diff options
Diffstat (limited to 'sound')
47 files changed, 623 insertions, 174 deletions
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 5d99436dfcae..0cda05c72f50 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -12,9 +12,11 @@ MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>"); MODULE_LICENSE("GPL v2"); #define OUI_WEISS 0x001c6a +#define OUI_LOUD 0x000ff2 #define DICE_CATEGORY_ID 0x04 #define WEISS_CATEGORY_ID 0x00 +#define LOUD_CATEGORY_ID 0x10 static int dice_interface_check(struct fw_unit *unit) { @@ -57,6 +59,8 @@ static int dice_interface_check(struct fw_unit *unit) } if (vendor == OUI_WEISS) category = WEISS_CATEGORY_ID; + else if (vendor == OUI_LOUD) + category = LOUD_CATEGORY_ID; else category = DICE_CATEGORY_ID; if (device->config_rom[3] != ((vendor << 8) | category) || diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8a7fbdcb4072..3b3658297070 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -312,6 +312,10 @@ enum { (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\ AZX_DCAPS_I915_POWERWELL) +#define AZX_DCAPS_INTEL_BROXTON \ + (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\ + AZX_DCAPS_I915_POWERWELL) + /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\ @@ -351,6 +355,8 @@ enum { ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c)) +#define IS_BROXTON(pci) ((pci)->device == 0x5a98) + static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", [AZX_DRIVER_PCH] = "HDA Intel PCH", @@ -502,15 +508,36 @@ static void azx_init_pci(struct azx *chip) } } +/* + * In BXT-P A0, HD-Audio DMA requests is later than expected, + * and makes an audio stream sensitive to system latencies when + * 24/32 bits are playing. + * Adjusting threshold of DMA fifo to force the DMA request + * sooner to improve latency tolerance at the expense of power. + */ +static void bxt_reduce_dma_latency(struct azx *chip) +{ + u32 val; + + val = azx_readl(chip, SKL_EM4L); + val &= (0x3 << 20); + azx_writel(chip, SKL_EM4L, val); +} + static void hda_intel_init_chip(struct azx *chip, bool full_reset) { struct hdac_bus *bus = azx_bus(chip); + struct pci_dev *pci = chip->pci; if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) snd_hdac_set_codec_wakeup(bus, true); azx_init_chip(chip, full_reset); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) snd_hdac_set_codec_wakeup(bus, false); + + /* reduce dma latency to avoid noise */ + if (IS_BROXTON(pci)) + bxt_reduce_dma_latency(chip); } /* calculate runtime delay from LPIB */ @@ -927,6 +954,36 @@ static int azx_resume(struct device *dev) } #endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */ +#ifdef CONFIG_PM_SLEEP +/* put codec down to D3 at hibernation for Intel SKL+; + * otherwise BIOS may still access the codec and screw up the driver + */ +#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170) +#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70) +#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) +#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) + +static int azx_freeze_noirq(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + + if (IS_SKL_PLUS(pci)) + pci_set_power_state(pci, PCI_D3hot); + + return 0; +} + +static int azx_thaw_noirq(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + + if (IS_SKL_PLUS(pci)) + pci_set_power_state(pci, PCI_D0); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + #ifdef CONFIG_PM static int azx_runtime_suspend(struct device *dev) { @@ -1036,6 +1093,10 @@ static int azx_runtime_idle(struct device *dev) static const struct dev_pm_ops azx_pm = { SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) +#ifdef CONFIG_PM_SLEEP + .freeze_noirq = azx_freeze_noirq, + .thaw_noirq = azx_thaw_noirq, +#endif SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, azx_runtime_idle) }; @@ -2124,6 +2185,9 @@ static const struct pci_device_id azx_ids[] = { /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, + /* Broxton-P(Apollolake) */ + { PCI_DEVICE(0x8086, 0x5a98), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index f8a12ca477f1..4ef2259f88ca 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -778,7 +778,8 @@ static const struct hda_pintbl alienware_pincfgs[] = { }; static const struct snd_pci_quirk ca0132_quirks[] = { - SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15", QUIRK_ALIENWARE), + SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE), + SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE), {} }; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c8b8ef5246a6..ef198903c0c3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -955,6 +955,7 @@ static int patch_conexant_auto(struct hda_codec *codec) */ static const struct hda_device_id snd_hda_id_conexant[] = { + HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto), @@ -972,9 +973,9 @@ static const struct hda_device_id snd_hda_id_conexant[] = { HDA_CODEC_ENTRY(0x14f150ac, "CX20652", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f150b8, "CX20664", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f150b9, "CX20665", patch_conexant_auto), - HDA_CODEC_ENTRY(0x14f150f1, "CX20721", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150f1, "CX21722", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f150f2, "CX20722", patch_conexant_auto), - HDA_CODEC_ENTRY(0x14f150f3, "CX20723", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150f3, "CX21724", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f150f4, "CX20724", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f1510f, "CX20751/2", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15110, "CX20751/2", patch_conexant_auto), diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 60cd9e700909..4b6fb668c91c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2352,6 +2352,12 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) struct hda_codec *codec = audio_ptr; int pin_nid = port + 0x04; + /* skip notification during system suspend (but not in runtime PM); + * the state will be updated at resume + */ + if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + return; + check_presence_and_report(codec, pin_nid); } @@ -2378,7 +2384,8 @@ static int patch_generic_hdmi(struct hda_codec *codec) * can cover the codec power request, and so need not set this flag. * For previous platforms, there is no such power well feature. */ - if (is_valleyview_plus(codec) || is_skylake(codec)) + if (is_valleyview_plus(codec) || is_skylake(codec) || + is_broxton(codec)) codec->core.link_power_control = 1; if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f7b065f9ac4..fe96428aa403 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -111,6 +111,7 @@ struct alc_spec { void (*power_hook)(struct hda_codec *codec); #endif void (*shutup)(struct hda_codec *codec); + void (*reboot_notify)(struct hda_codec *codec); int init_amp; int codec_variant; /* flag for other variants */ @@ -773,6 +774,25 @@ static inline void alc_shutup(struct hda_codec *codec) snd_hda_shutup_pins(codec); } +static void alc_reboot_notify(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (spec && spec->reboot_notify) + spec->reboot_notify(codec); + else + alc_shutup(codec); +} + +/* power down codec to D3 at reboot/shutdown; set as reboot_notify ops */ +static void alc_d3_at_reboot(struct hda_codec *codec) +{ + snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + msleep(10); +} + #define alc_free snd_hda_gen_free #ifdef CONFIG_PM @@ -818,7 +838,7 @@ static const struct hda_codec_ops alc_patch_ops = { .suspend = alc_suspend, .check_power_status = snd_hda_gen_check_power_status, #endif - .reboot_notify = alc_shutup, + .reboot_notify = alc_reboot_notify, }; @@ -1755,10 +1775,12 @@ enum { ALC889_FIXUP_MBA11_VREF, ALC889_FIXUP_MBA21_VREF, ALC889_FIXUP_MP11_VREF, + ALC889_FIXUP_MP41_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, ALC887_FIXUP_BASS_CHMAP, + ALC882_FIXUP_DISABLE_AAMIX, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -1842,7 +1864,7 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - static hda_nid_t nids[2] = { 0x14, 0x15 }; + static hda_nid_t nids[3] = { 0x14, 0x15, 0x19 }; int i; if (action != HDA_FIXUP_ACT_INIT) @@ -1920,6 +1942,8 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec, static void alc_fixup_bass_chmap(struct hda_codec *codec, const struct hda_fixup *fix, int action); +static void alc_fixup_disable_aamix(struct hda_codec *codec, + const struct hda_fixup *fix, int action); static const struct hda_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { @@ -2130,6 +2154,12 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC885_FIXUP_MACPRO_GPIO, }, + [ALC889_FIXUP_MP41_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mbp_vref, + .chained = true, + .chain_id = ALC885_FIXUP_MACPRO_GPIO, + }, [ALC882_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic, @@ -2151,6 +2181,10 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, }, + [ALC882_FIXUP_DISABLE_AAMIX] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2208,7 +2242,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), - SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 4,1/5,1", ALC889_FIXUP_MP41_VREF), SND_PCI_QUIRK(0x106b, 0x4300, "iMac 9,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), @@ -2218,6 +2252,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), + SND_PCI_QUIRK(0x1458, 0xa182, "Gigabyte Z170X-UD3", ALC882_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), @@ -4190,6 +4225,8 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->shutup = alc_no_shutup; /* reduce click noise */ + spec->reboot_notify = alc_d3_at_reboot; /* reduce noise */ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; codec->power_save_node = 0; /* avoid click noises */ snd_hda_apply_pincfgs(codec, pincfgs); @@ -4570,6 +4607,7 @@ enum { ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, + ALC292_FIXUP_TPT440, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, @@ -4585,8 +4623,11 @@ enum { ALC288_FIXUP_DISABLE_AAMIX, ALC292_FIXUP_DELL_E7X, ALC292_FIXUP_DISABLE_AAMIX, + ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, + ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, + ALC293_FIXUP_LENOVO_SPK_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5041,6 +5082,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC292_FIXUP_TPT440] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC292_FIXUP_TPT440_DOCK, + }, [ALC283_FIXUP_BXBT2807_MIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5140,6 +5187,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE }, + [ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE + }, [ALC292_FIXUP_DELL_E7X] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_dell_xps13, @@ -5167,6 +5220,23 @@ static const struct hda_fixup alc269_fixups[] = { {} } }, + [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Disable pass-through path for FRONT 14h */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x36}, + {0x20, AC_VERB_SET_PROC_COEF, 0x1737}, + {} + }, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, + [ALC293_FIXUP_LENOVO_SPK_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5180,8 +5250,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), + SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), + SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x05ca, "Dell Latitude E7240", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x05cb, "Dell Latitude E7440", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x05da, "Dell Vostro 5460", ALC290_FIXUP_SUBWOOFER), @@ -5199,11 +5271,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), @@ -5302,15 +5375,17 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440), SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2211, "Thinkpad W541", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -5320,6 +5395,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), @@ -5400,6 +5476,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"}, {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, + {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, {} }; @@ -6386,6 +6463,7 @@ static const struct hda_fixup alc662_fixups[] = { static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x1025, 0x022f, "Acer Aspire One", ALC662_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x0241, "Packard Bell DOTS", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 826122d8acee..2c7c5eb8b1e9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3110,6 +3110,29 @@ static void stac92hd71bxx_fixup_hp_hdx(struct hda_codec *codec, spec->gpio_led = 0x08; } +static bool is_hp_output(struct hda_codec *codec, hda_nid_t pin) +{ + unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, pin); + + /* count line-out, too, as BIOS sets often so */ + return get_defcfg_connect(pin_cfg) != AC_JACK_PORT_NONE && + (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT || + get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT); +} + +static void fixup_hp_headphone(struct hda_codec *codec, hda_nid_t pin) +{ + unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, pin); + + /* It was changed in the BIOS to just satisfy MS DTM. + * Lets turn it back into slaved HP + */ + pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE)) | + (AC_JACK_HP_OUT << AC_DEFCFG_DEVICE_SHIFT); + pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC | AC_DEFCFG_SEQUENCE))) | + 0x1f; + snd_hda_codec_set_pincfg(codec, pin, pin_cfg); +} static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -3119,22 +3142,12 @@ static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, if (action != HDA_FIXUP_ACT_PRE_PROBE) return; - if (hp_blike_system(codec->core.subsystem_id)) { - unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f); - if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT || - get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER || - get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT) { - /* It was changed in the BIOS to just satisfy MS DTM. - * Lets turn it back into slaved HP - */ - pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE)) - | (AC_JACK_HP_OUT << - AC_DEFCFG_DEVICE_SHIFT); - pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC - | AC_DEFCFG_SEQUENCE))) - | 0x1f; - snd_hda_codec_set_pincfg(codec, 0x0f, pin_cfg); - } + /* when both output A and F are assigned, these are supposedly + * dock and built-in headphones; fix both pin configs + */ + if (is_hp_output(codec, 0x0a) && is_hp_output(codec, 0x0f)) { + fixup_hp_headphone(codec, 0x0a); + fixup_hp_headphone(codec, 0x0f); } if (find_mute_led_cfg(codec, 1)) diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 714df906249e..41c31db65039 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -741,10 +741,11 @@ snd_rme96_playback_setrate(struct rme96 *rme96, { /* change to/from double-speed: reset the DAC (if available) */ snd_rme96_reset_dac(rme96); + return 1; /* need to restore volume */ } else { writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER); + return 0; } - return 0; } static int @@ -980,6 +981,7 @@ snd_rme96_playback_hw_params(struct snd_pcm_substream *substream, struct rme96 *rme96 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; int err, rate, dummy; + bool apply_dac_volume = false; runtime->dma_area = (void __force *)(rme96->iobase + RME96_IO_PLAY_BUFFER); @@ -993,24 +995,26 @@ snd_rme96_playback_hw_params(struct snd_pcm_substream *substream, { /* slave clock */ if ((int)params_rate(params) != rate) { - spin_unlock_irq(&rme96->lock); - return -EIO; - } - } else if ((err = snd_rme96_playback_setrate(rme96, params_rate(params))) < 0) { - spin_unlock_irq(&rme96->lock); - return err; - } - if ((err = snd_rme96_playback_setformat(rme96, params_format(params))) < 0) { - spin_unlock_irq(&rme96->lock); - return err; + err = -EIO; + goto error; + } + } else { + err = snd_rme96_playback_setrate(rme96, params_rate(params)); + if (err < 0) + goto error; + apply_dac_volume = err > 0; /* need to restore volume later? */ } + + err = snd_rme96_playback_setformat(rme96, params_format(params)); + if (err < 0) + goto error; snd_rme96_setframelog(rme96, params_channels(params), 1); if (rme96->capture_periodsize != 0) { if (params_period_size(params) << rme96->playback_frlog != rme96->capture_periodsize) { - spin_unlock_irq(&rme96->lock); - return -EBUSY; + err = -EBUSY; + goto error; } } rme96->playback_periodsize = @@ -1021,9 +1025,16 @@ snd_rme96_playback_hw_params(struct snd_pcm_substream *substream, rme96->wcreg &= ~(RME96_WCR_PRO | RME96_WCR_DOLBY | RME96_WCR_EMP); writel(rme96->wcreg |= rme96->wcreg_spdif_stream, rme96->iobase + RME96_IO_CONTROL_REGISTER); } + + err = 0; + error: spin_unlock_irq(&rme96->lock); - - return 0; + if (apply_dac_volume) { + usleep_range(3000, 10000); + snd_rme96_apply_dac_volume(rme96); + } + + return err; } static int diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5070a976adaa..93b400800905 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1023,24 +1023,18 @@ void arizona_init_dvfs(struct arizona_priv *priv) } EXPORT_SYMBOL_GPL(arizona_init_dvfs); -static unsigned int arizona_sysclk_48k_rates[] = { +static unsigned int arizona_opclk_ref_48k_rates[] = { 6144000, 12288000, 24576000, 49152000, - 73728000, - 98304000, - 147456000, }; -static unsigned int arizona_sysclk_44k1_rates[] = { +static unsigned int arizona_opclk_ref_44k1_rates[] = { 5644800, 11289600, 22579200, 45158400, - 67737600, - 90316800, - 135475200, }; static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk, @@ -1065,11 +1059,11 @@ static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk, } if (refclk % 8000) - rates = arizona_sysclk_44k1_rates; + rates = arizona_opclk_ref_44k1_rates; else - rates = arizona_sysclk_48k_rates; + rates = arizona_opclk_ref_48k_rates; - for (ref = 0; ref < ARRAY_SIZE(arizona_sysclk_48k_rates) && + for (ref = 0; ref < ARRAY_SIZE(arizona_opclk_ref_48k_rates) && rates[ref] <= refclk; ref++) { div = 1; while (rates[ref] / div >= freq && div < 32) { diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 969e337dc17c..afa6c5db9dcc 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -85,7 +85,15 @@ static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 300, 0); -static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; +static const struct { + int rate; + unsigned int val; +} deemph_settings[] = { + { 0, ES8328_DACCONTROL6_DEEMPH_OFF }, + { 32000, ES8328_DACCONTROL6_DEEMPH_32k }, + { 44100, ES8328_DACCONTROL6_DEEMPH_44_1k }, + { 48000, ES8328_DACCONTROL6_DEEMPH_48k }, +}; static int es8328_set_deemph(struct snd_soc_codec *codec) { @@ -97,21 +105,22 @@ static int es8328_set_deemph(struct snd_soc_codec *codec) * rate. */ if (es8328->deemph) { - best = 1; - for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { - if (abs(deemph_settings[i] - es8328->playback_fs) < - abs(deemph_settings[best] - es8328->playback_fs)) + best = 0; + for (i = 1; i < ARRAY_SIZE(deemph_settings); i++) { + if (abs(deemph_settings[i].rate - es8328->playback_fs) < + abs(deemph_settings[best].rate - es8328->playback_fs)) best = i; } - val = best << 1; + val = deemph_settings[best].val; } else { - val = 0; + val = ES8328_DACCONTROL6_DEEMPH_OFF; } dev_dbg(codec->dev, "Set deemphasis %d\n", val); - return snd_soc_update_bits(codec, ES8328_DACCONTROL6, 0x6, val); + return snd_soc_update_bits(codec, ES8328_DACCONTROL6, + ES8328_DACCONTROL6_DEEMPH_MASK, val); } static int es8328_get_deemph(struct snd_kcontrol *kcontrol, @@ -205,18 +214,18 @@ static const struct snd_kcontrol_new es8328_right_line_controls = /* Left Mixer */ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { - SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 8, 1, 0), - SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 7, 1, 0), - SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 8, 1, 0), - SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 7, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 7, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 6, 1, 0), + SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 7, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 6, 1, 0), }; /* Right Mixer */ static const struct snd_kcontrol_new es8328_right_mixer_controls[] = { - SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 8, 1, 0), - SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 7, 1, 0), - SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 8, 1, 0), - SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 7, 1, 0), + SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 7, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 6, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 7, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 6, 1, 0), }; static const char * const es8328_pga_sel[] = { diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h index cb36afe10c0e..156c748c89c7 100644 --- a/sound/soc/codecs/es8328.h +++ b/sound/soc/codecs/es8328.h @@ -153,6 +153,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_DACCONTROL6_CLICKFREE (1 << 3) #define ES8328_DACCONTROL6_DAC_INVR (1 << 4) #define ES8328_DACCONTROL6_DAC_INVL (1 << 5) +#define ES8328_DACCONTROL6_DEEMPH_MASK (3 << 6) #define ES8328_DACCONTROL6_DEEMPH_OFF (0 << 6) #define ES8328_DACCONTROL6_DEEMPH_32k (1 << 6) #define ES8328_DACCONTROL6_DEEMPH_44_1k (2 << 6) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 7fc7b4e3f444..c1b87c5800b1 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1271,6 +1271,36 @@ static int nau8825_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_SLEEP +static int nau8825_suspend(struct device *dev) +{ + struct i2c_client *client = to_i2c_client(dev); + struct nau8825 *nau8825 = dev_get_drvdata(dev); + + disable_irq(client->irq); + regcache_cache_only(nau8825->regmap, true); + regcache_mark_dirty(nau8825->regmap); + + return 0; +} + +static int nau8825_resume(struct device *dev) +{ + struct i2c_client *client = to_i2c_client(dev); + struct nau8825 *nau8825 = dev_get_drvdata(dev); + + regcache_cache_only(nau8825->regmap, false); + regcache_sync(nau8825->regmap); + enable_irq(client->irq); + + return 0; +} +#endif + +static const struct dev_pm_ops nau8825_pm = { + SET_SYSTEM_SLEEP_PM_OPS(nau8825_suspend, nau8825_resume) +}; + static const struct i2c_device_id nau8825_i2c_ids[] = { { "nau8825", 0 }, { } @@ -1297,6 +1327,7 @@ static struct i2c_driver nau8825_driver = { .name = "nau8825", .of_match_table = of_match_ptr(nau8825_of_ids), .acpi_match_table = ACPI_PTR(nau8825_acpi_match), + .pm = &nau8825_pm, }, .probe = nau8825_i2c_probe, .remove = nau8825_i2c_remove, diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index aca479fa7670..1dc68ab08a17 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -80,8 +80,10 @@ int rl6231_calc_dmic_clk(int rate) } for (i = 0; i < ARRAY_SIZE(div); i++) { - /* find divider that gives DMIC frequency below 3MHz */ - if (3000000 * div[i] >= rate) + if ((div[i] % 3) == 0) + continue; + /* find divider that gives DMIC frequency below 3.072MHz */ + if (3072000 * div[i] >= rate) return i; } diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 28132375e427..3e3c7f6be29d 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -245,7 +245,7 @@ struct rt5645_priv { struct snd_soc_jack *hp_jack; struct snd_soc_jack *mic_jack; struct snd_soc_jack *btn_jack; - struct delayed_work jack_detect_work; + struct delayed_work jack_detect_work, rcclock_work; struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; struct rt5645_eq_param_s *eq_param; @@ -565,12 +565,33 @@ static int rt5645_hweq_put(struct snd_kcontrol *kcontrol, .put = rt5645_hweq_put \ } +static int rt5645_spk_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + int ret; + + cancel_delayed_work_sync(&rt5645->rcclock_work); + + regmap_update_bits(rt5645->regmap, RT5645_MICBIAS, + RT5645_PWR_CLK25M_MASK, RT5645_PWR_CLK25M_PU); + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + + queue_delayed_work(system_power_efficient_wq, &rt5645->rcclock_work, + msecs_to_jiffies(200)); + + return ret; +} + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1), - SOC_DOUBLE_TLV("Speaker Playback Volume", RT5645_SPK_VOL, - RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), + SOC_DOUBLE_EXT_TLV("Speaker Playback Volume", RT5645_SPK_VOL, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, snd_soc_get_volsw, + rt5645_spk_put_volsw, out_vol_tlv), /* ClassD modulator Speaker Gain Ratio */ SOC_SINGLE_TLV("Speaker ClassD Playback Volume", RT5645_SPO_CLSD_RATIO, @@ -1498,7 +1519,7 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on) regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_write(codec, RT5645_DEPOP_M2, 0x1140); - msleep(40); + msleep(70); rt5645->hp_on = true; } else { /* depop parameters */ @@ -1646,9 +1667,13 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, RT5645_PWR_CLS_D_L, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L); + snd_soc_update_bits(codec, RT5645_GEN_CTRL3, + RT5645_DET_CLK_MASK, RT5645_DET_CLK_MODE1); break; case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5645_GEN_CTRL3, + RT5645_DET_CLK_MASK, RT5645_DET_CLK_DIS); snd_soc_write(codec, RT5645_EQ_CTRL2, 0); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | @@ -3122,6 +3147,15 @@ static void rt5645_jack_detect_work(struct work_struct *work) SND_JACK_BTN_2 | SND_JACK_BTN_3); } +static void rt5645_rcclock_work(struct work_struct *work) +{ + struct rt5645_priv *rt5645 = + container_of(work, struct rt5645_priv, rcclock_work.work); + + regmap_update_bits(rt5645->regmap, RT5645_MICBIAS, + RT5645_PWR_CLK25M_MASK, RT5645_PWR_CLK25M_PD); +} + static irqreturn_t rt5645_irq(int irq, void *data) { struct rt5645_priv *rt5645 = data; @@ -3348,6 +3382,27 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Reks"), }, }, + { + .ident = "Google Edgar", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Edgar"), + }, + }, + { + .ident = "Google Wizpig", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Wizpig"), + }, + }, + { + .ident = "Google Terra", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Terra"), + }, + }, { } }; @@ -3587,6 +3642,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); + INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work); if (rt5645->i2c->irq) { ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, @@ -3621,6 +3677,7 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) free_irq(i2c->irq, rt5645); cancel_delayed_work_sync(&rt5645->jack_detect_work); + cancel_delayed_work_sync(&rt5645->rcclock_work); snd_soc_unregister_codec(&i2c->dev); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 093e46d559fb..205e0715c99a 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2122,6 +2122,10 @@ enum { /* General Control3 (0xfc) */ #define RT5645_JD_PSV_MODE (0x1 << 12) #define RT5645_IRQ_CLK_GATE_CTRL (0x1 << 11) +#define RT5645_DET_CLK_MASK (0x3 << 9) +#define RT5645_DET_CLK_DIS (0x0 << 9) +#define RT5645_DET_CLK_MODE1 (0x1 << 9) +#define RT5645_DET_CLK_MODE2 (0x2 << 9) #define RT5645_MICINDET_MANU (0x1 << 7) #define RT5645_RING2_SLEEVE_GND (0x1 << 5) diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index dc2b46236c5c..3f1b0f1df809 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -973,12 +973,12 @@ #define RT5670_SCLK_SRC_MCLK (0x0 << 14) #define RT5670_SCLK_SRC_PLL1 (0x1 << 14) #define RT5670_SCLK_SRC_RCCLK (0x2 << 14) /* 15MHz */ -#define RT5670_PLL1_SRC_MASK (0x3 << 12) -#define RT5670_PLL1_SRC_SFT 12 -#define RT5670_PLL1_SRC_MCLK (0x0 << 12) -#define RT5670_PLL1_SRC_BCLK1 (0x1 << 12) -#define RT5670_PLL1_SRC_BCLK2 (0x2 << 12) -#define RT5670_PLL1_SRC_BCLK3 (0x3 << 12) +#define RT5670_PLL1_SRC_MASK (0x7 << 11) +#define RT5670_PLL1_SRC_SFT 11 +#define RT5670_PLL1_SRC_MCLK (0x0 << 11) +#define RT5670_PLL1_SRC_BCLK1 (0x1 << 11) +#define RT5670_PLL1_SRC_BCLK2 (0x2 << 11) +#define RT5670_PLL1_SRC_BCLK3 (0x3 << 11) #define RT5670_PLL1_PD_MASK (0x1 << 3) #define RT5670_PLL1_PD_SFT 3 #define RT5670_PLL1_PD_1 (0x0 << 3) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index b4cd7e3bf5f8..69d987a9935c 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1386,90 +1386,90 @@ static const struct snd_kcontrol_new rt5677_dac_r_mix[] = { }; static const struct snd_kcontrol_new rt5677_sto1_dac_l_mix[] = { - SOC_DAPM_SINGLE("ST L Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("ST L Switch", RT5677_STO1_DAC_MIXER, RT5677_M_ST_DAC1_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER, RT5677_M_DAC1_L_STO_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_STO1_DAC_MIXER, RT5677_M_DAC2_L_STO_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER, RT5677_M_DAC1_R_STO_L_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5677_sto1_dac_r_mix[] = { - SOC_DAPM_SINGLE("ST R Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("ST R Switch", RT5677_STO1_DAC_MIXER, RT5677_M_ST_DAC1_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER, RT5677_M_DAC1_R_STO_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_STO1_DAC_MIXER, RT5677_M_DAC2_R_STO_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER, RT5677_M_DAC1_L_STO_R_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5677_mono_dac_l_mix[] = { - SOC_DAPM_SINGLE("ST L Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("ST L Switch", RT5677_MONO_DAC_MIXER, RT5677_M_ST_DAC2_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_MONO_DAC_MIXER, RT5677_M_DAC1_L_MONO_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER, RT5677_M_DAC2_L_MONO_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER, RT5677_M_DAC2_R_MONO_L_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5677_mono_dac_r_mix[] = { - SOC_DAPM_SINGLE("ST R Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("ST R Switch", RT5677_MONO_DAC_MIXER, RT5677_M_ST_DAC2_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_MONO_DAC_MIXER, RT5677_M_DAC1_R_MONO_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER, RT5677_M_DAC2_R_MONO_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER, RT5677_M_DAC2_L_MONO_R_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5677_dd1_l_mix[] = { - SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix L Switch", RT5677_DD1_MIXER, RT5677_M_STO_L_DD1_L_SFT, 1, 1), - SOC_DAPM_SINGLE("Mono DAC Mix L Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix L Switch", RT5677_DD1_MIXER, RT5677_M_MONO_L_DD1_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC3 L Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC3 L Switch", RT5677_DD1_MIXER, RT5677_M_DAC3_L_DD1_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC3 R Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC3 R Switch", RT5677_DD1_MIXER, RT5677_M_DAC3_R_DD1_L_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5677_dd1_r_mix[] = { - SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix R Switch", RT5677_DD1_MIXER, RT5677_M_STO_R_DD1_R_SFT, 1, 1), - SOC_DAPM_SINGLE("Mono DAC Mix R Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix R Switch", RT5677_DD1_MIXER, RT5677_M_MONO_R_DD1_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC3 R Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC3 R Switch", RT5677_DD1_MIXER, RT5677_M_DAC3_R_DD1_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC3 L Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC3 L Switch", RT5677_DD1_MIXER, RT5677_M_DAC3_L_DD1_R_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5677_dd2_l_mix[] = { - SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix L Switch", RT5677_DD2_MIXER, RT5677_M_STO_L_DD2_L_SFT, 1, 1), - SOC_DAPM_SINGLE("Mono DAC Mix L Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix L Switch", RT5677_DD2_MIXER, RT5677_M_MONO_L_DD2_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC4 L Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC4 L Switch", RT5677_DD2_MIXER, RT5677_M_DAC4_L_DD2_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC4 R Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC4 R Switch", RT5677_DD2_MIXER, RT5677_M_DAC4_R_DD2_L_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5677_dd2_r_mix[] = { - SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix R Switch", RT5677_DD2_MIXER, RT5677_M_STO_R_DD2_R_SFT, 1, 1), - SOC_DAPM_SINGLE("Mono DAC Mix R Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix R Switch", RT5677_DD2_MIXER, RT5677_M_MONO_R_DD2_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC4 R Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC4 R Switch", RT5677_DD2_MIXER, RT5677_M_DAC4_R_DD2_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC4 L Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC4 L Switch", RT5677_DD2_MIXER, RT5677_M_DAC4_L_DD2_R_SFT, 1, 1), }; @@ -2596,6 +2596,21 @@ static int rt5677_vref_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5677_filter_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(50); + break; + + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT, 0, rt5677_set_pll1_event, SND_SOC_DAPM_PRE_PMU | @@ -3072,19 +3087,26 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* DAC Mixer */ SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5677_PWR_DIG2, - RT5677_PWR_DAC_S1F_BIT, 0, NULL, 0), + RT5677_PWR_DAC_S1F_BIT, 0, rt5677_filter_power_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("dac mono2 left filter", RT5677_PWR_DIG2, - RT5677_PWR_DAC_M2F_L_BIT, 0, NULL, 0), + RT5677_PWR_DAC_M2F_L_BIT, 0, rt5677_filter_power_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("dac mono2 right filter", RT5677_PWR_DIG2, - RT5677_PWR_DAC_M2F_R_BIT, 0, NULL, 0), + RT5677_PWR_DAC_M2F_R_BIT, 0, rt5677_filter_power_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("dac mono3 left filter", RT5677_PWR_DIG2, - RT5677_PWR_DAC_M3F_L_BIT, 0, NULL, 0), + RT5677_PWR_DAC_M3F_L_BIT, 0, rt5677_filter_power_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("dac mono3 right filter", RT5677_PWR_DIG2, - RT5677_PWR_DAC_M3F_R_BIT, 0, NULL, 0), + RT5677_PWR_DAC_M3F_R_BIT, 0, rt5677_filter_power_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("dac mono4 left filter", RT5677_PWR_DIG2, - RT5677_PWR_DAC_M4F_L_BIT, 0, NULL, 0), + RT5677_PWR_DAC_M4F_L_BIT, 0, rt5677_filter_power_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("dac mono4 right filter", RT5677_PWR_DIG2, - RT5677_PWR_DAC_M4F_R_BIT, 0, NULL, 0), + RT5677_PWR_DAC_M4F_R_BIT, 0, rt5677_filter_power_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, rt5677_sto1_dac_l_mix, ARRAY_SIZE(rt5677_sto1_dac_l_mix)), diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f540f82b1f27..08b40460663c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -189,6 +189,7 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); + msleep(400); break; case SND_SOC_DAPM_PRE_PMD: diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 056375339ea3..5380798883b5 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -229,7 +229,7 @@ SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL, SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL, 6, 1, 0), SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume", WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv), diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 39ebd7bf4f53..a7e79784fc16 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -365,8 +365,8 @@ static const struct reg_default wm8962_reg[] = { { 16924, 0x0059 }, /* R16924 - HDBASS_PG_1 */ { 16925, 0x999A }, /* R16925 - HDBASS_PG_0 */ - { 17048, 0x0083 }, /* R17408 - HPF_C_1 */ - { 17049, 0x98AD }, /* R17409 - HPF_C_0 */ + { 17408, 0x0083 }, /* R17408 - HPF_C_1 */ + { 17409, 0x98AD }, /* R17409 - HPF_C_0 */ { 17920, 0x007F }, /* R17920 - ADCL_RETUNE_C1_1 */ { 17921, 0xFFFF }, /* R17921 - ADCL_RETUNE_C1_0 */ diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 0a60677397b3..4c29bd2ae75c 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -574,6 +574,7 @@ static const struct regmap_config wm8974_regmap = { .max_register = WM8974_MONOMIX, .reg_defaults = wm8974_reg_defaults, .num_reg_defaults = ARRAY_SIZE(wm8974_reg_defaults), + .cache_type = REGCACHE_FLAT, }; static int wm8974_probe(struct snd_soc_codec *codec) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 4495a40a9468..2ccb8bccc9d4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -223,8 +223,8 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) /* wait for XDATA to be cleared */ cnt = 0; - while (!(mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) & - ~XRDATA) && (cnt < 100000)) + while ((mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) & XRDATA) && + (cnt < 100000)) cnt++; /* Release TX state machine */ @@ -681,8 +681,8 @@ static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai, } mcasp->tdm_slots = slots; - mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = rx_mask; - mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = tx_mask; + mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask; + mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = rx_mask; mcasp->slot_width = slot_width; return davinci_mcasp_set_ch_constraints(mcasp); @@ -908,6 +908,14 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRMOD(total_slots), FSRMOD(0x1FF)); + /* + * If McASP is set to be TX/RX synchronous and the playback is + * not running already we need to configure the TX slots in + * order to have correct FSX on the bus + */ + if (mcasp_is_synchronous(mcasp) && !mcasp->channels) + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(total_slots), FSXMOD(0x1FF)); } return 0; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 19c302b0d763..14dfdee05fd5 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -283,6 +283,8 @@ config SND_SOC_IMX_MC13783 config SND_SOC_FSL_ASOC_CARD tristate "Generic ASoC Sound Card with ASRC support" depends on OF && I2C + # enforce SND_SOC_FSL_ASOC_CARD=m if SND_AC97_CODEC=m: + depends on SND_AC97_CODEC || SND_AC97_CODEC=n select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_ESAI diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a4435f5e3be9..08b460ba06ef 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -454,7 +454,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx. * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx. */ - regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, + sai->synchronous[TX] ? FSL_SAI_CR2_SYNC : 0); regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0); @@ -504,6 +505,24 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); + + /* + * For sai master mode, after several open/close sai, + * there will be no frame clock, and can't recover + * anymore. Add software reset to fix this issue. + * This is a hardware bug, and will be fix in the + * next sai version. + */ + if (!sai->is_slave_mode) { + /* Software Reset for both Tx and Rx */ + regmap_write(sai->regmap, + FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, + FSL_SAI_RCSR, FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + } } break; default: diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 7b778ab85f8b..d430ef5a4f38 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -144,7 +144,7 @@ config SND_SOC_INTEL_SKYLAKE config SND_SOC_INTEL_SKL_RT286_MACH tristate "ASoC Audio driver for SKL with RT286 I2S mode" - depends on X86 && ACPI + depends on X86 && ACPI && I2C select SND_SOC_INTEL_SST select SND_SOC_INTEL_SKYLAKE select SND_SOC_RT286 diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index a7854c8fc523..ad4d0f82603e 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1248,5 +1248,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus) skl->resource.max_mcps = SKL_MAX_MCPS; skl->resource.max_mem = SKL_FW_MAX_MEM; + skl->tplg = fw; + return 0; } diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 5319529aedf7..caa69c4598a6 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -25,6 +25,7 @@ #include <linux/pci.h> #include <linux/pm_runtime.h> #include <linux/platform_device.h> +#include <linux/firmware.h> #include <sound/pcm.h> #include "skl.h" @@ -520,6 +521,9 @@ static void skl_remove(struct pci_dev *pci) struct hdac_ext_bus *ebus = pci_get_drvdata(pci); struct skl *skl = ebus_to_skl(ebus); + if (skl->tplg) + release_firmware(skl->tplg); + if (pci_dev_run_wake(pci)) pm_runtime_get_noresume(&pci->dev); pci_dev_put(pci); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index dd2e79ae45a8..a0709e344d44 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -68,6 +68,8 @@ struct skl { struct skl_dsp_resource resource; struct list_head ppl_list; struct list_head dapm_path_list; + + const struct firmware *tplg; }; #define skl_to_ebus(s) (&(s)->ebus) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index a38a3029062c..5a806da89f42 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -152,8 +152,10 @@ static int rk_spdif_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ret = regmap_update_bits(spdif->regmap, SPDIF_DMACR, - SPDIF_DMACR_TDE_ENABLE, - SPDIF_DMACR_TDE_ENABLE); + SPDIF_DMACR_TDE_ENABLE | + SPDIF_DMACR_TDL_MASK, + SPDIF_DMACR_TDE_ENABLE | + SPDIF_DMACR_TDL(16)); if (ret != 0) return ret; @@ -280,7 +282,7 @@ static int rk_spdif_probe(struct platform_device *pdev) int ret; match = of_match_node(rk_spdif_match, np); - if ((int) match->data == RK_SPDIF_RK3288) { + if (match->data == (void *)RK_SPDIF_RK3288) { struct regmap *grf; grf = syscon_regmap_lookup_by_phandle(np, "rockchip,grf"); diff --git a/sound/soc/rockchip/rockchip_spdif.h b/sound/soc/rockchip/rockchip_spdif.h index 07f86a21046a..3ef12770ae12 100644 --- a/sound/soc/rockchip/rockchip_spdif.h +++ b/sound/soc/rockchip/rockchip_spdif.h @@ -28,9 +28,9 @@ #define SPDIF_CFGR_VDW(x) (x << SPDIF_CFGR_VDW_SHIFT) #define SDPIF_CFGR_VDW_MASK (0xf << SPDIF_CFGR_VDW_SHIFT) -#define SPDIF_CFGR_VDW_16 SPDIF_CFGR_VDW(0x00) -#define SPDIF_CFGR_VDW_20 SPDIF_CFGR_VDW(0x01) -#define SPDIF_CFGR_VDW_24 SPDIF_CFGR_VDW(0x10) +#define SPDIF_CFGR_VDW_16 SPDIF_CFGR_VDW(0x0) +#define SPDIF_CFGR_VDW_20 SPDIF_CFGR_VDW(0x1) +#define SPDIF_CFGR_VDW_24 SPDIF_CFGR_VDW(0x2) /* * DMACR @@ -42,7 +42,7 @@ #define SPDIF_DMACR_TDL_SHIFT 0 #define SPDIF_DMACR_TDL(x) ((x) << SPDIF_DMACR_TDL_SHIFT) -#define SPDIF_DMACR_TDL_MASK (0x1f << SDPIF_DMACR_TDL_SHIFT) +#define SPDIF_DMACR_TDL_MASK (0x1f << SPDIF_DMACR_TDL_SHIFT) /* * XFER diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 76da7620904c..edcf4cc2e84f 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -235,7 +235,7 @@ static int rsnd_gen2_probe(struct platform_device *pdev, RSND_GEN_S_REG(SCU_SYS_STATUS0, 0x1c8), RSND_GEN_S_REG(SCU_SYS_INT_EN0, 0x1cc), RSND_GEN_S_REG(SCU_SYS_STATUS1, 0x1d0), - RSND_GEN_S_REG(SCU_SYS_INT_EN1, 0x1c4), + RSND_GEN_S_REG(SCU_SYS_INT_EN1, 0x1d4), RSND_GEN_M_REG(SRC_SWRSR, 0x200, 0x40), RSND_GEN_M_REG(SRC_SRCIR, 0x204, 0x40), RSND_GEN_M_REG(SRC_ADINR, 0x214, 0x40), diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 261b50217c48..68b439ed22d7 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -923,6 +923,7 @@ static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod, struct snd_soc_pcm_runtime *rtd) { struct rsnd_dai *rdai = rsnd_io_to_rdai(io); + struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io); struct rsnd_src *src = rsnd_mod_to_src(mod); int ret; @@ -937,6 +938,12 @@ static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod, return 0; /* + * SRC In doesn't work if DVC was enabled + */ + if (dvc && !rsnd_io_is_play(io)) + return 0; + + /* * enable sync convert */ ret = rsnd_kctrl_new_s(mod, io, rtd, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 24b096066a07..a1305f827a98 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -795,12 +795,12 @@ static void soc_resume_deferred(struct work_struct *work) dev_dbg(card->dev, "ASoC: resume work completed\n"); - /* userspace can access us now we are back as we were before */ - snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0); - /* Recheck all endpoints too, their state is affected by suspend */ dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); + + /* userspace can access us now we are back as we were before */ + snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0); } /* powers up audio subsystem after a suspend */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 016eba10b1ec..7d009428934a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2293,6 +2293,12 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) kfree(w); } +void snd_soc_dapm_reset_cache(struct snd_soc_dapm_context *dapm) +{ + dapm->path_sink_cache.widget = NULL; + dapm->path_source_cache.widget = NULL; +} + /* free all dapm widgets and resources */ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) { @@ -2303,6 +2309,7 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) continue; snd_soc_dapm_free_widget(w); } + snd_soc_dapm_reset_cache(dapm); } static struct snd_soc_dapm_widget *dapm_find_widget( diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index ecd38e52285a..2f67ba6d7a8f 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -404,7 +404,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); /** * snd_soc_put_volsw_sx - double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double mixer control that spans 2 registers. * diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 8d7ec80af51b..6963ba20991c 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -531,7 +531,7 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, /* TLV bytes controls need standard kcontrol info handler, * TLV callback and extended put/get handlers. */ - k->info = snd_soc_bytes_info; + k->info = snd_soc_bytes_info_ext; k->tlv.c = snd_soc_bytes_tlv_callback; ext_ops = tplg->bytes_ext_ops; @@ -1805,6 +1805,7 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm, snd_soc_tplg_widget_remove(w); snd_soc_dapm_free_widget(w); } + snd_soc_dapm_reset_cache(dapm); } EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove_all); diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 843f037a317d..5c2bc53f0a9b 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -669,6 +669,7 @@ static int uni_player_startup(struct snd_pcm_substream *substream, { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *player = priv->dai_data.uni; + player->substream = substream; player->clk_adj = 0; @@ -950,6 +951,8 @@ static void uni_player_shutdown(struct snd_pcm_substream *substream, if (player->state != UNIPERIF_STATE_STOPPED) /* Stop the player */ uni_player_stop(player); + + player->substream = NULL; } static int uni_player_parse_dt_clk_glue(struct platform_device *pdev, @@ -989,7 +992,7 @@ static int uni_player_parse_dt(struct platform_device *pdev, if (!info) return -ENOMEM; - if (of_property_read_u32(pnode, "version", &player->ver) || + if (of_property_read_u32(pnode, "st,version", &player->ver) || player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { dev_err(dev, "Unknown uniperipheral version "); return -EINVAL; @@ -998,13 +1001,13 @@ static int uni_player_parse_dt(struct platform_device *pdev, if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) info->underflow_enabled = 1; - if (of_property_read_u32(pnode, "uniperiph-id", &info->id)) { + if (of_property_read_u32(pnode, "st,uniperiph-id", &info->id)) { dev_err(dev, "uniperipheral id not defined"); return -EINVAL; } /* Read the device mode property */ - if (of_property_read_string(pnode, "mode", &mode)) { + if (of_property_read_string(pnode, "st,mode", &mode)) { dev_err(dev, "uniperipheral mode not defined"); return -EINVAL; } diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index f791239a3087..8a0eb2050169 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -316,7 +316,7 @@ static int uni_reader_parse_dt(struct platform_device *pdev, if (!info) return -ENOMEM; - if (of_property_read_u32(node, "version", &reader->ver) || + if (of_property_read_u32(node, "st,version", &reader->ver) || reader->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { dev_err(&pdev->dev, "Unknown uniperipheral version "); return -EINVAL; @@ -346,7 +346,6 @@ int uni_reader_init(struct platform_device *pdev, reader->hw = &uni_reader_pcm_hw; reader->dai_ops = &uni_reader_dai_ops; - dev_err(reader->dev, "%s: enter\n", __func__); ret = uni_reader_parse_dt(pdev, reader); if (ret < 0) { dev_err(reader->dev, "Failed to parse DeviceTree"); diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index bcbf4da168b6..1bb896d78d09 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -2,6 +2,7 @@ * Copyright 2014 Emilio López <emilio@elopez.com.ar> * Copyright 2014 Jon Smirl <jonsmirl@gmail.com> * Copyright 2015 Maxime Ripard <maxime.ripard@free-electrons.com> + * Copyright 2015 Adam Sampson <ats@offog.org> * * Based on the Allwinner SDK driver, released under the GPL. * @@ -404,7 +405,7 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mute = static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1); static const struct snd_kcontrol_new sun4i_codec_widgets[] = { - SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL, + SOC_SINGLE_TLV("Power Amplifier Volume", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0, sun4i_codec_pa_volume_scale), }; @@ -452,12 +453,12 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0), - /* Pre-Amplifier */ - SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL, + /* Power Amplifier */ + SND_SOC_DAPM_MIXER("Power Amplifier", SUN4I_CODEC_ADC_ACTL, SUN4I_CODEC_ADC_ACTL_PA_EN, 0, sun4i_codec_pa_mixer_controls, ARRAY_SIZE(sun4i_codec_pa_mixer_controls)), - SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SWITCH("Power Amplifier Mute", SND_SOC_NOPM, 0, 0, &sun4i_codec_pa_mute), SND_SOC_DAPM_OUTPUT("HP Right"), @@ -480,16 +481,16 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { { "Left Mixer", NULL, "Mixer Enable" }, { "Left Mixer", "Left DAC Playback Switch", "Left DAC" }, - /* Pre-Amplifier Mixer Routes */ - { "Pre-Amplifier", "Mixer Playback Switch", "Left Mixer" }, - { "Pre-Amplifier", "Mixer Playback Switch", "Right Mixer" }, - { "Pre-Amplifier", "DAC Playback Switch", "Left DAC" }, - { "Pre-Amplifier", "DAC Playback Switch", "Right DAC" }, + /* Power Amplifier Routes */ + { "Power Amplifier", "Mixer Playback Switch", "Left Mixer" }, + { "Power Amplifier", "Mixer Playback Switch", "Right Mixer" }, + { "Power Amplifier", "DAC Playback Switch", "Left DAC" }, + { "Power Amplifier", "DAC Playback Switch", "Right DAC" }, - /* PA -> HP path */ - { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" }, - { "HP Right", NULL, "Pre-Amplifier Mute" }, - { "HP Left", NULL, "Pre-Amplifier Mute" }, + /* Headphone Output Routes */ + { "Power Amplifier Mute", "Switch", "Power Amplifier" }, + { "HP Right", NULL, "Power Amplifier Mute" }, + { "HP Left", NULL, "Power Amplifier Mute" }, }; static struct snd_soc_codec_driver sun4i_codec_codec = { diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 7661616f3636..5b4c58c3e2c5 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -174,6 +174,8 @@ struct snd_usb_midi_in_endpoint { u8 running_status_length; } ports[0x10]; u8 seen_f5; + bool in_sysex; + u8 last_cin; u8 error_resubmit; int current_port; }; @@ -468,6 +470,39 @@ static void snd_usbmidi_maudio_broken_running_status_input( } /* + * QinHeng CH345 is buggy: every second packet inside a SysEx has not CIN 4 + * but the previously seen CIN, but still with three data bytes. + */ +static void ch345_broken_sysex_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) +{ + unsigned int i, cin, length; + + for (i = 0; i + 3 < buffer_length; i += 4) { + if (buffer[i] == 0 && i > 0) + break; + cin = buffer[i] & 0x0f; + if (ep->in_sysex && + cin == ep->last_cin && + (buffer[i + 1 + (cin == 0x6)] & 0x80) == 0) + cin = 0x4; +#if 0 + if (buffer[i + 1] == 0x90) { + /* + * Either a corrupted running status or a real note-on + * message; impossible to detect reliably. + */ + } +#endif + length = snd_usbmidi_cin_length[cin]; + snd_usbmidi_input_data(ep, 0, &buffer[i + 1], length); + ep->in_sysex = cin == 0x4; + if (!ep->in_sysex) + ep->last_cin = cin; + } +} + +/* * CME protocol: like the standard protocol, but SysEx commands are sent as a * single USB packet preceded by a 0x0F byte. */ @@ -660,6 +695,12 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = { .output_packet = snd_usbmidi_output_standard_packet, }; +static struct usb_protocol_ops snd_usbmidi_ch345_broken_sysex_ops = { + .input = ch345_broken_sysex_input, + .output = snd_usbmidi_standard_output, + .output_packet = snd_usbmidi_output_standard_packet, +}; + /* * AKAI MPD16 protocol: * @@ -1341,6 +1382,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi, * Various chips declare a packet size larger than 4 bytes, but * do not actually work with larger packets: */ + case USB_ID(0x0a67, 0x5011): /* Medeli DD305 */ case USB_ID(0x0a92, 0x1020): /* ESI M4U */ case USB_ID(0x1430, 0x474b): /* RedOctane GH MIDI INTERFACE */ case USB_ID(0x15ca, 0x0101): /* Textech USB Midi Cable */ @@ -2378,6 +2420,10 @@ int snd_usbmidi_create(struct snd_card *card, err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; + case QUIRK_MIDI_CH345: + umidi->usb_protocol_ops = &snd_usbmidi_ch345_broken_sysex_ops; + err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); + break; default: dev_err(&umidi->dev->dev, "invalid quirk type %d\n", quirk->type); diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index f494dced3c11..4f85757009b3 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1354,6 +1354,8 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, } } + snd_usb_mixer_fu_apply_quirk(state->mixer, cval, unitid, kctl); + range = (cval->max - cval->min) / cval->res; /* * Are there devices with volume range more than 255? I use a bit more diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 6a803eff87f7..ddca6547399b 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -348,13 +348,6 @@ static struct usbmix_name_map bose_companion5_map[] = { { 0 } /* terminator */ }; -/* Dragonfly DAC 1.2, the dB conversion factor is 1 instead of 256 */ -static struct usbmix_dB_map dragonfly_1_2_dB = {0, 5000}; -static struct usbmix_name_map dragonfly_1_2_map[] = { - { 7, NULL, .dB = &dragonfly_1_2_dB }, - { 0 } /* terminator */ -}; - /* * Control map entries */ @@ -470,11 +463,6 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x05a7, 0x1020), .map = bose_companion5_map, }, - { - /* Dragonfly DAC 1.2 */ - .id = USB_ID(0x21b4, 0x0081), - .map = dragonfly_1_2_map, - }, { 0 } /* terminator */ }; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index fe91184ce832..0ce888dceed0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -37,6 +37,7 @@ #include <sound/control.h> #include <sound/hwdep.h> #include <sound/info.h> +#include <sound/tlv.h> #include "usbaudio.h" #include "mixer.h" @@ -1825,3 +1826,39 @@ void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, } } +static void snd_dragonfly_quirk_db_scale(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl) +{ + /* Approximation using 10 ranges based on output measurement on hw v1.2. + * This seems close to the cubic mapping e.g. alsamixer uses. */ + static const DECLARE_TLV_DB_RANGE(scale, + 0, 1, TLV_DB_MINMAX_ITEM(-5300, -4970), + 2, 5, TLV_DB_MINMAX_ITEM(-4710, -4160), + 6, 7, TLV_DB_MINMAX_ITEM(-3884, -3710), + 8, 14, TLV_DB_MINMAX_ITEM(-3443, -2560), + 15, 16, TLV_DB_MINMAX_ITEM(-2475, -2324), + 17, 19, TLV_DB_MINMAX_ITEM(-2228, -2031), + 20, 26, TLV_DB_MINMAX_ITEM(-1910, -1393), + 27, 31, TLV_DB_MINMAX_ITEM(-1322, -1032), + 32, 40, TLV_DB_MINMAX_ITEM(-968, -490), + 41, 50, TLV_DB_MINMAX_ITEM(-441, 0), + ); + + usb_audio_info(mixer->chip, "applying DragonFly dB scale quirk\n"); + kctl->tlv.p = scale; + kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; +} + +void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, + struct usb_mixer_elem_info *cval, int unitid, + struct snd_kcontrol *kctl) +{ + switch (mixer->chip->usb_id) { + case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ + if (unitid == 7 && cval->min == 0 && cval->max == 50) + snd_dragonfly_quirk_db_scale(mixer, kctl); + break; + } +} + diff --git a/sound/usb/mixer_quirks.h b/sound/usb/mixer_quirks.h index bdbfab093816..177c329cd4dd 100644 --- a/sound/usb/mixer_quirks.h +++ b/sound/usb/mixer_quirks.h @@ -9,5 +9,9 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, int unitid); +void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, + struct usb_mixer_elem_info *cval, int unitid, + struct snd_kcontrol *kctl); + #endif /* SND_USB_MIXER_QUIRKS_H */ diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 1a1e2e4df35e..c60a776e815d 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2829,6 +2829,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), .idProduct = 0x1020, }, +/* QinHeng devices */ +{ + USB_DEVICE(0x1a86, 0x752d), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "QinHeng", + .product_name = "CH345", + .ifnum = 1, + .type = QUIRK_MIDI_CH345 + } +}, + /* KeithMcMillen Stringport */ { USB_DEVICE(0x1f38, 0x0001), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5ca80e7d30cd..b6c0c8e3b450 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -538,6 +538,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_CME] = create_any_midi_quirk, [QUIRK_MIDI_AKAI] = create_any_midi_quirk, [QUIRK_MIDI_FTDI] = create_any_midi_quirk, + [QUIRK_MIDI_CH345] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, @@ -1124,6 +1125,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ + case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ return true; } return false; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 15a12715bd05..b665d85555cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -95,6 +95,7 @@ enum quirk_type { QUIRK_MIDI_AKAI, QUIRK_MIDI_US122L, QUIRK_MIDI_FTDI, + QUIRK_MIDI_CH345, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UAXX, |