Merge branch 'topic/asoc' of...

Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.35

include/sound/tlv320aic3x.h

+/*
+ * Platform data for Texas Instruments TLV320AIC3x codec
+ *
+ * Author: Jarkko Nikula <jhnikula@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#ifndef __TLV320AIC3x_H__
+#define __TLV320AIC3x_H__
+
+struct aic3x_pdata {
+	int gpio_reset; /* < 0 if not used */
+};
+
+#endif
\ No newline at end of file

include/sound/tpa6130a2-plat.h

@@ -31,6 +31,7 @@ enum tpa_model {
 struct tpa6130a2_platform_data {
 	enum tpa_model id;
 	int power_gpio;
+	int limit_gain;
 };
 
 #endif

sound/soc/codecs/tlv320aic3x.c

@@ -38,6 +38,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
+#include <linux/gpio.h>
 #include <linux/regulator/consumer.h>
 #include <linux/platform_device.h>
 #include <sound/core.h>
@@ -47,6 +48,7 @@
 #include <sound/soc-dapm.h>
 #include <sound/initval.h>
 #include <sound/tlv.h>
+#include <sound/tlv320aic3x.h>
 
 #include "tlv320aic3x.h"
 
@@ -64,6 +66,7 @@ struct aic3x_priv {
 	struct regulator_bulk_data supplies[AIC3X_NUM_SUPPLIES];
 	unsigned int sysclk;
 	int master;
+	int gpio_reset;
 };
 
 /*
@@ -1278,6 +1281,10 @@ static int aic3x_unregister(struct aic3x_priv *aic3x)
 	snd_soc_unregister_dai(&aic3x_dai);
 	snd_soc_unregister_codec(&aic3x->codec);
 
+	if (aic3x->gpio_reset >= 0) {
+		gpio_set_value(aic3x->gpio_reset, 0);
+		gpio_free(aic3x->gpio_reset);
+	}
 	regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), aic3x->supplies);
 	regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies);
 
@@ -1302,6 +1309,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 {
 	struct snd_soc_codec *codec;
 	struct aic3x_priv *aic3x;
+	struct aic3x_pdata *pdata = i2c->dev.platform_data;
 	int ret, i;
 
 	aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL);
@@ -1318,6 +1326,15 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 
 	i2c_set_clientdata(i2c, aic3x);
 
+	aic3x->gpio_reset = -1;
+	if (pdata && pdata->gpio_reset >= 0) {
+		ret = gpio_request(pdata->gpio_reset, "tlv320aic3x reset");
+		if (ret != 0)
+			goto err_gpio;
+		aic3x->gpio_reset = pdata->gpio_reset;
+		gpio_direction_output(aic3x->gpio_reset, 0);
+	}
+
 	for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++)
 		aic3x->supplies[i].supply = aic3x_supply_names[i];
 
@@ -1335,11 +1352,19 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 		goto err_enable;
 	}
 
+	if (aic3x->gpio_reset >= 0) {
+		udelay(1);
+		gpio_set_value(aic3x->gpio_reset, 1);
+	}
+
 	return aic3x_register(codec);
 
 err_enable:
 	regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies);
 err_get:
+	if (aic3x->gpio_reset >= 0)
+		gpio_free(aic3x->gpio_reset);
+err_gpio:
 	kfree(aic3x);
 	return ret;
 }

sound/soc/codecs/tpa6130a2.c

@@ -46,6 +46,9 @@ static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = {
 	"AVdd",
 };
 
+#define TPA6130A2_GAIN_MAX	0x3f
+#define TPA6140A2_GAIN_MAX	0x1f
+
 /* This struct is used to save the context */
 struct tpa6130a2_data {
 	struct mutex mutex;
@@ -53,6 +56,8 @@ struct tpa6130a2_data {
 	struct regulator_bulk_data supplies[TPA6130A2_NUM_SUPPLIES];
 	int power_gpio;
 	unsigned char power_state;
+	enum tpa_model id;
+	int gain_limit;
 };
 
 static int tpa6130a2_i2c_read(int reg)
@@ -175,6 +180,40 @@ static int tpa6130a2_power(int power)
 	return ret;
 }
 
+static int tpa6130a2_info_volsw(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_info *uinfo)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct tpa6130a2_data *data;
+
+	BUG_ON(tpa6130a2_client == NULL);
+	data = i2c_get_clientdata(tpa6130a2_client);
+
+	mutex_lock(&data->mutex);
+	switch (mc->reg) {
+	case TPA6130A2_REG_VOL_MUTE:
+		if (data->gain_limit != mc->max)
+			mc->max = data->gain_limit;
+		break;
+	default:
+		dev_err(&tpa6130a2_client->dev,
+			"Invalid register: 0x02%x\n", mc->reg);
+		goto out;
+	}
+	if (unlikely(mc->max == 1))
+		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	else
+		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = mc->max;
+out:
+	mutex_unlock(&data->mutex);
+	return 0;
+}
+
 static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
@@ -238,6 +277,15 @@ static int tpa6130a2_set_reg(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
+#define SOC_SINGLE_EXT_TLV_TPA(xname, xreg, xshift, xmax, xinvert, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+	.tlv.p = (tlv_array), \
+	.info = tpa6130a2_info_volsw, \
+	.get = tpa6130a2_get_reg, .put = tpa6130a2_set_reg, \
+	.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
+
 /*
  * TPA6130 volume. From -59.5 to 4 dB with increasing step size when going
  * down in gain.
@@ -257,12 +305,24 @@ static const unsigned int tpa6130_tlv[] = {
 };
 
 static const struct snd_kcontrol_new tpa6130a2_controls[] = {
-	SOC_SINGLE_EXT_TLV("TPA6130A2 Headphone Playback Volume",
-		       TPA6130A2_REG_VOL_MUTE, 0, 0x3f, 0,
-		       tpa6130a2_get_reg, tpa6130a2_set_reg,
+	SOC_SINGLE_EXT_TLV_TPA("TPA6130A2 Headphone Playback Volume",
+			TPA6130A2_REG_VOL_MUTE, 0, TPA6130A2_GAIN_MAX, 0,
 			tpa6130_tlv),
 };
 
+static const unsigned int tpa6140_tlv[] = {
+	TLV_DB_RANGE_HEAD(3),
+	0, 8, TLV_DB_SCALE_ITEM(-5900, 400, 0),
+	9, 16, TLV_DB_SCALE_ITEM(-2500, 200, 0),
+	17, 31, TLV_DB_SCALE_ITEM(-1000, 100, 0),
+};
+
+static const struct snd_kcontrol_new tpa6140a2_controls[] = {
+	SOC_SINGLE_EXT_TLV_TPA("TPA6140A2 Headphone Playback Volume",
+			TPA6130A2_REG_VOL_MUTE, 1, TPA6140A2_GAIN_MAX, 0,
+			tpa6140_tlv),
+};
+
 /*
  * Enable or disable channel (left or right)
  * The bit number for mute and amplifier are the same per channel:
@@ -368,11 +428,20 @@ static const struct snd_soc_dapm_route audio_map[] = {
 
 int tpa6130a2_add_controls(struct snd_soc_codec *codec)
 {
+	struct	tpa6130a2_data *data;
+
+	BUG_ON(tpa6130a2_client == NULL);
+	data = i2c_get_clientdata(tpa6130a2_client);
+
 	snd_soc_dapm_new_controls(codec, tpa6130a2_dapm_widgets,
 				ARRAY_SIZE(tpa6130a2_dapm_widgets));
 
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
+	if (data->id == TPA6140A2)
+		return snd_soc_add_controls(codec, tpa6140a2_controls,
+						ARRAY_SIZE(tpa6140a2_controls));
+	else
 		return snd_soc_add_controls(codec, tpa6130a2_controls,
 						ARRAY_SIZE(tpa6130a2_controls));
 
@@ -407,6 +476,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client,
 
 	pdata = client->dev.platform_data;
 	data->power_gpio = pdata->power_gpio;
+	data->id = pdata->id;
 
 	mutex_init(&data->mutex);
 
@@ -425,20 +495,35 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client,
 		gpio_direction_output(data->power_gpio, 0);
 	}
 
-	switch (pdata->id) {
+	switch (data->id) {
 	case TPA6130A2:
 		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
 			data->supplies[i].supply = tpa6130a2_supply_names[i];
+		if (pdata->limit_gain > 0 &&
+		    pdata->limit_gain < TPA6130A2_GAIN_MAX)
+			data->gain_limit = pdata->limit_gain;
+		else
+			data->gain_limit = TPA6130A2_GAIN_MAX;
 		break;
 	case TPA6140A2:
 		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
 			data->supplies[i].supply = tpa6140a2_supply_names[i];;
+		if (pdata->limit_gain > 0 &&
+		    pdata->limit_gain < TPA6140A2_GAIN_MAX)
+			data->gain_limit = pdata->limit_gain;
+		else
+			data->gain_limit = TPA6140A2_GAIN_MAX;
 		break;
 	default:
 		dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n",
 			 pdata->id);
 		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
 			data->supplies[i].supply = tpa6130a2_supply_names[i];
+		if (pdata->limit_gain > 0 &&
+		    pdata->limit_gain < TPA6130A2_GAIN_MAX)
+			data->gain_limit = pdata->limit_gain;
+		else
+			data->gain_limit = TPA6130A2_GAIN_MAX;
 	}
 
 	ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies),

sound/soc/codecs/twl4030.c

@@ -123,6 +123,8 @@ struct twl4030_priv {
 	struct snd_soc_codec codec;
 
 	unsigned int codec_powered;
+
+	/* reference counts of AIF/APLL users */
 	unsigned int apll_enabled;
 
 	struct snd_pcm_substream *master_substream;
@@ -259,22 +261,22 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
 {
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
-	int status;
-
-	if (enable == twl4030->apll_enabled)
-		return;
+	int status = -1;
 
-	if (enable)
-		/* Enable PLL */
-		status = twl4030_codec_enable_resource(TWL4030_CODEC_RES_APLL);
-	else
-		/* Disable PLL */
-		status = twl4030_codec_disable_resource(TWL4030_CODEC_RES_APLL);
+	if (enable) {
+		twl4030->apll_enabled++;
+		if (twl4030->apll_enabled == 1)
+			status = twl4030_codec_enable_resource(
+							TWL4030_CODEC_RES_APLL);
+	} else {
+		twl4030->apll_enabled--;
+		if (!twl4030->apll_enabled)
+			status = twl4030_codec_disable_resource(
+							TWL4030_CODEC_RES_APLL);
+	}
 
 	if (status >= 0)
 		twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status);
-
-	twl4030->apll_enabled = enable;
 }
 
 static void twl4030_power_up(struct snd_soc_codec *codec)
@@ -672,6 +674,31 @@ static int apll_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int aif_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	u8 audio_if;
+
+	audio_if = twl4030_read_reg_cache(w->codec, TWL4030_REG_AUDIO_IF);
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		/* Enable AIF */
+		/* enable the PLL before we use it to clock the DAI */
+		twl4030_apll_enable(w->codec, 1);
+
+		twl4030_write(w->codec, TWL4030_REG_AUDIO_IF,
+						audio_if | TWL4030_AIF_EN);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		/* disable the DAI before we stop it's source PLL */
+		twl4030_write(w->codec, TWL4030_REG_AUDIO_IF,
+						audio_if &  ~TWL4030_AIF_EN);
+		twl4030_apll_enable(w->codec, 0);
+		break;
+	}
+	return 0;
+}
+
 static void headset_ramp(struct snd_soc_codec *codec, int ramp)
 {
 	struct snd_soc_device *socdev = codec->socdev;
@@ -1167,8 +1194,6 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("DIGIMIC1"),
 
 	/* Outputs */
-	SND_SOC_DAPM_OUTPUT("OUTL"),
-	SND_SOC_DAPM_OUTPUT("OUTR"),
 	SND_SOC_DAPM_OUTPUT("EARPIECE"),
 	SND_SOC_DAPM_OUTPUT("PREDRIVEL"),
 	SND_SOC_DAPM_OUTPUT("PREDRIVER"),
@@ -1180,6 +1205,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("HFR"),
 	SND_SOC_DAPM_OUTPUT("VIBRA"),
 
+	/* AIF and APLL clocks for running DAIs (including loopback) */
+	SND_SOC_DAPM_OUTPUT("Virtual HiFi OUT"),
+	SND_SOC_DAPM_INPUT("Virtual HiFi IN"),
+	SND_SOC_DAPM_OUTPUT("Virtual Voice OUT"),
+
 	/* DACs */
 	SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback",
 			SND_SOC_NOPM, 0, 0),
@@ -1243,7 +1273,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event,
 			    SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
 
-	SND_SOC_DAPM_SUPPLY("AIF Enable", TWL4030_REG_AUDIO_IF, 0, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("AIF Enable", SND_SOC_NOPM, 0, 0, aif_event,
+			    SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
 
 	/* Output MIXER controls */
 	/* Earpiece */
@@ -1373,10 +1404,6 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Digital Voice Playback Mixer", NULL, "DAC Voice"},
 
 	/* Supply for the digital part (APLL) */
-	{"Digital R1 Playback Mixer", NULL, "APLL Enable"},
-	{"Digital L1 Playback Mixer", NULL, "APLL Enable"},
-	{"Digital R2 Playback Mixer", NULL, "APLL Enable"},
-	{"Digital L2 Playback Mixer", NULL, "APLL Enable"},
 	{"Digital Voice Playback Mixer", NULL, "APLL Enable"},
 
 	{"Digital R1 Playback Mixer", NULL, "AIF Enable"},
@@ -1450,8 +1477,14 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Vibra Mux", "AudioR2", "DAC Right2"},
 
 	/* outputs */
-	{"OUTL", NULL, "Analog L2 Playback Mixer"},
-	{"OUTR", NULL, "Analog R2 Playback Mixer"},
+	/* Must be always connected (for AIF and APLL) */
+	{"Virtual HiFi OUT", NULL, "Digital L1 Playback Mixer"},
+	{"Virtual HiFi OUT", NULL, "Digital R1 Playback Mixer"},
+	{"Virtual HiFi OUT", NULL, "Digital L2 Playback Mixer"},
+	{"Virtual HiFi OUT", NULL, "Digital R2 Playback Mixer"},
+	/* Must be always connected (for APLL) */
+	{"Virtual Voice OUT", NULL, "Digital Voice Playback Mixer"},
+	/* Physical outputs */
 	{"EARPIECE", NULL, "Earpiece PGA"},
 	{"PREDRIVEL", NULL, "PredriveL PGA"},
 	{"PREDRIVER", NULL, "PredriveR PGA"},
@@ -1465,6 +1498,12 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"VIBRA", NULL, "Vibra Route"},
 
 	/* Capture path */
+	/* Must be always connected (for AIF and APLL) */
+	{"ADC Virtual Left1", NULL, "Virtual HiFi IN"},
+	{"ADC Virtual Right1", NULL, "Virtual HiFi IN"},
+	{"ADC Virtual Left2", NULL, "Virtual HiFi IN"},
+	{"ADC Virtual Right2", NULL, "Virtual HiFi IN"},
+	/* Physical inputs */
 	{"Analog Left", "Main Mic Capture Switch", "MAINMIC"},
 	{"Analog Left", "Headset Mic Capture Switch", "HSMIC"},
 	{"Analog Left", "AUXL Capture Switch", "AUXL"},
@@ -1497,11 +1536,6 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"ADC Virtual Left2", NULL, "TX2 Capture Route"},
 	{"ADC Virtual Right2", NULL, "TX2 Capture Route"},
 
-	{"ADC Virtual Left1", NULL, "APLL Enable"},
-	{"ADC Virtual Right1", NULL, "APLL Enable"},
-	{"ADC Virtual Left2", NULL, "APLL Enable"},
-	{"ADC Virtual Right2", NULL, "APLL Enable"},
-
 	{"ADC Virtual Left1", NULL, "AIF Enable"},
 	{"ADC Virtual Right1", NULL, "AIF Enable"},
 	{"ADC Virtual Left2", NULL, "AIF Enable"},

sound/soc/omap/Kconfig

@@ -18,6 +18,16 @@ config SND_OMAP_SOC_N810
 	help
 	  Say Y if you want to add support for SoC audio on Nokia N810.
 
+config SND_OMAP_SOC_RX51
+	tristate "SoC Audio support for Nokia RX-51"
+	depends on SND_OMAP_SOC && MACH_NOKIA_RX51
+	select OMAP_MCBSP
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC3X
+	help
+	  Say Y if you want to add support for SoC audio on Nokia RX-51
+	  hardware. This is also known as Nokia N900 product.
+
 config SND_OMAP_SOC_AMS_DELTA
 	tristate "SoC Audio support for Amstrad E3 (Delta) videophone"
 	depends on SND_OMAP_SOC && MACH_AMS_DELTA

sound/soc/omap/Makefile

@@ -9,6 +9,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o
 
 # OMAP Machine Support
 snd-soc-n810-objs := n810.o
+snd-soc-rx51-objs := rx51.o
 snd-soc-ams-delta-objs := ams-delta.o
 snd-soc-osk5912-objs := osk5912.o
 snd-soc-overo-objs := overo.o
@@ -22,6 +23,7 @@ snd-soc-zoom2-objs := zoom2.o
 snd-soc-igep0020-objs := igep0020.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
+obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o
 obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o
 obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
 obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o

sound/soc/omap/omap3pandora.c

@@ -188,8 +188,6 @@ static int omap3pandora_out_init(struct snd_soc_codec *codec)
 	int ret;
 
 	/* All TWL4030 output pins are floating */
-	snd_soc_dapm_nc_pin(codec, "OUTL");
-	snd_soc_dapm_nc_pin(codec, "OUTR");
 	snd_soc_dapm_nc_pin(codec, "EARPIECE");
 	snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
 	snd_soc_dapm_nc_pin(codec, "PREDRIVER");

sound/soc/omap/rx51.c

+/*
+ * rx51.c  --  SoC audio for Nokia RX-51
+ *
+ * Copyright (C) 2008 - 2009 Nokia Corporation
+ *
+ * Contact: Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ *          Eduardo Valentin <eduardo.valentin@nokia.com>
+ *          Jarkko Nikula <jhnikula@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/tlv320aic3x.h"
+
+/*
+ * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This
+ * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c
+ */
+#define RX51_SPEAKER_AMP_TWL_GPIO	(192 + 7)
+
+static int rx51_spk_func;
+static int rx51_dmic_func;
+
+static void rx51_ext_control(struct snd_soc_codec *codec)
+{
+	if (rx51_spk_func)
+		snd_soc_dapm_enable_pin(codec, "Ext Spk");
+	else
+		snd_soc_dapm_disable_pin(codec, "Ext Spk");
+	if (rx51_dmic_func)
+		snd_soc_dapm_enable_pin(codec, "DMic");
+	else
+		snd_soc_dapm_disable_pin(codec, "DMic");
+
+	snd_soc_dapm_sync(codec);
+}
+
+static int rx51_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->socdev->card->codec;
+
+	snd_pcm_hw_constraint_minmax(runtime,
+				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+	rx51_ext_control(codec);
+
+	return 0;
+}
+
+static int rx51_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	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;
+	int err;
+
+	/* Set codec DAI configuration */
+	err = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_DSP_A |
+				  SND_SOC_DAIFMT_IB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0)
+		return err;
+
+	/* Set cpu DAI configuration */
+	err = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_DSP_A |
+				  SND_SOC_DAIFMT_IB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0)
+		return err;
+
+	/* Set the codec system clock for DAC and ADC */
+	return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000,
+				      SND_SOC_CLOCK_IN);
+}
+
+static struct snd_soc_ops rx51_ops = {
+	.startup = rx51_startup,
+	.hw_params = rx51_hw_params,
+};
+
+static int rx51_get_spk(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = rx51_spk_func;
+
+	return 0;
+}
+
+static int rx51_set_spk(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (rx51_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	rx51_spk_func = ucontrol->value.integer.value[0];
+	rx51_ext_control(codec);
+
+	return 1;
+}
+
+static int rx51_spk_event(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *k, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		gpio_set_value(RX51_SPEAKER_AMP_TWL_GPIO, 1);
+	else
+		gpio_set_value(RX51_SPEAKER_AMP_TWL_GPIO, 0);
+
+	return 0;
+}
+
+static int rx51_get_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = rx51_dmic_func;
+
+	return 0;
+}
+
+static int rx51_set_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (rx51_dmic_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	rx51_dmic_func = ucontrol->value.integer.value[0];
+	rx51_ext_control(codec);
+
+	return 1;
+}
+
+static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event),
+	SND_SOC_DAPM_MIC("DMic", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Ext Spk", NULL, "HPLOUT"},
+	{"Ext Spk", NULL, "HPROUT"},
+
+	{"DMic Rate 64", NULL, "Mic Bias 2V"},
+	{"Mic Bias 2V", NULL, "DMic"},
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *input_function[] = {"ADC", "Digital Mic"};
+
+static const struct soc_enum rx51_enum[] = {
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
+};
+
+static const struct snd_kcontrol_new aic34_rx51_controls[] = {
+	SOC_ENUM_EXT("Speaker Function", rx51_enum[0],
+		     rx51_get_spk, rx51_set_spk),
+	SOC_ENUM_EXT("Input Select",  rx51_enum[1],
+		     rx51_get_input, rx51_set_input),
+};
+
+static int rx51_aic34_init(struct snd_soc_codec *codec)
+{
+	int err;
+
+	/* Set up NC codec pins */
+	snd_soc_dapm_nc_pin(codec, "MIC3L");
+	snd_soc_dapm_nc_pin(codec, "MIC3R");
+	snd_soc_dapm_nc_pin(codec, "LINE1R");
+
+	/* Add RX-51 specific controls */
+	err = snd_soc_add_controls(codec, aic34_rx51_controls,
+				   ARRAY_SIZE(aic34_rx51_controls));
+	if (err < 0)
+		return err;
+
+	/* Add RX-51 specific widgets */
+	snd_soc_dapm_new_controls(codec, aic34_dapm_widgets,
+				  ARRAY_SIZE(aic34_dapm_widgets));
+
+	/* Set up RX-51 specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link rx51_dai[] = {
+	{
+		.name = "TLV320AIC34",
+		.stream_name = "AIC34",
+		.cpu_dai = &omap_mcbsp_dai[0],
+		.codec_dai = &aic3x_dai,
+		.init = rx51_aic34_init,
+		.ops = &rx51_ops,
+	},
+};
+
+/* Audio private data */
+static struct aic3x_setup_data rx51_aic34_setup = {
+	.gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED,
+	.gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT,
+};
+
+/* Audio card */
+static struct snd_soc_card rx51_sound_card = {
+	.name = "RX-51",
+	.dai_link = rx51_dai,
+	.num_links = ARRAY_SIZE(rx51_dai),
+	.platform = &omap_soc_platform,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device rx51_snd_devdata = {
+	.card = &rx51_sound_card,
+	.codec_dev = &soc_codec_dev_aic3x,
+	.codec_data = &rx51_aic34_setup,
+};
+
+static struct platform_device *rx51_snd_device;
+
+static int __init rx51_soc_init(void)
+{
+	int err;
+
+	if (!machine_is_nokia_rx51())
+		return -ENODEV;
+
+	rx51_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!rx51_snd_device) {
+		err = -ENOMEM;
+		goto err1;
+	}
+
+	platform_set_drvdata(rx51_snd_device, &rx51_snd_devdata);
+	rx51_snd_devdata.dev = &rx51_snd_device->dev;
+	*(unsigned int *)rx51_dai[0].cpu_dai->private_data = 1; /* McBSP2 */
+
+	err = platform_device_add(rx51_snd_device);
+	if (err)
+		goto err2;
+
+	return 0;
+err2:
+	platform_device_put(rx51_snd_device);
+err1:
+
+	return err;
+}
+
+static void __exit rx51_soc_exit(void)
+{
+	platform_device_unregister(rx51_snd_device);
+}
+
+module_init(rx51_soc_init);
+module_exit(rx51_soc_exit);
+
+MODULE_AUTHOR("Nokia Corporation");
+MODULE_DESCRIPTION("ALSA SoC Nokia RX-51");
+MODULE_LICENSE("GPL");

sound/soc/omap/zoom2.c

@@ -181,9 +181,6 @@ static int zoom2_twl4030_init(struct snd_soc_codec *codec)
 	snd_soc_dapm_nc_pin(codec, "CARKITMIC");
 	snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
 	snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
-
-	snd_soc_dapm_nc_pin(codec, "OUTL");
-	snd_soc_dapm_nc_pin(codec, "OUTR");
 	snd_soc_dapm_nc_pin(codec, "EARPIECE");
 	snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
 	snd_soc_dapm_nc_pin(codec, "PREDRIVER");