/*
   * Copyright (C) 2010-2014 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 {
  	unsigned int mclk_freq;
  	struct platform_device *pdev;
  	struct platform_device *asrc_pdev;
  	u32 asrc_rate;
  	u32 asrc_format;
  };
  
  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;
  	u32 dai_format = 0;
  
  	dai_format = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
  		     SND_SOC_DAIFMT_CBS_CFS;
  	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
  		snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKT_EXTAL,
  			       priv->mclk_freq, SND_SOC_CLOCK_OUT);
  	else
  		snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKR_EXTAL,
  			       priv->mclk_freq, SND_SOC_CLOCK_OUT);
  	snd_soc_dai_set_sysclk(codec_dai, 0, priv->mclk_freq, SND_SOC_CLOCK_IN);
  
  	/* set cpu DAI configuration */
  	snd_soc_dai_set_fmt(cpu_dai, dai_format);
  	/* set i.MX active slot mask */
  	snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);
  
  	/* set codec DAI configuration */
  	snd_soc_dai_set_fmt(codec_dai, dai_format);
  	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;
  
  	if (priv->mclk_freq == 24576000) {
  		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
  ", 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,
  };
  
  /**
   * 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,
  };
  
  
  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"},
  	{"CPU-Playback",  NULL, "ASRC-Playback"},
  	{"Playback",  NULL, "CPU-Playback"},/* dai route for be and fe */
  	{"ASRC-Capture",  NULL, "CPU-Capture"},
  	{"CPU-Capture",  NULL, "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,
  	},
  	{
  		.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),
  };
  
  /*
   * 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;
  	struct platform_device *esai_pdev;
  	struct platform_device *asrc_pdev = NULL;
  	struct i2c_client *codec_dev;
  	struct imx_priv *priv = &card_priv;
  	struct clk *codec_clk = NULL;
  	int ret;
  	u32 width;
  
  	priv->pdev = pdev;
  	priv->asrc_pdev = NULL;
  
  	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
  ");
  		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
  ");
  		ret = -EINVAL;
  		goto fail;
  	}
  	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
  ");
  		ret = -EINVAL;
  		goto fail;
  	}
  
  	/*if there is no asrc controller, we only enable one device*/
  	if (!asrc_pdev) {
  		imx_cs42888_dai[0].codec_of_node   = codec_np;
  		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;
  	} else {
  		imx_cs42888_dai[0].codec_of_node   = codec_np;
  		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].codec_of_node   = codec_np;
  		imx_cs42888_dai[2].cpu_dai_name    = dev_name(&esai_pdev->dev);
  		snd_soc_card_imx_cs42888.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
  ");
  			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
  ");
  			ret = -EINVAL;
  			goto fail;
  		}
  
  		if (width == 24)
  			priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
  		else
  			priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
  	}
  
  	codec_clk = devm_clk_get(&codec_dev->dev, NULL);
  	if (IS_ERR(codec_clk)) {
  		ret = PTR_ERR(codec_clk);
  		dev_err(&codec_dev->dev, "failed to get codec clk: %d
  ", ret);
  		goto fail;
  	}
  	priv->mclk_freq = clk_get_rate(codec_clk);
  
  	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)
  ", 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",
  		.owner = THIS_MODULE,
  		.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");