[deliverable/linux.git] / sound / soc / intel / boards / cht_bsw_rt5672.c

/*
 *  cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms
 *                     Cherrytrail and Braswell, with RT5672 codec.
 *
 *  Copyright (C) 2014 Intel Corp
 *  Author: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
 *          Mengdong Lin <mengdong.lin@intel.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; 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/rt5670.h"
#include "../atom/sst-atom-controls.h"

/* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */
#define CHT_PLAT_CLK_3_HZ	19200000
#define CHT_CODEC_DAI	"rt5670-aif1"

static struct snd_soc_jack cht_bsw_headset;

/* Headset jack detection DAPM pins */
static struct snd_soc_jack_pin cht_bsw_headset_pins[] = {
	{
		.pin = "Headset Mic",
		.mask = SND_JACK_MICROPHONE,
	},
	{
		.pin = "Headphone",
		.mask = SND_JACK_HEADPHONE,
	},
};

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_ON(event)) {
		/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
		ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK,
				CHT_PLAT_CLK_3_HZ, 48000 * 512);
		if (ret < 0) {
			dev_err(card->dev, "can't set codec pll: %d\n", ret);
			return ret;
		}

		/* set codec sysclk source to PLL */
		ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1,
			48000 * 512, SND_SOC_CLOCK_IN);
		if (ret < 0) {
			dev_err(card->dev, "can't set codec sysclk: %d\n", ret);
			return ret;
		}
	} else {
		/* 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.
		 */
		snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK,
				       48000 * 512, SND_SOC_CLOCK_IN);
	}
	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_PRE_PMU |
			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, "SPOLP"},
	{"Ext Spk", NULL, "SPOLN"},
	{"Ext Spk", NULL, "SPORP"},
	{"Ext Spk", NULL, "SPORN"},
	{"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, RT5670_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;
	}

	/* set codec sysclk source to PLL */
	ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_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_dai *codec_dai = runtime->codec_dai;
	struct snd_soc_codec *codec = codec_dai->codec;

	/* 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;
	}

	/* Select codec ASRC clock source to track I2S1 clock, because codec
	 * is in slave mode and 100fs I2S format (BCLK = 100 * LRCLK) cannot
	 * be supported by RT5672. Otherwise, ASRC will be disabled and cause
	 * noise.
	 */
	rt5670_sel_asrc_clk_src(codec,
				RT5670_DA_STEREO_FILTER
				| RT5670_DA_MONO_L_FILTER
				| RT5670_DA_MONO_R_FILTER
				| RT5670_AD_STEREO_FILTER
				| RT5670_AD_MONO_L_FILTER
				| RT5670_AD_MONO_R_FILTER,
				RT5670_CLK_SEL_I2S1_ASRC);

        ret = snd_soc_card_jack_new(runtime->card, "Headset",
                SND_JACK_HEADSET | SND_JACK_BTN_0 |
                SND_JACK_BTN_1 | SND_JACK_BTN_2, &cht_bsw_headset,
                cht_bsw_headset_pins, ARRAY_SIZE(cht_bsw_headset_pins));
        if (ret)
                return ret;

	rt5670_set_jack_detect(codec, &cht_bsw_headset);
	return 0;
}

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 */
	params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
	return 0;
}

static int cht_aif1_startup(struct snd_pcm_substream *substream)
{
	return snd_pcm_hw_constraint_single(substream->runtime,
			SNDRV_PCM_HW_PARAM_RATE, 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[] = {
	/* Front End DAI links */
	[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",
		.nonatomic = true,
		.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",
	},

	/* Back End DAI links */
	{
		/* SSP2 - Codec */
		.name = "SSP2-Codec",
		.be_id = 1,
		.cpu_dai_name = "ssp2-port",
		.platform_name = "sst-mfld-platform",
		.no_pcm = 1,
		.nonatomic = true,
		.codec_dai_name = "rt5670-aif1",
		.codec_name = "i2c-10EC5670:00",
		.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,
		.dpcm_playback = 1,
		.dpcm_capture = 1,
		.ops = &cht_be_ssp2_ops,
	},
};

static int cht_suspend_pre(struct snd_soc_card *card)
{
	struct snd_soc_codec *codec;

	list_for_each_entry(codec, &card->codec_dev_list, card_list) {
		if (!strcmp(codec->component.name, "i2c-10EC5670:00")) {
			dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n");
			rt5670_jack_suspend(codec);
			break;
		}
	}
	return 0;
}

static int cht_resume_post(struct snd_soc_card *card)
{
	struct snd_soc_codec *codec;

	list_for_each_entry(codec, &card->codec_dev_list, card_list) {
		if (!strcmp(codec->component.name, "i2c-10EC5670:00")) {
			dev_dbg(codec->dev, "enabling jack detect for resume.\n");
			rt5670_jack_resume(codec);
			break;
		}
	}

	return 0;
}

/* SoC card */
static struct snd_soc_card snd_soc_card_cht = {
	.name = "cherrytrailcraudio",
	.owner = THIS_MODULE,
	.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),
	.suspend_pre = cht_suspend_pre,
	.resume_post = cht_resume_post,
};

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

	/* register the soc card */
	snd_soc_card_cht.dev = &pdev->dev;
	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-rt5672",
	},
	.probe = snd_cht_mc_probe,
};

module_platform_driver(snd_cht_mc_driver);

MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver");
MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:cht-bsw-rt5672");
Commit	Line	Data
026da220 ML	1	/*
	2	* cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms
	3	* Cherrytrail and Braswell, with RT5672 codec.
	4	*
	5	* Copyright (C) 2014 Intel Corp
	6	* Author: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
	7	* Mengdong Lin <mengdong.lin@intel.com>
	8	*
	9	* This program is free software; you can redistribute it and/or modify
	10	* it under the terms of the GNU General Public License as published by
	11	* the Free Software Foundation; version 2 of the License.
	12	*
	13	* This program is distributed in the hope that it will be useful, but
	14	* WITHOUT ANY WARRANTY; without even the implied warranty of
	15	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	16	* General Public License for more details.
	17	*/
	18
	19	#include <linux/module.h>
	20	#include <linux/platform_device.h>
	21	#include <linux/slab.h>
	22	#include <sound/pcm.h>
	23	#include <sound/pcm_params.h>
	24	#include <sound/soc.h>
6b3b58d9	25	#include <sound/jack.h>
e56c72d5	26	#include "../../codecs/rt5670.h"
b97169da	27	#include "../atom/sst-atom-controls.h"
026da220 ML	28
	29	/* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */
	30	#define CHT_PLAT_CLK_3_HZ 19200000
	31	#define CHT_CODEC_DAI "rt5670-aif1"
	32
6b3b58d9 JY	33	static struct snd_soc_jack cht_bsw_headset;
	34
	35	/* Headset jack detection DAPM pins */
	36	static struct snd_soc_jack_pin cht_bsw_headset_pins[] = {
	37	{
	38	.pin = "Headset Mic",
	39	.mask = SND_JACK_MICROPHONE,
	40	},
	41	{
	42	.pin = "Headphone",
	43	.mask = SND_JACK_HEADPHONE,
	44	},
	45	};
	46
026da220 ML	47	static inline struct snd_soc_dai cht_get_codec_dai(struct snd_soc_card card)
	48	{
	49	int i;
	50
	51	for (i = 0; i < card->num_rtd; i++) {
	52	struct snd_soc_pcm_runtime *rtd;
	53
	54	rtd = card->rtd + i;
	55	if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
	56	strlen(CHT_CODEC_DAI)))
	57	return rtd->codec_dai;
	58	}
	59	return NULL;
	60	}
	61
	62	static int platform_clock_control(struct snd_soc_dapm_widget *w,
	63	struct snd_kcontrol *k, int event)
	64	{
	65	struct snd_soc_dapm_context *dapm = w->dapm;
	66	struct snd_soc_card *card = dapm->card;
	67	struct snd_soc_dai *codec_dai;
8d0c38a3	68	int ret;
026da220 ML	69
	70	codec_dai = cht_get_codec_dai(card);
	71	if (!codec_dai) {
	72	dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
	73	return -EIO;
	74	}
	75
8d0c38a3 JY	76	if (SND_SOC_DAPM_EVENT_ON(event)) {
	77	/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
	78	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK,
	79	CHT_PLAT_CLK_3_HZ, 48000 * 512);
	80	if (ret < 0) {
	81	dev_err(card->dev, "can't set codec pll: %d\n", ret);
	82	return ret;
	83	}
026da220	84
8d0c38a3 JY	85	/* set codec sysclk source to PLL */
	86	ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1,
	87	48000 * 512, SND_SOC_CLOCK_IN);
	88	if (ret < 0) {
	89	dev_err(card->dev, "can't set codec sysclk: %d\n", ret);
	90	return ret;
	91	}
	92	} else {
	93	/* Set codec sysclk source to its internal clock because codec
	94	* PLL will be off when idle and MCLK will also be off by ACPI
	95	* when codec is runtime suspended. Codec needs clock for jack
	96	* detection and button press.
	97	*/
	98	snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK,
	99	48000 * 512, SND_SOC_CLOCK_IN);
	100	}
026da220 ML	101	return 0;
	102	}
	103
	104	static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
	105	SND_SOC_DAPM_HP("Headphone", NULL),
	106	SND_SOC_DAPM_MIC("Headset Mic", NULL),
	107	SND_SOC_DAPM_MIC("Int Mic", NULL),
	108	SND_SOC_DAPM_SPK("Ext Spk", NULL),
	109	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
8d0c38a3 JY	110	platform_clock_control, SND_SOC_DAPM_PRE_PMU \|
8d0c38a3 JY	111	SND_SOC_DAPM_POST_PMD),
026da220 ML	112	};
	113
	114	static const struct snd_soc_dapm_route cht_audio_map[] = {
	115	{"IN1P", NULL, "Headset Mic"},
	116	{"IN1N", NULL, "Headset Mic"},
	117	{"DMIC L1", NULL, "Int Mic"},
	118	{"DMIC R1", NULL, "Int Mic"},
	119	{"Headphone", NULL, "HPOL"},
	120	{"Headphone", NULL, "HPOR"},
	121	{"Ext Spk", NULL, "SPOLP"},
	122	{"Ext Spk", NULL, "SPOLN"},
	123	{"Ext Spk", NULL, "SPORP"},
	124	{"Ext Spk", NULL, "SPORN"},
	125	{"AIF1 Playback", NULL, "ssp2 Tx"},
	126	{"ssp2 Tx", NULL, "codec_out0"},
	127	{"ssp2 Tx", NULL, "codec_out1"},
	128	{"codec_in0", NULL, "ssp2 Rx"},
	129	{"codec_in1", NULL, "ssp2 Rx"},
	130	{"ssp2 Rx", NULL, "AIF1 Capture"},
	131	{"Headphone", NULL, "Platform Clock"},
	132	{"Headset Mic", NULL, "Platform Clock"},
	133	{"Int Mic", NULL, "Platform Clock"},
	134	{"Ext Spk", NULL, "Platform Clock"},
	135	};
	136
	137	static const struct snd_kcontrol_new cht_mc_controls[] = {
	138	SOC_DAPM_PIN_SWITCH("Headphone"),
	139	SOC_DAPM_PIN_SWITCH("Headset Mic"),
	140	SOC_DAPM_PIN_SWITCH("Int Mic"),
	141	SOC_DAPM_PIN_SWITCH("Ext Spk"),
	142	};
	143
	144	static int cht_aif1_hw_params(struct snd_pcm_substream *substream,
	145	struct snd_pcm_hw_params *params)
	146	{
	147	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	148	struct snd_soc_dai *codec_dai = rtd->codec_dai;
	149	int ret;
	150
	151	/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
	152	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK,
	153	CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
	154	if (ret < 0) {
	155	dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
	156	return ret;
	157	}
	158
	159	/* set codec sysclk source to PLL */
	160	ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1,
	161	params_rate(params) * 512,
	162	SND_SOC_CLOCK_IN);
	163	if (ret < 0) {
	164	dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
	165	return ret;
	166	}
	167	return 0;
	168	}
	169
	170	static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
	171	{
	172	int ret;
	173	struct snd_soc_dai *codec_dai = runtime->codec_dai;
eb55fab9	174	struct snd_soc_codec *codec = codec_dai->codec;
026da220 ML	175
	176	/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
	177	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
	178	if (ret < 0) {
	179	dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
	180	return ret;
	181	}
	182
eb55fab9 ML	183	/* Select codec ASRC clock source to track I2S1 clock, because codec
	184	* is in slave mode and 100fs I2S format (BCLK = 100 * LRCLK) cannot
	185	* be supported by RT5672. Otherwise, ASRC will be disabled and cause
	186	* noise.
	187	*/
	188	rt5670_sel_asrc_clk_src(codec,
	189	RT5670_DA_STEREO_FILTER
	190	\| RT5670_DA_MONO_L_FILTER
	191	\| RT5670_DA_MONO_R_FILTER
	192	\| RT5670_AD_STEREO_FILTER
	193	\| RT5670_AD_MONO_L_FILTER
	194	\| RT5670_AD_MONO_R_FILTER,
	195	RT5670_CLK_SEL_I2S1_ASRC);
6b3b58d9 JY	196
	197	ret = snd_soc_card_jack_new(runtime->card, "Headset",
	198	SND_JACK_HEADSET \| SND_JACK_BTN_0 \|
	199	SND_JACK_BTN_1 \| SND_JACK_BTN_2, &cht_bsw_headset,
	200	cht_bsw_headset_pins, ARRAY_SIZE(cht_bsw_headset_pins));
	201	if (ret)
	202	return ret;
	203
	204	rt5670_set_jack_detect(codec, &cht_bsw_headset);
026da220 ML	205	return 0;
	206	}
	207
	208	static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
	209	struct snd_pcm_hw_params *params)
	210	{
	211	struct snd_interval *rate = hw_param_interval(params,
	212	SNDRV_PCM_HW_PARAM_RATE);
	213	struct snd_interval *channels = hw_param_interval(params,
	214	SNDRV_PCM_HW_PARAM_CHANNELS);
	215
	216	/* The DSP will covert the FE rate to 48k, stereo, 24bits */
	217	rate->min = rate->max = 48000;
	218	channels->min = channels->max = 2;
	219
	220	/* set SSP2 to 24-bit */
369a9f5f	221	params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
026da220 ML	222	return 0;
	223	}
	224
026da220 ML	225	static int cht_aif1_startup(struct snd_pcm_substream *substream)
026da220 ML	226	{
a5be88f6 LPC	227	return snd_pcm_hw_constraint_single(substream->runtime,
a5be88f6 LPC	228	SNDRV_PCM_HW_PARAM_RATE, 48000);
026da220 ML	229	}
	230
	231	static struct snd_soc_ops cht_aif1_ops = {
	232	.startup = cht_aif1_startup,
	233	};
	234
	235	static struct snd_soc_ops cht_be_ssp2_ops = {
	236	.hw_params = cht_aif1_hw_params,
	237	};
	238
	239	static struct snd_soc_dai_link cht_dailink[] = {
	240	/* Front End DAI links */
	241	[MERR_DPCM_AUDIO] = {
	242	.name = "Audio Port",
	243	.stream_name = "Audio",
	244	.cpu_dai_name = "media-cpu-dai",
	245	.codec_dai_name = "snd-soc-dummy-dai",
	246	.codec_name = "snd-soc-dummy",
	247	.platform_name = "sst-mfld-platform",
76ca1c2c	248	.nonatomic = true,
026da220 ML	249	.dynamic = 1,
	250	.dpcm_playback = 1,
	251	.dpcm_capture = 1,
	252	.ops = &cht_aif1_ops,
	253	},
	254	[MERR_DPCM_COMPR] = {
	255	.name = "Compressed Port",
	256	.stream_name = "Compress",
	257	.cpu_dai_name = "compress-cpu-dai",
	258	.codec_dai_name = "snd-soc-dummy-dai",
	259	.codec_name = "snd-soc-dummy",
	260	.platform_name = "sst-mfld-platform",
	261	},
	262
	263	/* Back End DAI links */
	264	{
	265	/* SSP2 - Codec */
	266	.name = "SSP2-Codec",
	267	.be_id = 1,
	268	.cpu_dai_name = "ssp2-port",
	269	.platform_name = "sst-mfld-platform",
	270	.no_pcm = 1,
76ca1c2c	271	.nonatomic = true,
026da220 ML	272	.codec_dai_name = "rt5670-aif1",
	273	.codec_name = "i2c-10EC5670:00",
	274	.dai_fmt = SND_SOC_DAIFMT_DSP_B \| SND_SOC_DAIFMT_IB_NF
	275	\| SND_SOC_DAIFMT_CBS_CFS,
	276	.init = cht_codec_init,
	277	.be_hw_params_fixup = cht_codec_fixup,
026da220 ML	278	.dpcm_playback = 1,
	279	.dpcm_capture = 1,
	280	.ops = &cht_be_ssp2_ops,
	281	},
	282	};
	283
066d7b87 JY	284	static int cht_suspend_pre(struct snd_soc_card *card)
	285	{
	286	struct snd_soc_codec *codec;
	287
	288	list_for_each_entry(codec, &card->codec_dev_list, card_list) {
	289	if (!strcmp(codec->component.name, "i2c-10EC5670:00")) {
	290	dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n");
	291	rt5670_jack_suspend(codec);
	292	break;
	293	}
	294	}
	295	return 0;
	296	}
	297
	298	static int cht_resume_post(struct snd_soc_card *card)
	299	{
	300	struct snd_soc_codec *codec;
	301
	302	list_for_each_entry(codec, &card->codec_dev_list, card_list) {
	303	if (!strcmp(codec->component.name, "i2c-10EC5670:00")) {
	304	dev_dbg(codec->dev, "enabling jack detect for resume.\n");
	305	rt5670_jack_resume(codec);
	306	break;
	307	}
	308	}
	309
	310	return 0;
	311	}
	312
026da220 ML	313	/* SoC card */
	314	static struct snd_soc_card snd_soc_card_cht = {
	315	.name = "cherrytrailcraudio",
54d8697f	316	.owner = THIS_MODULE,
026da220 ML	317	.dai_link = cht_dailink,
	318	.num_links = ARRAY_SIZE(cht_dailink),
	319	.dapm_widgets = cht_dapm_widgets,
	320	.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
	321	.dapm_routes = cht_audio_map,
	322	.num_dapm_routes = ARRAY_SIZE(cht_audio_map),
	323	.controls = cht_mc_controls,
	324	.num_controls = ARRAY_SIZE(cht_mc_controls),
066d7b87 JY	325	.suspend_pre = cht_suspend_pre,
066d7b87 JY	326	.resume_post = cht_resume_post,
026da220 ML	327	};
	328
	329	static int snd_cht_mc_probe(struct platform_device *pdev)
	330	{
	331	int ret_val = 0;
	332
	333	/* register the soc card */
	334	snd_soc_card_cht.dev = &pdev->dev;
	335	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
	336	if (ret_val) {
	337	dev_err(&pdev->dev,
	338	"snd_soc_register_card failed %d\n", ret_val);
	339	return ret_val;
	340	}
	341	platform_set_drvdata(pdev, &snd_soc_card_cht);
	342	return ret_val;
	343	}
	344
	345	static struct platform_driver snd_cht_mc_driver = {
	346	.driver = {
026da220	347	.name = "cht-bsw-rt5672",
026da220 ML	348	},
	349	.probe = snd_cht_mc_probe,
	350	};
	351
	352	module_platform_driver(snd_cht_mc_driver);
	353
	354	MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver");
	355	MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin");
	356	MODULE_LICENSE("GPL v2");
	357	MODULE_ALIAS("platform:cht-bsw-rt5672");