[deliverable/linux.git] / sound / soc / fsl / fsl-asoc-card.c

/*
 * Freescale Generic ASoC Sound Card driver with ASRC
 *
 * Copyright (C) 2014 Freescale Semiconductor, Inc.
 *
 * Author: Nicolin Chen <nicoleotsuka@gmail.com>
 *
 * This file is licensed under the terms of the GNU General Public License
 * version 2. This program is licensed "as is" without any warranty of any
 * kind, whether express or implied.
 */

#include <linux/clk.h>
#include <linux/i2c.h>
#include <linux/module.h>
#include <linux/of_platform.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>

#include "fsl_esai.h"
#include "fsl_sai.h"
#include "imx-audmux.h"

#include "../codecs/sgtl5000.h"
#include "../codecs/wm8962.h"

#define RX 0
#define TX 1

/* Default DAI format without Master and Slave flag */
#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)

/**
 * CODEC private data
 *
 * @mclk_freq: Clock rate of MCLK
 * @mclk_id: MCLK (or main clock) id for set_sysclk()
 * @fll_id: FLL (or secordary clock) id for set_sysclk()
 * @pll_id: PLL id for set_pll()
 */
struct codec_priv {
	unsigned long mclk_freq;
	u32 mclk_id;
	u32 fll_id;
	u32 pll_id;
};

/**
 * CPU private data
 *
 * @sysclk_freq[2]: SYSCLK rates for set_sysclk()
 * @sysclk_dir[2]: SYSCLK directions for set_sysclk()
 * @sysclk_id[2]: SYSCLK ids for set_sysclk()
 * @slot_width: Slot width of each frame
 *
 * Note: [1] for tx and [0] for rx
 */
struct cpu_priv {
	unsigned long sysclk_freq[2];
	u32 sysclk_dir[2];
	u32 sysclk_id[2];
	u32 slot_width;
};

/**
 * Freescale Generic ASOC card private data
 *
 * @dai_link[3]: DAI link structure including normal one and DPCM link
 * @pdev: platform device pointer
 * @codec_priv: CODEC private data
 * @cpu_priv: CPU private data
 * @card: ASoC card structure
 * @sample_rate: Current sample rate
 * @sample_format: Current sample format
 * @asrc_rate: ASRC sample rate used by Back-Ends
 * @asrc_format: ASRC sample format used by Back-Ends
 * @dai_fmt: DAI format between CPU and CODEC
 * @name: Card name
 */

struct fsl_asoc_card_priv {
	struct snd_soc_dai_link dai_link[3];
	struct platform_device *pdev;
	struct codec_priv codec_priv;
	struct cpu_priv cpu_priv;
	struct snd_soc_card card;
	u32 sample_rate;
	u32 sample_format;
	u32 asrc_rate;
	u32 asrc_format;
	u32 dai_fmt;
	char name[32];
};

/**
 * This dapm route map exsits for DPCM link only.
 * The other routes shall go through Device Tree.
 */
static const struct snd_soc_dapm_route audio_map[] = {
	{"CPU-Playback",  NULL, "ASRC-Playback"},
	{"Playback",  NULL, "CPU-Playback"},
	{"ASRC-Capture",  NULL, "CPU-Capture"},
	{"CPU-Capture",  NULL, "Capture"},
};

/* Add all possible widgets into here without being redundant */
static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
	SND_SOC_DAPM_LINE("Line In Jack", NULL),
	SND_SOC_DAPM_HP("Headphone Jack", NULL),
	SND_SOC_DAPM_SPK("Ext Spk", NULL),
	SND_SOC_DAPM_MIC("Mic Jack", NULL),
	SND_SOC_DAPM_MIC("AMIC", NULL),
	SND_SOC_DAPM_MIC("DMIC", NULL),
};

static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
				   struct snd_pcm_hw_params *params)
{
	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
	struct cpu_priv *cpu_priv = &priv->cpu_priv;
	struct device *dev = rtd->card->dev;
	int ret;

	priv->sample_rate = params_rate(params);
	priv->sample_format = params_format(params);

	/*
	 * If codec-dai is DAI Master and all configurations are already in the
	 * set_bias_level(), bypass the remaining settings in hw_params().
	 * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
	 */
	if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM)
		return 0;

	/* Specific configurations of DAIs starts from here */
	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
				     cpu_priv->sysclk_freq[tx],
				     cpu_priv->sysclk_dir[tx]);
	if (ret) {
		dev_err(dev, "failed to set sysclk for cpu dai\n");
		return ret;
	}

	if (cpu_priv->slot_width) {
		ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2,
					       cpu_priv->slot_width);
		if (ret) {
			dev_err(dev, "failed to set TDM slot for cpu dai\n");
			return ret;
		}
	}

	return 0;
}

static struct snd_soc_ops fsl_asoc_card_ops = {
	.hw_params = fsl_asoc_card_hw_params,
};

static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
			      struct snd_pcm_hw_params *params)
{
	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
	struct snd_interval *rate;
	struct snd_mask *mask;

	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
	rate->max = rate->min = priv->asrc_rate;

	mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
	snd_mask_none(mask);
	snd_mask_set(mask, priv->asrc_format);

	return 0;
}

static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
	/* Default ASoC DAI Link*/
	{
		.name = "HiFi",
		.stream_name = "HiFi",
		.ops = &fsl_asoc_card_ops,
	},
	/* DPCM Link between Front-End and Back-End (Optional) */
	{
		.name = "HiFi-ASRC-FE",
		.stream_name = "HiFi-ASRC-FE",
		.codec_name = "snd-soc-dummy",
		.codec_dai_name = "snd-soc-dummy-dai",
		.dpcm_playback = 1,
		.dpcm_capture = 1,
		.dynamic = 1,
	},
	{
		.name = "HiFi-ASRC-BE",
		.stream_name = "HiFi-ASRC-BE",
		.platform_name = "snd-soc-dummy",
		.be_hw_params_fixup = be_hw_params_fixup,
		.ops = &fsl_asoc_card_ops,
		.dpcm_playback = 1,
		.dpcm_capture = 1,
		.no_pcm = 1,
	},
};

static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
					struct snd_soc_dapm_context *dapm,
					enum snd_soc_bias_level level)
{
	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
	struct codec_priv *codec_priv = &priv->codec_priv;
	struct device *dev = card->dev;
	unsigned int pll_out;
	int ret;

	if (dapm->dev != codec_dai->dev)
		return 0;

	switch (level) {
	case SND_SOC_BIAS_PREPARE:
		if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
			break;

		if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
			pll_out = priv->sample_rate * 384;
		else
			pll_out = priv->sample_rate * 256;

		ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
					  codec_priv->mclk_id,
					  codec_priv->mclk_freq, pll_out);
		if (ret) {
			dev_err(dev, "failed to start FLL: %d\n", ret);
			return ret;
		}

		ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
					     pll_out, SND_SOC_CLOCK_IN);
		if (ret) {
			dev_err(dev, "failed to set SYSCLK: %d\n", ret);
			return ret;
		}
		break;

	case SND_SOC_BIAS_STANDBY:
		if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
			break;

		ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
					     codec_priv->mclk_freq,
					     SND_SOC_CLOCK_IN);
		if (ret) {
			dev_err(dev, "failed to switch away from FLL: %d\n", ret);
			return ret;
		}

		ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
		if (ret) {
			dev_err(dev, "failed to stop FLL: %d\n", ret);
			return ret;
		}
		break;

	default:
		break;
	}

	return 0;
}

static int fsl_asoc_card_audmux_init(struct device_node *np,
				     struct fsl_asoc_card_priv *priv)
{
	struct device *dev = &priv->pdev->dev;
	u32 int_ptcr = 0, ext_ptcr = 0;
	int int_port, ext_port;
	int ret;

	ret = of_property_read_u32(np, "mux-int-port", &int_port);
	if (ret) {
		dev_err(dev, "mux-int-port missing or invalid\n");
		return ret;
	}
	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
	if (ret) {
		dev_err(dev, "mux-ext-port missing or invalid\n");
		return ret;
	}

	/*
	 * The port numbering in the hardware manual starts at 1, while
	 * the AUDMUX API expects it starts at 0.
	 */
	int_port--;
	ext_port--;

	/*
	 * Use asynchronous mode (6 wires) for all cases.
	 * If only 4 wires are needed, just set SSI into
	 * synchronous mode and enable 4 PADs in IOMUX.
	 */
	switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
	case SND_SOC_DAIFMT_CBM_CFM:
		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
			   IMX_AUDMUX_V2_PTCR_RFSDIR |
			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
			   IMX_AUDMUX_V2_PTCR_TFSDIR |
			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
		break;
	case SND_SOC_DAIFMT_CBM_CFS:
		int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
			   IMX_AUDMUX_V2_PTCR_RFSDIR |
			   IMX_AUDMUX_V2_PTCR_TFSDIR;
		break;
	case SND_SOC_DAIFMT_CBS_CFM:
		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
			   IMX_AUDMUX_V2_PTCR_RFSDIR |
			   IMX_AUDMUX_V2_PTCR_TFSDIR;
		ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
		break;
	case SND_SOC_DAIFMT_CBS_CFS:
		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
			   IMX_AUDMUX_V2_PTCR_RFSDIR |
			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
			   IMX_AUDMUX_V2_PTCR_TFSDIR |
			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
		break;
	default:
		return -EINVAL;
	}

	/* Asynchronous mode can not be set along with RCLKDIR */
	ret = imx_audmux_v2_configure_port(int_port, 0,
					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
	if (ret) {
		dev_err(dev, "audmux internal port setup failed\n");
		return ret;
	}

	ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
	if (ret) {
		dev_err(dev, "audmux internal port setup failed\n");
		return ret;
	}

	ret = imx_audmux_v2_configure_port(ext_port, 0,
					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
	if (ret) {
		dev_err(dev, "audmux external port setup failed\n");
		return ret;
	}

	ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
	if (ret) {
		dev_err(dev, "audmux external port setup failed\n");
		return ret;
	}

	return 0;
}

static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
{
	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
	struct codec_priv *codec_priv = &priv->codec_priv;
	struct device *dev = card->dev;
	int ret;

	ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
				     codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
	if (ret) {
		dev_err(dev, "failed to set sysclk in %s\n", __func__);
		return ret;
	}

	return 0;
}

static int fsl_asoc_card_probe(struct platform_device *pdev)
{
	struct device_node *cpu_np, *codec_np, *asrc_np;
	struct device_node *np = pdev->dev.of_node;
	struct platform_device *asrc_pdev = NULL;
	struct platform_device *cpu_pdev;
	struct fsl_asoc_card_priv *priv;
	struct i2c_client *codec_dev;
	struct clk *codec_clk;
	u32 width;
	int ret;

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

	cpu_np = of_parse_phandle(np, "audio-cpu", 0);
	/* Give a chance to old DT binding */
	if (!cpu_np)
		cpu_np = of_parse_phandle(np, "ssi-controller", 0);
	codec_np = of_parse_phandle(np, "audio-codec", 0);
	if (!cpu_np || !codec_np) {
		dev_err(&pdev->dev, "phandle missing or invalid\n");
		ret = -EINVAL;
		goto fail;
	}

	cpu_pdev = of_find_device_by_node(cpu_np);
	if (!cpu_pdev) {
		dev_err(&pdev->dev, "failed to find CPU DAI device\n");
		ret = -EINVAL;
		goto fail;
	}

	codec_dev = of_find_i2c_device_by_node(codec_np);
	if (!codec_dev) {
		dev_err(&pdev->dev, "failed to find codec platform device\n");
		ret = -EINVAL;
		goto fail;
	}

	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
	if (asrc_np)
		asrc_pdev = of_find_device_by_node(asrc_np);

	/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
	codec_clk = clk_get(&codec_dev->dev, NULL);
	if (!IS_ERR(codec_clk)) {
		priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
		clk_put(codec_clk);
	}

	/* Default sample rate and format, will be updated in hw_params() */
	priv->sample_rate = 44100;
	priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;

	/* Assign a default DAI format, and allow each card to overwrite it */
	priv->dai_fmt = DAI_FMT_BASE;

	/* Diversify the card configurations */
	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
		priv->card.set_bias_level = NULL;
		priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
		priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
		priv->cpu_priv.slot_width = 32;
		priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
	} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
		priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
		priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
		priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
		priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
		priv->codec_priv.pll_id = WM8962_FLL;
		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
	} else {
		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
		return -EINVAL;
	}

	/* Common settings for corresponding Freescale CPU DAI driver */
	if (strstr(cpu_np->name, "ssi")) {
		/* Only SSI needs to configure AUDMUX */
		ret = fsl_asoc_card_audmux_init(np, priv);
		if (ret) {
			dev_err(&pdev->dev, "failed to init audmux\n");
			goto asrc_fail;
		}
	} else if (strstr(cpu_np->name, "esai")) {
		priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
		priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
	} else if (strstr(cpu_np->name, "sai")) {
		priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
		priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
	}

	sprintf(priv->name, "%s-audio", codec_dev->name);

	/* Initialize sound card */
	priv->pdev = pdev;
	priv->card.dev = &pdev->dev;
	priv->card.name = priv->name;
	priv->card.dai_link = priv->dai_link;
	priv->card.dapm_routes = audio_map;
	priv->card.late_probe = fsl_asoc_card_late_probe;
	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
	priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
	priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);

	memcpy(priv->dai_link, fsl_asoc_card_dai,
	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));

	ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
	if (ret) {
		dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
		goto asrc_fail;
	}

	/* Normal DAI Link */
	priv->dai_link[0].cpu_of_node = cpu_np;
	priv->dai_link[0].codec_of_node = codec_np;
	priv->dai_link[0].codec_dai_name = codec_dev->name;
	priv->dai_link[0].platform_of_node = cpu_np;
	priv->dai_link[0].dai_fmt = priv->dai_fmt;
	priv->card.num_links = 1;

	if (asrc_pdev) {
		/* DPCM DAI Links only if ASRC exsits */
		priv->dai_link[1].cpu_of_node = asrc_np;
		priv->dai_link[1].platform_of_node = asrc_np;
		priv->dai_link[2].codec_dai_name = codec_dev->name;
		priv->dai_link[2].codec_of_node = codec_np;
		priv->dai_link[2].cpu_of_node = cpu_np;
		priv->dai_link[2].dai_fmt = priv->dai_fmt;
		priv->card.num_links = 3;

		ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
					   &priv->asrc_rate);
		if (ret) {
			dev_err(&pdev->dev, "failed to get output rate\n");
			ret = -EINVAL;
			goto asrc_fail;
		}

		ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
		if (ret) {
			dev_err(&pdev->dev, "failed to get output rate\n");
			ret = -EINVAL;
			goto asrc_fail;
		}

		if (width == 24)
			priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
		else
			priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
	}

	/* Finish card registering */
	platform_set_drvdata(pdev, priv);
	snd_soc_card_set_drvdata(&priv->card, priv);

	ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
	if (ret)
		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);

asrc_fail:
	of_node_put(asrc_np);
fail:
	of_node_put(codec_np);
	of_node_put(cpu_np);

	return ret;
}

static const struct of_device_id fsl_asoc_card_dt_ids[] = {
	{ .compatible = "fsl,imx-audio-cs42888", },
	{ .compatible = "fsl,imx-audio-sgtl5000", },
	{ .compatible = "fsl,imx-audio-wm8962", },
	{}
};

static struct platform_driver fsl_asoc_card_driver = {
	.probe = fsl_asoc_card_probe,
	.driver = {
		.name = "fsl-asoc-card",
		.pm = &snd_soc_pm_ops,
		.of_match_table = fsl_asoc_card_dt_ids,
	},
};
module_platform_driver(fsl_asoc_card_driver);

MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
MODULE_ALIAS("platform:fsl-asoc-card");
MODULE_LICENSE("GPL");
Commit	Line	Data
708b4351 NC	1	/*
	2	* Freescale Generic ASoC Sound Card driver with ASRC
	3	*
	4	* Copyright (C) 2014 Freescale Semiconductor, Inc.
	5	*
	6	* Author: Nicolin Chen <nicoleotsuka@gmail.com>
	7	*
	8	* This file is licensed under the terms of the GNU General Public License
	9	* version 2. This program is licensed "as is" without any warranty of any
	10	* kind, whether express or implied.
	11	*/
	12
	13	#include <linux/clk.h>
	14	#include <linux/i2c.h>
	15	#include <linux/module.h>
	16	#include <linux/of_platform.h>
	17	#include <sound/pcm_params.h>
	18	#include <sound/soc.h>
	19
	20	#include "fsl_esai.h"
	21	#include "fsl_sai.h"
	22	#include "imx-audmux.h"
	23
	24	#include "../codecs/sgtl5000.h"
	25	#include "../codecs/wm8962.h"
	26
	27	#define RX 0
	28	#define TX 1
	29
	30	/* Default DAI format without Master and Slave flag */
	31	#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S \| SND_SOC_DAIFMT_NB_NF)
	32
	33	/**
	34	* CODEC private data
	35	*
	36	* @mclk_freq: Clock rate of MCLK
	37	* @mclk_id: MCLK (or main clock) id for set_sysclk()
	38	* @fll_id: FLL (or secordary clock) id for set_sysclk()
	39	* @pll_id: PLL id for set_pll()
	40	*/
	41	struct codec_priv {
	42	unsigned long mclk_freq;
	43	u32 mclk_id;
	44	u32 fll_id;
	45	u32 pll_id;
	46	};
	47
	48	/**
	49	* CPU private data
	50	*
	51	* @sysclk_freq[2]: SYSCLK rates for set_sysclk()
	52	* @sysclk_dir[2]: SYSCLK directions for set_sysclk()
	53	* @sysclk_id[2]: SYSCLK ids for set_sysclk()
cb3fc1ff	54	* @slot_width: Slot width of each frame
708b4351 NC	55	*
	56	* Note: [1] for tx and [0] for rx
	57	*/
	58	struct cpu_priv {
	59	unsigned long sysclk_freq[2];
	60	u32 sysclk_dir[2];
	61	u32 sysclk_id[2];
cb3fc1ff	62	u32 slot_width;
708b4351 NC	63	};
	64
	65	/**
	66	* Freescale Generic ASOC card private data
	67	*
	68	* @dai_link[3]: DAI link structure including normal one and DPCM link
	69	* @pdev: platform device pointer
	70	* @codec_priv: CODEC private data
	71	* @cpu_priv: CPU private data
	72	* @card: ASoC card structure
	73	* @sample_rate: Current sample rate
	74	* @sample_format: Current sample format
	75	* @asrc_rate: ASRC sample rate used by Back-Ends
	76	* @asrc_format: ASRC sample format used by Back-Ends
	77	* @dai_fmt: DAI format between CPU and CODEC
	78	* @name: Card name
	79	*/
	80
	81	struct fsl_asoc_card_priv {
	82	struct snd_soc_dai_link dai_link[3];
	83	struct platform_device *pdev;
	84	struct codec_priv codec_priv;
	85	struct cpu_priv cpu_priv;
	86	struct snd_soc_card card;
	87	u32 sample_rate;
	88	u32 sample_format;
	89	u32 asrc_rate;
	90	u32 asrc_format;
	91	u32 dai_fmt;
	92	char name[32];
	93	};
	94
	95	/**
	96	* This dapm route map exsits for DPCM link only.
	97	* The other routes shall go through Device Tree.
	98	*/
	99	static const struct snd_soc_dapm_route audio_map[] = {
	100	{"CPU-Playback", NULL, "ASRC-Playback"},
	101	{"Playback", NULL, "CPU-Playback"},
	102	{"ASRC-Capture", NULL, "CPU-Capture"},
	103	{"CPU-Capture", NULL, "Capture"},
	104	};
	105
	106	/* Add all possible widgets into here without being redundant */
	107	static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
	108	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
	109	SND_SOC_DAPM_LINE("Line In Jack", NULL),
	110	SND_SOC_DAPM_HP("Headphone Jack", NULL),
	111	SND_SOC_DAPM_SPK("Ext Spk", NULL),
	112	SND_SOC_DAPM_MIC("Mic Jack", NULL),
	113	SND_SOC_DAPM_MIC("AMIC", NULL),
	114	SND_SOC_DAPM_MIC("DMIC", NULL),
	115	};
	116
	117	static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
	118	struct snd_pcm_hw_params *params)
	119	{
	120	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	121	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
	122	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
	123	struct cpu_priv *cpu_priv = &priv->cpu_priv;
	124	struct device *dev = rtd->card->dev;
	125	int ret;
	126
127	priv->sample_rate = params_rate(params);
128	priv->sample_format = params_format(params);
129
41282920 NC	130	/*
	131	* If codec-dai is DAI Master and all configurations are already in the
	132	* set_bias_level(), bypass the remaining settings in hw_params().
	133	* Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
	134	*/
	135	if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM)
708b4351 NC	136	return 0;
	137
	138	/* Specific configurations of DAIs starts from here */
	139	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
	140	cpu_priv->sysclk_freq[tx],
	141	cpu_priv->sysclk_dir[tx]);
	142	if (ret) {
	143	dev_err(dev, "failed to set sysclk for cpu dai\n");
	144	return ret;
	145	}
	146
cb3fc1ff NC	147	if (cpu_priv->slot_width) {
	148	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2,
	149	cpu_priv->slot_width);
	150	if (ret) {
	151	dev_err(dev, "failed to set TDM slot for cpu dai\n");
	152	return ret;
	153	}
	154	}
	155
708b4351 NC	156	return 0;
	157	}
	158
	159	static struct snd_soc_ops fsl_asoc_card_ops = {
	160	.hw_params = fsl_asoc_card_hw_params,
	161	};
	162
	163	static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
	164	struct snd_pcm_hw_params *params)
	165	{
	166	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
	167	struct snd_interval *rate;
	168	struct snd_mask *mask;
	169
	170	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
	171	rate->max = rate->min = priv->asrc_rate;
	172
	173	mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
	174	snd_mask_none(mask);
	175	snd_mask_set(mask, priv->asrc_format);
	176
	177	return 0;
	178	}
	179
	180	static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
	181	/* Default ASoC DAI Link*/
	182	{
	183	.name = "HiFi",
	184	.stream_name = "HiFi",
	185	.ops = &fsl_asoc_card_ops,
	186	},
	187	/* DPCM Link between Front-End and Back-End (Optional) */
	188	{
	189	.name = "HiFi-ASRC-FE",
	190	.stream_name = "HiFi-ASRC-FE",
	191	.codec_name = "snd-soc-dummy",
	192	.codec_dai_name = "snd-soc-dummy-dai",
	193	.dpcm_playback = 1,
	194	.dpcm_capture = 1,
	195	.dynamic = 1,
	196	},
	197	{
	198	.name = "HiFi-ASRC-BE",
	199	.stream_name = "HiFi-ASRC-BE",
	200	.platform_name = "snd-soc-dummy",
	201	.be_hw_params_fixup = be_hw_params_fixup,
	202	.ops = &fsl_asoc_card_ops,
	203	.dpcm_playback = 1,
	204	.dpcm_capture = 1,
	205	.no_pcm = 1,
	206	},
	207	};
	208
	209	static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
	210	struct snd_soc_dapm_context *dapm,
	211	enum snd_soc_bias_level level)
	212	{
	213	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
	214	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
	215	struct codec_priv *codec_priv = &priv->codec_priv;
	216	struct device *dev = card->dev;
	217	unsigned int pll_out;
	218	int ret;
	219
220	if (dapm->dev != codec_dai->dev)
221	return 0;
222
223	switch (level) {
224	case SND_SOC_BIAS_PREPARE:
225	if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
226	break;
227
228	if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
229	pll_out = priv->sample_rate * 384;
230	else
231	pll_out = priv->sample_rate * 256;
232
233	ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
234	codec_priv->mclk_id,
235	codec_priv->mclk_freq, pll_out);
236	if (ret) {
237	dev_err(dev, "failed to start FLL: %d\n", ret);
238	return ret;
239	}
240
241	ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
242	pll_out, SND_SOC_CLOCK_IN);
243	if (ret) {
244	dev_err(dev, "failed to set SYSCLK: %d\n", ret);
245	return ret;
246	}
247	break;
248
249	case SND_SOC_BIAS_STANDBY:
250	if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
251	break;
252
253	ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
254	codec_priv->mclk_freq,
255	SND_SOC_CLOCK_IN);
256	if (ret) {
257	dev_err(dev, "failed to switch away from FLL: %d\n", ret);
258	return ret;
259	}
260
261	ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
262	if (ret) {
263	dev_err(dev, "failed to stop FLL: %d\n", ret);
264	return ret;
265	}
266	break;
267
268	default:
269	break;
270	}
271
272	return 0;
273	}
274
275	static int fsl_asoc_card_audmux_init(struct device_node *np,
276	struct fsl_asoc_card_priv *priv)
277	{
278	struct device *dev = &priv->pdev->dev;
279	u32 int_ptcr = 0, ext_ptcr = 0;
280	int int_port, ext_port;
281	int ret;
282
283	ret = of_property_read_u32(np, "mux-int-port", &int_port);
284	if (ret) {
285	dev_err(dev, "mux-int-port missing or invalid\n");
286	return ret;
287	}
288	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
289	if (ret) {
290	dev_err(dev, "mux-ext-port missing or invalid\n");
291	return ret;
292	}
293
294	/*
295	* The port numbering in the hardware manual starts at 1, while
296	* the AUDMUX API expects it starts at 0.
297	*/
298	int_port--;
299	ext_port--;
300
301	/*
302	* Use asynchronous mode (6 wires) for all cases.
303	* If only 4 wires are needed, just set SSI into
304	* synchronous mode and enable 4 PADs in IOMUX.
305	*/
306	switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
307	case SND_SOC_DAIFMT_CBM_CFM:
308	int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 \| ext_port) \|
309	IMX_AUDMUX_V2_PTCR_RCSEL(8 \| ext_port) \|
310	IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) \|
311	IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) \|
312	IMX_AUDMUX_V2_PTCR_RFSDIR \|
313	IMX_AUDMUX_V2_PTCR_RCLKDIR \|
314	IMX_AUDMUX_V2_PTCR_TFSDIR \|
315	IMX_AUDMUX_V2_PTCR_TCLKDIR;
316	break;
317	case SND_SOC_DAIFMT_CBM_CFS:
318	int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 \| ext_port) \|
319	IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) \|
320	IMX_AUDMUX_V2_PTCR_RCLKDIR \|
321	IMX_AUDMUX_V2_PTCR_TCLKDIR;
322	ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 \| int_port) \|
323	IMX_AUDMUX_V2_PTCR_TFSEL(int_port) \|
324	IMX_AUDMUX_V2_PTCR_RFSDIR \|
325	IMX_AUDMUX_V2_PTCR_TFSDIR;
326	break;
327	case SND_SOC_DAIFMT_CBS_CFM:
328	int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 \| ext_port) \|
329	IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) \|
330	IMX_AUDMUX_V2_PTCR_RFSDIR \|
331	IMX_AUDMUX_V2_PTCR_TFSDIR;
332	ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 \| int_port) \|
333	IMX_AUDMUX_V2_PTCR_TCSEL(int_port) \|
334	IMX_AUDMUX_V2_PTCR_RCLKDIR \|
335	IMX_AUDMUX_V2_PTCR_TCLKDIR;
336	break;
337	case SND_SOC_DAIFMT_CBS_CFS:
338	ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 \| int_port) \|
339	IMX_AUDMUX_V2_PTCR_RCSEL(8 \| int_port) \|
340	IMX_AUDMUX_V2_PTCR_TFSEL(int_port) \|
341	IMX_AUDMUX_V2_PTCR_TCSEL(int_port) \|
342	IMX_AUDMUX_V2_PTCR_RFSDIR \|
343	IMX_AUDMUX_V2_PTCR_RCLKDIR \|
344	IMX_AUDMUX_V2_PTCR_TFSDIR \|
345	IMX_AUDMUX_V2_PTCR_TCLKDIR;
346	break;
347	default:
348	return -EINVAL;
349	}
350
351	/* Asynchronous mode can not be set along with RCLKDIR */
352	ret = imx_audmux_v2_configure_port(int_port, 0,
353	IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
354	if (ret) {
355	dev_err(dev, "audmux internal port setup failed\n");
356	return ret;
357	}
358
359	ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
360	IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
361	if (ret) {
362	dev_err(dev, "audmux internal port setup failed\n");
363	return ret;
364	}
365
366	ret = imx_audmux_v2_configure_port(ext_port, 0,
367	IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
368	if (ret) {
369	dev_err(dev, "audmux external port setup failed\n");
370	return ret;
371	}
372
373	ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
374	IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
375	if (ret) {
376	dev_err(dev, "audmux external port setup failed\n");
377	return ret;
378	}
379
380	return 0;
381	}
382
383	static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
384	{
385	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
386	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
387	struct codec_priv *codec_priv = &priv->codec_priv;
388	struct device *dev = card->dev;
389	int ret;
390
391	ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
392	codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
393	if (ret) {
394	dev_err(dev, "failed to set sysclk in %s\n", __func__);
395	return ret;
396	}
397
398	return 0;
399	}
400
401	static int fsl_asoc_card_probe(struct platform_device *pdev)
402	{
403	struct device_node cpu_np, codec_np, *asrc_np;
404	struct device_node *np = pdev->dev.of_node;
405	struct platform_device *asrc_pdev = NULL;
406	struct platform_device *cpu_pdev;
407	struct fsl_asoc_card_priv *priv;
408	struct i2c_client *codec_dev;
409	struct clk *codec_clk;
410	u32 width;
411	int ret;
412
413	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
414	if (!priv)
415	return -ENOMEM;
416
417	cpu_np = of_parse_phandle(np, "audio-cpu", 0);
418	/* Give a chance to old DT binding */
419	if (!cpu_np)
420	cpu_np = of_parse_phandle(np, "ssi-controller", 0);
421	codec_np = of_parse_phandle(np, "audio-codec", 0);
422	if (!cpu_np \|\| !codec_np) {
423	dev_err(&pdev->dev, "phandle missing or invalid\n");
424	ret = -EINVAL;
425	goto fail;
426	}
427
428	cpu_pdev = of_find_device_by_node(cpu_np);
429	if (!cpu_pdev) {
430	dev_err(&pdev->dev, "failed to find CPU DAI device\n");
431	ret = -EINVAL;
432	goto fail;
433	}
434
435	codec_dev = of_find_i2c_device_by_node(codec_np);
436	if (!codec_dev) {
437	dev_err(&pdev->dev, "failed to find codec platform device\n");
438	ret = -EINVAL;
439	goto fail;
440	}
441
442	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
443	if (asrc_np)
444	asrc_pdev = of_find_device_by_node(asrc_np);
445
446	/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
447	codec_clk = clk_get(&codec_dev->dev, NULL);
448	if (!IS_ERR(codec_clk)) {
449	priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
450	clk_put(codec_clk);
451	}
452
453	/* Default sample rate and format, will be updated in hw_params() */
454	priv->sample_rate = 44100;
455	priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
456
457	/* Assign a default DAI format, and allow each card to overwrite it */
458	priv->dai_fmt = DAI_FMT_BASE;
459
460	/* Diversify the card configurations */
461	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
462	priv->card.set_bias_level = NULL;
463	priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
464	priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
465	priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
466	priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
cb3fc1ff	467	priv->cpu_priv.slot_width = 32;
708b4351 NC	468	priv->dai_fmt \|= SND_SOC_DAIFMT_CBS_CFS;
	469	} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
	470	priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
	471	priv->dai_fmt \|= SND_SOC_DAIFMT_CBM_CFM;
	472	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
	473	priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
	474	priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
	475	priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
	476	priv->codec_priv.pll_id = WM8962_FLL;
	477	priv->dai_fmt \|= SND_SOC_DAIFMT_CBM_CFM;
	478	} else {
	479	dev_err(&pdev->dev, "unknown Device Tree compatible\n");
	480	return -EINVAL;
	481	}
	482
	483	/* Common settings for corresponding Freescale CPU DAI driver */
	484	if (strstr(cpu_np->name, "ssi")) {
	485	/* Only SSI needs to configure AUDMUX */
	486	ret = fsl_asoc_card_audmux_init(np, priv);
	487	if (ret) {
	488	dev_err(&pdev->dev, "failed to init audmux\n");
5f37671e	489	goto asrc_fail;
708b4351 NC	490	}
	491	} else if (strstr(cpu_np->name, "esai")) {
	492	priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
	493	priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
	494	} else if (strstr(cpu_np->name, "sai")) {
	495	priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
	496	priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
	497	}
	498
	499	sprintf(priv->name, "%s-audio", codec_dev->name);
	500
	501	/* Initialize sound card */
	502	priv->pdev = pdev;
	503	priv->card.dev = &pdev->dev;
	504	priv->card.name = priv->name;
	505	priv->card.dai_link = priv->dai_link;
	506	priv->card.dapm_routes = audio_map;
	507	priv->card.late_probe = fsl_asoc_card_late_probe;
	508	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
	509	priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
	510	priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
	511
	512	memcpy(priv->dai_link, fsl_asoc_card_dai,
	513	sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
	514
3185878a NC	515	ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
	516	if (ret) {
	517	dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
	518	goto asrc_fail;
	519	}
	520
708b4351 NC	521	/* Normal DAI Link */
	522	priv->dai_link[0].cpu_of_node = cpu_np;
	523	priv->dai_link[0].codec_of_node = codec_np;
	524	priv->dai_link[0].codec_dai_name = codec_dev->name;
	525	priv->dai_link[0].platform_of_node = cpu_np;
	526	priv->dai_link[0].dai_fmt = priv->dai_fmt;
	527	priv->card.num_links = 1;
	528
	529	if (asrc_pdev) {
	530	/* DPCM DAI Links only if ASRC exsits */
	531	priv->dai_link[1].cpu_of_node = asrc_np;
	532	priv->dai_link[1].platform_of_node = asrc_np;
	533	priv->dai_link[2].codec_dai_name = codec_dev->name;
	534	priv->dai_link[2].codec_of_node = codec_np;
	535	priv->dai_link[2].cpu_of_node = cpu_np;
	536	priv->dai_link[2].dai_fmt = priv->dai_fmt;
	537	priv->card.num_links = 3;
	538
	539	ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
	540	&priv->asrc_rate);
	541	if (ret) {
	542	dev_err(&pdev->dev, "failed to get output rate\n");
	543	ret = -EINVAL;
5f37671e	544	goto asrc_fail;
708b4351 NC	545	}
	546
	547	ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
	548	if (ret) {
	549	dev_err(&pdev->dev, "failed to get output rate\n");
	550	ret = -EINVAL;
5f37671e	551	goto asrc_fail;
708b4351 NC	552	}
	553
	554	if (width == 24)
	555	priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
	556	else
	557	priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
	558	}
	559
	560	/* Finish card registering */
	561	platform_set_drvdata(pdev, priv);
	562	snd_soc_card_set_drvdata(&priv->card, priv);
	563
	564	ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
	565	if (ret)
	566	dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
	567
5f37671e SW	568	asrc_fail:
5f37671e SW	569	of_node_put(asrc_np);
708b4351 NC	570	fail:
708b4351 NC	571	of_node_put(codec_np);
708b4351 NC	572	of_node_put(cpu_np);
	573
	574	return ret;
	575	}
	576
	577	static const struct of_device_id fsl_asoc_card_dt_ids[] = {
	578	{ .compatible = "fsl,imx-audio-cs42888", },
	579	{ .compatible = "fsl,imx-audio-sgtl5000", },
	580	{ .compatible = "fsl,imx-audio-wm8962", },
	581	{}
	582	};
	583
	584	static struct platform_driver fsl_asoc_card_driver = {
	585	.probe = fsl_asoc_card_probe,
	586	.driver = {
	587	.name = "fsl-asoc-card",
	588	.pm = &snd_soc_pm_ops,
	589	.of_match_table = fsl_asoc_card_dt_ids,
	590	},
	591	};
	592	module_platform_driver(fsl_asoc_card_driver);
	593
	594	MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
	595	MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
	596	MODULE_ALIAS("platform:fsl-asoc-card");
	597	MODULE_LICENSE("GPL");