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-rw-r--r--sound/drivers/aloop.c11
-rw-r--r--sound/pci/hda/hda_intel.c8
-rw-r--r--sound/pci/hda/patch_conexant.c1
-rw-r--r--sound/pci/hda/patch_realtek.c20
-rw-r--r--sound/soc/amd/raven/acp3x-i2s.c6
-rw-r--r--sound/soc/amd/raven/acp3x-pcm-dma.c6
-rw-r--r--sound/soc/bcm/cygnus-ssp.c4
-rw-r--r--sound/soc/codecs/adau17x1.c13
-rw-r--r--sound/soc/codecs/lpass-rx-macro.c2
-rw-r--r--sound/soc/codecs/lpass-tx-macro.c2
-rw-r--r--sound/soc/codecs/max98373-i2c.c1
-rw-r--r--sound/soc/codecs/max98373-sdw.c1
-rw-r--r--sound/soc/codecs/max98373.c2
-rw-r--r--sound/soc/codecs/rt1011.c16
-rw-r--r--sound/soc/codecs/rt1011.h1
-rw-r--r--sound/soc/codecs/tas2552.c4
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c12
-rw-r--r--sound/soc/codecs/wm8960.c8
-rw-r--r--sound/soc/fsl/fsl_esai.c8
-rw-r--r--sound/soc/intel/atom/sst-mfld-platform-pcm.c12
-rw-r--r--sound/soc/intel/boards/kbl_da7219_max98927.c38
-rw-r--r--sound/soc/intel/keembay/kmb_platform.c5
-rw-r--r--sound/soc/meson/axg-frddr.c27
-rw-r--r--sound/soc/sof/core.c8
-rw-r--r--sound/soc/sof/intel/apl.c3
-rw-r--r--sound/soc/sof/intel/cnl.c19
-rw-r--r--sound/soc/sof/intel/hda-dsp.c21
-rw-r--r--sound/soc/sof/intel/hda.h1
-rw-r--r--sound/soc/sof/intel/icl.c3
-rw-r--r--sound/soc/sof/intel/pci-tgl.c2
-rw-r--r--sound/soc/sof/intel/tgl.c18
-rw-r--r--sound/soc/sunxi/sun4i-codec.c5
-rw-r--r--sound/usb/quirks.c1
33 files changed, 198 insertions, 91 deletions
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 52637180af33..80b814b9922a 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -1571,6 +1571,14 @@ static int loopback_mixer_new(struct loopback *loopback, int notify)
return -ENOMEM;
kctl->id.device = dev;
kctl->id.subdevice = substr;
+
+ /* Add the control before copying the id so that
+ * the numid field of the id is set in the copy.
+ */
+ err = snd_ctl_add(card, kctl);
+ if (err < 0)
+ return err;
+
switch (idx) {
case ACTIVE_IDX:
setup->active_id = kctl->id;
@@ -1587,9 +1595,6 @@ static int loopback_mixer_new(struct loopback *loopback, int notify)
default:
break;
}
- err = snd_ctl_add(card, kctl);
- if (err < 0)
- return err;
}
}
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index b59b0f323d4e..79ade335c8a0 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -989,8 +989,12 @@ static int azx_prepare(struct device *dev)
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip;
+ if (!azx_is_pm_ready(card))
+ return 0;
+
chip = card->private_data;
chip->pm_prepared = 1;
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
flush_work(&azx_bus(chip)->unsol_work);
@@ -1005,7 +1009,11 @@ static void azx_complete(struct device *dev)
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip;
+ if (!azx_is_pm_ready(card))
+ return;
+
chip = card->private_data;
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
chip->pm_prepared = 0;
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index c20dad46a7c9..dfef9c17e140 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -944,6 +944,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x8402, "HP ProBook 645 G4", CXT_FIXUP_MUTE_LED_GPIO),
SND_PCI_QUIRK(0x103c, 0x8427, "HP ZBook Studio G5", CXT_FIXUP_HP_ZBOOK_MUTE_LED),
+ SND_PCI_QUIRK(0x103c, 0x844f, "HP ZBook Studio G5", CXT_FIXUP_HP_ZBOOK_MUTE_LED),
SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x8456, "HP Z2 G4 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x8457, "HP Z2 G4 mini", CXT_FIXUP_HP_MIC_NO_PRESENCE),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2c35d2df09d6..b07c5b5b4516 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3927,6 +3927,15 @@ static void alc271_fixup_dmic(struct hda_codec *codec,
snd_hda_sequence_write(codec, verbs);
}
+/* Fix the speaker amp after resume, etc */
+static void alc269vb_fixup_aspire_e1_coef(struct hda_codec *codec,
+ const struct hda_fixup *fix,
+ int action)
+{
+ if (action == HDA_FIXUP_ACT_INIT)
+ alc_update_coef_idx(codec, 0x0d, 0x6000, 0x6000);
+}
+
static void alc269_fixup_pcm_44k(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
@@ -5256,7 +5265,7 @@ static void alc_determine_headset_type(struct hda_codec *codec)
case 0x10ec0274:
case 0x10ec0294:
alc_process_coef_fw(codec, coef0274);
- msleep(80);
+ msleep(850);
val = alc_read_coef_idx(codec, 0x46);
is_ctia = (val & 0x00f0) == 0x00f0;
break;
@@ -5440,6 +5449,7 @@ static void alc_update_headset_jack_cb(struct hda_codec *codec,
struct hda_jack_callback *jack)
{
snd_hda_gen_hp_automute(codec, jack);
+ alc_update_headset_mode(codec);
}
static void alc_probe_headset_mode(struct hda_codec *codec)
@@ -6300,6 +6310,7 @@ enum {
ALC283_FIXUP_HEADSET_MIC,
ALC255_FIXUP_MIC_MUTE_LED,
ALC282_FIXUP_ASPIRE_V5_PINS,
+ ALC269VB_FIXUP_ASPIRE_E1_COEF,
ALC280_FIXUP_HP_GPIO4,
ALC286_FIXUP_HP_GPIO_LED,
ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY,
@@ -6978,6 +6989,10 @@ static const struct hda_fixup alc269_fixups[] = {
{ },
},
},
+ [ALC269VB_FIXUP_ASPIRE_E1_COEF] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc269vb_fixup_aspire_e1_coef,
+ },
[ALC280_FIXUP_HP_GPIO4] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc280_fixup_hp_gpio4,
@@ -7900,6 +7915,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
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, 0x0840, "Acer Aspire E1", ALC269VB_FIXUP_ASPIRE_E1_COEF),
SND_PCI_QUIRK(0x1025, 0x101c, "Acer Veriton N2510G", ALC269_FIXUP_LIFEBOOK),
SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC),
@@ -8057,6 +8073,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
ALC285_FIXUP_HP_GPIO_AMP_INIT),
SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED),
+ SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP),
@@ -8393,6 +8410,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{.id = ALC283_FIXUP_HEADSET_MIC, .name = "alc283-headset"},
{.id = ALC255_FIXUP_MIC_MUTE_LED, .name = "alc255-dell-mute"},
{.id = ALC282_FIXUP_ASPIRE_V5_PINS, .name = "aspire-v5"},
+ {.id = ALC269VB_FIXUP_ASPIRE_E1_COEF, .name = "aspire-e1-coef"},
{.id = ALC280_FIXUP_HP_GPIO4, .name = "hp-gpio4"},
{.id = ALC286_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"},
{.id = ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, .name = "hp-gpio2-hotkey"},
diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c
index 396cb7d619e2..de6f70d7ef36 100644
--- a/sound/soc/amd/raven/acp3x-i2s.c
+++ b/sound/soc/amd/raven/acp3x-i2s.c
@@ -264,8 +264,7 @@ static struct snd_soc_dai_driver acp3x_i2s_dai = {
.playback = {
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 |
- SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE |
- SNDRV_PCM_FMTBIT_S32_LE,
+ SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE,
.channels_min = 2,
.channels_max = 8,
.rate_min = 8000,
@@ -274,8 +273,7 @@ static struct snd_soc_dai_driver acp3x_i2s_dai = {
.capture = {
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 |
- SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE |
- SNDRV_PCM_FMTBIT_S32_LE,
+ SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE,
.channels_min = 2,
.channels_max = 2,
.rate_min = 8000,
diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c
index 417cda24030c..f22bb2bdf527 100644
--- a/sound/soc/amd/raven/acp3x-pcm-dma.c
+++ b/sound/soc/amd/raven/acp3x-pcm-dma.c
@@ -24,8 +24,7 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_playback = {
SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 |
- SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE |
- SNDRV_PCM_FMTBIT_S32_LE,
+ SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE,
.channels_min = 2,
.channels_max = 8,
.rates = SNDRV_PCM_RATE_8000_96000,
@@ -45,8 +44,7 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_capture = {
SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 |
- SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE |
- SNDRV_PCM_FMTBIT_S32_LE,
+ SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE,
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c
index fea296b41a43..ba03bb62ba96 100644
--- a/sound/soc/bcm/cygnus-ssp.c
+++ b/sound/soc/bcm/cygnus-ssp.c
@@ -1348,8 +1348,10 @@ static int cygnus_ssp_probe(struct platform_device *pdev)
&cygnus_ssp_dai[active_port_count]);
/* negative is err, 0 is active and good, 1 is disabled */
- if (err < 0)
+ if (err < 0) {
+ of_node_put(child_node);
return err;
+ }
else if (!err) {
dev_dbg(dev, "Activating DAI: %s\n",
cygnus_ssp_dai[active_port_count].name);
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 546ee8178038..8aae7ab74091 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -553,6 +553,7 @@ static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai,
{
struct adau *adau = snd_soc_component_get_drvdata(dai->component);
unsigned int ctrl0, ctrl1;
+ unsigned int ctrl0_mask;
int lrclk_pol;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -612,8 +613,16 @@ static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai,
if (lrclk_pol)
ctrl0 |= ADAU17X1_SERIAL_PORT0_LRCLK_POL;
- regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT0, ctrl0);
- regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT1, ctrl1);
+ /* Set the mask to update all relevant bits in ADAU17X1_SERIAL_PORT0 */
+ ctrl0_mask = ADAU17X1_SERIAL_PORT0_MASTER |
+ ADAU17X1_SERIAL_PORT0_LRCLK_POL |
+ ADAU17X1_SERIAL_PORT0_BCLK_POL |
+ ADAU17X1_SERIAL_PORT0_PULSE_MODE;
+
+ regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT0, ctrl0_mask,
+ ctrl0);
+ regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT1,
+ ADAU17X1_SERIAL_PORT1_DELAY_MASK, ctrl1);
adau->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c
index 4f1b569d7c47..b0ebfc8d180c 100644
--- a/sound/soc/codecs/lpass-rx-macro.c
+++ b/sound/soc/codecs/lpass-rx-macro.c
@@ -3549,7 +3549,7 @@ static int rx_macro_probe(struct platform_device *pdev)
/* set MCLK and NPL rates */
clk_set_rate(rx->clks[2].clk, MCLK_FREQ);
- clk_set_rate(rx->clks[3].clk, MCLK_FREQ);
+ clk_set_rate(rx->clks[3].clk, 2 * MCLK_FREQ);
ret = clk_bulk_prepare_enable(RX_NUM_CLKS_MAX, rx->clks);
if (ret)
diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c
index 4eede9ad57bf..acd2fbc0ca7c 100644
--- a/sound/soc/codecs/lpass-tx-macro.c
+++ b/sound/soc/codecs/lpass-tx-macro.c
@@ -1811,7 +1811,7 @@ static int tx_macro_probe(struct platform_device *pdev)
/* set MCLK and NPL rates */
clk_set_rate(tx->clks[2].clk, MCLK_FREQ);
- clk_set_rate(tx->clks[3].clk, MCLK_FREQ);
+ clk_set_rate(tx->clks[3].clk, 2 * MCLK_FREQ);
ret = clk_bulk_prepare_enable(TX_NUM_CLKS_MAX, tx->clks);
if (ret)
diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c
index 85f6865019d4..ddb6436835d7 100644
--- a/sound/soc/codecs/max98373-i2c.c
+++ b/sound/soc/codecs/max98373-i2c.c
@@ -446,6 +446,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg)
case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK:
case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK:
case MAX98373_R20B6_BDE_CUR_STATE_READBACK:
+ case MAX98373_R20FF_GLOBAL_SHDN:
case MAX98373_R21FF_REV_ID:
return true;
default:
diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c
index d8c47667a9ea..f3a12205cd48 100644
--- a/sound/soc/codecs/max98373-sdw.c
+++ b/sound/soc/codecs/max98373-sdw.c
@@ -220,6 +220,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg)
case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK:
case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK:
case MAX98373_R20B6_BDE_CUR_STATE_READBACK:
+ case MAX98373_R20FF_GLOBAL_SHDN:
case MAX98373_R21FF_REV_ID:
/* SoundWire Control Port Registers */
case MAX98373_R0040_SCP_INIT_STAT_1 ... MAX98373_R0070_SCP_FRAME_CTLR:
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index b1e43c26dd6b..e14fe98349a5 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -28,11 +28,13 @@ static int max98373_dac_event(struct snd_soc_dapm_widget *w,
regmap_update_bits(max98373->regmap,
MAX98373_R20FF_GLOBAL_SHDN,
MAX98373_GLOBAL_EN_MASK, 1);
+ usleep_range(30000, 31000);
break;
case SND_SOC_DAPM_POST_PMD:
regmap_update_bits(max98373->regmap,
MAX98373_R20FF_GLOBAL_SHDN,
MAX98373_GLOBAL_EN_MASK, 0);
+ usleep_range(30000, 31000);
max98373->tdm_mode = false;
break;
default:
diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c
index 238cb66f4c6a..faff2b558687 100644
--- a/sound/soc/codecs/rt1011.c
+++ b/sound/soc/codecs/rt1011.c
@@ -2236,18 +2236,9 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag)
dc_offset |= (value & 0xffff);
dev_info(dev, "Gain1 offset=0x%x\n", dc_offset);
- /* check the package info. */
- regmap_read(rt1011->regmap, RT1011_EFUSE_MATCH_DONE, &value);
- if (value & 0x4)
- rt1011->pack_id = 1;
-
if (cali_flag) {
- if (rt1011->pack_id)
- regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x292c);
- else
- regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x2925);
-
+ regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x2925);
/* Class D on */
regmap_write(rt1011->regmap, RT1011_CLASS_D_POS, 0x010e);
regmap_write(rt1011->regmap,
@@ -2373,10 +2364,7 @@ static void rt1011_calibration_work(struct work_struct *work)
rt1011_r0_load(rt1011);
}
- if (rt1011->pack_id)
- snd_soc_component_write(component, RT1011_ADC_SET_1, 0x292c);
- else
- snd_soc_component_write(component, RT1011_ADC_SET_1, 0x2925);
+ snd_soc_component_write(component, RT1011_ADC_SET_1, 0x2925);
}
static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device *dev)
diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h
index f3a9a96640f1..68fadc15fa8c 100644
--- a/sound/soc/codecs/rt1011.h
+++ b/sound/soc/codecs/rt1011.h
@@ -692,7 +692,6 @@ struct rt1011_priv {
unsigned int r0_reg, cali_done;
unsigned int r0_calib, temperature_calib;
int recv_spk_mode;
- unsigned int pack_id; /* 0: WLCSP; 1: QFN */
};
#endif /* end of _RT1011_H_ */
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index bd00c35116cd..700baa6314aa 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -730,8 +730,10 @@ static int tas2552_probe(struct i2c_client *client,
ret = devm_snd_soc_register_component(&client->dev,
&soc_component_dev_tas2552,
tas2552_dai, ARRAY_SIZE(tas2552_dai));
- if (ret < 0)
+ if (ret < 0) {
dev_err(&client->dev, "Failed to register component: %d\n", ret);
+ pm_runtime_get_noresume(&client->dev);
+ }
return ret;
}
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index f04f88c8d425..b689f26fc4be 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -577,12 +577,12 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = {
.window_start = 0,
.window_len = 128,
.range_min = 0,
- .range_max = AIC32X4_RMICPGAVOL,
+ .range_max = AIC32X4_REFPOWERUP,
},
};
const struct regmap_config aic32x4_regmap_config = {
- .max_register = AIC32X4_RMICPGAVOL,
+ .max_register = AIC32X4_REFPOWERUP,
.ranges = aic32x4_regmap_pages,
.num_ranges = ARRAY_SIZE(aic32x4_regmap_pages),
};
@@ -1243,6 +1243,10 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap)
if (ret)
goto err_disable_regulators;
+ ret = aic32x4_register_clocks(dev, aic32x4->mclk_name);
+ if (ret)
+ goto err_disable_regulators;
+
ret = devm_snd_soc_register_component(dev,
&soc_component_dev_aic32x4, &aic32x4_dai, 1);
if (ret) {
@@ -1250,10 +1254,6 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap)
goto err_disable_regulators;
}
- ret = aic32x4_register_clocks(dev, aic32x4->mclk_name);
- if (ret)
- goto err_disable_regulators;
-
return 0;
err_disable_regulators:
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 847ca16b9841..9e621a254392 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -697,7 +697,13 @@ int wm8960_configure_pll(struct snd_soc_component *component, int freq_in,
best_freq_out = -EINVAL;
*sysclk_idx = *dac_idx = *bclk_idx = -1;
- for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) {
+ /*
+ * From Datasheet, the PLL performs best when f2 is between
+ * 90MHz and 100MHz, the desired sysclk output is 11.2896MHz
+ * or 12.288MHz, then sysclkdiv = 2 is the best choice.
+ * So search sysclk_divs from 2 to 1 other than from 1 to 2.
+ */
+ for (i = ARRAY_SIZE(sysclk_divs) - 1; i >= 0; --i) {
if (sysclk_divs[i] == -1)
continue;
for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) {
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index c0d4f3c5dbb1..f356ae5925af 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -519,11 +519,13 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream,
ESAI_SAICR_SYNC, esai_priv->synchronous ?
ESAI_SAICR_SYNC : 0);
- /* Set a default slot number -- 2 */
+ /* Set slots count */
regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR,
- ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2));
+ ESAI_xCCR_xDC_MASK,
+ ESAI_xCCR_xDC(esai_priv->slots));
regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR,
- ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2));
+ ESAI_xCCR_xDC_MASK,
+ ESAI_xCCR_xDC(esai_priv->slots));
}
return 0;
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 9e9b05883557..4124aa2fc247 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -487,15 +487,15 @@ static struct snd_soc_dai_driver sst_platform_dai[] = {
.stream_name = "Headset Playback",
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
- .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
.stream_name = "Headset Capture",
.channels_min = 1,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
{
@@ -505,8 +505,8 @@ static struct snd_soc_dai_driver sst_platform_dai[] = {
.stream_name = "Deepbuffer Playback",
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
- .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
{
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
index 9dfe5bd9180d..4b7b4a044f81 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98927.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -284,12 +284,34 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
- struct snd_soc_dpcm *dpcm = container_of(
- params, struct snd_soc_dpcm, hw_params);
- struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link;
- struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link;
+ struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL;
/*
+ * The following loop will be called only for playback stream
+ * In this platform, there is only one playback device on every SSP
+ */
+ for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
+ rtd_dpcm = dpcm;
+ break;
+ }
+
+ /*
+ * This following loop will be called only for capture stream
+ * In this platform, there is only one capture device on every SSP
+ */
+ for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) {
+ rtd_dpcm = dpcm;
+ break;
+ }
+
+ if (!rtd_dpcm)
+ return -EINVAL;
+
+ /*
+ * The above 2 loops are mutually exclusive based on the stream direction,
+ * thus rtd_dpcm variable will never be overwritten
+ */
+ /*
* Topology for kblda7219m98373 & kblmax98373 supports only S24_LE,
* where as kblda7219m98927 & kblmax98927 supports S16_LE by default.
* Skipping the port wise FE and BE configuration for kblda7219m98373 &
@@ -311,9 +333,9 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
/*
* The ADSP will convert the FE rate to 48k, stereo, 24 bit
*/
- if (!strcmp(fe_dai_link->name, "Kbl Audio Port") ||
- !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
- !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
+ if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") ||
+ !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") ||
+ !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) {
rate->min = rate->max = 48000;
chan->min = chan->max = 2;
snd_mask_none(fmt);
@@ -324,7 +346,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
* The speaker on the SSP0 supports S16_LE and not S24_LE.
* thus changing the mask here
*/
- if (!strcmp(be_dai_link->name, "SSP0-Codec"))
+ if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec"))
snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
return 0;
diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c
index ef73392f8cff..a6fb74ba1c42 100644
--- a/sound/soc/intel/keembay/kmb_platform.c
+++ b/sound/soc/intel/keembay/kmb_platform.c
@@ -105,14 +105,15 @@ static unsigned int kmb_pcm_tx_fn(struct kmb_i2s_info *kmb_i2s,
void *buf = runtime->dma_area;
int i;
+ if (kmb_i2s->iec958_fmt)
+ hdmi_reformat_iec958(runtime, kmb_i2s, tx_ptr);
+
/* KMB i2s uses two separate L/R FIFO */
for (i = 0; i < kmb_i2s->fifo_th; i++) {
if (kmb_i2s->config.data_width == 16) {
writel(((u16(*)[2])buf)[tx_ptr][0], i2s_base + LRBR_LTHR(0));
writel(((u16(*)[2])buf)[tx_ptr][1], i2s_base + RRBR_RTHR(0));
} else {
- if (kmb_i2s->iec958_fmt)
- hdmi_reformat_iec958(runtime, kmb_i2s, tx_ptr);
writel(((u32(*)[2])buf)[tx_ptr][0], i2s_base + LRBR_LTHR(0));
writel(((u32(*)[2])buf)[tx_ptr][1], i2s_base + RRBR_RTHR(0));
}
diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c
index c3ae8ac30745..37f4bb3469b5 100644
--- a/sound/soc/meson/axg-frddr.c
+++ b/sound/soc/meson/axg-frddr.c
@@ -11,6 +11,7 @@
#include <linux/regmap.h>
#include <linux/module.h>
#include <linux/of_platform.h>
+#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dai.h>
@@ -46,11 +47,28 @@ static int g12a_frddr_dai_prepare(struct snd_pcm_substream *substream,
return 0;
}
+static int axg_frddr_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+ unsigned int period, depth, val;
+
+ period = params_period_bytes(params);
+
+ /* Trim the FIFO depth if the period is small to improve latency */
+ depth = min(period, fifo->depth);
+ val = (depth / AXG_FIFO_BURST) - 1;
+ regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH_MASK,
+ CTRL1_FRDDR_DEPTH(val));
+
+ return 0;
+}
+
static int axg_frddr_dai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
- unsigned int val;
int ret;
/* Enable pclk to access registers and clock the fifo ip */
@@ -61,11 +79,6 @@ static int axg_frddr_dai_startup(struct snd_pcm_substream *substream,
/* Apply single buffer mode to the interface */
regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_FRDDR_PP_MODE, 0);
- /* Use all fifo depth */
- val = (fifo->depth / AXG_FIFO_BURST) - 1;
- regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH_MASK,
- CTRL1_FRDDR_DEPTH(val));
-
return 0;
}
@@ -84,6 +97,7 @@ static int axg_frddr_pcm_new(struct snd_soc_pcm_runtime *rtd,
}
static const struct snd_soc_dai_ops axg_frddr_ops = {
+ .hw_params = axg_frddr_dai_hw_params,
.startup = axg_frddr_dai_startup,
.shutdown = axg_frddr_dai_shutdown,
};
@@ -157,6 +171,7 @@ static const struct axg_fifo_match_data axg_frddr_match_data = {
static const struct snd_soc_dai_ops g12a_frddr_ops = {
.prepare = g12a_frddr_dai_prepare,
+ .hw_params = axg_frddr_dai_hw_params,
.startup = axg_frddr_dai_startup,
.shutdown = axg_frddr_dai_shutdown,
};
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index f57e363b5974..3e4dd4a86363 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -405,7 +405,13 @@ int snd_sof_device_shutdown(struct device *dev)
{
struct snd_sof_dev *sdev = dev_get_drvdata(dev);
- return snd_sof_shutdown(sdev);
+ if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE))
+ cancel_work_sync(&sdev->probe_work);
+
+ if (sdev->fw_state == SOF_FW_BOOT_COMPLETE)
+ return snd_sof_shutdown(sdev);
+
+ return 0;
}
EXPORT_SYMBOL(snd_sof_device_shutdown);
diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c
index fc29b91b8932..c7ed2b3d6abc 100644
--- a/sound/soc/sof/intel/apl.c
+++ b/sound/soc/sof/intel/apl.c
@@ -27,9 +27,10 @@ static const struct snd_sof_debugfs_map apl_dsp_debugfs[] = {
/* apollolake ops */
const struct snd_sof_dsp_ops sof_apl_ops = {
- /* probe and remove */
+ /* probe/remove/shutdown */
.probe = hda_dsp_probe,
.remove = hda_dsp_remove,
+ .shutdown = hda_dsp_shutdown,
/* Register IO */
.write = sof_io_write,
diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c
index e38db519f38d..821f25fbcf08 100644
--- a/sound/soc/sof/intel/cnl.c
+++ b/sound/soc/sof/intel/cnl.c
@@ -232,9 +232,10 @@ void cnl_ipc_dump(struct snd_sof_dev *sdev)
/* cannonlake ops */
const struct snd_sof_dsp_ops sof_cnl_ops = {
- /* probe and remove */
+ /* probe/remove/shutdown */
.probe = hda_dsp_probe,
.remove = hda_dsp_remove,
+ .shutdown = hda_dsp_shutdown,
/* Register IO */
.write = sof_io_write,
@@ -349,22 +350,6 @@ const struct sof_intel_dsp_desc cnl_chip_info = {
};
EXPORT_SYMBOL_NS(cnl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
-const struct sof_intel_dsp_desc ehl_chip_info = {
- /* Elkhartlake */
- .cores_num = 4,
- .init_core_mask = 1,
- .host_managed_cores_mask = BIT(0),
- .ipc_req = CNL_DSP_REG_HIPCIDR,
- .ipc_req_mask = CNL_DSP_REG_HIPCIDR_BUSY,
- .ipc_ack = CNL_DSP_REG_HIPCIDA,
- .ipc_ack_mask = CNL_DSP_REG_HIPCIDA_DONE,
- .ipc_ctl = CNL_DSP_REG_HIPCCTL,
- .rom_init_timeout = 300,
- .ssp_count = ICL_SSP_COUNT,
- .ssp_base_offset = CNL_SSP_BASE_OFFSET,
-};
-EXPORT_SYMBOL_NS(ehl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
-
const struct sof_intel_dsp_desc jsl_chip_info = {
/* Jasperlake */
.cores_num = 2,
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index df4cb29b9a1b..623cf291e207 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -226,10 +226,17 @@ bool hda_dsp_core_is_enabled(struct snd_sof_dev *sdev,
val = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPCS);
- is_enable = (val & HDA_DSP_ADSPCS_CPA_MASK(core_mask)) &&
- (val & HDA_DSP_ADSPCS_SPA_MASK(core_mask)) &&
- !(val & HDA_DSP_ADSPCS_CRST_MASK(core_mask)) &&
- !(val & HDA_DSP_ADSPCS_CSTALL_MASK(core_mask));
+#define MASK_IS_EQUAL(v, m, field) ({ \
+ u32 _m = field(m); \
+ ((v) & _m) == _m; \
+})
+
+ is_enable = MASK_IS_EQUAL(val, core_mask, HDA_DSP_ADSPCS_CPA_MASK) &&
+ MASK_IS_EQUAL(val, core_mask, HDA_DSP_ADSPCS_SPA_MASK) &&
+ !(val & HDA_DSP_ADSPCS_CRST_MASK(core_mask)) &&
+ !(val & HDA_DSP_ADSPCS_CSTALL_MASK(core_mask));
+
+#undef MASK_IS_EQUAL
dev_dbg(sdev->dev, "DSP core(s) enabled? %d : core_mask %x\n",
is_enable, core_mask);
@@ -884,6 +891,12 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state)
return snd_sof_dsp_set_power_state(sdev, &target_dsp_state);
}
+int hda_dsp_shutdown(struct snd_sof_dev *sdev)
+{
+ sdev->system_suspend_target = SOF_SUSPEND_S3;
+ return snd_sof_suspend(sdev->dev);
+}
+
int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
{
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 5ef29dcc052d..4d44f8910393 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -517,6 +517,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev);
int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev);
int hda_dsp_runtime_resume(struct snd_sof_dev *sdev);
int hda_dsp_runtime_idle(struct snd_sof_dev *sdev);
+int hda_dsp_shutdown(struct snd_sof_dev *sdev);
int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev);
void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags);
void hda_ipc_dump(struct snd_sof_dev *sdev);
diff --git a/sound/soc/sof/intel/icl.c b/sound/soc/sof/intel/icl.c
index e9d5a0a58504..88a74be8a0c1 100644
--- a/sound/soc/sof/intel/icl.c
+++ b/sound/soc/sof/intel/icl.c
@@ -26,9 +26,10 @@ static const struct snd_sof_debugfs_map icl_dsp_debugfs[] = {
/* Icelake ops */
const struct snd_sof_dsp_ops sof_icl_ops = {
- /* probe and remove */
+ /* probe/remove/shutdown */
.probe = hda_dsp_probe,
.remove = hda_dsp_remove,
+ .shutdown = hda_dsp_shutdown,
/* Register IO */
.write = sof_io_write,
diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c
index 21f69204d5ad..88c3bf404dd7 100644
--- a/sound/soc/sof/intel/pci-tgl.c
+++ b/sound/soc/sof/intel/pci-tgl.c
@@ -66,7 +66,7 @@ static const struct sof_dev_desc ehl_desc = {
.default_tplg_path = "intel/sof-tplg",
.default_fw_filename = "sof-ehl.ri",
.nocodec_tplg_filename = "sof-ehl-nocodec.tplg",
- .ops = &sof_cnl_ops,
+ .ops = &sof_tgl_ops,
};
static const struct sof_dev_desc adls_desc = {
diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c
index 940f169047a0..2ed788304414 100644
--- a/sound/soc/sof/intel/tgl.c
+++ b/sound/soc/sof/intel/tgl.c
@@ -25,7 +25,7 @@ const struct snd_sof_dsp_ops sof_tgl_ops = {
/* probe/remove/shutdown */
.probe = hda_dsp_probe,
.remove = hda_dsp_remove,
- .shutdown = hda_dsp_remove,
+ .shutdown = hda_dsp_shutdown,
/* Register IO */
.write = sof_io_write,
@@ -156,6 +156,22 @@ const struct sof_intel_dsp_desc tglh_chip_info = {
};
EXPORT_SYMBOL_NS(tglh_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
+const struct sof_intel_dsp_desc ehl_chip_info = {
+ /* Elkhartlake */
+ .cores_num = 4,
+ .init_core_mask = 1,
+ .host_managed_cores_mask = BIT(0),
+ .ipc_req = CNL_DSP_REG_HIPCIDR,
+ .ipc_req_mask = CNL_DSP_REG_HIPCIDR_BUSY,
+ .ipc_ack = CNL_DSP_REG_HIPCIDA,
+ .ipc_ack_mask = CNL_DSP_REG_HIPCIDA_DONE,
+ .ipc_ctl = CNL_DSP_REG_HIPCCTL,
+ .rom_init_timeout = 300,
+ .ssp_count = ICL_SSP_COUNT,
+ .ssp_base_offset = CNL_SSP_BASE_OFFSET,
+};
+EXPORT_SYMBOL_NS(ehl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
+
const struct sof_intel_dsp_desc adls_chip_info = {
/* Alderlake-S */
.cores_num = 2,
diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
index 00b73b7444b7..6f3d9148a185 100644
--- a/sound/soc/sunxi/sun4i-codec.c
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -1364,6 +1364,7 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev)
return ERR_PTR(-ENOMEM);
card->dev = dev;
+ card->owner = THIS_MODULE;
card->name = "sun4i-codec";
card->dapm_widgets = sun4i_codec_card_dapm_widgets;
card->num_dapm_widgets = ARRAY_SIZE(sun4i_codec_card_dapm_widgets);
@@ -1396,6 +1397,7 @@ static struct snd_soc_card *sun6i_codec_create_card(struct device *dev)
return ERR_PTR(-ENOMEM);
card->dev = dev;
+ card->owner = THIS_MODULE;
card->name = "A31 Audio Codec";
card->dapm_widgets = sun6i_codec_card_dapm_widgets;
card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets);
@@ -1449,6 +1451,7 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev)
return ERR_PTR(-ENOMEM);
card->dev = dev;
+ card->owner = THIS_MODULE;
card->name = "A23 Audio Codec";
card->dapm_widgets = sun6i_codec_card_dapm_widgets;
card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets);
@@ -1487,6 +1490,7 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev)
return ERR_PTR(-ENOMEM);
card->dev = dev;
+ card->owner = THIS_MODULE;
card->name = "H3 Audio Codec";
card->dapm_widgets = sun6i_codec_card_dapm_widgets;
card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets);
@@ -1525,6 +1529,7 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev)
return ERR_PTR(-ENOMEM);
card->dev = dev;
+ card->owner = THIS_MODULE;
card->name = "V3s Audio Codec";
card->dapm_widgets = sun6i_codec_card_dapm_widgets;
card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets);
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index d3001fb18141..176437a441e6 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1521,6 +1521,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */
case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */
case USB_ID(0x413c, 0xa506): /* Dell AE515 sound bar */
+ case USB_ID(0x046d, 0x084c): /* Logitech ConferenceCam Connect */
return true;
}