[alsa-devel] [PATCH v5 2/2] ASoC: samsung: Add machine driver for Exynos5433 based TM2 board
Chanwoo Choi
cw00.choi at samsung.com
Fri Aug 19 03:52:08 CEST 2016
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 at 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 at samsung.com>
> [k.kozlowski: rebased on 4.1]
> Signed-off-by: Krzysztof Kozlowski <k.kozlowski at 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 at 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 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.
> + */
> +
> +#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 at samsung.com>");
> +MODULE_DESCRIPTION("ALSA SoC Exynos TM2 Audio Support");
> +MODULE_LICENSE("GPL v2");
> --
> 1.9.1
>
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