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-rw-r--r--sound/firewire/dice/dice.c4
-rw-r--r--sound/pci/hda/hda_intel.c7
-rw-r--r--sound/pci/hda/patch_conexant.c5
-rw-r--r--sound/pci/hda/patch_hdmi.c9
-rw-r--r--sound/pci/hda/patch_realtek.c23
-rw-r--r--sound/pci/hda/patch_sigmatel.c45
-rw-r--r--sound/soc/codecs/arizona.c16
-rw-r--r--sound/soc/codecs/es8328.c16
-rw-r--r--sound/soc/codecs/nau8825.c31
-rw-r--r--sound/soc/codecs/rl6231.c6
-rw-r--r--sound/soc/codecs/rt5645.c61
-rw-r--r--sound/soc/codecs/rt5670.h12
-rw-r--r--sound/soc/codecs/rt5677.c100
-rw-r--r--sound/soc/codecs/wm8960.c2
-rw-r--r--sound/soc/codecs/wm8962.c4
-rw-r--r--sound/soc/davinci/davinci-mcasp.c12
-rw-r--r--sound/soc/fsl/Kconfig2
-rw-r--r--sound/soc/fsl/fsl_sai.c3
-rw-r--r--sound/soc/intel/Kconfig2
-rw-r--r--sound/soc/intel/skylake/skl-topology.c1
-rw-r--r--sound/soc/rockchip/rockchip_spdif.c2
-rw-r--r--sound/soc/rockchip/rockchip_spdif.h6
-rw-r--r--sound/soc/sh/rcar/gen.c2
-rw-r--r--sound/soc/sh/rcar/src.c7
-rw-r--r--sound/soc/soc-core.c6
-rw-r--r--sound/soc/soc-dapm.c7
-rw-r--r--sound/soc/soc-ops.c2
-rw-r--r--sound/soc/soc-topology.c3
-rw-r--r--sound/soc/sti/uniperif_player.c9
-rw-r--r--sound/soc/sti/uniperif_reader.c3
-rw-r--r--sound/soc/sunxi/sun4i-codec.c27
-rw-r--r--sound/usb/midi.c46
-rw-r--r--sound/usb/quirks-table.h11
-rw-r--r--sound/usb/quirks.c1
-rw-r--r--sound/usb/usbaudio.h1
35 files changed, 370 insertions, 124 deletions
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 5d99436dfcae..0cda05c72f50 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -12,9 +12,11 @@ MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
MODULE_LICENSE("GPL v2");
#define OUI_WEISS 0x001c6a
+#define OUI_LOUD 0x000ff2
#define DICE_CATEGORY_ID 0x04
#define WEISS_CATEGORY_ID 0x00
+#define LOUD_CATEGORY_ID 0x10
static int dice_interface_check(struct fw_unit *unit)
{
@@ -57,6 +59,8 @@ static int dice_interface_check(struct fw_unit *unit)
}
if (vendor == OUI_WEISS)
category = WEISS_CATEGORY_ID;
+ else if (vendor == OUI_LOUD)
+ category = LOUD_CATEGORY_ID;
else
category = DICE_CATEGORY_ID;
if (device->config_rom[3] != ((vendor << 8) | category) ||
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 8a7fbdcb4072..963f82430938 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -312,6 +312,10 @@ enum {
(AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\
AZX_DCAPS_I915_POWERWELL)
+#define AZX_DCAPS_INTEL_BROXTON \
+ (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\
+ AZX_DCAPS_I915_POWERWELL)
+
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\
@@ -2124,6 +2128,9 @@ static const struct pci_device_id azx_ids[] = {
/* Sunrise Point-LP */
{ PCI_DEVICE(0x8086, 0x9d70),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE },
+ /* Broxton-P(Apollolake) */
+ { PCI_DEVICE(0x8086, 0x5a98),
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON },
/* Haswell */
{ PCI_DEVICE(0x8086, 0x0a0c),
.driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL },
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index c8b8ef5246a6..ef198903c0c3 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -955,6 +955,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
*/
static const struct hda_device_id snd_hda_id_conexant[] = {
+ HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto),
@@ -972,9 +973,9 @@ static const struct hda_device_id snd_hda_id_conexant[] = {
HDA_CODEC_ENTRY(0x14f150ac, "CX20652", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f150b8, "CX20664", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f150b9, "CX20665", patch_conexant_auto),
- HDA_CODEC_ENTRY(0x14f150f1, "CX20721", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150f1, "CX21722", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f150f2, "CX20722", patch_conexant_auto),
- HDA_CODEC_ENTRY(0x14f150f3, "CX20723", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150f3, "CX21724", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f150f4, "CX20724", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f1510f, "CX20751/2", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f15110, "CX20751/2", patch_conexant_auto),
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 60cd9e700909..4b6fb668c91c 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2352,6 +2352,12 @@ static void intel_pin_eld_notify(void *audio_ptr, int port)
struct hda_codec *codec = audio_ptr;
int pin_nid = port + 0x04;
+ /* skip notification during system suspend (but not in runtime PM);
+ * the state will be updated at resume
+ */
+ if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0)
+ return;
+
check_presence_and_report(codec, pin_nid);
}
@@ -2378,7 +2384,8 @@ static int patch_generic_hdmi(struct hda_codec *codec)
* can cover the codec power request, and so need not set this flag.
* For previous platforms, there is no such power well feature.
*/
- if (is_valleyview_plus(codec) || is_skylake(codec))
+ if (is_valleyview_plus(codec) || is_skylake(codec) ||
+ is_broxton(codec))
codec->core.link_power_control = 1;
if (is_haswell_plus(codec) || is_valleyview_plus(codec)) {
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2f7b065f9ac4..9bedf7c85e29 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1759,6 +1759,7 @@ enum {
ALC882_FIXUP_NO_PRIMARY_HP,
ALC887_FIXUP_ASUS_BASS,
ALC887_FIXUP_BASS_CHMAP,
+ ALC882_FIXUP_DISABLE_AAMIX,
};
static void alc889_fixup_coef(struct hda_codec *codec,
@@ -1920,6 +1921,8 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec,
static void alc_fixup_bass_chmap(struct hda_codec *codec,
const struct hda_fixup *fix, int action);
+static void alc_fixup_disable_aamix(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action);
static const struct hda_fixup alc882_fixups[] = {
[ALC882_FIXUP_ABIT_AW9D_MAX] = {
@@ -2151,6 +2154,10 @@ static const struct hda_fixup alc882_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_bass_chmap,
},
+ [ALC882_FIXUP_DISABLE_AAMIX] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_disable_aamix,
+ },
};
static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -2218,6 +2225,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1458, 0xa182, "Gigabyte Z170X-UD3", ALC882_FIXUP_DISABLE_AAMIX),
SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD),
SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD),
@@ -4587,6 +4595,7 @@ enum {
ALC292_FIXUP_DISABLE_AAMIX,
ALC298_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC275_FIXUP_DELL_XPS,
+ ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -5167,6 +5176,17 @@ static const struct hda_fixup alc269_fixups[] = {
{}
}
},
+ [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* Disable pass-through path for FRONT 14h */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x36},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x1737},
+ {}
+ },
+ .chained = true,
+ .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5180,8 +5200,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
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, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
+ SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X),
SND_PCI_QUIRK(0x1028, 0x05ca, "Dell Latitude E7240", ALC292_FIXUP_DELL_E7X),
SND_PCI_QUIRK(0x1028, 0x05cb, "Dell Latitude E7440", ALC292_FIXUP_DELL_E7X),
SND_PCI_QUIRK(0x1028, 0x05da, "Dell Vostro 5460", ALC290_FIXUP_SUBWOOFER),
@@ -5204,6 +5226,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
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, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE),
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),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 826122d8acee..2c7c5eb8b1e9 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3110,6 +3110,29 @@ static void stac92hd71bxx_fixup_hp_hdx(struct hda_codec *codec,
spec->gpio_led = 0x08;
}
+static bool is_hp_output(struct hda_codec *codec, hda_nid_t pin)
+{
+ unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, pin);
+
+ /* count line-out, too, as BIOS sets often so */
+ return get_defcfg_connect(pin_cfg) != AC_JACK_PORT_NONE &&
+ (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT ||
+ get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT);
+}
+
+static void fixup_hp_headphone(struct hda_codec *codec, hda_nid_t pin)
+{
+ unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, pin);
+
+ /* It was changed in the BIOS to just satisfy MS DTM.
+ * Lets turn it back into slaved HP
+ */
+ pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE)) |
+ (AC_JACK_HP_OUT << AC_DEFCFG_DEVICE_SHIFT);
+ pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC | AC_DEFCFG_SEQUENCE))) |
+ 0x1f;
+ snd_hda_codec_set_pincfg(codec, pin, pin_cfg);
+}
static void stac92hd71bxx_fixup_hp(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
@@ -3119,22 +3142,12 @@ static void stac92hd71bxx_fixup_hp(struct hda_codec *codec,
if (action != HDA_FIXUP_ACT_PRE_PROBE)
return;
- if (hp_blike_system(codec->core.subsystem_id)) {
- unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f);
- if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT ||
- get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER ||
- get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT) {
- /* It was changed in the BIOS to just satisfy MS DTM.
- * Lets turn it back into slaved HP
- */
- pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE))
- | (AC_JACK_HP_OUT <<
- AC_DEFCFG_DEVICE_SHIFT);
- pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC
- | AC_DEFCFG_SEQUENCE)))
- | 0x1f;
- snd_hda_codec_set_pincfg(codec, 0x0f, pin_cfg);
- }
+ /* when both output A and F are assigned, these are supposedly
+ * dock and built-in headphones; fix both pin configs
+ */
+ if (is_hp_output(codec, 0x0a) && is_hp_output(codec, 0x0f)) {
+ fixup_hp_headphone(codec, 0x0a);
+ fixup_hp_headphone(codec, 0x0f);
}
if (find_mute_led_cfg(codec, 1))
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 9929efc6b9aa..b3ea24d64c50 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1023,24 +1023,18 @@ void arizona_init_dvfs(struct arizona_priv *priv)
}
EXPORT_SYMBOL_GPL(arizona_init_dvfs);
-static unsigned int arizona_sysclk_48k_rates[] = {
+static unsigned int arizona_opclk_ref_48k_rates[] = {
6144000,
12288000,
24576000,
49152000,
- 73728000,
- 98304000,
- 147456000,
};
-static unsigned int arizona_sysclk_44k1_rates[] = {
+static unsigned int arizona_opclk_ref_44k1_rates[] = {
5644800,
11289600,
22579200,
45158400,
- 67737600,
- 90316800,
- 135475200,
};
static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk,
@@ -1065,11 +1059,11 @@ static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk,
}
if (refclk % 8000)
- rates = arizona_sysclk_44k1_rates;
+ rates = arizona_opclk_ref_44k1_rates;
else
- rates = arizona_sysclk_48k_rates;
+ rates = arizona_opclk_ref_48k_rates;
- for (ref = 0; ref < ARRAY_SIZE(arizona_sysclk_48k_rates) &&
+ for (ref = 0; ref < ARRAY_SIZE(arizona_opclk_ref_48k_rates) &&
rates[ref] <= refclk; ref++) {
div = 1;
while (rates[ref] / div >= freq && div < 32) {
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 969e337dc17c..84f5eb07a91b 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -205,18 +205,18 @@ static const struct snd_kcontrol_new es8328_right_line_controls =
/* Left Mixer */
static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
- SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 8, 1, 0),
- SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 7, 1, 0),
- SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 8, 1, 0),
- SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 7, 1, 0),
+ SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 7, 1, 0),
+ SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 6, 1, 0),
+ SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 7, 1, 0),
+ SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 6, 1, 0),
};
/* Right Mixer */
static const struct snd_kcontrol_new es8328_right_mixer_controls[] = {
- SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 8, 1, 0),
- SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 7, 1, 0),
- SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 8, 1, 0),
- SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 7, 1, 0),
+ SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 7, 1, 0),
+ SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 6, 1, 0),
+ SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 7, 1, 0),
+ SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 6, 1, 0),
};
static const char * const es8328_pga_sel[] = {
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
index 7fc7b4e3f444..c1b87c5800b1 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -1271,6 +1271,36 @@ static int nau8825_i2c_remove(struct i2c_client *client)
return 0;
}
+#ifdef CONFIG_PM_SLEEP
+static int nau8825_suspend(struct device *dev)
+{
+ struct i2c_client *client = to_i2c_client(dev);
+ struct nau8825 *nau8825 = dev_get_drvdata(dev);
+
+ disable_irq(client->irq);
+ regcache_cache_only(nau8825->regmap, true);
+ regcache_mark_dirty(nau8825->regmap);
+
+ return 0;
+}
+
+static int nau8825_resume(struct device *dev)
+{
+ struct i2c_client *client = to_i2c_client(dev);
+ struct nau8825 *nau8825 = dev_get_drvdata(dev);
+
+ regcache_cache_only(nau8825->regmap, false);
+ regcache_sync(nau8825->regmap);
+ enable_irq(client->irq);
+
+ return 0;
+}
+#endif
+
+static const struct dev_pm_ops nau8825_pm = {
+ SET_SYSTEM_SLEEP_PM_OPS(nau8825_suspend, nau8825_resume)
+};
+
static const struct i2c_device_id nau8825_i2c_ids[] = {
{ "nau8825", 0 },
{ }
@@ -1297,6 +1327,7 @@ static struct i2c_driver nau8825_driver = {
.name = "nau8825",
.of_match_table = of_match_ptr(nau8825_of_ids),
.acpi_match_table = ACPI_PTR(nau8825_acpi_match),
+ .pm = &nau8825_pm,
},
.probe = nau8825_i2c_probe,
.remove = nau8825_i2c_remove,
diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c
index aca479fa7670..1dc68ab08a17 100644
--- a/sound/soc/codecs/rl6231.c
+++ b/sound/soc/codecs/rl6231.c
@@ -80,8 +80,10 @@ int rl6231_calc_dmic_clk(int rate)
}
for (i = 0; i < ARRAY_SIZE(div); i++) {
- /* find divider that gives DMIC frequency below 3MHz */
- if (3000000 * div[i] >= rate)
+ if ((div[i] % 3) == 0)
+ continue;
+ /* find divider that gives DMIC frequency below 3.072MHz */
+ if (3072000 * div[i] >= rate)
return i;
}
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 28132375e427..ef76940f9dcb 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -245,7 +245,7 @@ struct rt5645_priv {
struct snd_soc_jack *hp_jack;
struct snd_soc_jack *mic_jack;
struct snd_soc_jack *btn_jack;
- struct delayed_work jack_detect_work;
+ struct delayed_work jack_detect_work, rcclock_work;
struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)];
struct rt5645_eq_param_s *eq_param;
@@ -565,12 +565,33 @@ static int rt5645_hweq_put(struct snd_kcontrol *kcontrol,
.put = rt5645_hweq_put \
}
+static int rt5645_spk_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+ struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component);
+ int ret;
+
+ cancel_delayed_work_sync(&rt5645->rcclock_work);
+
+ regmap_update_bits(rt5645->regmap, RT5645_MICBIAS,
+ RT5645_PWR_CLK25M_MASK, RT5645_PWR_CLK25M_PU);
+
+ ret = snd_soc_put_volsw(kcontrol, ucontrol);
+
+ queue_delayed_work(system_power_efficient_wq, &rt5645->rcclock_work,
+ msecs_to_jiffies(200));
+
+ return ret;
+}
+
static const struct snd_kcontrol_new rt5645_snd_controls[] = {
/* Speaker Output Volume */
SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL,
RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1),
- SOC_DOUBLE_TLV("Speaker Playback Volume", RT5645_SPK_VOL,
- RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv),
+ SOC_DOUBLE_EXT_TLV("Speaker Playback Volume", RT5645_SPK_VOL,
+ RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, snd_soc_get_volsw,
+ rt5645_spk_put_volsw, out_vol_tlv),
/* ClassD modulator Speaker Gain Ratio */
SOC_SINGLE_TLV("Speaker ClassD Playback Volume", RT5645_SPO_CLSD_RATIO,
@@ -1498,7 +1519,7 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on)
regmap_write(rt5645->regmap, RT5645_PR_BASE +
RT5645_MAMP_INT_REG2, 0xfc00);
snd_soc_write(codec, RT5645_DEPOP_M2, 0x1140);
- msleep(40);
+ msleep(70);
rt5645->hp_on = true;
} else {
/* depop parameters */
@@ -3122,6 +3143,15 @@ static void rt5645_jack_detect_work(struct work_struct *work)
SND_JACK_BTN_2 | SND_JACK_BTN_3);
}
+static void rt5645_rcclock_work(struct work_struct *work)
+{
+ struct rt5645_priv *rt5645 =
+ container_of(work, struct rt5645_priv, rcclock_work.work);
+
+ regmap_update_bits(rt5645->regmap, RT5645_MICBIAS,
+ RT5645_PWR_CLK25M_MASK, RT5645_PWR_CLK25M_PD);
+}
+
static irqreturn_t rt5645_irq(int irq, void *data)
{
struct rt5645_priv *rt5645 = data;
@@ -3348,6 +3378,27 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = {
DMI_MATCH(DMI_PRODUCT_NAME, "Reks"),
},
},
+ {
+ .ident = "Google Edgar",
+ .callback = strago_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_NAME, "Edgar"),
+ },
+ },
+ {
+ .ident = "Google Wizpig",
+ .callback = strago_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_NAME, "Wizpig"),
+ },
+ },
+ {
+ .ident = "Google Terra",
+ .callback = strago_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_NAME, "Terra"),
+ },
+ },
{ }
};
@@ -3587,6 +3638,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
}
INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work);
+ INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work);
if (rt5645->i2c->irq) {
ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq,
@@ -3621,6 +3673,7 @@ static int rt5645_i2c_remove(struct i2c_client *i2c)
free_irq(i2c->irq, rt5645);
cancel_delayed_work_sync(&rt5645->jack_detect_work);
+ cancel_delayed_work_sync(&rt5645->rcclock_work);
snd_soc_unregister_codec(&i2c->dev);
regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies);
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index dc2b46236c5c..3f1b0f1df809 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -973,12 +973,12 @@
#define RT5670_SCLK_SRC_MCLK (0x0 << 14)
#define RT5670_SCLK_SRC_PLL1 (0x1 << 14)
#define RT5670_SCLK_SRC_RCCLK (0x2 << 14) /* 15MHz */
-#define RT5670_PLL1_SRC_MASK (0x3 << 12)
-#define RT5670_PLL1_SRC_SFT 12
-#define RT5670_PLL1_SRC_MCLK (0x0 << 12)
-#define RT5670_PLL1_SRC_BCLK1 (0x1 << 12)
-#define RT5670_PLL1_SRC_BCLK2 (0x2 << 12)
-#define RT5670_PLL1_SRC_BCLK3 (0x3 << 12)
+#define RT5670_PLL1_SRC_MASK (0x7 << 11)
+#define RT5670_PLL1_SRC_SFT 11
+#define RT5670_PLL1_SRC_MCLK (0x0 << 11)
+#define RT5670_PLL1_SRC_BCLK1 (0x1 << 11)
+#define RT5670_PLL1_SRC_BCLK2 (0x2 << 11)
+#define RT5670_PLL1_SRC_BCLK3 (0x3 << 11)
#define RT5670_PLL1_PD_MASK (0x1 << 3)
#define RT5670_PLL1_PD_SFT 3
#define RT5670_PLL1_PD_1 (0x0 << 3)
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index b4cd7e3bf5f8..69d987a9935c 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -1386,90 +1386,90 @@ static const struct snd_kcontrol_new rt5677_dac_r_mix[] = {
};
static const struct snd_kcontrol_new rt5677_sto1_dac_l_mix[] = {
- SOC_DAPM_SINGLE("ST L Switch", RT5677_STO1_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("ST L Switch", RT5677_STO1_DAC_MIXER,
RT5677_M_ST_DAC1_L_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER,
RT5677_M_DAC1_L_STO_L_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_STO1_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_STO1_DAC_MIXER,
RT5677_M_DAC2_L_STO_L_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER,
RT5677_M_DAC1_R_STO_L_SFT, 1, 1),
};
static const struct snd_kcontrol_new rt5677_sto1_dac_r_mix[] = {
- SOC_DAPM_SINGLE("ST R Switch", RT5677_STO1_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("ST R Switch", RT5677_STO1_DAC_MIXER,
RT5677_M_ST_DAC1_R_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER,
RT5677_M_DAC1_R_STO_R_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_STO1_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_STO1_DAC_MIXER,
RT5677_M_DAC2_R_STO_R_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER,
RT5677_M_DAC1_L_STO_R_SFT, 1, 1),
};
static const struct snd_kcontrol_new rt5677_mono_dac_l_mix[] = {
- SOC_DAPM_SINGLE("ST L Switch", RT5677_MONO_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("ST L Switch", RT5677_MONO_DAC_MIXER,
RT5677_M_ST_DAC2_L_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_MONO_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_MONO_DAC_MIXER,
RT5677_M_DAC1_L_MONO_L_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER,
RT5677_M_DAC2_L_MONO_L_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER,
RT5677_M_DAC2_R_MONO_L_SFT, 1, 1),
};
static const struct snd_kcontrol_new rt5677_mono_dac_r_mix[] = {
- SOC_DAPM_SINGLE("ST R Switch", RT5677_MONO_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("ST R Switch", RT5677_MONO_DAC_MIXER,
RT5677_M_ST_DAC2_R_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_MONO_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_MONO_DAC_MIXER,
RT5677_M_DAC1_R_MONO_R_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER,
RT5677_M_DAC2_R_MONO_R_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER,
RT5677_M_DAC2_L_MONO_R_SFT, 1, 1),
};
static const struct snd_kcontrol_new rt5677_dd1_l_mix[] = {
- SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5677_DD1_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix L Switch", RT5677_DD1_MIXER,
RT5677_M_STO_L_DD1_L_SFT, 1, 1),
- SOC_DAPM_SINGLE("Mono DAC Mix L Switch", RT5677_DD1_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix L Switch", RT5677_DD1_MIXER,
RT5677_M_MONO_L_DD1_L_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC3 L Switch", RT5677_DD1_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC3 L Switch", RT5677_DD1_MIXER,
RT5677_M_DAC3_L_DD1_L_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC3 R Switch", RT5677_DD1_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC3 R Switch", RT5677_DD1_MIXER,
RT5677_M_DAC3_R_DD1_L_SFT, 1, 1),
};
static const struct snd_kcontrol_new rt5677_dd1_r_mix[] = {
- SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5677_DD1_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix R Switch", RT5677_DD1_MIXER,
RT5677_M_STO_R_DD1_R_SFT, 1, 1),
- SOC_DAPM_SINGLE("Mono DAC Mix R Switch", RT5677_DD1_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix R Switch", RT5677_DD1_MIXER,
RT5677_M_MONO_R_DD1_R_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC3 R Switch", RT5677_DD1_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC3 R Switch", RT5677_DD1_MIXER,
RT5677_M_DAC3_R_DD1_R_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC3 L Switch", RT5677_DD1_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC3 L Switch", RT5677_DD1_MIXER,
RT5677_M_DAC3_L_DD1_R_SFT, 1, 1),
};
static const struct snd_kcontrol_new rt5677_dd2_l_mix[] = {
- SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5677_DD2_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix L Switch", RT5677_DD2_MIXER,
RT5677_M_STO_L_DD2_L_SFT, 1, 1),
- SOC_DAPM_SINGLE("Mono DAC Mix L Switch", RT5677_DD2_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix L Switch", RT5677_DD2_MIXER,
RT5677_M_MONO_L_DD2_L_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC4 L Switch", RT5677_DD2_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC4 L Switch", RT5677_DD2_MIXER,
RT5677_M_DAC4_L_DD2_L_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC4 R Switch", RT5677_DD2_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC4 R Switch", RT5677_DD2_MIXER,
RT5677_M_DAC4_R_DD2_L_SFT, 1, 1),
};
static const struct snd_kcontrol_new rt5677_dd2_r_mix[] = {
- SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5677_DD2_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix R Switch", RT5677_DD2_MIXER,
RT5677_M_STO_R_DD2_R_SFT, 1, 1),
- SOC_DAPM_SINGLE("Mono DAC Mix R Switch", RT5677_DD2_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix R Switch", RT5677_DD2_MIXER,
RT5677_M_MONO_R_DD2_R_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC4 R Switch", RT5677_DD2_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC4 R Switch", RT5677_DD2_MIXER,
RT5677_M_DAC4_R_DD2_R_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC4 L Switch", RT5677_DD2_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC4 L Switch", RT5677_DD2_MIXER,
RT5677_M_DAC4_L_DD2_R_SFT, 1, 1),
};
@@ -2596,6 +2596,21 @@ static int rt5677_vref_event(struct snd_soc_dapm_widget *w,
return 0;
}
+static int rt5677_filter_power_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ msleep(50);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT,
0, rt5677_set_pll1_event, SND_SOC_DAPM_PRE_PMU |
@@ -3072,19 +3087,26 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
/* DAC Mixer */
SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5677_PWR_DIG2,
- RT5677_PWR_DAC_S1F_BIT, 0, NULL, 0),
+ RT5677_PWR_DAC_S1F_BIT, 0, rt5677_filter_power_event,
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("dac mono2 left filter", RT5677_PWR_DIG2,
- RT5677_PWR_DAC_M2F_L_BIT, 0, NULL, 0),
+ RT5677_PWR_DAC_M2F_L_BIT, 0, rt5677_filter_power_event,
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("dac mono2 right filter", RT5677_PWR_DIG2,
- RT5677_PWR_DAC_M2F_R_BIT, 0, NULL, 0),
+ RT5677_PWR_DAC_M2F_R_BIT, 0, rt5677_filter_power_event,
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("dac mono3 left filter", RT5677_PWR_DIG2,
- RT5677_PWR_DAC_M3F_L_BIT, 0, NULL, 0),
+ RT5677_PWR_DAC_M3F_L_BIT, 0, rt5677_filter_power_event,
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("dac mono3 right filter", RT5677_PWR_DIG2,
- RT5677_PWR_DAC_M3F_R_BIT, 0, NULL, 0),
+ RT5677_PWR_DAC_M3F_R_BIT, 0, rt5677_filter_power_event,
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("dac mono4 left filter", RT5677_PWR_DIG2,
- RT5677_PWR_DAC_M4F_L_BIT, 0, NULL, 0),
+ RT5677_PWR_DAC_M4F_L_BIT, 0, rt5677_filter_power_event,
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("dac mono4 right filter", RT5677_PWR_DIG2,
- RT5677_PWR_DAC_M4F_R_BIT, 0, NULL, 0),
+ RT5677_PWR_DAC_M4F_R_BIT, 0, rt5677_filter_power_event,
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0,
rt5677_sto1_dac_l_mix, ARRAY_SIZE(rt5677_sto1_dac_l_mix)),
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 056375339ea3..5380798883b5 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -229,7 +229,7 @@ SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
6, 1, 0),
SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
- 7, 1, 0),
+ 7, 1, 1),
SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv),
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 39ebd7bf4f53..a7e79784fc16 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -365,8 +365,8 @@ static const struct reg_default wm8962_reg[] = {
{ 16924, 0x0059 }, /* R16924 - HDBASS_PG_1 */
{ 16925, 0x999A }, /* R16925 - HDBASS_PG_0 */
- { 17048, 0x0083 }, /* R17408 - HPF_C_1 */
- { 17049, 0x98AD }, /* R17409 - HPF_C_0 */
+ { 17408, 0x0083 }, /* R17408 - HPF_C_1 */
+ { 17409, 0x98AD }, /* R17409 - HPF_C_0 */
{ 17920, 0x007F }, /* R17920 - ADCL_RETUNE_C1_1 */
{ 17921, 0xFFFF }, /* R17921 - ADCL_RETUNE_C1_0 */
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 4495a40a9468..c1c9c2e3525b 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -681,8 +681,8 @@ static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai,
}
mcasp->tdm_slots = slots;
- mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = rx_mask;
- mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = tx_mask;
+ mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask;
+ mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = rx_mask;
mcasp->slot_width = slot_width;
return davinci_mcasp_set_ch_constraints(mcasp);
@@ -908,6 +908,14 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream,
mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
FSRMOD(total_slots), FSRMOD(0x1FF));
+ /*
+ * If McASP is set to be TX/RX synchronous and the playback is
+ * not running already we need to configure the TX slots in
+ * order to have correct FSX on the bus
+ */
+ if (mcasp_is_synchronous(mcasp) && !mcasp->channels)
+ mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
+ FSXMOD(total_slots), FSXMOD(0x1FF));
}
return 0;
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 19c302b0d763..14dfdee05fd5 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -283,6 +283,8 @@ config SND_SOC_IMX_MC13783
config SND_SOC_FSL_ASOC_CARD
tristate "Generic ASoC Sound Card with ASRC support"
depends on OF && I2C
+ # enforce SND_SOC_FSL_ASOC_CARD=m if SND_AC97_CODEC=m:
+ depends on SND_AC97_CODEC || SND_AC97_CODEC=n
select SND_SOC_IMX_AUDMUX
select SND_SOC_IMX_PCM_DMA
select SND_SOC_FSL_ESAI
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index a4435f5e3be9..ffd5f9acc849 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -454,7 +454,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
* Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx.
* Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx.
*/
- regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0);
+ regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC,
+ sai->synchronous[TX] ? FSL_SAI_CR2_SYNC : 0);
regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC,
sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0);
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 7b778ab85f8b..d430ef5a4f38 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -144,7 +144,7 @@ config SND_SOC_INTEL_SKYLAKE
config SND_SOC_INTEL_SKL_RT286_MACH
tristate "ASoC Audio driver for SKL with RT286 I2S mode"
- depends on X86 && ACPI
+ depends on X86 && ACPI && I2C
select SND_SOC_INTEL_SST
select SND_SOC_INTEL_SKYLAKE
select SND_SOC_RT286
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index a7854c8fc523..ffea427aeca8 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -1240,6 +1240,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus)
*/
ret = snd_soc_tplg_component_load(&platform->component,
&skl_tplg_ops, fw, 0);
+ release_firmware(fw);
if (ret < 0) {
dev_err(bus->dev, "tplg component load failed%d\n", ret);
return -EINVAL;
diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c
index a38a3029062c..ac72ff5055bb 100644
--- a/sound/soc/rockchip/rockchip_spdif.c
+++ b/sound/soc/rockchip/rockchip_spdif.c
@@ -280,7 +280,7 @@ static int rk_spdif_probe(struct platform_device *pdev)
int ret;
match = of_match_node(rk_spdif_match, np);
- if ((int) match->data == RK_SPDIF_RK3288) {
+ if (match->data == (void *)RK_SPDIF_RK3288) {
struct regmap *grf;
grf = syscon_regmap_lookup_by_phandle(np, "rockchip,grf");
diff --git a/sound/soc/rockchip/rockchip_spdif.h b/sound/soc/rockchip/rockchip_spdif.h
index 07f86a21046a..921b4095fb92 100644
--- a/sound/soc/rockchip/rockchip_spdif.h
+++ b/sound/soc/rockchip/rockchip_spdif.h
@@ -28,9 +28,9 @@
#define SPDIF_CFGR_VDW(x) (x << SPDIF_CFGR_VDW_SHIFT)
#define SDPIF_CFGR_VDW_MASK (0xf << SPDIF_CFGR_VDW_SHIFT)
-#define SPDIF_CFGR_VDW_16 SPDIF_CFGR_VDW(0x00)
-#define SPDIF_CFGR_VDW_20 SPDIF_CFGR_VDW(0x01)
-#define SPDIF_CFGR_VDW_24 SPDIF_CFGR_VDW(0x10)
+#define SPDIF_CFGR_VDW_16 SPDIF_CFGR_VDW(0x0)
+#define SPDIF_CFGR_VDW_20 SPDIF_CFGR_VDW(0x1)
+#define SPDIF_CFGR_VDW_24 SPDIF_CFGR_VDW(0x2)
/*
* DMACR
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 76da7620904c..edcf4cc2e84f 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -235,7 +235,7 @@ static int rsnd_gen2_probe(struct platform_device *pdev,
RSND_GEN_S_REG(SCU_SYS_STATUS0, 0x1c8),
RSND_GEN_S_REG(SCU_SYS_INT_EN0, 0x1cc),
RSND_GEN_S_REG(SCU_SYS_STATUS1, 0x1d0),
- RSND_GEN_S_REG(SCU_SYS_INT_EN1, 0x1c4),
+ RSND_GEN_S_REG(SCU_SYS_INT_EN1, 0x1d4),
RSND_GEN_M_REG(SRC_SWRSR, 0x200, 0x40),
RSND_GEN_M_REG(SRC_SRCIR, 0x204, 0x40),
RSND_GEN_M_REG(SRC_ADINR, 0x214, 0x40),
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 261b50217c48..68b439ed22d7 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -923,6 +923,7 @@ static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod,
struct snd_soc_pcm_runtime *rtd)
{
struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
+ struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io);
struct rsnd_src *src = rsnd_mod_to_src(mod);
int ret;
@@ -937,6 +938,12 @@ static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod,
return 0;
/*
+ * SRC In doesn't work if DVC was enabled
+ */
+ if (dvc && !rsnd_io_is_play(io))
+ return 0;
+
+ /*
* enable sync convert
*/
ret = rsnd_kctrl_new_s(mod, io, rtd,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 24b096066a07..a1305f827a98 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -795,12 +795,12 @@ static void soc_resume_deferred(struct work_struct *work)
dev_dbg(card->dev, "ASoC: resume work completed\n");
- /* userspace can access us now we are back as we were before */
- snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0);
-
/* Recheck all endpoints too, their state is affected by suspend */
dapm_mark_endpoints_dirty(card);
snd_soc_dapm_sync(&card->dapm);
+
+ /* userspace can access us now we are back as we were before */
+ snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0);
}
/* powers up audio subsystem after a suspend */
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 016eba10b1ec..7d009428934a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2293,6 +2293,12 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w)
kfree(w);
}
+void snd_soc_dapm_reset_cache(struct snd_soc_dapm_context *dapm)
+{
+ dapm->path_sink_cache.widget = NULL;
+ dapm->path_source_cache.widget = NULL;
+}
+
/* free all dapm widgets and resources */
static void dapm_free_widgets(struct snd_soc_dapm_context *dapm)
{
@@ -2303,6 +2309,7 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm)
continue;
snd_soc_dapm_free_widget(w);
}
+ snd_soc_dapm_reset_cache(dapm);
}
static struct snd_soc_dapm_widget *dapm_find_widget(
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index ecd38e52285a..2f67ba6d7a8f 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -404,7 +404,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx);
/**
* snd_soc_put_volsw_sx - double mixer set callback
* @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
*
* Callback to set the value of a double mixer control that spans 2 registers.
*
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 8d7ec80af51b..6963ba20991c 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -531,7 +531,7 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr,
/* TLV bytes controls need standard kcontrol info handler,
* TLV callback and extended put/get handlers.
*/
- k->info = snd_soc_bytes_info;
+ k->info = snd_soc_bytes_info_ext;
k->tlv.c = snd_soc_bytes_tlv_callback;
ext_ops = tplg->bytes_ext_ops;
@@ -1805,6 +1805,7 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm,
snd_soc_tplg_widget_remove(w);
snd_soc_dapm_free_widget(w);
}
+ snd_soc_dapm_reset_cache(dapm);
}
EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove_all);
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index 843f037a317d..5c2bc53f0a9b 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -669,6 +669,7 @@ static int uni_player_startup(struct snd_pcm_substream *substream,
{
struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
struct uniperif *player = priv->dai_data.uni;
+ player->substream = substream;
player->clk_adj = 0;
@@ -950,6 +951,8 @@ static void uni_player_shutdown(struct snd_pcm_substream *substream,
if (player->state != UNIPERIF_STATE_STOPPED)
/* Stop the player */
uni_player_stop(player);
+
+ player->substream = NULL;
}
static int uni_player_parse_dt_clk_glue(struct platform_device *pdev,
@@ -989,7 +992,7 @@ static int uni_player_parse_dt(struct platform_device *pdev,
if (!info)
return -ENOMEM;
- if (of_property_read_u32(pnode, "version", &player->ver) ||
+ if (of_property_read_u32(pnode, "st,version", &player->ver) ||
player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
dev_err(dev, "Unknown uniperipheral version ");
return -EINVAL;
@@ -998,13 +1001,13 @@ static int uni_player_parse_dt(struct platform_device *pdev,
if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
info->underflow_enabled = 1;
- if (of_property_read_u32(pnode, "uniperiph-id", &info->id)) {
+ if (of_property_read_u32(pnode, "st,uniperiph-id", &info->id)) {
dev_err(dev, "uniperipheral id not defined");
return -EINVAL;
}
/* Read the device mode property */
- if (of_property_read_string(pnode, "mode", &mode)) {
+ if (of_property_read_string(pnode, "st,mode", &mode)) {
dev_err(dev, "uniperipheral mode not defined");
return -EINVAL;
}
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index f791239a3087..8a0eb2050169 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -316,7 +316,7 @@ static int uni_reader_parse_dt(struct platform_device *pdev,
if (!info)
return -ENOMEM;
- if (of_property_read_u32(node, "version", &reader->ver) ||
+ if (of_property_read_u32(node, "st,version", &reader->ver) ||
reader->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
dev_err(&pdev->dev, "Unknown uniperipheral version ");
return -EINVAL;
@@ -346,7 +346,6 @@ int uni_reader_init(struct platform_device *pdev,
reader->hw = &uni_reader_pcm_hw;
reader->dai_ops = &uni_reader_dai_ops;
- dev_err(reader->dev, "%s: enter\n", __func__);
ret = uni_reader_parse_dt(pdev, reader);
if (ret < 0) {
dev_err(reader->dev, "Failed to parse DeviceTree");
diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
index bcbf4da168b6..1bb896d78d09 100644
--- a/sound/soc/sunxi/sun4i-codec.c
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -2,6 +2,7 @@
* Copyright 2014 Emilio López <emilio@elopez.com.ar>
* Copyright 2014 Jon Smirl <jonsmirl@gmail.com>
* Copyright 2015 Maxime Ripard <maxime.ripard@free-electrons.com>
+ * Copyright 2015 Adam Sampson <ats@offog.org>
*
* Based on the Allwinner SDK driver, released under the GPL.
*
@@ -404,7 +405,7 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mute =
static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1);
static const struct snd_kcontrol_new sun4i_codec_widgets[] = {
- SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL,
+ SOC_SINGLE_TLV("Power Amplifier Volume", SUN4I_CODEC_DAC_ACTL,
SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0,
sun4i_codec_pa_volume_scale),
};
@@ -452,12 +453,12 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL,
SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0),
- /* Pre-Amplifier */
- SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL,
+ /* Power Amplifier */
+ SND_SOC_DAPM_MIXER("Power Amplifier", SUN4I_CODEC_ADC_ACTL,
SUN4I_CODEC_ADC_ACTL_PA_EN, 0,
sun4i_codec_pa_mixer_controls,
ARRAY_SIZE(sun4i_codec_pa_mixer_controls)),
- SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0,
+ SND_SOC_DAPM_SWITCH("Power Amplifier Mute", SND_SOC_NOPM, 0, 0,
&sun4i_codec_pa_mute),
SND_SOC_DAPM_OUTPUT("HP Right"),
@@ -480,16 +481,16 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = {
{ "Left Mixer", NULL, "Mixer Enable" },
{ "Left Mixer", "Left DAC Playback Switch", "Left DAC" },
- /* Pre-Amplifier Mixer Routes */
- { "Pre-Amplifier", "Mixer Playback Switch", "Left Mixer" },
- { "Pre-Amplifier", "Mixer Playback Switch", "Right Mixer" },
- { "Pre-Amplifier", "DAC Playback Switch", "Left DAC" },
- { "Pre-Amplifier", "DAC Playback Switch", "Right DAC" },
+ /* Power Amplifier Routes */
+ { "Power Amplifier", "Mixer Playback Switch", "Left Mixer" },
+ { "Power Amplifier", "Mixer Playback Switch", "Right Mixer" },
+ { "Power Amplifier", "DAC Playback Switch", "Left DAC" },
+ { "Power Amplifier", "DAC Playback Switch", "Right DAC" },
- /* PA -> HP path */
- { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" },
- { "HP Right", NULL, "Pre-Amplifier Mute" },
- { "HP Left", NULL, "Pre-Amplifier Mute" },
+ /* Headphone Output Routes */
+ { "Power Amplifier Mute", "Switch", "Power Amplifier" },
+ { "HP Right", NULL, "Power Amplifier Mute" },
+ { "HP Left", NULL, "Power Amplifier Mute" },
};
static struct snd_soc_codec_driver sun4i_codec_codec = {
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 7661616f3636..5b4c58c3e2c5 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 */
@@ -2378,6 +2420,10 @@ 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/quirks-table.h b/sound/usb/quirks-table.h
index 1a1e2e4df35e..c60a776e815d 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2829,6 +2829,17 @@ 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 5ca80e7d30cd..7016ad898187 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -538,6 +538,7 @@ 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,
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 15a12715bd05..b665d85555cb 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -95,6 +95,7 @@ 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,