max_burst_size_in_ms + * amount of data in one initial burst to fill up the host DMA buffer. + * Consequent DMA burst sizes are shorter and their length can vary. + * To make sure that userspace allocate large enough ALSA buffer we need + * to place a constraint on the buffer time. + * + * On capture the DMA will transfer 1ms chunks. + * + * Exact dsp_max_burst_size_in_ms constraint is racy, so set the + * constraint to a minimum of 2x dsp_max_burst_size_in_ms. + */ + if (spcm->stream[direction].dsp_max_burst_size_in_ms) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_TIME, + spcm->stream[direction].dsp_max_burst_size_in_ms * USEC_PER_MSEC * 2, + UINT_MAX); + /* binding pcm substream to hda stream */ substream->runtime->private_data = &dsp_stream->hstream; + + /* + * Reset the llp cache values (they are used for LLP compensation in + * case the counter is not reset) + */ + dsp_stream->pplcllpl = 0; + dsp_stream->pplcllpu = 0; + return 0; } diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index b387b1a69d7e..0c189d3b19c1 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -1063,3 +1063,73 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, return pos; } + +#define merge_u64(u32_u, u32_l) (((u64)(u32_u) << 32) | (u32_l)) + +/** + * hda_dsp_get_stream_llp - Retrieve the LLP (Linear Link Position) of the stream + * @sdev: SOF device + * @component: ASoC component + * @substream: PCM substream + * + * Returns the raw Linear Link Position value + */ +u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct hdac_stream *hstream = substream->runtime->private_data; + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); + u32 llp_l, llp_u; + + /* + * The pplc_addr have been calculated during probe in + * hda_dsp_stream_init(): + * pplc_addr = sdev->bar[HDA_DSP_PP_BAR] + + * SOF_HDA_PPLC_BASE + + * SOF_HDA_PPLC_MULTI * total_stream + + * SOF_HDA_PPLC_INTERVAL * stream_index + * + * Use this pre-calculated address to avoid repeated re-calculation. + */ + llp_l = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL); + llp_u = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU); + + /* Compensate the LLP counter with the saved offset */ + if (hext_stream->pplcllpl || hext_stream->pplcllpu) + return merge_u64(llp_u, llp_l) - + merge_u64(hext_stream->pplcllpu, hext_stream->pplcllpl); + + return merge_u64(llp_u, llp_l); +} + +/** + * hda_dsp_get_stream_ldp - Retrieve the LDP (Linear DMA Position) of the stream + * @sdev: SOF device + * @component: ASoC component + * @substream: PCM substream + * + * Returns the raw Linear Link Position value + */ +u64 hda_dsp_get_stream_ldp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct hdac_stream *hstream = substream->runtime->private_data; + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); + u32 ldp_l, ldp_u; + + /* + * The pphc_addr have been calculated during probe in + * hda_dsp_stream_init(): + * pphc_addr = sdev->bar[HDA_DSP_PP_BAR] + + * SOF_HDA_PPHC_BASE + + * SOF_HDA_PPHC_INTERVAL * stream_index + * + * Use this pre-calculated address to avoid repeated re-calculation. + */ + ldp_l = readl(hext_stream->pphc_addr + AZX_REG_PPHCLDPL); + ldp_u = readl(hext_stream->pphc_addr + AZX_REG_PPHCLDPU); + + return ((u64)ldp_u << 32) | ldp_l; +} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index b36eb7c78913..81a1d4606d3c 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -662,6 +662,12 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev); snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, int direction, bool can_sleep); +u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); +u64 hda_dsp_get_stream_ldp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags); diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 7ae017a00184..aeb4350cce6b 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -29,15 +29,17 @@ static const struct snd_sof_debugfs_map lnl_dsp_debugfs[] = { }; /* this helps allows the DSP to setup DMIC/SSP */ -static int hdac_bus_offload_dmic_ssp(struct hdac_bus *bus) +static int hdac_bus_offload_dmic_ssp(struct hdac_bus *bus, bool enable) { int ret; - ret = hdac_bus_eml_enable_offload(bus, true, AZX_REG_ML_LEPTR_ID_INTEL_SSP, true); + ret = hdac_bus_eml_enable_offload(bus, true, + AZX_REG_ML_LEPTR_ID_INTEL_SSP, enable); if (ret < 0) return ret; - ret = hdac_bus_eml_enable_offload(bus, true, AZX_REG_ML_LEPTR_ID_INTEL_DMIC, true); + ret = hdac_bus_eml_enable_offload(bus, true, + AZX_REG_ML_LEPTR_ID_INTEL_DMIC, enable); if (ret < 0) return ret; @@ -52,7 +54,19 @@ static int lnl_hda_dsp_probe(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev)); + return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true); +} + +static void lnl_hda_dsp_remove(struct snd_sof_dev *sdev) +{ + int ret; + + ret = hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), false); + if (ret < 0) + dev_warn(sdev->dev, + "Failed to disable offload for DMIC/SSP: %d\n", ret); + + hda_dsp_remove(sdev); } static int lnl_hda_dsp_resume(struct snd_sof_dev *sdev) @@ -63,7 +77,7 @@ static int lnl_hda_dsp_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev)); + return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true); } static int lnl_hda_dsp_runtime_resume(struct snd_sof_dev *sdev) @@ -74,7 +88,7 @@ static int lnl_hda_dsp_runtime_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev)); + return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true); } static int lnl_dsp_post_fw_run(struct snd_sof_dev *sdev) @@ -97,9 +111,11 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) /* common defaults */ memcpy(&sof_lnl_ops, &sof_hda_common_ops, sizeof(struct snd_sof_dsp_ops)); - /* probe */ - if (!sdev->dspless_mode_selected) + /* probe/remove */ + if (!sdev->dspless_mode_selected) { sof_lnl_ops.probe = lnl_hda_dsp_probe; + sof_lnl_ops.remove = lnl_hda_dsp_remove; + } /* shutdown */ sof_lnl_ops.shutdown = hda_dsp_shutdown; @@ -134,8 +150,6 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) sof_lnl_ops.runtime_resume = lnl_hda_dsp_runtime_resume; } - sof_lnl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position; - /* dsp core get/put */ sof_lnl_ops.core_get = mtl_dsp_core_get; sof_lnl_ops.core_put = mtl_dsp_core_put; diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index df05dc77b8d5..060c34988e90 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -626,18 +626,6 @@ static int mtl_dsp_disable_interrupts(struct snd_sof_dev *sdev) return mtl_enable_interrupts(sdev, false); } -u64 mtl_dsp_get_stream_hda_link_position(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - struct hdac_stream *hstream = substream->runtime->private_data; - u32 llp_l, llp_u; - - llp_l = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, MTL_PPLCLLPL(hstream->index)); - llp_u = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, MTL_PPLCLLPU(hstream->index)); - return ((u64)llp_u << 32) | llp_l; -} - int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core) { const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm; @@ -707,8 +695,6 @@ int sof_mtl_ops_init(struct snd_sof_dev *sdev) sof_mtl_ops.core_get = mtl_dsp_core_get; sof_mtl_ops.core_put = mtl_dsp_core_put; - sof_mtl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position; - sdev->private = kzalloc(sizeof(struct sof_ipc4_fw_data), GFP_KERNEL); if (!sdev->private) return -ENOMEM; diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h index cc5a1f46fd09..ea8c1b83f712 100644 --- a/sound/soc/sof/intel/mtl.h +++ b/sound/soc/sof/intel/mtl.h @@ -6,12 +6,6 @@ * Copyright(c) 2020-2022 Intel Corporation. All rights reserved. */ -/* HDA Registers */ -#define MTL_PPLCLLPL_BASE 0x948 -#define MTL_PPLCLLPU_STRIDE 0x10 -#define MTL_PPLCLLPL(x) (MTL_PPLCLLPL_BASE + (x) * MTL_PPLCLLPU_STRIDE) -#define MTL_PPLCLLPU(x) (MTL_PPLCLLPL_BASE + 0x4 + (x) * MTL_PPLCLLPU_STRIDE) - /* DSP Registers */ #define MTL_HFDSSCS 0x1000 #define MTL_HFDSSCS_SPA_MASK BIT(16) @@ -103,9 +97,5 @@ int mtl_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id); void mtl_ipc_dump(struct snd_sof_dev *sdev); -u64 mtl_dsp_get_stream_hda_link_position(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream); - int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core); int mtl_dsp_core_put(struct snd_sof_dev *sdev, int core); diff --git a/sound/soc/sof/ipc4-mtrace.c b/sound/soc/sof/ipc4-mtrace.c index 9f1e33ee8826..0e04bea9432d 100644 --- a/sound/soc/sof/ipc4-mtrace.c +++ b/sound/soc/sof/ipc4-mtrace.c @@ -4,6 +4,7 @@ #include <linux/debugfs.h> #include <linux/sched/signal.h> +#include <linux/sched/clock.h> #include <sound/sof/ipc4/header.h> #include "sof-priv.h" #include "ipc4-priv.h" @@ -412,7 +413,6 @@ static int ipc4_mtrace_enable(struct snd_sof_dev *sdev) const struct sof_ipc_ops *iops = sdev->ipc->ops; struct sof_ipc4_msg msg; u64 system_time; - ktime_t kt; int ret; if (priv->mtrace_state != SOF_MTRACE_DISABLED) @@ -424,9 +424,12 @@ static int ipc4_mtrace_enable(struct snd_sof_dev *sdev) msg.primary |= SOF_IPC4_MOD_INSTANCE(SOF_IPC4_MOD_INIT_BASEFW_INSTANCE_ID); msg.extension = SOF_IPC4_MOD_EXT_MSG_PARAM_ID(SOF_IPC4_FW_PARAM_SYSTEM_TIME); - /* The system time is in usec, UTC, epoch is 1601-01-01 00:00:00 */ - kt = ktime_add_us(ktime_get_real(), FW_EPOCH_DELTA * USEC_PER_SEC); - system_time = ktime_to_us(kt); + /* + * local_clock() is used to align with dmesg, so both kernel and firmware logs have + * the same base and a minor delta due to the IPC. system time is in us format but + * local_clock() returns the time in ns, so convert to ns. + */ + system_time = div64_u64(local_clock(), NSEC_PER_USEC); msg.data_size = sizeof(system_time); msg.data_ptr = &system_time; ret = iops->set_get_data(sdev, &msg, msg.data_size, true); diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 0f332c8cdbe6..e915f9f87a6c 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -15,6 +15,28 @@ #include "ipc4-topology.h" #include "ipc4-fw-reg.h" +/** + * struct sof_ipc4_timestamp_info - IPC4 timestamp info + * @host_copier: the host copier of the pcm stream + * @dai_copier: the dai copier of the pcm stream + * @stream_start_offset: reported by fw in memory window (converted to frames) + * @stream_end_offset: reported by fw in memory window (converted to frames) + * @llp_offset: llp offset in memory window + * @boundary: wrap boundary should be used for the LLP frame counter + * @delay: Calculated and stored in pointer callback. The stored value is + * returned in the delay callback. + */ +struct sof_ipc4_timestamp_info { + struct sof_ipc4_copier *host_copier; + struct sof_ipc4_copier *dai_copier; + u64 stream_start_offset; + u64 stream_end_offset; + u32 llp_offset; + + u64 boundary; + snd_pcm_sframes_t delay; +}; + static int sof_ipc4_set_multi_pipeline_state(struct snd_sof_dev *sdev, u32 state, struct ipc4_pipeline_set_state_data *trigger_list) { @@ -423,8 +445,19 @@ static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component, } /* return if this is the final state */ - if (state == SOF_IPC4_PIPE_PAUSED) + if (state == SOF_IPC4_PIPE_PAUSED) { + struct sof_ipc4_timestamp_info *time_info; + + /* + * Invalidate the stream_start_offset to make sure that it is + * going to be updated if the stream resumes + */ + time_info = spcm->stream[substream->stream].private; + if (time_info) + time_info->stream_start_offset = SOF_IPC4_INVALID_STREAM_POSITION; + goto free; + } skip_pause_transition: /* else set the RUNNING/RESET state in the DSP */ ret = sof_ipc4_set_multi_pipeline_state(sdev, state, trigger_list); @@ -464,14 +497,12 @@ static int sof_ipc4_pcm_trigger(struct snd_soc_component *component, /* determine the pipeline state */ switch (cmd) { - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - state = SOF_IPC4_PIPE_PAUSED; - break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: state = SOF_IPC4_PIPE_RUNNING; break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: state = SOF_IPC4_PIPE_PAUSED; @@ -703,6 +734,10 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm if (abi_version < SOF_IPC4_FW_REGS_ABI_VER) support_info = false; + /* For delay reporting the get_host_byte_counter callback is needed */ + if (!sof_ops(sdev) || !sof_ops(sdev)->get_host_byte_counter) + support_info = false; + for_each_pcm_streams(stream) { pipeline_list = &spcm->stream[stream].pipeline_list; @@ -835,7 +870,6 @@ static int sof_ipc4_get_stream_start_offset(struct snd_sof_dev *sdev, struct sof_ipc4_copier *host_copier = time_info->host_copier; struct sof_ipc4_copier *dai_copier = time_info->dai_copier; struct sof_ipc4_pipeline_registers ppl_reg; - u64 stream_start_position; u32 dai_sample_size; u32 ch, node_index; u32 offset; @@ -852,38 +886,51 @@ static int sof_ipc4_get_stream_start_offset(struct snd_sof_dev *sdev, if (ppl_reg.stream_start_offset == SOF_IPC4_INVALID_STREAM_POSITION) return -EINVAL; - stream_start_position = ppl_reg.stream_start_offset; ch = dai_copier->data.out_format.fmt_cfg; ch = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(ch); dai_sample_size = (dai_copier->data.out_format.bit_depth >> 3) * ch; - /* convert offset to sample count */ - do_div(stream_start_position, dai_sample_size); - time_info->stream_start_offset = stream_start_position; + + /* convert offsets to frame count */ + time_info->stream_start_offset = ppl_reg.stream_start_offset; + do_div(time_info->stream_start_offset, dai_sample_size); + time_info->stream_end_offset = ppl_reg.stream_end_offset; + do_div(time_info->stream_end_offset, dai_sample_size); + + /* + * Calculate the wrap boundary need to be used for delay calculation + * The host counter is in bytes, it will wrap earlier than the frames + * based link counter. + */ + time_info->boundary = div64_u64(~((u64)0), + frames_to_bytes(substream->runtime, 1)); + /* Initialize the delay value to 0 (no delay) */ + time_info->delay = 0; return 0; } -static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, - struct snd_pcm_substream *substream) +static int sof_ipc4_pcm_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + snd_pcm_uframes_t *pointer) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sof_ipc4_timestamp_info *time_info; struct sof_ipc4_llp_reading_slot llp; - snd_pcm_uframes_t head_ptr, tail_ptr; + snd_pcm_uframes_t head_cnt, tail_cnt; struct snd_sof_pcm_stream *stream; + u64 dai_cnt, host_cnt, host_ptr; struct snd_sof_pcm *spcm; - u64 tmp_ptr; int ret; spcm = snd_sof_find_spcm_dai(component, rtd); if (!spcm) - return 0; + return -EOPNOTSUPP; stream = &spcm->stream[substream->stream]; time_info = stream->private; if (!time_info) - return 0; + return -EOPNOTSUPP; /* * stream_start_offset is updated to memory window by FW based on @@ -893,45 +940,116 @@ static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, if (time_info->stream_start_offset == SOF_IPC4_INVALID_STREAM_POSITION) { ret = sof_ipc4_get_stream_start_offset(sdev, substream, stream, time_info); if (ret < 0) - return 0; + return -EOPNOTSUPP; } + /* For delay calculation we need the host counter */ + host_cnt = snd_sof_pcm_get_host_byte_counter(sdev, component, substream); + host_ptr = host_cnt; + + /* convert the host_cnt to frames */ + host_cnt = div64_u64(host_cnt, frames_to_bytes(substream->runtime, 1)); + /* - * HDaudio links don't support the LLP counter reported by firmware - * the link position is read directly from hardware registers. + * If the LLP counter is not reported by firmware in the SRAM window + * then read the dai (link) counter via host accessible means if + * available. */ if (!time_info->llp_offset) { - tmp_ptr = snd_sof_pcm_get_stream_position(sdev, component, substream); - if (!tmp_ptr) - return 0; + dai_cnt = snd_sof_pcm_get_dai_frame_counter(sdev, component, substream); + if (!dai_cnt) + return -EOPNOTSUPP; } else { sof_mailbox_read(sdev, time_info->llp_offset, &llp, sizeof(llp)); - tmp_ptr = ((u64)llp.reading.llp_u << 32) | llp.reading.llp_l; + dai_cnt = ((u64)llp.reading.llp_u << 32) | llp.reading.llp_l; } + dai_cnt += time_info->stream_end_offset; - /* In two cases dai dma position is not accurate + /* In two cases dai dma counter is not accurate * (1) dai pipeline is started before host pipeline - * (2) multiple streams mixed into one. Each stream has the same dai dma position + * (2) multiple streams mixed into one. Each stream has the same dai dma + * counter * - * Firmware calculates correct stream_start_offset for all cases including above two. - * Driver subtracts stream_start_offset from dai dma position to get accurate one + * Firmware calculates correct stream_start_offset for all cases + * including above two. + * Driver subtracts stream_start_offset from dai dma counter to get + * accurate one */ - tmp_ptr -= time_info->stream_start_offset; - /* Calculate the delay taking into account that both pointer can wrap */ - div64_u64_rem(tmp_ptr, substream->runtime->boundary, &tmp_ptr); + /* + * On stream start the dai counter might not yet have reached the + * stream_start_offset value which means that no frames have left the + * DSP yet from the audio stream (on playback, capture streams have + * offset of 0 as we start capturing right away). + * In this case we need to adjust the distance between the counters by + * increasing the host counter by (offset - dai_counter). + * Otherwise the dai_counter needs to be adjusted to reflect the number + * of valid frames passed on the DAI side. + * + * The delay is the difference between the counters on the two + * sides of the DSP. + */ + if (dai_cnt < time_info->stream_start_offset) { + host_cnt += time_info->stream_start_offset - dai_cnt; + dai_cnt = 0; + } else { + dai_cnt -= time_info->stream_start_offset; + } + + /* Wrap the dai counter at the boundary where the host counter wraps */ + div64_u64_rem(dai_cnt, time_info->boundary, &dai_cnt); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - head_ptr = substream->runtime->status->hw_ptr; - tail_ptr = tmp_ptr; + head_cnt = host_cnt; + tail_cnt = dai_cnt; } else { - head_ptr = tmp_ptr; - tail_ptr = substream->runtime->status->hw_ptr; + head_cnt = dai_cnt; + tail_cnt = host_cnt; + } + + if (head_cnt < tail_cnt) { + time_info->delay = time_info->boundary - tail_cnt + head_cnt; + goto out; } - if (head_ptr < tail_ptr) - return substream->runtime->boundary - tail_ptr + head_ptr; + time_info->delay = head_cnt - tail_cnt; + +out: + /* + * Convert the host byte counter to PCM pointer which wraps in buffer + * and it is in frames + */ + div64_u64_rem(host_ptr, snd_pcm_lib_buffer_bytes(substream), &host_ptr); + *pointer = bytes_to_frames(substream->runtime, host_ptr); + + return 0; +} + +static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct sof_ipc4_timestamp_info *time_info; + struct snd_sof_pcm_stream *stream; + struct snd_sof_pcm *spcm; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return 0; + + stream = &spcm->stream[substream->stream]; + time_info = stream->private; + /* + * Report the stored delay value calculated in the pointer callback. + * In the unlikely event that the calculation was skipped/aborted, the + * default 0 delay returned. + */ + if (time_info) + return time_info->delay; + + /* No delay information available, report 0 as delay */ + return 0; - return head_ptr - tail_ptr; } const struct sof_ipc_pcm_ops ipc4_pcm_ops = { @@ -941,6 +1059,7 @@ const struct sof_ipc_pcm_ops ipc4_pcm_ops = { .dai_link_fixup = sof_ipc4_pcm_dai_link_fixup, .pcm_setup = sof_ipc4_pcm_setup, .pcm_free = sof_ipc4_pcm_free, + .pointer = sof_ipc4_pcm_pointer, .delay = sof_ipc4_pcm_delay, .ipc_first_on_start = true, .platform_stop_during_hw_free = true, diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index f3b908b093f9..afed618a15f0 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -92,20 +92,6 @@ struct sof_ipc4_fw_data { struct mutex pipeline_state_mutex; /* protect pipeline triggers, ref counts and states */ }; -/** - * struct sof_ipc4_timestamp_info - IPC4 timestamp info - * @host_copier: the host copier of the pcm stream - * @dai_copier: the dai copier of the pcm stream - * @stream_start_offset: reported by fw in memory window - * @llp_offset: llp offset in memory window - */ -struct sof_ipc4_timestamp_info { - struct sof_ipc4_copier *host_copier; - struct sof_ipc4_copier *dai_copier; - u64 stream_start_offset; - u32 llp_offset; -}; - extern const struct sof_ipc_fw_loader_ops ipc4_loader_ops; extern const struct sof_ipc_tplg_ops ipc4_tplg_ops; extern const struct sof_ipc_tplg_control_ops tplg_ipc4_control_ops; diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index da4a83afb87a..5cca05842126 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -412,8 +412,9 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget) struct sof_ipc4_available_audio_format *available_fmt; struct snd_soc_component *scomp = swidget->scomp; struct sof_ipc4_copier *ipc4_copier; + struct snd_sof_pcm *spcm; int node_type = 0; - int ret; + int ret, dir; ipc4_copier = kzalloc(sizeof(*ipc4_copier), GFP_KERNEL); if (!ipc4_copier) @@ -447,6 +448,25 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget) } dev_dbg(scomp->dev, "host copier '%s' node_type %u\n", swidget->widget->name, node_type); + spcm = snd_sof_find_spcm_comp(scomp, swidget->comp_id, &dir); + if (!spcm) + goto skip_gtw_cfg; + + if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + struct snd_sof_pcm_stream *sps = &spcm->stream[dir]; + + sof_update_ipc_object(scomp, &sps->dsp_max_burst_size_in_ms, + SOF_COPIER_DEEP_BUFFER_TOKENS, + swidget->tuples, + swidget->num_tuples, sizeof(u32), 1); + /* Set default DMA buffer size if it is not specified in topology */ + if (!sps->dsp_max_burst_size_in_ms) + sps->dsp_max_burst_size_in_ms = SOF_IPC4_MIN_DMA_BUFFER_SIZE; + } else { + /* Capture data is copied from DSP to host in 1ms bursts */ + spcm->stream[dir].dsp_max_burst_size_in_ms = 1; + } + skip_gtw_cfg: ipc4_copier->gtw_attr = kzalloc(sizeof(*ipc4_copier->gtw_attr), GFP_KERNEL); if (!ipc4_copier->gtw_attr) { @@ -1356,6 +1376,7 @@ static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_s int sample_rate, channel_count; int bit_depth, ret; u32 nhlt_type; + int dev_type = 0; /* convert to NHLT type */ switch (linktype) { @@ -1371,18 +1392,30 @@ static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_s &bit_depth); if (ret < 0) return ret; + + /* + * We need to know the type of the external device attached to a SSP + * port to retrieve the blob from NHLT. However, device type is not + * specified in topology. + * Query the type for the port and then pass that information back + * to the blob lookup function. + */ + dev_type = intel_nhlt_ssp_device_type(sdev->dev, ipc4_data->nhlt, + dai_index); + if (dev_type < 0) + return dev_type; break; default: return 0; } - dev_dbg(sdev->dev, "dai index %d nhlt type %d direction %d\n", - dai_index, nhlt_type, dir); + dev_dbg(sdev->dev, "dai index %d nhlt type %d direction %d dev type %d\n", + dai_index, nhlt_type, dir, dev_type); /* find NHLT blob with matching params */ cfg = intel_nhlt_get_endpoint_blob(sdev->dev, ipc4_data->nhlt, dai_index, nhlt_type, bit_depth, bit_depth, channel_count, sample_rate, - dir, 0); + dir, dev_type); if (!cfg) { dev_err(sdev->dev, diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 6cf21e829e07..3cd748e13460 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -523,12 +523,26 @@ static inline int snd_sof_pcm_platform_ack(struct snd_sof_dev *sdev, return 0; } -static inline u64 snd_sof_pcm_get_stream_position(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream) +static inline u64 +snd_sof_pcm_get_dai_frame_counter(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) { - if (sof_ops(sdev) && sof_ops(sdev)->get_stream_position) - return sof_ops(sdev)->get_stream_position(sdev, component, substream); + if (sof_ops(sdev) && sof_ops(sdev)->get_dai_frame_counter) + return sof_ops(sdev)->get_dai_frame_counter(sdev, component, + substream); + + return 0; +} + +static inline u64 +snd_sof_pcm_get_host_byte_counter(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + if (sof_ops(sdev) && sof_ops(sdev)->get_host_byte_counter) + return sof_ops(sdev)->get_host_byte_counter(sdev, component, + substream); return 0; } diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 33d576b17647..f03cee94bce6 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -388,13 +388,21 @@ static snd_pcm_uframes_t sof_pcm_pointer(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + const struct sof_ipc_pcm_ops *pcm_ops = sof_ipc_get_ops(sdev, pcm); struct snd_sof_pcm *spcm; snd_pcm_uframes_t host, dai; + int ret = -EOPNOTSUPP; /* nothing to do for BE */ if (rtd->dai_link->no_pcm) return 0; + if (pcm_ops && pcm_ops->pointer) + ret = pcm_ops->pointer(component, substream, &host); + + if (ret != -EOPNOTSUPP) + return ret ? ret : host; + /* use dsp ops pointer callback directly if set */ if (sof_ops(sdev)->pcm_pointer) return sof_ops(sdev)->pcm_pointer(sdev, substream); diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 9ea2ac5adac7..86bbb531e142 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -103,7 +103,10 @@ struct snd_sof_dai_config_data { * additional memory in the SOF PCM stream structure * @pcm_free: Function pointer for PCM free that can be used for freeing any * additional memory in the SOF PCM stream structure - * @delay: Function pointer for pcm delay calculation + * @pointer: Function pointer for pcm pointer + * Note: the @pointer callback may return -EOPNOTSUPP which should be + * handled in a same way as if the callback is not provided + * @delay: Function pointer for pcm delay reporting * @reset_hw_params_during_stop: Flag indicating whether the hw_params should be reset during the * STOP pcm trigger * @ipc_first_on_start: Send IPC before invoking platform trigger during @@ -124,6 +127,9 @@ struct sof_ipc_pcm_ops { int (*dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params); int (*pcm_setup)(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm); void (*pcm_free)(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm); + int (*pointer)(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + snd_pcm_uframes_t *pointer); snd_pcm_sframes_t (*delay)(struct snd_soc_component *component, struct snd_pcm_substream *substream); bool reset_hw_params_during_stop; @@ -322,6 +328,7 @@ struct snd_sof_pcm_stream { struct work_struct period_elapsed_work; struct snd_soc_dapm_widget_list *list; /* list of connected DAPM widgets */ bool d0i3_compatible; /* DSP can be in D0I3 when this pcm is opened */ + unsigned int dsp_max_burst_size_in_ms; /* The maximum size of the host DMA burst in ms */ /* * flag to indicate that the DSP pipelines should be kept * active or not while suspending the stream diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index d453a4ce3b21..d3c436f82604 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -262,13 +262,25 @@ struct snd_sof_dsp_ops { int (*pcm_ack)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); /* optional */ /* - * optional callback to retrieve the link DMA position for the substream - * when the position is not reported in the shared SRAM windows but - * instead from a host-accessible hardware counter. + * optional callback to retrieve the number of frames left/arrived from/to + * the DSP on the DAI side (link/codec/DMIC/etc). + * + * The callback is used when the firmware does not provide this information + * via the shared SRAM window and it can be retrieved by host. */ - u64 (*get_stream_position)(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream); /* optional */ + u64 (*get_dai_frame_counter)(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); /* optional */ + + /* + * Optional callback to retrieve the number of bytes left/arrived from/to + * the DSP on the host side (bytes between host ALSA buffer and DSP). + * + * The callback is needed for ALSA delay reporting. + */ + u64 (*get_host_byte_counter)(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); /* optional */ /* host read DSP stream data */ int (*ipc_msg_data)(struct snd_sof_dev *sdev, diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index b67617b68e50..f4437015d43a 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -202,7 +202,7 @@ int line6_send_raw_message_async(struct usb_line6 *line6, const char *buffer, struct urb *urb; /* create message: */ - msg = kmalloc(sizeof(struct message), GFP_ATOMIC); + msg = kzalloc(sizeof(struct message), GFP_ATOMIC); if (msg == NULL) return -ENOMEM; @@ -688,7 +688,7 @@ static int line6_init_cap_control(struct usb_line6 *line6) int ret; /* initialize USB buffers: */ - line6->buffer_listen = kmalloc(LINE6_BUFSIZE_LISTEN, GFP_KERNEL); + line6->buffer_listen = kzalloc(LINE6_BUFSIZE_LISTEN, GFP_KERNEL); if (!line6->buffer_listen) return -ENOMEM; @@ -697,7 +697,7 @@ static int line6_init_cap_control(struct usb_line6 *line6) return -ENOMEM; if (line6->properties->capabilities & LINE6_CAP_CONTROL_MIDI) { - line6->buffer_message = kmalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL); + line6->buffer_message = kzalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL); if (!line6->buffer_message) return -ENOMEM; |