[PATCH v4 4/4] ASoC: samsung: Add machine driver for Exynos5433 based TM2 board

[Sylwester Nawrocki]

This patch adds the sound machine driver for TM2 and TM2E board.
Speaker and headphone playback, Main Mic capture, Bluetooth,
Voice call and external accessory are supported.

Signed-off-by: Inha Song <ideal.song@xxxxxxxxxxx>
[k.kozlowski: rebased on 4.1]
Signed-off-by: Krzysztof Kozlowski <k.kozlowski@xxxxxxxxxxx>
[s.nawrocki: rebased to 4.7, adjustment to the ASoC core changes,
 removed unused ops and direct calls to the max98504 function,
 added parsing of "audio-amplifier" and "audio-codec"
 properties, added TDM API calls, switched to gpiod API]
Signed-off-by: Sylwester Nawrocki <s.nawrocki@xxxxxxxxxxx>
---
Changes since v3:
 - removed SND_SOC_SAMSUNG_AUDSS from Kconfig.

Changes since v2:
 - added missing Kconfig dependencies.

Changes since initial version:
 - added PDM Tx channels setup through TDM API
 - adaptation to renamed 'samsung,model', 'samsung,i2s-controller',
   'samsung,speaker-amplifier' properties,
 - removed some dev_dbg() calls,
 - cleaned up mic-bias GPIO handling and switched to gpiod API,
 - added parsing of 'audio-codec' property,
 - initialized codec_of_node of dai_link instead of codec_name,
 - switched to using clock, clock-names properties from the wm5110
   codec node,
 - fixed error paths in probe() (of_node reference counting).
---
 sound/soc/samsung/Kconfig      |   9 +
 sound/soc/samsung/Makefile     |   2 +
 sound/soc/samsung/tm2_wm5110.c | 579 +++++++++++++++++++++++++++++++++++++++++
 3 files changed, 590 insertions(+)
 create mode 100644 sound/soc/samsung/tm2_wm5110.c

diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 78baa26..a711605 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -237,3 +237,12 @@ config SND_SOC_ARNDALE_RT5631_ALC5631
         depends on SND_SOC_SAMSUNG && I2C
         select SND_SAMSUNG_I2S
         select SND_SOC_RT5631
+
+config SND_SOC_SAMSUNG_TM2_WM5110
+	tristate "SoC I2S Audio support for WM5110 on TM2 board"
+	depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER
+	select SND_SOC_MAX98504
+	select SND_SOC_WM5110
+	select SND_SAMSUNG_I2S
+	help
+	  Say Y if you want to add support for SoC audio on the TM2 board.
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 052fe71..c15a759 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -45,6 +45,7 @@ snd-soc-littlemill-objs := littlemill.o
 snd-soc-bells-objs := bells.o
 snd-soc-odroidx2-max98090-objs := odroidx2_max98090.o
 snd-soc-arndale-rt5631-objs := arndale_rt5631.o
+snd-soc-tm2-wm5110-objs := tm2_wm5110.o
 
 obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
@@ -71,3 +72,4 @@ obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o
 obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o
 obj-$(CONFIG_SND_SOC_ODROIDX2) += snd-soc-odroidx2-max98090.o
 obj-$(CONFIG_SND_SOC_ARNDALE_RT5631_ALC5631) += snd-soc-arndale-rt5631.o
+obj-$(CONFIG_SND_SOC_SAMSUNG_TM2_WM5110) += snd-soc-tm2-wm5110.o
diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c
new file mode 100644
index 0000000..9728b3c
--- /dev/null
+++ b/sound/soc/samsung/tm2_wm5110.c
@@ -0,0 +1,579 @@
+/*
+ * Copyright (C) 2015 - 2016 Samsung Electronics Co., Ltd.
+ *
+ * Authors: Inha Song <ideal.song@xxxxxxxxxxx>
+ *          Sylwester Nawrocki <s.nawrocki@xxxxxxxxxxx>
+ *
+ * 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.
+ */
+
+#include <linux/clk.h>
+#include <linux/gpio.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "i2s.h"
+#include "../codecs/wm5110.h"
+
+struct tm2_machine_priv {
+	struct snd_soc_codec *codec;
+	struct clk *codec_mclk1;
+	struct clk *codec_mclk2;
+
+	unsigned int sysclk_rate;
+
+	struct gpio_desc *gpio_mic_bias;
+};
+
+static int tm2_start_sysclk(struct snd_soc_card *card)
+{
+	struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_codec *codec = priv->codec;
+	unsigned long mclk_rate = clk_get_rate(priv->codec_mclk1);
+	int ret;
+
+	ret = clk_prepare_enable(priv->codec_mclk1);
+	if (ret < 0) {
+		dev_err(card->dev, "Failed to enable mclk: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_codec_set_pll(codec, WM5110_FLL1,
+				    ARIZONA_FLL_SRC_MCLK1,
+				    mclk_rate,
+				    priv->sysclk_rate);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to start FLL: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_codec_set_pll(codec, WM5110_FLL1_REFCLK,
+				    ARIZONA_FLL_SRC_MCLK1,
+				    mclk_rate,
+				    priv->sysclk_rate);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set FLL1 Source: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK,
+				       ARIZONA_CLK_SRC_FLL1,
+				       priv->sysclk_rate,
+				       SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set SYSCLK Source: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int tm2_stop_sysclk(struct snd_soc_card *card)
+{
+	struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_codec *codec = priv->codec;
+	int ret;
+
+	ret = snd_soc_codec_set_pll(codec, WM5110_FLL1, 0, 0, 0);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to stop FLL: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK,
+				       ARIZONA_CLK_SRC_FLL1, 0, 0);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to stop SYSCLK: %d\n", ret);
+		return ret;
+	}
+
+	clk_disable_unprepare(priv->codec_mclk1);
+
+	return 0;
+}
+
+static int tm2_aif1_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 tm2_machine_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	int ret;
+
+	switch (params_rate(params)) {
+	case 4000:
+	case 8000:
+	case 12000:
+	case 16000:
+	case 24000:
+	case 32000:
+	case 48000:
+	case 96000:
+	case 192000:
+		/* Highest possible SYSCLK frequency: 147.456MHz */
+		priv->sysclk_rate = 147456000U;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+	case 88200:
+	case 176400:
+		/* Highest possible SYSCLK frequency: 135.4752 MHz */
+		priv->sysclk_rate = 135475200U;
+		break;
+	default:
+		dev_err(codec->dev, "Not supported sample rate: %d\n",
+			params_rate(params));
+		return -EINVAL;
+	}
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, ARIZONA_CLK_SYSCLK, 0, 0);
+	if (ret < 0) {
+		dev_err(codec_dai->dev, "Failed to set SYSCLK: %d\n", ret);
+		return ret;
+	}
+
+	return tm2_start_sysclk(rtd->card);
+}
+
+static struct snd_soc_ops tm2_aif1_ops = {
+	.hw_params = tm2_aif1_hw_params,
+};
+
+static int tm2_aif2_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 tm2_machine_priv *priv =	snd_soc_card_get_drvdata(rtd->card);
+	unsigned long mclk_rate = clk_get_rate(priv->codec_mclk1);
+	unsigned int asyncclk_rate;
+	int ret;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 12000:
+	case 16000:
+		/* Highest possible ASYNCCLK frequency: 49.152MHz */
+		asyncclk_rate = 49152000U;
+		break;
+	case 11025:
+		/* Highest possible ASYNCCLK frequency: 45.1584 MHz */
+		asyncclk_rate = 45158400U;
+		break;
+	default:
+		dev_err(codec->dev, "Not supported sample rate: %d\n",
+			params_rate(params));
+		return -EINVAL;
+	}
+
+	ret = snd_soc_codec_set_pll(codec, WM5110_FLL2,
+				    ARIZONA_FLL_SRC_MCLK1,
+				    mclk_rate,
+				    asyncclk_rate);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to start FLL: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_codec_set_pll(codec, WM5110_FLL2_REFCLK,
+				    ARIZONA_FLL_SRC_MCLK1,
+				    mclk_rate,
+				    asyncclk_rate);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set FLL1 Source: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, ARIZONA_CLK_ASYNCCLK, 0, 0);
+
+	if (ret < 0) {
+		dev_err(codec_dai->dev, "Failed to set ASYNCCLK: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_ASYNCCLK,
+				       ARIZONA_CLK_SRC_FLL2,
+				       asyncclk_rate,
+				       SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set ASYNCCLK Source: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops tm2_aif2_ops = {
+	.hw_params = tm2_aif2_hw_params,
+};
+
+static int tm2_mic_bias(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_card *card = w->dapm->card;
+	struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		gpiod_set_value_cansleep(priv->gpio_mic_bias,  1);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		gpiod_set_value_cansleep(priv->gpio_mic_bias,  0);
+		break;
+	}
+
+	return 0;
+}
+
+static int tm2_set_bias_level(struct snd_soc_card *card,
+				struct snd_soc_dapm_context *dapm,
+				enum snd_soc_bias_level level)
+{
+	struct snd_soc_pcm_runtime *rtd;
+
+	rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name);
+
+	if (dapm->dev != rtd->codec_dai->dev)
+		return 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_STANDBY:
+		if (card->dapm.bias_level == SND_SOC_BIAS_OFF)
+			tm2_start_sysclk(card);
+		break;
+	case SND_SOC_BIAS_OFF:
+		tm2_stop_sysclk(card);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	default:
+		break;
+	}
+
+	card->dapm.bias_level = level;
+
+	return 0;
+}
+
+static struct snd_soc_aux_dev tm2_speaker_amp_dev;
+
+static int tm2_late_probe(struct snd_soc_card *card)
+{
+	struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai_link_component dlc = { 0 };
+	struct snd_soc_dai *amp_pdm_dai;
+	struct snd_soc_pcm_runtime *rtd;
+	unsigned int ch_map[] = { 0, 1 };
+	int ret;
+
+	rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name);
+	priv->codec = rtd->codec;
+
+	/* 32 kHz must be enabled for jack detection */
+	if (!IS_ERR(priv->codec_mclk2))
+		clk_prepare_enable(priv->codec_mclk2);
+
+	dlc.of_node = tm2_speaker_amp_dev.codec_of_node;
+	amp_pdm_dai = snd_soc_find_dai(&dlc);
+	if (!amp_pdm_dai)
+		return -ENODEV;
+
+	/* Set the MAX98504 V/I sense PDM Tx DAI channel mapping */
+	ret = snd_soc_dai_set_channel_map(amp_pdm_dai, ARRAY_SIZE(ch_map),
+					  ch_map, 0, NULL);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_tdm_slot(amp_pdm_dai, 0x3, 0x0, 2, 16);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static int tm2_suspend_post(struct snd_soc_card *card)
+{
+	return tm2_stop_sysclk(card);
+}
+
+static int tm2_resume_pre(struct snd_soc_card *card)
+{
+	return tm2_start_sysclk(card);
+}
+
+static const struct snd_kcontrol_new tm2_controls[] = {
+	SOC_DAPM_PIN_SWITCH("HP"),
+	SOC_DAPM_PIN_SWITCH("SPK"),
+	SOC_DAPM_PIN_SWITCH("RCV"),
+	SOC_DAPM_PIN_SWITCH("VPS"),
+	SOC_DAPM_PIN_SWITCH("HDMI"),
+
+	SOC_DAPM_PIN_SWITCH("Main Mic"),
+	SOC_DAPM_PIN_SWITCH("Sub Mic"),
+	SOC_DAPM_PIN_SWITCH("Third Mic"),
+
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+};
+
+const struct snd_soc_dapm_widget tm2_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("HP", NULL),
+	SND_SOC_DAPM_SPK("SPK", NULL),
+	SND_SOC_DAPM_SPK("RCV", NULL),
+	SND_SOC_DAPM_LINE("VPS", NULL),
+	SND_SOC_DAPM_LINE("HDMI", NULL),
+
+	SND_SOC_DAPM_MIC("Main Mic", tm2_mic_bias),
+	SND_SOC_DAPM_MIC("Sub Mic", NULL),
+	SND_SOC_DAPM_MIC("Third Mic", NULL),
+
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+};
+
+static const struct snd_soc_component_driver tm2_component = {
+	.name	= "tm2-audio",
+};
+
+static struct snd_soc_dai_driver tm2_ext_dai[] = {
+	{
+		.name = "Voice call",
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 4,
+			.rate_min = 8000,
+			.rate_max = 48000,
+			.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+					SNDRV_PCM_RATE_48000),
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 4,
+			.rate_min = 8000,
+			.rate_max = 48000,
+			.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+					SNDRV_PCM_RATE_48000),
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+	},
+	{
+		.name = "Bluetooth",
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 4,
+			.rate_min = 8000,
+			.rate_max = 16000,
+			.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000),
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rate_min = 8000,
+			.rate_max = 16000,
+			.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000),
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+	},
+};
+
+static struct snd_soc_dai_link tm2_dai_links[] = {
+	{
+		.name		= "WM5110 AIF1",
+		.stream_name	= "HiFi Primary",
+		.codec_dai_name = "wm5110-aif1",
+		.ops		= &tm2_aif1_ops,
+		.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM,
+	}, {
+		.name		= "WM5110 Voice",
+		.stream_name	= "Voice call",
+		.codec_dai_name = "wm5110-aif2",
+		.ops		= &tm2_aif2_ops,
+		.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM,
+		.ignore_suspend = 1,
+	}, {
+		.name		= "WM5110 BT",
+		.stream_name	= "Bluetooth",
+		.codec_dai_name = "wm5110-aif3",
+		.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM,
+		.ignore_suspend = 1,
+	}
+};
+
+static struct snd_soc_card tm2_card = {
+	.owner			= THIS_MODULE,
+
+	.dai_link		= tm2_dai_links,
+	.num_links		= ARRAY_SIZE(tm2_dai_links),
+	.controls		= tm2_controls,
+	.num_controls		= ARRAY_SIZE(tm2_controls),
+	.dapm_widgets		= tm2_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(tm2_dapm_widgets),
+
+	.aux_dev		= &tm2_speaker_amp_dev,
+	.num_aux_devs		= 1,
+
+	.late_probe		= tm2_late_probe,
+
+	.set_bias_level		= tm2_set_bias_level,
+
+	.suspend_post		= tm2_suspend_post,
+	.resume_pre		= tm2_resume_pre,
+};
+
+static int tm2_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct snd_soc_card *card = &tm2_card;
+	struct tm2_machine_priv *priv;
+	struct device_node *cpu_dai_node, *codec_dai_node;
+	int ret, i;
+
+	if (!dev->of_node) {
+		dev_err(dev, "DT node is missing\n");
+		return -ENODEV;
+	}
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	snd_soc_card_set_drvdata(card, priv);
+	card->dev = dev;
+
+	priv->gpio_mic_bias = devm_gpiod_get(dev, "mic-bias",
+						GPIOF_OUT_INIT_LOW);
+	if (IS_ERR(priv->gpio_mic_bias)) {
+		dev_err(dev, "Failed to get mic bias gpio\n");
+		return PTR_ERR(priv->gpio_mic_bias);
+	}
+
+	ret = snd_soc_of_parse_card_name(card, "model");
+	if (ret < 0) {
+		dev_err(dev, "Card name is not specified\n");
+		return ret;
+	}
+
+	ret = snd_soc_of_parse_audio_routing(card, "samsung,audio-routing");
+	if (ret < 0) {
+		dev_err(dev, "Audio routing is not specified or invalid\n");
+		return ret;
+	}
+
+	card->aux_dev[0].codec_of_node = of_parse_phandle(dev->of_node,
+							"audio-amplifier", 0);
+	if (!card->aux_dev[0].codec_of_node) {
+		dev_err(dev, "audio-amplifier property invalid or missing\n");
+		return -EINVAL;
+	}
+
+	cpu_dai_node = of_parse_phandle(dev->of_node, "i2s-controller", 0);
+	if (!cpu_dai_node) {
+		dev_err(dev, "i2s-controllers property invalid or missing\n");
+		ret = -EINVAL;
+		goto err_put_amp;
+	}
+
+	codec_dai_node = of_parse_phandle(dev->of_node, "audio-codec", 0);
+	if (!codec_dai_node) {
+		dev_err(dev, "audio-codec property invalid or missing\n");
+		ret = -EINVAL;
+		goto err_put_cpu_dai;
+	}
+
+	for (i = 0; i < card->num_links; i++) {
+		card->dai_link[i].cpu_dai_name = NULL;
+		card->dai_link[i].cpu_name = NULL;
+		card->dai_link[i].platform_name = NULL;
+		card->dai_link[i].codec_of_node = codec_dai_node;
+		card->dai_link[i].cpu_of_node = cpu_dai_node;
+		card->dai_link[i].platform_of_node = cpu_dai_node;
+	}
+
+	priv->codec_mclk1 = of_clk_get_by_name(codec_dai_node, "mclk1");
+	if (IS_ERR(priv->codec_mclk1)) {
+		dev_err(dev, "Failed to get mclk1 clock\n");
+		ret = PTR_ERR(priv->codec_mclk1);
+		goto err_put_codec_dai;
+	}
+
+	/* mclk2 is optional */
+	priv->codec_mclk2 = of_clk_get_by_name(codec_dai_node, "mclk2");
+	if (IS_ERR(priv->codec_mclk2))
+		dev_info(dev, "Not using mclk2 clock\n");
+
+	ret = devm_snd_soc_register_component(dev, &tm2_component,
+				tm2_ext_dai, ARRAY_SIZE(tm2_ext_dai));
+	if (ret < 0) {
+		dev_err(dev, "Failed to register component: %d\n", ret);
+		goto err_put_mclk;
+	}
+
+	ret = devm_snd_soc_register_card(dev, card);
+	if (ret < 0) {
+		dev_err(dev, "Failed to register card: %d\n", ret);
+		goto err_put_mclk;
+	}
+
+	return 0;
+
+err_put_mclk:
+	clk_put(priv->codec_mclk1);
+	if (!IS_ERR(priv->codec_mclk2))
+		clk_put(priv->codec_mclk2);
+err_put_codec_dai:
+	of_node_put(codec_dai_node);
+err_put_cpu_dai:
+	of_node_put(cpu_dai_node);
+err_put_amp:
+	of_node_put(card->aux_dev[0].codec_of_node);
+	return ret;
+}
+
+static int tm2_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &tm2_card;
+	struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card);
+
+	clk_put(priv->codec_mclk1);
+	if (!IS_ERR(priv->codec_mclk2))
+		clk_put(priv->codec_mclk2);
+
+	of_node_put(card->dai_link[0].codec_of_node);
+	of_node_put(card->dai_link[0].cpu_of_node);
+	of_node_put(card->aux_dev[0].codec_of_node);
+
+	return 0;
+}
+
+static const struct of_device_id tm2_of_match[] = {
+	{ .compatible = "samsung,tm2-audio" },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, tm2_of_match);
+
+static struct platform_driver tm2_driver = {
+	.driver = {
+		.name		= "tm2-audio",
+		.pm		= &snd_soc_pm_ops,
+		.of_match_table	= tm2_of_match,
+	},
+	.probe	= tm2_probe,
+	.remove	= tm2_remove,
+};
+
+module_platform_driver(tm2_driver);
+
+MODULE_AUTHOR("Inha Song <ideal.song@xxxxxxxxxxx>");
+MODULE_DESCRIPTION("ALSA SoC Exynos TM2 Audio Support");
+MODULE_LICENSE("GPL v2");
-- 
1.9.1