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-rw-r--r--sound/hda/ext/hdac_ext_bus.c1
-rw-r--r--sound/pci/hda/hda_codec.c4
-rw-r--r--sound/pci/hda/patch_cirrus.c1
-rw-r--r--sound/pci/hda/patch_conexant.c1
-rw-r--r--sound/pci/hda/patch_realtek.c1
-rw-r--r--sound/pci/hda/patch_sigmatel.c6
-rw-r--r--sound/soc/au1x/db1200.c4
-rw-r--r--sound/soc/codecs/rt298.c26
-rw-r--r--sound/soc/codecs/rt5645.c6
-rw-r--r--sound/soc/codecs/rt5645.h16
-rw-r--r--sound/soc/codecs/sgtl5000.c6
-rw-r--r--sound/soc/codecs/tas2552.c2
-rw-r--r--sound/soc/codecs/tlv320aic3x.c19
-rw-r--r--sound/soc/codecs/wm8962.c7
-rw-r--r--sound/soc/dwc/designware_i2s.c19
-rw-r--r--sound/soc/fsl/imx-ssi.c19
-rw-r--r--sound/soc/soc-ops.c28
-rw-r--r--sound/synth/emux/emux_oss.c3
18 files changed, 113 insertions, 56 deletions
diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c
index 4449d1a99089..2433f7c81472 100644
--- a/sound/hda/ext/hdac_ext_bus.c
+++ b/sound/hda/ext/hdac_ext_bus.c
@@ -19,6 +19,7 @@
#include <linux/module.h>
#include <linux/slab.h>
+#include <linux/io.h>
#include <sound/hdaudio_ext.h>
MODULE_DESCRIPTION("HDA extended core");
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 37f43a1b34ef..a249d5486889 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3367,10 +3367,8 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec)
int dev, err;
err = snd_hda_codec_parse_pcms(codec);
- if (err < 0) {
- snd_hda_codec_reset(codec);
+ if (err < 0)
return err;
- }
/* attach a new PCM streams */
list_for_each_entry(cpcm, &codec->pcm_list_head, list) {
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 584a0343ab0c..85813de26da8 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -633,6 +633,7 @@ static const struct snd_pci_quirk cs4208_mac_fixup_tbl[] = {
SND_PCI_QUIRK(0x106b, 0x5e00, "MacBookPro 11,2", CS4208_MBP11),
SND_PCI_QUIRK(0x106b, 0x7100, "MacBookAir 6,1", CS4208_MBA6),
SND_PCI_QUIRK(0x106b, 0x7200, "MacBookAir 6,2", CS4208_MBA6),
+ SND_PCI_QUIRK(0x106b, 0x7b00, "MacBookPro 12,1", CS4208_MBP11),
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index ca03c40609fc..2f0ec7c45fc7 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -819,6 +819,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD),
+ 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, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index afec6dc9f91f..16b8dcba5c12 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5306,6 +5306,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 9d947aef2c8b..def5cc8dff02 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4520,7 +4520,11 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
return err;
spec = codec->spec;
- codec->power_save_node = 1;
+ /* enable power_save_node only for new 92HD89xx chips, as it causes
+ * click noises on old 92HD73xx chips.
+ */
+ if ((codec->core.vendor_id & 0xfffffff0) != 0x111d7670)
+ codec->power_save_node = 1;
spec->linear_tone_beep = 0;
spec->gen.mixer_nid = 0x1d;
spec->have_spdif_mux = 1;
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 58c3164802b8..8c907ebea189 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -129,6 +129,8 @@ static struct snd_soc_dai_link db1300_i2s_dai = {
.cpu_dai_name = "au1xpsc_i2s.2",
.platform_name = "au1xpsc-pcm.2",
.codec_name = "wm8731.0-001b",
+ .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.ops = &db1200_i2s_wm8731_ops,
};
@@ -146,6 +148,8 @@ static struct snd_soc_dai_link db1550_i2s_dai = {
.cpu_dai_name = "au1xpsc_i2s.3",
.platform_name = "au1xpsc-pcm.3",
.codec_name = "wm8731.0-001b",
+ .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.ops = &db1200_i2s_wm8731_ops,
};
diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c
index 3c2f0f8d6266..f823eb502367 100644
--- a/sound/soc/codecs/rt298.c
+++ b/sound/soc/codecs/rt298.c
@@ -50,24 +50,24 @@ struct rt298_priv {
};
static struct reg_default rt298_index_def[] = {
- { 0x01, 0xaaaa },
- { 0x02, 0x8aaa },
+ { 0x01, 0xa5a8 },
+ { 0x02, 0x8e95 },
{ 0x03, 0x0002 },
- { 0x04, 0xaf01 },
- { 0x08, 0x000d },
- { 0x09, 0xd810 },
- { 0x0a, 0x0120 },
+ { 0x04, 0xaf67 },
+ { 0x08, 0x200f },
+ { 0x09, 0xd010 },
+ { 0x0a, 0x0100 },
{ 0x0b, 0x0000 },
{ 0x0d, 0x2800 },
- { 0x0f, 0x0000 },
- { 0x19, 0x0a17 },
+ { 0x0f, 0x0022 },
+ { 0x19, 0x0217 },
{ 0x20, 0x0020 },
{ 0x33, 0x0208 },
{ 0x46, 0x0300 },
- { 0x49, 0x0004 },
- { 0x4f, 0x50e9 },
- { 0x50, 0x2000 },
- { 0x63, 0x2902 },
+ { 0x49, 0x4004 },
+ { 0x4f, 0x50c9 },
+ { 0x50, 0x3000 },
+ { 0x63, 0x1b02 },
{ 0x67, 0x1111 },
{ 0x68, 0x1016 },
{ 0x69, 0x273f },
@@ -1214,7 +1214,7 @@ static int rt298_i2c_probe(struct i2c_client *i2c,
mdelay(10);
if (!rt298->pdata.gpio2_en)
- regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0x4000);
+ regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0x40);
else
regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0);
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 268a28bd1df4..5c101af0ac63 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -519,11 +519,11 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = {
RT5645_L_VOL_SFT + 1, RT5645_R_VOL_SFT + 1, 63, 0, adc_vol_tlv),
/* ADC Boost Volume Control */
- SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5645_ADC_BST_VOL1,
+ SOC_DOUBLE_TLV("ADC Boost Capture Volume", RT5645_ADC_BST_VOL1,
RT5645_STO1_ADC_L_BST_SFT, RT5645_STO1_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
- SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5645_ADC_BST_VOL1,
- RT5645_STO2_ADC_L_BST_SFT, RT5645_STO2_ADC_R_BST_SFT, 3, 0,
+ SOC_DOUBLE_TLV("Mono ADC Boost Capture Volume", RT5645_ADC_BST_VOL2,
+ RT5645_MONO_ADC_L_BST_SFT, RT5645_MONO_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
/* I2S2 function select */
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index 0e4cfc6ac649..8c964cfb120d 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -39,8 +39,8 @@
#define RT5645_STO1_ADC_DIG_VOL 0x1c
#define RT5645_MONO_ADC_DIG_VOL 0x1d
#define RT5645_ADC_BST_VOL1 0x1e
-/* Mixer - D-D */
#define RT5645_ADC_BST_VOL2 0x20
+/* Mixer - D-D */
#define RT5645_STO1_ADC_MIXER 0x27
#define RT5645_MONO_ADC_MIXER 0x28
#define RT5645_AD_DA_MIXER 0x29
@@ -315,12 +315,14 @@
#define RT5645_STO1_ADC_R_BST_SFT 12
#define RT5645_STO1_ADC_COMP_MASK (0x3 << 10)
#define RT5645_STO1_ADC_COMP_SFT 10
-#define RT5645_STO2_ADC_L_BST_MASK (0x3 << 8)
-#define RT5645_STO2_ADC_L_BST_SFT 8
-#define RT5645_STO2_ADC_R_BST_MASK (0x3 << 6)
-#define RT5645_STO2_ADC_R_BST_SFT 6
-#define RT5645_STO2_ADC_COMP_MASK (0x3 << 4)
-#define RT5645_STO2_ADC_COMP_SFT 4
+
+/* ADC Boost Volume Control (0x20) */
+#define RT5645_MONO_ADC_L_BST_MASK (0x3 << 14)
+#define RT5645_MONO_ADC_L_BST_SFT 14
+#define RT5645_MONO_ADC_R_BST_MASK (0x3 << 12)
+#define RT5645_MONO_ADC_R_BST_SFT 12
+#define RT5645_MONO_ADC_COMP_MASK (0x3 << 10)
+#define RT5645_MONO_ADC_COMP_SFT 10
/* Stereo2 ADC Mixer Control (0x26) */
#define RT5645_STO2_ADC_SRC_MASK (0x1 << 15)
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index bfda25ef0dd4..f540f82b1f27 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1376,8 +1376,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT);
snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL,
- SGTL5000_BIAS_R_MASK,
- sgtl5000->micbias_voltage << SGTL5000_BIAS_R_SHIFT);
+ SGTL5000_BIAS_VOLT_MASK,
+ sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT);
/*
* disable DAP
* TODO:
@@ -1549,7 +1549,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
else {
sgtl5000->micbias_voltage = 0;
dev_err(&client->dev,
- "Unsuitable MicBias resistor\n");
+ "Unsuitable MicBias voltage\n");
}
} else {
sgtl5000->micbias_voltage = 0;
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index e3a0bca28bcf..cc1d3981fa4b 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -549,7 +549,7 @@ static struct snd_soc_dai_driver tas2552_dai[] = {
/*
* DAC digital volumes. From -7 to 24 dB in 1 dB steps
*/
-static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 0);
+static DECLARE_TLV_DB_SCALE(dac_tlv, -700, 100, 0);
static const char * const tas2552_din_source_select[] = {
"Muted",
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 1a82b19b2644..8739126a1f6f 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1509,14 +1509,17 @@ static int aic3x_init(struct snd_soc_codec *codec)
snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL);
snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL);
- /* Line2 to HP Bypass default volume, disconnect from Output Mixer */
- snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL);
- snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL);
- snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL);
- snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL);
- /* Line2 Line Out default volume, disconnect from Output Mixer */
- snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL);
- snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL);
+ /* On tlv320aic3104, these registers are reserved and must not be written */
+ if (aic3x->model != AIC3X_MODEL_3104) {
+ /* Line2 to HP Bypass default volume, disconnect from Output Mixer */
+ snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL);
+ snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL);
+ snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL);
+ snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL);
+ /* Line2 Line Out default volume, disconnect from Output Mixer */
+ snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL);
+ snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL);
+ }
switch (aic3x->model) {
case AIC3X_MODEL_3X:
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 293e47a6ff59..39ebd7bf4f53 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3760,7 +3760,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_wm8962, &wm8962_dai, 1);
if (ret < 0)
- goto err_enable;
+ goto err_pm_runtime;
regcache_cache_only(wm8962->regmap, true);
@@ -3769,6 +3769,8 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
return 0;
+err_pm_runtime:
+ pm_runtime_disable(&i2c->dev);
err_enable:
regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
err:
@@ -3778,6 +3780,7 @@ err:
static int wm8962_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
+ pm_runtime_disable(&client->dev);
return 0;
}
@@ -3805,6 +3808,8 @@ static int wm8962_runtime_resume(struct device *dev)
wm8962_reset(wm8962);
+ regcache_mark_dirty(wm8962->regmap);
+
/* SYSCLK defaults to on; make sure it is off so we can safely
* write to registers if the device is declocked.
*/
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index a3e97b46b64e..ba34252b7bba 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -131,23 +131,32 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream)
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
for (i = 0; i < 4; i++)
- i2s_write_reg(dev->i2s_base, TOR(i), 0);
+ i2s_read_reg(dev->i2s_base, TOR(i));
} else {
for (i = 0; i < 4; i++)
- i2s_write_reg(dev->i2s_base, ROR(i), 0);
+ i2s_read_reg(dev->i2s_base, ROR(i));
}
}
static void i2s_start(struct dw_i2s_dev *dev,
struct snd_pcm_substream *substream)
{
-
+ u32 i, irq;
i2s_write_reg(dev->i2s_base, IER, 1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ for (i = 0; i < 4; i++) {
+ irq = i2s_read_reg(dev->i2s_base, IMR(i));
+ i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30);
+ }
i2s_write_reg(dev->i2s_base, ITER, 1);
- else
+ } else {
+ for (i = 0; i < 4; i++) {
+ irq = i2s_read_reg(dev->i2s_base, IMR(i));
+ i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03);
+ }
i2s_write_reg(dev->i2s_base, IRER, 1);
+ }
i2s_write_reg(dev->i2s_base, CER, 1);
}
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 48b2d24dd1f0..b95132e2f9dc 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -95,7 +95,8 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
/* data on rising edge of bclk, frame low 1clk before data */
- strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0;
+ strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSI |
+ SSI_STCR_TEFS;
scr |= SSI_SCR_NET;
if (ssi->flags & IMX_SSI_USE_I2S_SLAVE) {
scr &= ~SSI_I2S_MODE_MASK;
@@ -104,33 +105,31 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
break;
case SND_SOC_DAIFMT_LEFT_J:
/* data on rising edge of bclk, frame high with data */
- strcr |= SSI_STCR_TXBIT0;
+ strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP;
break;
case SND_SOC_DAIFMT_DSP_B:
/* data on rising edge of bclk, frame high with data */
- strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0;
+ strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSL;
break;
case SND_SOC_DAIFMT_DSP_A:
/* data on rising edge of bclk, frame high 1clk before data */
- strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS;
+ strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSL |
+ SSI_STCR_TEFS;
break;
}
/* DAI clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_IB_IF:
- strcr |= SSI_STCR_TFSI;
- strcr &= ~SSI_STCR_TSCKP;
+ strcr ^= SSI_STCR_TSCKP | SSI_STCR_TFSI;
break;
case SND_SOC_DAIFMT_IB_NF:
- strcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI);
+ strcr ^= SSI_STCR_TSCKP;
break;
case SND_SOC_DAIFMT_NB_IF:
- strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP;
+ strcr ^= SSI_STCR_TFSI;
break;
case SND_SOC_DAIFMT_NB_NF:
- strcr &= ~SSI_STCR_TFSI;
- strcr |= SSI_STCR_TSCKP;
break;
}
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 100d92b5b77e..05977ae1ff2a 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -207,6 +207,34 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
EXPORT_SYMBOL_GPL(snd_soc_info_volsw);
/**
+ * snd_soc_info_volsw_sx - Mixer info callback for SX TLV controls
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a single mixer control, or a double
+ * mixer control that spans 2 registers of the SX TLV type. SX TLV controls
+ * have a range that represents both positive and negative values either side
+ * of zero but without a sign bit.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_volsw_sx(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ snd_soc_info_volsw(kcontrol, uinfo);
+ /* Max represents the number of levels in an SX control not the
+ * maximum value, so add the minimum value back on
+ */
+ uinfo->value.integer.max += mc->min;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_sx);
+
+/**
* snd_soc_get_volsw - single mixer get callback
* @kcontrol: mixer control
* @ucontrol: control element information
diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c
index 82e350e9501c..ac75816ada7c 100644
--- a/sound/synth/emux/emux_oss.c
+++ b/sound/synth/emux/emux_oss.c
@@ -69,7 +69,8 @@ snd_emux_init_seq_oss(struct snd_emux *emu)
struct snd_seq_oss_reg *arg;
struct snd_seq_device *dev;
- if (snd_seq_device_new(emu->card, 0, SNDRV_SEQ_DEV_ID_OSS,
+ /* using device#1 here for avoiding conflicts with OPL3 */
+ if (snd_seq_device_new(emu->card, 1, SNDRV_SEQ_DEV_ID_OSS,
sizeof(struct snd_seq_oss_reg), &dev) < 0)
return;
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-26 17:34:25 +0000