linux/sound/soc/intel/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 "../codecs/rt5670.h"
#include "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 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;

	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.
	 */
	snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK,
			       0, 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_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;

	/* 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;
	}

	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 */
	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[] = {
	/* 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",
		.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",
	},

	/* 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,
		.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,
		.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 = "cherrytrailcraudio",
	.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;

	/* 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 = {
		.owner = THIS_MODULE,
		.name = "cht-bsw-rt5672",
		.pm = &snd_soc_pm_ops,
	},
	.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");