Merge branch 'for-2.6.29' into for-2.6.30
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
index 46f9684..9e67632 100644
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -116,6 +116,9 @@
SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
ARRAY_SIZE(wm8731_output_mixer_controls)),
+If you dont want the mixer elements prefixed with the name of the mixer widget,
+you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
+as for SND_SOC_DAPM_MIXER.
2.3 Platform/Machine domain Widgets
-----------------------------------
diff --git a/arch/arm/mach-pxa/e740.c b/arch/arm/mach-pxa/e740.c
index 6d48e00..a6fff78 100644
--- a/arch/arm/mach-pxa/e740.c
+++ b/arch/arm/mach-pxa/e740.c
@@ -135,6 +135,11 @@
/* IrDA */
GPIO38_GPIO | MFP_LPM_DRIVE_HIGH,
+ /* Audio power control */
+ GPIO16_GPIO, /* AC97 codec AVDD2 supply (analogue power) */
+ GPIO40_GPIO, /* Mic amp power */
+ GPIO41_GPIO, /* Headphone amp power */
+
/* PC Card */
GPIO8_GPIO, /* CD0 */
GPIO44_GPIO, /* CD1 */
diff --git a/arch/arm/mach-pxa/e750.c b/arch/arm/mach-pxa/e750.c
index be1ab8e..665066f 100644
--- a/arch/arm/mach-pxa/e750.c
+++ b/arch/arm/mach-pxa/e750.c
@@ -133,6 +133,11 @@
/* IrDA */
GPIO38_GPIO | MFP_LPM_DRIVE_HIGH,
+ /* Audio power control */
+ GPIO4_GPIO, /* Headphone amp power */
+ GPIO7_GPIO, /* Speaker amp power */
+ GPIO37_GPIO, /* Headphone detect */
+
/* PC Card */
GPIO8_GPIO, /* CD0 */
GPIO44_GPIO, /* CD1 */
diff --git a/arch/arm/mach-pxa/include/mach/eseries-gpio.h b/arch/arm/mach-pxa/include/mach/eseries-gpio.h
index efbd2aa..f3e5509 100644
--- a/arch/arm/mach-pxa/include/mach/eseries-gpio.h
+++ b/arch/arm/mach-pxa/include/mach/eseries-gpio.h
@@ -45,6 +45,21 @@
/* e7xx IrDA power control */
#define GPIO_E7XX_IR_OFF 38
+/* e740 audio control GPIOs */
+#define GPIO_E740_WM9705_nAVDD2 16
+#define GPIO_E740_MIC_ON 40
+#define GPIO_E740_AMP_ON 41
+
+/* e750 audio control GPIOs */
+#define GPIO_E750_HP_AMP_OFF 4
+#define GPIO_E750_SPK_AMP_OFF 7
+#define GPIO_E750_HP_DETECT 37
+
+/* e800 audio control GPIOs */
+#define GPIO_E800_HP_DETECT 81
+#define GPIO_E800_HP_AMP_OFF 82
+#define GPIO_E800_SPK_AMP_ON 83
+
/* ASIC related GPIOs */
#define GPIO_ESERIES_TMIO_IRQ 5
#define GPIO_ESERIES_TMIO_PCLR 19
diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h
index af95a1d..d899dc0 100644
--- a/include/linux/mfd/wm8350/audio.h
+++ b/include/linux/mfd/wm8350/audio.h
@@ -490,6 +490,7 @@
/*
* R231 (0xE7) - Jack Status
*/
+#define WM8350_JACK_L_LVL 0x0800
#define WM8350_JACK_R_LVL 0x0400
/*
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 93a4edb..0accdba 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -76,6 +76,11 @@
wcontrols, wncontrols)\
{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols}
+#define SND_SOC_DAPM_MIXER_NAMED_CTL(wname, wreg, wshift, winvert, \
+ wcontrols, wncontrols)\
+{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \
+ .shift = wshift, .invert = winvert, .kcontrols = wcontrols, \
+ .num_kcontrols = wncontrols}
#define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrols = NULL, .num_kcontrols = 0}
@@ -101,6 +106,11 @@
{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \
.event = wevent, .event_flags = wflags}
+#define SND_SOC_DAPM_MIXER_NAMED_CTL_E(wname, wreg, wshift, winvert, \
+ wcontrols, wncontrols, wevent, wflags) \
+{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
+ .invert = winvert, .kcontrols = wcontrols, \
+ .num_kcontrols = wncontrols, .event = wevent, .event_flags = wflags}
#define SND_SOC_DAPM_MICBIAS_E(wname, wreg, wshift, winvert, wevent, wflags) \
{ .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrols = NULL, .num_kcontrols = 0, \
@@ -250,10 +260,10 @@
int snd_soc_dapm_sys_add(struct device *dev);
/* dapm audio pin control and status */
-int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin);
-int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin);
-int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin);
-int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin);
+int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin);
+int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin);
+int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin);
+int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin);
int snd_soc_dapm_sync(struct snd_soc_codec *codec);
/* dapm widget types */
@@ -263,6 +273,7 @@
snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */
snd_soc_dapm_value_mux, /* selects 1 analog signal from many inputs */
snd_soc_dapm_mixer, /* mixes several analog signals together */
+ snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */
snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */
snd_soc_dapm_adc, /* analog to digital converter */
snd_soc_dapm_dac, /* digital to analog converter */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 24593ac..7039343 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -154,6 +154,8 @@
SND_SOC_BIAS_OFF,
};
+struct snd_jack;
+struct snd_soc_card;
struct snd_soc_device;
struct snd_soc_pcm_stream;
struct snd_soc_ops;
@@ -164,6 +166,8 @@
struct snd_soc_codec;
struct soc_enum;
struct snd_soc_ac97_ops;
+struct snd_soc_jack;
+struct snd_soc_jack_pin;
typedef int (*hw_write_t)(void *,const char* ,int);
typedef int (*hw_read_t)(void *,char* ,int);
@@ -184,6 +188,13 @@
int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
const struct snd_pcm_hardware *hw);
+/* Jack reporting */
+int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type,
+ struct snd_soc_jack *jack);
+void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask);
+int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
+ struct snd_soc_jack_pin *pins);
+
/* codec IO */
#define snd_soc_read(codec, reg) codec->read(codec, reg)
#define snd_soc_write(codec, reg, value) codec->write(codec, reg, value)
@@ -203,6 +214,8 @@
*/
struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
void *data, char *long_name);
+int snd_soc_add_controls(struct snd_soc_codec *codec,
+ const struct snd_kcontrol_new *controls, int num_controls);
int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
@@ -237,6 +250,27 @@
int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
+/**
+ * struct snd_soc_jack_pin - Describes a pin to update based on jack detection
+ *
+ * @pin: name of the pin to update
+ * @mask: bits to check for in reported jack status
+ * @invert: if non-zero then pin is enabled when status is not reported
+ */
+struct snd_soc_jack_pin {
+ struct list_head list;
+ const char *pin;
+ int mask;
+ bool invert;
+};
+
+struct snd_soc_jack {
+ struct snd_jack *jack;
+ struct snd_soc_card *card;
+ struct list_head pins;
+ int status;
+};
+
/* SoC PCM stream information */
struct snd_soc_pcm_stream {
char *stream_name;
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index ef025c6..3d2bb6f 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -6,6 +6,7 @@
tristate "ALSA for SoC audio support"
select SND_PCM
select AC97_BUS if SND_SOC_AC97_BUS
+ select SND_JACK if INPUT=y || INPUT=SND
---help---
If you want ASoC support, you should say Y here and also to the
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 86a9b1f..0237879 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,4 @@
-snd-soc-core-objs := soc-core.o soc-dapm.o
+snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
obj-$(CONFIG_SND_SOC) += codecs/
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 3dcdc4e..9ef6b96 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -347,7 +347,7 @@
vma->vm_end - vma->vm_start, vma->vm_page_prot);
}
-struct snd_pcm_ops atmel_pcm_ops = {
+static struct snd_pcm_ops atmel_pcm_ops = {
.open = atmel_pcm_open,
.close = atmel_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index bc8d654..30490a2 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -305,7 +305,7 @@
return 0;
}
-struct snd_pcm_ops au1xpsc_pcm_ops = {
+static struct snd_pcm_ops au1xpsc_pcm_ops = {
.open = au1xpsc_pcm_open,
.close = au1xpsc_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 8067cfa..8cfed1a 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -297,7 +297,7 @@
}
#endif
-struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
+static struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
.open = bf5xx_pcm_open,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = bf5xx_pcm_hw_params,
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index 53d290b..1318c4f 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -184,7 +184,7 @@
return 0 ;
}
-struct snd_pcm_ops bf5xx_pcm_i2s_ops = {
+static struct snd_pcm_ops bf5xx_pcm_i2s_ops = {
.open = bf5xx_pcm_open,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = bf5xx_pcm_hw_params,
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index d0e0d69..cb5fcd6 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -34,6 +34,7 @@
select SND_SOC_WM8903 if I2C
select SND_SOC_WM8971 if I2C
select SND_SOC_WM8990 if I2C
+ select SND_SOC_WM9705 if SND_SOC_AC97_BUS
select SND_SOC_WM9712 if SND_SOC_AC97_BUS
select SND_SOC_WM9713 if SND_SOC_AC97_BUS
help
@@ -144,6 +145,9 @@
config SND_SOC_WM8990
tristate
+config SND_SOC_WM9705
+ tristate
+
config SND_SOC_WM9712
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index c4ddc9a..3664cdc 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -23,6 +23,7 @@
snd-soc-wm8903-objs := wm8903.o
snd-soc-wm8971-objs := wm8971.o
snd-soc-wm8990-objs := wm8990.o
+snd-soc-wm9705-objs := wm9705.o
snd-soc-wm9712-objs := wm9712.o
snd-soc-wm9713-objs := wm9713.o
@@ -51,5 +52,7 @@
obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o
obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o
obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o
+obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o
+obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o
obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index fb53e65..89d4127 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -123,7 +123,6 @@
snd_soc_free_pcms(socdev);
err:
- kfree(socdev->codec->reg_cache);
kfree(socdev->codec);
socdev->codec = NULL;
return ret;
@@ -138,7 +137,6 @@
return 0;
snd_soc_free_pcms(socdev);
- kfree(socdev->codec->reg_cache);
kfree(socdev->codec);
return 0;
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 73fdbb4..faf3587 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -93,20 +93,6 @@
SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
};
-/* add non dapm controls */
-static int ad1980_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(ad1980_snd_ac97_controls); i++) {
- err = snd_ctl_add(codec->card, snd_soc_cnew(
- &ad1980_snd_ac97_controls[i], codec, NULL));
- if (err < 0)
- return err;
- }
- return 0;
-}
-
static unsigned int ac97_read(struct snd_soc_codec *codec,
unsigned int reg)
{
@@ -123,7 +109,7 @@
default:
reg = reg >> 1;
- if (reg >= (ARRAY_SIZE(ad1980_reg)))
+ if (reg >= ARRAY_SIZE(ad1980_reg))
return -EINVAL;
return cache[reg];
@@ -137,7 +123,7 @@
soc_ac97_ops.write(codec->ac97, reg, val);
reg = reg >> 1;
- if (reg < (ARRAY_SIZE(ad1980_reg)))
+ if (reg < ARRAY_SIZE(ad1980_reg))
cache[reg] = val;
return 0;
@@ -269,7 +255,8 @@
ext_status = ac97_read(codec, AC97_EXTENDED_STATUS);
ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
- ad1980_add_controls(codec);
+ snd_soc_add_controls(codec, ad1980_snd_ac97_controls,
+ ARRAY_SIZE(ad1980_snd_ac97_controls));
ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "ad1980: failed to register card\n");
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 81300d8d..f17c363 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -155,21 +155,6 @@
SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0),
};
-/* add non dapm controls */
-static int ak4535_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&ak4535_snd_controls[i], codec, NULL));
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
/* Mono 1 Mixer */
static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = {
SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0),
@@ -510,7 +495,8 @@
/* power on device */
ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- ak4535_add_controls(codec);
+ snd_soc_add_controls(codec, ak4535_snd_controls,
+ ARRAY_SIZE(ak4535_snd_controls));
ak4535_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0) {
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index cac3736..ec7fe3b 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -151,21 +151,6 @@
SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]),
};
-/* add non dapm controls */
-static int ssm2602_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL));
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
/* Output Mixer */
static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0),
@@ -622,7 +607,8 @@
APANA_ENABLE_MIC_BOOST);
ssm2602_write(codec, SSM2602_PWR, 0);
- ssm2602_add_controls(codec);
+ snd_soc_add_controls(codec, ssm2602_snd_controls,
+ ARRAY_SIZE(ssm2602_snd_controls));
ssm2602_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0) {
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index cfdea00..a0e47c1 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -183,24 +183,6 @@
SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
};
-/* add non dapm controls */
-static int tlv320aic23_add_controls(struct snd_soc_codec *codec)
-{
-
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&tlv320aic23_snd_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
-
- return 0;
-
-}
-
/* PGA Mixer controls for Line and Mic switch */
static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
@@ -718,7 +700,8 @@
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1);
- tlv320aic23_add_controls(codec);
+ snd_soc_add_controls(codec, tlv320aic23_snd_controls,
+ ARRAY_SIZE(tlv320aic23_snd_controls));
tlv320aic23_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0) {
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index b47a749..36ab019 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -311,22 +311,6 @@
SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
};
-/* add non dapm controls */
-static int aic3x_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(aic3x_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&aic3x_snd_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
/* Left DAC Mux */
static const struct snd_kcontrol_new aic3x_left_dac_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]);
@@ -1224,7 +1208,8 @@
aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4);
aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4);
- aic3x_add_controls(codec);
+ snd_soc_add_controls(codec, aic3x_snd_controls,
+ ARRAY_SIZE(aic3x_snd_controls));
aic3x_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0) {
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index ea370a4..f530c1e 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -125,6 +125,9 @@
{
u8 *cache = codec->reg_cache;
+ if (reg >= TWL4030_CACHEREGNUM)
+ return -EIO;
+
return cache[reg];
}
@@ -670,22 +673,6 @@
0, 3, 5, 0, input_gain_tlv),
};
-/* add non dapm controls */
-static int twl4030_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&twl4030_snd_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
/* Left channel inputs */
SND_SOC_DAPM_INPUT("MAINMIC"),
@@ -1233,7 +1220,8 @@
/* power on device */
twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- twl4030_add_controls(codec);
+ snd_soc_add_controls(codec, twl4030_snd_controls,
+ ARRAY_SIZE(twl4030_snd_controls));
twl4030_add_widgets(codec);
ret = snd_soc_init_card(socdev);
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index a2c5064..277825d 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -431,39 +431,6 @@
SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
};
-static int uda134x_add_controls(struct snd_soc_codec *codec)
-{
- int err, i, n;
- const struct snd_kcontrol_new *ctrls;
- struct uda134x_platform_data *pd = codec->control_data;
-
- switch (pd->model) {
- case UDA134X_UDA1340:
- case UDA134X_UDA1344:
- n = ARRAY_SIZE(uda1340_snd_controls);
- ctrls = uda1340_snd_controls;
- break;
- case UDA134X_UDA1341:
- n = ARRAY_SIZE(uda1341_snd_controls);
- ctrls = uda1341_snd_controls;
- break;
- default:
- printk(KERN_ERR "%s unkown codec type: %d",
- __func__, pd->model);
- return -EINVAL;
- }
-
- for (i = 0; i < n; i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&ctrls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
struct snd_soc_dai uda134x_dai = {
.name = "UDA134X",
/* playback capabilities */
@@ -572,7 +539,22 @@
goto pcm_err;
}
- ret = uda134x_add_controls(codec);
+ switch (pd->model) {
+ case UDA134X_UDA1340:
+ case UDA134X_UDA1344:
+ ret = snd_soc_add_controls(codec, uda1340_snd_controls,
+ ARRAY_SIZE(uda1340_snd_controls));
+ break;
+ case UDA134X_UDA1341:
+ ret = snd_soc_add_controls(codec, uda1341_snd_controls,
+ ARRAY_SIZE(uda1341_snd_controls));
+ break;
+ default:
+ printk(KERN_ERR "%s unkown codec type: %d",
+ __func__, pd->model);
+ return -EINVAL;
+ }
+
if (ret < 0) {
printk(KERN_ERR "UDA134X: failed to register controls\n");
goto pcm_err;
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index e6bf084..a957b43 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -271,21 +271,6 @@
SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0),
};
-/* add non dapm controls */
-static int uda1380_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL));
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
/* Input mux */
static const struct snd_kcontrol_new uda1380_input_mux_control =
SOC_DAPM_ENUM("Route", uda1380_input_sel_enum);
@@ -675,7 +660,8 @@
}
/* uda1380 init */
- uda1380_add_controls(codec);
+ snd_soc_add_controls(codec, uda1380_snd_controls,
+ ARRAY_SIZE(uda1380_snd_controls));
uda1380_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0) {
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index e3989d4..2e0db29 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -51,10 +51,17 @@
u16 mute;
};
+struct wm8350_jack_data {
+ struct snd_soc_jack *jack;
+ int report;
+};
+
struct wm8350_data {
struct snd_soc_codec codec;
struct wm8350_output out1;
struct wm8350_output out2;
+ struct wm8350_jack_data hpl;
+ struct wm8350_jack_data hpr;
struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
};
@@ -775,21 +782,6 @@
{"Beep", NULL, "IN3R PGA"},
};
-static int wm8350_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8350_snd_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
static int wm8350_add_widgets(struct snd_soc_codec *codec)
{
int ret;
@@ -1328,6 +1320,95 @@
return 0;
}
+static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data)
+{
+ struct wm8350_data *priv = data;
+ u16 reg;
+ int report;
+ int mask;
+ struct wm8350_jack_data *jack = NULL;
+
+ switch (irq) {
+ case WM8350_IRQ_CODEC_JCK_DET_L:
+ jack = &priv->hpl;
+ mask = WM8350_JACK_L_LVL;
+ break;
+
+ case WM8350_IRQ_CODEC_JCK_DET_R:
+ jack = &priv->hpr;
+ mask = WM8350_JACK_R_LVL;
+ break;
+
+ default:
+ BUG();
+ }
+
+ if (!jack->jack) {
+ dev_warn(wm8350->dev, "Jack interrupt called with no jack\n");
+ return;
+ }
+
+ /* Debounce */
+ msleep(200);
+
+ reg = wm8350_reg_read(wm8350, WM8350_JACK_PIN_STATUS);
+ if (reg & mask)
+ report = jack->report;
+ else
+ report = 0;
+
+ snd_soc_jack_report(jack->jack, report, jack->report);
+}
+
+/**
+ * wm8350_hp_jack_detect - Enable headphone jack detection.
+ *
+ * @codec: WM8350 codec
+ * @which: left or right jack detect signal
+ * @jack: jack to report detection events on
+ * @report: value to report
+ *
+ * Enables the headphone jack detection of the WM8350.
+ */
+int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
+ struct snd_soc_jack *jack, int report)
+{
+ struct wm8350_data *priv = codec->private_data;
+ struct wm8350 *wm8350 = codec->control_data;
+ int irq;
+ int ena;
+
+ switch (which) {
+ case WM8350_JDL:
+ priv->hpl.jack = jack;
+ priv->hpl.report = report;
+ irq = WM8350_IRQ_CODEC_JCK_DET_L;
+ ena = WM8350_JDL_ENA;
+ break;
+
+ case WM8350_JDR:
+ priv->hpr.jack = jack;
+ priv->hpr.report = report;
+ irq = WM8350_IRQ_CODEC_JCK_DET_R;
+ ena = WM8350_JDR_ENA;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
+ wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena);
+
+ /* Sync status */
+ wm8350_hp_jack_handler(wm8350, irq, priv);
+
+ wm8350_unmask_irq(wm8350, irq);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect);
+
static struct snd_soc_codec *wm8350_codec;
static int wm8350_probe(struct platform_device *pdev)
@@ -1381,13 +1462,21 @@
wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
+ wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
+ wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+ wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L,
+ wm8350_hp_jack_handler, priv);
+ wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R,
+ wm8350_hp_jack_handler, priv);
+
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
dev_err(&pdev->dev, "failed to create pcms\n");
return ret;
}
- wm8350_add_controls(codec);
+ snd_soc_add_controls(codec, wm8350_snd_controls,
+ ARRAY_SIZE(wm8350_snd_controls));
wm8350_add_widgets(codec);
wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -1411,8 +1500,21 @@
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
struct wm8350 *wm8350 = codec->control_data;
+ struct wm8350_data *priv = codec->private_data;
int ret;
+ wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
+ WM8350_JDL_ENA | WM8350_JDR_ENA);
+ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
+
+ wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
+ wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+ wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
+ wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+
+ priv->hpl.jack = NULL;
+ priv->hpr.jack = NULL;
+
/* cancel any work waiting to be queued. */
ret = cancel_delayed_work(&codec->delayed_work);
diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h
index cc2887a..d11bd92 100644
--- a/sound/soc/codecs/wm8350.h
+++ b/sound/soc/codecs/wm8350.h
@@ -17,4 +17,12 @@
extern struct snd_soc_dai wm8350_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8350;
+enum wm8350_jack {
+ WM8350_JDL = 1,
+ WM8350_JDR = 2,
+};
+
+int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
+ struct snd_soc_jack *jack, int report);
+
#endif
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 40f8238..abe7cce 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -171,22 +171,6 @@
SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1),
};
-/* add non dapm controls */
-static int wm8510_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8510_snd_controls[i], codec,
- NULL));
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
/* Speaker Output Mixer */
static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0),
@@ -656,7 +640,8 @@
/* power on device */
codec->bias_level = SND_SOC_BIAS_OFF;
wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- wm8510_add_controls(codec);
+ snd_soc_add_controls(codec, wm8510_snd_controls,
+ ARRAY_SIZE(wm8510_snd_controls));
wm8510_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0) {
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index d004e58..3faf0e7 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -200,7 +200,7 @@
unsigned int reg)
{
u16 *cache = codec->reg_cache;
- BUG_ON(reg > ARRAY_SIZE(wm8580_reg));
+ BUG_ON(reg >= ARRAY_SIZE(wm8580_reg));
return cache[reg];
}
@@ -223,7 +223,7 @@
{
u8 data[2];
- BUG_ON(reg > ARRAY_SIZE(wm8580_reg));
+ BUG_ON(reg >= ARRAY_SIZE(wm8580_reg));
/* Registers are 9 bits wide */
value &= 0x1ff;
@@ -330,20 +330,6 @@
SOC_SINGLE("ADC High-Pass Filter Switch", WM8580_ADC_CONTROL1, 4, 1, 0),
};
-/* Add non-DAPM controls */
-static int wm8580_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(wm8580_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8580_snd_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
- return 0;
-}
static const struct snd_soc_dapm_widget wm8580_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DAC1", "Playback", WM8580_PWRDN1, 2, 1),
SND_SOC_DAPM_DAC("DAC2", "Playback", WM8580_PWRDN1, 3, 1),
@@ -866,7 +852,8 @@
goto pcm_err;
}
- wm8580_add_controls(codec);
+ snd_soc_add_controls(codec, wm8580_snd_controls,
+ ARRAY_SIZE(wm8580_snd_controls));
wm8580_add_widgets(codec);
ret = snd_soc_init_card(socdev);
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 80b1198..f90dc52 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -47,7 +47,7 @@
unsigned int reg)
{
u16 *cache = codec->reg_cache;
- BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+ BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults));
return cache[reg];
}
@@ -55,7 +55,7 @@
u16 reg, unsigned int value)
{
u16 *cache = codec->reg_cache;
- BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+ BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults));
cache[reg] = value;
}
@@ -92,21 +92,6 @@
SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0),
};
-static int wm8728_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8728_snd_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
/*
* DAPM controls.
*/
@@ -330,7 +315,8 @@
/* power on device */
wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- wm8728_add_controls(codec);
+ snd_soc_add_controls(codec, wm8728_snd_controls,
+ ARRAY_SIZE(wm8728_snd_controls));
wm8728_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0) {
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index c444b9f..96d6e1a 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -129,22 +129,6 @@
SOC_ENUM("Playback De-emphasis", wm8731_enum[1]),
};
-/* add non dapm controls */
-static int wm8731_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8731_snd_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
/* Output Mixer */
static const struct snd_kcontrol_new wm8731_output_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
@@ -543,7 +527,8 @@
reg = wm8731_read_reg_cache(codec, WM8731_RINVOL);
wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100);
- wm8731_add_controls(codec);
+ snd_soc_add_controls(codec, wm8731_snd_controls,
+ ARRAY_SIZE(wm8731_snd_controls));
wm8731_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0) {
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 5997fa6..1578569 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -231,21 +231,6 @@
};
-/* add non dapm controls */
-static int wm8750_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8750_snd_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
- return 0;
-}
-
/*
* DAPM Controls
*/
@@ -816,7 +801,8 @@
reg = wm8750_read_reg_cache(codec, WM8750_RINVOL);
wm8750_write(codec, WM8750_RINVOL, reg | 0x0100);
- wm8750_add_controls(codec);
+ snd_soc_add_controls(codec, wm8750_snd_controls,
+ ARRAY_SIZE(wm8750_snd_controls));
wm8750_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0) {
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 6c21b50..5a1c1fc 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -97,7 +97,7 @@
unsigned int reg)
{
u16 *cache = codec->reg_cache;
- if (reg < 1 || reg > (ARRAY_SIZE(wm8753_reg) + 1))
+ if (reg < 1 || reg >= (ARRAY_SIZE(wm8753_reg) + 1))
return -1;
return cache[reg - 1];
}
@@ -109,7 +109,7 @@
unsigned int reg, unsigned int value)
{
u16 *cache = codec->reg_cache;
- if (reg < 1 || reg > 0x3f)
+ if (reg < 1 || reg >= (ARRAY_SIZE(wm8753_reg) + 1))
return;
cache[reg - 1] = value;
}
@@ -339,21 +339,6 @@
SOC_ENUM("ROUT2 Phase", wm8753_enum[28]),
};
-/* add non dapm controls */
-static int wm8753_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8753_snd_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
- return 0;
-}
-
/*
* _DAPM_ Controls
*/
@@ -1603,7 +1588,8 @@
reg = wm8753_read_reg_cache(codec, WM8753_RINVOL);
wm8753_write(codec, WM8753_RINVOL, reg | 0x0100);
- wm8753_add_controls(codec);
+ snd_soc_add_controls(codec, wm8753_snd_controls,
+ ARRAY_SIZE(wm8753_snd_controls));
wm8753_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0) {
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 6767de1..1e08d4f 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -517,22 +517,6 @@
};
-/* add non dapm controls */
-static int wm8900_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(wm8900_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8900_snd_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
static const struct snd_kcontrol_new wm8900_dapm_loutput2_control =
SOC_DAPM_SINGLE("LINEOUT2L Switch", WM8900_REG_POWER3, 6, 1, 0);
@@ -1439,7 +1423,8 @@
goto pcm_err;
}
- wm8900_add_controls(codec);
+ snd_soc_add_controls(codec, wm8900_snd_controls,
+ ARRAY_SIZE(wm8900_snd_controls));
wm8900_add_widgets(codec);
ret = snd_soc_init_card(socdev);
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index bde7454..6ff34b9 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -744,21 +744,6 @@
0, 63, 0, out_tlv),
};
-static int wm8903_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(wm8903_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8903_snd_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
static const struct snd_kcontrol_new linput_mode_mux =
SOC_DAPM_ENUM("Left Input Mode Mux", linput_mode_enum);
@@ -1737,7 +1722,8 @@
goto err;
}
- wm8903_add_controls(socdev->codec);
+ snd_soc_add_controls(socdev->codec, wm8903_snd_controls,
+ ARRAY_SIZE(wm8903_snd_controls));
wm8903_add_widgets(socdev->codec);
ret = snd_soc_init_card(socdev);
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 88ead7f..c8bd9b0 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -195,21 +195,6 @@
SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0),
};
-/* add non-DAPM controls */
-static int wm8971_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(wm8971_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8971_snd_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
- return 0;
-}
-
/*
* DAPM Controls
*/
@@ -745,7 +730,8 @@
reg = wm8971_read_reg_cache(codec, WM8971_RINVOL);
wm8971_write(codec, WM8971_RINVOL, reg | 0x0100);
- wm8971_add_controls(codec);
+ snd_soc_add_controls(codec, wm8971_snd_controls,
+ ARRAY_SIZE(wm8971_snd_controls));
wm8971_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0) {
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 5b5afc1..f93c095 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -116,7 +116,7 @@
unsigned int reg)
{
u16 *cache = codec->reg_cache;
- BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1);
+ BUG_ON(reg >= ARRAY_SIZE(wm8990_reg));
return cache[reg];
}
@@ -129,7 +129,7 @@
u16 *cache = codec->reg_cache;
/* Reset register and reserved registers are uncached */
- if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1)
+ if (reg == 0 || reg >= ARRAY_SIZE(wm8990_reg))
return;
cache[reg] = value;
@@ -417,21 +417,6 @@
};
-/* add non dapm controls */
-static int wm8990_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(wm8990_snd_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8990_snd_controls[i], codec,
- NULL));
- if (err < 0)
- return err;
- }
- return 0;
-}
-
/*
* _DAPM_ Controls
*/
@@ -1460,7 +1445,8 @@
wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
- wm8990_add_controls(codec);
+ snd_soc_add_controls(codec, wm8990_snd_controls,
+ ARRAY_SIZE(wm8990_snd_controls));
wm8990_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0) {
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
new file mode 100644
index 0000000..5e1937a
--- /dev/null
+++ b/sound/soc/codecs/wm9705.c
@@ -0,0 +1,410 @@
+/*
+ * wm9705.c -- ALSA Soc WM9705 codec support
+ *
+ * Copyright 2008 Ian Molton <spyro@f2s.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; Version 2 of the License only.
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+/*
+ * WM9705 register cache
+ */
+static const u16 wm9705_reg[] = {
+ 0x6150, 0x8000, 0x8000, 0x8000, /* 0x0 */
+ 0x0000, 0x8000, 0x8008, 0x8008, /* 0x8 */
+ 0x8808, 0x8808, 0x8808, 0x8808, /* 0x10 */
+ 0x8808, 0x0000, 0x8000, 0x0000, /* 0x18 */
+ 0x0000, 0x0000, 0x0000, 0x000f, /* 0x20 */
+ 0x0605, 0x0000, 0xbb80, 0x0000, /* 0x28 */
+ 0x0000, 0xbb80, 0x0000, 0x0000, /* 0x30 */
+ 0x0000, 0x2000, 0x0000, 0x0000, /* 0x38 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 0x40 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 0x48 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 0x50 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 0x58 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 0x60 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 0x68 */
+ 0x0000, 0x0808, 0x0000, 0x0006, /* 0x70 */
+ 0x0000, 0x0000, 0x574d, 0x4c05, /* 0x78 */
+};
+
+static const struct snd_kcontrol_new wm9705_snd_ac97_controls[] = {
+ SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1),
+ SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1),
+ SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
+ SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
+ SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
+ SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1),
+ SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1),
+ SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
+ SOC_SINGLE("PCBeep Playback Volume", AC97_PC_BEEP, 1, 15, 1),
+ SOC_SINGLE("Phone Playback Volume", AC97_PHONE, 0, 31, 1),
+ SOC_DOUBLE("Line Playback Volume", AC97_LINE, 8, 0, 31, 1),
+ SOC_DOUBLE("CD Playback Volume", AC97_CD, 8, 0, 31, 1),
+ SOC_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1),
+ SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 6, 1, 0),
+ SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0),
+ SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1),
+};
+
+static const char *wm9705_mic[] = {"Mic 1", "Mic 2"};
+static const char *wm9705_rec_sel[] = {"Mic", "CD", "NC", "NC",
+ "Line", "Stereo Mix", "Mono Mix", "Phone"};
+
+static const struct soc_enum wm9705_enum_mic =
+ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, wm9705_mic);
+static const struct soc_enum wm9705_enum_rec_l =
+ SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9705_rec_sel);
+static const struct soc_enum wm9705_enum_rec_r =
+ SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9705_rec_sel);
+
+/* Headphone Mixer */
+static const struct snd_kcontrol_new wm9705_hp_mixer_controls[] = {
+ SOC_DAPM_SINGLE("PCBeep Playback Switch", AC97_PC_BEEP, 15, 1, 1),
+ SOC_DAPM_SINGLE("CD Playback Switch", AC97_CD, 15, 1, 1),
+ SOC_DAPM_SINGLE("Mic Playback Switch", AC97_MIC, 15, 1, 1),
+ SOC_DAPM_SINGLE("Phone Playback Switch", AC97_PHONE, 15, 1, 1),
+ SOC_DAPM_SINGLE("Line Playback Switch", AC97_LINE, 15, 1, 1),
+};
+
+/* Mic source */
+static const struct snd_kcontrol_new wm9705_mic_src_controls =
+ SOC_DAPM_ENUM("Route", wm9705_enum_mic);
+
+/* Capture source */
+static const struct snd_kcontrol_new wm9705_capture_selectl_controls =
+ SOC_DAPM_ENUM("Route", wm9705_enum_rec_l);
+static const struct snd_kcontrol_new wm9705_capture_selectr_controls =
+ SOC_DAPM_ENUM("Route", wm9705_enum_rec_r);
+
+/* DAPM widgets */
+static const struct snd_soc_dapm_widget wm9705_dapm_widgets[] = {
+ SND_SOC_DAPM_MUX("Mic Source", SND_SOC_NOPM, 0, 0,
+ &wm9705_mic_src_controls),
+ SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0,
+ &wm9705_capture_selectl_controls),
+ SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0,
+ &wm9705_capture_selectr_controls),
+ SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_MIXER_NAMED_CTL("HP Mixer", SND_SOC_NOPM, 0, 0,
+ &wm9705_hp_mixer_controls[0],
+ ARRAY_SIZE(wm9705_hp_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_PGA("Headphone PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Speaker PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Line PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Line out PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mono PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Phone PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mic PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PCBEEP PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("CD PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("ADC PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("HPOUTL"),
+ SND_SOC_DAPM_OUTPUT("HPOUTR"),
+ SND_SOC_DAPM_OUTPUT("LOUT"),
+ SND_SOC_DAPM_OUTPUT("ROUT"),
+ SND_SOC_DAPM_OUTPUT("MONOOUT"),
+ SND_SOC_DAPM_INPUT("PHONE"),
+ SND_SOC_DAPM_INPUT("LINEINL"),
+ SND_SOC_DAPM_INPUT("LINEINR"),
+ SND_SOC_DAPM_INPUT("CDINL"),
+ SND_SOC_DAPM_INPUT("CDINR"),
+ SND_SOC_DAPM_INPUT("PCBEEP"),
+ SND_SOC_DAPM_INPUT("MIC1"),
+ SND_SOC_DAPM_INPUT("MIC2"),
+};
+
+/* Audio map
+ * WM9705 has no switches to disable the route from the inputs to the HP mixer
+ * so in order to prevent active inputs from forcing the audio outputs to be
+ * constantly enabled, we use the mutes on those inputs to simulate such
+ * controls.
+ */
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* HP mixer */
+ {"HP Mixer", "PCBeep Playback Switch", "PCBEEP PGA"},
+ {"HP Mixer", "CD Playback Switch", "CD PGA"},
+ {"HP Mixer", "Mic Playback Switch", "Mic PGA"},
+ {"HP Mixer", "Phone Playback Switch", "Phone PGA"},
+ {"HP Mixer", "Line Playback Switch", "Line PGA"},
+ {"HP Mixer", NULL, "Left DAC"},
+ {"HP Mixer", NULL, "Right DAC"},
+
+ /* mono mixer */
+ {"Mono Mixer", NULL, "HP Mixer"},
+
+ /* outputs */
+ {"Headphone PGA", NULL, "HP Mixer"},
+ {"HPOUTL", NULL, "Headphone PGA"},
+ {"HPOUTR", NULL, "Headphone PGA"},
+ {"Line out PGA", NULL, "HP Mixer"},
+ {"LOUT", NULL, "Line out PGA"},
+ {"ROUT", NULL, "Line out PGA"},
+ {"Mono PGA", NULL, "Mono Mixer"},
+ {"MONOOUT", NULL, "Mono PGA"},
+
+ /* inputs */
+ {"CD PGA", NULL, "CDINL"},
+ {"CD PGA", NULL, "CDINR"},
+ {"Line PGA", NULL, "LINEINL"},
+ {"Line PGA", NULL, "LINEINR"},
+ {"Phone PGA", NULL, "PHONE"},
+ {"Mic Source", "Mic 1", "MIC1"},
+ {"Mic Source", "Mic 2", "MIC2"},
+ {"Mic PGA", NULL, "Mic Source"},
+ {"PCBEEP PGA", NULL, "PCBEEP"},
+
+ /* Left capture selector */
+ {"Left Capture Source", "Mic", "Mic Source"},
+ {"Left Capture Source", "CD", "CDINL"},
+ {"Left Capture Source", "Line", "LINEINL"},
+ {"Left Capture Source", "Stereo Mix", "HP Mixer"},
+ {"Left Capture Source", "Mono Mix", "HP Mixer"},
+ {"Left Capture Source", "Phone", "PHONE"},
+
+ /* Right capture source */
+ {"Right Capture Source", "Mic", "Mic Source"},
+ {"Right Capture Source", "CD", "CDINR"},
+ {"Right Capture Source", "Line", "LINEINR"},
+ {"Right Capture Source", "Stereo Mix", "HP Mixer"},
+ {"Right Capture Source", "Mono Mix", "HP Mixer"},
+ {"Right Capture Source", "Phone", "PHONE"},
+
+ {"ADC PGA", NULL, "Left Capture Source"},
+ {"ADC PGA", NULL, "Right Capture Source"},
+
+ /* ADC's */
+ {"Left ADC", NULL, "ADC PGA"},
+ {"Right ADC", NULL, "ADC PGA"},
+};
+
+static int wm9705_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets,
+ ARRAY_SIZE(wm9705_dapm_widgets));
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_widgets(codec);
+
+ return 0;
+}
+
+/* We use a register cache to enhance read performance. */
+static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+
+ switch (reg) {
+ case AC97_RESET:
+ case AC97_VENDOR_ID1:
+ case AC97_VENDOR_ID2:
+ return soc_ac97_ops.read(codec->ac97, reg);
+ default:
+ reg = reg >> 1;
+
+ if (reg >= (ARRAY_SIZE(wm9705_reg)))
+ return -EIO;
+
+ return cache[reg];
+ }
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val)
+{
+ u16 *cache = codec->reg_cache;
+
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ reg = reg >> 1;
+ if (reg < (ARRAY_SIZE(wm9705_reg)))
+ cache[reg] = val;
+
+ return 0;
+}
+
+static int ac97_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg;
+ u16 vra;
+
+ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ reg = AC97_PCM_FRONT_DAC_RATE;
+ else
+ reg = AC97_PCM_LR_ADC_RATE;
+
+ return ac97_write(codec, reg, runtime->rate);
+}
+
+#define WM9705_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+struct snd_soc_dai wm9705_dai[] = {
+ {
+ .name = "AC97 HiFi",
+ .ac97_control = 1,
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM9705_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "HiFi Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM9705_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = {
+ .prepare = ac97_prepare,
+ },
+ },
+ {
+ .name = "AC97 Aux",
+ .playback = {
+ .stream_name = "Aux Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = WM9705_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ }
+};
+EXPORT_SYMBOL_GPL(wm9705_dai);
+
+static int wm9705_reset(struct snd_soc_codec *codec)
+{
+ if (soc_ac97_ops.reset) {
+ soc_ac97_ops.reset(codec->ac97);
+ if (ac97_read(codec, 0) == wm9705_reg[0])
+ return 0; /* Success */
+ }
+
+ return -EIO;
+}
+
+static int wm9705_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ printk(KERN_INFO "WM9705 SoC Audio Codec\n");
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->codec == NULL)
+ return -ENOMEM;
+ codec = socdev->codec;
+ mutex_init(&codec->mutex);
+
+ codec->reg_cache = kmemdup(wm9705_reg, sizeof(wm9705_reg), GFP_KERNEL);
+ if (codec->reg_cache == NULL) {
+ ret = -ENOMEM;
+ goto cache_err;
+ }
+ codec->reg_cache_size = sizeof(wm9705_reg);
+ codec->reg_cache_step = 2;
+
+ codec->name = "WM9705";
+ codec->owner = THIS_MODULE;
+ codec->dai = wm9705_dai;
+ codec->num_dai = ARRAY_SIZE(wm9705_dai);
+ codec->write = ac97_write;
+ codec->read = ac97_read;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+ if (ret < 0) {
+ printk(KERN_ERR "wm9705: failed to register AC97 codec\n");
+ goto codec_err;
+ }
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0)
+ goto pcm_err;
+
+ ret = wm9705_reset(codec);
+ if (ret)
+ goto reset_err;
+
+ snd_soc_add_controls(codec, wm9705_snd_ac97_controls,
+ ARRAY_SIZE(wm9705_snd_ac97_controls));
+ wm9705_add_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm9705: failed to register card\n");
+ goto pcm_err;
+ }
+
+ return 0;
+
+reset_err:
+ snd_soc_free_pcms(socdev);
+pcm_err:
+ snd_soc_free_ac97_codec(codec);
+codec_err:
+ kfree(codec->reg_cache);
+cache_err:
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+ return ret;
+}
+
+static int wm9705_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec == NULL)
+ return 0;
+
+ snd_soc_dapm_free(socdev);
+ snd_soc_free_pcms(socdev);
+ snd_soc_free_ac97_codec(codec);
+ kfree(codec->reg_cache);
+ kfree(codec);
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm9705 = {
+ .probe = wm9705_soc_probe,
+ .remove = wm9705_soc_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705);
+
+MODULE_DESCRIPTION("ASoC WM9705 driver");
+MODULE_AUTHOR("Ian Molton");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/wm9705.h b/sound/soc/codecs/wm9705.h
new file mode 100644
index 0000000..d380f11
--- /dev/null
+++ b/sound/soc/codecs/wm9705.h
@@ -0,0 +1,14 @@
+/*
+ * wm9705.h -- WM9705 Soc Audio driver
+ */
+
+#ifndef _WM9705_H
+#define _WM9705_H
+
+#define WM9705_DAI_AC97_HIFI 0
+#define WM9705_DAI_AC97_AUX 1
+
+extern struct snd_soc_dai wm9705_dai[2];
+extern struct snd_soc_codec_device soc_codec_dev_wm9705;
+
+#endif
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index af83d62..4dc90d6 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -154,21 +154,6 @@
SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0),
};
-/* add non dapm controls */
-static int wm9712_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm9712_snd_ac97_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
- return 0;
-}
-
/* We have to create a fake left and right HP mixers because
* the codec only has a single control that is shared by both channels.
* This makes it impossible to determine the audio path.
@@ -467,7 +452,7 @@
else {
reg = reg >> 1;
- if (reg > (ARRAY_SIZE(wm9712_reg)))
+ if (reg >= (ARRAY_SIZE(wm9712_reg)))
return -EIO;
return cache[reg];
@@ -481,7 +466,7 @@
soc_ac97_ops.write(codec->ac97, reg, val);
reg = reg >> 1;
- if (reg <= (ARRAY_SIZE(wm9712_reg)))
+ if (reg < (ARRAY_SIZE(wm9712_reg)))
cache[reg] = val;
return 0;
@@ -698,7 +683,8 @@
ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- wm9712_add_controls(codec);
+ snd_soc_add_controls(codec, wm9712_snd_ac97_controls,
+ ARRAY_SIZE(wm9712_snd_ac97_controls));
wm9712_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0) {
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index f3ca8aa..0e60e16 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -32,7 +32,6 @@
struct wm9713_priv {
u32 pll_in; /* PLL input frequency */
- u32 pll_out; /* PLL output frequency */
};
static unsigned int ac97_read(struct snd_soc_codec *codec,
@@ -190,21 +189,6 @@
SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
};
-/* add non dapm controls */
-static int wm9713_add_controls(struct snd_soc_codec *codec)
-{
- int err, i;
-
- for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm9713_snd_ac97_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
- return 0;
-}
-
/* We have to create a fake left and right HP mixers because
* the codec only has a single control that is shared by both channels.
* This makes it impossible to determine the audio path using the current
@@ -636,7 +620,7 @@
else {
reg = reg >> 1;
- if (reg > (ARRAY_SIZE(wm9713_reg)))
+ if (reg >= (ARRAY_SIZE(wm9713_reg)))
return -EIO;
return cache[reg];
@@ -650,7 +634,7 @@
if (reg < 0x7c)
soc_ac97_ops.write(codec->ac97, reg, val);
reg = reg >> 1;
- if (reg <= (ARRAY_SIZE(wm9713_reg)))
+ if (reg < (ARRAY_SIZE(wm9713_reg)))
cache[reg] = val;
return 0;
@@ -738,13 +722,13 @@
struct _pll_div pll_div;
/* turn PLL off ? */
- if (freq_in == 0 || freq_out == 0) {
+ if (freq_in == 0) {
/* disable PLL power and select ext source */
reg = ac97_read(codec, AC97_HANDSET_RATE);
ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080);
reg = ac97_read(codec, AC97_EXTENDED_MID);
ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200);
- wm9713->pll_out = 0;
+ wm9713->pll_in = 0;
return 0;
}
@@ -788,7 +772,6 @@
ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff);
reg = ac97_read(codec, AC97_HANDSET_RATE);
ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f);
- wm9713->pll_out = freq_out;
wm9713->pll_in = freq_in;
/* wait 10ms AC97 link frames for the link to stabilise */
@@ -1164,8 +1147,8 @@
wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* do we need to re-start the PLL ? */
- if (wm9713->pll_out)
- wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out);
+ if (wm9713->pll_in)
+ wm9713_set_pll(codec, 0, wm9713->pll_in, 0);
/* only synchronise the codec if warm reset failed */
if (ret == 0) {
@@ -1245,7 +1228,8 @@
reg = ac97_read(codec, AC97_CD) & 0x7fff;
ac97_write(codec, AC97_CD, reg);
- wm9713_add_controls(codec);
+ snd_soc_add_controls(codec, wm9713_snd_ac97_controls,
+ ARRAY_SIZE(wm9713_snd_ac97_controls));
wm9713_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0)
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 366049d..7af3b5b 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -286,7 +286,7 @@
runtime->dma_bytes);
}
-struct snd_pcm_ops davinci_pcm_ops = {
+static struct snd_pcm_ops davinci_pcm_ops = {
.open = davinci_pcm_open,
.close = davinci_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
index 4935d1b..50baef1 100644
--- a/sound/soc/davinci/davinci-sffsdr.c
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -25,7 +25,9 @@
#include <asm/dma.h>
#include <asm/mach-types.h>
+#ifdef CONFIG_SFFSDR_FPGA
#include <asm/plat-sffsdr/sffsdr-fpga.h>
+#endif
#include <mach/mcbsp.h>
#include <mach/edma.h>
@@ -43,6 +45,17 @@
int fs;
int ret = 0;
+ /* Fsref can be 32000, 44100 or 48000. */
+ fs = params_rate(params);
+
+#ifndef CONFIG_SFFSDR_FPGA
+ /* Without the FPGA module, the Fs is fixed at 44100 Hz */
+ if (fs != 44100) {
+ pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n");
+ return -EINVAL;
+ }
+#endif
+
/* Set cpu DAI configuration:
* CLKX and CLKR are the inputs for the Sample Rate Generator.
* FSX and FSR are outputs, driven by the sample Rate Generator. */
@@ -53,12 +66,13 @@
if (ret < 0)
return ret;
- /* Fsref can be 32000, 44100 or 48000. */
- fs = params_rate(params);
-
pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs);
+#ifndef CONFIG_SFFSDR_FPGA
+ return 0;
+#else
return sffsdr_fpga_set_codec_fs(fs);
+#endif
}
static struct snd_soc_ops sffsdr_ops = {
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 95c12b2..c7c78c3 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,17 +1,17 @@
config SND_SOC_OF_SIMPLE
tristate
+# ASoC platform support for the Freescale MPC8610 SOC. This compiles drivers
+# for the SSI and the Elo DMA controller. You will still need to select
+# a platform driver and a codec driver.
config SND_SOC_MPC8610
- bool "ALSA SoC support for the MPC8610 SOC"
- depends on MPC8610_HPCD
- default y if MPC8610
- help
- Say Y if you want to add support for codecs attached to the SSI
- device on an MPC8610.
+ tristate
+ depends on MPC8610
config SND_SOC_MPC8610_HPCD
- bool "ALSA SoC support for the Freescale MPC8610 HPCD board"
- depends on SND_SOC_MPC8610
+ tristate "ALSA SoC support for the Freescale MPC8610 HPCD board"
+ depends on MPC8610_HPCD
+ select SND_SOC_MPC8610
select SND_SOC_CS4270
select SND_SOC_CS4270_VD33_ERRATA
default y if MPC8610_HPCD
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 035da4a..f85134c 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -2,10 +2,13 @@
obj-$(CONFIG_SND_SOC_OF_SIMPLE) += soc-of-simple.o
# MPC8610 HPCD Machine Support
-obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o
+snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o
+obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o
# MPC8610 Platform Support
-obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
+snd-soc-fsl-ssi-objs := fsl_ssi.o
+snd-soc-fsl-dma-objs := fsl_dma.o
+obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index b0362df..607a38c 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -264,7 +264,7 @@
runtime->dma_bytes);
}
-struct snd_pcm_ops omap_pcm_ops = {
+static struct snd_pcm_ops omap_pcm_ops = {
.open = omap_pcm_open,
.close = omap_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index f82e106..958ac3f 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -61,6 +61,24 @@
Say Y if you want to add support for SoC audio on Sharp
Zaurus SL-C6000x models (Tosa).
+config SND_PXA2XX_SOC_E740
+ tristate "SoC AC97 Audio support for e740"
+ depends on SND_PXA2XX_SOC && MACH_E740
+ select SND_SOC_WM9705
+ select SND_PXA2XX_SOC_AC97
+ help
+ Say Y if you want to add support for SoC audio on the
+ toshiba e740 PDA
+
+config SND_PXA2XX_SOC_E750
+ tristate "SoC AC97 Audio support for e750"
+ depends on SND_PXA2XX_SOC && MACH_E750
+ select SND_SOC_WM9705
+ select SND_PXA2XX_SOC_AC97
+ help
+ Say Y if you want to add support for SoC audio on the
+ toshiba e750 PDA
+
config SND_PXA2XX_SOC_E800
tristate "SoC AC97 Audio support for e800"
depends on SND_PXA2XX_SOC && MACH_E800
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 08a9f27..97a51a8 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -13,6 +13,8 @@
snd-soc-corgi-objs := corgi.o
snd-soc-poodle-objs := poodle.o
snd-soc-tosa-objs := tosa.o
+snd-soc-e740-objs := e740_wm9705.o
+snd-soc-e750-objs := e750_wm9705.o
snd-soc-e800-objs := e800_wm9712.o
snd-soc-spitz-objs := spitz.o
snd-soc-em-x270-objs := em-x270.o
@@ -22,6 +24,8 @@
obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E740) += snd-soc-e740.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o
obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
new file mode 100644
index 0000000..ac36176
--- /dev/null
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -0,0 +1,213 @@
+/*
+ * e740-wm9705.c -- SoC audio for e740
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; version 2 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm9705.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+
+#define E740_AUDIO_OUT 1
+#define E740_AUDIO_IN 2
+
+static int e740_audio_power;
+
+static void e740_sync_audio_power(int status)
+{
+ gpio_set_value(GPIO_E740_WM9705_nAVDD2, !status);
+ gpio_set_value(GPIO_E740_AMP_ON, (status & E740_AUDIO_OUT) ? 1 : 0);
+ gpio_set_value(GPIO_E740_MIC_ON, (status & E740_AUDIO_IN) ? 1 : 0);
+}
+
+static int e740_mic_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ e740_audio_power |= E740_AUDIO_IN;
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ e740_audio_power &= ~E740_AUDIO_IN;
+
+ e740_sync_audio_power(e740_audio_power);
+
+ return 0;
+}
+
+static int e740_output_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ e740_audio_power |= E740_AUDIO_OUT;
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ e740_audio_power &= ~E740_AUDIO_OUT;
+
+ e740_sync_audio_power(e740_audio_power);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget e740_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+ SND_SOC_DAPM_PGA_E("Output Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e740_output_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("Mic Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e740_mic_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Output Amp", NULL, "LOUT"},
+ {"Output Amp", NULL, "ROUT"},
+ {"Output Amp", NULL, "MONOOUT"},
+
+ {"Speaker", NULL, "Output Amp"},
+ {"Headphone Jack", NULL, "Output Amp"},
+
+ {"MIC1", NULL, "Mic Amp"},
+ {"Mic Amp", NULL, "Mic (Internal)"},
+};
+
+static int e740_ac97_init(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_nc_pin(codec, "HPOUTL");
+ snd_soc_dapm_nc_pin(codec, "HPOUTR");
+ snd_soc_dapm_nc_pin(codec, "PHONE");
+ snd_soc_dapm_nc_pin(codec, "LINEINL");
+ snd_soc_dapm_nc_pin(codec, "LINEINR");
+ snd_soc_dapm_nc_pin(codec, "CDINL");
+ snd_soc_dapm_nc_pin(codec, "CDINR");
+ snd_soc_dapm_nc_pin(codec, "PCBEEP");
+ snd_soc_dapm_nc_pin(codec, "MIC2");
+
+ snd_soc_dapm_new_controls(codec, e740_dapm_widgets,
+ ARRAY_SIZE(e740_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static struct snd_soc_dai_link e740_dai[] = {
+ {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+ .codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI],
+ .init = e740_ac97_init,
+ },
+ {
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+ .codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX],
+ },
+};
+
+static struct snd_soc_card e740 = {
+ .name = "Toshiba e740",
+ .platform = &pxa2xx_soc_platform,
+ .dai_link = e740_dai,
+ .num_links = ARRAY_SIZE(e740_dai),
+};
+
+static struct snd_soc_device e740_snd_devdata = {
+ .card = &e740,
+ .codec_dev = &soc_codec_dev_wm9705,
+};
+
+static struct platform_device *e740_snd_device;
+
+static int __init e740_init(void)
+{
+ int ret;
+
+ if (!machine_is_e740())
+ return -ENODEV;
+
+ ret = gpio_request(GPIO_E740_MIC_ON, "Mic amp");
+ if (ret)
+ return ret;
+
+ ret = gpio_request(GPIO_E740_AMP_ON, "Output amp");
+ if (ret)
+ goto free_mic_amp_gpio;
+
+ ret = gpio_request(GPIO_E740_WM9705_nAVDD2, "Audio power");
+ if (ret)
+ goto free_op_amp_gpio;
+
+ /* Disable audio */
+ ret = gpio_direction_output(GPIO_E740_MIC_ON, 0);
+ if (ret)
+ goto free_apwr_gpio;
+ ret = gpio_direction_output(GPIO_E740_AMP_ON, 0);
+ if (ret)
+ goto free_apwr_gpio;
+ ret = gpio_direction_output(GPIO_E740_WM9705_nAVDD2, 1);
+ if (ret)
+ goto free_apwr_gpio;
+
+ e740_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!e740_snd_device) {
+ ret = -ENOMEM;
+ goto free_apwr_gpio;
+ }
+
+ platform_set_drvdata(e740_snd_device, &e740_snd_devdata);
+ e740_snd_devdata.dev = &e740_snd_device->dev;
+ ret = platform_device_add(e740_snd_device);
+
+ if (!ret)
+ return 0;
+
+/* Fail gracefully */
+ platform_device_put(e740_snd_device);
+free_apwr_gpio:
+ gpio_free(GPIO_E740_WM9705_nAVDD2);
+free_op_amp_gpio:
+ gpio_free(GPIO_E740_AMP_ON);
+free_mic_amp_gpio:
+ gpio_free(GPIO_E740_MIC_ON);
+
+ return ret;
+}
+
+static void __exit e740_exit(void)
+{
+ platform_device_unregister(e740_snd_device);
+}
+
+module_init(e740_init);
+module_exit(e740_exit);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e740");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
new file mode 100644
index 0000000..20fbdcf
--- /dev/null
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -0,0 +1,189 @@
+/*
+ * e750-wm9705.c -- SoC audio for e750
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; version 2 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm9705.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static int e750_spk_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ gpio_set_value(GPIO_E750_SPK_AMP_OFF, 0);
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ gpio_set_value(GPIO_E750_SPK_AMP_OFF, 1);
+
+ return 0;
+}
+
+static int e750_hp_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ gpio_set_value(GPIO_E750_HP_AMP_OFF, 0);
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ gpio_set_value(GPIO_E750_HP_AMP_OFF, 1);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget e750_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+ SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e750_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e750_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Headphone Amp", NULL, "HPOUTL"},
+ {"Headphone Amp", NULL, "HPOUTR"},
+ {"Headphone Jack", NULL, "Headphone Amp"},
+
+ {"Speaker Amp", NULL, "MONOOUT"},
+ {"Speaker", NULL, "Speaker Amp"},
+
+ {"MIC1", NULL, "Mic (Internal)"},
+};
+
+static int e750_ac97_init(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_nc_pin(codec, "LOUT");
+ snd_soc_dapm_nc_pin(codec, "ROUT");
+ snd_soc_dapm_nc_pin(codec, "PHONE");
+ snd_soc_dapm_nc_pin(codec, "LINEINL");
+ snd_soc_dapm_nc_pin(codec, "LINEINR");
+ snd_soc_dapm_nc_pin(codec, "CDINL");
+ snd_soc_dapm_nc_pin(codec, "CDINR");
+ snd_soc_dapm_nc_pin(codec, "PCBEEP");
+ snd_soc_dapm_nc_pin(codec, "MIC2");
+
+ snd_soc_dapm_new_controls(codec, e750_dapm_widgets,
+ ARRAY_SIZE(e750_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static struct snd_soc_dai_link e750_dai[] = {
+ {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+ .codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI],
+ .init = e750_ac97_init,
+ /* use ops to check startup state */
+ },
+ {
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+ .codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX],
+ },
+};
+
+static struct snd_soc_card e750 = {
+ .name = "Toshiba e750",
+ .platform = &pxa2xx_soc_platform,
+ .dai_link = e750_dai,
+ .num_links = ARRAY_SIZE(e750_dai),
+};
+
+static struct snd_soc_device e750_snd_devdata = {
+ .card = &e750,
+ .codec_dev = &soc_codec_dev_wm9705,
+};
+
+static struct platform_device *e750_snd_device;
+
+static int __init e750_init(void)
+{
+ int ret;
+
+ if (!machine_is_e750())
+ return -ENODEV;
+
+ ret = gpio_request(GPIO_E750_HP_AMP_OFF, "Headphone amp");
+ if (ret)
+ return ret;
+
+ ret = gpio_request(GPIO_E750_SPK_AMP_OFF, "Speaker amp");
+ if (ret)
+ goto free_hp_amp_gpio;
+
+ ret = gpio_direction_output(GPIO_E750_HP_AMP_OFF, 1);
+ if (ret)
+ goto free_spk_amp_gpio;
+
+ ret = gpio_direction_output(GPIO_E750_SPK_AMP_OFF, 1);
+ if (ret)
+ goto free_spk_amp_gpio;
+
+ e750_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!e750_snd_device) {
+ ret = -ENOMEM;
+ goto free_spk_amp_gpio;
+ }
+
+ platform_set_drvdata(e750_snd_device, &e750_snd_devdata);
+ e750_snd_devdata.dev = &e750_snd_device->dev;
+ ret = platform_device_add(e750_snd_device);
+
+ if (!ret)
+ return 0;
+
+/* Fail gracefully */
+ platform_device_put(e750_snd_device);
+free_spk_amp_gpio:
+ gpio_free(GPIO_E750_SPK_AMP_OFF);
+free_hp_amp_gpio:
+ gpio_free(GPIO_E750_HP_AMP_OFF);
+
+ return ret;
+}
+
+static void __exit e750_exit(void)
+{
+ platform_device_unregister(e750_snd_device);
+ gpio_free(GPIO_E750_SPK_AMP_OFF);
+ gpio_free(GPIO_E750_HP_AMP_OFF);
+}
+
+module_init(e750_init);
+module_exit(e750_exit);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e750");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 2e3386d..78a1770 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -1,8 +1,6 @@
/*
* e800-wm9712.c -- SoC audio for e800
*
- * Based on tosa.c
- *
* Copyright 2007 (c) Ian Molton <spyro@f2s.com>
*
* This program is free software; you can redistribute it and/or modify it
@@ -13,31 +11,96 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
-#include <linux/device.h>
+#include <linux/gpio.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
-#include <asm/mach-types.h>
#include <mach/pxa-regs.h>
#include <mach/hardware.h>
#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include <asm/mach-types.h>
#include "../codecs/wm9712.h"
#include "pxa2xx-pcm.h"
#include "pxa2xx-ac97.h"
-static struct snd_soc_card e800;
+static int e800_spk_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ gpio_set_value(GPIO_E800_SPK_AMP_ON, 1);
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ gpio_set_value(GPIO_E800_SPK_AMP_ON, 0);
+
+ return 0;
+}
+
+static int e800_hp_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ gpio_set_value(GPIO_E800_HP_AMP_OFF, 0);
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ gpio_set_value(GPIO_E800_HP_AMP_OFF, 1);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget e800_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Mic (Internal1)", NULL),
+ SND_SOC_DAPM_MIC("Mic (Internal2)", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e800_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e800_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Headphone Jack", NULL, "HPOUTL"},
+ {"Headphone Jack", NULL, "HPOUTR"},
+ {"Headphone Jack", NULL, "Headphone Amp"},
+
+ {"Speaker Amp", NULL, "MONOOUT"},
+ {"Speaker", NULL, "Speaker Amp"},
+
+ {"MIC1", NULL, "Mic (Internal1)"},
+ {"MIC2", NULL, "Mic (Internal2)"},
+};
+
+static int e800_ac97_init(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, e800_dapm_widgets,
+ ARRAY_SIZE(e800_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
static struct snd_soc_dai_link e800_dai[] = {
-{
- .name = "AC97 Aux",
- .stream_name = "AC97 Aux",
- .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
- .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
-},
+ {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+ .init = e800_ac97_init,
+ },
+ {
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+ },
};
static struct snd_soc_card e800 = {
@@ -61,6 +124,22 @@
if (!machine_is_e800())
return -ENODEV;
+ ret = gpio_request(GPIO_E800_HP_AMP_OFF, "Headphone amp");
+ if (ret)
+ return ret;
+
+ ret = gpio_request(GPIO_E800_SPK_AMP_ON, "Speaker amp");
+ if (ret)
+ goto free_hp_amp_gpio;
+
+ ret = gpio_direction_output(GPIO_E800_HP_AMP_OFF, 1);
+ if (ret)
+ goto free_spk_amp_gpio;
+
+ ret = gpio_direction_output(GPIO_E800_SPK_AMP_ON, 1);
+ if (ret)
+ goto free_spk_amp_gpio;
+
e800_snd_device = platform_device_alloc("soc-audio", -1);
if (!e800_snd_device)
return -ENOMEM;
@@ -69,8 +148,15 @@
e800_snd_devdata.dev = &e800_snd_device->dev;
ret = platform_device_add(e800_snd_device);
- if (ret)
- platform_device_put(e800_snd_device);
+ if (!ret)
+ return 0;
+
+/* Fail gracefully */
+ platform_device_put(e800_snd_device);
+free_spk_amp_gpio:
+ gpio_free(GPIO_E800_SPK_AMP_ON);
+free_hp_amp_gpio:
+ gpio_free(GPIO_E800_HP_AMP_OFF);
return ret;
}
@@ -78,6 +164,8 @@
static void __exit e800_exit(void)
{
platform_device_unregister(e800_snd_device);
+ gpio_free(GPIO_E800_SPK_AMP_ON);
+ gpio_free(GPIO_E800_HP_AMP_OFF);
}
module_init(e800_init);
@@ -86,4 +174,4 @@
/* Module information */
MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
MODULE_DESCRIPTION("ALSA SoC driver for e800");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index f8e9ecd..8541b67 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -14,6 +14,7 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/device.h>
+#include <linux/clk.h>
#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -26,6 +27,17 @@
#include "pxa2xx-ac97.h"
#include "pxa-ssp.h"
+/*
+ * There is a physical switch SW15 on the board which changes the MCLK
+ * for the WM9713 between the standard AC97 master clock and the
+ * output of the CLK_POUT signal from the PXA.
+ */
+static int clk_pout;
+module_param(clk_pout, int, 0);
+MODULE_PARM_DESC(clk_pout, "Use CLK_POUT as WM9713 MCLK (SW15 on board).");
+
+static struct clk *pout;
+
static struct snd_soc_card zylonite;
static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = {
@@ -61,10 +73,8 @@
static int zylonite_wm9713_init(struct snd_soc_codec *codec)
{
- /* Currently we only support use of the AC97 clock here. If
- * CLK_POUT is selected by SW15 then the clock API will need
- * to be used to request and enable it here.
- */
+ if (clk_pout)
+ snd_soc_dai_set_pll(&codec->dai[0], 0, clk_get_rate(pout), 0);
snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
ARRAY_SIZE(zylonite_dapm_widgets));
@@ -85,7 +95,6 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
- unsigned int pll_out = 0;
unsigned int acds = 0;
unsigned int wm9713_div = 0;
int ret = 0;
@@ -93,16 +102,13 @@
switch (params_rate(params)) {
case 8000:
wm9713_div = 12;
- pll_out = 2048000;
break;
case 16000:
wm9713_div = 6;
- pll_out = 4096000;
break;
case 48000:
default:
wm9713_div = 2;
- pll_out = 12288000;
acds = 1;
break;
}
@@ -123,10 +129,6 @@
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out);
- if (ret < 0)
- return ret;
-
ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds);
if (ret < 0)
return ret;
@@ -135,11 +137,12 @@
if (ret < 0)
return ret;
- /* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs
- * to be set instead.
- */
- ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
- WM9713_PCMDIV(wm9713_div));
+ if (clk_pout)
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV,
+ WM9713_PCMDIV(wm9713_div));
+ else
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
+ WM9713_PCMDIV(wm9713_div));
if (ret < 0)
return ret;
@@ -173,8 +176,72 @@
},
};
+static int zylonite_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ if (clk_pout) {
+ pout = clk_get(NULL, "CLK_POUT");
+ if (IS_ERR(pout)) {
+ dev_err(&pdev->dev, "Unable to obtain CLK_POUT: %ld\n",
+ PTR_ERR(pout));
+ return PTR_ERR(pout);
+ }
+
+ ret = clk_enable(pout);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n",
+ ret);
+ clk_put(pout);
+ return ret;
+ }
+
+ dev_dbg(&pdev->dev, "MCLK enabled at %luHz\n",
+ clk_get_rate(pout));
+ }
+
+ return 0;
+}
+
+static int zylonite_remove(struct platform_device *pdev)
+{
+ if (clk_pout) {
+ clk_disable(pout);
+ clk_put(pout);
+ }
+
+ return 0;
+}
+
+static int zylonite_suspend_post(struct platform_device *pdev,
+ pm_message_t state)
+{
+ if (clk_pout)
+ clk_disable(pout);
+
+ return 0;
+}
+
+static int zylonite_resume_pre(struct platform_device *pdev)
+{
+ int ret = 0;
+
+ if (clk_pout) {
+ ret = clk_enable(pout);
+ if (ret != 0)
+ dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n",
+ ret);
+ }
+
+ return ret;
+}
+
static struct snd_soc_card zylonite = {
.name = "Zylonite",
+ .probe = &zylonite_probe,
+ .remove = &zylonite_remove,
+ .suspend_post = &zylonite_suspend_post,
+ .resume_pre = &zylonite_resume_pre,
.platform = &pxa2xx_soc_platform,
.dai_link = zylonite_dai,
.num_links = ARRAY_SIZE(zylonite_dai),
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 55fdb4a..8313d52 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1495,6 +1495,37 @@
EXPORT_SYMBOL_GPL(snd_soc_cnew);
/**
+ * snd_soc_add_controls - add an array of controls to a codec.
+ * Convienience function to add a list of controls. Many codecs were
+ * duplicating this code.
+ *
+ * @codec: codec to add controls to
+ * @controls: array of controls to add
+ * @num_controls: number of elements in the array
+ *
+ * Return 0 for success, else error.
+ */
+int snd_soc_add_controls(struct snd_soc_codec *codec,
+ const struct snd_kcontrol_new *controls, int num_controls)
+{
+ struct snd_card *card = codec->card;
+ int err, i;
+
+ for (i = 0; i < num_controls; i++) {
+ const struct snd_kcontrol_new *control = &controls[i];
+ err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL));
+ if (err < 0) {
+ dev_err(codec->dev, "%s: Failed to add %s\n",
+ codec->name, control->name);
+ return err;
+ }
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_add_controls);
+
+/**
* snd_soc_info_enum_double - enumerated double mixer info callback
* @kcontrol: mixer control
* @uinfo: control element information
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index a2f1da8..54b4564 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -54,14 +54,15 @@
static int dapm_up_seq[] = {
snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic,
snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_dac,
- snd_soc_dapm_mixer, snd_soc_dapm_pga, snd_soc_dapm_adc, snd_soc_dapm_hp,
- snd_soc_dapm_spk, snd_soc_dapm_post
+ snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_pga,
+ snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post
};
+
static int dapm_down_seq[] = {
snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
- snd_soc_dapm_pga, snd_soc_dapm_mixer, snd_soc_dapm_dac, snd_soc_dapm_mic,
- snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_value_mux,
- snd_soc_dapm_post
+ snd_soc_dapm_pga, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_mixer,
+ snd_soc_dapm_dac, snd_soc_dapm_mic, snd_soc_dapm_micbias,
+ snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_post
};
static int dapm_status = 1;
@@ -101,7 +102,8 @@
{
switch (w->id) {
case snd_soc_dapm_switch:
- case snd_soc_dapm_mixer: {
+ case snd_soc_dapm_mixer:
+ case snd_soc_dapm_mixer_named_ctl: {
int val;
struct soc_mixer_control *mc = (struct soc_mixer_control *)
w->kcontrols[i].private_value;
@@ -323,15 +325,33 @@
if (path->name != (char*)w->kcontrols[i].name)
continue;
- /* add dapm control with long name */
- name_len = 2 + strlen(w->name)
- + strlen(w->kcontrols[i].name);
+ /* add dapm control with long name.
+ * for dapm_mixer this is the concatenation of the
+ * mixer and kcontrol name.
+ * for dapm_mixer_named_ctl this is simply the
+ * kcontrol name.
+ */
+ name_len = strlen(w->kcontrols[i].name) + 1;
+ if (w->id == snd_soc_dapm_mixer)
+ name_len += 1 + strlen(w->name);
+
path->long_name = kmalloc(name_len, GFP_KERNEL);
+
if (path->long_name == NULL)
return -ENOMEM;
- snprintf(path->long_name, name_len, "%s %s",
- w->name, w->kcontrols[i].name);
+ switch (w->id) {
+ case snd_soc_dapm_mixer:
+ default:
+ snprintf(path->long_name, name_len, "%s %s",
+ w->name, w->kcontrols[i].name);
+ break;
+ case snd_soc_dapm_mixer_named_ctl:
+ snprintf(path->long_name, name_len, "%s",
+ w->kcontrols[i].name);
+ break;
+ }
+
path->long_name[name_len - 1] = '\0';
path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
@@ -687,6 +707,7 @@
case snd_soc_dapm_adc:
case snd_soc_dapm_pga:
case snd_soc_dapm_mixer:
+ case snd_soc_dapm_mixer_named_ctl:
if (w->name) {
in = is_connected_input_ep(w);
dapm_clear_walk(w->codec);
@@ -760,6 +781,7 @@
int found = 0;
if (widget->id != snd_soc_dapm_mixer &&
+ widget->id != snd_soc_dapm_mixer_named_ctl &&
widget->id != snd_soc_dapm_switch)
return -ENODEV;
@@ -813,6 +835,7 @@
case snd_soc_dapm_adc:
case snd_soc_dapm_pga:
case snd_soc_dapm_mixer:
+ case snd_soc_dapm_mixer_named_ctl:
if (w->name)
count += sprintf(buf + count, "%s: %s\n",
w->name, w->power ? "On":"Off");
@@ -876,7 +899,7 @@
}
static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec,
- char *pin, int status)
+ const char *pin, int status)
{
struct snd_soc_dapm_widget *w;
@@ -991,6 +1014,7 @@
break;
case snd_soc_dapm_switch:
case snd_soc_dapm_mixer:
+ case snd_soc_dapm_mixer_named_ctl:
ret = dapm_connect_mixer(codec, wsource, wsink, path, control);
if (ret != 0)
goto err;
@@ -1068,6 +1092,7 @@
switch(w->id) {
case snd_soc_dapm_switch:
case snd_soc_dapm_mixer:
+ case snd_soc_dapm_mixer_named_ctl:
dapm_new_mixer(codec, w);
break;
case snd_soc_dapm_mux:
@@ -1549,7 +1574,7 @@
* 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_codec *codec, char *pin)
+int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin)
{
return snd_soc_dapm_set_pin(codec, pin, 1);
}
@@ -1564,7 +1589,7 @@
* 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_codec *codec, char *pin)
+int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin)
{
return snd_soc_dapm_set_pin(codec, pin, 0);
}
@@ -1584,7 +1609,7 @@
* 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(struct snd_soc_codec *codec, char *pin)
+int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin)
{
return snd_soc_dapm_set_pin(codec, pin, 0);
}
@@ -1599,7 +1624,7 @@
*
* Returns 1 for connected otherwise 0.
*/
-int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin)
+int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin)
{
struct snd_soc_dapm_widget *w;
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
new file mode 100644
index 0000000..8cc00c3
--- /dev/null
+++ b/sound/soc/soc-jack.c
@@ -0,0 +1,138 @@
+/*
+ * soc-jack.c -- ALSA SoC jack handling
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <sound/jack.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+/**
+ * snd_soc_jack_new - Create a new jack
+ * @card: ASoC card
+ * @id: an identifying string for this jack
+ * @type: a bitmask of enum snd_jack_type values that can be detected by
+ * this jack
+ * @jack: structure to use for the jack
+ *
+ * Creates a new jack object.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ * On success jack will be initialised.
+ */
+int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type,
+ struct snd_soc_jack *jack)
+{
+ jack->card = card;
+ INIT_LIST_HEAD(&jack->pins);
+
+ return snd_jack_new(card->socdev->codec->card, id, type, &jack->jack);
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_new);
+
+/**
+ * snd_soc_jack_report - Report the current status for a jack
+ *
+ * @jack: the jack
+ * @status: a bitmask of enum snd_jack_type values that are currently detected.
+ * @mask: a bitmask of enum snd_jack_type values that being reported.
+ *
+ * If configured using snd_soc_jack_add_pins() then the associated
+ * DAPM pins will be enabled or disabled as appropriate and DAPM
+ * synchronised.
+ *
+ * Note: This function uses mutexes and should be called from a
+ * context which can sleep (such as a workqueue).
+ */
+void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
+{
+ struct snd_soc_codec *codec = jack->card->socdev->codec;
+ struct snd_soc_jack_pin *pin;
+ int enable;
+ int oldstatus;
+
+ if (!jack) {
+ WARN_ON_ONCE(!jack);
+ return;
+ }
+
+ mutex_lock(&codec->mutex);
+
+ oldstatus = jack->status;
+
+ jack->status &= ~mask;
+ jack->status |= status;
+
+ /* The DAPM sync is expensive enough to be worth skipping */
+ if (jack->status == oldstatus)
+ goto out;
+
+ list_for_each_entry(pin, &jack->pins, list) {
+ enable = pin->mask & status;
+
+ if (pin->invert)
+ enable = !enable;
+
+ if (enable)
+ snd_soc_dapm_enable_pin(codec, pin->pin);
+ else
+ snd_soc_dapm_disable_pin(codec, pin->pin);
+ }
+
+ snd_soc_dapm_sync(codec);
+
+ snd_jack_report(jack->jack, status);
+
+out:
+ mutex_unlock(&codec->mutex);
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_report);
+
+/**
+ * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack
+ *
+ * @jack: ASoC jack
+ * @count: Number of pins
+ * @pins: Array of pins
+ *
+ * After this function has been called the DAPM pins specified in the
+ * pins array will have their status updated to reflect the current
+ * state of the jack whenever the jack status is updated.
+ */
+int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
+ struct snd_soc_jack_pin *pins)
+{
+ int i;
+
+ for (i = 0; i < count; i++) {
+ if (!pins[i].pin) {
+ printk(KERN_ERR "No name for pin %d\n", i);
+ return -EINVAL;
+ }
+ if (!pins[i].mask) {
+ printk(KERN_ERR "No mask for pin %d (%s)\n", i,
+ pins[i].pin);
+ return -EINVAL;
+ }
+
+ INIT_LIST_HEAD(&pins[i].list);
+ list_add(&(pins[i].list), &jack->pins);
+ }
+
+ /* Update to reflect the last reported status; canned jack
+ * implementations are likely to set their state before the
+ * card has an opportunity to associate pins.
+ */
+ snd_soc_jack_report(jack, 0, 0);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins);