[alsa-devel] [PATCH v7 RESEND 2/2] ASoC: samsung: Add machine driver for Exynos5433 based TM2 board
Sylwester Nawrocki
s.nawrocki at samsung.com
Wed Nov 2 17:05:45 CET 2016
This patch adds the sound machine driver for the TM2 and TM2E boards.
Speaker and headphone playback, Main Mic capture, Bluetooth, Voice
call and external accessory are supported.
Signed-off-by: Inha Song <ideal.song at samsung.com>
[k.kozlowski: rebased on 4.1]
Signed-off-by: Krzysztof Kozlowski <krzk at kernel.org>
[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 at samsung.com>
Reviewed-by: Charles Keepax <ckeepax at opensource.wolfsonmicro.com>
---
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 f6023b4..6b5b048 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -229,4 +229,13 @@ config SND_SOC_ARNDALE_RT5631_ALC5631
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.
+
endif #SND_SOC_SAMSUNG
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 5d03f5c..4444b9f 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -44,6 +44,7 @@ snd-soc-lowland-objs := lowland.o
snd-soc-littlemill-objs := littlemill.o
snd-soc-bells-objs := bells.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
@@ -69,3 +70,4 @@ obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o
obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o
obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o
obj-$(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 at samsung.com>
+ * Sylwester Nawrocki <s.nawrocki at samsung.com>
+ *
+ * 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 at samsung.com>");
+MODULE_DESCRIPTION("ALSA SoC Exynos TM2 Audio Support");
+MODULE_LICENSE("GPL v2");
--
1.9.1
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