Re: [alsa-devel] [PATCH v5 2/2] ASoC: samsung: Add machine driver for Exynos5433 based TM2 board

Dear all,

I tested this patch with TM2 dt patches[1] based on v4.8-rc2. The playback is well working. [1] https://lkml.org/lkml/2016/8/16/61 : [PATCH 0/7] arm64: dts: Add the dts file for Exynos5433 and TM/TM2E board

Tested-by: Chanwoo Choi cw00.choi@samsung.com

Best Regards, Chanwoo Choi

On 2016년 08월 09일 23:21, Sylwester Nawrocki wrote:

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@samsung.com [k.kozlowski: rebased on 4.1] Signed-off-by: Krzysztof Kozlowski k.kozlowski@samsung.com [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@samsung.com

---

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 | 604 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 615 insertions(+) create mode 100644 sound/soc/samsung/tm2_wm5110.c

diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 7b722b0..1bed8a5 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -229,3 +229,12 @@ 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 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..16c48fb --- /dev/null +++ b/sound/soc/samsung/tm2_wm5110.c @@ -0,0 +1,604 @@ +/*

• • Copyright (C) 2015 - 2016 Samsung Electronics Co., Ltd.
• • Authors: Inha Song ideal.song@samsung.com

• •      Sylwester Nawrocki <s.nawrocki@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.
• */

+#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"

+#define TM2_DAI_AIF1 0 +#define TM2_DAI_AIF2 1

+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_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;
  

• }
• 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_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;
• 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_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;
• /* 32 kHz must be enabled for jack detection */
• if (!IS_ERR(priv->codec_mclk2))

• clk_prepare_enable(priv->codec_mclk2);
  

• 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;
• 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 int tm2_suspend_late(struct device *dev) +{

• struct snd_soc_card *card = dev_get_drvdata(dev);
• return tm2_stop_sysclk(card);

+}

+static int tm2_resume_early(struct device *dev) +{

• struct snd_soc_card *card = dev_get_drvdata(dev);
• return tm2_start_sysclk(card);

+}

+const struct dev_pm_ops tm2_pm_ops = {

• .suspend = snd_soc_suspend,
• .suspend_late = tm2_suspend_late,
• .resume = snd_soc_resume,
• .resume_early = tm2_resume_early,
• .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,
• .remove = tm2_remove,

+}; +module_platform_driver(tm2_driver);

+MODULE_AUTHOR("Inha Song ideal.song@samsung.com"); +MODULE_DESCRIPTION("ALSA SoC Exynos TM2 Audio Support");

+MODULE_LICENSE("GPL v2");

1.9.1

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