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-rw-r--r--sound/arm/Kconfig15
-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.c44
-rw-r--r--sound/pci/hda/patch_sigmatel.c6
-rw-r--r--sound/soc/au1x/db1200.c4
-rw-r--r--sound/soc/codecs/adav80x.c1
-rw-r--r--sound/soc/codecs/arizona.c49
-rw-r--r--sound/soc/codecs/arizona.h1
-rw-r--r--sound/soc/codecs/rt5640.c40
-rw-r--r--sound/soc/codecs/sgtl5000.c4
-rw-r--r--sound/soc/dwc/designware_i2s.c4
-rw-r--r--sound/soc/pxa/Kconfig2
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c4
-rw-r--r--sound/soc/samsung/arndale_rt5631.c10
-rw-r--r--sound/soc/soc-ops.c28
-rw-r--r--sound/synth/emux/emux_oss.c3
-rw-r--r--sound/usb/midi.c46
-rw-r--r--sound/usb/mixer.c2
-rw-r--r--sound/usb/quirks-table.h11
-rw-r--r--sound/usb/quirks.c2
-rw-r--r--sound/usb/usbaudio.h1
23 files changed, 202 insertions, 81 deletions
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index 885683a3b0bd..e0406211716b 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -9,6 +9,14 @@ menuconfig SND_ARM
Drivers that are implemented on ASoC can be found in
"ALSA for SoC audio support" section.
+config SND_PXA2XX_LIB
+ tristate
+ select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
+ select SND_DMAENGINE_PCM
+
+config SND_PXA2XX_LIB_AC97
+ bool
+
if SND_ARM
config SND_ARMAACI
@@ -21,13 +29,6 @@ config SND_PXA2XX_PCM
tristate
select SND_PCM
-config SND_PXA2XX_LIB
- tristate
- select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
-
-config SND_PXA2XX_LIB_AC97
- bool
-
config SND_PXA2XX_AC97
tristate "AC97 driver for the Intel PXA2xx chip"
depends on ARCH_PXA
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 36e8f1236637..57197bef5f5b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3833,10 +3833,8 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec)
return -EINVAL;
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 3a24f7739aaa..b791529bf31c 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -634,6 +634,7 @@ 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 06cc9d57ba3d..488f4c7be33e 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 91f6928560e1..57bb5a559f8e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1134,7 +1134,7 @@ static const struct hda_fixup alc880_fixups[] = {
/* override all pins as BIOS on old Amilo is broken */
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
- { 0x14, 0x0121411f }, /* HP */
+ { 0x14, 0x0121401f }, /* HP */
{ 0x15, 0x99030120 }, /* speaker */
{ 0x16, 0x99030130 }, /* bass speaker */
{ 0x17, 0x411111f0 }, /* N/A */
@@ -1154,7 +1154,7 @@ static const struct hda_fixup alc880_fixups[] = {
/* almost compatible with FUJITSU, but no bass and SPDIF */
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
- { 0x14, 0x0121411f }, /* HP */
+ { 0x14, 0x0121401f }, /* HP */
{ 0x15, 0x99030120 }, /* speaker */
{ 0x16, 0x411111f0 }, /* N/A */
{ 0x17, 0x411111f0 }, /* N/A */
@@ -1363,7 +1363,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = {
SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810),
SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM),
SND_PCI_QUIRK(0x1631, 0xe011, "PB 13201056", ALC880_FIXUP_6ST_AUTOMUTE),
- SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_FIXUP_F1734),
+ SND_PCI_QUIRK(0x1734, 0x107c, "FSC Amilo M1437", ALC880_FIXUP_FUJITSU),
SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FIXUP_FUJITSU),
SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_FIXUP_F1734),
SND_PCI_QUIRK(0x1734, 0x10b0, "FSC Amilo Pi1556", ALC880_FIXUP_FUJITSU),
@@ -4182,6 +4182,24 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec,
}
}
+/* fixup for Thinkpad docks: add dock pins, avoid HP parser fixup */
+static void alc_fixup_tpt440_dock(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ static const struct hda_pintbl pincfgs[] = {
+ { 0x16, 0x21211010 }, /* dock headphone */
+ { 0x19, 0x21a11010 }, /* dock mic */
+ { }
+ };
+ struct alc_spec *spec = codec->spec;
+
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
+ codec->power_save_node = 0; /* avoid click noises */
+ snd_hda_apply_pincfgs(codec, pincfgs);
+ }
+}
+
static void alc_shutup_dell_xps13(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -4507,7 +4525,6 @@ enum {
ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC,
ALC293_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC292_FIXUP_TPT440_DOCK,
- ALC292_FIXUP_TPT440_DOCK2,
ALC283_FIXUP_BXBT2807_MIC,
ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED,
ALC282_FIXUP_ASPIRE_V5_PINS,
@@ -4972,17 +4989,7 @@ static const struct hda_fixup alc269_fixups[] = {
},
[ALC292_FIXUP_TPT440_DOCK] = {
.type = HDA_FIXUP_FUNC,
- .v.func = alc269_fixup_pincfg_no_hp_to_lineout,
- .chained = true,
- .chain_id = ALC292_FIXUP_TPT440_DOCK2
- },
- [ALC292_FIXUP_TPT440_DOCK2] = {
- .type = HDA_FIXUP_PINS,
- .v.pins = (const struct hda_pintbl[]) {
- { 0x16, 0x21211010 }, /* dock headphone */
- { 0x19, 0x21a11010 }, /* dock mic */
- { }
- },
+ .v.func = alc_fixup_tpt440_dock,
.chained = true,
.chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST
},
@@ -5118,8 +5125,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
- SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
+ SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
+ SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
+ SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
+ SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -5223,6 +5233,7 @@ 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),
@@ -6454,6 +6465,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05fe, "Dell XPS 15", ALC668_FIXUP_DELL_XPS13),
SND_PCI_QUIRK(0x1028, 0x060a, "Dell XPS 13", ALC668_FIXUP_DELL_XPS13),
+ SND_PCI_QUIRK(0x1028, 0x060d, "Dell M3800", ALC668_FIXUP_DELL_XPS13),
SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0696, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 25f0f45e6640..b1bc66783974 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4522,7 +4522,11 @@ 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 c75995f2779c..b914a08258ea 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/adav80x.c b/sound/soc/codecs/adav80x.c
index 4373ada95648..3a91a00fb973 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -864,7 +864,6 @@ const struct regmap_config adav80x_regmap_config = {
.val_bits = 8,
.pad_bits = 1,
.reg_bits = 7,
- .read_flag_mask = 0x01,
.max_register = ADAV80X_PLL_OUTE,
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index eff4b4d512b7..ee91edcf3cb0 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1610,17 +1610,6 @@ int arizona_init_dai(struct arizona_priv *priv, int id)
}
EXPORT_SYMBOL_GPL(arizona_init_dai);
-static irqreturn_t arizona_fll_clock_ok(int irq, void *data)
-{
- struct arizona_fll *fll = data;
-
- arizona_fll_dbg(fll, "clock OK\n");
-
- complete(&fll->ok);
-
- return IRQ_HANDLED;
-}
-
static struct {
unsigned int min;
unsigned int max;
@@ -1902,17 +1891,18 @@ static int arizona_is_enabled_fll(struct arizona_fll *fll)
static int arizona_enable_fll(struct arizona_fll *fll)
{
struct arizona *arizona = fll->arizona;
- unsigned long time_left;
bool use_sync = false;
int already_enabled = arizona_is_enabled_fll(fll);
struct arizona_fll_cfg cfg;
+ int i;
+ unsigned int val;
if (already_enabled < 0)
return already_enabled;
if (already_enabled) {
/* Facilitate smooth refclk across the transition */
- regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x7,
+ regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9,
ARIZONA_FLL1_GAIN_MASK, 0);
regmap_update_bits_async(fll->arizona->regmap, fll->base + 1,
ARIZONA_FLL1_FREERUN,
@@ -1964,9 +1954,6 @@ static int arizona_enable_fll(struct arizona_fll *fll)
if (!already_enabled)
pm_runtime_get(arizona->dev);
- /* Clear any pending completions */
- try_wait_for_completion(&fll->ok);
-
regmap_update_bits_async(arizona->regmap, fll->base + 1,
ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA);
if (use_sync)
@@ -1978,10 +1965,24 @@ static int arizona_enable_fll(struct arizona_fll *fll)
regmap_update_bits_async(arizona->regmap, fll->base + 1,
ARIZONA_FLL1_FREERUN, 0);
- time_left = wait_for_completion_timeout(&fll->ok,
- msecs_to_jiffies(250));
- if (time_left == 0)
+ arizona_fll_dbg(fll, "Waiting for FLL lock...\n");
+ val = 0;
+ for (i = 0; i < 15; i++) {
+ if (i < 5)
+ usleep_range(200, 400);
+ else
+ msleep(20);
+
+ regmap_read(arizona->regmap,
+ ARIZONA_INTERRUPT_RAW_STATUS_5,
+ &val);
+ if (val & (ARIZONA_FLL1_CLOCK_OK_STS << (fll->id - 1)))
+ break;
+ }
+ if (i == 15)
arizona_fll_warn(fll, "Timed out waiting for lock\n");
+ else
+ arizona_fll_dbg(fll, "FLL locked (%d polls)\n", i);
return 0;
}
@@ -2066,11 +2067,8 @@ EXPORT_SYMBOL_GPL(arizona_set_fll);
int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq,
int ok_irq, struct arizona_fll *fll)
{
- int ret;
unsigned int val;
- init_completion(&fll->ok);
-
fll->id = id;
fll->base = base;
fll->arizona = arizona;
@@ -2092,13 +2090,6 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq,
snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name),
"FLL%d clock OK", id);
- ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name,
- arizona_fll_clock_ok, fll);
- if (ret != 0) {
- dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n",
- id, ret);
- }
-
regmap_update_bits(arizona->regmap, fll->base + 1,
ARIZONA_FLL1_FREERUN, 0);
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 11ff899b0272..14e8485b5585 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -233,7 +233,6 @@ struct arizona_fll {
int id;
unsigned int base;
unsigned int vco_mult;
- struct completion ok;
unsigned int fout;
int sync_src;
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 178e55d4d481..06317f7d945f 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -985,6 +985,35 @@ static int rt5640_hp_event(struct snd_soc_dapm_widget *w,
return 0;
}
+static int rt5640_lout_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ hp_amp_power_on(codec);
+ snd_soc_update_bits(codec, RT5640_PWR_ANLG1,
+ RT5640_PWR_LM, RT5640_PWR_LM);
+ snd_soc_update_bits(codec, RT5640_OUTPUT,
+ RT5640_L_MUTE | RT5640_R_MUTE, 0);
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_update_bits(codec, RT5640_OUTPUT,
+ RT5640_L_MUTE | RT5640_R_MUTE,
+ RT5640_L_MUTE | RT5640_R_MUTE);
+ snd_soc_update_bits(codec, RT5640_PWR_ANLG1,
+ RT5640_PWR_LM, 0);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -1180,13 +1209,16 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
0, rt5640_spo_l_mix, ARRAY_SIZE(rt5640_spo_l_mix)),
SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0,
0, rt5640_spo_r_mix, ARRAY_SIZE(rt5640_spo_r_mix)),
- SND_SOC_DAPM_MIXER("LOUT MIX", RT5640_PWR_ANLG1, RT5640_PWR_LM_BIT, 0,
+ SND_SOC_DAPM_MIXER("LOUT MIX", SND_SOC_NOPM, 0, 0,
rt5640_lout_mix, ARRAY_SIZE(rt5640_lout_mix)),
SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM,
0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0,
rt5640_hp_event,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PGA_S("LOUT amp", 1, SND_SOC_NOPM, 0, 0,
+ rt5640_lout_event,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1,
RT5640_PWR_HP_L_BIT, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1,
@@ -1501,8 +1533,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
{"HP R Playback", "Switch", "HP Amp"},
{"HPOL", NULL, "HP L Playback"},
{"HPOR", NULL, "HP R Playback"},
- {"LOUTL", NULL, "LOUT MIX"},
- {"LOUTR", NULL, "LOUT MIX"},
+
+ {"LOUT amp", NULL, "LOUT MIX"},
+ {"LOUTL", NULL, "LOUT amp"},
+ {"LOUTR", NULL, "LOUT amp"},
};
static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = {
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 3593a1496056..3a29c0ac5d8a 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1339,8 +1339,8 @@ 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:
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index a3e97b46b64e..0d28e3b356f6 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -131,10 +131,10 @@ 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));
}
}
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 39cea80846c3..f2bf8661dd21 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -1,7 +1,6 @@
config SND_PXA2XX_SOC
tristate "SoC Audio for the Intel PXA2xx chip"
depends on ARCH_PXA
- select SND_ARM
select SND_PXA2XX_LIB
help
Say Y or M if you want to add support for codecs attached to
@@ -25,7 +24,6 @@ config SND_PXA2XX_AC97
config SND_PXA2XX_SOC_AC97
tristate
select AC97_BUS
- select SND_ARM
select SND_PXA2XX_LIB_AC97
select SND_SOC_AC97_BUS
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 1f6054650991..9e4b04e0fbd1 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -49,7 +49,7 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
.reset = pxa2xx_ac97_cold_reset,
};
-static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12;
+static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11;
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
.addr = __PREG(PCDR),
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
@@ -57,7 +57,7 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
.filter_data = &pxa2xx_ac97_pcm_stereo_in_req,
};
-static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11;
+static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12;
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
.addr = __PREG(PCDR),
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c
index 8bf2e2c4bafb..9e371eb3e4fa 100644
--- a/sound/soc/samsung/arndale_rt5631.c
+++ b/sound/soc/samsung/arndale_rt5631.c
@@ -116,15 +116,6 @@ static int arndale_audio_probe(struct platform_device *pdev)
return ret;
}
-static int arndale_audio_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
-}
-
static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = {
{ .compatible = "samsung,arndale-rt5631", },
{ .compatible = "samsung,arndale-alc5631", },
@@ -139,7 +130,6 @@ static struct platform_driver arndale_audio_driver = {
.of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match),
},
.probe = arndale_audio_probe,
- .remove = arndale_audio_remove,
};
module_platform_driver(arndale_audio_driver);
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;
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 417ebb11cf48..bec63e0d2605 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -174,6 +174,8 @@ struct snd_usb_midi_in_endpoint {
u8 running_status_length;
} ports[0x10];
u8 seen_f5;
+ bool in_sysex;
+ u8 last_cin;
u8 error_resubmit;
int current_port;
};
@@ -468,6 +470,39 @@ static void snd_usbmidi_maudio_broken_running_status_input(
}
/*
+ * QinHeng CH345 is buggy: every second packet inside a SysEx has not CIN 4
+ * but the previously seen CIN, but still with three data bytes.
+ */
+static void ch345_broken_sysex_input(struct snd_usb_midi_in_endpoint *ep,
+ uint8_t *buffer, int buffer_length)
+{
+ unsigned int i, cin, length;
+
+ for (i = 0; i + 3 < buffer_length; i += 4) {
+ if (buffer[i] == 0 && i > 0)
+ break;
+ cin = buffer[i] & 0x0f;
+ if (ep->in_sysex &&
+ cin == ep->last_cin &&
+ (buffer[i + 1 + (cin == 0x6)] & 0x80) == 0)
+ cin = 0x4;
+#if 0
+ if (buffer[i + 1] == 0x90) {
+ /*
+ * Either a corrupted running status or a real note-on
+ * message; impossible to detect reliably.
+ */
+ }
+#endif
+ length = snd_usbmidi_cin_length[cin];
+ snd_usbmidi_input_data(ep, 0, &buffer[i + 1], length);
+ ep->in_sysex = cin == 0x4;
+ if (!ep->in_sysex)
+ ep->last_cin = cin;
+ }
+}
+
+/*
* CME protocol: like the standard protocol, but SysEx commands are sent as a
* single USB packet preceded by a 0x0F byte.
*/
@@ -660,6 +695,12 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = {
.output_packet = snd_usbmidi_output_standard_packet,
};
+static struct usb_protocol_ops snd_usbmidi_ch345_broken_sysex_ops = {
+ .input = ch345_broken_sysex_input,
+ .output = snd_usbmidi_standard_output,
+ .output_packet = snd_usbmidi_output_standard_packet,
+};
+
/*
* AKAI MPD16 protocol:
*
@@ -1341,6 +1382,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi,
* Various chips declare a packet size larger than 4 bytes, but
* do not actually work with larger packets:
*/
+ case USB_ID(0x0a67, 0x5011): /* Medeli DD305 */
case USB_ID(0x0a92, 0x1020): /* ESI M4U */
case USB_ID(0x1430, 0x474b): /* RedOctane GH MIDI INTERFACE */
case USB_ID(0x15ca, 0x0101): /* Textech USB Midi Cable */
@@ -2375,6 +2417,10 @@ int snd_usbmidi_create(struct snd_card *card,
err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
break;
+ case QUIRK_MIDI_CH345:
+ umidi->usb_protocol_ops = &snd_usbmidi_ch345_broken_sysex_ops;
+ err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
+ break;
default:
dev_err(&umidi->dev->dev, "invalid quirk type %d\n",
quirk->type);
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 8b7e391dd0b8..cd8ed2e393a2 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -2522,7 +2522,7 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list)
for (c = 0; c < MAX_CHANNELS; c++) {
if (!(cval->cmask & (1 << c)))
continue;
- if (cval->cached & (1 << c)) {
+ if (cval->cached & (1 << (c + 1))) {
err = snd_usb_set_cur_mix_value(cval, c + 1, idx,
cval->cache_val[idx]);
if (err < 0)
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index e4756651a52c..ecc2a4ea014d 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2820,6 +2820,17 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.idProduct = 0x1020,
},
+/* QinHeng devices */
+{
+ USB_DEVICE(0x1a86, 0x752d),
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .vendor_name = "QinHeng",
+ .product_name = "CH345",
+ .ifnum = 1,
+ .type = QUIRK_MIDI_CH345
+ }
+},
+
/* KeithMcMillen Stringport */
{
USB_DEVICE(0x1f38, 0x0001),
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 00ebc0ca008e..eef9b8e4b949 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -535,6 +535,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
[QUIRK_MIDI_CME] = create_any_midi_quirk,
[QUIRK_MIDI_AKAI] = create_any_midi_quirk,
[QUIRK_MIDI_FTDI] = create_any_midi_quirk,
+ [QUIRK_MIDI_CH345] = create_any_midi_quirk,
[QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk,
[QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk,
[QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk,
@@ -1271,6 +1272,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
case USB_ID(0x20b1, 0x000a): /* Gustard DAC-X20U */
case USB_ID(0x20b1, 0x2009): /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */
case USB_ID(0x20b1, 0x2023): /* JLsounds I2SoverUSB */
+ case USB_ID(0x20b1, 0x3023): /* Aune X1S 32BIT/384 DSD DAC */
if (fp->altsetting == 3)
return SNDRV_PCM_FMTBIT_DSD_U32_BE;
break;
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 91d0380431b4..991aa84491cd 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -94,6 +94,7 @@ enum quirk_type {
QUIRK_MIDI_AKAI,
QUIRK_MIDI_US122L,
QUIRK_MIDI_FTDI,
+ QUIRK_MIDI_CH345,
QUIRK_AUDIO_STANDARD_INTERFACE,
QUIRK_AUDIO_FIXED_ENDPOINT,
QUIRK_AUDIO_EDIROL_UAXX,
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-17 07:12:48 +0000