text: // SPDX-License-Identifier: GPL-2.0
  //
  // Freescale Generic ASoC Sound Card driver with ASRC
  //
  // Copyright (C) 2014 Freescale Semiconductor, Inc.
  //
  // Author: Nicolin Chen <nicoleotsuka@gmail.com>
  
  #include <linux/clk.h>
  #include <linux/i2c.h>
  #include <linux/module.h>
  #include <linux/of_platform.h>
  #if IS_ENABLED(CONFIG_SND_AC97_CODEC)
  #include <sound/ac97_codec.h>
  #endif
  #include <sound/pcm_params.h>
  #include <sound/soc.h>
  #include <sound/jack.h>
  #include <sound/simple_card_utils.h>
  
  #include "fsl_esai.h"
  #include "fsl_sai.h"
  #include "imx-audmux.h"
  
  #include "../codecs/sgtl5000.h"
  #include "../codecs/wm8962.h"
  #include "../codecs/wm8960.h"
  #include "../codecs/wm8994.h"
  
  #define CS427x_SYSCLK_MCLK 0
  
  #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)
  
  /**
   * struct codec_priv - CODEC private data
   * @mclk_freq: Clock rate of MCLK
   * @free_freq: Clock rate of MCLK for hw_free()
   * @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;
  	unsigned long free_freq;
  	u32 mclk_id;
  	u32 fll_id;
  	u32 pll_id;
  };
  
  /**
   * struct cpu_priv - CPU private data
   * @sysclk_freq: SYSCLK rates for set_sysclk()
   * @sysclk_dir: SYSCLK directions for set_sysclk()
   * @sysclk_id: 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;
  };
  
  /**
   * struct fsl_asoc_card_priv - Freescale Generic ASOC card private data
   * @dai_link: DAI link structure including normal one and DPCM link
   * @hp_jack: Headphone Jack structure
   * @mic_jack: Microphone Jack structure
   * @pdev: platform device pointer
   * @codec_priv: CODEC private data
   * @cpu_priv: CPU private data
   * @card: ASoC card structure
   * @streams: Mask of current active streams
   * @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 asoc_simple_jack hp_jack;
  	struct asoc_simple_jack mic_jack;
  	struct platform_device *pdev;
  	struct codec_priv codec_priv;
  	struct cpu_priv cpu_priv;
  	struct snd_soc_card card;
  	u8 streams;
  	u32 sample_rate;
  	snd_pcm_format_t sample_format;
  	u32 asrc_rate;
  	snd_pcm_format_t asrc_format;
  	u32 dai_fmt;
  	char name[32];
  };
  
  /*
   * This dapm route map exists for DPCM link only.
   * The other routes shall go through Device Tree.
   *
   * Note: keep all ASRC routes in the second half
   *	 to drop them easily for non-ASRC cases.
   */
  static const struct snd_soc_dapm_route audio_map[] = {
  	/* 1st half -- Normal DAPM routes */
  	{"Playback",  NULL, "CPU-Playback"},
  	{"CPU-Capture",  NULL, "Capture"},
  	/* 2nd half -- ASRC DAPM routes */
  	{"CPU-Playback",  NULL, "ASRC-Playback"},
  	{"ASRC-Capture",  NULL, "CPU-Capture"},
  };
  
  static const struct snd_soc_dapm_route audio_map_ac97[] = {
  	/* 1st half -- Normal DAPM routes */
  	{"Playback",  NULL, "AC97 Playback"},
  	{"AC97 Capture",  NULL, "Capture"},
  	/* 2nd half -- ASRC DAPM routes */
  	{"AC97 Playback",  NULL, "ASRC-Playback"},
  	{"ASRC-Capture",  NULL, "AC97 Capture"},
  };
  
  static const struct snd_soc_dapm_route audio_map_tx[] = {
  	/* 1st half -- Normal DAPM routes */
  	{"Playback",  NULL, "CPU-Playback"},
  	/* 2nd half -- ASRC DAPM routes */
  	{"CPU-Playback",  NULL, "ASRC-Playback"},
  };
  
  static const struct snd_soc_dapm_route audio_map_rx[] = {
  	/* 1st half -- Normal DAPM routes */
  	{"CPU-Capture",  NULL, "Capture"},
  	/* 2nd half -- ASRC DAPM routes */
  	{"ASRC-Capture",  NULL, "CPU-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 bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
  {
  	return priv->dai_fmt == SND_SOC_DAIFMT_AC97;
  }
  
  static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
  				   struct snd_pcm_hw_params *params)
  {
  	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
  	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
  	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
  	struct codec_priv *codec_priv = &priv->codec_priv;
  	struct cpu_priv *cpu_priv = &priv->cpu_priv;
  	struct device *dev = rtd->card->dev;
  	unsigned int pll_out;
  	int ret;
  
  	priv->sample_rate = params_rate(params);
  	priv->sample_format = params_format(params);
  	priv->streams |= BIT(substream->stream);
  
  	if (fsl_asoc_card_is_ac97(priv))
  		return 0;
  
  	/* Specific configurations of DAIs starts from here */
  	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx],
  				     cpu_priv->sysclk_freq[tx],
  				     cpu_priv->sysclk_dir[tx]);
  	if (ret && ret != -ENOTSUPP) {
  		dev_err(dev, "failed to set sysclk for cpu dai\n");
  		goto fail;
  	}
  
  	if (cpu_priv->slot_width) {
  		ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2,
  					       cpu_priv->slot_width);
  		if (ret && ret != -ENOTSUPP) {
  			dev_err(dev, "failed to set TDM slot for cpu dai\n");
  			goto fail;
  		}
  	}
  
  	/* Specific configuration for PLL */
  	if (codec_priv->pll_id && codec_priv->fll_id) {
  		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(asoc_rtd_to_codec(rtd, 0),
  					  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);
  			goto fail;
  		}
  
  		ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
  					     codec_priv->fll_id,
  					     pll_out, SND_SOC_CLOCK_IN);
  
  		if (ret && ret != -ENOTSUPP) {
  			dev_err(dev, "failed to set SYSCLK: %d\n", ret);
  			goto fail;
  		}
  	}
  
  	return 0;
  
  fail:
  	priv->streams &= ~BIT(substream->stream);
  	return ret;
  }
  
  static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream)
  {
  	struct snd_soc_pcm_runtime *rtd = substream->private_data;
  	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
  	struct codec_priv *codec_priv = &priv->codec_priv;
  	struct device *dev = rtd->card->dev;
  	int ret;
  
  	priv->streams &= ~BIT(substream->stream);
  
  	if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) {
  		/* Force freq to be free_freq to avoid error message in codec */
  		ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
  					     codec_priv->mclk_id,
  					     codec_priv->free_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(asoc_rtd_to_codec(rtd, 0),
  					  codec_priv->pll_id, 0, 0, 0);
  		if (ret && ret != -ENOTSUPP) {
  			dev_err(dev, "failed to stop FLL: %d\n", ret);
  			return ret;
  		}
  	}
  
  	return 0;
  }
  
  static const struct snd_soc_ops fsl_asoc_card_ops = {
  	.hw_params = fsl_asoc_card_hw_params,
  	.hw_free = fsl_asoc_card_hw_free,
  };
  
  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_format(mask, priv->asrc_format);
  
  	return 0;
  }
  
  SND_SOC_DAILINK_DEFS(hifi,
  	DAILINK_COMP_ARRAY(COMP_EMPTY()),
  	DAILINK_COMP_ARRAY(COMP_EMPTY()),
  	DAILINK_COMP_ARRAY(COMP_EMPTY()));
  
  SND_SOC_DAILINK_DEFS(hifi_fe,
  	DAILINK_COMP_ARRAY(COMP_EMPTY()),
  	DAILINK_COMP_ARRAY(COMP_DUMMY()),
  	DAILINK_COMP_ARRAY(COMP_EMPTY()));
  
  SND_SOC_DAILINK_DEFS(hifi_be,
  	DAILINK_COMP_ARRAY(COMP_EMPTY()),
  	DAILINK_COMP_ARRAY(COMP_EMPTY()),
  	DAILINK_COMP_ARRAY(COMP_DUMMY()));
  
  static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
  	/* Default ASoC DAI Link*/
  	{
  		.name = "HiFi",
  		.stream_name = "HiFi",
  		.ops = &fsl_asoc_card_ops,
  		SND_SOC_DAILINK_REG(hifi),
  	},
  	/* DPCM Link between Front-End and Back-End (Optional) */
  	{
  		.name = "HiFi-ASRC-FE",
  		.stream_name = "HiFi-ASRC-FE",
  		.dpcm_playback = 1,
  		.dpcm_capture = 1,
  		.dynamic = 1,
  		SND_SOC_DAILINK_REG(hifi_fe),
  	},
  	{
  		.name = "HiFi-ASRC-BE",
  		.stream_name = "HiFi-ASRC-BE",
  		.be_hw_params_fixup = be_hw_params_fixup,
  		.ops = &fsl_asoc_card_ops,
  		.dpcm_playback = 1,
  		.dpcm_capture = 1,
  		.no_pcm = 1,
  		SND_SOC_DAILINK_REG(hifi_be),
  	},
  };
  
  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 except AC97.
  	 * 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:
  		if (!fsl_asoc_card_is_ac97(priv))
  			return -EINVAL;
  	}
  
  	if (fsl_asoc_card_is_ac97(priv)) {
  		int_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
  			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
  			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
  		ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
  			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
  			   IMX_AUDMUX_V2_PTCR_TFSDIR;
  	}
  
  	/* Asynchronous mode can not be set along with RCLKDIR */
  	if (!fsl_asoc_card_is_ac97(priv)) {
  		unsigned int pdcr =
  				IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port);
  
  		ret = imx_audmux_v2_configure_port(int_port, 0,
  						   pdcr);
  		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;
  	}
  
  	if (!fsl_asoc_card_is_ac97(priv)) {
  		unsigned int pdcr =
  				IMX_AUDMUX_V2_PDCR_RXDSEL(int_port);
  
  		ret = imx_audmux_v2_configure_port(ext_port, 0,
  						   pdcr);
  		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 hp_jack_event(struct notifier_block *nb, unsigned long event,
  			 void *data)
  {
  	struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
  	struct snd_soc_dapm_context *dapm = &jack->card->dapm;
  
  	if (event & SND_JACK_HEADPHONE)
  		/* Disable speaker if headphone is plugged in */
  		snd_soc_dapm_disable_pin(dapm, "Ext Spk");
  	else
  		snd_soc_dapm_enable_pin(dapm, "Ext Spk");
  
  	return 0;
  }
  
  static struct notifier_block hp_jack_nb = {
  	.notifier_call = hp_jack_event,
  };
  
  static int mic_jack_event(struct notifier_block *nb, unsigned long event,
  			  void *data)
  {
  	struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
  	struct snd_soc_dapm_context *dapm = &jack->card->dapm;
  
  	if (event & SND_JACK_MICROPHONE)
  		/* Disable dmic if microphone is plugged in */
  		snd_soc_dapm_disable_pin(dapm, "DMIC");
  	else
  		snd_soc_dapm_enable_pin(dapm, "DMIC");
  
  	return 0;
  }
  
  static struct notifier_block mic_jack_nb = {
  	.notifier_call = mic_jack_event,
  };
  
  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_pcm_runtime *rtd = list_first_entry(
  			&card->rtd_list, struct snd_soc_pcm_runtime, list);
  	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
  	struct codec_priv *codec_priv = &priv->codec_priv;
  	struct device *dev = card->dev;
  	int ret;
  
  	if (fsl_asoc_card_is_ac97(priv)) {
  #if IS_ENABLED(CONFIG_SND_AC97_CODEC)
  		struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
  		struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component);
  
  		/*
  		 * Use slots 3/4 for S/PDIF so SSI won't try to enable
  		 * other slots and send some samples there
  		 * due to SLOTREQ bits for S/PDIF received from codec
  		 */
  		snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
  				     AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4);
  #endif
  
  		return 0;
  	}
  
  	ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
  				     codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
  	if (ret && ret != -ENOTSUPP) {
  		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 device_node *bitclkmaster = NULL;
  	struct device_node *framemaster = NULL;
  	struct platform_device *cpu_pdev;
  	struct fsl_asoc_card_priv *priv;
  	struct device *codec_dev = NULL;
  	const char *codec_dai_name;
  	const char *codec_dev_name;
  	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);
  	if (!cpu_np) {
  		dev_err(&pdev->dev, "CPU 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_np = of_parse_phandle(np, "audio-codec", 0);
  	if (codec_np) {
  		struct platform_device *codec_pdev;
  		struct i2c_client *codec_i2c;
  
  		codec_i2c = of_find_i2c_device_by_node(codec_np);
  		if (codec_i2c) {
  			codec_dev = &codec_i2c->dev;
  			codec_dev_name = codec_i2c->name;
  		}
  		if (!codec_dev) {
  			codec_pdev = of_find_device_by_node(codec_np);
  			if (codec_pdev) {
  				codec_dev = &codec_pdev->dev;
  				codec_dev_name = codec_pdev->name;
  			}
  		}
  	}
  
  	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 */
  	if (codec_dev) {
  		struct clk *codec_clk = clk_get(codec_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;
  
  	memcpy(priv->dai_link, fsl_asoc_card_dai,
  	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
  
  	priv->card.dapm_routes = audio_map;
  	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
  	/* Diversify the card configurations */
  	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
  		codec_dai_name = "cs42888";
  		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-cs427x")) {
  		codec_dai_name = "cs4271-hifi";
  		priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK;
  		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
  	} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
  		codec_dai_name = "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-tlv320aic32x4")) {
  		codec_dai_name = "tlv320aic32x4-hifi";
  		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
  	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
  		codec_dai_name = "wm8962";
  		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 if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
  		codec_dai_name = "wm8960-hifi";
  		priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
  		priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
  		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
  	} else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
  		codec_dai_name = "ac97-hifi";
  		priv->dai_fmt = SND_SOC_DAIFMT_AC97;
  		priv->card.dapm_routes = audio_map_ac97;
  		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97);
  	} else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) {
  		codec_dai_name = "fsl-mqs-dai";
  		priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J |
  				SND_SOC_DAIFMT_CBS_CFS |
  				SND_SOC_DAIFMT_NB_NF;
  		priv->dai_link[1].dpcm_capture = 0;
  		priv->dai_link[2].dpcm_capture = 0;
  		priv->card.dapm_routes = audio_map_tx;
  		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
  	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) {
  		codec_dai_name = "wm8524-hifi";
  		priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
  		priv->dai_link[1].dpcm_capture = 0;
  		priv->dai_link[2].dpcm_capture = 0;
  		priv->cpu_priv.slot_width = 32;
  		priv->card.dapm_routes = audio_map_tx;
  		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
  	} else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) {
  		codec_dai_name = "si476x-codec";
  		priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
  		priv->card.dapm_routes = audio_map_rx;
  		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx);
  	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8958")) {
  		codec_dai_name = "wm8994-aif1";
  		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
  		priv->codec_priv.mclk_id = WM8994_FLL_SRC_MCLK1;
  		priv->codec_priv.fll_id = WM8994_SYSCLK_FLL1;
  		priv->codec_priv.pll_id = WM8994_FLL1;
  		priv->codec_priv.free_freq = priv->codec_priv.mclk_freq;
  		priv->card.dapm_routes = NULL;
  		priv->card.num_dapm_routes = 0;
  	} else {
  		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
  		ret = -EINVAL;
  		goto asrc_fail;
  	}
  
  	/* Format info from DT is optional. */
  	snd_soc_daifmt_parse_clock_provider_as_phandle(np, NULL, &bitclkmaster, &framemaster);
  	if (bitclkmaster || framemaster) {
  		unsigned int daifmt = snd_soc_daifmt_parse_format(np, NULL);
  
  		if (codec_np == bitclkmaster)
  			daifmt |= (codec_np == framemaster) ?
  				SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS;
  		else
  			daifmt |= (codec_np == framemaster) ?
  				SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS;
  
  		/* Override dai_fmt with value from DT */
  		priv->dai_fmt = daifmt;
  	}
  
  	/* Change direction according to format */
  	if (priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) {
  		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN;
  		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN;
  	}
  
  	of_node_put(bitclkmaster);
  	of_node_put(framemaster);
  
  	if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
  		dev_dbg(&pdev->dev, "failed to find codec device\n");
  		ret = -EPROBE_DEFER;
  		goto asrc_fail;
  	}
  
  	/* Common settings for corresponding Freescale CPU DAI driver */
  	if (of_node_name_eq(cpu_np, "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 (of_node_name_eq(cpu_np, "esai")) {
  		struct clk *esai_clk = clk_get(&cpu_pdev->dev, "extal");
  
  		if (!IS_ERR(esai_clk)) {
  			priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(esai_clk);
  			priv->cpu_priv.sysclk_freq[RX] = clk_get_rate(esai_clk);
  			clk_put(esai_clk);
  		} else if (PTR_ERR(esai_clk) == -EPROBE_DEFER) {
  			ret = -EPROBE_DEFER;
  			goto asrc_fail;
  		}
  
  		priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
  		priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
  	} else if (of_node_name_eq(cpu_np, "sai")) {
  		priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
  		priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
  	}
  
  	/* Initialize sound card */
  	priv->pdev = pdev;
  	priv->card.dev = &pdev->dev;
  	priv->card.owner = THIS_MODULE;
  	ret = snd_soc_of_parse_card_name(&priv->card, "model");
  	if (ret) {
  		snprintf(priv->name, sizeof(priv->name), "%s-audio",
  			 fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name);
  		priv->card.name = priv->name;
  	}
  	priv->card.dai_link = priv->dai_link;
  	priv->card.late_probe = fsl_asoc_card_late_probe;
  	priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
  	priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
  
  	/* Drop the second half of DAPM routes -- ASRC */
  	if (!asrc_pdev)
  		priv->card.num_dapm_routes /= 2;
  
  	if (of_property_read_bool(np, "audio-routing")) {
  		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].cpus->of_node = cpu_np;
  	priv->dai_link[0].codecs->dai_name = codec_dai_name;
  
  	if (!fsl_asoc_card_is_ac97(priv))
  		priv->dai_link[0].codecs->of_node = codec_np;
  	else {
  		u32 idx;
  
  		ret = of_property_read_u32(cpu_np, "cell-index", &idx);
  		if (ret) {
  			dev_err(&pdev->dev,
  				"cannot get CPU index property\n");
  			goto asrc_fail;
  		}
  
  		priv->dai_link[0].codecs->name =
  				devm_kasprintf(&pdev->dev, GFP_KERNEL,
  					       "ac97-codec.%u",
  					       (unsigned int)idx);
  		if (!priv->dai_link[0].codecs->name) {
  			ret = -ENOMEM;
  			goto asrc_fail;
  		}
  	}
  
  	priv->dai_link[0].platforms->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].cpus->of_node = asrc_np;
  		priv->dai_link[1].platforms->of_node = asrc_np;
  		priv->dai_link[2].codecs->dai_name = codec_dai_name;
  		priv->dai_link[2].codecs->of_node = codec_np;
  		priv->dai_link[2].codecs->name =
  				priv->dai_link[0].codecs->name;
  		priv->dai_link[2].cpus->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-format",
  					   &priv->asrc_format);
  		if (ret) {
  			/* Fallback to old binding; translate to asrc_format */
  			ret = of_property_read_u32(asrc_np, "fsl,asrc-width",
  						   &width);
  			if (ret) {
  				dev_err(&pdev->dev,
  					"failed to decide output format\n");
  				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) {
  		if (ret != -EPROBE_DEFER)
  			dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
  		goto asrc_fail;
  	}
  
  	/*
  	 * Properties "hp-det-gpio" and "mic-det-gpio" are optional, and
  	 * asoc_simple_init_jack uses these properties for creating
  	 * Headphone Jack and Microphone Jack.
  	 *
  	 * The notifier is initialized in snd_soc_card_jack_new(), then
  	 * snd_soc_jack_notifier_register can be called.
  	 */
  	if (of_property_read_bool(np, "hp-det-gpio")) {
  		ret = asoc_simple_init_jack(&priv->card, &priv->hp_jack,
  					    1, NULL, "Headphone Jack");
  		if (ret)
  			goto asrc_fail;
  
  		snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb);
  	}
  
  	if (of_property_read_bool(np, "mic-det-gpio")) {
  		ret = asoc_simple_init_jack(&priv->card, &priv->mic_jack,
  					    0, NULL, "Mic Jack");
  		if (ret)
  			goto asrc_fail;
  
  		snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb);
  	}
  
  asrc_fail:
  	of_node_put(asrc_np);
  	of_node_put(codec_np);
  	put_device(&cpu_pdev->dev);
  fail:
  	of_node_put(cpu_np);
  
  	return ret;
  }
  
  static const struct of_device_id fsl_asoc_card_dt_ids[] = {
  	{ .compatible = "fsl,imx-audio-ac97", },
  	{ .compatible = "fsl,imx-audio-cs42888", },
  	{ .compatible = "fsl,imx-audio-cs427x", },
  	{ .compatible = "fsl,imx-audio-tlv320aic32x4", },
  	{ .compatible = "fsl,imx-audio-sgtl5000", },
  	{ .compatible = "fsl,imx-audio-wm8962", },
  	{ .compatible = "fsl,imx-audio-wm8960", },
  	{ .compatible = "fsl,imx-audio-mqs", },
  	{ .compatible = "fsl,imx-audio-wm8524", },
  	{ .compatible = "fsl,imx-audio-si476x", },
  	{ .compatible = "fsl,imx-audio-wm8958", },
  	{}
  };
  MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
  
  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");
  