54 files changed, 1568 insertions, 453 deletions
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
index 46f9684d0b29..9e6763264a2e 100644
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -116,6 +116,9 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
 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 6d48e00f4f0b..a6fff782e7a8 100644
--- a/arch/arm/mach-pxa/e740.c
+++ b/arch/arm/mach-pxa/e740.c
@@ -135,6 +135,11 @@ static unsigned long e740_pin_config[] __initdata = {
        /* 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 be1ab8edb973..665066fd280e 100644
--- a/arch/arm/mach-pxa/e750.c
+++ b/arch/arm/mach-pxa/e750.c
@@ -133,6 +133,11 @@ static unsigned long e750_pin_config[] __initdata = {
        /* 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 efbd2aa9ecec..f3e5509820d7 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 af95a1d2f3a1..d899dc0223ba 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 93a4edb148b5..0accdba211f9 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_set_bias_level(struct snd_soc_device *socdev,
 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_enable_pin(struct snd_soc_codec *codec, const char *pin);
-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);
-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);
-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);
 int snd_soc_dapm_sync(struct snd_soc_codec *codec);
 /* dapm widget types */
@@ -263,6 +273,7 @@ enum snd_soc_dapm_type {
        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 24593ac3ea19..7039343e8a78 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -154,6 +154,8 @@ enum snd_soc_bias_level {
        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_platform;
 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_init_card(struct snd_soc_device *socdev);
 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 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec);
 */
 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_get_volsw_s8(struct snd_kcontrol *kcontrol,
 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 ef025c66cc66..3d2bb6fc6dcc 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -6,6 +6,7 @@ menuconfig SND_SOC
        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 86a9b1f5b0f3..0237879fd412 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 3dcdc4e3cfa0..9ef6b96373f5 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -347,7 +347,7 @@ static int atmel_pcm_mmap(struct snd_pcm_substream *substream,
                       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 bc8d654576c0..30490a259148 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -305,7 +305,7 @@ static int au1xpsc_pcm_close(struct snd_pcm_substream *substream)
        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 8067cfafa3a7..8cfed1a5dcbe 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -297,7 +297,7 @@ static	int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
 }
 #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 53d290b3ea47..1318c4f627b7 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -184,7 +184,7 @@ static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream,
        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 d0e0d691ae51..cb5fcd605acc 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -34,6 +34,7 @@ config SND_SOC_ALL_CODECS
        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_WM8971
 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 c4ddc9aa2bbd..3664cdc300b2 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -23,6 +23,7 @@ snd-soc-wm8900-objs := wm8900.o
 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-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(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 fb53e6511af2..89d41277616d 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -123,7 +123,6 @@ bus_err:
        snd_soc_free_pcms(socdev);
 err:
-        kfree(socdev->codec->reg_cache);
        kfree(socdev->codec);
        socdev->codec = NULL;
        return ret;
@@ -138,7 +137,6 @@ static int ac97_soc_remove(struct platform_device *pdev)
                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 73fdbb4d4a3d..faf358758e13 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -93,20 +93,6 @@ SOC_ENUM("Capture Source", ad1980_cap_src),
 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 @@ static unsigned int ac97_read(struct snd_soc_codec *codec,
        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 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
        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 @@ static int ad1980_soc_probe(struct platform_device *pdev)
        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 81300d8d42ca..f17c363cb1db 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -155,21 +155,6 @@ static const struct snd_kcontrol_new ak4535_snd_controls[] = {
        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 @@ static int ak4535_init(struct snd_soc_device *socdev)
        /* 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 cac373616768..ec7fe3b7b0cb 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -151,21 +151,6 @@ SOC_ENUM("Capture Source", ssm2602_enum[0]),
 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 @@ static int ssm2602_init(struct snd_soc_device *socdev)
                        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 cfdea007c4cb..a0e47c1dcd64 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -183,24 +183,6 @@ static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
        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 @@ static int tlv320aic23_init(struct snd_soc_device *socdev)
        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 b47a749c5ea2..36ab0198ca3f 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -311,22 +311,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
        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 @@ static int aic3x_init(struct snd_soc_device *socdev)
        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 ea370a4f86d5..f530c1e6d9e8 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -125,6 +125,9 @@ static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec,
 {
        u8 *cache = codec->reg_cache;
+        if (reg >= TWL4030_CACHEREGNUM)
+                return -EIO;
        return cache[reg];
 }
@@ -670,22 +673,6 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
                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 @@ static int twl4030_init(struct snd_soc_device *socdev)
        /* 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 a2c5064a774b..277825d155a6 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -431,39 +431,6 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
 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 @@ static int uda134x_soc_probe(struct platform_device *pdev)
                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 e6bf0844fbf3..a957b4365b9d 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -271,21 +271,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = {
        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 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
        }
        /* 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 e3989d406f54..2e0db29b4998 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -51,10 +51,17 @@ struct wm8350_output {
        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 @@ static const struct snd_soc_dapm_route audio_map[] = {
        {"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 @@ static int wm8350_resume(struct platform_device *pdev)
        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 @@ static int wm8350_probe(struct platform_device *pdev)
        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 @@ static int wm8350_remove(struct platform_device *pdev)
        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 cc2887aa6c38..d11bd9288cf9 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 40f8238df717..abe7cce87714 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -171,22 +171,6 @@ SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST,  8, 1, 0),
 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 @@ static int wm8510_init(struct snd_soc_device *socdev)
        /* 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 d004e5845298..3faf0e70ce10 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -200,7 +200,7 @@ static inline unsigned int wm8580_read_reg_cache(struct snd_soc_codec *codec,
        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 @@ static int wm8580_write(struct snd_soc_codec *codec, unsigned int reg,
 {
        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_DOUBLE("ADC Mute Switch", WM8580_ADC_CONTROL1, 0, 1, 1, 0),
 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 @@ static int wm8580_init(struct snd_soc_device *socdev)
                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 80b11983e137..f90dc52e975d 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -47,7 +47,7 @@ static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec,
        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 @@ static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec,
        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_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL,
 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 @@ static int wm8728_init(struct snd_soc_device *socdev)
        /* 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 c444b9f2701e..96d6e1aeaf43 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -129,22 +129,6 @@ SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0),
 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 @@ static int wm8731_init(struct snd_soc_device *socdev)
        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 5997fa68e0d5..1578569793a2 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -231,21 +231,6 @@ SOC_SINGLE("Mono Playback Volume", WM8750_MOUTV, 0, 127, 0),
 };
-/* 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 @@ static int wm8750_init(struct snd_soc_device *socdev)
        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 6c21b50c9375..5a1c1fca120f 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -97,7 +97,7 @@ static inline unsigned int wm8753_read_reg_cache(struct snd_soc_codec *codec,
        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 @@ static inline void wm8753_write_reg_cache(struct snd_soc_codec *codec,
        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("ADC Data Select", wm8753_enum[27]),
 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 @@ static int wm8753_init(struct snd_soc_device *socdev)
        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 6767de10ded0..1e08d4f065f2 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -517,22 +517,6 @@ SOC_SINGLE("LINEOUT2 LP -12dB", WM8900_REG_LOUTMIXCTL1,
 };
-/* 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 @@ static int wm8900_probe(struct platform_device *pdev)
                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 bde74546db4a..6ff34b957dce 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -744,21 +744,6 @@ SOC_DOUBLE_R_TLV("Speaker Volume",
                 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 @@ static int wm8903_probe(struct platform_device *pdev)
                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 88ead7f8dd98..c8bd9b06f330 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -195,21 +195,6 @@ static const struct snd_kcontrol_new wm8971_snd_controls[] = {
        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 @@ static int wm8971_init(struct snd_soc_device *socdev)
        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 5b5afc144478..f93c0955ed9d 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -116,7 +116,7 @@ static inline unsigned int wm8990_read_reg_cache(struct snd_soc_codec *codec,
        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 @@ static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec,
        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 @@ SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
 };
-/* 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 @@ static int wm8990_init(struct snd_soc_device *socdev)
        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 000000000000..5e1937ac0b5e
--- /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 000000000000..d380f110f9e2
--- /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 af83d629078a..4dc90d67530e 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -154,21 +154,6 @@ SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
 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 @@ static unsigned int ac97_read(struct snd_soc_codec *codec,
        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 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
        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 @@ static int wm9712_soc_probe(struct platform_device *pdev)
        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 f3ca8aaf0139..0e60e16973d4 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 Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
 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 @@ static unsigned int ac97_read(struct snd_soc_codec *codec,
        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 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
        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 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
        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 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
        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 @@ static int wm9713_soc_resume(struct platform_device *pdev)
        wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
        /* do we need to re-start the PLL ? */
-        if (wm9713->pll_out)
+        if (wm9713->pll_in)
-                wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out);
+                wm9713_set_pll(codec, 0, wm9713->pll_in, 0);
        /* only synchronise the codec if warm reset failed */
        if (ret == 0) {
@@ -1245,7 +1228,8 @@ static int wm9713_soc_probe(struct platform_device *pdev)
        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 366049d8578c..7af3b5b3a53d 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -286,7 +286,7 @@ static int davinci_pcm_mmap(struct snd_pcm_substream *substream,
                                     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 4935d1bcbd8d..50baef1fe5b4 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 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream,
        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 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream,
        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 95c12b26fe37..c7c78c39cfed 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"
+        tristate
-        depends on MPC8610_HPCD
+        depends on MPC8610
-        default y if MPC8610
-        help
-          Say Y if you want to add support for codecs attached to the SSI
-          device on an MPC8610.
 config SND_SOC_MPC8610_HPCD
-        bool "ALSA SoC support for the Freescale MPC8610 HPCD board"
+        tristate "ALSA SoC support for the Freescale MPC8610 HPCD board"
-        depends on SND_SOC_MPC8610
+        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 035da4afec34..f85134c86387 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 b0362dfd5b71..607a38c7ae48 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -264,7 +264,7 @@ static int omap_pcm_mmap(struct snd_pcm_substream *substream,
                                     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 f82e10699471..958ac3fe15d1 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -61,6 +61,24 @@ config SND_PXA2XX_SOC_TOSA
          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 08a9f2797729..97a51a8c936c 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -13,6 +13,8 @@ obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
 snd-soc-corgi-objs := corgi.o
 snd-soc-poodle-objs := poodle.o
 snd-soc-tosa-objs := tosa.o
+snd-soc-e740-objs := e740_wm9705.o
+snd-soc-e750-objs := e750_wm9705.o
 snd-soc-e800-objs := e800_wm9712.o
 snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
@@ -22,6 +24,8 @@ snd-soc-zylonite-objs := zylonite.o
 obj-$(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 000000000000..ac3617651734
--- /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 000000000000..20fbdcfa9f78
--- /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 2e3386dfa0f0..78a1770b986c 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);
-static struct snd_soc_dai_link e800_dai[] = {
+        return 0;
+}
+static int e800_hp_amp_event(struct snd_soc_dapm_widget *w,
+                                struct snd_kcontrol *kcontrol, int event)
 {
-        .name = "AC97 Aux",
+        if (event & SND_SOC_DAPM_PRE_PMU)
-        .stream_name = "AC97 Aux",
+                gpio_set_value(GPIO_E800_HP_AMP_OFF, 0);
-        .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+        else if (event & SND_SOC_DAPM_POST_PMD)
-        .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+                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",
+                .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 @@ static int __init e800_init(void)
        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 @@ static int __init e800_init(void)
        e800_snd_devdata.dev = &e800_snd_device->dev;
        ret = platform_device_add(e800_snd_device);
-        if (ret)
+        if (!ret)
-                platform_device_put(e800_snd_device);
+                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 int __init e800_init(void)
 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_exit(e800_exit);
 /* 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 f8e9ecd589d3..8541b679f6eb 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 const struct snd_soc_dapm_route audio_map[] = {
 static int zylonite_wm9713_init(struct snd_soc_codec *codec)
 {
-        /* Currently we only support use of the AC97 clock here.  If
+        if (clk_pout)
-         * CLK_POUT is selected by SW15 then the clock API will need
+                snd_soc_dai_set_pll(&codec->dai[0], 0, clk_get_rate(pout), 0);
-         * to be used to request and enable it here.
-         */
        snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
                                  ARRAY_SIZE(zylonite_dapm_widgets));
@@ -85,7 +95,6 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
        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 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
        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 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
        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 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
        if (ret < 0)
                return ret;
-        /* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs
+        if (clk_pout)
-         * to be set instead.
+                ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV,
-         */
+                                             WM9713_PCMDIV(wm9713_div));
-        ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
+        else
-                                     WM9713_PCMDIV(wm9713_div));
+                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 struct snd_soc_dai_link zylonite_dai[] = {
 },
 };
+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 55fdb4abb179..8313d52a6e8c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1495,6 +1495,37 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
 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 a2f1da8b4646..54b4564b82b4 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_mixer, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_pga,
-        snd_soc_dapm_spk, snd_soc_dapm_post
+        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_pga, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_mixer,
-        snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_value_mux,
+        snd_soc_dapm_dac, snd_soc_dapm_mic, snd_soc_dapm_micbias,
-        snd_soc_dapm_post
+        snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_post
 };
 static int dapm_status = 1;
@@ -101,7 +102,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 {
        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 @@ static int dapm_new_mixer(struct snd_soc_codec *codec,
                        if (path->name != (char*)w->kcontrols[i].name)
                                continue;
-                        /* add dapm control with long name */
+                        /* add dapm control with long name.
-                        name_len = 2 + strlen(w->name)
+                         * for dapm_mixer this is the concatenation of the
-                                + strlen(w->kcontrols[i].name);
+                         * 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",
+                        switch (w->id) {
-                                 w->name, w->kcontrols[i].name);
+                        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 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action)
                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 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
        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 @@ static ssize_t dapm_widget_show(struct device *dev,
                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 void dapm_free_widgets(struct snd_soc_codec *codec)
 }
 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 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
                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 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
                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 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
 * 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 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
 * 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 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
 * 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 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin);
 *
 * 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 000000000000..8cc00c3cdf34
--- /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);