ASoC: Add mc13783 codec

Signed-off-by: Philippe Rétornaz <philippe.retornaz@epfl.ch>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

4 files changed, 834 insertions, 0 deletions

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index f8035bd09af6..22c686444633 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MAX9850 if I2C
 	select SND_SOC_MAX9768 if I2C
 	select SND_SOC_MAX9877 if I2C
+	select SND_SOC_MC13783 if MFD_MC13XXX
 	select SND_SOC_ML26124 if I2C
 	select SND_SOC_PCM3008
 	select SND_SOC_RT5631 if I2C
@@ -444,6 +445,9 @@ config SND_SOC_MAX9768
 config SND_SOC_MAX9877
 	tristate
 
+config SND_SOC_MC13783
+	tristate
+
 config SND_SOC_ML26124
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index d11ab901b075..a9663e9c375b 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -31,6 +31,7 @@ snd-soc-max9768-objs := max9768.o
 snd-soc-max98088-objs := max98088.o
 snd-soc-max98095-objs := max98095.o
 snd-soc-max9850-objs := max9850.o
+snd-soc-mc13783-objs := mc13783.o
 snd-soc-ml26124-objs := ml26124.o
 snd-soc-pcm3008-objs := pcm3008.o
 snd-soc-rt5631-objs := rt5631.o
@@ -140,6 +141,7 @@ obj-$(CONFIG_SND_SOC_MAX9768)	+= snd-soc-max9768.o
 obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
 obj-$(CONFIG_SND_SOC_MAX98095)	+= snd-soc-max98095.o
 obj-$(CONFIG_SND_SOC_MAX9850)	+= snd-soc-max9850.o
+obj-$(CONFIG_SND_SOC_MC13783)	+= snd-soc-mc13783.o
 obj-$(CONFIG_SND_SOC_ML26124)	+= snd-soc-ml26124.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
 obj-$(CONFIG_SND_SOC_RT5631)	+= snd-soc-rt5631.o
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
new file mode 100644
index 000000000000..50fa38b9d183
--- /dev/null
+++ b/sound/soc/codecs/mc13783.c
@@ -0,0 +1,800 @@
+/*
+ * Copyright 2008 Juergen Beisert, kernel@pengutronix.de
+ * Copyright 2009 Sascha Hauer, s.hauer@pengutronix.de
+ * Copyright 2012 Philippe Retornaz, philippe.retornaz@epfl.ch
+ *
+ * Initial development of this code was funded by
+ * Phytec Messtechnik GmbH, http://www.phytec.de
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ * 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 Street, Fifth Floor, Boston,
+ * MA  02110-1301, USA.
+ */
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/mfd/mc13xxx.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/soc-dapm.h>
+
+#include "mc13783.h"
+
+#define MC13783_AUDIO_RX0	36
+#define MC13783_AUDIO_RX1	37
+#define MC13783_AUDIO_TX	38
+#define MC13783_SSI_NETWORK	39
+#define MC13783_AUDIO_CODEC	40
+#define MC13783_AUDIO_DAC	41
+
+#define AUDIO_RX0_ALSPEN		(1 << 5)
+#define AUDIO_RX0_ALSPSEL		(1 << 7)
+#define AUDIO_RX0_ADDCDC		(1 << 21)
+#define AUDIO_RX0_ADDSTDC		(1 << 22)
+#define AUDIO_RX0_ADDRXIN		(1 << 23)
+
+#define AUDIO_RX1_PGARXEN		(1 << 0);
+#define AUDIO_RX1_PGASTEN		(1 << 5)
+#define AUDIO_RX1_ARXINEN		(1 << 10)
+
+#define AUDIO_TX_AMC1REN		(1 << 5)
+#define AUDIO_TX_AMC1LEN		(1 << 7)
+#define AUDIO_TX_AMC2EN			(1 << 9)
+#define AUDIO_TX_ATXINEN		(1 << 11)
+#define AUDIO_TX_RXINREC		(1 << 13)
+
+#define SSI_NETWORK_CDCTXRXSLOT(x)	(((x) & 0x3) << 2)
+#define SSI_NETWORK_CDCTXSECSLOT(x)	(((x) & 0x3) << 4)
+#define SSI_NETWORK_CDCRXSECSLOT(x)	(((x) & 0x3) << 6)
+#define SSI_NETWORK_CDCRXSECGAIN(x)	(((x) & 0x3) << 8)
+#define SSI_NETWORK_CDCSUMGAIN(x)	(1 << 10)
+#define SSI_NETWORK_CDCFSDLY(x)		(1 << 11)
+#define SSI_NETWORK_DAC_SLOTS_8		(1 << 12)
+#define SSI_NETWORK_DAC_SLOTS_4		(2 << 12)
+#define SSI_NETWORK_DAC_SLOTS_2		(3 << 12)
+#define SSI_NETWORK_DAC_SLOT_MASK	(3 << 12)
+#define SSI_NETWORK_DAC_RXSLOT_0_1	(0 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_2_3	(1 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_4_5	(2 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_6_7	(3 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_MASK	(3 << 14)
+#define SSI_NETWORK_STDCRXSECSLOT(x)	(((x) & 0x3) << 16)
+#define SSI_NETWORK_STDCRXSECGAIN(x)	(((x) & 0x3) << 18)
+#define SSI_NETWORK_STDCSUMGAIN		(1 << 20)
+
+/*
+ * MC13783_AUDIO_CODEC and MC13783_AUDIO_DAC mostly share the same
+ * register layout
+ */
+#define AUDIO_SSI_SEL			(1 << 0)
+#define AUDIO_CLK_SEL			(1 << 1)
+#define AUDIO_CSM			(1 << 2)
+#define AUDIO_BCL_INV			(1 << 3)
+#define AUDIO_CFS_INV			(1 << 4)
+#define AUDIO_CFS(x)			(((x) & 0x3) << 5)
+#define AUDIO_CLK(x)			(((x) & 0x7) << 7)
+#define AUDIO_C_EN			(1 << 11)
+#define AUDIO_C_CLK_EN			(1 << 12)
+#define AUDIO_C_RESET			(1 << 15)
+
+#define AUDIO_CODEC_CDCFS8K16K		(1 << 10)
+#define AUDIO_DAC_CFS_DLY_B		(1 << 10)
+
+struct mc13783_priv {
+	struct snd_soc_codec codec;
+	struct mc13xxx *mc13xxx;
+
+	enum mc13783_ssi_port adc_ssi_port;
+	enum mc13783_ssi_port dac_ssi_port;
+};
+
+static unsigned int mc13783_read(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+	unsigned int value = 0;
+
+	mc13xxx_lock(priv->mc13xxx);
+
+	mc13xxx_reg_read(priv->mc13xxx, reg, &value);
+
+	mc13xxx_unlock(priv->mc13xxx);
+
+	return value;
+}
+
+static int mc13783_write(struct snd_soc_codec *codec,
+	unsigned int reg, unsigned int value)
+{
+	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	mc13xxx_lock(priv->mc13xxx);
+
+	ret = mc13xxx_reg_write(priv->mc13xxx, reg, value);
+
+	mc13xxx_unlock(priv->mc13xxx);
+
+	return ret;
+}
+
+/* Mapping between sample rates and register value */
+static unsigned int mc13783_rates[] = {
+	8000, 11025, 12000, 16000,
+	22050, 24000, 32000, 44100,
+	48000, 64000, 96000
+};
+
+static int mc13783_pcm_hw_params_dac(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	unsigned int rate = params_rate(params);
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(mc13783_rates); i++) {
+		if (rate == mc13783_rates[i]) {
+			snd_soc_update_bits(codec, MC13783_AUDIO_DAC,
+					0xf << 17, i << 17);
+			return 0;
+		}
+	}
+
+	return -EINVAL;
+}
+
+static int mc13783_pcm_hw_params_codec(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	unsigned int rate = params_rate(params);
+	unsigned int val;
+
+	switch (rate) {
+	case 8000:
+		val = 0;
+		break;
+	case 16000:
+		val = AUDIO_CODEC_CDCFS8K16K;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, MC13783_AUDIO_CODEC, AUDIO_CODEC_CDCFS8K16K,
+			val);
+
+	return 0;
+}
+
+static int mc13783_pcm_hw_params_sync(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return mc13783_pcm_hw_params_dac(substream, params, dai);
+	else
+		return mc13783_pcm_hw_params_codec(substream, params, dai);
+}
+
+static int mc13783_set_fmt(struct snd_soc_dai *dai, unsigned int fmt,
+			unsigned int reg)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned int val = 0;
+	unsigned int mask = AUDIO_CFS(3) | AUDIO_BCL_INV | AUDIO_CFS_INV |
+				AUDIO_CSM | AUDIO_C_CLK_EN | AUDIO_C_RESET;
+
+
+	/* DAI mode */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		val |= AUDIO_CFS(2);
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		val |= AUDIO_CFS(1);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* DAI clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		val |= AUDIO_BCL_INV;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		val |= AUDIO_BCL_INV | AUDIO_CFS_INV;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		val |= AUDIO_CFS_INV;
+		break;
+	}
+
+	/* DAI clock master masks */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		val |= AUDIO_C_CLK_EN;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		val |= AUDIO_CSM;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+	case SND_SOC_DAIFMT_CBS_CFM:
+		return -EINVAL;
+	}
+
+	val |= AUDIO_C_RESET;
+
+	snd_soc_update_bits(codec, reg, mask, val);
+
+	return 0;
+}
+
+static int mc13783_set_fmt_async(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	if (dai->id == MC13783_ID_STEREO_DAC)
+		return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC);
+	else
+		return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC);
+}
+
+static int mc13783_set_fmt_sync(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	int ret;
+
+	ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC);
+	if (ret)
+		return ret;
+
+	/*
+	 * In synchronous mode force the voice codec into slave mode
+	 * so that the clock / framesync from the stereo DAC is used
+	 */
+	fmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+	fmt |= SND_SOC_DAIFMT_CBS_CFS;
+	ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC);
+
+	return ret;
+}
+
+static int mc13783_sysclk[] = {
+	13000000,
+	15360000,
+	16800000,
+	-1,
+	26000000,
+	-1, /* 12000000, invalid for voice codec */
+	-1, /* 3686400, invalid for voice codec */
+	33600000,
+};
+
+static int mc13783_set_sysclk(struct snd_soc_dai *dai,
+				  int clk_id, unsigned int freq, int dir,
+				  unsigned int reg)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	int clk;
+	unsigned int val = 0;
+	unsigned int mask = AUDIO_CLK(0x7) | AUDIO_CLK_SEL;
+
+	for (clk = 0; clk < ARRAY_SIZE(mc13783_sysclk); clk++) {
+		if (mc13783_sysclk[clk] < 0)
+			continue;
+		if (mc13783_sysclk[clk] == freq)
+			break;
+	}
+
+	if (clk == ARRAY_SIZE(mc13783_sysclk))
+		return -EINVAL;
+
+	if (clk_id == MC13783_CLK_CLIB)
+		val |= AUDIO_CLK_SEL;
+
+	val |= AUDIO_CLK(clk);
+
+	snd_soc_update_bits(codec, reg, mask, val);
+
+	return 0;
+}
+
+static int mc13783_set_sysclk_dac(struct snd_soc_dai *dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC);
+}
+
+static int mc13783_set_sysclk_codec(struct snd_soc_dai *dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC);
+}
+
+static int mc13783_set_sysclk_sync(struct snd_soc_dai *dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	int ret;
+
+	ret = mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC);
+	if (ret)
+		return ret;
+
+	return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC);
+}
+
+static int mc13783_set_tdm_slot_dac(struct snd_soc_dai *dai,
+	unsigned int tx_mask, unsigned int rx_mask, int slots,
+	int slot_width)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned int val = 0;
+	unsigned int mask = SSI_NETWORK_DAC_SLOT_MASK |
+				SSI_NETWORK_DAC_RXSLOT_MASK;
+
+	switch (slots) {
+	case 2:
+		val |= SSI_NETWORK_DAC_SLOTS_2;
+		break;
+	case 4:
+		val |= SSI_NETWORK_DAC_SLOTS_4;
+		break;
+	case 8:
+		val |= SSI_NETWORK_DAC_SLOTS_8;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (rx_mask) {
+	case 0xfffffffc:
+		val |= SSI_NETWORK_DAC_RXSLOT_0_1;
+		break;
+	case 0xfffffff3:
+		val |= SSI_NETWORK_DAC_RXSLOT_2_3;
+		break;
+	case 0xffffffcf:
+		val |= SSI_NETWORK_DAC_RXSLOT_4_5;
+		break;
+	case 0xffffff3f:
+		val |= SSI_NETWORK_DAC_RXSLOT_6_7;
+		break;
+	default:
+		return -EINVAL;
+	};
+
+	snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val);
+
+	return 0;
+}
+
+static int mc13783_set_tdm_slot_codec(struct snd_soc_dai *dai,
+	unsigned int tx_mask, unsigned int rx_mask, int slots,
+	int slot_width)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned int val = 0;
+	unsigned int mask = 0x3f;
+
+	if (slots != 4)
+		return -EINVAL;
+
+	if (tx_mask != 0xfffffffc)
+		return -EINVAL;
+
+	val |= (0x00 << 2);	/* primary timeslot RX/TX(?) is 0 */
+	val |= (0x01 << 4);	/* secondary timeslot TX is 1 */
+
+	snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val);
+
+	return 0;
+}
+
+static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai,
+	unsigned int tx_mask, unsigned int rx_mask, int slots,
+	int slot_width)
+{
+	int ret;
+
+	ret = mc13783_set_tdm_slot_dac(dai, tx_mask, rx_mask, slots,
+			slot_width);
+	if (ret)
+		return ret;
+
+	ret = mc13783_set_tdm_slot_codec(dai, tx_mask, rx_mask, slots,
+			slot_width);
+
+	return ret;
+}
+
+static const struct snd_kcontrol_new mc1l_amp_ctl =
+	SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0);
+
+static const struct snd_kcontrol_new mc1r_amp_ctl =
+	SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0);
+
+static const struct snd_kcontrol_new mc2_amp_ctl =
+	SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0);
+
+static const struct snd_kcontrol_new atx_amp_ctl =
+	SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0);
+
+
+/* Virtual mux. The chip does the input selection automatically
+ * as soon as we enable one input. */
+static const char * const adcl_enum_text[] = {
+	"MC1L", "RXINL",
+};
+
+static const struct soc_enum adcl_enum =
+	SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcl_enum_text), adcl_enum_text);
+
+static const struct snd_kcontrol_new left_input_mux =
+	SOC_DAPM_ENUM_VIRT("Route", adcl_enum);
+
+static const char * const adcr_enum_text[] = {
+	"MC1R", "MC2", "RXINR", "TXIN",
+};
+
+static const struct soc_enum adcr_enum =
+	SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcr_enum_text), adcr_enum_text);
+
+static const struct snd_kcontrol_new right_input_mux =
+	SOC_DAPM_ENUM_VIRT("Route", adcr_enum);
+
+static const struct snd_kcontrol_new samp_ctl =
+	SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0);
+
+static const struct snd_kcontrol_new lamp_ctl =
+	SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0);
+
+static const struct snd_kcontrol_new hlamp_ctl =
+	SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0);
+
+static const struct snd_kcontrol_new hramp_ctl =
+	SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0);
+
+static const struct snd_kcontrol_new llamp_ctl =
+	SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0);
+
+static const struct snd_kcontrol_new lramp_ctl =
+	SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0);
+
+static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = {
+/* Input */
+	SND_SOC_DAPM_INPUT("MC1LIN"),
+	SND_SOC_DAPM_INPUT("MC1RIN"),
+	SND_SOC_DAPM_INPUT("MC2IN"),
+	SND_SOC_DAPM_INPUT("RXINR"),
+	SND_SOC_DAPM_INPUT("RXINL"),
+	SND_SOC_DAPM_INPUT("TXIN"),
+
+	SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0),
+
+	SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl),
+	SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl),
+	SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl),
+	SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl),
+
+	SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0,
+			      &left_input_mux),
+	SND_SOC_DAPM_VIRT_MUX("PGA Right Input Mux", SND_SOC_NOPM, 0, 0,
+			      &right_input_mux),
+
+	SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0),
+	SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0),
+
+/* Output */
+	SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("RXOUTL"),
+	SND_SOC_DAPM_OUTPUT("RXOUTR"),
+	SND_SOC_DAPM_OUTPUT("HSL"),
+	SND_SOC_DAPM_OUTPUT("HSR"),
+	SND_SOC_DAPM_OUTPUT("LSP"),
+	SND_SOC_DAPM_OUTPUT("SP"),
+
+	SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl),
+	SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl),
+	SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl),
+	SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl),
+	SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl),
+	SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl),
+	SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0),
+	SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0),
+};
+
+static struct snd_soc_dapm_route mc13783_routes[] = {
+/* Input */
+	{ "MC1L Amp", NULL, "MC1LIN"},
+	{ "MC1R Amp", NULL, "MC1RIN" },
+	{ "MC2 Amp", NULL, "MC2IN" },
+	{ "TXIN Amp", NULL, "TXIN"},
+
+	{ "PGA Left Input Mux", "MC1L", "MC1L Amp" },
+	{ "PGA Left Input Mux", "RXINL", "RXINL"},
+	{ "PGA Right Input Mux", "MC1R", "MC1R Amp" },
+	{ "PGA Right Input Mux", "MC2",  "MC2 Amp"},
+	{ "PGA Right Input Mux", "TXIN", "TXIN Amp"},
+	{ "PGA Right Input Mux", "RXINR", "RXINR"},
+
+	{ "PGA Left Input", NULL, "PGA Left Input Mux"},
+	{ "PGA Right Input", NULL, "PGA Right Input Mux"},
+
+	{ "ADC", NULL, "PGA Left Input"},
+	{ "ADC", NULL, "PGA Right Input"},
+	{ "ADC", NULL, "ADC_Reset"},
+
+/* Output */
+	{ "HSL", NULL, "Headset Amp Left" },
+	{ "HSR", NULL, "Headset Amp Right"},
+	{ "RXOUTL", NULL, "Line out Amp Left"},
+	{ "RXOUTR", NULL, "Line out Amp Right"},
+	{ "SP", NULL, "Speaker Amp"},
+	{ "Speaker Amp", NULL, "DAC PGA"},
+	{ "LSP", NULL, "DAC PGA"},
+	{ "Headset Amp Left", NULL, "DAC PGA"},
+	{ "Headset Amp Right", NULL, "DAC PGA"},
+	{ "Line out Amp Left", NULL, "DAC PGA"},
+	{ "Line out Amp Right", NULL, "DAC PGA"},
+	{ "DAC PGA", NULL, "DAC"},
+	{ "DAC", NULL, "DAC_E"},
+};
+
+static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix",
+						"Mono", "Mono Mix"};
+
+static const struct soc_enum mc13783_enum_3d_mixer =
+	SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 16, ARRAY_SIZE(mc13783_3d_mixer),
+			mc13783_3d_mixer);
+
+static struct snd_kcontrol_new mc13783_control_list[] = {
+	SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0),
+	SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0),
+	SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0),
+	SOC_ENUM("3D Control", mc13783_enum_3d_mixer),
+};
+
+static int mc13783_probe(struct snd_soc_codec *codec)
+{
+	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	mc13xxx_lock(priv->mc13xxx);
+
+	/* these are the reset values */
+	mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX0, 0x25893);
+	mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX1, 0x00d35A);
+	mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_TX, 0x420000);
+	mc13xxx_reg_write(priv->mc13xxx, MC13783_SSI_NETWORK, 0x013060);
+	mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_CODEC, 0x180027);
+	mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_DAC, 0x0e0004);
+
+	snd_soc_add_codec_controls(codec, mc13783_control_list,
+					ARRAY_SIZE(mc13783_control_list));
+
+	snd_soc_dapm_new_controls(dapm, mc13783_dapm_widgets,
+					ARRAY_SIZE(mc13783_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, mc13783_routes,
+					ARRAY_SIZE(mc13783_routes));
+
+	if (priv->adc_ssi_port == MC13783_SSI1_PORT)
+		mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC,
+				AUDIO_SSI_SEL, 0);
+	else
+		mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC,
+				0, AUDIO_SSI_SEL);
+
+	if (priv->dac_ssi_port == MC13783_SSI1_PORT)
+		mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC,
+				AUDIO_SSI_SEL, 0);
+	else
+		mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC,
+				0, AUDIO_SSI_SEL);
+
+	mc13xxx_unlock(priv->mc13xxx);
+
+	return 0;
+}
+
+static int mc13783_remove(struct snd_soc_codec *codec)
+{
+	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	mc13xxx_lock(priv->mc13xxx);
+
+	/* Make sure VAUDIOON is off */
+	mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_RX0, 0x3, 0);
+
+	mc13xxx_unlock(priv->mc13xxx);
+
+	return 0;
+}
+
+#define MC13783_RATES_RECORD (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000)
+
+#define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops mc13783_ops_dac = {
+	.hw_params	= mc13783_pcm_hw_params_dac,
+	.set_fmt	= mc13783_set_fmt_async,
+	.set_sysclk	= mc13783_set_sysclk_dac,
+	.set_tdm_slot	= mc13783_set_tdm_slot_dac,
+};
+
+static struct snd_soc_dai_ops mc13783_ops_codec = {
+	.hw_params	= mc13783_pcm_hw_params_codec,
+	.set_fmt	= mc13783_set_fmt_async,
+	.set_sysclk	= mc13783_set_sysclk_codec,
+	.set_tdm_slot	= mc13783_set_tdm_slot_codec,
+};
+
+/*
+ * The mc13783 has two SSI ports, both of them can be routed either
+ * to the voice codec or the stereo DAC. When two different SSI ports
+ * are used for the voice codec and the stereo DAC we can do different
+ * formats and sysclock settings for playback and capture
+ * (mc13783-hifi-playback and mc13783-hifi-capture). Using the same port
+ * forces us to use symmetric rates (mc13783-hifi).
+ */
+static struct snd_soc_dai_driver mc13783_dai_async[] = {
+	{
+		.name = "mc13783-hifi-playback",
+		.id = MC13783_ID_STEREO_DAC,
+		.playback = {
+			.stream_name = "Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_96000,
+			.formats = MC13783_FORMATS,
+		},
+		.ops = &mc13783_ops_dac,
+	}, {
+		.name = "mc13783-hifi-capture",
+		.id = MC13783_ID_STEREO_CODEC,
+		.capture = {
+			.stream_name = "Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MC13783_RATES_RECORD,
+			.formats = MC13783_FORMATS,
+		},
+		.ops = &mc13783_ops_codec,
+	},
+};
+
+static struct snd_soc_dai_ops mc13783_ops_sync = {
+	.hw_params	= mc13783_pcm_hw_params_sync,
+	.set_fmt	= mc13783_set_fmt_sync,
+	.set_sysclk	= mc13783_set_sysclk_sync,
+	.set_tdm_slot	= mc13783_set_tdm_slot_sync,
+};
+
+static struct snd_soc_dai_driver mc13783_dai_sync[] = {
+	{
+		.name = "mc13783-hifi",
+		.id = MC13783_ID_SYNC,
+		.playback = {
+			.stream_name = "Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_96000,
+			.formats = MC13783_FORMATS,
+		},
+		.capture = {
+			.stream_name = "Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MC13783_RATES_RECORD,
+			.formats = MC13783_FORMATS,
+		},
+		.ops = &mc13783_ops_sync,
+		.symmetric_rates = 1,
+	}
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_mc13783 = {
+	.probe		= mc13783_probe,
+	.remove		= mc13783_remove,
+	.read		= mc13783_read,
+	.write		= mc13783_write,
+};
+
+static int mc13783_codec_probe(struct platform_device *pdev)
+{
+	struct mc13xxx *mc13xxx;
+	struct mc13783_priv *priv;
+	struct mc13xxx_codec_platform_data *pdata = pdev->dev.platform_data;
+	int ret;
+
+	mc13xxx = dev_get_drvdata(pdev->dev.parent);
+
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (priv == NULL)
+		return -ENOMEM;
+
+	dev_set_drvdata(&pdev->dev, priv);
+	priv->mc13xxx = mc13xxx;
+	if (pdata) {
+		priv->adc_ssi_port = pdata->adc_ssi_port;
+		priv->dac_ssi_port = pdata->dac_ssi_port;
+	} else {
+		priv->adc_ssi_port = MC13783_SSI1_PORT;
+		priv->dac_ssi_port = MC13783_SSI2_PORT;
+	}
+
+	if (priv->adc_ssi_port == priv->dac_ssi_port)
+		ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
+			mc13783_dai_sync, ARRAY_SIZE(mc13783_dai_sync));
+	else
+		ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
+			mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async));
+
+	if (ret)
+		goto err_register_codec;
+
+	return 0;
+
+err_register_codec:
+	dev_err(&pdev->dev, "register codec failed with %d\n", ret);
+
+	return ret;
+}
+
+static int mc13783_codec_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver mc13783_codec_driver = {
+	.driver = {
+		   .name = "mc13783-codec",
+		   .owner = THIS_MODULE,
+		   },
+	.probe = mc13783_codec_probe,
+	.remove = __devexit_p(mc13783_codec_remove),
+};
+
+static __init int mc13783_init(void)
+{
+	return platform_driver_register(&mc13783_codec_driver);
+}
+
+static __exit void mc13783_exit(void)
+{
+	platform_driver_unregister(&mc13783_codec_driver);
+}
+
+module_init(mc13783_init);
+module_exit(mc13783_exit);
+
+MODULE_DESCRIPTION("ASoC MC13783 driver");
+MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>");
+MODULE_AUTHOR("Philippe Retornaz <philippe.retornaz@epfl.ch>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/mc13783.h b/sound/soc/codecs/mc13783.h
new file mode 100644
index 000000000000..3a6d1993a217
--- /dev/null
+++ b/sound/soc/codecs/mc13783.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright 2008 Juergen Beisert, kernel@pengutronix.de
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ * 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 Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ */
+
+#ifndef MC13783_MIXER_H
+#define MC13783_MIXER_H
+
+#define MC13783_CLK_CLIA	1
+#define MC13783_CLK_CLIB	2
+
+#define MC13783_ID_STEREO_DAC	1
+#define MC13783_ID_STEREO_CODEC	2
+#define MC13783_ID_SYNC		3
+
+#endif /* MC13783_MIXER_H */