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authorMark Brown <broonie@kernel.org>2020-10-06 16:19:24 +0100
committerMark Brown <broonie@kernel.org>2020-10-06 16:19:24 +0100
commitfd6b519a30a7179026d22c98d6bf10bb5ca8ca27 (patch)
treee4a6bc4d8548a5b8db7148a18eb257b84e72d48b /sound
parent43499134f50a77844f0503df2c995f43b858f4c3 (diff)
parent856deb866d16e29bd65952e0289066f6078af773 (diff)
Merge tag 'v5.9-rc5' into asoc-5.10
Linux 5.9-rc5
Diffstat (limited to 'sound')
-rw-r--r--sound/core/oss/mulaw.c4
-rw-r--r--sound/firewire/digi00x/digi00x.c5
-rw-r--r--sound/firewire/tascam/tascam.c33
-rw-r--r--sound/hda/hdac_device.c2
-rw-r--r--sound/hda/intel-dsp-config.c10
-rw-r--r--sound/pci/ca0106/ca0106_main.c3
-rw-r--r--sound/pci/hda/hda_intel.c7
-rw-r--r--sound/pci/hda/hda_tegra.c7
-rw-r--r--sound/pci/hda/patch_hdmi.c7
-rw-r--r--sound/pci/hda/patch_realtek.c46
-rw-r--r--sound/ppc/snd_ps3.c4
-rw-r--r--sound/soc/atmel/mchp-i2s-mcc.c2
-rw-r--r--sound/soc/codecs/jz4770.c2
-rw-r--r--sound/soc/codecs/pcm186x.c2
-rw-r--r--sound/soc/fsl/fsl_ssi.c2
-rw-r--r--sound/soc/hisilicon/hi6210-i2s.c4
-rw-r--r--sound/soc/intel/boards/bytcht_es8316.c2
-rw-r--r--sound/soc/intel/boards/bytcr_rt5651.c4
-rw-r--r--sound/soc/intel/skylake/skl-pcm.c2
-rw-r--r--sound/soc/meson/axg-tdm-interface.c10
-rw-r--r--sound/soc/pxa/pxa-ssp.c2
-rw-r--r--sound/soc/rockchip/rockchip_pdm.c6
-rw-r--r--sound/soc/samsung/i2s.c2
-rw-r--r--sound/soc/soc-core.c2
-rw-r--r--sound/soc/soc-topology.c4
-rw-r--r--sound/soc/sof/intel/hda-dai.c4
-rw-r--r--sound/soc/sof/pcm.c4
-rw-r--r--sound/soc/ti/davinci-i2s.c2
-rw-r--r--sound/soc/ti/n810.c2
-rw-r--r--sound/soc/ti/omap-dmic.c4
-rw-r--r--sound/soc/ti/omap-mcpdm.c8
-rw-r--r--sound/soc/ti/rx51.c2
-rw-r--r--sound/soc/zte/zx-i2s.c4
-rw-r--r--sound/soc/zte/zx-spdif.c2
-rw-r--r--sound/usb/pcm.c2
-rw-r--r--sound/usb/quirks-table.h78
-rw-r--r--sound/usb/quirks.c11
-rw-r--r--sound/usb/usbaudio.h1
-rw-r--r--sound/x86/Kconfig2
39 files changed, 229 insertions, 71 deletions
diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c
index 3788906421a7..fe27034f2846 100644
--- a/sound/core/oss/mulaw.c
+++ b/sound/core/oss/mulaw.c
@@ -329,8 +329,8 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug,
snd_BUG();
return -EINVAL;
}
- if (snd_BUG_ON(!snd_pcm_format_linear(format->format)))
- return -ENXIO;
+ if (!snd_pcm_format_linear(format->format))
+ return -EINVAL;
err = snd_pcm_plugin_build(plug, "Mu-Law<->linear conversion",
src_format, dst_format,
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
index c84b913a9fe0..ab8408966ec3 100644
--- a/sound/firewire/digi00x/digi00x.c
+++ b/sound/firewire/digi00x/digi00x.c
@@ -14,6 +14,7 @@ MODULE_LICENSE("GPL v2");
#define VENDOR_DIGIDESIGN 0x00a07e
#define MODEL_CONSOLE 0x000001
#define MODEL_RACK 0x000002
+#define SPEC_VERSION 0x000001
static int name_card(struct snd_dg00x *dg00x)
{
@@ -175,14 +176,18 @@ static const struct ieee1394_device_id snd_dg00x_id_table[] = {
/* Both of 002/003 use the same ID. */
{
.match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_VERSION |
IEEE1394_MATCH_MODEL_ID,
.vendor_id = VENDOR_DIGIDESIGN,
+ .version = SPEC_VERSION,
.model_id = MODEL_CONSOLE,
},
{
.match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_VERSION |
IEEE1394_MATCH_MODEL_ID,
.vendor_id = VENDOR_DIGIDESIGN,
+ .version = SPEC_VERSION,
.model_id = MODEL_RACK,
},
{}
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index 5dac0d9fc58e..75f2edd8e78f 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -39,9 +39,6 @@ static const struct snd_tscm_spec model_specs[] = {
.midi_capture_ports = 2,
.midi_playback_ports = 4,
},
- // This kernel module doesn't support FE-8 because the most of features
- // can be implemented in userspace without any specific support of this
- // module.
};
static int identify_model(struct snd_tscm *tscm)
@@ -211,11 +208,39 @@ static void snd_tscm_remove(struct fw_unit *unit)
}
static const struct ieee1394_device_id snd_tscm_id_table[] = {
+ // Tascam, FW-1884.
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_SPECIFIER_ID |
+ IEEE1394_MATCH_VERSION,
+ .vendor_id = 0x00022e,
+ .specifier_id = 0x00022e,
+ .version = 0x800000,
+ },
+ // Tascam, FE-8 (.version = 0x800001)
+ // This kernel module doesn't support FE-8 because the most of features
+ // can be implemented in userspace without any specific support of this
+ // module.
+ //
+ // .version = 0x800002 is unknown.
+ //
+ // Tascam, FW-1082.
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_SPECIFIER_ID |
+ IEEE1394_MATCH_VERSION,
+ .vendor_id = 0x00022e,
+ .specifier_id = 0x00022e,
+ .version = 0x800003,
+ },
+ // Tascam, FW-1804.
{
.match_flags = IEEE1394_MATCH_VENDOR_ID |
- IEEE1394_MATCH_SPECIFIER_ID,
+ IEEE1394_MATCH_SPECIFIER_ID |
+ IEEE1394_MATCH_VERSION,
.vendor_id = 0x00022e,
.specifier_id = 0x00022e,
+ .version = 0x800004,
},
{}
};
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index 333220f0f8af..3e9e9ac804f6 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -127,6 +127,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_init);
void snd_hdac_device_exit(struct hdac_device *codec)
{
pm_runtime_put_noidle(&codec->dev);
+ /* keep balance of runtime PM child_count in parent device */
+ pm_runtime_set_suspended(&codec->dev);
snd_hdac_bus_remove_device(codec->bus, codec);
kfree(codec->vendor_name);
kfree(codec->chip_name);
diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c
index 99aec7349167..1c5114dedda9 100644
--- a/sound/hda/intel-dsp-config.c
+++ b/sound/hda/intel-dsp-config.c
@@ -54,7 +54,7 @@ static const struct config_entry config_table[] = {
#endif
/*
* Apollolake (Broxton-P)
- * the legacy HDaudio driver is used except on Up Squared (SOF) and
+ * the legacy HDAudio driver is used except on Up Squared (SOF) and
* Chromebooks (SST)
*/
#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE)
@@ -89,7 +89,7 @@ static const struct config_entry config_table[] = {
},
#endif
/*
- * Skylake and Kabylake use legacy HDaudio driver except for Google
+ * Skylake and Kabylake use legacy HDAudio driver except for Google
* Chromebooks (SST)
*/
@@ -135,7 +135,7 @@ static const struct config_entry config_table[] = {
#endif
/*
- * Geminilake uses legacy HDaudio driver except for Google
+ * Geminilake uses legacy HDAudio driver except for Google
* Chromebooks
*/
/* Geminilake */
@@ -157,7 +157,7 @@ static const struct config_entry config_table[] = {
/*
* CoffeeLake, CannonLake, CometLake, IceLake, TigerLake use legacy
- * HDaudio driver except for Google Chromebooks and when DMICs are
+ * HDAudio driver except for Google Chromebooks and when DMICs are
* present. Two cases are required since Coreboot does not expose NHLT
* tables.
*
@@ -391,7 +391,7 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci)
if (pci->class == 0x040300)
return SND_INTEL_DSP_DRIVER_LEGACY;
if (pci->class != 0x040100 && pci->class != 0x040380) {
- dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class);
+ dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDAudio legacy driver\n", pci->class);
return SND_INTEL_DSP_DRIVER_LEGACY;
}
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 70d775ff967e..c189f70c82cb 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -537,7 +537,8 @@ static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id,
else
/* Power down */
chip->spi_dac_reg[reg] |= bit;
- return snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]);
+ if (snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]) != 0)
+ return -ENXIO;
}
return 0;
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e34a4d5d047c..36a9dbc33aa0 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2127,9 +2127,10 @@ static int azx_probe(struct pci_dev *pci,
*/
if (dmic_detect) {
err = snd_intel_dsp_driver_probe(pci);
- if (err != SND_INTEL_DSP_DRIVER_ANY &&
- err != SND_INTEL_DSP_DRIVER_LEGACY)
+ if (err != SND_INTEL_DSP_DRIVER_ANY && err != SND_INTEL_DSP_DRIVER_LEGACY) {
+ dev_dbg(&pci->dev, "HDAudio driver not selected, aborting probe\n");
return -ENODEV;
+ }
} else {
dev_warn(&pci->dev, "dmic_detect option is deprecated, pass snd-intel-dspcfg.dsp_driver=1 option instead\n");
}
@@ -2745,8 +2746,6 @@ static const struct pci_device_id azx_ids[] = {
.driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI },
/* Zhaoxin */
{ PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN },
- /* Loongson */
- { PCI_DEVICE(0x0014, 0x7a07), .driver_data = AZX_DRIVER_GENERIC },
{ 0, }
};
MODULE_DEVICE_TABLE(pci, azx_ids);
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index c94553bcca88..70164d1428d4 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -179,6 +179,10 @@ static int __maybe_unused hda_tegra_runtime_suspend(struct device *dev)
struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip);
if (chip && chip->running) {
+ /* enable controller wake up event */
+ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
+ STATESTS_INT_MASK);
+
azx_stop_chip(chip);
azx_enter_link_reset(chip);
}
@@ -200,6 +204,9 @@ static int __maybe_unused hda_tegra_runtime_resume(struct device *dev)
if (chip && chip->running) {
hda_tegra_init(hda);
azx_init_chip(chip, 1);
+ /* disable controller wake up event*/
+ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
+ ~STATESTS_INT_MASK);
}
return 0;
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index b8c8490e568b..402050088090 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2794,6 +2794,7 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec,
hda_nid_t cvt_nid)
{
if (per_pin) {
+ haswell_verify_D0(codec, per_pin->cvt_nid, per_pin->pin_nid);
snd_hda_set_dev_select(codec, per_pin->pin_nid,
per_pin->dev_id);
intel_verify_pin_cvt_connect(codec, per_pin);
@@ -3734,6 +3735,7 @@ static int tegra_hdmi_build_pcms(struct hda_codec *codec)
static int patch_tegra_hdmi(struct hda_codec *codec)
{
+ struct hdmi_spec *spec;
int err;
err = patch_generic_hdmi(codec);
@@ -3741,6 +3743,10 @@ static int patch_tegra_hdmi(struct hda_codec *codec)
return err;
codec->patch_ops.build_pcms = tegra_hdmi_build_pcms;
+ spec = codec->spec;
+ spec->chmap.ops.chmap_cea_alloc_validate_get_type =
+ nvhdmi_chmap_cea_alloc_validate_get_type;
+ spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate;
return 0;
}
@@ -4263,6 +4269,7 @@ HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI", patch_i915_glk_hdmi),
HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi),
HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi),
HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi),
+HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi),
HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi),
HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi),
HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a1fa983d2a94..c521a1f17096 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2475,6 +2475,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
+ SND_PCI_QUIRK(0x1462, 0x9c37, "MSI X570-A PRO", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
@@ -5867,6 +5868,39 @@ static void alc275_fixup_gpio4_off(struct hda_codec *codec,
}
}
+/* Quirk for Thinkpad X1 7th and 8th Gen
+ * The following fixed routing needed
+ * DAC1 (NID 0x02) -> Speaker (NID 0x14); some eq applied secretly
+ * DAC2 (NID 0x03) -> Bass (NID 0x17) & Headphone (NID 0x21); sharing a DAC
+ * DAC3 (NID 0x06) -> Unused, due to the lack of volume amp
+ */
+static void alc285_fixup_thinkpad_x1_gen7(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ static const hda_nid_t conn[] = { 0x02, 0x03 }; /* exclude 0x06 */
+ static const hda_nid_t preferred_pairs[] = {
+ 0x14, 0x02, 0x17, 0x03, 0x21, 0x03, 0
+ };
+ struct alc_spec *spec = codec->spec;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn);
+ spec->gen.preferred_dacs = preferred_pairs;
+ break;
+ case HDA_FIXUP_ACT_BUILD:
+ /* The generic parser creates somewhat unintuitive volume ctls
+ * with the fixed routing above, and the shared DAC2 may be
+ * confusing for PA.
+ * Rename those to unique names so that PA doesn't touch them
+ * and use only Master volume.
+ */
+ rename_ctl(codec, "Front Playback Volume", "DAC1 Playback Volume");
+ rename_ctl(codec, "Bass Speaker Playback Volume", "DAC2 Playback Volume");
+ break;
+ }
+}
+
static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec,
const struct hda_fixup *fix,
int action)
@@ -6135,6 +6169,7 @@ enum {
ALC289_FIXUP_DUAL_SPK,
ALC294_FIXUP_SPK2_TO_DAC1,
ALC294_FIXUP_ASUS_DUAL_SPK,
+ ALC285_FIXUP_THINKPAD_X1_GEN7,
ALC285_FIXUP_THINKPAD_HEADSET_JACK,
ALC294_FIXUP_ASUS_HPE,
ALC294_FIXUP_ASUS_COEF_1B,
@@ -7280,11 +7315,17 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC294_FIXUP_SPK2_TO_DAC1
},
+ [ALC285_FIXUP_THINKPAD_X1_GEN7] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_thinkpad_x1_gen7,
+ .chained = true,
+ .chain_id = ALC269_FIXUP_THINKPAD_ACPI
+ },
[ALC285_FIXUP_THINKPAD_HEADSET_JACK] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_headset_jack,
.chained = true,
- .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1
+ .chain_id = ALC285_FIXUP_THINKPAD_X1_GEN7
},
[ALC294_FIXUP_ASUS_HPE] = {
.type = HDA_FIXUP_VERBS,
@@ -7695,7 +7736,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
- SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+ SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+ SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index b8161a08f2ca..58bb49fff184 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -227,14 +227,14 @@ static int snd_ps3_program_dma(struct snd_ps3_card_info *card,
switch (filltype) {
case SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL:
silent = 1;
- /* intentionally fall thru */
+ fallthrough;
case SND_PS3_DMA_FILLTYPE_FIRSTFILL:
ch0_kick_event = PS3_AUDIO_KICK_EVENT_ALWAYS;
break;
case SND_PS3_DMA_FILLTYPE_SILENT_RUNNING:
silent = 1;
- /* intentionally fall thru */
+ fallthrough;
case SND_PS3_DMA_FILLTYPE_RUNNING:
ch0_kick_event = PS3_AUDIO_KICK_EVENT_SERIALOUT0_EMPTY;
break;
diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c
index 3cb63886195f..04acc18f2d72 100644
--- a/sound/soc/atmel/mchp-i2s-mcc.c
+++ b/sound/soc/atmel/mchp-i2s-mcc.c
@@ -536,7 +536,7 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream,
/* cpu is BCLK master */
mrb |= MCHP_I2SMCC_MRB_CLKSEL_INT;
set_divs = 1;
- /* fall through */
+ fallthrough;
case SND_SOC_DAIFMT_CBM_CFM:
/* cpu is slave */
mra |= MCHP_I2SMCC_MRA_MODE_SLAVE;
diff --git a/sound/soc/codecs/jz4770.c b/sound/soc/codecs/jz4770.c
index c0a28f06b09a..298689a07168 100644
--- a/sound/soc/codecs/jz4770.c
+++ b/sound/soc/codecs/jz4770.c
@@ -202,7 +202,7 @@ static int jz4770_codec_set_bias_level(struct snd_soc_component *codec,
REG_CR_VIC_SB_SLEEP, REG_CR_VIC_SB_SLEEP);
regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_VIC,
REG_CR_VIC_SB, REG_CR_VIC_SB);
- /* fall-through */
+ fallthrough;
default:
break;
}
diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c
index f0da55901dcb..b8845f45549e 100644
--- a/sound/soc/codecs/pcm186x.c
+++ b/sound/soc/codecs/pcm186x.c
@@ -401,7 +401,7 @@ static int pcm186x_set_fmt(struct snd_soc_dai *dai, unsigned int format)
break;
case SND_SOC_DAIFMT_DSP_A:
priv->tdm_offset += 1;
- /* fall through */
+ fallthrough;
/* DSP_A uses the same basic config as DSP_B
* except we need to shift the TDM output by one BCK cycle
*/
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index d8b9c6547142..404be27c15fe 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -898,7 +898,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt)
"missing baudclk for master mode\n");
return -EINVAL;
}
- /* fall through */
+ fallthrough;
case SND_SOC_DAIFMT_CBM_CFS:
ssi->i2s_net |= SSI_SCR_I2S_MODE_MASTER;
break;
diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c
index fd5dcd6b9f85..907f5f1f7b44 100644
--- a/sound/soc/hisilicon/hi6210-i2s.c
+++ b/sound/soc/hisilicon/hi6210-i2s.c
@@ -261,13 +261,13 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream,
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_U16_LE:
signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
- /* fall through */
+ fallthrough;
case SNDRV_PCM_FORMAT_S16_LE:
bits = HII2S_BITS_16;
break;
case SNDRV_PCM_FORMAT_U24_LE:
signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
- /* fall through */
+ fallthrough;
case SNDRV_PCM_FORMAT_S24_LE:
bits = HII2S_BITS_24;
break;
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index 9c09b44b4d33..7ed869bf1a92 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -572,7 +572,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev)
break;
default:
dev_err(dev, "get speaker GPIO failed: %d\n", ret);
- /* fall through */
+ fallthrough;
case -EPROBE_DEFER:
return ret;
}
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 4e2897596cea..688b5e0a49e3 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -1009,7 +1009,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
default:
dev_err(&pdev->dev, "Failed to get ext-amp-enable GPIO: %d\n",
ret_val);
- /* fall through */
+ fallthrough;
case -EPROBE_DEFER:
put_device(codec_dev);
return ret_val;
@@ -1029,7 +1029,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
default:
dev_err(&pdev->dev, "Failed to get hp-detect GPIO: %d\n",
ret_val);
- /* fall through */
+ fallthrough;
case -EPROBE_DEFER:
put_device(codec_dev);
return ret_val;
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 5dee55e9546b..bbe8d782e0af 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -488,7 +488,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
stream->lpib);
snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
}
- /* fall through */
+ fallthrough;
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index 36df30915378..c8664ab80d45 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -58,17 +58,17 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
switch (slot_width) {
case 0:
slot_width = 32;
- /* Fall-through */
+ fallthrough;
case 32:
fmt |= SNDRV_PCM_FMTBIT_S32_LE;
- /* Fall-through */
+ fallthrough;
case 24:
fmt |= SNDRV_PCM_FMTBIT_S24_LE;
fmt |= SNDRV_PCM_FMTBIT_S20_LE;
- /* Fall-through */
+ fallthrough;
case 16:
fmt |= SNDRV_PCM_FMTBIT_S16_LE;
- /* Fall-through */
+ fallthrough;
case 8:
fmt |= SNDRV_PCM_FMTBIT_S8;
break;
@@ -133,7 +133,7 @@ static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
case SND_SOC_DAIFMT_CBS_CFM:
case SND_SOC_DAIFMT_CBM_CFS:
dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n");
- /* Fall-through */
+ fallthrough;
default:
return -EINVAL;
}
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index d1e09ade0190..c4e7307a4437 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -488,7 +488,7 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv)
case SND_SOC_DAIFMT_DSP_A:
sspsp |= SSPSP_FSRT;
- /* fall through */
+ fallthrough;
case SND_SOC_DAIFMT_DSP_B:
sscr0 |= SSCR0_MOD | SSCR0_PSP;
sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c
index 1707414cfa92..5adb293d0435 100644
--- a/sound/soc/rockchip/rockchip_pdm.c
+++ b/sound/soc/rockchip/rockchip_pdm.c
@@ -229,13 +229,13 @@ static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream,
switch (params_channels(params)) {
case 8:
val |= PDM_PATH3_EN;
- /* fallthrough */
+ fallthrough;
case 6:
val |= PDM_PATH2_EN;
- /* fallthrough */
+ fallthrough;
case 4:
val |= PDM_PATH1_EN;
- /* fallthrough */
+ fallthrough;
case 2:
val |= PDM_PATH0_EN;
break;
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 80ecb5c7fed0..df53d4ea808f 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -733,7 +733,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream,
switch (params_channels(params)) {
case 6:
val |= MOD_DC2_EN;
- /* Fall through */
+ fallthrough;
case 4:
val |= MOD_DC1_EN;
break;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 62370774e3ce..ea3986a46c12 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -616,7 +616,7 @@ int snd_soc_suspend(struct device *dev)
"ASoC: idle_bias_off CODEC on over suspend\n");
break;
}
- /* fall through */
+ fallthrough;
case SND_SOC_BIAS_OFF:
snd_soc_component_suspend(component);
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 63086fa8b861..c5ef432a023b 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1069,7 +1069,7 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count,
ec->hdr.name);
goto err_denum;
}
- /* fall through */
+ fallthrough;
case SND_SOC_TPLG_CTL_ENUM:
case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
@@ -1457,7 +1457,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create(
ec->hdr.name);
goto err_se;
}
- /* fall through */
+ fallthrough;
case SND_SOC_TPLG_CTL_ENUM:
case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index df1c6997cb4e..c6cb8c212eca 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -310,7 +310,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream,
return ret;
}
- /* fallthrough */
+ fallthrough;
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
snd_hdac_ext_link_stream_start(link_dev);
@@ -333,7 +333,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream,
link_dev->link_prepared = 0;
- /* fallthrough */
+ fallthrough;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
snd_hdac_ext_link_stream_clear(link_dev);
break;
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 4c5082b7eea9..cbac6f17c52f 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -361,7 +361,7 @@ static int sof_pcm_trigger(struct snd_soc_component *component,
return ret;
}
- /* fallthrough */
+ fallthrough;
case SNDRV_PCM_TRIGGER_START:
if (spcm->stream[substream->stream].suspend_ignored) {
/*
@@ -386,7 +386,7 @@ static int sof_pcm_trigger(struct snd_soc_component *component,
spcm->stream[substream->stream].suspend_ignored = true;
return 0;
}
- /* fallthrough */
+ fallthrough;
case SNDRV_PCM_TRIGGER_STOP:
stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP;
ipc_first = true;
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index d89b5c928c4d..dd34504c09ba 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -289,7 +289,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
* rate is lowered.
*/
inv_fs = true;
- /* fall through */
+ fallthrough;
case SND_SOC_DAIFMT_DSP_A:
dev->mode = MOD_DSP_A;
break;
diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c
index 2802a33b9c5f..ed217b34f846 100644
--- a/sound/soc/ti/n810.c
+++ b/sound/soc/ti/n810.c
@@ -46,7 +46,7 @@ static void n810_ext_control(struct snd_soc_dapm_context *dapm)
switch (n810_jack_func) {
case N810_JACK_HS:
line1l = 1;
- /* fall through */
+ fallthrough;
case N810_JACK_HP:
hp = 1;
break;
diff --git a/sound/soc/ti/omap-dmic.c b/sound/soc/ti/omap-dmic.c
index 01abf1be5d78..a26588e9c3bc 100644
--- a/sound/soc/ti/omap-dmic.c
+++ b/sound/soc/ti/omap-dmic.c
@@ -203,10 +203,10 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
switch (channels) {
case 6:
dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE;
- /* fall through */
+ fallthrough;
case 4:
dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE;
- /* fall through */
+ fallthrough;
case 2:
dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE;
break;
diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c
index d482b62f314a..fafb2998ad0d 100644
--- a/sound/soc/ti/omap-mcpdm.c
+++ b/sound/soc/ti/omap-mcpdm.c
@@ -309,19 +309,19 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
/* up to 3 channels for capture */
return -EINVAL;
link_mask |= 1 << 4;
- /* fall through */
+ fallthrough;
case 4:
if (stream == SNDRV_PCM_STREAM_CAPTURE)
/* up to 3 channels for capture */
return -EINVAL;
link_mask |= 1 << 3;
- /* fall through */
+ fallthrough;
case 3:
link_mask |= 1 << 2;
- /* fall through */
+ fallthrough;
case 2:
link_mask |= 1 << 1;
- /* fall through */
+ fallthrough;
case 1:
link_mask |= 1 << 0;
break;
diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c
index 2176a95201bf..a2629ccc1dc8 100644
--- a/sound/soc/ti/rx51.c
+++ b/sound/soc/ti/rx51.c
@@ -55,7 +55,7 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
break;
case RX51_JACK_HS:
hs = 1;
- /* fall through */
+ fallthrough;
case RX51_JACK_HP:
hp = 1;
break;
diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c
index 568cde64ff8b..1c1a44e08a67 100644
--- a/sound/soc/zte/zx-i2s.c
+++ b/sound/soc/zte/zx-i2s.c
@@ -294,7 +294,7 @@ static int zx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
zx_i2s_rx_dma_en(zx_i2s->reg_base, true);
else
zx_i2s_tx_dma_en(zx_i2s->reg_base, true);
- /* fall thru */
+ fallthrough;
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
if (capture)
@@ -308,7 +308,7 @@ static int zx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
zx_i2s_rx_dma_en(zx_i2s->reg_base, false);
else
zx_i2s_tx_dma_en(zx_i2s->reg_base, false);
- /* fall thru */
+ fallthrough;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
if (capture)
diff --git a/sound/soc/zte/zx-spdif.c b/sound/soc/zte/zx-spdif.c
index a3a07c0730e6..b4168bd532b7 100644
--- a/sound/soc/zte/zx-spdif.c
+++ b/sound/soc/zte/zx-spdif.c
@@ -218,7 +218,7 @@ static int zx_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
val = readl_relaxed(zx_spdif->reg_base + ZX_FIFOCTRL);
val |= ZX_FIFOCTRL_TX_FIFO_RST;
writel_relaxed(val, zx_spdif->reg_base + ZX_FIFOCTRL);
- /* fall thru */
+ fallthrough;
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
zx_spdif_cfg_tx(zx_spdif->reg_base, true);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 5600751803cf..b401ee894e1b 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -369,11 +369,13 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
case USB_ID(0x07fd, 0x0008): /* MOTU M Series */
case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */
case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */
+ case USB_ID(0x0499, 0x172f): /* Steinberg UR22C */
case USB_ID(0x0d9a, 0x00df): /* RTX6001 */
ep = 0x81;
ifnum = 2;
goto add_sync_ep_from_ifnum;
case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */
+ case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */
ep = 0x82;
ifnum = 0;
goto add_sync_ep_from_ifnum;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index f4fb002e3ef4..23eafd50126f 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2827,14 +2827,24 @@ YAMAHA_DEVICE(0x7010, "UB99"),
/* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */
{
USB_DEVICE(0x17aa, 0x1046),
- QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Rear",
- "Lenovo-ThinkStation-P620-Rear"),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Lenovo",
+ .product_name = "ThinkStation P620 Rear",
+ .profile_name = "Lenovo-ThinkStation-P620-Rear",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_SETUP_DISABLE_AUTOSUSPEND
+ }
},
/* Lenovo ThinkStation P620 Internal Speaker + Front Headset */
{
USB_DEVICE(0x17aa, 0x104d),
- QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Main",
- "Lenovo-ThinkStation-P620-Main"),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Lenovo",
+ .product_name = "ThinkStation P620 Main",
+ .profile_name = "Lenovo-ThinkStation-P620-Main",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_SETUP_DISABLE_AUTOSUSPEND
+ }
},
/* Native Instruments MK2 series */
@@ -3549,14 +3559,40 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
{
/*
* Pioneer DJ DJM-250MK2
- * PCM is 8 channels out @ 48 fixed (endpoints 0x01).
- * The output from computer to the mixer is usable.
+ * PCM is 8 channels out @ 48 fixed (endpoint 0x01)
+ * and 8 channels in @ 48 fixed (endpoint 0x82).
+ *
+ * Both playback and recording is working, even simultaneously.
*
- * The input (phono or line to computer) is not working.
- * It should be at endpoint 0x82 and probably also 8 channels,
- * but it seems that it works only with Pioneer proprietary software.
- * Even on officially supported OS, the Audacity was unable to record
- * and Mixxx to recognize the control vinyls.
+ * Playback channels could be mapped to:
+ * - CH1
+ * - CH2
+ * - AUX
+ *
+ * Recording channels could be mapped to:
+ * - Post CH1 Fader
+ * - Post CH2 Fader
+ * - Cross Fader A
+ * - Cross Fader B
+ * - MIC
+ * - AUX
+ * - REC OUT
+ *
+ * There is remaining problem with recording directly from PHONO/LINE.
+ * If we map a channel to:
+ * - CH1 Control Tone PHONO
+ * - CH1 Control Tone LINE
+ * - CH2 Control Tone PHONO
+ * - CH2 Control Tone LINE
+ * it is silent.
+ * There is no signal even on other operating systems with official drivers.
+ * The signal appears only when a supported application is started.
+ * This needs to be investigated yet...
+ * (there is quite a lot communication on the USB in both directions)
+ *
+ * In current version this mixer could be used for playback
+ * and for recording from vinyls (through Post CH* Fader)
+ * but not for DVS (Digital Vinyl Systems) like in Mixxx.
*/
USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017),
.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
@@ -3580,6 +3616,26 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
.rate_max = 48000,
.nr_rates = 1,
.rate_table = (unsigned int[]) { 48000 }
+ }
+ },
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S24_3LE,
+ .channels = 8, // inputs
+ .iface = 0,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .endpoint = 0x82,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC|
+ USB_ENDPOINT_SYNC_ASYNC|
+ USB_ENDPOINT_USAGE_IMPLICIT_FB,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .nr_rates = 1,
+ .rate_table = (unsigned int[]) { 48000 }
}
},
{
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index abf99b814a0f..75bbdc691243 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -518,6 +518,15 @@ static int setup_fmt_after_resume_quirk(struct snd_usb_audio *chip,
return 1; /* Continue with creating streams and mixer */
}
+static int setup_disable_autosuspend(struct snd_usb_audio *chip,
+ struct usb_interface *iface,
+ struct usb_driver *driver,
+ const struct snd_usb_audio_quirk *quirk)
+{
+ driver->supports_autosuspend = 0;
+ return 1; /* Continue with creating streams and mixer */
+}
+
/*
* audio-interface quirks
*
@@ -557,6 +566,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
[QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk,
[QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk,
[QUIRK_SETUP_FMT_AFTER_RESUME] = setup_fmt_after_resume_quirk,
+ [QUIRK_SETUP_DISABLE_AUTOSUSPEND] = setup_disable_autosuspend,
};
if (quirk->type < QUIRK_TYPE_COUNT) {
@@ -1493,6 +1503,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
set_format_emu_quirk(subs, fmt);
break;
case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */
+ case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */
pioneer_djm_set_format_quirk(subs);
break;
case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index b91c4c0807ec..6839915a0128 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -102,6 +102,7 @@ enum quirk_type {
QUIRK_AUDIO_ALIGN_TRANSFER,
QUIRK_AUDIO_STANDARD_MIXER,
QUIRK_SETUP_FMT_AFTER_RESUME,
+ QUIRK_SETUP_DISABLE_AUTOSUSPEND,
QUIRK_TYPE_COUNT
};
diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig
index 77777192f650..4ffcc5e623c2 100644
--- a/sound/x86/Kconfig
+++ b/sound/x86/Kconfig
@@ -9,7 +9,7 @@ menuconfig SND_X86
if SND_X86
config HDMI_LPE_AUDIO
- tristate "HDMI audio without HDaudio on Intel Atom platforms"
+ tristate "HDMI audio without HDAudio on Intel Atom platforms"
depends on DRM_I915
select SND_PCM
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