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 v6:
- removed unused variables.
Changes since v5:
- dropped requesting and managing of the CODEC's clocks,
- removed driver remove() handler,
- changed pm_ops to use prepare/complete rather than
late_suspend/early_resume.
Changes since v4 (addressing review comments from Charles):
- changed the order of WM5110_FLL{1,2}, WM5110_FLL{1,2}_REFCLK setting,
- ARIZONA_CLK_SYSCLK, ARIZONA_CLK_ASYNCCLK setting moved to late_probe,
- added tm2_aif2_hw_free callback for disabling FLL2,
- removed unneded card->dapm.bias_level assignment in tm2_mic_bias callback,
- suspend_late, resume_early dev_pm_ops used instead of suspend_post,
resume_pre struct snd_soc_card callbacks.
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 | 552 +++++++++++++++++++++++++++++++++++++++++
3 files changed, 563 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..5cdf7d1
--- /dev/null
+++ b/sound/soc/samsung/tm2_wm5110.c
@@ -0,0 +1,552 @@
+/*
+ * 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"
+
+/*
+ * The source clock is XCLKOUT with its mux set to the external fixed rate
+ * oscillator (XXTI).
+ */
+#define MCLK_RATE 24000000U
+
+#define TM2_DAI_AIF1 0
+#define TM2_DAI_AIF2 1
+
+struct tm2_machine_priv {
+ struct snd_soc_codec *codec;
+ 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;
+ int 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_pll(codec, WM5110_FLL1,
+ ARIZONA_FLL_SRC_MCLK1,
+ MCLK_RATE,
+ priv->sysclk_rate);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to start FLL1: %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 FLL1: %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;
+ }
+
+ 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_codec *codec = rtd->codec;
+ struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+
+ 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;
+ }
+
+ 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_codec *codec = rtd->codec;
+ 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_REFCLK,
+ ARIZONA_FLL_SRC_MCLK1,
+ MCLK_RATE,
+ asyncclk_rate);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set FLL2 source: %d\n", ret);
+ return ret;
+ }
+
+ 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 FLL2: %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 int tm2_aif2_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->codec;
+ int ret;
+
+ /* disable FLL2 */
+ ret = snd_soc_codec_set_pll(codec, WM5110_FLL2, ARIZONA_FLL_SRC_MCLK1,
+ 0, 0);
+ if (ret < 0)
+ dev_err(codec->dev, "Failed to stop FLL2: %d\n", ret);
+
+ return ret;
+}
+
+static struct snd_soc_ops tm2_aif2_ops = {
+ .hw_params = tm2_aif2_hw_params,
+ .hw_free = tm2_aif2_hw_free,
+};
+
+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;
+ default:
+ break;
+ }
+
+ 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 };
+ unsigned int ch_map[] = { 0, 1 };
+ struct snd_soc_dai *amp_pdm_dai;
+ struct snd_soc_pcm_runtime *rtd;
+ struct snd_soc_dai *aif1_dai;
+ struct snd_soc_dai *aif2_dai;
+ int ret;
+
+ rtd = snd_soc_get_pcm_runtime(card, card->dai_link[TM2_DAI_AIF1].name);
+ aif1_dai = rtd->codec_dai;
+ priv->codec = rtd->codec;
+
+ ret = snd_soc_dai_set_sysclk(aif1_dai, ARIZONA_CLK_SYSCLK, 0, 0);
+ if (ret < 0) {
+ dev_err(aif1_dai->dev, "Failed to set SYSCLK: %d\n", ret);
+ return ret;
+ }
+
+ rtd = snd_soc_get_pcm_runtime(card, card->dai_link[TM2_DAI_AIF2].name);
+ aif2_dai = rtd->codec_dai;
+
+ ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0);
+ if (ret < 0) {
+ dev_err(aif2_dai->dev, "Failed to set ASYNCCLK: %d\n", ret);
+ return ret;
+ }
+
+ 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 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,
+};
+
+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;
+
+ 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 amp_node_put;
+ }
+
+ 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 cpu_dai_node_put;
+ }
+
+ 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;
+ }
+
+ 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 codec_dai_node_put;
+ }
+
+ ret = devm_snd_soc_register_card(dev, card);
+ if (ret < 0) {
+ dev_err(dev, "Failed to register card: %d\n", ret);
+ goto codec_dai_node_put;
+ }
+
+codec_dai_node_put:
+ of_node_put(codec_dai_node);
+cpu_dai_node_put:
+ of_node_put(cpu_dai_node);
+amp_node_put:
+ of_node_put(card->aux_dev[0].codec_of_node);
+ return ret;
+}
+
+static int tm2_pm_prepare(struct device *dev)
+{
+ struct snd_soc_card *card = dev_get_drvdata(dev);
+
+ return tm2_stop_sysclk(card);
+}
+
+static void tm2_pm_complete(struct device *dev)
+{
+ struct snd_soc_card *card = dev_get_drvdata(dev);
+
+ tm2_start_sysclk(card);
+}
+
+const struct dev_pm_ops tm2_pm_ops = {
+ .prepare = tm2_pm_prepare,
+ .suspend = snd_soc_suspend,
+ .resume = snd_soc_resume,
+ .complete = tm2_pm_complete,
+ .freeze = snd_soc_suspend,
+ .thaw = snd_soc_resume,
+ .poweroff = snd_soc_poweroff,
+ .restore = snd_soc_resume,
+};
+
+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 = &tm2_pm_ops,
+ .of_match_table = tm2_of_match,
+ },
+ .probe = tm2_probe,
+};
+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
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