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-rw-r--r--sound/pci/hda/hda_codec.c34
-rw-r--r--sound/pci/hda/hda_generic.c8
-rw-r--r--sound/pci/hda/hda_generic.h1
-rw-r--r--sound/pci/hda/hda_intel.c2
-rw-r--r--sound/pci/hda/patch_analog.c27
-rw-r--r--sound/pci/hda/patch_conexant.c3
-rw-r--r--sound/pci/hda/patch_realtek.c20
-rw-r--r--sound/pci/hda/patch_sigmatel.c27
-rw-r--r--sound/pci/hda/thinkpad_helper.c1
-rw-r--r--sound/soc/blackfin/Kconfig11
-rw-r--r--sound/soc/codecs/ad1980.c4
-rw-r--r--sound/soc/codecs/da9055.c11
-rw-r--r--sound/soc/codecs/isabelle.c52
-rw-r--r--sound/soc/codecs/max98090.c21
-rw-r--r--sound/soc/codecs/rt5640.c1
-rw-r--r--sound/soc/codecs/sta32x.c74
-rw-r--r--sound/soc/codecs/wm8770.c4
-rw-r--r--sound/soc/codecs/wm8900.c44
-rw-r--r--sound/soc/codecs/wm8993.c1
-rw-r--r--sound/soc/codecs/wm8994.c135
-rw-r--r--sound/soc/davinci/davinci-evm.c1
-rw-r--r--sound/soc/davinci/davinci-mcasp.c83
-rw-r--r--sound/soc/fsl/fsl_esai.c4
-rw-r--r--sound/soc/fsl/fsl_esai.h2
-rw-r--r--sound/soc/fsl/imx-mc13783.c1
-rw-r--r--sound/soc/fsl/imx-sgtl5000.c10
-rw-r--r--sound/soc/fsl/imx-wm8962.c11
-rw-r--r--sound/soc/samsung/Kconfig6
-rw-r--r--sound/soc/soc-dapm.c139
-rw-r--r--sound/soc/txx9/txx9aclc-ac97.c8
-rw-r--r--sound/usb/Kconfig1
31 files changed, 465 insertions, 282 deletions
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index ec4536c8d8d4..dafcf82139e2 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -932,7 +932,7 @@ int snd_hda_bus_new(struct snd_card *card,
}
EXPORT_SYMBOL_GPL(snd_hda_bus_new);
-#ifdef CONFIG_SND_HDA_GENERIC
+#if IS_ENABLED(CONFIG_SND_HDA_GENERIC)
#define is_generic_config(codec) \
(codec->modelname && !strcmp(codec->modelname, "generic"))
#else
@@ -1339,23 +1339,15 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid)
/*
* Dynamic symbol binding for the codec parsers
*/
-#ifdef MODULE
-#define load_parser_sym(sym) ((int (*)(struct hda_codec *))symbol_request(sym))
-#define unload_parser_addr(addr) symbol_put_addr(addr)
-#else
-#define load_parser_sym(sym) (sym)
-#define unload_parser_addr(addr) do {} while (0)
-#endif
#define load_parser(codec, sym) \
- ((codec)->parser = load_parser_sym(sym))
+ ((codec)->parser = (int (*)(struct hda_codec *))symbol_request(sym))
static void unload_parser(struct hda_codec *codec)
{
- if (codec->parser) {
- unload_parser_addr(codec->parser);
- codec->parser = NULL;
- }
+ if (codec->parser)
+ symbol_put_addr(codec->parser);
+ codec->parser = NULL;
}
/*
@@ -1570,7 +1562,7 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec)
EXPORT_SYMBOL_GPL(snd_hda_codec_update_widgets);
-#ifdef CONFIG_SND_HDA_CODEC_HDMI
+#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI)
/* if all audio out widgets are digital, let's assume the codec as a HDMI/DP */
static bool is_likely_hdmi_codec(struct hda_codec *codec)
{
@@ -1620,12 +1612,20 @@ int snd_hda_codec_configure(struct hda_codec *codec)
patch = codec->preset->patch;
if (!patch) {
unload_parser(codec); /* to be sure */
- if (is_likely_hdmi_codec(codec))
+ if (is_likely_hdmi_codec(codec)) {
+#if IS_MODULE(CONFIG_SND_HDA_CODEC_HDMI)
patch = load_parser(codec, snd_hda_parse_hdmi_codec);
-#ifdef CONFIG_SND_HDA_GENERIC
- if (!patch)
+#elif IS_BUILTIN(CONFIG_SND_HDA_CODEC_HDMI)
+ patch = snd_hda_parse_hdmi_codec;
+#endif
+ }
+ if (!patch) {
+#if IS_MODULE(CONFIG_SND_HDA_GENERIC)
patch = load_parser(codec, snd_hda_parse_generic_codec);
+#elif IS_BUILTIN(CONFIG_SND_HDA_GENERIC)
+ patch = snd_hda_parse_generic_codec;
#endif
+ }
if (!patch) {
printk(KERN_ERR "hda-codec: No codec parser is available\n");
return -ENODEV;
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 8321a97d5c05..d9a09bdd09db 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -3269,7 +3269,7 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol,
mutex_unlock(&codec->control_mutex);
snd_hda_codec_flush_cache(codec); /* flush the updates */
if (err >= 0 && spec->cap_sync_hook)
- spec->cap_sync_hook(codec, ucontrol);
+ spec->cap_sync_hook(codec, kcontrol, ucontrol);
return err;
}
@@ -3390,7 +3390,7 @@ static int cap_single_sw_put(struct snd_kcontrol *kcontrol,
return ret;
if (spec->cap_sync_hook)
- spec->cap_sync_hook(codec, ucontrol);
+ spec->cap_sync_hook(codec, kcontrol, ucontrol);
return ret;
}
@@ -3795,7 +3795,7 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx,
return 0;
snd_hda_activate_path(codec, path, true, false);
if (spec->cap_sync_hook)
- spec->cap_sync_hook(codec, NULL);
+ spec->cap_sync_hook(codec, NULL, NULL);
path_power_down_sync(codec, old_path);
return 1;
}
@@ -5270,7 +5270,7 @@ static void init_input_src(struct hda_codec *codec)
}
if (spec->cap_sync_hook)
- spec->cap_sync_hook(codec, NULL);
+ spec->cap_sync_hook(codec, NULL, NULL);
}
/* set right pin controls for digital I/O */
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 07f767231c9f..c908afbe4d94 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -274,6 +274,7 @@ struct hda_gen_spec {
void (*init_hook)(struct hda_codec *codec);
void (*automute_hook)(struct hda_codec *codec);
void (*cap_sync_hook)(struct hda_codec *codec,
+ struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
/* PCM hooks */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index fa2879a21a50..e354ab1ec20f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -198,7 +198,7 @@ MODULE_DESCRIPTION("Intel HDA driver");
#endif
#if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO)
-#ifdef CONFIG_SND_HDA_CODEC_HDMI
+#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI)
#define SUPPORT_VGA_SWITCHEROO
#endif
#endif
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 7a426ed491f2..df3652ad15ef 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -244,6 +244,19 @@ static void ad_fixup_inv_jack_detect(struct hda_codec *codec,
}
}
+/* Toshiba Satellite L40 implements EAPD in a standard way unlike others */
+static void ad1986a_fixup_eapd(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct ad198x_spec *spec = codec->spec;
+
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ codec->inv_eapd = 0;
+ spec->gen.keep_eapd_on = 1;
+ spec->eapd_nid = 0x1b;
+ }
+}
+
enum {
AD1986A_FIXUP_INV_JACK_DETECT,
AD1986A_FIXUP_ULTRA,
@@ -251,6 +264,7 @@ enum {
AD1986A_FIXUP_3STACK,
AD1986A_FIXUP_LAPTOP,
AD1986A_FIXUP_LAPTOP_IMIC,
+ AD1986A_FIXUP_EAPD,
};
static const struct hda_fixup ad1986a_fixups[] = {
@@ -311,6 +325,10 @@ static const struct hda_fixup ad1986a_fixups[] = {
.chained_before = 1,
.chain_id = AD1986A_FIXUP_LAPTOP,
},
+ [AD1986A_FIXUP_EAPD] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = ad1986a_fixup_eapd,
+ },
};
static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
@@ -318,6 +336,7 @@ static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK),
SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK),
SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_FIXUP_3STACK),
+ SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba Satellite L40", AD1986A_FIXUP_EAPD),
SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_FIXUP_LAPTOP),
SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG),
SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA),
@@ -472,6 +491,8 @@ static int ad1983_add_spdif_mux_ctl(struct hda_codec *codec)
static int patch_ad1983(struct hda_codec *codec)
{
struct ad198x_spec *spec;
+ static hda_nid_t conn_0c[] = { 0x08 };
+ static hda_nid_t conn_0d[] = { 0x09 };
int err;
err = alloc_ad_spec(codec);
@@ -479,8 +500,14 @@ static int patch_ad1983(struct hda_codec *codec)
return err;
spec = codec->spec;
+ spec->gen.mixer_nid = 0x0e;
spec->gen.beep_nid = 0x10;
set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
+
+ /* limit the loopback routes not to confuse the parser */
+ snd_hda_override_conn_list(codec, 0x0c, ARRAY_SIZE(conn_0c), conn_0c);
+ snd_hda_override_conn_list(codec, 0x0d, ARRAY_SIZE(conn_0d), conn_0d);
+
err = ad198x_parse_auto_config(codec, false);
if (err < 0)
goto error;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 4e0ec146553d..bcf91bea3317 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3291,7 +3291,8 @@ static void cxt_update_headset_mode(struct hda_codec *codec)
}
static void cxt_update_headset_mode_hook(struct hda_codec *codec,
- struct snd_ctl_elem_value *ucontrol)
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
cxt_update_headset_mode(codec);
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 56a8f1876603..a9a83b85517a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -708,7 +708,8 @@ static void alc_inv_dmic_sync(struct hda_codec *codec, bool force)
}
static void alc_inv_dmic_hook(struct hda_codec *codec,
- struct snd_ctl_elem_value *ucontrol)
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
alc_inv_dmic_sync(codec, false);
}
@@ -1821,6 +1822,7 @@ enum {
ALC889_FIXUP_IMAC91_VREF,
ALC889_FIXUP_MBA11_VREF,
ALC889_FIXUP_MBA21_VREF,
+ ALC889_FIXUP_MP11_VREF,
ALC882_FIXUP_INV_DMIC,
ALC882_FIXUP_NO_PRIMARY_HP,
ALC887_FIXUP_ASUS_BASS,
@@ -2190,6 +2192,12 @@ static const struct hda_fixup alc882_fixups[] = {
.chained = true,
.chain_id = ALC889_FIXUP_MBP_VREF,
},
+ [ALC889_FIXUP_MP11_VREF] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc889_fixup_mba11_vref,
+ .chained = true,
+ .chain_id = ALC885_FIXUP_MACPRO_GPIO,
+ },
[ALC882_FIXUP_INV_DMIC] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_inv_dmic_0x12,
@@ -2253,7 +2261,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF),
SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC889_FIXUP_MBP_VREF),
SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF),
- SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_FIXUP_MACPRO_GPIO),
+ SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC889_FIXUP_MP11_VREF),
SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_FIXUP_MACPRO_GPIO),
SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_FIXUP_MACPRO_GPIO),
SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC889_FIXUP_MBP_VREF),
@@ -3211,7 +3219,8 @@ static void alc269_fixup_hp_gpio_mute_hook(void *private_data, int enabled)
/* turn on/off mic-mute LED per capture hook */
static void alc269_fixup_hp_gpio_mic_mute_hook(struct hda_codec *codec,
- struct snd_ctl_elem_value *ucontrol)
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct alc_spec *spec = codec->spec;
unsigned int oldval = spec->gpio_led;
@@ -3521,7 +3530,8 @@ static void alc_update_headset_mode(struct hda_codec *codec)
}
static void alc_update_headset_mode_hook(struct hda_codec *codec,
- struct snd_ctl_elem_value *ucontrol)
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
alc_update_headset_mode(codec);
}
@@ -4322,6 +4332,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101),
+ SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
@@ -5096,6 +5107,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x060a, "Dell XPS 13", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_AUTO_MUTE),
SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_AUTO_MUTE),
SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6998cf29b9bc..7311badf6a94 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -194,7 +194,7 @@ struct sigmatel_spec {
int default_polarity;
unsigned int mic_mute_led_gpio; /* capture mute LED GPIO */
- bool mic_mute_led_on; /* current mic mute state */
+ unsigned int mic_enabled; /* current mic mute state (bitmask) */
/* stream */
unsigned int stream_delay;
@@ -324,19 +324,26 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask,
/* hook for controlling mic-mute LED GPIO */
static void stac_capture_led_hook(struct hda_codec *codec,
- struct snd_ctl_elem_value *ucontrol)
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct sigmatel_spec *spec = codec->spec;
- bool mute;
+ unsigned int mask;
+ bool cur_mute, prev_mute;
- if (!ucontrol)
+ if (!kcontrol || !ucontrol)
return;
- mute = !(ucontrol->value.integer.value[0] ||
- ucontrol->value.integer.value[1]);
- if (spec->mic_mute_led_on != mute) {
- spec->mic_mute_led_on = mute;
- if (mute)
+ mask = 1U << snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ prev_mute = !spec->mic_enabled;
+ if (ucontrol->value.integer.value[0] ||
+ ucontrol->value.integer.value[1])
+ spec->mic_enabled |= mask;
+ else
+ spec->mic_enabled &= ~mask;
+ cur_mute = !spec->mic_enabled;
+ if (cur_mute != prev_mute) {
+ if (cur_mute)
spec->gpio_data |= spec->mic_mute_led_gpio;
else
spec->gpio_data &= ~spec->mic_mute_led_gpio;
@@ -4462,7 +4469,7 @@ static void stac_setup_gpio(struct hda_codec *codec)
if (spec->mic_mute_led_gpio) {
spec->gpio_mask |= spec->mic_mute_led_gpio;
spec->gpio_dir |= spec->mic_mute_led_gpio;
- spec->mic_mute_led_on = true;
+ spec->mic_enabled = 0;
spec->gpio_data |= spec->mic_mute_led_gpio;
spec->gen.cap_sync_hook = stac_capture_led_hook;
diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c
index 5799fbc24c28..8fe3b8c18ed4 100644
--- a/sound/pci/hda/thinkpad_helper.c
+++ b/sound/pci/hda/thinkpad_helper.c
@@ -39,6 +39,7 @@ static void update_tpacpi_mute_led(void *private_data, int enabled)
}
static void update_tpacpi_micmute_led(struct hda_codec *codec,
+ struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
if (!ucontrol || !led_set_func)
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 54f74f8cbb75..4544d8eb1452 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -11,7 +11,7 @@ config SND_BF5XX_I2S
config SND_BF5XX_SOC_SSM2602
tristate "SoC SSM2602 Audio Codec Add-On Card support"
- depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
+ depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
select SND_BF5XX_SOC_I2S if !BF60x
select SND_BF6XX_SOC_I2S if BF60x
select SND_SOC_SSM2602
@@ -21,10 +21,9 @@ config SND_BF5XX_SOC_SSM2602
config SND_SOC_BFIN_EVAL_ADAU1701
tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards"
- depends on SND_BF5XX_I2S
+ depends on SND_BF5XX_I2S && I2C
select SND_BF5XX_SOC_I2S
select SND_SOC_ADAU1701
- select I2C
help
Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ
board connected to one of the Blackfin evaluation boards like the
@@ -45,7 +44,7 @@ config SND_SOC_BFIN_EVAL_ADAU1373
config SND_SOC_BFIN_EVAL_ADAV80X
tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards"
- depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
+ depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
select SND_BF5XX_SOC_I2S
select SND_SOC_ADAV80X
help
@@ -58,7 +57,7 @@ config SND_SOC_BFIN_EVAL_ADAV80X
config SND_BF5XX_SOC_AD1836
tristate "SoC AD1836 Audio support for BF5xx"
- depends on SND_BF5XX_I2S
+ depends on SND_BF5XX_I2S && SPI_MASTER
select SND_BF5XX_SOC_I2S
select SND_SOC_AD1836
help
@@ -66,7 +65,7 @@ config SND_BF5XX_SOC_AD1836
config SND_BF5XX_SOC_AD193X
tristate "SoC AD193X Audio support for Blackfin"
- depends on SND_BF5XX_I2S
+ depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
select SND_BF5XX_SOC_I2S
select SND_SOC_AD193X
help
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 7257a8885f42..34d965a4a040 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -57,8 +57,8 @@ static const u16 ad1980_reg[] = {
static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line",
"Stereo Mix", "Mono Mix", "Phone"};
-static const struct soc_enum ad1980_cap_src =
- SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 7, ad1980_rec_sel);
+static SOC_ENUM_DOUBLE_DECL(ad1980_cap_src,
+ AC97_REC_SEL, 8, 0, ad1980_rec_sel);
static const struct snd_kcontrol_new ad1980_snd_ac97_controls[] = {
SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1),
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index 52b79a487ac7..422812613a28 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -1523,8 +1523,15 @@ static int da9055_remove(struct i2c_client *client)
return 0;
}
+/*
+ * DO NOT change the device Ids. The naming is intentionally specific as both
+ * the CODEC and PMIC parts of this chip are instantiated separately as I2C
+ * devices (both have configurable I2C addresses, and are to all intents and
+ * purposes separate). As a result there are specific DA9055 Ids for CODEC
+ * and PMIC, which must be different to operate together.
+ */
static const struct i2c_device_id da9055_i2c_id[] = {
- { "da9055", 0 },
+ { "da9055-codec", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, da9055_i2c_id);
@@ -1532,7 +1539,7 @@ MODULE_DEVICE_TABLE(i2c, da9055_i2c_id);
/* I2C codec control layer */
static struct i2c_driver da9055_i2c_driver = {
.driver = {
- .name = "da9055",
+ .name = "da9055-codec",
.owner = THIS_MODULE,
},
.probe = da9055_i2c_probe,
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index 5839048ec467..cb736ddc446d 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -140,13 +140,17 @@ static const char *isabelle_rx1_texts[] = {"VRX1", "ARX1"};
static const char *isabelle_rx2_texts[] = {"VRX2", "ARX2"};
static const struct soc_enum isabelle_rx1_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, 1, isabelle_rx1_texts),
- SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, 1, isabelle_rx1_texts),
+ SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3,
+ ARRAY_SIZE(isabelle_rx1_texts), isabelle_rx1_texts),
+ SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5,
+ ARRAY_SIZE(isabelle_rx1_texts), isabelle_rx1_texts),
};
static const struct soc_enum isabelle_rx2_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, 1, isabelle_rx2_texts),
- SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, 1, isabelle_rx2_texts),
+ SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2,
+ ARRAY_SIZE(isabelle_rx2_texts), isabelle_rx2_texts),
+ SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4,
+ ARRAY_SIZE(isabelle_rx2_texts), isabelle_rx2_texts),
};
/* Headset DAC playback switches */
@@ -161,13 +165,17 @@ static const char *isabelle_atx_texts[] = {"AMIC1", "DMIC"};
static const char *isabelle_vtx_texts[] = {"AMIC2", "DMIC"};
static const struct soc_enum isabelle_atx_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, 1, isabelle_atx_texts),
- SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_atx_texts),
+ SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7,
+ ARRAY_SIZE(isabelle_atx_texts), isabelle_atx_texts),
+ SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0,
+ ARRAY_SIZE(isabelle_atx_texts), isabelle_atx_texts),
};
static const struct soc_enum isabelle_vtx_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, 1, isabelle_vtx_texts),
- SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_vtx_texts),
+ SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6,
+ ARRAY_SIZE(isabelle_vtx_texts), isabelle_vtx_texts),
+ SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0,
+ ARRAY_SIZE(isabelle_vtx_texts), isabelle_vtx_texts),
};
static const struct snd_kcontrol_new atx_mux_controls =
@@ -183,17 +191,13 @@ static const char *isabelle_amic1_texts[] = {
/* Left analog microphone selection */
static const char *isabelle_amic2_texts[] = {"Sub Mic", "Aux/FM Right"};
-static const struct soc_enum isabelle_amic1_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 5,
- ARRAY_SIZE(isabelle_amic1_texts),
- isabelle_amic1_texts),
-};
+static SOC_ENUM_SINGLE_DECL(isabelle_amic1_enum,
+ ISABELLE_AMIC_CFG_REG, 5,
+ isabelle_amic1_texts);
-static const struct soc_enum isabelle_amic2_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 4,
- ARRAY_SIZE(isabelle_amic2_texts),
- isabelle_amic2_texts),
-};
+static SOC_ENUM_SINGLE_DECL(isabelle_amic2_enum,
+ ISABELLE_AMIC_CFG_REG, 4,
+ isabelle_amic2_texts);
static const struct snd_kcontrol_new amic1_control =
SOC_DAPM_ENUM("Route", isabelle_amic1_enum);
@@ -206,16 +210,20 @@ static const char *isabelle_st_audio_texts[] = {"ATX1", "ATX2"};
static const char *isabelle_st_voice_texts[] = {"VTX1", "VTX2"};
static const struct soc_enum isabelle_st_audio_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, 1,
+ SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7,
+ ARRAY_SIZE(isabelle_st_audio_texts),
isabelle_st_audio_texts),
- SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, 1,
+ SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7,
+ ARRAY_SIZE(isabelle_st_audio_texts),
isabelle_st_audio_texts),
};
static const struct soc_enum isabelle_st_voice_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, 1,
+ SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7,
+ ARRAY_SIZE(isabelle_st_voice_texts),
isabelle_st_voice_texts),
- SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, 1,
+ SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7,
+ ARRAY_SIZE(isabelle_st_voice_texts),
isabelle_st_voice_texts),
};
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 51f9b3d16b41..9f714ea86613 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -336,6 +336,7 @@ static bool max98090_readable_register(struct device *dev, unsigned int reg)
case M98090_REG_RECORD_TDM_SLOT:
case M98090_REG_SAMPLE_RATE:
case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E:
+ case M98090_REG_REVISION_ID:
return true;
default:
return false;
@@ -1769,16 +1770,6 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
switch (level) {
case SND_SOC_BIAS_ON:
- if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
- ret = regcache_sync(max98090->regmap);
-
- if (ret != 0) {
- dev_err(codec->dev,
- "Failed to sync cache: %d\n", ret);
- return ret;
- }
- }
-
if (max98090->jack_state == M98090_JACK_STATE_HEADSET) {
/*
* Set to normal bias level.
@@ -1792,6 +1783,16 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ ret = regcache_sync(max98090->regmap);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to sync cache: %d\n", ret);
+ return ret;
+ }
+ }
+ break;
+
case SND_SOC_BIAS_OFF:
/* Set internal pull-up to lowest power mode */
snd_soc_update_bits(codec, M98090_REG_JACK_DETECT,
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index a3fb41179636..886924934aa5 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -2093,6 +2093,7 @@ MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id);
#ifdef CONFIG_ACPI
static struct acpi_device_id rt5640_acpi_match[] = {
{ "INT33CA", 0 },
+ { "10EC5640", 0 },
{ },
};
MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match);
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 06edb396e733..ea78c172538c 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -187,42 +187,42 @@ static const unsigned int sta32x_limiter_drc_release_tlv[] = {
13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0),
};
-static const struct soc_enum sta32x_drc_ac_enum =
- SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT,
- 2, sta32x_drc_ac);
-static const struct soc_enum sta32x_auto_eq_enum =
- SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT,
- 3, sta32x_auto_eq_mode);
-static const struct soc_enum sta32x_auto_gc_enum =
- SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT,
- 4, sta32x_auto_gc_mode);
-static const struct soc_enum sta32x_auto_xo_enum =
- SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT,
- 16, sta32x_auto_xo_mode);
-static const struct soc_enum sta32x_preset_eq_enum =
- SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT,
- 32, sta32x_preset_eq_mode);
-static const struct soc_enum sta32x_limiter_ch1_enum =
- SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT,
- 3, sta32x_limiter_select);
-static const struct soc_enum sta32x_limiter_ch2_enum =
- SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT,
- 3, sta32x_limiter_select);
-static const struct soc_enum sta32x_limiter_ch3_enum =
- SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT,
- 3, sta32x_limiter_select);
-static const struct soc_enum sta32x_limiter1_attack_rate_enum =
- SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT,
- 16, sta32x_limiter_attack_rate);
-static const struct soc_enum sta32x_limiter2_attack_rate_enum =
- SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT,
- 16, sta32x_limiter_attack_rate);
-static const struct soc_enum sta32x_limiter1_release_rate_enum =
- SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT,
- 16, sta32x_limiter_release_rate);
-static const struct soc_enum sta32x_limiter2_release_rate_enum =
- SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT,
- 16, sta32x_limiter_release_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_drc_ac_enum,
+ STA32X_CONFD, STA32X_CONFD_DRC_SHIFT,
+ sta32x_drc_ac);
+static SOC_ENUM_SINGLE_DECL(sta32x_auto_eq_enum,
+ STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT,
+ sta32x_auto_eq_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_auto_gc_enum,
+ STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT,
+ sta32x_auto_gc_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_auto_xo_enum,
+ STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT,
+ sta32x_auto_xo_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_preset_eq_enum,
+ STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT,
+ sta32x_preset_eq_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch1_enum,
+ STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT,
+ sta32x_limiter_select);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch2_enum,
+ STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT,
+ sta32x_limiter_select);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch3_enum,
+ STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT,
+ sta32x_limiter_select);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_attack_rate_enum,
+ STA32X_L1AR, STA32X_LxA_SHIFT,
+ sta32x_limiter_attack_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_attack_rate_enum,
+ STA32X_L2AR, STA32X_LxA_SHIFT,
+ sta32x_limiter_attack_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_release_rate_enum,
+ STA32X_L1AR, STA32X_LxR_SHIFT,
+ sta32x_limiter_release_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum,
+ STA32X_L2AR, STA32X_LxR_SHIFT,
+ sta32x_limiter_release_rate);
/* byte array controls for setting biquad, mixer, scaling coefficients;
* for biquads all five coefficients need to be set in one go,
@@ -331,7 +331,7 @@ static int sta32x_sync_coef_shadow(struct snd_soc_codec *codec)
static int sta32x_cache_sync(struct snd_soc_codec *codec)
{
- struct sta32x_priv *sta32x = codec->control_data;
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
unsigned int mute;
int rc;
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index 89a18d82f303..5bce21013485 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -196,8 +196,8 @@ static const char *ain_text[] = {
"AIN5", "AIN6", "AIN7", "AIN8"
};
-static const struct soc_enum ain_enum =
- SOC_ENUM_DOUBLE(WM8770_ADCMUX, 0, 4, 8, ain_text);
+static SOC_ENUM_DOUBLE_DECL(ain_enum,
+ WM8770_ADCMUX, 0, 4, ain_text);
static const struct snd_kcontrol_new ain_mux =
SOC_DAPM_ENUM("Capture Mux", ain_enum);
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index e98bc7038a08..43c2201cb901 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -304,53 +304,53 @@ static const DECLARE_TLV_DB_SCALE(adc_tlv, -7200, 75, 1);
static const char *mic_bias_level_txt[] = { "0.9*AVDD", "0.65*AVDD" };
-static const struct soc_enum mic_bias_level =
-SOC_ENUM_SINGLE(WM8900_REG_INCTL, 8, 2, mic_bias_level_txt);
+static SOC_ENUM_SINGLE_DECL(mic_bias_level,
+ WM8900_REG_INCTL, 8, mic_bias_level_txt);
static const char *dac_mute_rate_txt[] = { "Fast", "Slow" };
-static const struct soc_enum dac_mute_rate =
-SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 7, 2, dac_mute_rate_txt);
+static SOC_ENUM_SINGLE_DECL(dac_mute_rate,
+ WM8900_REG_DACCTRL, 7, dac_mute_rate_txt);
static const char *dac_deemphasis_txt[] = {
"Disabled", "32kHz", "44.1kHz", "48kHz"
};
-static const struct soc_enum dac_deemphasis =
-SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 4, 4, dac_deemphasis_txt);
+static SOC_ENUM_SINGLE_DECL(dac_deemphasis,
+ WM8900_REG_DACCTRL, 4, dac_deemphasis_txt);
static const char *adc_hpf_cut_txt[] = {
"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"
};
-static const struct soc_enum adc_hpf_cut =
-SOC_ENUM_SINGLE(WM8900_REG_ADCCTRL, 5, 4, adc_hpf_cut_txt);
+static SOC_ENUM_SINGLE_DECL(adc_hpf_cut,
+ WM8900_REG_ADCCTRL, 5, adc_hpf_cut_txt);
static const char *lr_txt[] = {
"Left", "Right"
};
-static const struct soc_enum aifl_src =
-SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 15, 2, lr_txt);
+static SOC_ENUM_SINGLE_DECL(aifl_src,
+ WM8900_REG_AUDIO1, 15, lr_txt);
-static const struct soc_enum aifr_src =
-SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 14, 2, lr_txt);
+static SOC_ENUM_SINGLE_DECL(aifr_src,
+ WM8900_REG_AUDIO1, 14, lr_txt);
-static const struct soc_enum dacl_src =
-SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 15, 2, lr_txt);
+static SOC_ENUM_SINGLE_DECL(dacl_src,
+ WM8900_REG_AUDIO2, 15, lr_txt);
-static const struct soc_enum dacr_src =
-SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 14, 2, lr_txt);
+static SOC_ENUM_SINGLE_DECL(dacr_src,
+ WM8900_REG_AUDIO2, 14, lr_txt);
static const char *sidetone_txt[] = {
"Disabled", "Left ADC", "Right ADC"
};
-static const struct soc_enum dacl_sidetone =
-SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 2, 3, sidetone_txt);
+static SOC_ENUM_SINGLE_DECL(dacl_sidetone,
+ WM8900_REG_SIDETONE, 2, sidetone_txt);
-static const struct soc_enum dacr_sidetone =
-SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 0, 3, sidetone_txt);
+static SOC_ENUM_SINGLE_DECL(dacr_sidetone,
+ WM8900_REG_SIDETONE, 0, sidetone_txt);
static const struct snd_kcontrol_new wm8900_snd_controls[] = {
SOC_ENUM("Mic Bias Level", mic_bias_level),
@@ -496,8 +496,8 @@ SOC_DAPM_SINGLE("RINPUT3 Switch", WM8900_REG_INCTL, 0, 1, 0),
static const char *wm8900_lp_mux[] = { "Disabled", "Enabled" };
-static const struct soc_enum wm8900_lineout2_lp_mux =
-SOC_ENUM_SINGLE(WM8900_REG_LOUTMIXCTL1, 1, 2, wm8900_lp_mux);
+static SOC_ENUM_SINGLE_DECL(wm8900_lineout2_lp_mux,
+ WM8900_REG_LOUTMIXCTL1, 1, wm8900_lp_mux);
static const struct snd_kcontrol_new wm8900_lineout2_lp =
SOC_DAPM_ENUM("Route", wm8900_lineout2_lp_mux);
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 433d59a0f3ef..2ee23a39622c 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1562,7 +1562,6 @@ static int wm8993_remove(struct snd_soc_codec *codec)
struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF);
- regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies);
return 0;
}
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index b9be9cbc4603..adb72063d44e 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -265,21 +265,21 @@ static const char *sidetone_hpf_text[] = {
"2.7kHz", "1.35kHz", "675Hz", "370Hz", "180Hz", "90Hz", "45Hz"
};
-static const struct soc_enum sidetone_hpf =
- SOC_ENUM_SINGLE(WM8994_SIDETONE, 7, 7, sidetone_hpf_text);
+static SOC_ENUM_SINGLE_DECL(sidetone_hpf,
+ WM8994_SIDETONE, 7, sidetone_hpf_text);
static const char *adc_hpf_text[] = {
"HiFi", "Voice 1", "Voice 2", "Voice 3"
};
-static const struct soc_enum aif1adc1_hpf =
- SOC_ENUM_SINGLE(WM8994_AIF1_ADC1_FILTERS, 13, 4, adc_hpf_text);
+static SOC_ENUM_SINGLE_DECL(aif1adc1_hpf,
+ WM8994_AIF1_ADC1_FILTERS, 13, adc_hpf_text);
-static const struct soc_enum aif1adc2_hpf =
- SOC_ENUM_SINGLE(WM8994_AIF1_ADC2_FILTERS, 13, 4, adc_hpf_text);
+static SOC_ENUM_SINGLE_DECL(aif1adc2_hpf,
+ WM8994_AIF1_ADC2_FILTERS, 13, adc_hpf_text);
-static const struct soc_enum aif2adc_hpf =
- SOC_ENUM_SINGLE(WM8994_AIF2_ADC_FILTERS, 13, 4, adc_hpf_text);
+static SOC_ENUM_SINGLE_DECL(aif2adc_hpf,
+ WM8994_AIF2_ADC_FILTERS, 13, adc_hpf_text);
static const DECLARE_TLV_DB_SCALE(aif_tlv, 0, 600, 0);
static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
@@ -501,39 +501,39 @@ static const char *aif_chan_src_text[] = {
"Left", "Right"
};
-static const struct soc_enum aif1adcl_src =
- SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_1, 15, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif1adcl_src,
+ WM8994_AIF1_CONTROL_1, 15, aif_chan_src_text);
-static const struct soc_enum aif1adcr_src =
- SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_1, 14, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif1adcr_src,
+ WM8994_AIF1_CONTROL_1, 14, aif_chan_src_text);
-static const struct soc_enum aif2adcl_src =
- SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_1, 15, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2adcl_src,
+ WM8994_AIF2_CONTROL_1, 15, aif_chan_src_text);
-static const struct soc_enum aif2adcr_src =
- SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_1, 14, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2adcr_src,
+ WM8994_AIF2_CONTROL_1, 14, aif_chan_src_text);
-static const struct soc_enum aif1dacl_src =
- SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 15, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif1dacl_src,
+ WM8994_AIF1_CONTROL_2, 15, aif_chan_src_text);
-static const struct soc_enum aif1dacr_src =
- SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 14, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif1dacr_src,
+ WM8994_AIF1_CONTROL_2, 14, aif_chan_src_text);
-static const struct soc_enum aif2dacl_src =
- SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 15, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2dacl_src,
+ WM8994_AIF2_CONTROL_2, 15, aif_chan_src_text);
-static const struct soc_enum aif2dacr_src =
- SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 14, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2dacr_src,
+ WM8994_AIF2_CONTROL_2, 14, aif_chan_src_text);
static const char *osr_text[] = {
"Low Power", "High Performance",
};
-static const struct soc_enum dac_osr =
- SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 0, 2, osr_text);
+static SOC_ENUM_SINGLE_DECL(dac_osr,
+ WM8994_OVERSAMPLING, 0, osr_text);
-static const struct soc_enum adc_osr =
- SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 1, 2, osr_text);
+static SOC_ENUM_SINGLE_DECL(adc_osr,
+ WM8994_OVERSAMPLING, 1, osr_text);
static const struct snd_kcontrol_new wm8994_snd_controls[] = {
SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME,
@@ -690,17 +690,20 @@ static const char *wm8958_ng_text[] = {
"30ms", "125ms", "250ms", "500ms",
};
-static const struct soc_enum wm8958_aif1dac1_ng_hold =
- SOC_ENUM_SINGLE(WM8958_AIF1_DAC1_NOISE_GATE,
- WM8958_AIF1DAC1_NG_THR_SHIFT, 4, wm8958_ng_text);
+static SOC_ENUM_SINGLE_DECL(wm8958_aif1dac1_ng_hold,
+ WM8958_AIF1_DAC1_NOISE_GATE,
+ WM8958_AIF1DAC1_NG_THR_SHIFT,
+ wm8958_ng_text);
-static const struct soc_enum wm8958_aif1dac2_ng_hold =
- SOC_ENUM_SINGLE(WM8958_AIF1_DAC2_NOISE_GATE,
- WM8958_AIF1DAC2_NG_THR_SHIFT, 4, wm8958_ng_text);
+static SOC_ENUM_SINGLE_DECL(wm8958_aif1dac2_ng_hold,
+ WM8958_AIF1_DAC2_NOISE_GATE,
+ WM8958_AIF1DAC2_NG_THR_SHIFT,
+ wm8958_ng_text);
-static const struct soc_enum wm8958_aif2dac_ng_hold =
- SOC_ENUM_SINGLE(WM8958_AIF2_DAC_NOISE_GATE,
- WM8958_AIF2DAC_NG_THR_SHIFT, 4, wm8958_ng_text);
+static SOC_ENUM_SINGLE_DECL(wm8958_aif2dac_ng_hold,
+ WM8958_AIF2_DAC_NOISE_GATE,
+ WM8958_AIF2DAC_NG_THR_SHIFT,
+ wm8958_ng_text);
static const struct snd_kcontrol_new wm8958_snd_controls[] = {
SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv),
@@ -1341,8 +1344,8 @@ static const char *adc_mux_text[] = {
"DMIC",
};
-static const struct soc_enum adc_enum =
- SOC_ENUM_SINGLE(0, 0, 2, adc_mux_text);
+static SOC_ENUM_SINGLE_DECL(adc_enum,
+ 0, 0, adc_mux_text);
static const struct snd_kcontrol_new adcl_mux =
SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum);
@@ -1478,14 +1481,14 @@ static const char *sidetone_text[] = {
"ADC/DMIC1", "DMIC2",
};
-static const struct soc_enum sidetone1_enum =
- SOC_ENUM_SINGLE(WM8994_SIDETONE, 0, 2, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(sidetone1_enum,
+ WM8994_SIDETONE, 0, sidetone_text);
static const struct snd_kcontrol_new sidetone1_mux =
SOC_DAPM_ENUM("Left Sidetone Mux", sidetone1_enum);
-static const struct soc_enum sidetone2_enum =
- SOC_ENUM_SINGLE(WM8994_SIDETONE, 1, 2, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(sidetone2_enum,
+ WM8994_SIDETONE, 1, sidetone_text);
static const struct snd_kcontrol_new sidetone2_mux =
SOC_DAPM_ENUM("Right Sidetone Mux", sidetone2_enum);
@@ -1498,22 +1501,24 @@ static const char *loopback_text[] = {
"None", "ADCDAT",
};
-static const struct soc_enum aif1_loopback_enum =
- SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, WM8994_AIF1_LOOPBACK_SHIFT, 2,
- loopback_text);
+static SOC_ENUM_SINGLE_DECL(aif1_loopback_enum,
+ WM8994_AIF1_CONTROL_2,
+ WM8994_AIF1_LOOPBACK_SHIFT,
+ loopback_text);
static const struct snd_kcontrol_new aif1_loopback =
SOC_DAPM_ENUM("AIF1 Loopback", aif1_loopback_enum);
-static const struct soc_enum aif2_loopback_enum =
- SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, WM8994_AIF2_LOOPBACK_SHIFT, 2,
- loopback_text);
+static SOC_ENUM_SINGLE_DECL(aif2_loopback_enum,
+ WM8994_AIF2_CONTROL_2,
+ WM8994_AIF2_LOOPBACK_SHIFT,
+ loopback_text);
static const struct snd_kcontrol_new aif2_loopback =
SOC_DAPM_ENUM("AIF2 Loopback", aif2_loopback_enum);
-static const struct soc_enum aif1dac_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text);
+static SOC_ENUM_SINGLE_DECL(aif1dac_enum,
+ WM8994_POWER_MANAGEMENT_6, 0, aif1dac_text);
static const struct snd_kcontrol_new aif1dac_mux =
SOC_DAPM_ENUM("AIF1DAC Mux", aif1dac_enum);
@@ -1522,8 +1527,8 @@ static const char *aif2dac_text[] = {
"AIF2DACDAT", "AIF3DACDAT",
};
-static const struct soc_enum aif2dac_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 1, 2, aif2dac_text);
+static SOC_ENUM_SINGLE_DECL(aif2dac_enum,
+ WM8994_POWER_MANAGEMENT_6, 1, aif2dac_text);
static const struct snd_kcontrol_new aif2dac_mux =
SOC_DAPM_ENUM("AIF2DAC Mux", aif2dac_enum);
@@ -1532,8 +1537,8 @@ static const char *aif2adc_text[] = {
"AIF2ADCDAT", "AIF3DACDAT",
};
-static const struct soc_enum aif2adc_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 2, 2, aif2adc_text);
+static SOC_ENUM_SINGLE_DECL(aif2adc_enum,
+ WM8994_POWER_MANAGEMENT_6, 2, aif2adc_text);
static const struct snd_kcontrol_new aif2adc_mux =
SOC_DAPM_ENUM("AIF2ADC Mux", aif2adc_enum);
@@ -1542,14 +1547,14 @@ static const char *aif3adc_text[] = {
"AIF1ADCDAT", "AIF2ADCDAT", "AIF2DACDAT", "Mono PCM",
};
-static const struct soc_enum wm8994_aif3adc_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 3, aif3adc_text);
+static SOC_ENUM_SINGLE_DECL(wm8994_aif3adc_enum,
+ WM8994_POWER_MANAGEMENT_6, 3, aif3adc_text);
static const struct snd_kcontrol_new wm8994_aif3adc_mux =
SOC_DAPM_ENUM("AIF3ADC Mux", wm8994_aif3adc_enum);
-static const struct soc_enum wm8958_aif3adc_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 4, aif3adc_text);
+static SOC_ENUM_SINGLE_DECL(wm8958_aif3adc_enum,
+ WM8994_POWER_MANAGEMENT_6, 3, aif3adc_text);
static const struct snd_kcontrol_new wm8958_aif3adc_mux =
SOC_DAPM_ENUM("AIF3ADC Mux", wm8958_aif3adc_enum);
@@ -1558,8 +1563,8 @@ static const char *mono_pcm_out_text[] = {
"None", "AIF2ADCL", "AIF2ADCR",
};
-static const struct soc_enum mono_pcm_out_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 9, 3, mono_pcm_out_text);
+static SOC_ENUM_SINGLE_DECL(mono_pcm_out_enum,
+ WM8994_POWER_MANAGEMENT_6, 9, mono_pcm_out_text);
static const struct snd_kcontrol_new mono_pcm_out_mux =
SOC_DAPM_ENUM("Mono PCM Out Mux", mono_pcm_out_enum);
@@ -1569,14 +1574,14 @@ static const char *aif2dac_src_text[] = {
};
/* Note that these two control shouldn't be simultaneously switched to AIF3 */
-static const struct soc_enum aif2dacl_src_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 7, 2, aif2dac_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2dacl_src_enum,
+ WM8994_POWER_MANAGEMENT_6, 7, aif2dac_src_text);
static const struct snd_kcontrol_new aif2dacl_src_mux =
SOC_DAPM_ENUM("AIF2DACL Mux", aif2dacl_src_enum);
-static const struct soc_enum aif2dacr_src_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 8, 2, aif2dac_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2dacr_src_enum,
+ WM8994_POWER_MANAGEMENT_6, 8, aif2dac_src_text);
static const struct snd_kcontrol_new aif2dacr_src_mux =
SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum);
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 70ff3772079f..5e3bc3c6801a 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -399,6 +399,7 @@ static struct platform_driver davinci_evm_driver = {
.driver = {
.name = "davinci_evm",
.owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
.of_match_table = of_match_ptr(davinci_evm_dt_ids),
},
};
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index b7858bfa0295..670afa29e30d 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -263,7 +263,9 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+ int ret = 0;
+ pm_runtime_get_sync(mcasp->dev);
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_B:
case SND_SOC_DAIFMT_AC97:
@@ -317,7 +319,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
break;
default:
- return -EINVAL;
+ ret = -EINVAL;
+ goto out;
}
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -354,10 +357,12 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
break;
default:
- return -EINVAL;
+ ret = -EINVAL;
+ break;
}
-
- return 0;
+out:
+ pm_runtime_put_sync(mcasp->dev);
+ return ret;
}
static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
@@ -448,7 +453,7 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
return 0;
}
-static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream,
+static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
int channels)
{
int i;
@@ -524,12 +529,18 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream,
return 0;
}
-static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream)
+static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream)
{
int i, active_slots;
u32 mask = 0;
u32 busel = 0;
+ if ((mcasp->tdm_slots < 2) || (mcasp->tdm_slots > 32)) {
+ dev_err(mcasp->dev, "tdm slot %d not supported\n",
+ mcasp->tdm_slots);
+ return -EINVAL;
+ }
+
active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots;
for (i = 0; i < active_slots; i++)
mask |= (1 << i);
@@ -539,35 +550,21 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream)
if (!mcasp->dat_port)
busel = TXSEL;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- /* bit stream is MSB first with no delay */
- /* DSP_B mode */
- mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
- mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
-
- if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32))
- mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
- FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF));
- else
- printk(KERN_ERR "playback tdm slot %d not supported\n",
- mcasp->tdm_slots);
- } else {
- /* bit stream is MSB first with no delay */
- /* DSP_B mode */
- mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
-
- if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32))
- mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
- FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF));
- else
- printk(KERN_ERR "capture tdm slot %d not supported\n",
- mcasp->tdm_slots);
- }
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
+ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
+ mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
+ FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF));
+
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
+ mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
+ mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
+ FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF));
+
+ return 0;
}
/* S/PDIF */
-static void davinci_hw_dit_param(struct davinci_mcasp *mcasp)
+static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp)
{
/* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0
and LSB first */
@@ -589,6 +586,8 @@ static void davinci_hw_dit_param(struct davinci_mcasp *mcasp)
/* Enable the DIT */
mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN);
+
+ return 0;
}
static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
@@ -605,13 +604,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
u8 slots = mcasp->tdm_slots;
u8 active_serializers;
int channels;
+ int ret;
struct snd_interval *pcm_channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
channels = pcm_channels->min;
active_serializers = (channels + slots - 1) / slots;
- if (davinci_hw_common_param(mcasp, substream->stream, channels) == -EINVAL)
+ if (mcasp_common_hw_param(mcasp, substream->stream, channels) == -EINVAL)
return -EINVAL;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
fifo_level = mcasp->txnumevt * active_serializers;
@@ -619,9 +619,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
fifo_level = mcasp->rxnumevt * active_serializers;
if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
- davinci_hw_dit_param(mcasp);
+ ret = mcasp_dit_hw_param(mcasp);
else
- davinci_hw_param(mcasp, substream->stream);
+ ret = mcasp_i2s_hw_param(mcasp, substream->stream);
+
+ if (ret)
+ return ret;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_U8:
@@ -678,19 +681,9 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- ret = pm_runtime_get_sync(mcasp->dev);
- if (IS_ERR_VALUE(ret))
- dev_err(mcasp->dev, "pm_runtime_get_sync() failed\n");
davinci_mcasp_start(mcasp, substream->stream);
break;
-
case SNDRV_PCM_TRIGGER_SUSPEND:
- davinci_mcasp_stop(mcasp, substream->stream);
- ret = pm_runtime_put_sync(mcasp->dev);
- if (IS_ERR_VALUE(ret))
- dev_err(mcasp->dev, "pm_runtime_put_sync() failed\n");
- break;
-
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
davinci_mcasp_stop(mcasp, substream->stream);
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index d0c72ed261e7..c84026c99134 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -326,7 +326,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA,
ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask));
regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB,
- ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(tx_mask));
+ ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask));
regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR,
ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots));
@@ -334,7 +334,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA,
ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask));
regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB,
- ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(rx_mask));
+ ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask));
esai_priv->slot_width = slot_width;
diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h
index 9c9f957fcae1..75e14033e8d8 100644
--- a/sound/soc/fsl/fsl_esai.h
+++ b/sound/soc/fsl/fsl_esai.h
@@ -322,7 +322,7 @@
#define ESAI_xSMB_xS_SHIFT 0
#define ESAI_xSMB_xS_WIDTH 16
#define ESAI_xSMB_xS_MASK (((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT)
-#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMA_xS_MASK)
+#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMB_xS_MASK)
/* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */
#define ESAI_PRRC_PDC_SHIFT 0
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index 79cee782dbbf..a2fd7321b5a9 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -160,7 +160,6 @@ static struct platform_driver imx_mc13783_audio_driver = {
.driver = {
.name = "imx_mc13783",
.owner = THIS_MODULE,
- .pm = &snd_soc_pm_ops,
},
.probe = imx_mc13783_probe,
.remove = imx_mc13783_remove
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index f2beae78969f..1cb22dd034eb 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -33,8 +33,7 @@ struct imx_sgtl5000_data {
static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd)
{
- struct imx_sgtl5000_data *data = container_of(rtd->card,
- struct imx_sgtl5000_data, card);
+ struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(rtd->card);
struct device *dev = rtd->card->dev;
int ret;
@@ -159,13 +158,15 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
data->card.dapm_widgets = imx_sgtl5000_dapm_widgets;
data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets);
+ platform_set_drvdata(pdev, &data->card);
+ snd_soc_card_set_drvdata(&data->card, data);
+
ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
goto fail;
}
- platform_set_drvdata(pdev, data);
of_node_put(ssi_np);
of_node_put(codec_np);
@@ -184,7 +185,8 @@ fail:
static int imx_sgtl5000_remove(struct platform_device *pdev)
{
- struct imx_sgtl5000_data *data = platform_get_drvdata(pdev);
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(card);
clk_put(data->codec_clk);
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 3fd76bc391de..3a3d17ce6ba4 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -71,7 +71,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
{
struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
struct imx_priv *priv = &card_priv;
- struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev);
+ struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
struct device *dev = &priv->pdev->dev;
unsigned int pll_out;
int ret;
@@ -137,7 +137,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card)
{
struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
struct imx_priv *priv = &card_priv;
- struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev);
+ struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
struct device *dev = &priv->pdev->dev;
int ret;
@@ -264,13 +264,15 @@ static int imx_wm8962_probe(struct platform_device *pdev)
data->card.late_probe = imx_wm8962_late_probe;
data->card.set_bias_level = imx_wm8962_set_bias_level;
+ platform_set_drvdata(pdev, &data->card);
+ snd_soc_card_set_drvdata(&data->card, data);
+
ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
goto clk_fail;
}
- platform_set_drvdata(pdev, data);
of_node_put(ssi_np);
of_node_put(codec_np);
@@ -289,7 +291,8 @@ fail:
static int imx_wm8962_remove(struct platform_device *pdev)
{
- struct imx_wm8962_data *data = platform_get_drvdata(pdev);
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
if (!IS_ERR(data->codec_clk))
clk_disable_unprepare(data->codec_clk);
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 454f41cfc828..350757400391 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -59,7 +59,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750
select SND_SOC_WM8750
select SND_S3C2412_SOC_I2S
help
- Sat Y if you want to add support for SoC audio on the Jive.
+ Say Y if you want to add support for SoC audio on the Jive.
config SND_SOC_SAMSUNG_SMDK_WM8580
tristate "SoC I2S Audio support for WM8580 on SMDK"
@@ -145,11 +145,11 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380
config SND_SOC_SAMSUNG_SMDK_WM9713
tristate "SoC AC97 Audio support for SMDK with WM9713"
- depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210)
+ depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110)
select SND_SOC_WM9713
select SND_SAMSUNG_AC97
help
- Sat Y if you want to add support for SoC audio on the SMDK.
+ Say Y if you want to add support for SoC audio on the SMDK.
config SND_SOC_SMARTQ
tristate "SoC I2S Audio support for SmartQ board"
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index dc8ff13187f7..b9dc6acbba8c 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1218,7 +1218,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
ret = regulator_allow_bypass(w->regulator, false);
if (ret != 0)
dev_warn(w->dapm->dev,
- "ASoC: Failed to bypass %s: %d\n",
+ "ASoC: Failed to unbypass %s: %d\n",
w->name, ret);
}
@@ -1228,7 +1228,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
ret = regulator_allow_bypass(w->regulator, true);
if (ret != 0)
dev_warn(w->dapm->dev,
- "ASoC: Failed to unbypass %s: %d\n",
+ "ASoC: Failed to bypass %s: %d\n",
w->name, ret);
}
@@ -3210,15 +3210,11 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
const char *pin = (const char *)kcontrol->private_value;
- mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
-
if (ucontrol->value.integer.value[0])
snd_soc_dapm_enable_pin(&card->dapm, pin);
else
snd_soc_dapm_disable_pin(&card->dapm, pin);
- mutex_unlock(&card->dapm_mutex);
-
snd_soc_dapm_sync(&card->dapm);
return 0;
}
@@ -3248,7 +3244,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
ret = regulator_allow_bypass(w->regulator, true);
if (ret != 0)
dev_warn(w->dapm->dev,
- "ASoC: Failed to unbypass %s: %d\n",
+ "ASoC: Failed to bypass %s: %d\n",
w->name, ret);
}
break;
@@ -3767,23 +3763,52 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
}
/**
+ * snd_soc_dapm_enable_pin_unlocked - enable pin.
+ * @dapm: DAPM context
+ * @pin: pin name
+ *
+ * Enables input/output pin and its parents or children widgets iff there is
+ * a valid audio route and active audio stream.
+ *
+ * Requires external locking.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_enable_pin_unlocked(struct snd_soc_dapm_context *dapm,
+ const char *pin)
+{
+ return snd_soc_dapm_set_pin(dapm, pin, 1);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin_unlocked);
+
+/**
* snd_soc_dapm_enable_pin - enable pin.
* @dapm: DAPM context
* @pin: pin name
*
* Enables input/output pin and its parents or children widgets iff there is
* a valid audio route and active audio stream.
+ *
* NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
* do any widget power switching.
*/
int snd_soc_dapm_enable_pin(struct snd_soc_dapm_context *dapm, const char *pin)
{
- return snd_soc_dapm_set_pin(dapm, pin, 1);
+ int ret;
+
+ mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+ ret = snd_soc_dapm_set_pin(dapm, pin, 1);
+
+ mutex_unlock(&dapm->card->dapm_mutex);
+
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
/**
- * snd_soc_dapm_force_enable_pin - force a pin to be enabled
+ * snd_soc_dapm_force_enable_pin_unlocked - force a pin to be enabled
* @dapm: DAPM context
* @pin: pin name
*
@@ -3791,11 +3816,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
* intended for use with microphone bias supplies used in microphone
* jack detection.
*
+ * Requires external locking.
+ *
* NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
* do any widget power switching.
*/
-int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
- const char *pin)
+int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm,
+ const char *pin)
{
struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true);
@@ -3811,25 +3838,103 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
return 0;
}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin_unlocked);
+
+/**
+ * snd_soc_dapm_force_enable_pin - force a pin to be enabled
+ * @dapm: DAPM context
+ * @pin: pin name
+ *
+ * Enables input/output pin regardless of any other state. This is
+ * intended for use with microphone bias supplies used in microphone
+ * jack detection.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
+ const char *pin)
+{
+ int ret;
+
+ mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+ ret = snd_soc_dapm_force_enable_pin_unlocked(dapm, pin);
+
+ mutex_unlock(&dapm->card->dapm_mutex);
+
+ return ret;
+}
EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin);
/**
+ * snd_soc_dapm_disable_pin_unlocked - disable pin.
+ * @dapm: DAPM context
+ * @pin: pin name
+ *
+ * Disables input/output pin and its parents or children widgets.
+ *
+ * Requires external locking.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_disable_pin_unlocked(struct snd_soc_dapm_context *dapm,
+ const char *pin)
+{
+ return snd_soc_dapm_set_pin(dapm, pin, 0);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin_unlocked);
+
+/**
* snd_soc_dapm_disable_pin - disable pin.
* @dapm: DAPM context
* @pin: pin name
*
* Disables input/output pin and its parents or children widgets.
+ *
* NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
* do any widget power switching.
*/
int snd_soc_dapm_disable_pin(struct snd_soc_dapm_context *dapm,
const char *pin)
{
- return snd_soc_dapm_set_pin(dapm, pin, 0);
+ int ret;
+
+ mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+ ret = snd_soc_dapm_set_pin(dapm, pin, 0);
+
+ mutex_unlock(&dapm->card->dapm_mutex);
+
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
/**
+ * snd_soc_dapm_nc_pin_unlocked - permanently disable pin.
+ * @dapm: DAPM context
+ * @pin: pin name
+ *
+ * Marks the specified pin as being not connected, disabling it along
+ * any parent or child widgets. At present this is identical to
+ * snd_soc_dapm_disable_pin() but in future it will be extended to do
+ * additional things such as disabling controls which only affect
+ * paths through the pin.
+ *
+ * Requires external locking.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_nc_pin_unlocked(struct snd_soc_dapm_context *dapm,
+ const char *pin)
+{
+ return snd_soc_dapm_set_pin(dapm, pin, 0);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin_unlocked);
+
+/**
* snd_soc_dapm_nc_pin - permanently disable pin.
* @dapm: DAPM context
* @pin: pin name
@@ -3845,7 +3950,15 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
*/
int snd_soc_dapm_nc_pin(struct snd_soc_dapm_context *dapm, const char *pin)
{
- return snd_soc_dapm_set_pin(dapm, pin, 0);
+ int ret;
+
+ mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+ ret = snd_soc_dapm_set_pin(dapm, pin, 0);
+
+ mutex_unlock(&dapm->card->dapm_mutex);
+
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin);
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
index e0305a148568..9edd68db9f48 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -183,14 +183,16 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev)
irq = platform_get_irq(pdev, 0);
if (irq < 0)
return irq;
+
+ drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL);
+ if (!drvdata)
+ return -ENOMEM;
+
r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
drvdata->base = devm_ioremap_resource(&pdev->dev, r);
if (IS_ERR(drvdata->base))
return PTR_ERR(drvdata->base);
- drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL);
- if (!drvdata)
- return -ENOMEM;
platform_set_drvdata(pdev, drvdata);
drvdata->physbase = r->start;
if (sizeof(drvdata->physbase) > sizeof(r->start) &&
diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig
index de9408b83f75..e05a86b7c0da 100644
--- a/sound/usb/Kconfig
+++ b/sound/usb/Kconfig
@@ -14,6 +14,7 @@ config SND_USB_AUDIO
select SND_HWDEP
select SND_RAWMIDI
select SND_PCM
+ select BITREVERSE
help
Say Y here to include support for USB audio and USB MIDI
devices.
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-13 03:17:46 +0100