sound/soc/intel/cht_bsw_rt5645.c - kernel/msm-4.9 - Gitiles

 /*
  *  cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms
  *                     Cherrytrail and Braswell, with RT5645 codec.
  *
  *  Copyright (C) 2015 Intel Corp
  *  Author: Fang, Yang A <yang.a.fang@intel.com>
  *	        N,Harshapriya <harshapriya.n@intel.com>
  *  This file is modified from cht_bsw_rt5672.c
  *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  *
  *  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; version 2 of the License.
  *
  *  This program is distributed in the hope that it will be useful, but
  *  WITHOUT ANY WARRANTY; without even the implied warranty of
  *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  *  General Public License for more details.
  *
  * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  */

 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/jack.h>
 #include "../codecs/rt5645.h"
 #include "sst-atom-controls.h"

 #define CHT_PLAT_CLK_3_HZ	19200000
 #define CHT_CODEC_DAI	"rt5645-aif1"

 struct cht_mc_private {
 	struct snd_soc_jack hp_jack;
 	struct snd_soc_jack mic_jack;
 };

 static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
 {
 	int i;

 	for (i = 0; i < card->num_rtd; i++) {
 		struct snd_soc_pcm_runtime *rtd;

 		rtd = card->rtd + i;
 		if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
 			     strlen(CHT_CODEC_DAI)))
 			return rtd->codec_dai;
 	}
 	return NULL;
 }

 static int platform_clock_control(struct snd_soc_dapm_widget *w,
 		struct snd_kcontrol *k, int  event)
 {
 	struct snd_soc_dapm_context *dapm = w->dapm;
 	struct snd_soc_card *card = dapm->card;
 	struct snd_soc_dai *codec_dai;
 	int ret;

 	codec_dai = cht_get_codec_dai(card);
 	if (!codec_dai) {
 		dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
 		return -EIO;
 	}

 	if (!SND_SOC_DAPM_EVENT_OFF(event))
 		return 0;

 	/* Set codec sysclk source to its internal clock because codec PLL will
 	 * be off when idle and MCLK will also be off by ACPI when codec is
 	 * runtime suspended. Codec needs clock for jack detection and button
 	 * press.
 	 */
 	ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK,
 			0, SND_SOC_CLOCK_IN);
 	if (ret < 0) {
 		dev_err(card->dev, "can't set codec sysclk: %d\n", ret);
 		return ret;
 	}

 	return 0;
 }

 static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
 	SND_SOC_DAPM_HP("Headphone", NULL),
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
 	SND_SOC_DAPM_MIC("Int Mic", NULL),
 	SND_SOC_DAPM_SPK("Ext Spk", NULL),
 	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
 			platform_clock_control, SND_SOC_DAPM_POST_PMD),
 };

 static const struct snd_soc_dapm_route cht_audio_map[] = {
 	{"IN1P", NULL, "Headset Mic"},
 	{"IN1N", NULL, "Headset Mic"},
 	{"DMIC L1", NULL, "Int Mic"},
 	{"DMIC R1", NULL, "Int Mic"},
 	{"Headphone", NULL, "HPOL"},
 	{"Headphone", NULL, "HPOR"},
 	{"Ext Spk", NULL, "SPOL"},
 	{"Ext Spk", NULL, "SPOR"},
 	{"AIF1 Playback", NULL, "ssp2 Tx"},
 	{"ssp2 Tx", NULL, "codec_out0"},
 	{"ssp2 Tx", NULL, "codec_out1"},
 	{"codec_in0", NULL, "ssp2 Rx" },
 	{"codec_in1", NULL, "ssp2 Rx" },
 	{"ssp2 Rx", NULL, "AIF1 Capture"},
 	{"Headphone", NULL, "Platform Clock"},
 	{"Headset Mic", NULL, "Platform Clock"},
 	{"Int Mic", NULL, "Platform Clock"},
 	{"Ext Spk", NULL, "Platform Clock"},
 };

 static const struct snd_kcontrol_new cht_mc_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Headphone"),
 	SOC_DAPM_PIN_SWITCH("Headset Mic"),
 	SOC_DAPM_PIN_SWITCH("Int Mic"),
 	SOC_DAPM_PIN_SWITCH("Ext Spk"),
 };

 static int cht_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;
 	int ret;

 	/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
 	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
 				  CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
 	if (ret < 0) {
 		dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
 		return ret;
 	}

 	ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1,
 				params_rate(params) * 512, SND_SOC_CLOCK_IN);
 	if (ret < 0) {
 		dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
 		return ret;
 	}

 	return 0;
 }

 static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
 {
 	int ret;
 	struct snd_soc_codec *codec = runtime->codec;
 	struct snd_soc_dai *codec_dai = runtime->codec_dai;
 	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);

 	/* Select clk_i2s1_asrc as ASRC clock source */
 	rt5645_sel_asrc_clk_src(codec,
 				RT5645_DA_STEREO_FILTER |
 				RT5645_DA_MONO_L_FILTER |
 				RT5645_DA_MONO_R_FILTER |
 				RT5645_AD_STEREO_FILTER,
 				RT5645_CLK_SEL_I2S1_ASRC);

 	/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
 	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
 	if (ret < 0) {
 		dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
 		return ret;
 	}

 	ret = snd_soc_jack_new(codec, "Headphone Jack",
 				SND_JACK_HEADPHONE,
 				&ctx->hp_jack);
 	if (ret) {
 		dev_err(runtime->dev, "HP jack creation failed %d\n", ret);
 		return ret;
 	}

 	ret = snd_soc_jack_new(codec, "Mic Jack",
 				SND_JACK_MICROPHONE,
 				&ctx->mic_jack);
 	if (ret) {
 		dev_err(runtime->dev, "Mic jack creation failed %d\n", ret);
 		return ret;
 	}

 	rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack);

 	return ret;
 }

 static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
 			    struct snd_pcm_hw_params *params)
 {
 	struct snd_interval *rate = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_RATE);
 	struct snd_interval *channels = hw_param_interval(params,
 						SNDRV_PCM_HW_PARAM_CHANNELS);

 	/* The DSP will covert the FE rate to 48k, stereo, 24bits */
 	rate->min = rate->max = 48000;
 	channels->min = channels->max = 2;

 	/* set SSP2 to 24-bit */
 	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
 				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
 				    SNDRV_PCM_FORMAT_S24_LE);
 	return 0;
 }

 static unsigned int rates_48000[] = {
 	48000,
 };

 static struct snd_pcm_hw_constraint_list constraints_48000 = {
 	.count = ARRAY_SIZE(rates_48000),
 	.list  = rates_48000,
 };

 static int cht_aif1_startup(struct snd_pcm_substream *substream)
 {
 	return snd_pcm_hw_constraint_list(substream->runtime, 0,
 			SNDRV_PCM_HW_PARAM_RATE,
 			&constraints_48000);
 }

 static struct snd_soc_ops cht_aif1_ops = {
 	.startup = cht_aif1_startup,
 };

 static struct snd_soc_ops cht_be_ssp2_ops = {
 	.hw_params = cht_aif1_hw_params,
 };

 static struct snd_soc_dai_link cht_dailink[] = {
 	[MERR_DPCM_AUDIO] = {
 		.name = "Audio Port",
 		.stream_name = "Audio",
 		.cpu_dai_name = "media-cpu-dai",
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.codec_name = "snd-soc-dummy",
 		.platform_name = "sst-mfld-platform",
 		.ignore_suspend = 1,
 		.dynamic = 1,
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 		.ops = &cht_aif1_ops,
 	},
 	[MERR_DPCM_COMPR] = {
 		.name = "Compressed Port",
 		.stream_name = "Compress",
 		.cpu_dai_name = "compress-cpu-dai",
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.codec_name = "snd-soc-dummy",
 		.platform_name = "sst-mfld-platform",
 	},
 	/* CODEC<->CODEC link */
 	/* back ends */
 	{
 		.name = "SSP2-Codec",
 		.be_id = 1,
 		.cpu_dai_name = "ssp2-port",
 		.platform_name = "sst-mfld-platform",
 		.no_pcm = 1,
 		.codec_dai_name = "rt5645-aif1",
 		.codec_name = "i2c-10EC5645:00",
 		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
 					| SND_SOC_DAIFMT_CBS_CFS,
 		.init = cht_codec_init,
 		.be_hw_params_fixup = cht_codec_fixup,
 		.ignore_suspend = 1,
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 		.ops = &cht_be_ssp2_ops,
 	},
 };

 /* SoC card */
 static struct snd_soc_card snd_soc_card_cht = {
 	.name = "chtrt5645",
 	.dai_link = cht_dailink,
 	.num_links = ARRAY_SIZE(cht_dailink),
 	.dapm_widgets = cht_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
 	.dapm_routes = cht_audio_map,
 	.num_dapm_routes = ARRAY_SIZE(cht_audio_map),
 	.controls = cht_mc_controls,
 	.num_controls = ARRAY_SIZE(cht_mc_controls),
 };

 static int snd_cht_mc_probe(struct platform_device *pdev)
 {
 	int ret_val = 0;
 	struct cht_mc_private *drv;

 	drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC);
 	if (!drv)
 		return -ENOMEM;

 	snd_soc_card_cht.dev = &pdev->dev;
 	snd_soc_card_set_drvdata(&snd_soc_card_cht, drv);
 	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
 	if (ret_val) {
 		dev_err(&pdev->dev,
 			"snd_soc_register_card failed %d\n", ret_val);
 		return ret_val;
 	}
 	platform_set_drvdata(pdev, &snd_soc_card_cht);
 	return ret_val;
 }

 static struct platform_driver snd_cht_mc_driver = {
 	.driver = {
 		.name = "cht-bsw-rt5645",
 		.pm = &snd_soc_pm_ops,
 	},
 	.probe = snd_cht_mc_probe,
 };

 module_platform_driver(snd_cht_mc_driver)

 MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver");
 MODULE_AUTHOR("Fang, Yang A,N,Harshapriya");
 MODULE_LICENSE("GPL v2");
 MODULE_ALIAS("platform:cht-bsw-rt5645");
	/*
	* cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms
	* Cherrytrail and Braswell, with RT5645 codec.
	*
	* Copyright (C) 2015 Intel Corp
	* Author: Fang, Yang A <yang.a.fang@intel.com>
	* N,Harshapriya <harshapriya.n@intel.com>
	* This file is modified from cht_bsw_rt5672.c
	* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
	*
	* 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; version 2 of the License.
	*
	* This program is distributed in the hope that it will be useful, but
	* WITHOUT ANY WARRANTY; without even the implied warranty of
	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	* General Public License for more details.
	*
	* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
	*/

	#include <linux/module.h>
	#include <linux/platform_device.h>
	#include <linux/slab.h>
	#include <sound/pcm.h>
	#include <sound/pcm_params.h>
	#include <sound/soc.h>
	#include <sound/jack.h>
	#include "../codecs/rt5645.h"
	#include "sst-atom-controls.h"

	#define CHT_PLAT_CLK_3_HZ 19200000
	#define CHT_CODEC_DAI "rt5645-aif1"

	struct cht_mc_private {
	struct snd_soc_jack hp_jack;
	struct snd_soc_jack mic_jack;
	};

	static inline struct snd_soc_dai cht_get_codec_dai(struct snd_soc_card card)
	{
	int i;

	for (i = 0; i < card->num_rtd; i++) {
	struct snd_soc_pcm_runtime *rtd;

	rtd = card->rtd + i;
	if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
	strlen(CHT_CODEC_DAI)))
	return rtd->codec_dai;
	}
	return NULL;
	}

	static int platform_clock_control(struct snd_soc_dapm_widget *w,
	struct snd_kcontrol *k, int event)
	{
	struct snd_soc_dapm_context *dapm = w->dapm;
	struct snd_soc_card *card = dapm->card;
	struct snd_soc_dai *codec_dai;
	int ret;

	codec_dai = cht_get_codec_dai(card);
	if (!codec_dai) {
	dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
	return -EIO;
	}

	if (!SND_SOC_DAPM_EVENT_OFF(event))
	return 0;

	/* Set codec sysclk source to its internal clock because codec PLL will
	* be off when idle and MCLK will also be off by ACPI when codec is
	* runtime suspended. Codec needs clock for jack detection and button
	* press.
	*/
	ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK,
	0, SND_SOC_CLOCK_IN);
	if (ret < 0) {
	dev_err(card->dev, "can't set codec sysclk: %d\n", ret);
	return ret;
	}

	return 0;
	}

	static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
	SND_SOC_DAPM_HP("Headphone", NULL),
	SND_SOC_DAPM_MIC("Headset Mic", NULL),
	SND_SOC_DAPM_MIC("Int Mic", NULL),
	SND_SOC_DAPM_SPK("Ext Spk", NULL),
	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
	platform_clock_control, SND_SOC_DAPM_POST_PMD),
	};

	static const struct snd_soc_dapm_route cht_audio_map[] = {
	{"IN1P", NULL, "Headset Mic"},
	{"IN1N", NULL, "Headset Mic"},
	{"DMIC L1", NULL, "Int Mic"},
	{"DMIC R1", NULL, "Int Mic"},
	{"Headphone", NULL, "HPOL"},
	{"Headphone", NULL, "HPOR"},
	{"Ext Spk", NULL, "SPOL"},
	{"Ext Spk", NULL, "SPOR"},
	{"AIF1 Playback", NULL, "ssp2 Tx"},
	{"ssp2 Tx", NULL, "codec_out0"},
	{"ssp2 Tx", NULL, "codec_out1"},
	{"codec_in0", NULL, "ssp2 Rx" },
	{"codec_in1", NULL, "ssp2 Rx" },
	{"ssp2 Rx", NULL, "AIF1 Capture"},
	{"Headphone", NULL, "Platform Clock"},
	{"Headset Mic", NULL, "Platform Clock"},
	{"Int Mic", NULL, "Platform Clock"},
	{"Ext Spk", NULL, "Platform Clock"},
	};

	static const struct snd_kcontrol_new cht_mc_controls[] = {
	SOC_DAPM_PIN_SWITCH("Headphone"),
	SOC_DAPM_PIN_SWITCH("Headset Mic"),
	SOC_DAPM_PIN_SWITCH("Int Mic"),
	SOC_DAPM_PIN_SWITCH("Ext Spk"),
	};

	static int cht_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;
	int ret;

	/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
	CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
	if (ret < 0) {
	dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
	return ret;
	}

	ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1,
	params_rate(params) * 512, SND_SOC_CLOCK_IN);
	if (ret < 0) {
	dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
	return ret;
	}

	return 0;
	}

	static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
	{
	int ret;
	struct snd_soc_codec *codec = runtime->codec;
	struct snd_soc_dai *codec_dai = runtime->codec_dai;
	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);

	/* Select clk_i2s1_asrc as ASRC clock source */
	rt5645_sel_asrc_clk_src(codec,
	RT5645_DA_STEREO_FILTER \|
	RT5645_DA_MONO_L_FILTER \|
	RT5645_DA_MONO_R_FILTER \|
	RT5645_AD_STEREO_FILTER,
	RT5645_CLK_SEL_I2S1_ASRC);

	/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
	if (ret < 0) {
	dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
	return ret;
	}

	ret = snd_soc_jack_new(codec, "Headphone Jack",
	SND_JACK_HEADPHONE,
	&ctx->hp_jack);
	if (ret) {
	dev_err(runtime->dev, "HP jack creation failed %d\n", ret);
	return ret;
	}

	ret = snd_soc_jack_new(codec, "Mic Jack",
	SND_JACK_MICROPHONE,
	&ctx->mic_jack);
	if (ret) {
	dev_err(runtime->dev, "Mic jack creation failed %d\n", ret);
	return ret;
	}

	rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack);

	return ret;
	}

	static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
	struct snd_pcm_hw_params *params)
	{
	struct snd_interval *rate = hw_param_interval(params,
	SNDRV_PCM_HW_PARAM_RATE);
	struct snd_interval *channels = hw_param_interval(params,
	SNDRV_PCM_HW_PARAM_CHANNELS);

	/* The DSP will covert the FE rate to 48k, stereo, 24bits */
	rate->min = rate->max = 48000;
	channels->min = channels->max = 2;

	/* set SSP2 to 24-bit */
	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
	SNDRV_PCM_HW_PARAM_FIRST_MASK],
	SNDRV_PCM_FORMAT_S24_LE);
	return 0;
	}

	static unsigned int rates_48000[] = {
	48000,
	};

	static struct snd_pcm_hw_constraint_list constraints_48000 = {
	.count = ARRAY_SIZE(rates_48000),
	.list = rates_48000,
	};

	static int cht_aif1_startup(struct snd_pcm_substream *substream)
	{
	return snd_pcm_hw_constraint_list(substream->runtime, 0,
	SNDRV_PCM_HW_PARAM_RATE,
	&constraints_48000);
	}

	static struct snd_soc_ops cht_aif1_ops = {
	.startup = cht_aif1_startup,
	};

	static struct snd_soc_ops cht_be_ssp2_ops = {
	.hw_params = cht_aif1_hw_params,
	};

	static struct snd_soc_dai_link cht_dailink[] = {
	[MERR_DPCM_AUDIO] = {
	.name = "Audio Port",
	.stream_name = "Audio",
	.cpu_dai_name = "media-cpu-dai",
	.codec_dai_name = "snd-soc-dummy-dai",
	.codec_name = "snd-soc-dummy",
	.platform_name = "sst-mfld-platform",
	.ignore_suspend = 1,
	.dynamic = 1,
	.dpcm_playback = 1,
	.dpcm_capture = 1,
	.ops = &cht_aif1_ops,
	},
	[MERR_DPCM_COMPR] = {
	.name = "Compressed Port",
	.stream_name = "Compress",
	.cpu_dai_name = "compress-cpu-dai",
	.codec_dai_name = "snd-soc-dummy-dai",
	.codec_name = "snd-soc-dummy",
	.platform_name = "sst-mfld-platform",
	},
	/* CODEC<->CODEC link */
	/* back ends */
	{
	.name = "SSP2-Codec",
	.be_id = 1,
	.cpu_dai_name = "ssp2-port",
	.platform_name = "sst-mfld-platform",
	.no_pcm = 1,
	.codec_dai_name = "rt5645-aif1",
	.codec_name = "i2c-10EC5645:00",
	.dai_fmt = SND_SOC_DAIFMT_DSP_B \| SND_SOC_DAIFMT_IB_NF
	\| SND_SOC_DAIFMT_CBS_CFS,
	.init = cht_codec_init,
	.be_hw_params_fixup = cht_codec_fixup,
	.ignore_suspend = 1,
	.dpcm_playback = 1,
	.dpcm_capture = 1,
	.ops = &cht_be_ssp2_ops,
	},
	};

	/* SoC card */
	static struct snd_soc_card snd_soc_card_cht = {
	.name = "chtrt5645",
	.dai_link = cht_dailink,
	.num_links = ARRAY_SIZE(cht_dailink),
	.dapm_widgets = cht_dapm_widgets,
	.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
	.dapm_routes = cht_audio_map,
	.num_dapm_routes = ARRAY_SIZE(cht_audio_map),
	.controls = cht_mc_controls,
	.num_controls = ARRAY_SIZE(cht_mc_controls),
	};

	static int snd_cht_mc_probe(struct platform_device *pdev)
	{
	int ret_val = 0;
	struct cht_mc_private *drv;

	drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC);
	if (!drv)
	return -ENOMEM;

	snd_soc_card_cht.dev = &pdev->dev;
	snd_soc_card_set_drvdata(&snd_soc_card_cht, drv);
	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
	if (ret_val) {
	dev_err(&pdev->dev,
	"snd_soc_register_card failed %d\n", ret_val);
	return ret_val;
	}
	platform_set_drvdata(pdev, &snd_soc_card_cht);
	return ret_val;
	}

	static struct platform_driver snd_cht_mc_driver = {
	.driver = {
	.name = "cht-bsw-rt5645",
	.pm = &snd_soc_pm_ops,
	},
	.probe = snd_cht_mc_probe,
	};

	module_platform_driver(snd_cht_mc_driver)

	MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver");
	MODULE_AUTHOR("Fang, Yang A,N,Harshapriya");
	MODULE_LICENSE("GPL v2");
	MODULE_ALIAS("platform:cht-bsw-rt5645");