sound/soc/fsl/imx-cs42888.c - linux-imx - Git at Google

 /*
  * Copyright (C) 2010-2016 Freescale Semiconductor, Inc. All Rights Reserved.
  */

 /*
  * The code contained herein is licensed under the GNU General Public
  * License. You may obtain a copy of the GNU General Public License
  * Version 2 or later at the following locations:
  *
  * http://www.opensource.org/licenses/gpl-license.html
  * http://www.gnu.org/copyleft/gpl.html
  */

 #include <linux/module.h>
 #include <linux/of.h>
 #include <linux/of_platform.h>
 #include <linux/slab.h>
 #include <linux/device.h>
 #include <linux/i2c.h>
 #include <linux/clk.h>
 #include <linux/delay.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/soc.h>
 #include <sound/initval.h>
 #include <sound/pcm_params.h>

 #include "fsl_esai.h"

 #define CODEC_CLK_EXTER_OSC   1
 #define CODEC_CLK_ESAI_HCKT   2
 #define SUPPORT_RATE_NUM    10

 struct imx_priv {
 	struct clk *codec_clk;
 	struct clk *esai_clk;
 	unsigned int mclk_freq;
 	unsigned int esai_freq;
 	struct platform_device *pdev;
 	struct platform_device *asrc_pdev;
 	u32 asrc_rate;
 	u32 asrc_format;
 	bool is_codec_master;
 	bool is_codec_rpmsg;
 	bool is_stream_in_use[2];
 	bool is_stream_tdm[2];
 };

 static struct imx_priv card_priv;

 static int imx_cs42888_surround_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 *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct imx_priv *priv = &card_priv;
 	struct device *dev = &priv->pdev->dev;
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	u32 channels = params_channels(params);
 	u32 max_tdm_rate;
 	u32 dai_format;
 	int ret = 0;

 	priv->is_stream_tdm[tx] = channels > 1 && channels % 2;
 	dai_format = SND_SOC_DAIFMT_NB_NF |
 		(priv->is_stream_tdm[tx] ? SND_SOC_DAIFMT_DSP_A :
 					SND_SOC_DAIFMT_LEFT_J);

 	priv->is_stream_in_use[tx] = true;

 	if (priv->is_stream_in_use[!tx] &&
 		(priv->is_stream_tdm[tx] != priv->is_stream_tdm[!tx])) {

 		dev_err(dev, "Don't support different fmt for tx & rx\n");
 		return -EINVAL;
 	}

 	priv->mclk_freq = clk_get_rate(priv->codec_clk);
 	priv->esai_freq = clk_get_rate(priv->esai_clk);

 	if (priv->is_codec_master) {
 		/* TDM is not supported by codec in master mode */
 		if (priv->is_stream_tdm[tx]) {
 			dev_err(dev, "%d channels are not supported in codec master mode\n",
 				channels);
 			return -EINVAL;
 		}
 		dai_format |= SND_SOC_DAIFMT_CBM_CFM;
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKT_EXTAL,
 				       priv->mclk_freq, SND_SOC_CLOCK_IN);
 		else
 			ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKR_EXTAL,
 				       priv->mclk_freq, SND_SOC_CLOCK_IN);
 		if (ret) {
 			dev_err(dev, "failed to set cpu sysclk: %d\n", ret);
 			return ret;
 		}

 		ret = snd_soc_dai_set_sysclk(codec_dai, 0,
 					priv->mclk_freq, SND_SOC_CLOCK_OUT);
 		if (ret) {
 			dev_err(dev, "failed to set codec sysclk: %d\n", ret);
 			return ret;
 		}

 	} else {
 		dai_format |= SND_SOC_DAIFMT_CBS_CFS;
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKT_EXTAL,
 				       priv->mclk_freq, SND_SOC_CLOCK_OUT);
 		else
 			ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKR_EXTAL,
 				       priv->mclk_freq, SND_SOC_CLOCK_OUT);
 		if (ret) {
 			dev_err(dev, "failed to set cpu sysclk: %d\n", ret);
 			return ret;
 		}

 		ret = snd_soc_dai_set_sysclk(codec_dai, 0,
 					priv->mclk_freq, SND_SOC_CLOCK_IN);
 		if (ret) {
 			dev_err(dev, "failed to set codec sysclk: %d\n", ret);
 			return ret;
 		}
 	}

 	/* set cpu DAI configuration */
 	ret = snd_soc_dai_set_fmt(cpu_dai, dai_format);
 	if (ret) {
 		dev_err(dev, "failed to set cpu dai fmt: %d\n", ret);
 		return ret;
 	}
 	/* set i.MX active slot mask */
 	if (priv->is_stream_tdm[tx]) {
 		/* 2 required by ESAI BCLK divisors, 8 slots, 32 width */
 		if (priv->is_codec_master)
 			max_tdm_rate = priv->mclk_freq / (8*32);
 		else
 			max_tdm_rate = priv->esai_freq / (2*8*32);
 		if (params_rate(params) > max_tdm_rate) {
 			dev_err(dev,
 				"maximum supported sampling rate for %d channels is %dKHz\n",
 				channels, max_tdm_rate / 1000);
 			return -EINVAL;
 		}

 		/*
 		 * Per datasheet, the codec expects 8 slots and 32 bits
 		 * for every slot in TDM mode.
 		 */
 		snd_soc_dai_set_tdm_slot(cpu_dai,
 					 BIT(channels) - 1, BIT(channels) - 1,
 					 8, 32);
 	} else
 		snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);

 	/* set codec DAI configuration */
 	ret = snd_soc_dai_set_fmt(codec_dai, dai_format);
 	if (ret) {
 		dev_err(dev, "failed to set codec dai fmt: %d\n", ret);
 		return ret;
 	}
 	return 0;
 }

 static int imx_cs42888_surround_hw_free(struct snd_pcm_substream *substream)
 {
 	struct imx_priv *priv = &card_priv;
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;

 	priv->is_stream_in_use[tx] = false;

 	return 0;
 }

 static int imx_cs42888_surround_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	static struct snd_pcm_hw_constraint_list constraint_rates;
 	struct imx_priv *priv = &card_priv;
 	struct device *dev = &priv->pdev->dev;
 	static u32 support_rates[SUPPORT_RATE_NUM];
 	int ret;

 	priv->mclk_freq = clk_get_rate(priv->codec_clk);

 	if (priv->mclk_freq % 12288000 == 0) {
 		support_rates[0] = 48000;
 		support_rates[1] = 96000;
 		support_rates[2] = 192000;
 		constraint_rates.list = support_rates;
 		constraint_rates.count = 3;

 		ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
 							&constraint_rates);
 		if (ret)
 			return ret;
 	} else
 		dev_warn(dev, "mclk may be not supported %d\n", priv->mclk_freq);

 	return 0;
 }

 static struct snd_soc_ops imx_cs42888_surround_ops = {
 	.startup = imx_cs42888_surround_startup,
 	.hw_params = imx_cs42888_surround_hw_params,
 	.hw_free = imx_cs42888_surround_hw_free,
 };

 /**
  * imx_cs42888_surround_startup() is to set constrain for hw parameter, but
  * backend use same runtime as frontend, for p2p backend need to use different
  * parameter, so backend can't use the startup.
  */
 static struct snd_soc_ops imx_cs42888_surround_ops_be = {
 	.hw_params = imx_cs42888_surround_hw_params,
 	.hw_free = imx_cs42888_surround_hw_free,
 };


 static const struct snd_soc_dapm_widget imx_cs42888_dapm_widgets[] = {
 	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
 	SND_SOC_DAPM_LINE("Line In Jack", NULL),
 };

 static const struct snd_soc_dapm_route audio_map[] = {
 	/* Line out jack */
 	{"Line Out Jack", NULL, "AOUT1L"},
 	{"Line Out Jack", NULL, "AOUT1R"},
 	{"Line Out Jack", NULL, "AOUT2L"},
 	{"Line Out Jack", NULL, "AOUT2R"},
 	{"Line Out Jack", NULL, "AOUT3L"},
 	{"Line Out Jack", NULL, "AOUT3R"},
 	{"Line Out Jack", NULL, "AOUT4L"},
 	{"Line Out Jack", NULL, "AOUT4R"},
 	{"AIN1L", NULL, "Line In Jack"},
 	{"AIN1R", NULL, "Line In Jack"},
 	{"AIN2L", NULL, "Line In Jack"},
 	{"AIN2R", NULL, "Line In Jack"},
 	{"Playback",  NULL, "CPU-Playback"},/* dai route for be and fe */
 	{"CPU-Capture",  NULL, "Capture"},
 	{"CPU-Playback",  NULL, "ASRC-Playback"},
 	{"ASRC-Capture",  NULL, "CPU-Capture"},
 };

 static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 				struct snd_pcm_hw_params *params) {

 	struct imx_priv *priv = &card_priv;
 	struct snd_interval *rate;
 	struct snd_mask *mask;

 	if (!priv->asrc_pdev)
 		return -EINVAL;

 	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 imx_cs42888_dai[] = {
 	{
 		.name = "HiFi",
 		.stream_name = "HiFi",
 		.codec_dai_name = "cs42888",
 		.ops = &imx_cs42888_surround_ops,
 		.ignore_pmdown_time = 1,
 	},
 	{
 		.name = "HiFi-ASRC-FE",
 		.stream_name = "HiFi-ASRC-FE",
 		.codec_name = "snd-soc-dummy",
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.dynamic = 1,
 		.ignore_pmdown_time = 1,
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 		.dpcm_merged_chan = 1,
 	},
 	{
 		.name = "HiFi-ASRC-BE",
 		.stream_name = "HiFi-ASRC-BE",
 		.codec_dai_name = "cs42888",
 		.platform_name = "snd-soc-dummy",
 		.no_pcm = 1,
 		.ignore_pmdown_time = 1,
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 		.ops = &imx_cs42888_surround_ops_be,
 		.be_hw_params_fixup = be_hw_params_fixup,
 	},
 };

 static struct snd_soc_card snd_soc_card_imx_cs42888 = {
 	.name = "cs42888-audio",
 	.dai_link = imx_cs42888_dai,
 	.dapm_widgets = imx_cs42888_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(imx_cs42888_dapm_widgets),
 	.dapm_routes = audio_map,
 	.num_dapm_routes = ARRAY_SIZE(audio_map),
 	.owner = THIS_MODULE,
 };

 /*
  * This function will register the snd_soc_pcm_link drivers.
  */
 static int imx_cs42888_probe(struct platform_device *pdev)
 {
 	struct device_node *esai_np, *codec_np;
 	struct device_node *asrc_np = NULL;
 	struct platform_device *esai_pdev;
 	struct platform_device *asrc_pdev = NULL;
 	struct imx_priv *priv = &card_priv;
 	int ret;
 	u32 width;

 	priv->pdev = pdev;
 	priv->asrc_pdev = NULL;

 	if (of_property_read_bool(pdev->dev.of_node, "codec-rpmsg"))
 		priv->is_codec_rpmsg = true;

 	esai_np = of_parse_phandle(pdev->dev.of_node, "esai-controller", 0);
 	codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
 	if (!esai_np || !codec_np) {
 		dev_err(&pdev->dev, "phandle missing or invalid\n");
 		ret = -EINVAL;
 		goto fail;
 	}

 	asrc_np = of_parse_phandle(pdev->dev.of_node, "asrc-controller", 0);
 	if (asrc_np) {
 		asrc_pdev = of_find_device_by_node(asrc_np);
 		priv->asrc_pdev = asrc_pdev;
 	}

 	esai_pdev = of_find_device_by_node(esai_np);
 	if (!esai_pdev) {
 		dev_err(&pdev->dev, "failed to find ESAI platform device\n");
 		ret = -EINVAL;
 		goto fail;
 	}

 	if (priv->is_codec_rpmsg) {
 		struct platform_device *codec_dev;

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

 		priv->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
 		if (IS_ERR(priv->codec_clk)) {
 			ret = PTR_ERR(priv->codec_clk);
 			dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret);
 			goto fail;
 		}

 	} else {
 		struct i2c_client *codec_dev;

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

 		priv->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
 		if (IS_ERR(priv->codec_clk)) {
 			ret = PTR_ERR(priv->codec_clk);
 			dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret);
 			goto fail;
 		}
 	}

 	if (priv->is_codec_rpmsg) {
 		imx_cs42888_dai[0].codec_name     = "rpmsg-audio-codec-cs42888";
 		imx_cs42888_dai[0].codec_dai_name = "cs42888";
 	} else {
 		imx_cs42888_dai[0].codec_of_node   = codec_np;
 	}

 	/*if there is no asrc controller, we only enable one device*/
 	if (!asrc_pdev) {
 		imx_cs42888_dai[0].cpu_dai_name    = dev_name(&esai_pdev->dev);
 		imx_cs42888_dai[0].platform_of_node = esai_np;
 		snd_soc_card_imx_cs42888.num_links = 1;
 		snd_soc_card_imx_cs42888.num_dapm_routes =
 			ARRAY_SIZE(audio_map) - 2;
 	} else {
 		imx_cs42888_dai[0].cpu_dai_name    = dev_name(&esai_pdev->dev);
 		imx_cs42888_dai[0].platform_of_node = esai_np;
 		imx_cs42888_dai[1].cpu_of_node    = asrc_np;
 		imx_cs42888_dai[1].platform_of_node   = asrc_np;
 		imx_cs42888_dai[2].cpu_dai_name    = dev_name(&esai_pdev->dev);
 		snd_soc_card_imx_cs42888.num_links = 3;

 		if (priv->is_codec_rpmsg) {
 			imx_cs42888_dai[2].codec_name     = "rpmsg-audio-codec-cs42888";
 			imx_cs42888_dai[2].codec_dai_name = "cs42888";
 		} else {
 			imx_cs42888_dai[2].codec_of_node   = codec_np;
 		}

 		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 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 fail;
 		}

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

 	priv->esai_clk = devm_clk_get(&esai_pdev->dev, "extal");
 	if (IS_ERR(priv->esai_clk)) {
 		ret = PTR_ERR(priv->esai_clk);
 		dev_err(&esai_pdev->dev, "failed to get cpu clk: %d\n", ret);
 		goto fail;
 	}

 	priv->is_codec_master = false;
 	if (of_property_read_bool(pdev->dev.of_node, "codec-master"))
 		priv->is_codec_master = true;

 	snd_soc_card_imx_cs42888.dev = &pdev->dev;

 	platform_set_drvdata(pdev, &snd_soc_card_imx_cs42888);

 	ret = snd_soc_register_card(&snd_soc_card_imx_cs42888);
 	if (ret)
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
 fail:
 	if (asrc_np)
 		of_node_put(asrc_np);
 	if (esai_np)
 		of_node_put(esai_np);
 	if (codec_np)
 		of_node_put(codec_np);
 	return ret;
 }

 static int imx_cs42888_remove(struct platform_device *pdev)
 {
 	snd_soc_unregister_card(&snd_soc_card_imx_cs42888);
 	return 0;
 }

 static const struct of_device_id imx_cs42888_dt_ids[] = {
 	{ .compatible = "fsl,imx-audio-cs42888", },
 	{ /* sentinel */ }
 };

 static struct platform_driver imx_cs42888_driver = {
 	.probe = imx_cs42888_probe,
 	.remove = imx_cs42888_remove,
 	.driver = {
 		.name = "imx-cs42888",
 		.pm = &snd_soc_pm_ops,
 		.of_match_table = imx_cs42888_dt_ids,
 	},
 };
 module_platform_driver(imx_cs42888_driver);

 MODULE_AUTHOR("Freescale Semiconductor, Inc.");
 MODULE_DESCRIPTION("ALSA SoC cs42888 Machine Layer Driver");
 MODULE_ALIAS("platform:imx-cs42888");
 MODULE_LICENSE("GPL");
	/*
	* Copyright (C) 2010-2016 Freescale Semiconductor, Inc. All Rights Reserved.
	*/

	/*
	* The code contained herein is licensed under the GNU General Public
	* License. You may obtain a copy of the GNU General Public License
	* Version 2 or later at the following locations:
	*
	* http://www.opensource.org/licenses/gpl-license.html
	* http://www.gnu.org/copyleft/gpl.html
	*/

	#include <linux/module.h>
	#include <linux/of.h>
	#include <linux/of_platform.h>
	#include <linux/slab.h>
	#include <linux/device.h>
	#include <linux/i2c.h>
	#include <linux/clk.h>
	#include <linux/delay.h>
	#include <sound/core.h>
	#include <sound/pcm.h>
	#include <sound/soc.h>
	#include <sound/initval.h>
	#include <sound/pcm_params.h>

	#include "fsl_esai.h"

	#define CODEC_CLK_EXTER_OSC 1
	#define CODEC_CLK_ESAI_HCKT 2
	#define SUPPORT_RATE_NUM 10

	struct imx_priv {
	struct clk *codec_clk;
	struct clk *esai_clk;
	unsigned int mclk_freq;
	unsigned int esai_freq;
	struct platform_device *pdev;
	struct platform_device *asrc_pdev;
	u32 asrc_rate;
	u32 asrc_format;
	bool is_codec_master;
	bool is_codec_rpmsg;
	bool is_stream_in_use[2];
	bool is_stream_tdm[2];
	};

	static struct imx_priv card_priv;

	static int imx_cs42888_surround_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 *cpu_dai = rtd->cpu_dai;
	struct snd_soc_dai *codec_dai = rtd->codec_dai;
	struct imx_priv *priv = &card_priv;
	struct device *dev = &priv->pdev->dev;
	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
	u32 channels = params_channels(params);
	u32 max_tdm_rate;
	u32 dai_format;
	int ret = 0;

	priv->is_stream_tdm[tx] = channels > 1 && channels % 2;
	dai_format = SND_SOC_DAIFMT_NB_NF \|
	(priv->is_stream_tdm[tx] ? SND_SOC_DAIFMT_DSP_A :
	SND_SOC_DAIFMT_LEFT_J);

	priv->is_stream_in_use[tx] = true;

	if (priv->is_stream_in_use[!tx] &&
	(priv->is_stream_tdm[tx] != priv->is_stream_tdm[!tx])) {

	dev_err(dev, "Don't support different fmt for tx & rx\n");
	return -EINVAL;
	}

	priv->mclk_freq = clk_get_rate(priv->codec_clk);
	priv->esai_freq = clk_get_rate(priv->esai_clk);

	if (priv->is_codec_master) {
	/* TDM is not supported by codec in master mode */
	if (priv->is_stream_tdm[tx]) {
	dev_err(dev, "%d channels are not supported in codec master mode\n",
	channels);
	return -EINVAL;
	}
	dai_format \|= SND_SOC_DAIFMT_CBM_CFM;
	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
	ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKT_EXTAL,
	priv->mclk_freq, SND_SOC_CLOCK_IN);
	else
	ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKR_EXTAL,
	priv->mclk_freq, SND_SOC_CLOCK_IN);
	if (ret) {
	dev_err(dev, "failed to set cpu sysclk: %d\n", ret);
	return ret;
	}

	ret = snd_soc_dai_set_sysclk(codec_dai, 0,
	priv->mclk_freq, SND_SOC_CLOCK_OUT);
	if (ret) {
	dev_err(dev, "failed to set codec sysclk: %d\n", ret);
	return ret;
	}

	} else {
	dai_format \|= SND_SOC_DAIFMT_CBS_CFS;
	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
	ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKT_EXTAL,
	priv->mclk_freq, SND_SOC_CLOCK_OUT);
	else
	ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKR_EXTAL,
	priv->mclk_freq, SND_SOC_CLOCK_OUT);
	if (ret) {
	dev_err(dev, "failed to set cpu sysclk: %d\n", ret);
	return ret;
	}

	ret = snd_soc_dai_set_sysclk(codec_dai, 0,
	priv->mclk_freq, SND_SOC_CLOCK_IN);
	if (ret) {
	dev_err(dev, "failed to set codec sysclk: %d\n", ret);
	return ret;
	}
	}

	/* set cpu DAI configuration */
	ret = snd_soc_dai_set_fmt(cpu_dai, dai_format);
	if (ret) {
	dev_err(dev, "failed to set cpu dai fmt: %d\n", ret);
	return ret;
	}
	/* set i.MX active slot mask */
	if (priv->is_stream_tdm[tx]) {
	/* 2 required by ESAI BCLK divisors, 8 slots, 32 width */
	if (priv->is_codec_master)
	max_tdm_rate = priv->mclk_freq / (8*32);
	else
	max_tdm_rate = priv->esai_freq / (2832);
	if (params_rate(params) > max_tdm_rate) {
	dev_err(dev,
	"maximum supported sampling rate for %d channels is %dKHz\n",
	channels, max_tdm_rate / 1000);
	return -EINVAL;
	}

	/*
	* Per datasheet, the codec expects 8 slots and 32 bits
	* for every slot in TDM mode.
	*/
	snd_soc_dai_set_tdm_slot(cpu_dai,
	BIT(channels) - 1, BIT(channels) - 1,
	8, 32);
	} else
	snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);

	/* set codec DAI configuration */
	ret = snd_soc_dai_set_fmt(codec_dai, dai_format);
	if (ret) {
	dev_err(dev, "failed to set codec dai fmt: %d\n", ret);
	return ret;
	}
	return 0;
	}

	static int imx_cs42888_surround_hw_free(struct snd_pcm_substream *substream)
	{
	struct imx_priv *priv = &card_priv;
	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;

	priv->is_stream_in_use[tx] = false;

	return 0;
	}

	static int imx_cs42888_surround_startup(struct snd_pcm_substream *substream)
	{
	struct snd_pcm_runtime *runtime = substream->runtime;
	static struct snd_pcm_hw_constraint_list constraint_rates;
	struct imx_priv *priv = &card_priv;
	struct device *dev = &priv->pdev->dev;
	static u32 support_rates[SUPPORT_RATE_NUM];
	int ret;

	priv->mclk_freq = clk_get_rate(priv->codec_clk);

	if (priv->mclk_freq % 12288000 == 0) {
	support_rates[0] = 48000;
	support_rates[1] = 96000;
	support_rates[2] = 192000;
	constraint_rates.list = support_rates;
	constraint_rates.count = 3;

	ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
	&constraint_rates);
	if (ret)
	return ret;
	} else
	dev_warn(dev, "mclk may be not supported %d\n", priv->mclk_freq);

	return 0;
	}

	static struct snd_soc_ops imx_cs42888_surround_ops = {
	.startup = imx_cs42888_surround_startup,
	.hw_params = imx_cs42888_surround_hw_params,
	.hw_free = imx_cs42888_surround_hw_free,
	};

	/**
	* imx_cs42888_surround_startup() is to set constrain for hw parameter, but
	* backend use same runtime as frontend, for p2p backend need to use different
	* parameter, so backend can't use the startup.
	*/
	static struct snd_soc_ops imx_cs42888_surround_ops_be = {
	.hw_params = imx_cs42888_surround_hw_params,
	.hw_free = imx_cs42888_surround_hw_free,
	};


	static const struct snd_soc_dapm_widget imx_cs42888_dapm_widgets[] = {
	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
	SND_SOC_DAPM_LINE("Line In Jack", NULL),
	};

	static const struct snd_soc_dapm_route audio_map[] = {
	/* Line out jack */
	{"Line Out Jack", NULL, "AOUT1L"},
	{"Line Out Jack", NULL, "AOUT1R"},
	{"Line Out Jack", NULL, "AOUT2L"},
	{"Line Out Jack", NULL, "AOUT2R"},
	{"Line Out Jack", NULL, "AOUT3L"},
	{"Line Out Jack", NULL, "AOUT3R"},
	{"Line Out Jack", NULL, "AOUT4L"},
	{"Line Out Jack", NULL, "AOUT4R"},
	{"AIN1L", NULL, "Line In Jack"},
	{"AIN1R", NULL, "Line In Jack"},
	{"AIN2L", NULL, "Line In Jack"},
	{"AIN2R", NULL, "Line In Jack"},
	{"Playback", NULL, "CPU-Playback"},/* dai route for be and fe */
	{"CPU-Capture", NULL, "Capture"},
	{"CPU-Playback", NULL, "ASRC-Playback"},
	{"ASRC-Capture", NULL, "CPU-Capture"},
	};

	static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
	struct snd_pcm_hw_params *params) {

	struct imx_priv *priv = &card_priv;
	struct snd_interval *rate;
	struct snd_mask *mask;

	if (!priv->asrc_pdev)
	return -EINVAL;

	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 imx_cs42888_dai[] = {
	{
	.name = "HiFi",
	.stream_name = "HiFi",
	.codec_dai_name = "cs42888",
	.ops = &imx_cs42888_surround_ops,
	.ignore_pmdown_time = 1,
	},
	{
	.name = "HiFi-ASRC-FE",
	.stream_name = "HiFi-ASRC-FE",
	.codec_name = "snd-soc-dummy",
	.codec_dai_name = "snd-soc-dummy-dai",
	.dynamic = 1,
	.ignore_pmdown_time = 1,
	.dpcm_playback = 1,
	.dpcm_capture = 1,
	.dpcm_merged_chan = 1,
	},
	{
	.name = "HiFi-ASRC-BE",
	.stream_name = "HiFi-ASRC-BE",
	.codec_dai_name = "cs42888",
	.platform_name = "snd-soc-dummy",
	.no_pcm = 1,
	.ignore_pmdown_time = 1,
	.dpcm_playback = 1,
	.dpcm_capture = 1,
	.ops = &imx_cs42888_surround_ops_be,
	.be_hw_params_fixup = be_hw_params_fixup,
	},
	};

	static struct snd_soc_card snd_soc_card_imx_cs42888 = {
	.name = "cs42888-audio",
	.dai_link = imx_cs42888_dai,
	.dapm_widgets = imx_cs42888_dapm_widgets,
	.num_dapm_widgets = ARRAY_SIZE(imx_cs42888_dapm_widgets),
	.dapm_routes = audio_map,
	.num_dapm_routes = ARRAY_SIZE(audio_map),
	.owner = THIS_MODULE,
	};

	/*
	* This function will register the snd_soc_pcm_link drivers.
	*/
	static int imx_cs42888_probe(struct platform_device *pdev)
	{
	struct device_node esai_np, codec_np;
	struct device_node *asrc_np = NULL;
	struct platform_device *esai_pdev;
	struct platform_device *asrc_pdev = NULL;
	struct imx_priv *priv = &card_priv;
	int ret;
	u32 width;

	priv->pdev = pdev;
	priv->asrc_pdev = NULL;

	if (of_property_read_bool(pdev->dev.of_node, "codec-rpmsg"))
	priv->is_codec_rpmsg = true;

	esai_np = of_parse_phandle(pdev->dev.of_node, "esai-controller", 0);
	codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
	if (!esai_np \|\| !codec_np) {
	dev_err(&pdev->dev, "phandle missing or invalid\n");
	ret = -EINVAL;
	goto fail;
	}

	asrc_np = of_parse_phandle(pdev->dev.of_node, "asrc-controller", 0);
	if (asrc_np) {
	asrc_pdev = of_find_device_by_node(asrc_np);
	priv->asrc_pdev = asrc_pdev;
	}

	esai_pdev = of_find_device_by_node(esai_np);
	if (!esai_pdev) {
	dev_err(&pdev->dev, "failed to find ESAI platform device\n");
	ret = -EINVAL;
	goto fail;
	}

	if (priv->is_codec_rpmsg) {
	struct platform_device *codec_dev;

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

	priv->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
	if (IS_ERR(priv->codec_clk)) {
	ret = PTR_ERR(priv->codec_clk);
	dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret);
	goto fail;
	}

	} else {
	struct i2c_client *codec_dev;

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

	priv->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
	if (IS_ERR(priv->codec_clk)) {
	ret = PTR_ERR(priv->codec_clk);
	dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret);
	goto fail;
	}
	}

	if (priv->is_codec_rpmsg) {
	imx_cs42888_dai[0].codec_name = "rpmsg-audio-codec-cs42888";
	imx_cs42888_dai[0].codec_dai_name = "cs42888";
	} else {
	imx_cs42888_dai[0].codec_of_node = codec_np;
	}

	/if there is no asrc controller, we only enable one device/
	if (!asrc_pdev) {
	imx_cs42888_dai[0].cpu_dai_name = dev_name(&esai_pdev->dev);
	imx_cs42888_dai[0].platform_of_node = esai_np;
	snd_soc_card_imx_cs42888.num_links = 1;
	snd_soc_card_imx_cs42888.num_dapm_routes =
	ARRAY_SIZE(audio_map) - 2;
	} else {
	imx_cs42888_dai[0].cpu_dai_name = dev_name(&esai_pdev->dev);
	imx_cs42888_dai[0].platform_of_node = esai_np;
	imx_cs42888_dai[1].cpu_of_node = asrc_np;
	imx_cs42888_dai[1].platform_of_node = asrc_np;
	imx_cs42888_dai[2].cpu_dai_name = dev_name(&esai_pdev->dev);
	snd_soc_card_imx_cs42888.num_links = 3;

	if (priv->is_codec_rpmsg) {
	imx_cs42888_dai[2].codec_name = "rpmsg-audio-codec-cs42888";
	imx_cs42888_dai[2].codec_dai_name = "cs42888";
	} else {
	imx_cs42888_dai[2].codec_of_node = codec_np;
	}

	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 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 fail;
	}

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

	priv->esai_clk = devm_clk_get(&esai_pdev->dev, "extal");
	if (IS_ERR(priv->esai_clk)) {
	ret = PTR_ERR(priv->esai_clk);
	dev_err(&esai_pdev->dev, "failed to get cpu clk: %d\n", ret);
	goto fail;
	}

	priv->is_codec_master = false;
	if (of_property_read_bool(pdev->dev.of_node, "codec-master"))
	priv->is_codec_master = true;

	snd_soc_card_imx_cs42888.dev = &pdev->dev;

	platform_set_drvdata(pdev, &snd_soc_card_imx_cs42888);

	ret = snd_soc_register_card(&snd_soc_card_imx_cs42888);
	if (ret)
	dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
	fail:
	if (asrc_np)
	of_node_put(asrc_np);
	if (esai_np)
	of_node_put(esai_np);
	if (codec_np)
	of_node_put(codec_np);
	return ret;
	}

	static int imx_cs42888_remove(struct platform_device *pdev)
	{
	snd_soc_unregister_card(&snd_soc_card_imx_cs42888);
	return 0;
	}

	static const struct of_device_id imx_cs42888_dt_ids[] = {
	{ .compatible = "fsl,imx-audio-cs42888", },
	{ /* sentinel */ }
	};

	static struct platform_driver imx_cs42888_driver = {
	.probe = imx_cs42888_probe,
	.remove = imx_cs42888_remove,
	.driver = {
	.name = "imx-cs42888",
	.pm = &snd_soc_pm_ops,
	.of_match_table = imx_cs42888_dt_ids,
	},
	};
	module_platform_driver(imx_cs42888_driver);

	MODULE_AUTHOR("Freescale Semiconductor, Inc.");
	MODULE_DESCRIPTION("ALSA SoC cs42888 Machine Layer Driver");
	MODULE_ALIAS("platform:imx-cs42888");
	MODULE_LICENSE("GPL");