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

11 files changed, 619 insertions, 143 deletions

diff --git a/include/sound/tlv320aic3x.h b/include/sound/tlv320aic3x.h
new file mode 100644
index 000000000000..b1a5f34e5cfa
--- /dev/null
+++ b/include/sound/tlv320aic3x.h
@@ -0,0 +1,17 @@
+/*
+ * 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
diff --git a/include/sound/tpa6130a2-plat.h b/include/sound/tpa6130a2-plat.h
index e29fde6b5cbe..426f62767dab 100644
--- a/include/sound/tpa6130a2-plat.h
+++ b/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
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 584bc1e67f76..d57372be7a96 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/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;
 }
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 3eddaec728c1..ad5e2636c944 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -61,6 +61,8 @@
 #define US_TO_SAMPLES(rate, us) \
         (rate / (1000000 / us))
 
+static void dac33_calculate_times(struct snd_pcm_substream *substream);
+static int dac33_prepare_chip(struct snd_pcm_substream *substream);
 
 static struct snd_soc_codec *tlv320dac33_codec;
 
@@ -91,6 +93,7 @@ struct tlv320dac33_priv {
         struct work_struct work;
         struct snd_soc_codec codec;
         struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES];
+        struct snd_pcm_substream *substream;
         int power_gpio;
         int chip_power;
         int irq;
@@ -284,45 +287,47 @@ static int dac33_write16(struct snd_soc_codec *codec, unsigned int reg,
         return ret;
 }
 
-static void dac33_restore_regs(struct snd_soc_codec *codec)
+static void dac33_init_chip(struct snd_soc_codec *codec)
 {
         struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
-        u8 *cache = codec->reg_cache;
-        u8 data[2];
-        int i, ret;
 
-        if (!dac33->chip_power)
+        if (unlikely(!dac33->chip_power))
                 return;
 
-        for (i = DAC33_PWR_CTRL; i <= DAC33_INTP_CTRL_B; i++) {
-                data[0] = i;
-                data[1] = cache[i];
-                /* Skip the read only registers */
-                if ((i >= DAC33_INT_OSC_STATUS &&
-                                i <= DAC33_INT_OSC_FREQ_RAT_READ_B) ||
-                    (i >= DAC33_FIFO_WPTR_MSB && i <= DAC33_FIFO_IRQ_FLAG) ||
-                    i == DAC33_DAC_STATUS_FLAGS ||
-                    i == DAC33_SRC_EST_REF_CLK_RATIO_A ||
-                    i == DAC33_SRC_EST_REF_CLK_RATIO_B)
-                        continue;
-                ret = codec->hw_write(codec->control_data, data, 2);
-                if (ret != 2)
-                        dev_err(codec->dev, "Write failed (%d)\n", ret);
-        }
-        for (i = DAC33_LDAC_PWR_CTRL; i <= DAC33_LINEL_TO_LLO_VOL; i++) {
-                data[0] = i;
-                data[1] = cache[i];
-                ret = codec->hw_write(codec->control_data, data, 2);
-                if (ret != 2)
-                        dev_err(codec->dev, "Write failed (%d)\n", ret);
-        }
-        for (i = DAC33_LINER_TO_RLO_VOL; i <= DAC33_OSC_TRIM; i++) {
-                data[0] = i;
-                data[1] = cache[i];
-                ret = codec->hw_write(codec->control_data, data, 2);
-                if (ret != 2)
-                        dev_err(codec->dev, "Write failed (%d)\n", ret);
-        }
+        /* 44-46: DAC Control Registers */
+        /* A : DAC sample rate Fsref/1.5 */
+        dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(0));
+        /* B : DAC src=normal, not muted */
+        dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT |
+                                             DAC33_DACSRCL_LEFT);
+        /* C : (defaults) */
+        dac33_write(codec, DAC33_DAC_CTRL_C, 0x00);
+
+        /* 73 : volume soft stepping control,
+         clock source = internal osc (?) */
+        dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, DAC33_VOLCLKEN);
+
+        dac33_write(codec, DAC33_PWR_CTRL, DAC33_PDNALLB);
+
+        /* Restore only selected registers (gains mostly) */
+        dac33_write(codec, DAC33_LDAC_DIG_VOL_CTRL,
+                    dac33_read_reg_cache(codec, DAC33_LDAC_DIG_VOL_CTRL));
+        dac33_write(codec, DAC33_RDAC_DIG_VOL_CTRL,
+                    dac33_read_reg_cache(codec, DAC33_RDAC_DIG_VOL_CTRL));
+
+        dac33_write(codec, DAC33_LINEL_TO_LLO_VOL,
+                    dac33_read_reg_cache(codec, DAC33_LINEL_TO_LLO_VOL));
+        dac33_write(codec, DAC33_LINER_TO_RLO_VOL,
+                    dac33_read_reg_cache(codec, DAC33_LINER_TO_RLO_VOL));
+}
+
+static inline void dac33_read_id(struct snd_soc_codec *codec)
+{
+        u8 reg;
+
+        dac33_read(codec, DAC33_DEVICE_ID_MSB, &reg);
+        dac33_read(codec, DAC33_DEVICE_ID_LSB, &reg);
+        dac33_read(codec, DAC33_DEVICE_REV_ID, &reg);
 }
 
 static inline void dac33_soft_power(struct snd_soc_codec *codec, int power)
@@ -341,9 +346,17 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power)
 static int dac33_hard_power(struct snd_soc_codec *codec, int power)
 {
         struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
-        int ret;
+        int ret = 0;
 
         mutex_lock(&dac33->mutex);
+
+        /* Safety check */
+        if (unlikely(power == dac33->chip_power)) {
+                dev_warn(codec->dev, "Trying to set the same power state: %s\n",
+                        power ? "ON" : "OFF");
+                goto exit;
+        }
+
         if (power) {
                 ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies),
                                           dac33->supplies);
@@ -357,11 +370,6 @@ static int dac33_hard_power(struct snd_soc_codec *codec, int power)
                         gpio_set_value(dac33->power_gpio, 1);
 
                 dac33->chip_power = 1;
-
-                /* Restore registers */
-                dac33_restore_regs(codec);
-
-                dac33_soft_power(codec, 1);
         } else {
                 dac33_soft_power(codec, 0);
                 if (dac33->power_gpio >= 0)
@@ -383,6 +391,22 @@ exit:
         return ret;
 }
 
+static int playback_event(struct snd_soc_dapm_widget *w,
+                struct snd_kcontrol *kcontrol, int event)
+{
+        struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(w->codec);
+
+        switch (event) {
+        case SND_SOC_DAPM_PRE_PMU:
+                if (likely(dac33->substream)) {
+                        dac33_calculate_times(dac33->substream);
+                        dac33_prepare_chip(dac33->substream);
+                }
+                break;
+        }
+        return 0;
+}
+
 static int dac33_get_nsample(struct snd_kcontrol *kcontrol,
                          struct snd_ctl_elem_value *ucontrol)
 {
@@ -512,6 +536,8 @@ static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = {
                          DAC33_OUT_AMP_PWR_CTRL, 6, 3, 3, 0),
         SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amp Power",
                          DAC33_OUT_AMP_PWR_CTRL, 4, 3, 3, 0),
+
+        SND_SOC_DAPM_PRE("Prepare Playback", playback_event),
 };
 
 static const struct snd_soc_dapm_route audio_map[] = {
@@ -554,18 +580,18 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec,
                 break;
         case SND_SOC_BIAS_STANDBY:
                 if (codec->bias_level == SND_SOC_BIAS_OFF) {
+                        /* Coming from OFF, switch on the codec */
                         ret = dac33_hard_power(codec, 1);
                         if (ret != 0)
                                 return ret;
-                }
 
-                dac33_soft_power(codec, 0);
+                        dac33_init_chip(codec);
+                }
                 break;
         case SND_SOC_BIAS_OFF:
                 ret = dac33_hard_power(codec, 0);
                 if (ret != 0)
                         return ret;
-
                 break;
         }
         codec->bias_level = level;
@@ -708,6 +734,31 @@ static void dac33_oscwait(struct snd_soc_codec *codec)
                         "internal oscillator calibration failed\n");
 }
 
+static int dac33_startup(struct snd_pcm_substream *substream,
+                           struct snd_soc_dai *dai)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_device *socdev = rtd->socdev;
+        struct snd_soc_codec *codec = socdev->card->codec;
+        struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
+
+        /* Stream started, save the substream pointer */
+        dac33->substream = substream;
+
+        return 0;
+}
+
+static void dac33_shutdown(struct snd_pcm_substream *substream,
+                             struct snd_soc_dai *dai)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_device *socdev = rtd->socdev;
+        struct snd_soc_codec *codec = socdev->card->codec;
+        struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
+
+        dac33->substream = NULL;
+}
+
 static int dac33_hw_params(struct snd_pcm_substream *substream,
                            struct snd_pcm_hw_params *params,
                            struct snd_soc_dai *dai)
@@ -791,6 +842,16 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
         }
 
         mutex_lock(&dac33->mutex);
+
+        if (!dac33->chip_power) {
+                /*
+                 * Chip is not powered yet.
+                 * Do the init in the dac33_set_bias_level later.
+                 */
+                mutex_unlock(&dac33->mutex);
+                return 0;
+        }
+
         dac33_soft_power(codec, 0);
         dac33_soft_power(codec, 1);
 
@@ -997,15 +1058,6 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
 
 }
 
-static int dac33_pcm_prepare(struct snd_pcm_substream *substream,
-                             struct snd_soc_dai *dai)
-{
-        dac33_calculate_times(substream);
-        dac33_prepare_chip(substream);
-
-        return 0;
-}
-
 static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
                              struct snd_soc_dai *dai)
 {
@@ -1269,35 +1321,6 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai,
         return 0;
 }
 
-static void dac33_init_chip(struct snd_soc_codec *codec)
-{
-        /* 44-46: DAC Control Registers */
-        /* A : DAC sample rate Fsref/1.5 */
-        dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(0));
-        /* B : DAC src=normal, not muted */
-        dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT |
-                                             DAC33_DACSRCL_LEFT);
-        /* C : (defaults) */
-        dac33_write(codec, DAC33_DAC_CTRL_C, 0x00);
-
-        /* 64-65 : L&R DAC power control
-         Line In -> OUT 1V/V Gain, DAC -> OUT 4V/V Gain*/
-        dac33_write(codec, DAC33_LDAC_PWR_CTRL, DAC33_LROUT_GAIN(2));
-        dac33_write(codec, DAC33_RDAC_PWR_CTRL, DAC33_LROUT_GAIN(2));
-
-        /* 73 : volume soft stepping control,
-         clock source = internal osc (?) */
-        dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, DAC33_VOLCLKEN);
-
-        /* 66 : LOP/LOM Modes */
-        dac33_write(codec, DAC33_OUT_AMP_CM_CTRL, 0xff);
-
-        /* 68 : LOM inverted from LOP */
-        dac33_write(codec, DAC33_OUT_AMP_CTRL, (3<<2));
-
-        dac33_write(codec, DAC33_PWR_CTRL, DAC33_PDNALLB);
-}
-
 static int dac33_soc_probe(struct platform_device *pdev)
 {
         struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -1311,11 +1334,6 @@ static int dac33_soc_probe(struct platform_device *pdev)
         socdev->card->codec = codec;
         dac33 = snd_soc_codec_get_drvdata(codec);
 
-        /* Power up the codec */
-        dac33_hard_power(codec, 1);
-        /* Set default configuration */
-        dac33_init_chip(codec);
-
         /* register pcms */
         ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
         if (ret < 0) {
@@ -1332,12 +1350,6 @@ static int dac33_soc_probe(struct platform_device *pdev)
 
         dac33_add_widgets(codec);
 
-        /* power on device */
-        dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-        /* Bias level configuration has enabled regulator an extra time */
-        regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies);
-
         return 0;
 
 pcm_err:
@@ -1374,6 +1386,8 @@ static int dac33_soc_resume(struct platform_device *pdev)
         struct snd_soc_codec *codec = socdev->card->codec;
 
         dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+        if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+                dac33_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
         dac33_set_bias_level(codec, codec->suspend_bias_level);
 
         return 0;
@@ -1392,8 +1406,9 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320dac33);
 #define DAC33_FORMATS   SNDRV_PCM_FMTBIT_S16_LE
 
 static struct snd_soc_dai_ops dac33_dai_ops = {
+        .startup        = dac33_startup,
+        .shutdown       = dac33_shutdown,
         .hw_params      = dac33_hw_params,
-        .prepare        = dac33_pcm_prepare,
         .trigger        = dac33_pcm_trigger,
         .delay          = dac33_dai_delay,
         .set_sysclk     = dac33_set_dai_sysclk,
@@ -1447,6 +1462,7 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
         codec->hw_write = (hw_write_t) i2c_master_send;
         codec->bias_level = SND_SOC_BIAS_OFF;
         codec->set_bias_level = dac33_set_bias_level;
+        codec->idle_bias_off = 1;
         codec->dai = &dac33_dai;
         codec->num_dai = 1;
         codec->reg_cache_size = ARRAY_SIZE(dac33_reg);
@@ -1487,8 +1503,6 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
                         goto error_gpio;
                 }
                 gpio_direction_output(dac33->power_gpio, 0);
-        } else {
-                dac33->chip_power = 1;
         }
 
         /* Check if the IRQ number is valid and request it */
@@ -1526,12 +1540,14 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
                 goto err_get;
         }
 
-        ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies),
-                                    dac33->supplies);
+        /* Read the tlv320dac33 ID registers */
+        ret = dac33_hard_power(codec, 1);
         if (ret != 0) {
-                dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
-                goto err_enable;
+                dev_err(codec->dev, "Failed to power up codec: %d\n", ret);
+                goto error_codec;
         }
+        dac33_read_id(codec);
+        dac33_hard_power(codec, 0);
 
         ret = snd_soc_register_codec(codec);
         if (ret != 0) {
@@ -1546,14 +1562,9 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
                 goto error_codec;
         }
 
-        /* Shut down the codec for now */
-        dac33_hard_power(codec, 0);
-
         return ret;
 
 error_codec:
-        regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies);
-err_enable:
         regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies);
 err_get:
         if (dac33->irq >= 0) {
@@ -1577,7 +1588,9 @@ static int __devexit dac33_i2c_remove(struct i2c_client *client)
         struct tlv320dac33_priv *dac33;
 
         dac33 = i2c_get_clientdata(client);
-        dac33_hard_power(&dac33->codec, 0);
+
+        if (unlikely(dac33->chip_power))
+                dac33_hard_power(&dac33->codec, 0);
 
         if (dac33->power_gpio >= 0)
                 gpio_free(dac33->power_gpio);
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 958d49c969ac..31f67b527ca1 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/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 @@ exit:
         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.
@@ -257,10 +305,22 @@ 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,
-                       tpa6130_tlv),
+        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),
 };
 
 /*
@@ -368,13 +428,22 @@ 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));
 
-        return snd_soc_add_controls(codec, tpa6130a2_controls,
-                                ARRAY_SIZE(tpa6130a2_controls));
+        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));
 
 }
 EXPORT_SYMBOL_GPL(tpa6130a2_add_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),
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 2e025a3a2618..b717a03dfacf 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/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"},
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index f11963c21873..83be4a76d2bb 100644
--- a/sound/soc/omap/Kconfig
+++ b/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
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 0bc00ca14b37..3a75755f25e4 100644
--- a/sound/soc/omap/Makefile
+++ b/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
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index de10f76baded..87ce842fa2e8 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/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");
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
new file mode 100644
index 000000000000..47d831ef2dbb
--- /dev/null
+++ b/sound/soc/omap/rx51.c
@@ -0,0 +1,294 @@
+/*
+ * 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");
diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c
index f90a2ac888cf..50a94ee76ecc 100644
--- a/sound/soc/omap/zoom2.c
+++ b/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");