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authorDaiane Angolini <daiane.angolini@foundries.io>2022-11-07 08:19:55 -0300
committerDaiane Angolini <daiane.angolini@foundries.io>2022-11-07 08:20:24 -0300
commit1db947108c27f7769c873f02c7af67e212d880a2 (patch)
tree2c2e32d8ad5c6d53acf93b22a825dc7a91b9cb2e /sound
parent5abdee26320fc69a8a445cc5944f6792648d2714 (diff)
parent374bf3fc1f53c6611c4ad98d11863344d375e2be (diff)
Merge tag 'v5.15.61' into 5.15-2.1.x-imx
This is the 5.15.61 stable release Conflicts: arch/arm/boot/dts/imx6ul.dtsi drivers/media/platform/nxp/imx-jpeg/mxc-jpeg.c drivers/media/platform/nxp/imx-jpeg/mxc-jpeg.h Signed-off-by: Daiane Angolini <daiane.angolini@foundries.io>
Diffstat (limited to 'sound')
-rw-r--r--sound/pci/hda/patch_cirrus.c1
-rw-r--r--sound/pci/hda/patch_conexant.c11
-rw-r--r--sound/pci/hda/patch_realtek.c15
-rw-r--r--sound/soc/atmel/mchp-spdifrx.c9
-rw-r--r--sound/soc/codecs/cros_ec_codec.c1
-rw-r--r--sound/soc/codecs/da7210.c2
-rw-r--r--sound/soc/codecs/msm8916-wcd-digital.c46
-rw-r--r--sound/soc/codecs/mt6359-accdet.c1
-rw-r--r--sound/soc/codecs/mt6359.c1
-rw-r--r--sound/soc/codecs/wcd9335.c81
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c5
-rw-r--r--sound/soc/fsl/fsl_asrc.c6
-rw-r--r--sound/soc/fsl/fsl_easrc.c9
-rw-r--r--sound/soc/fsl/fsl_easrc.h2
-rw-r--r--sound/soc/fsl/imx-audmux.c2
-rw-r--r--sound/soc/fsl/imx-card.c22
-rw-r--r--sound/soc/generic/audio-graph-card.c4
-rw-r--r--sound/soc/mediatek/mt6797/mt6797-mt6351.c6
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c10
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-rt5650.c9
-rw-r--r--sound/soc/qcom/lpass-cpu.c1
-rw-r--r--sound/soc/qcom/qdsp6/q6adm.c2
-rw-r--r--sound/soc/samsung/aries_wm8994.c6
-rw-r--r--sound/soc/samsung/h1940_uda1380.c2
-rw-r--r--sound/soc/samsung/rx1950_uda1380.c4
-rw-r--r--sound/usb/bcd2000/bcd2000.c3
-rw-r--r--sound/usb/quirks.c2
27 files changed, 161 insertions, 102 deletions
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 678fbcaf2a3b..6807b4708a17 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -395,6 +395,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = {
/* codec SSID */
SND_PCI_QUIRK(0x106b, 0x0600, "iMac 14,1", CS420X_IMAC27_122),
+ SND_PCI_QUIRK(0x106b, 0x0900, "iMac 12,1", CS420X_IMAC27_122),
SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81),
SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122),
SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101),
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 8c68e6e1387e..2bc9274e0960 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -222,6 +222,7 @@ enum {
CXT_PINCFG_LEMOTE_A1205,
CXT_PINCFG_COMPAQ_CQ60,
CXT_FIXUP_STEREO_DMIC,
+ CXT_PINCFG_LENOVO_NOTEBOOK,
CXT_FIXUP_INC_MIC_BOOST,
CXT_FIXUP_HEADPHONE_MIC_PIN,
CXT_FIXUP_HEADPHONE_MIC,
@@ -772,6 +773,14 @@ static const struct hda_fixup cxt_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = cxt_fixup_stereo_dmic,
},
+ [CXT_PINCFG_LENOVO_NOTEBOOK] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1a, 0x05d71030 },
+ { }
+ },
+ .chain_id = CXT_FIXUP_STEREO_DMIC,
+ },
[CXT_FIXUP_INC_MIC_BOOST] = {
.type = HDA_FIXUP_FUNC,
.v.func = cxt5066_increase_mic_boost,
@@ -971,7 +980,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
- SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_PINCFG_LENOVO_NOTEBOOK),
SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo G50-70", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f3fa35b302ce..f7ab620d0e30 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6721,6 +6721,7 @@ enum {
ALC269_FIXUP_LIMIT_INT_MIC_BOOST,
ALC269VB_FIXUP_ASUS_ZENBOOK,
ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A,
+ ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE,
ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED,
ALC269VB_FIXUP_ORDISSIMO_EVE2,
ALC283_FIXUP_CHROME_BOOK,
@@ -7297,6 +7298,15 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC269VB_FIXUP_ASUS_ZENBOOK,
},
+ [ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x18, 0x01a110f0 }, /* use as headset mic */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
+ },
[ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc269_fixup_limit_int_mic_boost,
@@ -8852,6 +8862,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT),
SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED),
SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO),
+ SND_PCI_QUIRK(0x103c, 0x86e7, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1),
+ SND_PCI_QUIRK(0x103c, 0x86e8, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1),
SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
SND_PCI_QUIRK(0x103c, 0x8720, "HP EliteBook x360 1040 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED),
@@ -8867,6 +8879,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
ALC285_FIXUP_HP_GPIO_AMP_INIT),
SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation",
ALC285_FIXUP_HP_GPIO_AMP_INIT),
+ SND_PCI_QUIRK(0x103c, 0x8786, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED),
SND_PCI_QUIRK(0x103c, 0x8787, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED),
SND_PCI_QUIRK(0x103c, 0x8788, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED),
SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED),
@@ -8917,6 +8930,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC),
SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC),
+ SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC),
SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK),
SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A),
@@ -8992,6 +9006,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x4041, "Clevo NV4[15]PZ", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1558, 0x40a1, "Clevo NL40GU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1558, 0x40c1, "Clevo NL40[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1558, 0x40d1, "Clevo NL41DU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
diff --git a/sound/soc/atmel/mchp-spdifrx.c b/sound/soc/atmel/mchp-spdifrx.c
index bcd4f3e4fb0f..2a62d9a2fa0d 100644
--- a/sound/soc/atmel/mchp-spdifrx.c
+++ b/sound/soc/atmel/mchp-spdifrx.c
@@ -288,15 +288,17 @@ static void mchp_spdifrx_isr_blockend_en(struct mchp_spdifrx_dev *dev)
spin_unlock_irqrestore(&dev->blockend_lock, flags);
}
-/* called from atomic context only */
+/* called from atomic/non-atomic context */
static void mchp_spdifrx_isr_blockend_dis(struct mchp_spdifrx_dev *dev)
{
- spin_lock(&dev->blockend_lock);
+ unsigned long flags;
+
+ spin_lock_irqsave(&dev->blockend_lock, flags);
dev->blockend_refcount--;
/* don't enable BLOCKEND interrupt if it's already enabled */
if (dev->blockend_refcount == 0)
regmap_write(dev->regmap, SPDIFRX_IDR, SPDIFRX_IR_BLOCKEND);
- spin_unlock(&dev->blockend_lock);
+ spin_unlock_irqrestore(&dev->blockend_lock, flags);
}
static irqreturn_t mchp_spdif_interrupt(int irq, void *dev_id)
@@ -575,6 +577,7 @@ static int mchp_spdifrx_subcode_ch_get(struct mchp_spdifrx_dev *dev,
if (ret <= 0) {
dev_dbg(dev->dev, "user data for channel %d timeout\n",
channel);
+ mchp_spdifrx_isr_blockend_dis(dev);
return ret;
}
diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c
index a201d652aca2..8823edc35113 100644
--- a/sound/soc/codecs/cros_ec_codec.c
+++ b/sound/soc/codecs/cros_ec_codec.c
@@ -994,6 +994,7 @@ static int cros_ec_codec_platform_probe(struct platform_device *pdev)
dev_dbg(dev, "ap_shm_phys_addr=%#llx len=%#x\n",
priv->ap_shm_phys_addr, priv->ap_shm_len);
}
+ of_node_put(node);
}
#endif
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 8af344b2fdbf..d75d15006f64 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -1336,6 +1336,8 @@ static int __init da7210_modinit(void)
int ret = 0;
#if IS_ENABLED(CONFIG_I2C)
ret = i2c_add_driver(&da7210_i2c_driver);
+ if (ret)
+ return ret;
#endif
#if defined(CONFIG_SPI_MASTER)
ret = spi_register_driver(&da7210_spi_driver);
diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c
index 20a07c92b2fc..098a58990f07 100644
--- a/sound/soc/codecs/msm8916-wcd-digital.c
+++ b/sound/soc/codecs/msm8916-wcd-digital.c
@@ -328,8 +328,8 @@ static const struct snd_kcontrol_new rx1_mix2_inp1_mux = SOC_DAPM_ENUM(
static const struct snd_kcontrol_new rx2_mix2_inp1_mux = SOC_DAPM_ENUM(
"RX2 MIX2 INP1 Mux", rx2_mix2_inp1_chain_enum);
-/* Digital Gain control -38.4 dB to +38.4 dB in 0.3 dB steps */
-static const DECLARE_TLV_DB_SCALE(digital_gain, -3840, 30, 0);
+/* Digital Gain control -84 dB to +40 dB in 1 dB steps */
+static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400);
/* Cutoff Freq for High Pass Filter at -3dB */
static const char * const hpf_cutoff_text[] = {
@@ -510,15 +510,15 @@ static int wcd_iir_filter_info(struct snd_kcontrol *kcontrol,
static const struct snd_kcontrol_new msm8916_wcd_digital_snd_controls[] = {
SOC_SINGLE_S8_TLV("RX1 Digital Volume", LPASS_CDC_RX1_VOL_CTL_B2_CTL,
- -128, 127, digital_gain),
+ -84, 40, digital_gain),
SOC_SINGLE_S8_TLV("RX2 Digital Volume", LPASS_CDC_RX2_VOL_CTL_B2_CTL,
- -128, 127, digital_gain),
+ -84, 40, digital_gain),
SOC_SINGLE_S8_TLV("RX3 Digital Volume", LPASS_CDC_RX3_VOL_CTL_B2_CTL,
- -128, 127, digital_gain),
+ -84, 40, digital_gain),
SOC_SINGLE_S8_TLV("TX1 Digital Volume", LPASS_CDC_TX1_VOL_CTL_GAIN,
- -128, 127, digital_gain),
+ -84, 40, digital_gain),
SOC_SINGLE_S8_TLV("TX2 Digital Volume", LPASS_CDC_TX2_VOL_CTL_GAIN,
- -128, 127, digital_gain),
+ -84, 40, digital_gain),
SOC_ENUM("TX1 HPF Cutoff", tx1_hpf_cutoff_enum),
SOC_ENUM("TX2 HPF Cutoff", tx2_hpf_cutoff_enum),
SOC_SINGLE("TX1 HPF Switch", LPASS_CDC_TX1_MUX_CTL, 3, 1, 0),
@@ -553,22 +553,22 @@ static const struct snd_kcontrol_new msm8916_wcd_digital_snd_controls[] = {
WCD_IIR_FILTER_CTL("IIR2 Band3", IIR2, BAND3),
WCD_IIR_FILTER_CTL("IIR2 Band4", IIR2, BAND4),
WCD_IIR_FILTER_CTL("IIR2 Band5", IIR2, BAND5),
- SOC_SINGLE_SX_TLV("IIR1 INP1 Volume", LPASS_CDC_IIR1_GAIN_B1_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("IIR1 INP2 Volume", LPASS_CDC_IIR1_GAIN_B2_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("IIR1 INP3 Volume", LPASS_CDC_IIR1_GAIN_B3_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("IIR1 INP4 Volume", LPASS_CDC_IIR1_GAIN_B4_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("IIR2 INP1 Volume", LPASS_CDC_IIR2_GAIN_B1_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("IIR2 INP2 Volume", LPASS_CDC_IIR2_GAIN_B2_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("IIR2 INP3 Volume", LPASS_CDC_IIR2_GAIN_B3_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("IIR2 INP4 Volume", LPASS_CDC_IIR2_GAIN_B4_CTL,
- 0, -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP1 Volume", LPASS_CDC_IIR1_GAIN_B1_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP2 Volume", LPASS_CDC_IIR1_GAIN_B2_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP3 Volume", LPASS_CDC_IIR1_GAIN_B3_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP4 Volume", LPASS_CDC_IIR1_GAIN_B4_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR2 INP1 Volume", LPASS_CDC_IIR2_GAIN_B1_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR2 INP2 Volume", LPASS_CDC_IIR2_GAIN_B2_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR2 INP3 Volume", LPASS_CDC_IIR2_GAIN_B3_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR2 INP4 Volume", LPASS_CDC_IIR2_GAIN_B4_CTL,
+ -84, 40, digital_gain),
};
diff --git a/sound/soc/codecs/mt6359-accdet.c b/sound/soc/codecs/mt6359-accdet.c
index 6d3d170144a0..c190628e2905 100644
--- a/sound/soc/codecs/mt6359-accdet.c
+++ b/sound/soc/codecs/mt6359-accdet.c
@@ -675,6 +675,7 @@ static int mt6359_accdet_parse_dt(struct mt6359_accdet *priv)
sizeof(struct three_key_threshold));
}
+ of_node_put(node);
dev_warn(priv->dev, "accdet caps=%x\n", priv->caps);
return 0;
diff --git a/sound/soc/codecs/mt6359.c b/sound/soc/codecs/mt6359.c
index 2d6a4a29b850..cf1febe54bcd 100644
--- a/sound/soc/codecs/mt6359.c
+++ b/sound/soc/codecs/mt6359.c
@@ -2780,6 +2780,7 @@ static int mt6359_parse_dt(struct mt6359_priv *priv)
ret = of_property_read_u32(np, "mediatek,mic-type-2",
&priv->mux_select[MUX_MIC_TYPE_2]);
+ of_node_put(np);
if (ret) {
dev_info(priv->dev,
"%s() failed to read mic-type-2, use default (%d)\n",
diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c
index bc5d68c53e5a..3a13afbfb74b 100644
--- a/sound/soc/codecs/wcd9335.c
+++ b/sound/soc/codecs/wcd9335.c
@@ -2252,51 +2252,42 @@ static int wcd9335_rx_hph_mode_put(struct snd_kcontrol *kc,
static const struct snd_kcontrol_new wcd9335_snd_controls[] = {
/* -84dB min - 40dB max */
- SOC_SINGLE_SX_TLV("RX0 Digital Volume", WCD9335_CDC_RX0_RX_VOL_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX1 Digital Volume", WCD9335_CDC_RX1_RX_VOL_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX2 Digital Volume", WCD9335_CDC_RX2_RX_VOL_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX3 Digital Volume", WCD9335_CDC_RX3_RX_VOL_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX4 Digital Volume", WCD9335_CDC_RX4_RX_VOL_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX5 Digital Volume", WCD9335_CDC_RX5_RX_VOL_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX6 Digital Volume", WCD9335_CDC_RX6_RX_VOL_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX7 Digital Volume", WCD9335_CDC_RX7_RX_VOL_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX8 Digital Volume", WCD9335_CDC_RX8_RX_VOL_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX0 Mix Digital Volume",
- WCD9335_CDC_RX0_RX_VOL_MIX_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX1 Mix Digital Volume",
- WCD9335_CDC_RX1_RX_VOL_MIX_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX2 Mix Digital Volume",
- WCD9335_CDC_RX2_RX_VOL_MIX_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX3 Mix Digital Volume",
- WCD9335_CDC_RX3_RX_VOL_MIX_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX4 Mix Digital Volume",
- WCD9335_CDC_RX4_RX_VOL_MIX_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX5 Mix Digital Volume",
- WCD9335_CDC_RX5_RX_VOL_MIX_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX6 Mix Digital Volume",
- WCD9335_CDC_RX6_RX_VOL_MIX_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX7 Mix Digital Volume",
- WCD9335_CDC_RX7_RX_VOL_MIX_CTL,
- 0, -84, 40, digital_gain),
- SOC_SINGLE_SX_TLV("RX8 Mix Digital Volume",
- WCD9335_CDC_RX8_RX_VOL_MIX_CTL,
- 0, -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX0 Digital Volume", WCD9335_CDC_RX0_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX1 Digital Volume", WCD9335_CDC_RX1_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX2 Digital Volume", WCD9335_CDC_RX2_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX3 Digital Volume", WCD9335_CDC_RX3_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX4 Digital Volume", WCD9335_CDC_RX4_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX5 Digital Volume", WCD9335_CDC_RX5_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX6 Digital Volume", WCD9335_CDC_RX6_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX7 Digital Volume", WCD9335_CDC_RX7_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX8 Digital Volume", WCD9335_CDC_RX8_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX0 Mix Digital Volume", WCD9335_CDC_RX0_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX1 Mix Digital Volume", WCD9335_CDC_RX1_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX2 Mix Digital Volume", WCD9335_CDC_RX2_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX3 Mix Digital Volume", WCD9335_CDC_RX3_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX4 Mix Digital Volume", WCD9335_CDC_RX4_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX5 Mix Digital Volume", WCD9335_CDC_RX5_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX6 Mix Digital Volume", WCD9335_CDC_RX6_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX7 Mix Digital Volume", WCD9335_CDC_RX7_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX8 Mix Digital Volume", WCD9335_CDC_RX8_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
SOC_ENUM("RX INT0_1 HPF cut off", cf_int0_1_enum),
SOC_ENUM("RX INT0_2 HPF cut off", cf_int0_2_enum),
SOC_ENUM("RX INT1_1 HPF cut off", cf_int1_1_enum),
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 0aaf8aa88794..0f3a6290403d 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -700,6 +700,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
struct platform_device *client_pdev[2];
const char *codec_dai_name;
const char *codec_dev_name;
+ u32 asrc_fmt = 0;
u32 width;
int ret;
@@ -1013,8 +1014,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
goto asrc_fail;
}
- ret = of_property_read_u32(asrc_np, "fsl,asrc-format",
- &priv->asrc_format);
+ ret = of_property_read_u32(asrc_np, "fsl,asrc-format", &asrc_fmt);
+ priv->asrc_format = (__force snd_pcm_format_t)asrc_fmt;
if (ret) {
/* Fallback to old binding; translate to asrc_format */
ret = of_property_read_u32(asrc_np, "fsl,asrc-width",
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index b05fdfb7d336..3538c9268ab4 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -1075,6 +1075,7 @@ static int fsl_asrc_probe(struct platform_device *pdev)
struct resource *res;
void __iomem *regs;
int irq, ret, i;
+ u32 asrc_fmt = 0;
u32 map_idx;
char tmp[16];
u32 width;
@@ -1195,7 +1196,8 @@ static int fsl_asrc_probe(struct platform_device *pdev)
return ret;
}
- ret = of_property_read_u32(np, "fsl,asrc-format", &asrc->asrc_format);
+ ret = of_property_read_u32(np, "fsl,asrc-format", &asrc_fmt);
+ asrc->asrc_format = (__force snd_pcm_format_t)asrc_fmt;
if (ret) {
ret = of_property_read_u32(np, "fsl,asrc-width", &width);
if (ret) {
@@ -1218,7 +1220,7 @@ static int fsl_asrc_probe(struct platform_device *pdev)
}
}
- if (!(FSL_ASRC_FORMATS & (1ULL << asrc->asrc_format))) {
+ if (!(FSL_ASRC_FORMATS & pcm_format_to_bits(asrc->asrc_format))) {
dev_warn(&pdev->dev, "unsupported width, use default S24_LE\n");
asrc->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
}
diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c
index 442015b86e46..91d0b833970f 100644
--- a/sound/soc/fsl/fsl_easrc.c
+++ b/sound/soc/fsl/fsl_easrc.c
@@ -476,7 +476,8 @@ static int fsl_easrc_prefilter_config(struct fsl_asrc *easrc,
struct fsl_asrc_pair *ctx;
struct device *dev;
u32 inrate, outrate, offset = 0;
- u32 in_s_rate, out_s_rate, in_s_fmt, out_s_fmt;
+ u32 in_s_rate, out_s_rate;
+ snd_pcm_format_t in_s_fmt, out_s_fmt;
int ret, i;
if (!easrc)
@@ -1875,6 +1876,7 @@ static int fsl_easrc_probe(struct platform_device *pdev)
struct resource *res;
struct device_node *np;
void __iomem *regs;
+ u32 asrc_fmt = 0;
int ret, irq;
int width;
@@ -1936,7 +1938,8 @@ static int fsl_easrc_probe(struct platform_device *pdev)
return ret;
}
- ret = of_property_read_u32(np, "fsl,asrc-format", &easrc->asrc_format);
+ ret = of_property_read_u32(np, "fsl,asrc-format", &asrc_fmt);
+ easrc->asrc_format = (__force snd_pcm_format_t)asrc_fmt;
if (ret) {
ret = of_property_read_u32(np, "fsl,asrc-width", &width);
if (ret) {
@@ -1959,7 +1962,7 @@ static int fsl_easrc_probe(struct platform_device *pdev)
}
}
- if (!(FSL_EASRC_FORMATS & (1ULL << easrc->asrc_format))) {
+ if (!(FSL_EASRC_FORMATS & (pcm_format_to_bits(easrc->asrc_format)))) {
dev_warn(dev, "unsupported format, switching to S24_LE\n");
easrc->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
}
diff --git a/sound/soc/fsl/fsl_easrc.h b/sound/soc/fsl/fsl_easrc.h
index 30620d56252c..5b8469757c12 100644
--- a/sound/soc/fsl/fsl_easrc.h
+++ b/sound/soc/fsl/fsl_easrc.h
@@ -569,7 +569,7 @@ struct fsl_easrc_io_params {
unsigned int access_len;
unsigned int fifo_wtmk;
unsigned int sample_rate;
- unsigned int sample_format;
+ snd_pcm_format_t sample_format;
unsigned int norm_rate;
};
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index dfa05d40b276..a8e5e0f57faf 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -298,7 +298,7 @@ static int imx_audmux_probe(struct platform_device *pdev)
audmux_clk = NULL;
}
- audmux_type = (enum imx_audmux_type)of_device_get_match_data(&pdev->dev);
+ audmux_type = (uintptr_t)of_device_get_match_data(&pdev->dev);
switch (audmux_type) {
case IMX31_AUDMUX:
diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c
index c494bed8f98e..36b54675ce7f 100644
--- a/sound/soc/fsl/imx-card.c
+++ b/sound/soc/fsl/imx-card.c
@@ -17,6 +17,9 @@
#include "fsl_sai.h"
+#define IMX_CARD_MCLK_22P5792MHZ 22579200
+#define IMX_CARD_MCLK_24P576MHZ 24576000
+
enum codec_type {
CODEC_DUMMY = 0,
CODEC_AK5558 = 1,
@@ -115,7 +118,7 @@ struct imx_card_data {
struct snd_soc_card card;
int num_dapm_routes;
u32 asrc_rate;
- u32 asrc_format;
+ snd_pcm_format_t asrc_format;
};
static struct imx_akcodec_fs_mul ak4458_fs_mul[] = {
@@ -353,9 +356,14 @@ static int imx_aif_hw_params(struct snd_pcm_substream *substream,
mclk_freq = akcodec_get_mclk_rate(substream, params, slots, slot_width);
else
mclk_freq = params_rate(params) * slots * slot_width;
- /* Use the maximum freq from DSD512 (512*44100 = 22579200) */
- if (format_is_dsd(params))
- mclk_freq = 22579200;
+
+ if (format_is_dsd(params)) {
+ /* Use the maximum freq from DSD512 (512*44100 = 22579200) */
+ if (!(params_rate(params) % 11025))
+ mclk_freq = IMX_CARD_MCLK_22P5792MHZ;
+ else
+ mclk_freq = IMX_CARD_MCLK_24P576MHZ;
+ }
ret = snd_soc_dai_set_sysclk(cpu_dai, link_data->cpu_sysclk_id, mclk_freq,
SND_SOC_CLOCK_OUT);
@@ -471,7 +479,7 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
snd_mask_none(mask);
- snd_mask_set(mask, data->asrc_format);
+ snd_mask_set(mask, (__force unsigned int)data->asrc_format);
return 0;
}
@@ -490,6 +498,7 @@ static int imx_card_parse_of(struct imx_card_data *data)
struct dai_link_data *link_data;
struct of_phandle_args args;
int ret, num_links;
+ u32 asrc_fmt = 0;
u32 width;
ret = snd_soc_of_parse_card_name(card, "model");
@@ -636,7 +645,8 @@ static int imx_card_parse_of(struct imx_card_data *data)
goto err;
}
- ret = of_property_read_u32(args.np, "fsl,asrc-format", &data->asrc_format);
+ ret = of_property_read_u32(args.np, "fsl,asrc-format", &asrc_fmt);
+ data->asrc_format = (__force snd_pcm_format_t)asrc_fmt;
if (ret) {
/* Fallback to old binding; translate to asrc_format */
ret = of_property_read_u32(args.np, "fsl,asrc-width", &width);
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 546f6fd0609e..28cbcdb56857 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -158,8 +158,10 @@ static int asoc_simple_parse_dai(struct device_node *ep,
* if he unbinded CPU or Codec.
*/
ret = snd_soc_get_dai_name(&args, &dlc->dai_name);
- if (ret < 0)
+ if (ret < 0) {
+ of_node_put(node);
return ret;
+ }
dlc->of_node = node;
diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c
index 496f32bcfb5e..d2f6213a6bfc 100644
--- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c
+++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c
@@ -217,7 +217,8 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev)
if (!codec_node) {
dev_err(&pdev->dev,
"Property 'audio-codec' missing or invalid\n");
- return -EINVAL;
+ ret = -EINVAL;
+ goto put_platform_node;
}
for_each_card_prelinks(card, i, dai_link) {
if (dai_link->codecs->name)
@@ -230,6 +231,9 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
+ of_node_put(codec_node);
+put_platform_node:
+ of_node_put(platform_node);
return ret;
}
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
index c8e4e85e1057..94a9bbf144d1 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
@@ -256,14 +256,16 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
if (!mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node) {
dev_err(&pdev->dev,
"Property 'audio-codec' missing or invalid\n");
- return -EINVAL;
+ ret = -EINVAL;
+ goto put_node;
}
mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node =
of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1);
if (!mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node) {
dev_err(&pdev->dev,
"Property 'audio-codec' missing or invalid\n");
- return -EINVAL;
+ ret = -EINVAL;
+ goto put_node;
}
mt8173_rt5650_rt5676_codec_conf[0].dlc.of_node =
mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node;
@@ -276,7 +278,8 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
if (!mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codecs->of_node) {
dev_err(&pdev->dev,
"Property 'audio-codec' missing or invalid\n");
- return -EINVAL;
+ ret = -EINVAL;
+ goto put_node;
}
card->dev = &pdev->dev;
@@ -286,6 +289,7 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
+put_node:
of_node_put(platform_node);
return ret;
}
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
index e168d31f4445..1de9dab218c6 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
@@ -280,7 +280,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
if (!mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node) {
dev_err(&pdev->dev,
"Property 'audio-codec' missing or invalid\n");
- return -EINVAL;
+ ret = -EINVAL;
+ goto put_platform_node;
}
mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node =
mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node;
@@ -293,7 +294,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
dev_err(&pdev->dev,
"%s codec_capture_dai name fail %d\n",
__func__, ret);
- return ret;
+ goto put_platform_node;
}
mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[1].dai_name =
codec_capture_dai;
@@ -315,7 +316,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
if (!mt8173_rt5650_dais[DAI_LINK_HDMI_I2S].codecs->of_node) {
dev_err(&pdev->dev,
"Property 'audio-codec' missing or invalid\n");
- return -EINVAL;
+ ret = -EINVAL;
+ goto put_platform_node;
}
card->dev = &pdev->dev;
@@ -324,6 +326,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
+put_platform_node:
of_node_put(platform_node);
return ret;
}
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index 3bd9eb3cc688..5e89d280e355 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -880,6 +880,7 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev)
dsp_of_node = of_parse_phandle(pdev->dev.of_node, "qcom,adsp", 0);
if (dsp_of_node) {
dev_err(dev, "DSP exists and holds audio resources\n");
+ of_node_put(dsp_of_node);
return -EBUSY;
}
diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c
index 3d831b635524..4ae97afe9062 100644
--- a/sound/soc/qcom/qdsp6/q6adm.c
+++ b/sound/soc/qcom/qdsp6/q6adm.c
@@ -217,7 +217,7 @@ static struct q6copp *q6adm_alloc_copp(struct q6adm *adm, int port_idx)
idx = find_first_zero_bit(&adm->copp_bitmap[port_idx],
MAX_COPPS_PER_PORT);
- if (idx > MAX_COPPS_PER_PORT)
+ if (idx >= MAX_COPPS_PER_PORT)
return ERR_PTR(-EBUSY);
c = kzalloc(sizeof(*c), GFP_ATOMIC);
diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c
index 83acbe57b248..a0825da9fff9 100644
--- a/sound/soc/samsung/aries_wm8994.c
+++ b/sound/soc/samsung/aries_wm8994.c
@@ -628,8 +628,10 @@ static int aries_audio_probe(struct platform_device *pdev)
return -EINVAL;
codec = of_get_child_by_name(dev->of_node, "codec");
- if (!codec)
- return -EINVAL;
+ if (!codec) {
+ ret = -EINVAL;
+ goto out;
+ }
for_each_card_prelinks(card, i, dai_link) {
dai_link->codecs->of_node = of_parse_phandle(codec,
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index c994e67d1eaf..ca086243fcfd 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -8,7 +8,7 @@
// Based on version from Arnaud Patard <arnaud.patard@rtp-net.org>
#include <linux/types.h>
-#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
#include <linux/module.h>
#include <sound/soc.h>
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 6ea1c8cc9167..2820097b00b9 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -128,7 +128,7 @@ static int rx1950_startup(struct snd_pcm_substream *substream)
&hw_rates);
}
-struct gpio_desc *gpiod_speaker_power;
+static struct gpio_desc *gpiod_speaker_power;
static int rx1950_spk_power(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
@@ -227,7 +227,7 @@ static int rx1950_probe(struct platform_device *pdev)
return devm_snd_soc_register_card(dev, &rx1950_asoc);
}
-struct platform_driver rx1950_audio = {
+static struct platform_driver rx1950_audio = {
.driver = {
.name = "rx1950-audio",
.pm = &snd_soc_pm_ops,
diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c
index cd4a0bc6d278..7aec0a95c609 100644
--- a/sound/usb/bcd2000/bcd2000.c
+++ b/sound/usb/bcd2000/bcd2000.c
@@ -348,7 +348,8 @@ static int bcd2000_init_midi(struct bcd2000 *bcd2k)
static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k,
struct usb_interface *interface)
{
- /* usb_kill_urb not necessary, urb is aborted automatically */
+ usb_kill_urb(bcd2k->midi_out_urb);
+ usb_kill_urb(bcd2k->midi_in_urb);
usb_free_urb(bcd2k->midi_out_urb);
usb_free_urb(bcd2k->midi_in_urb);
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 968d90caeefa..168fd802d70b 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1843,6 +1843,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER),
DEVICE_FLG(0x1395, 0x740a, /* Sennheiser DECT */
QUIRK_FLAG_GET_SAMPLE_RATE),
+ DEVICE_FLG(0x1397, 0x0507, /* Behringer UMC202HD */
+ QUIRK_FLAG_PLAYBACK_FIRST | QUIRK_FLAG_GENERIC_IMPLICIT_FB),
DEVICE_FLG(0x1397, 0x0508, /* Behringer UMC204HD */
QUIRK_FLAG_PLAYBACK_FIRST | QUIRK_FLAG_GENERIC_IMPLICIT_FB),
DEVICE_FLG(0x1397, 0x0509, /* Behringer UMC404HD */