Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into next/boards

The asoc branch that was already merged into v3.4 contains some
board-level changes that conflict with patches we already have
here, so pull in that branch to resolve the conflicts.

Conflicts:
	arch/arm/mach-imx/mach-imx27_visstrim_m10.c
	arch/arm/mach-omap2/board-omap4panda.c

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
[olof: Amended fix for mismerge as reported by Kevin Hilman]
Signed-off-by: Olof Johansson <olof@lixom.net>

11 files changed, 390 insertions, 318 deletions

diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index fb1bf2581efb..47b23fea20c2 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -97,16 +97,19 @@ config SND_OMAP_SOC_SDP3430
 	  Say Y if you want to add support for SoC audio on Texas Instruments
 	  SDP3430.
 
-config SND_OMAP_SOC_SDP4430
-	tristate "SoC Audio support for Texas Instruments SDP4430"
-	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP
+config SND_OMAP_SOC_OMAP_ABE_TWL6040
+	tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec"
+	depends on TWL4030_CORE && SND_OMAP_SOC && ARCH_OMAP4
 	select SND_OMAP_SOC_DMIC
 	select SND_OMAP_SOC_MCPDM
 	select SND_SOC_TWL6040
 	select SND_SOC_DMIC
 	help
-	  Say Y if you want to add support for SoC audio on Texas Instruments
-	  SDP4430.
+	  Say Y if you want to add support for SoC audio on OMAP boards using
+	  ABE and twl6040 codec. This driver currently supports:
+	  - SDP4430/Blaze boards
+	  - PandaBoard (4430)
+	  - PandaBoardES (4460)
 
 config SND_OMAP_SOC_OMAP4_HDMI
 	tristate "SoC Audio support for Texas Instruments OMAP4 HDMI"
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 1fd723fb559d..123ac18303e5 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -20,7 +20,7 @@ snd-soc-overo-objs := overo.o
 snd-soc-omap3evm-objs := omap3evm.o
 snd-soc-am3517evm-objs := am3517evm.o
 snd-soc-sdp3430-objs := sdp3430.o
-snd-soc-sdp4430-objs := sdp4430.o
+snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o
 snd-soc-omap3pandora-objs := omap3pandora.o
 snd-soc-omap3beagle-objs := omap3beagle.o
 snd-soc-zoom2-objs := zoom2.o
@@ -36,7 +36,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o
 obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
-obj-$(CONFIG_SND_OMAP_SOC_SDP4430) += snd-soc-sdp4430.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
 obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index f610260065ba..41586b26ce90 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -544,7 +544,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
 	snd_soc_dapm_disable_pin(dapm, "AGCOUT");
 
 	/* Add virtual switch */
-	ret = snd_soc_add_controls(codec, ams_delta_audio_controls,
+	ret = snd_soc_add_codec_controls(codec, ams_delta_audio_controls,
 					ARRAY_SIZE(ams_delta_audio_controls));
 	if (ret)
 		dev_warn(card->dev,
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 597be412f1e4..c292bf0fd19c 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -55,9 +55,8 @@ static int n810_spk_func;
 static int n810_jack_func;
 static int n810_dmic_func;
 
-static void n810_ext_control(struct snd_soc_codec *codec)
+static void n810_ext_control(struct snd_soc_dapm_context *dapm)
 {
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int hp = 0, line1l = 0;
 
 	switch (n810_jack_func) {
@@ -102,7 +101,7 @@ static int n810_startup(struct snd_pcm_substream *substream)
 	snd_pcm_hw_constraint_minmax(runtime,
 				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
 
-	n810_ext_control(codec);
+	n810_ext_control(&codec->dapm);
 	return clk_enable(sys_clkout2);
 }
 
@@ -142,13 +141,13 @@ static int n810_get_spk(struct snd_kcontrol *kcontrol,
 static int n810_set_spk(struct snd_kcontrol *kcontrol,
 			struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card =  snd_kcontrol_chip(kcontrol);
 
 	if (n810_spk_func == ucontrol->value.integer.value[0])
 		return 0;
 
 	n810_spk_func = ucontrol->value.integer.value[0];
-	n810_ext_control(codec);
+	n810_ext_control(&card->dapm);
 
 	return 1;
 }
@@ -164,13 +163,13 @@ static int n810_get_jack(struct snd_kcontrol *kcontrol,
 static int n810_set_jack(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card =  snd_kcontrol_chip(kcontrol);
 
 	if (n810_jack_func == ucontrol->value.integer.value[0])
 		return 0;
 
 	n810_jack_func = ucontrol->value.integer.value[0];
-	n810_ext_control(codec);
+	n810_ext_control(&card->dapm);
 
 	return 1;
 }
@@ -186,13 +185,13 @@ static int n810_get_input(struct snd_kcontrol *kcontrol,
 static int n810_set_input(struct snd_kcontrol *kcontrol,
 			  struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card =  snd_kcontrol_chip(kcontrol);
 
 	if (n810_dmic_func == ucontrol->value.integer.value[0])
 		return 0;
 
 	n810_dmic_func = ucontrol->value.integer.value[0];
-	n810_ext_control(codec);
+	n810_ext_control(&card->dapm);
 
 	return 1;
 }
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
new file mode 100644
index 000000000000..93bb8eee22b3
--- /dev/null
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -0,0 +1,349 @@
+/*
+ * omap-abe-twl6040.c  --  SoC audio for TI OMAP based boards with ABE and
+ *			   twl6040 codec
+ *
+ * Author: Misael Lopez Cruz <misael.lopez@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.
+ *
+ * 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/clk.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/twl6040.h>
+#include <linux/platform_data/omap-abe-twl6040.h>
+#include <linux/module.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+#include <plat/hardware.h>
+#include <plat/mux.h>
+
+#include "omap-dmic.h"
+#include "omap-mcpdm.h"
+#include "omap-pcm.h"
+#include "../codecs/twl6040.h"
+
+static int omap_abe_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->codec_dai;
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_card *card = codec->card;
+	struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
+	int clk_id, freq;
+	int ret;
+
+	clk_id = twl6040_get_clk_id(rtd->codec);
+	if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
+		freq = pdata->mclk_freq;
+	else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
+		freq = 32768;
+	else
+		return -EINVAL;
+
+	/* set the codec mclk */
+	ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
+				SND_SOC_CLOCK_IN);
+	if (ret) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+	return ret;
+}
+
+static struct snd_soc_ops omap_abe_ops = {
+	.hw_params = omap_abe_hw_params,
+};
+
+static int omap_abe_dmic_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 *cpu_dai = rtd->cpu_dai;
+	int ret = 0;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
+				     19200000, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set DMIC cpu system clock\n");
+		return ret;
+	}
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
+				     SND_SOC_CLOCK_OUT);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set DMIC output clock\n");
+		return ret;
+	}
+	return 0;
+}
+
+static struct snd_soc_ops omap_abe_dmic_ops = {
+	.hw_params = omap_abe_dmic_hw_params,
+};
+
+/* Headset jack */
+static struct snd_soc_jack hs_jack;
+
+/*Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{
+		.pin = "Headset Mic",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headset Stereophone",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+/* SDP4430 machine DAPM */
+static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
+	/* Outputs */
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_SPK("Earphone Spk", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+	SND_SOC_DAPM_SPK("Vibrator", NULL),
+
+	/* Inputs */
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Main Handset Mic", NULL),
+	SND_SOC_DAPM_MIC("Sub Handset Mic", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Routings for outputs */
+	{"Headset Stereophone", NULL, "HSOL"},
+	{"Headset Stereophone", NULL, "HSOR"},
+
+	{"Earphone Spk", NULL, "EP"},
+
+	{"Ext Spk", NULL, "HFL"},
+	{"Ext Spk", NULL, "HFR"},
+
+	{"Line Out", NULL, "AUXL"},
+	{"Line Out", NULL, "AUXR"},
+
+	{"Vibrator", NULL, "VIBRAL"},
+	{"Vibrator", NULL, "VIBRAR"},
+
+	/* Routings for inputs */
+	{"HSMIC", NULL, "Headset Mic"},
+	{"Headset Mic", NULL, "Headset Mic Bias"},
+
+	{"MAINMIC", NULL, "Main Handset Mic"},
+	{"Main Handset Mic", NULL, "Main Mic Bias"},
+
+	{"SUBMIC", NULL, "Sub Handset Mic"},
+	{"Sub Handset Mic", NULL, "Main Mic Bias"},
+
+	{"AFML", NULL, "Line In"},
+	{"AFMR", NULL, "Line In"},
+};
+
+static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm,
+					  int connected, char *pin)
+{
+	if (!connected)
+		snd_soc_dapm_disable_pin(dapm, pin);
+}
+
+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_dapm_context *dapm = &codec->dapm;
+	struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
+	int hs_trim;
+	int ret = 0;
+
+	/* Disable not connected paths if not used */
+	twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone");
+	twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
+	twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
+	twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
+	twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator");
+	twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
+	twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
+	twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
+	twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In");
+
+	/*
+	 * Configure McPDM offset cancellation based on the HSOTRIM value from
+	 * twl6040.
+	 */
+	hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM);
+	omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
+					TWL6040_HSF_TRIM_RIGHT(hs_trim));
+
+	/* Headset jack detection only if it is supported */
+	if (pdata->jack_detection) {
+		ret = snd_soc_jack_new(codec, "Headset Jack",
+					SND_JACK_HEADSET, &hs_jack);
+		if (ret)
+			return ret;
+
+		ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+					hs_jack_pins);
+		twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Digital Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route dmic_audio_map[] = {
+	{"DMic", NULL, "Digital Mic"},
+	{"Digital Mic", NULL, "Digital Mic1 Bias"},
+};
+
+static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
+				ARRAY_SIZE(dmic_dapm_widgets));
+	if (ret)
+		return ret;
+
+	return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
+				ARRAY_SIZE(dmic_audio_map));
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link twl6040_dmic_dai[] = {
+	{
+		.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,
+	},
+	{
+		.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,
+	},
+};
+
+static struct snd_soc_dai_link twl6040_only_dai[] = {
+	{
+		.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,
+	},
+};
+
+/* Audio machine driver */
+static struct snd_soc_card omap_abe_card = {
+	.owner = THIS_MODULE,
+
+	.dapm_widgets = twl6040_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static __devinit int omap_abe_probe(struct platform_device *pdev)
+{
+	struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev);
+	struct snd_soc_card *card = &omap_abe_card;
+	int ret;
+
+	card->dev = &pdev->dev;
+
+	if (!pdata) {
+		dev_err(&pdev->dev, "Missing pdata\n");
+		return -ENODEV;
+	}
+
+	if (pdata->card_name) {
+		card->name = pdata->card_name;
+	} else {
+		dev_err(&pdev->dev, "Card name is not provided\n");
+		return -ENODEV;
+	}
+
+	if (!pdata->mclk_freq) {
+		dev_err(&pdev->dev, "MCLK frequency missing\n");
+		return -ENODEV;
+	}
+
+	if (pdata->has_dmic) {
+		card->dai_link = twl6040_dmic_dai;
+		card->num_links = ARRAY_SIZE(twl6040_dmic_dai);
+	} else {
+		card->dai_link = twl6040_only_dai;
+		card->num_links = ARRAY_SIZE(twl6040_only_dai);
+	}
+
+	ret = snd_soc_register_card(card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+
+	return ret;
+}
+
+static int __devexit omap_abe_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static struct platform_driver omap_abe_driver = {
+	.driver = {
+		.name = "omap-abe-twl6040",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = omap_abe_probe,
+	.remove = __devexit_p(omap_abe_remove),
+};
+
+module_platform_driver(omap_abe_driver);
+
+MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:omap-abe-twl6040");
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 0855c1cfa7fd..4dcb5a7e40e8 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -113,12 +113,10 @@ static int omap_dmic_dai_startup(struct snd_pcm_substream *substream,
 
 	mutex_lock(&dmic->mutex);
 
-	if (!dai->active) {
-		snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24);
+	if (!dai->active)
 		dmic->active = 1;
-	} else {
+	else
 		ret = -EBUSY;
-	}
 
 	mutex_unlock(&dmic->mutex);
 
@@ -445,6 +443,7 @@ static struct snd_soc_dai_driver omap_dmic_dai = {
 		.channels_max = 6,
 		.rates = SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000,
 		.formats = SNDRV_PCM_FMTBIT_S32_LE,
+		.sig_bits = 24,
 	},
 	.ops = &omap_dmic_dai_ops,
 };
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 017371913ec3..1287b870f221 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -744,17 +744,17 @@ static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = {
 				      omap_mcbsp3_set_st_ch1_volume),
 };
 
-int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id)
+int omap_mcbsp_st_add_controls(struct snd_soc_dai *dai)
 {
 	if (!cpu_is_omap34xx())
 		return -ENODEV;
 
-	switch (mcbsp_id) {
+	switch (dai->id) {
 	case 1: /* McBSP 2 */
-		return snd_soc_add_controls(codec, omap_mcbsp2_st_controls,
+		return snd_soc_add_dai_controls(dai, omap_mcbsp2_st_controls,
 					ARRAY_SIZE(omap_mcbsp2_st_controls));
 	case 2: /* McBSP 3 */
-		return snd_soc_add_controls(codec, omap_mcbsp3_st_controls,
+		return snd_soc_add_dai_controls(dai, omap_mcbsp3_st_controls,
 					ARRAY_SIZE(omap_mcbsp3_st_controls));
 	default:
 		break;
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index 65cde9d3807b..476fe2add703 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -59,6 +59,6 @@ enum omap_mcbsp_div {
 #define NUM_LINKS	5
 #endif
 
-int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id);
+int omap_mcbsp_st_add_controls(struct snd_soc_dai *dai);
 
 #endif
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 0e25df4fa9e5..39705561131a 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -419,12 +419,14 @@ static struct snd_soc_dai_driver omap_mcpdm_dai = {
 		.channels_max = 5,
 		.rates = OMAP_MCPDM_RATES,
 		.formats = OMAP_MCPDM_FORMATS,
+		.sig_bits = 24,
 	},
 	.capture = {
 		.channels_min = 1,
 		.channels_max = 3,
 		.rates = OMAP_MCPDM_RATES,
 		.formats = OMAP_MCPDM_FORMATS,
+		.sig_bits = 24,
 	},
 	.ops = &omap_mcpdm_dai_ops,
 };
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index fada6ef43eea..58936c730a87 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -59,9 +59,8 @@ static int rx51_spk_func;
 static int rx51_dmic_func;
 static int rx51_jack_func;
 
-static void rx51_ext_control(struct snd_soc_codec *codec)
+static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
 {
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int hp = 0, hs = 0, tvout = 0;
 
 	switch (rx51_jack_func) {
@@ -102,11 +101,11 @@ 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->codec;
+	struct snd_soc_card *card = rtd->card;
 
 	snd_pcm_hw_constraint_minmax(runtime,
 				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
-	rx51_ext_control(codec);
+	rx51_ext_control(&card->dapm);
 
 	return 0;
 }
@@ -138,13 +137,13 @@ static int rx51_get_spk(struct snd_kcontrol *kcontrol,
 static int rx51_set_spk(struct snd_kcontrol *kcontrol,
 			struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card = 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);
+	rx51_ext_control(&card->dapm);
 
 	return 1;
 }
@@ -184,13 +183,13 @@ static int rx51_get_input(struct snd_kcontrol *kcontrol,
 static int rx51_set_input(struct snd_kcontrol *kcontrol,
 			  struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card = 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);
+	rx51_ext_control(&card->dapm);
 
 	return 1;
 }
@@ -206,13 +205,13 @@ static int rx51_get_jack(struct snd_kcontrol *kcontrol,
 static int rx51_set_jack(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
 
 	if (rx51_jack_func == ucontrol->value.integer.value[0])
 		return 0;
 
 	rx51_jack_func = ucontrol->value.integer.value[0];
-	rx51_ext_control(codec);
+	rx51_ext_control(&card->dapm);
 
 	return 1;
 }
@@ -297,7 +296,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
 	snd_soc_dapm_nc_pin(dapm, "LINE1R");
 
 	/* Add RX-51 specific controls */
-	err = snd_soc_add_controls(codec, aic34_rx51_controls,
+	err = snd_soc_add_card_controls(rtd->card, aic34_rx51_controls,
 				   ARRAY_SIZE(aic34_rx51_controls));
 	if (err < 0)
 		return err;
@@ -314,7 +313,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
 		return err;
 	snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42);
 
-	err = omap_mcbsp_st_add_controls(codec, 1);
+	err = omap_mcbsp_st_add_controls(rtd->cpu_dai);
 	if (err < 0)
 		return err;
 
@@ -335,7 +334,7 @@ static int rx51_aic34b_init(struct snd_soc_dapm_context *dapm)
 {
 	int err;
 
-	err = snd_soc_add_controls(dapm->codec, aic34_rx51_controlsb,
+	err = snd_soc_add_card_controls(dapm->card, aic34_rx51_controlsb,
 				   ARRAY_SIZE(aic34_rx51_controlsb));
 	if (err < 0)
 		return err;
diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c
deleted file mode 100644
index 175ba9a04edf..000000000000
--- a/sound/soc/omap/sdp4430.c
+++ /dev/null
@@ -1,279 +0,0 @@
-/*
- * sdp4430.c  --  SoC audio for TI OMAP4430 SDP
- *
- * Author: Misael Lopez Cruz <x0052729@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.
- *
- * 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/clk.h>
-#include <linux/platform_device.h>
-#include <linux/mfd/twl6040.h>
-#include <linux/module.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include <asm/mach-types.h>
-#include <plat/hardware.h>
-#include <plat/mux.h>
-
-#include "omap-dmic.h"
-#include "omap-mcpdm.h"
-#include "omap-pcm.h"
-#include "../codecs/twl6040.h"
-
-static int sdp4430_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->codec_dai;
-	int clk_id, freq;
-	int ret;
-
-	clk_id = twl6040_get_clk_id(rtd->codec);
-	if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
-		freq = 38400000;
-	else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
-		freq = 32768;
-	else
-		return -EINVAL;
-
-	/* set the codec mclk */
-	ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
-				SND_SOC_CLOCK_IN);
-	if (ret) {
-		printk(KERN_ERR "can't set codec system clock\n");
-		return ret;
-	}
-	return ret;
-}
-
-static struct snd_soc_ops sdp4430_ops = {
-	.hw_params = sdp4430_hw_params,
-};
-
-static int sdp4430_dmic_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 *cpu_dai = rtd->cpu_dai;
-	int ret = 0;
-
-	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
-				     19200000, SND_SOC_CLOCK_IN);
-	if (ret < 0) {
-		printk(KERN_ERR "can't set DMIC cpu system clock\n");
-		return ret;
-	}
-	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
-				     SND_SOC_CLOCK_OUT);
-	if (ret < 0) {
-		printk(KERN_ERR "can't set DMIC output clock\n");
-		return ret;
-	}
-	return 0;
-}
-
-static struct snd_soc_ops sdp4430_dmic_ops = {
-	.hw_params = sdp4430_dmic_hw_params,
-};
-
-/* Headset jack */
-static struct snd_soc_jack hs_jack;
-
-/*Headset jack detection DAPM pins */
-static struct snd_soc_jack_pin hs_jack_pins[] = {
-	{
-		.pin = "Headset Mic",
-		.mask = SND_JACK_MICROPHONE,
-	},
-	{
-		.pin = "Headset Stereophone",
-		.mask = SND_JACK_HEADPHONE,
-	},
-};
-
-/* SDP4430 machine DAPM */
-static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = {
-	SND_SOC_DAPM_MIC("Ext Mic", NULL),
-	SND_SOC_DAPM_SPK("Ext Spk", NULL),
-	SND_SOC_DAPM_MIC("Headset Mic", NULL),
-	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
-	SND_SOC_DAPM_SPK("Earphone Spk", NULL),
-	SND_SOC_DAPM_INPUT("FM Stereo In"),
-};
-
-static const struct snd_soc_dapm_route audio_map[] = {
-	/* External Mics: MAINMIC, SUBMIC with bias*/
-	{"MAINMIC", NULL, "Main Mic Bias"},
-	{"SUBMIC", NULL, "Main Mic Bias"},
-	{"Main Mic Bias", NULL, "Ext Mic"},
-
-	/* External Speakers: HFL, HFR */
-	{"Ext Spk", NULL, "HFL"},
-	{"Ext Spk", NULL, "HFR"},
-
-	/* Headset Mic: HSMIC with bias */
-	{"HSMIC", NULL, "Headset Mic Bias"},
-	{"Headset Mic Bias", NULL, "Headset Mic"},
-
-	/* Headset Stereophone (Headphone): HSOL, HSOR */
-	{"Headset Stereophone", NULL, "HSOL"},
-	{"Headset Stereophone", NULL, "HSOR"},
-
-	/* Earphone speaker */
-	{"Earphone Spk", NULL, "EP"},
-
-	/* Aux/FM Stereo In: AFML, AFMR */
-	{"AFML", NULL, "FM Stereo In"},
-	{"AFMR", NULL, "FM Stereo In"},
-};
-
-static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	int ret, hs_trim;
-
-	/*
-	 * Configure McPDM offset cancellation based on the HSOTRIM value from
-	 * twl6040.
-	 */
-	hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM);
-	omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
-					TWL6040_HSF_TRIM_RIGHT(hs_trim));
-
-	/* Headset jack detection */
-	ret = snd_soc_jack_new(codec, "Headset Jack",
-				SND_JACK_HEADSET, &hs_jack);
-	if (ret)
-		return ret;
-
-	ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
-				hs_jack_pins);
-
-	if (machine_is_omap_4430sdp())
-		twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
-	else
-		snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET);
-
-	return ret;
-}
-
-static const struct snd_soc_dapm_widget sdp4430_dmic_dapm_widgets[] = {
-	SND_SOC_DAPM_MIC("Digital Mic", NULL),
-};
-
-static const struct snd_soc_dapm_route dmic_audio_map[] = {
-	{"DMic", NULL, "Digital Mic1 Bias"},
-	{"Digital Mic1 Bias", NULL, "Digital Mic"},
-};
-
-static int sdp4430_dmic_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int ret;
-
-	ret = snd_soc_dapm_new_controls(dapm, sdp4430_dmic_dapm_widgets,
-				ARRAY_SIZE(sdp4430_dmic_dapm_widgets));
-	if (ret)
-		return ret;
-
-	return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
-				ARRAY_SIZE(dmic_audio_map));
-}
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link sdp4430_dai[] = {
-	{
-		.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 = sdp4430_twl6040_init,
-		.ops = &sdp4430_ops,
-	},
-	{
-		.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 = sdp4430_dmic_init,
-		.ops = &sdp4430_dmic_ops,
-	},
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_sdp4430 = {
-	.name = "SDP4430",
-	.owner = THIS_MODULE,
-	.dai_link = sdp4430_dai,
-	.num_links = ARRAY_SIZE(sdp4430_dai),
-
-	.dapm_widgets = sdp4430_twl6040_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(sdp4430_twl6040_dapm_widgets),
-	.dapm_routes = audio_map,
-	.num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
-static struct platform_device *sdp4430_snd_device;
-
-static int __init sdp4430_soc_init(void)
-{
-	int ret;
-
-	if (!machine_is_omap_4430sdp())
-		return -ENODEV;
-	printk(KERN_INFO "SDP4430 SoC init\n");
-
-	sdp4430_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!sdp4430_snd_device) {
-		printk(KERN_ERR "Platform device allocation failed\n");
-		return -ENOMEM;
-	}
-
-	platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430);
-
-	ret = platform_device_add(sdp4430_snd_device);
-	if (ret)
-		goto err;
-
-	return 0;
-
-err:
-	printk(KERN_ERR "Unable to add platform device\n");
-	platform_device_put(sdp4430_snd_device);
-	return ret;
-}
-module_init(sdp4430_soc_init);
-
-static void __exit sdp4430_soc_exit(void)
-{
-	platform_device_unregister(sdp4430_snd_device);
-}
-module_exit(sdp4430_soc_exit);
-
-MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
-MODULE_DESCRIPTION("ALSA SoC SDP4430");
-MODULE_LICENSE("GPL");
-