17 files changed, 305 insertions, 226 deletions

diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index e00659351a4e..d44463a7b0fa 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810
 
 config SND_OMAP_SOC_RX51
         tristate "SoC Audio support for Nokia RX-51"
-        depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST)
+        depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) && I2C
         select SND_OMAP_SOC_MCBSP
         select SND_SOC_TLV320AIC3X
         select SND_SOC_TPA6130A2
@@ -37,7 +37,7 @@ config SND_OMAP_SOC_RX51
 
 config SND_OMAP_SOC_AMS_DELTA
         tristate "SoC Audio support for Amstrad E3 (Delta) videophone"
-        depends on SND_OMAP_SOC && MACH_AMS_DELTA
+        depends on SND_OMAP_SOC && MACH_AMS_DELTA && TTY
         select SND_OMAP_SOC_MCBSP
         select SND_SOC_CX20442
         help
diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
index 994dcf345975..25a33e9d417a 100644
--- a/sound/soc/omap/am3517evm.c
+++ b/sound/soc/omap/am3517evm.c
@@ -77,7 +77,7 @@ static struct snd_soc_dai_link am3517evm_dai = {
         .stream_name = "AIC23",
         .cpu_dai_name = "omap-mcbsp.1",
         .codec_dai_name = "tlv320aic23-hifi",
-        .platform_name = "omap-pcm-audio",
+        .platform_name = "omap-mcbsp.1",
         .codec_name = "tlv320aic23-codec.2-001a",
         .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
                    SND_SOC_DAIFMT_CBM_CFM,
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 56a5219c0a00..bb243c663e6b 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -38,7 +38,6 @@
 #include "omap-mcbsp.h"
 #include "../codecs/cx20442.h"
 
-
 /* Board specific DAPM widgets */
 static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
         /* Handset */
@@ -90,17 +89,23 @@ static const unsigned short ams_delta_audio_mode_pins[] = {
 
 static unsigned short ams_delta_audio_agc;
 
+/*
+ * Used for passing a codec structure pointer
+ * from the board initialization code to the tty line discipline.
+ */
+static struct snd_soc_codec *cx20442_codec;
+
 static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
                                         struct snd_ctl_elem_value *ucontrol)
 {
-        struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
-        struct snd_soc_dapm_context *dapm = &codec->dapm;
+        struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+        struct snd_soc_dapm_context *dapm = &card->dapm;
         struct soc_enum *control = (struct soc_enum *)kcontrol->private_value;
         unsigned short pins;
         int pin, changed = 0;
 
         /* Refuse any mode changes if we are not able to control the codec. */
-        if (!codec->hw_write)
+        if (!cx20442_codec->hw_write)
                 return -EUNATCH;
 
         if (ucontrol->value.enumerated.item[0] >= control->items)
@@ -166,8 +171,8 @@ static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
 static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
                                         struct snd_ctl_elem_value *ucontrol)
 {
-        struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
-        struct snd_soc_dapm_context *dapm = &codec->dapm;
+        struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+        struct snd_soc_dapm_context *dapm = &card->dapm;
         unsigned short pins, mode;
 
         pins = ((snd_soc_dapm_get_pin_status(dapm, "Mouthpiece") <<
@@ -270,12 +275,6 @@ static void cx81801_timeout(unsigned long data)
                 ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0);
 }
 
-/*
- * Used for passing a codec structure pointer
- * from the board initialization code to the tty line discipline.
- */
-static struct snd_soc_codec *cx20442_codec;
-
 /* Line discipline .open() */
 static int cx81801_open(struct tty_struct *tty)
 {
@@ -302,7 +301,7 @@ static int cx81801_open(struct tty_struct *tty)
 static void cx81801_close(struct tty_struct *tty)
 {
         struct snd_soc_codec *codec = tty->disc_data;
-        struct snd_soc_dapm_context *dapm = &codec->dapm;
+        struct snd_soc_dapm_context *dapm = &codec->card->dapm;
 
         del_timer_sync(&cx81801_timer);
 
@@ -475,15 +474,14 @@ static void ams_delta_shutdown(struct snd_pcm_substream *substream)
 
 static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
 {
-        struct snd_soc_codec *codec = rtd->codec;
-        struct snd_soc_dapm_context *dapm = &codec->dapm;
         struct snd_soc_dai *codec_dai = rtd->codec_dai;
         struct snd_soc_card *card = rtd->card;
+        struct snd_soc_dapm_context *dapm = &card->dapm;
         int ret;
         /* Codec is ready, now add/activate board specific controls */
 
         /* Store a pointer to the codec structure for tty ldisc use */
-        cx20442_codec = codec;
+        cx20442_codec = rtd->codec;
 
         /* Set up digital mute if not provided by the codec */
         if (!codec_dai->driver->ops) {
@@ -520,41 +518,12 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
                 return 0;
         }
 
-        /* Add board specific DAPM widgets and routes */
-        ret = snd_soc_dapm_new_controls(dapm, ams_delta_dapm_widgets,
-                                        ARRAY_SIZE(ams_delta_dapm_widgets));
-        if (ret) {
-                dev_warn(card->dev,
-                                "Failed to register DAPM controls, "
-                                "will continue without any.\n");
-                return 0;
-        }
-
-        ret = snd_soc_dapm_add_routes(dapm, ams_delta_audio_map,
-                                        ARRAY_SIZE(ams_delta_audio_map));
-        if (ret) {
-                dev_warn(card->dev,
-                                "Failed to set up DAPM routes, "
-                                "will continue with codec default map.\n");
-                return 0;
-        }
-
         /* Set up initial pin constellation */
         snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
-        snd_soc_dapm_enable_pin(dapm, "Earpiece");
-        snd_soc_dapm_enable_pin(dapm, "Microphone");
         snd_soc_dapm_disable_pin(dapm, "Speaker");
         snd_soc_dapm_disable_pin(dapm, "AGCIN");
         snd_soc_dapm_disable_pin(dapm, "AGCOUT");
 
-        /* Add virtual switch */
-        ret = snd_soc_add_codec_controls(codec, ams_delta_audio_controls,
-                                        ARRAY_SIZE(ams_delta_audio_controls));
-        if (ret)
-                dev_warn(card->dev,
-                                "Failed to register audio mode control, "
-                                "will continue without it.\n");
-
         return 0;
 }
 
@@ -565,7 +534,7 @@ static struct snd_soc_dai_link ams_delta_dai_link = {
         .cpu_dai_name = "omap-mcbsp.1",
         .codec_dai_name = "cx20442-voice",
         .init = ams_delta_cx20442_init,
-        .platform_name = "omap-pcm-audio",
+        .platform_name = "omap-mcbsp.1",
         .codec_name = "cx20442-codec",
         .ops = &ams_delta_ops,
 };
@@ -576,6 +545,13 @@ static struct snd_soc_card ams_delta_audio_card = {
         .owner = THIS_MODULE,
         .dai_link = &ams_delta_dai_link,
         .num_links = 1,
+
+        .controls = ams_delta_audio_controls,
+        .num_controls = ARRAY_SIZE(ams_delta_audio_controls),
+        .dapm_widgets = ams_delta_dapm_widgets,
+        .num_dapm_widgets = ARRAY_SIZE(ams_delta_dapm_widgets),
+        .dapm_routes = ams_delta_audio_map,
+        .num_dapm_routes = ARRAY_SIZE(ams_delta_audio_map),
 };
 
 /* Module init/exit */
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index fd4d9c809e50..5d7f9cebe041 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -278,7 +278,7 @@ static struct snd_soc_dai_link n810_dai = {
         .name = "TLV320AIC33",
         .stream_name = "AIC33",
         .cpu_dai_name = "omap-mcbsp.2",
-        .platform_name = "omap-pcm-audio",
+        .platform_name = "omap-mcbsp.2",
         .codec_name = "tlv320aic3x-codec.2-0018",
         .codec_dai_name = "tlv320aic3x-hifi",
         .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 024dafc3e298..cec836ed0c01 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -47,8 +47,7 @@ static int omap_abe_hw_params(struct snd_pcm_substream *substream,
 {
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct snd_soc_dai *codec_dai = rtd->codec_dai;
-        struct snd_soc_codec *codec = rtd->codec;
-        struct snd_soc_card *card = codec->card;
+        struct snd_soc_card *card = rtd->card;
         struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
         int clk_id, freq;
         int ret;
@@ -168,7 +167,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
 {
         struct snd_soc_codec *codec = rtd->codec;
-        struct snd_soc_card *card = codec->card;
+        struct snd_soc_card *card = rtd->card;
         struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
         int hs_trim;
         int ret = 0;
@@ -214,9 +213,7 @@ static struct snd_soc_dai_link abe_twl6040_dai_links[] = {
         {
                 .name = "TWL6040",
                 .stream_name = "TWL6040",
-                .cpu_dai_name = "omap-mcpdm",
                 .codec_dai_name = "twl6040-legacy",
-                .platform_name = "omap-pcm-audio",
                 .codec_name = "twl6040-codec",
                 .init = omap_abe_twl6040_init,
                 .ops = &omap_abe_ops,
@@ -224,9 +221,7 @@ static struct snd_soc_dai_link abe_twl6040_dai_links[] = {
         {
                 .name = "DMIC",
                 .stream_name = "DMIC Capture",
-                .cpu_dai_name = "omap-dmic",
                 .codec_dai_name = "dmic-hifi",
-                .platform_name = "omap-pcm-audio",
                 .codec_name = "dmic-codec",
                 .init = omap_abe_dmic_init,
                 .ops = &omap_abe_dmic_ops,
@@ -281,14 +276,14 @@ static int omap_abe_probe(struct platform_device *pdev)
                 dev_err(&pdev->dev, "McPDM node is not provided\n");
                 return -EINVAL;
         }
-        abe_twl6040_dai_links[0].cpu_dai_name  = NULL;
         abe_twl6040_dai_links[0].cpu_of_node = dai_node;
+        abe_twl6040_dai_links[0].platform_of_node = dai_node;
 
         dai_node = of_parse_phandle(node, "ti,dmic", 0);
         if (dai_node) {
                 num_links = 2;
-                abe_twl6040_dai_links[1].cpu_dai_name  = NULL;
                 abe_twl6040_dai_links[1].cpu_of_node = dai_node;
+                abe_twl6040_dai_links[1].platform_of_node = dai_node;
 
                 priv->dmic_codec_dev = platform_device_register_simple(
                                                 "dmic-codec", -1, NULL, 0);
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 1bd531d718f9..53da041896c4 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -42,6 +42,7 @@
 #include <sound/dmaengine_pcm.h>
 
 #include "omap-dmic.h"
+#include "omap-pcm.h"
 
 struct omap_dmic {
         struct device *dev;
@@ -113,7 +114,6 @@ static int omap_dmic_dai_startup(struct snd_pcm_substream *substream,
 
         mutex_unlock(&dmic->mutex);
 
-        snd_soc_dai_set_dma_data(dai, substream, &dmic->dma_data);
         return ret;
 }
 
@@ -417,6 +417,9 @@ static int omap_dmic_probe(struct snd_soc_dai *dai)
 
         /* Configure DMIC threshold value */
         dmic->threshold = OMAP_DMIC_THRES_MAX - 3;
+
+        snd_soc_dai_init_dma_data(dai, NULL, &dmic->dma_data);
+
         return 0;
 }
 
@@ -492,6 +495,10 @@ static int asoc_dmic_probe(struct platform_device *pdev)
         if (ret)
                 goto err_put_clk;
 
+        ret = omap_pcm_platform_register(&pdev->dev);
+        if (ret)
+                goto err_put_clk;
+
         return 0;
 
 err_put_clk:
diff --git a/sound/soc/omap/omap-hdmi-card.c b/sound/soc/omap/omap-hdmi-card.c
index 7e66e9cba5a8..f649fe84b629 100644
--- a/sound/soc/omap/omap-hdmi-card.c
+++ b/sound/soc/omap/omap-hdmi-card.c
@@ -33,7 +33,7 @@ static struct snd_soc_dai_link omap_hdmi_dai = {
         .name = "HDMI",
         .stream_name = "HDMI",
         .cpu_dai_name = "omap-hdmi-audio-dai",
-        .platform_name = "omap-pcm-audio",
+        .platform_name = "omap-hdmi-audio-dai",
         .codec_name = "hdmi-audio-codec",
         .codec_dai_name = "hdmi-hifi",
 };
diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c
index ced3b88b44d4..537a1ec8ad61 100644
--- a/sound/soc/omap/omap-hdmi.c
+++ b/sound/soc/omap/omap-hdmi.c
@@ -36,6 +36,7 @@
 #include <video/omapdss.h>
 
 #include "omap-hdmi.h"
+#include "omap-pcm.h"
 
 #define DRV_NAME "omap-hdmi-audio-dai"
 
@@ -324,7 +325,10 @@ static int omap_hdmi_probe(struct platform_device *pdev)
         ret = snd_soc_register_component(&pdev->dev, &omap_hdmi_component,
                                          &omap_hdmi_dai, 1);
 
-        return ret;
+        if (ret)
+                return ret;
+
+        return omap_pcm_platform_register(&pdev->dev);
 }
 
 static int omap_hdmi_remove(struct platform_device *pdev)
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 6c19bba23570..71d226626f7c 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -38,6 +38,7 @@
 #include <linux/platform_data/asoc-ti-mcbsp.h>
 #include "mcbsp.h"
 #include "omap-mcbsp.h"
+#include "omap-pcm.h"
 
 #define OMAP_MCBSP_RATES        (SNDRV_PCM_RATE_8000_96000)
 
@@ -149,9 +150,6 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
                                            SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2);
         }
 
-        snd_soc_dai_set_dma_data(cpu_dai, substream,
-                                 &mcbsp->dma_data[substream->stream]);
-
         return err;
 }
 
@@ -559,6 +557,10 @@ static int omap_mcbsp_probe(struct snd_soc_dai *dai)
 
         pm_runtime_enable(mcbsp->dev);
 
+        snd_soc_dai_init_dma_data(dai,
+                                  &mcbsp->dma_data[SNDRV_PCM_STREAM_PLAYBACK],
+                                  &mcbsp->dma_data[SNDRV_PCM_STREAM_CAPTURE]);
+
         return 0;
 }
 
@@ -691,7 +693,7 @@ OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 1 Volume", \
 OMAP_MCBSP_ST_CONTROLS(2);
 OMAP_MCBSP_ST_CONTROLS(3);
 
-int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd)
+int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id)
 {
         struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
         struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
@@ -701,7 +703,7 @@ int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd)
                 return 0;
         }
 
-        switch (mcbsp->id) {
+        switch (port_id) {
         case 2: /* McBSP 2 */
                 return snd_soc_add_dai_controls(cpu_dai,
                                         omap_mcbsp2_st_controls,
@@ -711,6 +713,7 @@ int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd)
                                         omap_mcbsp3_st_controls,
                                         ARRAY_SIZE(omap_mcbsp3_st_controls));
         default:
+                dev_err(mcbsp->dev, "Port %d not supported\n", port_id);
                 break;
         }
 
@@ -799,11 +802,15 @@ static int asoc_mcbsp_probe(struct platform_device *pdev)
         platform_set_drvdata(pdev, mcbsp);
 
         ret = omap_mcbsp_init(pdev);
-        if (!ret)
-                return snd_soc_register_component(&pdev->dev, &omap_mcbsp_component,
-                                                  &omap_mcbsp_dai, 1);
+        if (ret)
+                return ret;
+
+        ret = snd_soc_register_component(&pdev->dev, &omap_mcbsp_component,
+                                         &omap_mcbsp_dai, 1);
+        if (ret)
+                return ret;
 
-        return ret;
+        return omap_pcm_platform_register(&pdev->dev);
 }
 
 static int asoc_mcbsp_remove(struct platform_device *pdev)
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index ba8386a0d8dc..2e3369c27be3 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -39,6 +39,6 @@ enum omap_mcbsp_div {
         OMAP_MCBSP_CLKGDV,              /* Sample rate generator divider */
 };
 
-int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd);
+int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id);
 
 #endif
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 2f5b1536477e..d8ebb52645a9 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -42,6 +42,7 @@
 #include <sound/dmaengine_pcm.h>
 
 #include "omap-mcpdm.h"
+#include "omap-pcm.h"
 
 struct mcpdm_link_config {
         u32 link_mask; /* channel mask for the direction */
@@ -265,9 +266,6 @@ static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream,
         }
         mutex_unlock(&mcpdm->mutex);
 
-        snd_soc_dai_set_dma_data(dai, substream,
-                                 &mcpdm->dma_data[substream->stream]);
-
         return 0;
 }
 
@@ -406,6 +404,11 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai)
         mcpdm->config[SNDRV_PCM_STREAM_PLAYBACK].threshold = 2;
         mcpdm->config[SNDRV_PCM_STREAM_CAPTURE].threshold =
                                                         MCPDM_UP_THRES_MAX - 3;
+
+        snd_soc_dai_init_dma_data(dai,
+                                  &mcpdm->dma_data[SNDRV_PCM_STREAM_PLAYBACK],
+                                  &mcpdm->dma_data[SNDRV_PCM_STREAM_CAPTURE]);
+
         return ret;
 }
 
@@ -460,6 +463,7 @@ static int asoc_mcpdm_probe(struct platform_device *pdev)
 {
         struct omap_mcpdm *mcpdm;
         struct resource *res;
+        int ret;
 
         mcpdm = devm_kzalloc(&pdev->dev, sizeof(struct omap_mcpdm), GFP_KERNEL);
         if (!mcpdm)
@@ -490,9 +494,13 @@ static int asoc_mcpdm_probe(struct platform_device *pdev)
 
         mcpdm->dev = &pdev->dev;
 
-        return devm_snd_soc_register_component(&pdev->dev,
+        ret =  devm_snd_soc_register_component(&pdev->dev,
                                                &omap_mcpdm_component,
                                                &omap_mcpdm_dai, 1);
+        if (ret)
+                return ret;
+
+        return omap_pcm_platform_register(&pdev->dev);
 }
 
 static const struct of_device_id omap_mcpdm_of_match[] = {
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 07b8b7bc9d20..8d809f8509c8 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -232,31 +232,12 @@ static struct snd_soc_platform_driver omap_soc_platform = {
         .pcm_free       = omap_pcm_free_dma_buffers,
 };
 
-static int omap_pcm_probe(struct platform_device *pdev)
+int omap_pcm_platform_register(struct device *dev)
 {
-        return snd_soc_register_platform(&pdev->dev,
-                        &omap_soc_platform);
+        return devm_snd_soc_register_platform(dev, &omap_soc_platform);
 }
-
-static int omap_pcm_remove(struct platform_device *pdev)
-{
-        snd_soc_unregister_platform(&pdev->dev);
-        return 0;
-}
-
-static struct platform_driver omap_pcm_driver = {
-        .driver = {
-                        .name = "omap-pcm-audio",
-                        .owner = THIS_MODULE,
-        },
-
-        .probe = omap_pcm_probe,
-        .remove = omap_pcm_remove,
-};
-
-module_platform_driver(omap_pcm_driver);
+EXPORT_SYMBOL_GPL(omap_pcm_platform_register);
 
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
 MODULE_DESCRIPTION("OMAP PCM DMA module");
 MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:omap-pcm-audio");
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
new file mode 100644
index 000000000000..c1d2f31d71e9
--- /dev/null
+++ b/sound/soc/omap/omap-pcm.h
@@ -0,0 +1,30 @@
+/*
+ * omap-pcm.h - OMAP PCM driver
+ *
+ * Copyright (C) 2014 Texas Instruments, Inc.
+ *
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.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.
+ */
+
+#ifndef __OMAP_PCM_H__
+#define __OMAP_PCM_H__
+
+#if IS_ENABLED(CONFIG_SND_OMAP_SOC)
+int omap_pcm_platform_register(struct device *dev);
+#else
+static inline int omap_pcm_platform_register(struct device *dev)
+{
+        return 0;
+}
+#endif /* CONFIG_SND_OMAP_SOC */
+
+#endif /* __OMAP_PCM_H__ */
diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c
index 6a8d6b5f160d..64141db311b2 100644
--- a/sound/soc/omap/omap-twl4030.c
+++ b/sound/soc/omap/omap-twl4030.c
@@ -55,8 +55,7 @@ static int omap_twl4030_hw_params(struct snd_pcm_substream *substream,
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct snd_soc_dai *codec_dai = rtd->codec_dai;
         struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-        struct snd_soc_codec *codec = rtd->codec;
-        struct snd_soc_card *card = codec->card;
+        struct snd_soc_card *card = rtd->card;
         unsigned int fmt;
         int ret;
 
@@ -179,7 +178,7 @@ static inline void twl4030_disconnect_pin(struct snd_soc_dapm_context *dapm,
 static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd)
 {
         struct snd_soc_codec *codec = rtd->codec;
-        struct snd_soc_card *card = codec->card;
+        struct snd_soc_card *card = rtd->card;
         struct snd_soc_dapm_context *dapm = &codec->dapm;
         struct omap_tw4030_pdata *pdata = dev_get_platdata(card->dev);
         struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card);
@@ -239,7 +238,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
                 .stream_name = "TWL4030 HiFi",
                 .cpu_dai_name = "omap-mcbsp.2",
                 .codec_dai_name = "twl4030-hifi",
-                .platform_name = "omap-pcm-audio",
+                .platform_name = "omap-mcbsp.2",
                 .codec_name = "twl4030-codec",
                 .init = omap_twl4030_init,
                 .ops = &omap_twl4030_ops,
@@ -249,7 +248,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
                 .stream_name = "TWL4030 Voice",
                 .cpu_dai_name = "omap-mcbsp.3",
                 .codec_dai_name = "twl4030-voice",
-                .platform_name = "omap-pcm-audio",
+                .platform_name = "omap-mcbsp.2",
                 .codec_name = "twl4030-codec",
                 .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
                            SND_SOC_DAIFMT_CBM_CFM,
@@ -299,12 +298,18 @@ static int omap_twl4030_probe(struct platform_device *pdev)
                 omap_twl4030_dai_links[0].cpu_dai_name  = NULL;
                 omap_twl4030_dai_links[0].cpu_of_node = dai_node;
 
+                omap_twl4030_dai_links[0].platform_name  = NULL;
+                omap_twl4030_dai_links[0].platform_of_node = dai_node;
+
                 dai_node = of_parse_phandle(node, "ti,mcbsp-voice", 0);
                 if (!dai_node) {
                         card->num_links = 1;
                 } else {
                         omap_twl4030_dai_links[1].cpu_dai_name  = NULL;
                         omap_twl4030_dai_links[1].cpu_of_node = dai_node;
+
+                        omap_twl4030_dai_links[1].platform_name  = NULL;
+                        omap_twl4030_dai_links[1].platform_of_node = dai_node;
                 }
 
                 priv->jack_detect = of_get_named_gpio(node,
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index cf604a2faa18..076bec606d78 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -121,7 +121,7 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
  *  |A| <~~clk~~+
  *  |P| <--- TWL4030 <--------- Line In and MICs
  */
-static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget omap3pandora_dapm_widgets[] = {
         SND_SOC_DAPM_DAC_E("PCM DAC", "HiFi Playback", SND_SOC_NOPM,
                            0, 0, omap3pandora_dac_event,
                            SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -130,22 +130,18 @@ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
                            SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
         SND_SOC_DAPM_HP("Headphone Jack", NULL),
         SND_SOC_DAPM_LINE("Line Out", NULL),
-};
 
-static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
         SND_SOC_DAPM_MIC("Mic (internal)", NULL),
         SND_SOC_DAPM_MIC("Mic (external)", NULL),
         SND_SOC_DAPM_LINE("Line In", NULL),
 };
 
-static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
+static const struct snd_soc_dapm_route omap3pandora_map[] = {
         {"PCM DAC", NULL, "APLL Enable"},
         {"Headphone Amplifier", NULL, "PCM DAC"},
         {"Line Out", NULL, "PCM DAC"},
         {"Headphone Jack", NULL, "Headphone Amplifier"},
-};
 
-static const struct snd_soc_dapm_route omap3pandora_in_map[] = {
         {"AUXL", NULL, "Line In"},
         {"AUXR", NULL, "Line In"},
 
@@ -160,7 +156,6 @@ static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd)
 {
         struct snd_soc_codec *codec = rtd->codec;
         struct snd_soc_dapm_context *dapm = &codec->dapm;
-        int ret;
 
         /* All TWL4030 output pins are floating */
         snd_soc_dapm_nc_pin(dapm, "EARPIECE");
@@ -174,20 +169,13 @@ static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd)
         snd_soc_dapm_nc_pin(dapm, "HFR");
         snd_soc_dapm_nc_pin(dapm, "VIBRA");
 
-        ret = snd_soc_dapm_new_controls(dapm, omap3pandora_out_dapm_widgets,
-                                ARRAY_SIZE(omap3pandora_out_dapm_widgets));
-        if (ret < 0)
-                return ret;
-
-        return snd_soc_dapm_add_routes(dapm, omap3pandora_out_map,
-                ARRAY_SIZE(omap3pandora_out_map));
+        return 0;
 }
 
 static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd)
 {
         struct snd_soc_codec *codec = rtd->codec;
         struct snd_soc_dapm_context *dapm = &codec->dapm;
-        int ret;
 
         /* Not comnnected */
         snd_soc_dapm_nc_pin(dapm, "HSMIC");
@@ -195,13 +183,7 @@ static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd)
         snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
         snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
 
-        ret = snd_soc_dapm_new_controls(dapm, omap3pandora_in_dapm_widgets,
-                                ARRAY_SIZE(omap3pandora_in_dapm_widgets));
-        if (ret < 0)
-                return ret;
-
-        return snd_soc_dapm_add_routes(dapm, omap3pandora_in_map,
-                ARRAY_SIZE(omap3pandora_in_map));
+        return 0;
 }
 
 static struct snd_soc_ops omap3pandora_ops = {
@@ -215,7 +197,7 @@ static struct snd_soc_dai_link omap3pandora_dai[] = {
                 .stream_name = "HiFi Out",
                 .cpu_dai_name = "omap-mcbsp.2",
                 .codec_dai_name = "twl4030-hifi",
-                .platform_name = "omap-pcm-audio",
+                .platform_name = "omap-mcbsp.2",
                 .codec_name = "twl4030-codec",
                 .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
                            SND_SOC_DAIFMT_CBS_CFS,
@@ -226,7 +208,7 @@ static struct snd_soc_dai_link omap3pandora_dai[] = {
                 .stream_name = "Line/Mic In",
                 .cpu_dai_name = "omap-mcbsp.4",
                 .codec_dai_name = "twl4030-hifi",
-                .platform_name = "omap-pcm-audio",
+                .platform_name = "omap-mcbsp.4",
                 .codec_name = "twl4030-codec",
                 .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
                            SND_SOC_DAIFMT_CBS_CFS,
@@ -241,6 +223,11 @@ static struct snd_soc_card snd_soc_card_omap3pandora = {
         .owner = THIS_MODULE,
         .dai_link = omap3pandora_dai,
         .num_links = ARRAY_SIZE(omap3pandora_dai),
+
+        .dapm_widgets = omap3pandora_dapm_widgets,
+        .num_dapm_widgets = ARRAY_SIZE(omap3pandora_dapm_widgets),
+        .dapm_routes = omap3pandora_map,
+        .num_dapm_routes = ARRAY_SIZE(omap3pandora_map),
 };
 
 static struct platform_device *omap3pandora_snd_device;
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index d03e57da7708..aa4053bf6710 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -96,7 +96,7 @@ static struct snd_soc_dai_link osk_dai = {
         .stream_name = "AIC23",
         .cpu_dai_name = "omap-mcbsp.1",
         .codec_dai_name = "tlv320aic23-hifi",
-        .platform_name = "omap-pcm-audio",
+        .platform_name = "omap-mcbsp.1",
         .codec_name = "tlv320aic23-codec",
         .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
                    SND_SOC_DAIFMT_CBM_CFM,
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 7fb3d4b10370..6951dc812055 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -26,6 +26,7 @@
 #include <linux/delay.h>
 #include <linux/gpio.h>
 #include <linux/platform_device.h>
+#include <linux/gpio/consumer.h>
 #include <linux/module.h>
 #include <sound/core.h>
 #include <sound/jack.h>
@@ -38,15 +39,6 @@
 
 #include "omap-mcbsp.h"
 
-#define RX51_TVOUT_SEL_GPIO             40
-#define RX51_JACK_DETECT_GPIO           177
-#define RX51_ECI_SW_GPIO                182
-/*
- * 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)
-
 enum {
         RX51_JACK_DISABLED,
         RX51_JACK_TVOUT,                /* tv-out with stereo output */
@@ -54,12 +46,21 @@ enum {
         RX51_JACK_HS,                   /* headset: stereo output with mic */
 };
 
+struct rx51_audio_pdata {
+        struct gpio_desc *tvout_selection_gpio;
+        struct gpio_desc *jack_detection_gpio;
+        struct gpio_desc *eci_sw_gpio;
+        struct gpio_desc *speaker_amp_gpio;
+};
+
 static int rx51_spk_func;
 static int rx51_dmic_func;
 static int rx51_jack_func;
 
 static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
 {
+        struct snd_soc_card *card = dapm->card;
+        struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
         int hp = 0, hs = 0, tvout = 0;
 
         switch (rx51_jack_func) {
@@ -93,7 +94,7 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
         else
                 snd_soc_dapm_disable_pin_unlocked(dapm, "HS Mic");
 
-        gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout);
+        gpiod_set_value(pdata->tvout_selection_gpio, tvout);
 
         snd_soc_dapm_sync_unlocked(dapm);
 
@@ -154,10 +155,12 @@ static int rx51_set_spk(struct snd_kcontrol *kcontrol,
 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_cansleep(RX51_SPEAKER_AMP_TWL_GPIO, 1);
-        else
-                gpio_set_value_cansleep(RX51_SPEAKER_AMP_TWL_GPIO, 0);
+        struct snd_soc_dapm_context *dapm = w->dapm;
+        struct snd_soc_card *card = dapm->card;
+        struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
+
+        gpiod_set_raw_value_cansleep(pdata->speaker_amp_gpio,
+                                     !!SND_SOC_DAPM_EVENT_ON(event));
 
         return 0;
 }
@@ -223,7 +226,6 @@ static struct snd_soc_jack rx51_av_jack;
 
 static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = {
         {
-                .gpio = RX51_JACK_DETECT_GPIO,
                 .name = "avdet-gpio",
                 .report = SND_JACK_HEADSET,
                 .invert = 1,
@@ -237,9 +239,6 @@ static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = {
         SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event),
         SND_SOC_DAPM_MIC("HS Mic", NULL),
         SND_SOC_DAPM_LINE("FM Transmitter", NULL),
-};
-
-static const struct snd_soc_dapm_widget aic34_dapm_widgetsb[] = {
         SND_SOC_DAPM_SPK("Earphone", NULL),
 };
 
@@ -253,9 +252,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 
         {"DMic Rate 64", NULL, "Mic Bias"},
         {"Mic Bias", NULL, "DMic"},
-};
 
-static const struct snd_soc_dapm_route audio_mapb[] = {
         {"b LINE2R", NULL, "MONO_LOUT"},
         {"Earphone", NULL, "b HPLOUT"},
 
@@ -263,9 +260,11 @@ static const struct snd_soc_dapm_route audio_mapb[] = {
         {"b Mic Bias", NULL, "HS Mic"}
 };
 
-static const char *spk_function[] = {"Off", "On"};
-static const char *input_function[] = {"ADC", "Digital Mic"};
-static const char *jack_function[] = {"Off", "TV-OUT", "Headphone", "Headset"};
+static const char * const spk_function[] = {"Off", "On"};
+static const char * const input_function[] = {"ADC", "Digital Mic"};
+static const char * const jack_function[] = {
+        "Off", "TV-OUT", "Headphone", "Headset"
+};
 
 static const struct soc_enum rx51_enum[] = {
         SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
@@ -281,15 +280,15 @@ static const struct snd_kcontrol_new aic34_rx51_controls[] = {
         SOC_ENUM_EXT("Jack Function", rx51_enum[2],
                      rx51_get_jack, rx51_set_jack),
         SOC_DAPM_PIN_SWITCH("FM Transmitter"),
-};
-
-static const struct snd_kcontrol_new aic34_rx51_controlsb[] = {
         SOC_DAPM_PIN_SWITCH("Earphone"),
 };
 
 static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
 {
         struct snd_soc_codec *codec = rtd->codec;
+        struct snd_soc_card *card = rtd->card;
+        struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
+
         struct snd_soc_dapm_context *dapm = &codec->dapm;
         int err;
 
@@ -298,57 +297,41 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
         snd_soc_dapm_nc_pin(dapm, "MIC3R");
         snd_soc_dapm_nc_pin(dapm, "LINE1R");
 
-        /* Add RX-51 specific controls */
-        err = snd_soc_add_card_controls(rtd->card, aic34_rx51_controls,
-                                   ARRAY_SIZE(aic34_rx51_controls));
-        if (err < 0)
-                return err;
-
-        /* Add RX-51 specific widgets */
-        snd_soc_dapm_new_controls(dapm, aic34_dapm_widgets,
-                                  ARRAY_SIZE(aic34_dapm_widgets));
-
-        /* Set up RX-51 specific audio path audio_map */
-        snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
         err = tpa6130a2_add_controls(codec);
-        if (err < 0)
+        if (err < 0) {
+                dev_err(card->dev, "Failed to add TPA6130A2 controls\n");
                 return err;
+        }
         snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42);
 
-        err = omap_mcbsp_st_add_controls(rtd);
-        if (err < 0)
+        err = omap_mcbsp_st_add_controls(rtd, 2);
+        if (err < 0) {
+                dev_err(card->dev, "Failed to add MCBSP controls\n");
                 return err;
+        }
 
         /* AV jack detection */
         err = snd_soc_jack_new(codec, "AV Jack",
                                SND_JACK_HEADSET | SND_JACK_VIDEOOUT,
                                &rx51_av_jack);
-        if (err)
+        if (err) {
+                dev_err(card->dev, "Failed to add AV Jack\n");
                 return err;
+        }
+
+        /* prepare gpio for snd_soc_jack_add_gpios */
+        rx51_av_jack_gpios[0].gpio = desc_to_gpio(pdata->jack_detection_gpio);
+        devm_gpiod_put(card->dev, pdata->jack_detection_gpio);
+
         err = snd_soc_jack_add_gpios(&rx51_av_jack,
                                      ARRAY_SIZE(rx51_av_jack_gpios),
                                      rx51_av_jack_gpios);
-
-        return err;
-}
-
-static int rx51_aic34b_init(struct snd_soc_dapm_context *dapm)
-{
-        int err;
-
-        err = snd_soc_add_card_controls(dapm->card, aic34_rx51_controlsb,
-                                   ARRAY_SIZE(aic34_rx51_controlsb));
-        if (err < 0)
+        if (err) {
+                dev_err(card->dev, "Failed to add GPIOs\n");
                 return err;
+        }
 
-        err = snd_soc_dapm_new_controls(dapm, aic34_dapm_widgetsb,
-                                        ARRAY_SIZE(aic34_dapm_widgetsb));
-        if (err < 0)
-                return 0;
-
-        return snd_soc_dapm_add_routes(dapm, audio_mapb,
-                                       ARRAY_SIZE(audio_mapb));
+        return err;
 }
 
 /* Digital audio interface glue - connects codec <--> CPU */
@@ -358,7 +341,7 @@ static struct snd_soc_dai_link rx51_dai[] = {
                 .stream_name = "AIC34",
                 .cpu_dai_name = "omap-mcbsp.2",
                 .codec_dai_name = "tlv320aic3x-hifi",
-                .platform_name = "omap-pcm-audio",
+                .platform_name = "omap-mcbsp.2",
                 .codec_name = "tlv320aic3x-codec.2-0018",
                 .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
                            SND_SOC_DAIFMT_CBM_CFM,
@@ -371,7 +354,6 @@ static struct snd_soc_aux_dev rx51_aux_dev[] = {
         {
                 .name = "TLV320AIC34b",
                 .codec_name = "tlv320aic3x-codec.2-0019",
-                .init = rx51_aic34b_init,
         },
 };
 
@@ -392,63 +374,160 @@ static struct snd_soc_card rx51_sound_card = {
         .num_aux_devs = ARRAY_SIZE(rx51_aux_dev),
         .codec_conf = rx51_codec_conf,
         .num_configs = ARRAY_SIZE(rx51_codec_conf),
-};
 
-static struct platform_device *rx51_snd_device;
+        .controls = aic34_rx51_controls,
+        .num_controls = ARRAY_SIZE(aic34_rx51_controls),
+        .dapm_widgets = aic34_dapm_widgets,
+        .num_dapm_widgets = ARRAY_SIZE(aic34_dapm_widgets),
+        .dapm_routes = audio_map,
+        .num_dapm_routes = ARRAY_SIZE(audio_map),
+};
 
-static int __init rx51_soc_init(void)
+static int rx51_soc_probe(struct platform_device *pdev)
 {
+        struct rx51_audio_pdata *pdata;
+        struct device_node *np = pdev->dev.of_node;
+        struct snd_soc_card *card = &rx51_sound_card;
         int err;
 
         if (!machine_is_nokia_rx51() && !of_machine_is_compatible("nokia,omap3-n900"))
                 return -ENODEV;
 
-        err = gpio_request_one(RX51_TVOUT_SEL_GPIO,
-                               GPIOF_DIR_OUT | GPIOF_INIT_LOW, "tvout_sel");
-        if (err)
-                goto err_gpio_tvout_sel;
-        err = gpio_request_one(RX51_ECI_SW_GPIO,
-                               GPIOF_DIR_OUT | GPIOF_INIT_HIGH, "eci_sw");
-        if (err)
-                goto err_gpio_eci_sw;
-
-        rx51_snd_device = platform_device_alloc("soc-audio", -1);
-        if (!rx51_snd_device) {
-                err = -ENOMEM;
-                goto err1;
+        card->dev = &pdev->dev;
+
+        if (np) {
+                struct device_node *dai_node;
+
+                dai_node = of_parse_phandle(np, "nokia,cpu-dai", 0);
+                if (!dai_node) {
+                        dev_err(&pdev->dev, "McBSP node is not provided\n");
+                        return -EINVAL;
+                }
+                rx51_dai[0].cpu_dai_name = NULL;
+                rx51_dai[0].platform_name = NULL;
+                rx51_dai[0].cpu_of_node = dai_node;
+                rx51_dai[0].platform_of_node = dai_node;
+
+                dai_node = of_parse_phandle(np, "nokia,audio-codec", 0);
+                if (!dai_node) {
+                        dev_err(&pdev->dev, "Codec node is not provided\n");
+                        return -EINVAL;
+                }
+                rx51_dai[0].codec_name = NULL;
+                rx51_dai[0].codec_of_node = dai_node;
+
+                dai_node = of_parse_phandle(np, "nokia,audio-codec", 1);
+                if (!dai_node) {
+                        dev_err(&pdev->dev, "Auxiliary Codec node is not provided\n");
+                        return -EINVAL;
+                }
+                rx51_aux_dev[0].codec_name = NULL;
+                rx51_aux_dev[0].codec_of_node = dai_node;
+                rx51_codec_conf[0].dev_name = NULL;
+                rx51_codec_conf[0].of_node = dai_node;
+
+                dai_node = of_parse_phandle(np, "nokia,headphone-amplifier", 0);
+                if (!dai_node) {
+                        dev_err(&pdev->dev, "Headphone amplifier node is not provided\n");
+                        return -EINVAL;
+                }
+
+                /* TODO: tpa6130a2a driver supports only a single instance, so
+                 * this driver ignores the headphone-amplifier node for now.
+                 * It's already mandatory in the DT binding to be future proof.
+                 */
         }
 
-        platform_set_drvdata(rx51_snd_device, &rx51_sound_card);
+        pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);
+        if (pdata == NULL) {
+                dev_err(card->dev, "failed to create private data\n");
+                return -ENOMEM;
+        }
+        snd_soc_card_set_drvdata(card, pdata);
 
-        err = platform_device_add(rx51_snd_device);
-        if (err)
-                goto err2;
+        pdata->tvout_selection_gpio = devm_gpiod_get(card->dev,
+                                                     "tvout-selection");
+        if (IS_ERR(pdata->tvout_selection_gpio)) {
+                dev_err(card->dev, "could not get tvout selection gpio\n");
+                return PTR_ERR(pdata->tvout_selection_gpio);
+        }
 
-        return 0;
-err2:
-        platform_device_put(rx51_snd_device);
-err1:
-        gpio_free(RX51_ECI_SW_GPIO);
-err_gpio_eci_sw:
-        gpio_free(RX51_TVOUT_SEL_GPIO);
-err_gpio_tvout_sel:
+        err = gpiod_direction_output(pdata->tvout_selection_gpio, 0);
+        if (err) {
+                dev_err(card->dev, "could not setup tvout selection gpio\n");
+                return err;
+        }
 
-        return err;
+        pdata->jack_detection_gpio = devm_gpiod_get(card->dev,
+                                                    "jack-detection");
+        if (IS_ERR(pdata->jack_detection_gpio)) {
+                dev_err(card->dev, "could not get jack detection gpio\n");
+                return PTR_ERR(pdata->jack_detection_gpio);
+        }
+
+        pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch");
+        if (IS_ERR(pdata->eci_sw_gpio)) {
+                dev_err(card->dev, "could not get eci switch gpio\n");
+                return PTR_ERR(pdata->eci_sw_gpio);
+        }
+
+        err = gpiod_direction_output(pdata->eci_sw_gpio, 1);
+        if (err) {
+                dev_err(card->dev, "could not setup eci switch gpio\n");
+                return err;
+        }
+
+        pdata->speaker_amp_gpio = devm_gpiod_get(card->dev,
+                                                 "speaker-amplifier");
+        if (IS_ERR(pdata->speaker_amp_gpio)) {
+                dev_err(card->dev, "could not get speaker enable gpio\n");
+                return PTR_ERR(pdata->speaker_amp_gpio);
+        }
+
+        err = gpiod_direction_output(pdata->speaker_amp_gpio, 0);
+        if (err) {
+                dev_err(card->dev, "could not setup speaker enable gpio\n");
+                return err;
+        }
+
+        err = devm_snd_soc_register_card(card->dev, card);
+        if (err) {
+                dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", err);
+                return err;
+        }
+
+        return 0;
 }
 
-static void __exit rx51_soc_exit(void)
+static int rx51_soc_remove(struct platform_device *pdev)
 {
         snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios),
                                 rx51_av_jack_gpios);
 
-        platform_device_unregister(rx51_snd_device);
-        gpio_free(RX51_ECI_SW_GPIO);
-        gpio_free(RX51_TVOUT_SEL_GPIO);
+        return 0;
 }
 
-module_init(rx51_soc_init);
-module_exit(rx51_soc_exit);
+#if defined(CONFIG_OF)
+static const struct of_device_id rx51_audio_of_match[] = {
+        { .compatible = "nokia,n900-audio", },
+        {},
+};
+MODULE_DEVICE_TABLE(of, rx51_audio_of_match);
+#endif
+
+static struct platform_driver rx51_soc_driver = {
+        .driver = {
+                .name = "rx51-audio",
+                .owner = THIS_MODULE,
+                .of_match_table = of_match_ptr(rx51_audio_of_match),
+        },
+        .probe = rx51_soc_probe,
+        .remove = rx51_soc_remove,
+};
+
+module_platform_driver(rx51_soc_driver);
 
 MODULE_AUTHOR("Nokia Corporation");
 MODULE_DESCRIPTION("ALSA SoC Nokia RX-51");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:rx51-audio");