[Feature]add MT2731_MP2_MR2_SVN388 baseline version

Change-Id: Ief04314834b31e27effab435d3ca8ba33b499059
diff --git a/src/kernel/linux/v4.14/sound/soc/codecs/tlv320aic23.c b/src/kernel/linux/v4.14/sound/soc/codecs/tlv320aic23.c
new file mode 100644
index 0000000..3d42138
--- /dev/null
+++ b/src/kernel/linux/v4.14/sound/soc/codecs/tlv320aic23.c
@@ -0,0 +1,617 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver
+ *
+ * Author:      Arun KS, <arunks@mistralsolutions.com>
+ * Copyright:   (C) 2008 Mistral Solutions Pvt Ltd.,
+ *
+ * Based on sound/soc/codecs/wm8731.c by Richard Purdie
+ *
+ * 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.
+ *
+ * Notes:
+ *  The AIC23 is a driver for a low power stereo audio
+ *  codec tlv320aic23
+ *
+ *  The machine layer should disable unsupported inputs/outputs by
+ *  snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+
+#include "tlv320aic23.h"
+
+/*
+ * AIC23 register cache
+ */
+static const struct reg_default tlv320aic23_reg[] = {
+	{  0, 0x0097 },
+	{  1, 0x0097 },
+	{  2, 0x00F9 },
+	{  3, 0x00F9 },
+	{  4, 0x001A },
+	{  5, 0x0004 },
+	{  6, 0x0007 },
+	{  7, 0x0001 },
+	{  8, 0x0020 },
+	{  9, 0x0000 },
+};
+
+const struct regmap_config tlv320aic23_regmap = {
+	.reg_bits = 7,
+	.val_bits = 9,
+
+	.max_register = TLV320AIC23_RESET,
+	.reg_defaults = tlv320aic23_reg,
+	.num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg),
+	.cache_type = REGCACHE_RBTREE,
+};
+EXPORT_SYMBOL(tlv320aic23_regmap);
+
+static const char *rec_src_text[] = { "Line", "Mic" };
+static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+
+static SOC_ENUM_SINGLE_DECL(rec_src_enum,
+			    TLV320AIC23_ANLG, 2, rec_src_text);
+
+static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
+SOC_DAPM_ENUM("Input Select", rec_src_enum);
+
+static SOC_ENUM_SINGLE_DECL(tlv320aic23_rec_src,
+			    TLV320AIC23_ANLG, 2, rec_src_text);
+static SOC_ENUM_SINGLE_DECL(tlv320aic23_deemph,
+			    TLV320AIC23_DIGT, 1, deemph_text);
+
+static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0);
+static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0);
+static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0);
+
+static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	u16 val, reg;
+
+	val = (ucontrol->value.integer.value[0] & 0x07);
+
+	/* linear conversion to userspace
+	* 000	=	-6db
+	* 001	=	-9db
+	* 010	=	-12db
+	* 011	=	-18db (Min)
+	* 100	=	0db (Max)
+	*/
+	val = (val >= 4) ? 4  : (3 - val);
+
+	reg = snd_soc_read(codec, TLV320AIC23_ANLG) & (~0x1C0);
+	snd_soc_write(codec, TLV320AIC23_ANLG, reg | (val << 6));
+
+	return 0;
+}
+
+static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	u16 val;
+
+	val = snd_soc_read(codec, TLV320AIC23_ANLG) & (0x1C0);
+	val = val >> 6;
+	val = (val >= 4) ? 4  : (3 -  val);
+	ucontrol->value.integer.value[0] = val;
+	return 0;
+
+}
+
+static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL,
+			 TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv),
+	SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1),
+	SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL,
+		     TLV320AIC23_RINVOL, 7, 1, 0),
+	SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL,
+			 TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv),
+	SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1),
+	SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0),
+	SOC_SINGLE_EXT_TLV("Sidetone Volume", TLV320AIC23_ANLG, 6, 4, 0,
+			   snd_soc_tlv320aic23_get_volsw,
+			   snd_soc_tlv320aic23_put_volsw, sidetone_vol_tlv),
+	SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
+};
+
+/* PGA Mixer controls for Line and Mic switch */
+static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
+	SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0),
+	SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1),
+	SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1),
+	SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0,
+			 &tlv320aic23_rec_src_mux_controls),
+	SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1,
+			   &tlv320aic23_output_mixer_controls[0],
+			   ARRAY_SIZE(tlv320aic23_output_mixer_controls)),
+	SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("LHPOUT"),
+	SND_SOC_DAPM_OUTPUT("RHPOUT"),
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+	SND_SOC_DAPM_OUTPUT("ROUT"),
+
+	SND_SOC_DAPM_INPUT("LLINEIN"),
+	SND_SOC_DAPM_INPUT("RLINEIN"),
+
+	SND_SOC_DAPM_INPUT("MICIN"),
+};
+
+static const struct snd_soc_dapm_route tlv320aic23_intercon[] = {
+	/* Output Mixer */
+	{"Output Mixer", "Line Bypass Switch", "Line Input"},
+	{"Output Mixer", "Playback Switch", "DAC"},
+	{"Output Mixer", "Mic Sidetone Switch", "Mic Input"},
+
+	/* Outputs */
+	{"RHPOUT", NULL, "Output Mixer"},
+	{"LHPOUT", NULL, "Output Mixer"},
+	{"LOUT", NULL, "Output Mixer"},
+	{"ROUT", NULL, "Output Mixer"},
+
+	/* Inputs */
+	{"Line Input", NULL, "LLINEIN"},
+	{"Line Input", NULL, "RLINEIN"},
+	{"Mic Input", NULL, "MICIN"},
+
+	/* input mux */
+	{"Capture Source", "Line", "Line Input"},
+	{"Capture Source", "Mic", "Mic Input"},
+	{"ADC", NULL, "Capture Source"},
+
+};
+
+/* AIC23 driver data */
+struct aic23 {
+	struct regmap *regmap;
+	int mclk;
+	int requested_adc;
+	int requested_dac;
+};
+
+/*
+ * Common Crystals used
+ * 11.2896 Mhz /128 = *88.2k  /192 = 58.8k
+ * 12.0000 Mhz /125 = *96k    /136 = 88.235K
+ * 12.2880 Mhz /128 = *96k    /192 = 64k
+ * 16.9344 Mhz /128 = 132.3k /192 = *88.2k
+ * 18.4320 Mhz /128 = 144k   /192 = *96k
+ */
+
+/*
+ * Normal BOSR 0-256/2 = 128, 1-384/2 = 192
+ * USB BOSR 0-250/2 = 125, 1-272/2 = 136
+ */
+static const int bosr_usb_divisor_table[] = {
+	128, 125, 192, 136
+};
+#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7))
+#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11)        | (1<<15))
+static const unsigned short sr_valid_mask[] = {
+	LOWER_GROUP|UPPER_GROUP,	/* Normal, bosr - 0*/
+	LOWER_GROUP,			/* Usb, bosr - 0*/
+	LOWER_GROUP|UPPER_GROUP,	/* Normal, bosr - 1*/
+	UPPER_GROUP,			/* Usb, bosr - 1*/
+};
+/*
+ * Every divisor is a factor of 11*12
+ */
+#define SR_MULT (11*12)
+#define A(x) (SR_MULT/x)
+static const unsigned char sr_adc_mult_table[] = {
+	A(2), A(2), A(12), A(12),  0, 0, A(3), A(1),
+	A(2), A(2), A(11), A(11),  0, 0, 0, A(1)
+};
+static const unsigned char sr_dac_mult_table[] = {
+	A(2), A(12), A(2), A(12),  0, 0, A(3), A(1),
+	A(2), A(11), A(2), A(11),  0, 0, 0, A(1)
+};
+
+static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
+		int dac, int dac_l, int dac_h, int need_dac)
+{
+	if ((adc >= adc_l) && (adc <= adc_h) &&
+			(dac >= dac_l) && (dac <= dac_h)) {
+		int diff_adc = need_adc - adc;
+		int diff_dac = need_dac - dac;
+		return abs(diff_adc) + abs(diff_dac);
+	}
+	return UINT_MAX;
+}
+
+static int find_rate(int mclk, u32 need_adc, u32 need_dac)
+{
+	int i, j;
+	int best_i = -1;
+	int best_j = -1;
+	int best_div = 0;
+	unsigned best_score = UINT_MAX;
+	int adc_l, adc_h, dac_l, dac_h;
+
+	need_adc *= SR_MULT;
+	need_dac *= SR_MULT;
+	/*
+	 * rates given are +/- 1/32
+	 */
+	adc_l = need_adc - (need_adc >> 5);
+	adc_h = need_adc + (need_adc >> 5);
+	dac_l = need_dac - (need_dac >> 5);
+	dac_h = need_dac + (need_dac >> 5);
+	for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) {
+		int base = mclk / bosr_usb_divisor_table[i];
+		int mask = sr_valid_mask[i];
+		for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table);
+				j++, mask >>= 1) {
+			int adc;
+			int dac;
+			int score;
+			if ((mask & 1) == 0)
+				continue;
+			adc = base * sr_adc_mult_table[j];
+			dac = base * sr_dac_mult_table[j];
+			score = get_score(adc, adc_l, adc_h, need_adc,
+					dac, dac_l, dac_h, need_dac);
+			if (best_score > score) {
+				best_score = score;
+				best_i = i;
+				best_j = j;
+				best_div = 0;
+			}
+			score = get_score((adc >> 1), adc_l, adc_h, need_adc,
+					(dac >> 1), dac_l, dac_h, need_dac);
+			/* prefer to have a /2 */
+			if ((score != UINT_MAX) && (best_score >= score)) {
+				best_score = score;
+				best_i = i;
+				best_j = j;
+				best_div = 1;
+			}
+		}
+	}
+	return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT);
+}
+
+#ifdef DEBUG
+static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk,
+		u32 *sample_rate_adc, u32 *sample_rate_dac)
+{
+	int src = snd_soc_read(codec, TLV320AIC23_SRATE);
+	int sr = (src >> 2) & 0x0f;
+	int val = (mclk / bosr_usb_divisor_table[src & 3]);
+	int adc = (val * sr_adc_mult_table[sr]) / SR_MULT;
+	int dac = (val * sr_dac_mult_table[sr]) / SR_MULT;
+	if (src & TLV320AIC23_CLKIN_HALF) {
+		adc >>= 1;
+		dac >>= 1;
+	}
+	*sample_rate_adc = adc;
+	*sample_rate_dac = dac;
+}
+#endif
+
+static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
+		u32 sample_rate_adc, u32 sample_rate_dac)
+{
+	/* Search for the right sample rate */
+	int data = find_rate(mclk, sample_rate_adc, sample_rate_dac);
+	if (data < 0) {
+		printk(KERN_ERR "%s:Invalid rate %u,%u requested\n",
+				__func__, sample_rate_adc, sample_rate_dac);
+		return -EINVAL;
+	}
+	snd_soc_write(codec, TLV320AIC23_SRATE, data);
+#ifdef DEBUG
+	{
+		u32 adc, dac;
+		get_current_sample_rates(codec, mclk, &adc, &dac);
+		printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n",
+			adc, dac, data);
+	}
+#endif
+	return 0;
+}
+
+static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 iface_reg;
+	int ret;
+	struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
+	u32 sample_rate_adc = aic23->requested_adc;
+	u32 sample_rate_dac = aic23->requested_dac;
+	u32 sample_rate = params_rate(params);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		aic23->requested_dac = sample_rate_dac = sample_rate;
+		if (!sample_rate_adc)
+			sample_rate_adc = sample_rate;
+	} else {
+		aic23->requested_adc = sample_rate_adc = sample_rate;
+		if (!sample_rate_dac)
+			sample_rate_dac = sample_rate;
+	}
+	ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc,
+			sample_rate_dac);
+	if (ret < 0)
+		return ret;
+
+	iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
+
+	switch (params_width(params)) {
+	case 16:
+		break;
+	case 20:
+		iface_reg |= (0x01 << 2);
+		break;
+	case 24:
+		iface_reg |= (0x02 << 2);
+		break;
+	case 32:
+		iface_reg |= (0x03 << 2);
+		break;
+	}
+	snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+
+	return 0;
+}
+
+static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
+				   struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	/* set active */
+	snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001);
+
+	return 0;
+}
+
+static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
+
+	/* deactivate */
+	if (!snd_soc_codec_is_active(codec)) {
+		udelay(50);
+		snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
+	}
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		aic23->requested_dac = 0;
+	else
+		aic23->requested_adc = 0;
+}
+
+static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 reg;
+
+	reg = snd_soc_read(codec, TLV320AIC23_DIGT);
+	if (mute)
+		reg |= TLV320AIC23_DACM_MUTE;
+
+	else
+		reg &= ~TLV320AIC23_DACM_MUTE;
+
+	snd_soc_write(codec, TLV320AIC23_DIGT, reg);
+
+	return 0;
+}
+
+static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
+				   unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface_reg;
+
+	iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & (~0x03);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		iface_reg |= TLV320AIC23_MS_MASTER;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		iface_reg &= ~TLV320AIC23_MS_MASTER;
+		break;
+	default:
+		return -EINVAL;
+
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface_reg |= TLV320AIC23_FOR_I2S;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface_reg |= TLV320AIC23_LRP_ON;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface_reg |= TLV320AIC23_FOR_DSP;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface_reg |= TLV320AIC23_FOR_LJUST;
+		break;
+	default:
+		return -EINVAL;
+
+	}
+
+	snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+
+	return 0;
+}
+
+static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				      int clk_id, unsigned int freq, int dir)
+{
+	struct aic23 *aic23 = snd_soc_dai_get_drvdata(codec_dai);
+	aic23->mclk = freq;
+	return 0;
+}
+
+static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
+				      enum snd_soc_bias_level level)
+{
+	u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		/* vref/mid, osc on, dac unmute */
+		reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \
+			TLV320AIC23_DAC_OFF);
+		snd_soc_write(codec, TLV320AIC23_PWR, reg);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* everything off except vref/vmid, */
+		snd_soc_write(codec, TLV320AIC23_PWR,
+			      reg | TLV320AIC23_CLK_OFF);
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* everything off, dac mute, inactive */
+		snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
+		snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff);
+		break;
+	}
+	return 0;
+}
+
+#define AIC23_RATES	SNDRV_PCM_RATE_8000_96000
+#define AIC23_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static const struct snd_soc_dai_ops tlv320aic23_dai_ops = {
+	.prepare	= tlv320aic23_pcm_prepare,
+	.hw_params	= tlv320aic23_hw_params,
+	.shutdown	= tlv320aic23_shutdown,
+	.digital_mute	= tlv320aic23_mute,
+	.set_fmt	= tlv320aic23_set_dai_fmt,
+	.set_sysclk	= tlv320aic23_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver tlv320aic23_dai = {
+	.name = "tlv320aic23-hifi",
+	.playback = {
+		     .stream_name = "Playback",
+		     .channels_min = 2,
+		     .channels_max = 2,
+		     .rates = AIC23_RATES,
+		     .formats = AIC23_FORMATS,},
+	.capture = {
+		    .stream_name = "Capture",
+		    .channels_min = 2,
+		    .channels_max = 2,
+		    .rates = AIC23_RATES,
+		    .formats = AIC23_FORMATS,},
+	.ops = &tlv320aic23_dai_ops,
+};
+
+static int tlv320aic23_resume(struct snd_soc_codec *codec)
+{
+	struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
+	regcache_mark_dirty(aic23->regmap);
+	regcache_sync(aic23->regmap);
+
+	return 0;
+}
+
+static int tlv320aic23_codec_probe(struct snd_soc_codec *codec)
+{
+	/* Reset codec */
+	snd_soc_write(codec, TLV320AIC23_RESET, 0);
+
+	snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
+
+	/* Unmute input */
+	snd_soc_update_bits(codec, TLV320AIC23_LINVOL,
+			    TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED);
+
+	snd_soc_update_bits(codec, TLV320AIC23_RINVOL,
+			    TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED);
+
+	snd_soc_update_bits(codec, TLV320AIC23_ANLG,
+			    TLV320AIC23_BYPASS_ON | TLV320AIC23_MICM_MUTED,
+			    0);
+
+	/* Default output volume */
+	snd_soc_write(codec, TLV320AIC23_LCHNVOL,
+		      TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK);
+	snd_soc_write(codec, TLV320AIC23_RCHNVOL,
+		      TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK);
+
+	snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1);
+
+	return 0;
+}
+
+static const struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = {
+	.probe = tlv320aic23_codec_probe,
+	.resume = tlv320aic23_resume,
+	.set_bias_level = tlv320aic23_set_bias_level,
+	.suspend_bias_off = true,
+
+	.component_driver = {
+		.controls		= tlv320aic23_snd_controls,
+		.num_controls		= ARRAY_SIZE(tlv320aic23_snd_controls),
+		.dapm_widgets		= tlv320aic23_dapm_widgets,
+		.num_dapm_widgets	= ARRAY_SIZE(tlv320aic23_dapm_widgets),
+		.dapm_routes		= tlv320aic23_intercon,
+		.num_dapm_routes	= ARRAY_SIZE(tlv320aic23_intercon),
+	},
+};
+
+int tlv320aic23_probe(struct device *dev, struct regmap *regmap)
+{
+	struct aic23 *aic23;
+
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
+
+	aic23 = devm_kzalloc(dev, sizeof(struct aic23), GFP_KERNEL);
+	if (aic23 == NULL)
+		return -ENOMEM;
+
+	aic23->regmap = regmap;
+
+	dev_set_drvdata(dev, aic23);
+
+	return snd_soc_register_codec(dev, &soc_codec_dev_tlv320aic23,
+				      &tlv320aic23_dai, 1);
+}
+EXPORT_SYMBOL(tlv320aic23_probe);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_LICENSE("GPL");