// 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"
#include "../codecs/tlv320aic31xx.h"
#include "../codecs/nau8822.h"
#include "../codecs/wm8904.h"

#define DRIVER_NAME "fsl-asoc-card"

#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: Main clock of the CODEC
 * @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 {
        struct clk *mclk;
        unsigned long mclk_freq;
        unsigned long free_freq;
        u32 mclk_id;
        int fll_id;
        int 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()
 * @sysclk_ratio: SYSCLK ratio on sample rate
 * @slot_width: Slot width of each frame
 * @slot_num: Number of slots 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 sysclk_ratio[2];
        u32 slot_width;
        u32 slot_num;
};

/**
 * 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 simple_util_jack hp_jack;
        struct simple_util_jack mic_jack;
        struct platform_device *pdev;
        struct codec_priv codec_priv[2];
        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 */
        {"AC97 Playback",  NULL, "CPU AC97 Playback"},
        {"CPU AC97 Capture",  NULL, "AC97 Capture"},
        /* 2nd half -- ASRC DAPM routes */
        {"CPU AC97 Playback",  NULL, "ASRC-Playback"},
        {"ASRC-Capture",  NULL, "CPU 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 = snd_soc_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;
        struct snd_soc_dai *codec_dai;
        struct cpu_priv *cpu_priv = &priv->cpu_priv;
        struct device *dev = rtd->card->dev;
        unsigned int pll_out, sysclk_freq;
        int codec_idx;
        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;

        if (!cpu_priv->sysclk_freq[tx] && cpu_priv->sysclk_ratio[tx])
                sysclk_freq = priv->sample_rate * cpu_priv->sysclk_ratio[tx];
        else
                sysclk_freq = cpu_priv->sysclk_freq[tx];

        /* Specific configurations of DAIs starts from here */
        ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx],
                                     sysclk_freq,
                                     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) {
                if (!cpu_priv->slot_num)
                        cpu_priv->slot_num = 2;

                ret = snd_soc_dai_set_tdm_slot(snd_soc_rtd_to_cpu(rtd, 0), 0x3, 0x3,
                                               cpu_priv->slot_num,
                                               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 */
        for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) {
                codec_priv = &priv->codec_priv[codec_idx];

                if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
                        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);
                                goto fail;
                        }

                        ret = snd_soc_dai_set_sysclk(codec_dai,
                                                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 = snd_soc_substream_to_rtd(substream);
        struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
        struct codec_priv *codec_priv;
        struct snd_soc_dai *codec_dai;
        struct device *dev = rtd->card->dev;
        int codec_idx;
        int ret;

        priv->streams &= ~BIT(substream->stream);

        for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) {
                codec_priv = &priv->codec_priv[codec_idx];

                if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
                        /* Force freq to be free_freq to avoid error message in codec */
                        ret = snd_soc_dai_set_sysclk(codec_dai,
                                                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(codec_dai,
                                                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;
}

static const 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",
                .dynamic = 1,
        },
        {
                .name = "HiFi-ASRC-BE",
                .stream_name = "HiFi-ASRC-BE",
                .be_hw_params_fixup = be_hw_params_fixup,
                .ops = &fsl_asoc_card_ops,
                .no_pcm = 1,
        },
};

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_CLOCK_PROVIDER_MASK) {
        case SND_SOC_DAIFMT_CBP_CFP:
                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_CBP_CFC:
                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_CBC_CFP:
                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_CBC_CFC:
                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 fsl_asoc_card_spdif_init(struct device_node *codec_np[],
                                    struct device_node *cpu_np,
                                    const char *codec_dai_name[],
                                    struct fsl_asoc_card_priv *priv)
{
        struct device *dev = &priv->pdev->dev;
        struct device_node *np = dev->of_node;

        if (!of_node_name_eq(cpu_np, "spdif")) {
                dev_err(dev, "CPU phandle invalid, should be an SPDIF device\n");
                return -EINVAL;
        }

        priv->dai_link[0].playback_only = true;
        priv->dai_link[0].capture_only = true;

        for (int i = 0; i < 2; i++) {
                if (!codec_np[i])
                        break;

                if (of_device_is_compatible(codec_np[i], "linux,spdif-dit")) {
                        priv->dai_link[0].capture_only = false;
                        codec_dai_name[i] = "dit-hifi";
                } else if (of_device_is_compatible(codec_np[i], "linux,spdif-dir")) {
                        priv->dai_link[0].playback_only = false;
                        codec_dai_name[i] = "dir-hifi";
                }
        }

        // Old SPDIF DT binding
        if (!codec_np[0]) {
                codec_dai_name[0] = snd_soc_dummy_dlc.dai_name;
                if (of_property_read_bool(np, "spdif-out"))
                        priv->dai_link[0].capture_only = false;
                if (of_property_read_bool(np, "spdif-in"))
                        priv->dai_link[0].playback_only = false;
        }

        if (priv->dai_link[0].playback_only && priv->dai_link[0].capture_only) {
                dev_err(dev, "no enabled S/PDIF DAI link\n");
                return -EINVAL;
        }

        if (priv->dai_link[0].playback_only) {
                priv->dai_link[1].playback_only = true;
                priv->dai_link[2].playback_only = true;
                priv->card.dapm_routes = audio_map_tx;
                priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
        } else if (priv->dai_link[0].capture_only) {
                priv->dai_link[1].capture_only = true;
                priv->dai_link[2].capture_only = true;
                priv->card.dapm_routes = audio_map_rx;
                priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx);
        }

        // No DAPM routes with old bindings and dummy codec
        if (!codec_np[0]) {
                priv->card.dapm_routes = NULL;
                priv->card.num_dapm_routes = 0;
        }

        if (codec_np[0] && codec_np[1]) {
                priv->dai_link[0].num_codecs = 2;
                priv->dai_link[2].num_codecs = 2;
        }

        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 = snd_soc_card_to_dapm(jack->card);

        if (event & SND_JACK_HEADPHONE)
                /* Disable speaker if headphone is plugged in */
                return snd_soc_dapm_disable_pin(dapm, "Ext Spk");
        else
                return snd_soc_dapm_enable_pin(dapm, "Ext Spk");
}

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 = snd_soc_card_to_dapm(jack->card);

        if (event & SND_JACK_MICROPHONE)
                /* Disable dmic if microphone is plugged in */
                return snd_soc_dapm_disable_pin(dapm, "DMIC");
        else
                return snd_soc_dapm_enable_pin(dapm, "DMIC");
}

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;
        struct codec_priv *codec_priv;
        struct device *dev = card->dev;
        int codec_idx;
        int ret;

        if (fsl_asoc_card_is_ac97(priv)) {
#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
                struct snd_soc_component *component = snd_soc_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;
        }

        for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) {
                codec_priv = &priv->codec_priv[codec_idx];

                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;
                }

                if (!IS_ERR_OR_NULL(codec_priv->mclk))
                        clk_prepare_enable(codec_priv->mclk);
        }

        return 0;
}

static int fsl_asoc_card_probe(struct platform_device *pdev)
{
        struct device_node *cpu_np, *asrc_np;
        struct snd_soc_dai_link_component *codec_comp;
        struct device_node *codec_np[2];
        struct device_node *np = pdev->dev.of_node;
        struct platform_device *asrc_pdev = NULL;
        struct device_node *bitclkprovider = NULL;
        struct device_node *frameprovider = NULL;
        struct platform_device *cpu_pdev;
        struct fsl_asoc_card_priv *priv;
        struct device *codec_dev[2] = { NULL, NULL };
        struct snd_soc_dai_link_component *dlc;
        const char *codec_dai_name[2];
        const char *codec_dev_name[2];
        u32 asrc_fmt = 0;
        int codec_idx;
        u32 width;
        int ret;

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

        priv->pdev = pdev;

        cpu_np = of_parse_phandle(np, "audio-cpu", 0);
        /* Give a chance to old DT bindings */
        if (!cpu_np)
                cpu_np = of_parse_phandle(np, "ssi-controller", 0);
        if (!cpu_np)
                cpu_np = of_parse_phandle(np, "spdif-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[0] = of_parse_phandle(np, "audio-codec", 0);
        codec_np[1] = of_parse_phandle(np, "audio-codec", 1);

        for (codec_idx = 0; codec_idx < 2; codec_idx++) {
                if (codec_np[codec_idx]) {
                        struct platform_device *codec_pdev;
                        struct i2c_client *codec_i2c;

                        codec_i2c = of_find_i2c_device_by_node(codec_np[codec_idx]);
                        if (codec_i2c) {
                                codec_dev[codec_idx] = &codec_i2c->dev;
                                codec_dev_name[codec_idx] = codec_i2c->name;
                        }
                        if (!codec_dev[codec_idx]) {
                                codec_pdev = of_find_device_by_node(codec_np[codec_idx]);
                                if (codec_pdev) {
                                        codec_dev[codec_idx] = &codec_pdev->dev;
                                        codec_dev_name[codec_idx] = 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 */
        for (codec_idx = 0; codec_idx < 2; codec_idx++) {
                if (codec_dev[codec_idx]) {
                        struct clk *codec_clk = clk_get(codec_dev[codec_idx], NULL);

                        if (!IS_ERR(codec_clk)) {
                                priv->codec_priv[codec_idx].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));
        /*
         * "Default ASoC DAI Link": 1 cpus, 2 codecs, 1 platforms
         * "DPCM Link Front-End":  1 cpus, 1 codecs (dummy), 1 platforms
         * "DPCM Link Back-End": 1 cpus, 2 codecs
         * totally 10 components
         */
        dlc = devm_kcalloc(&pdev->dev, 10, sizeof(*dlc), GFP_KERNEL);
        if (!dlc) {
                ret = -ENOMEM;
                goto asrc_fail;
        }

        priv->dai_link[0].cpus = &dlc[0];
        priv->dai_link[0].num_cpus = 1;
        priv->dai_link[0].codecs = &dlc[1];
        priv->dai_link[0].num_codecs = 1;
        priv->dai_link[0].platforms = &dlc[3];
        priv->dai_link[0].num_platforms = 1;

        priv->dai_link[1].cpus = &dlc[4];
        priv->dai_link[1].num_cpus = 1;
        priv->dai_link[1].codecs = &dlc[5];
        priv->dai_link[1].num_codecs = 0; /* dummy */
        priv->dai_link[1].platforms = &dlc[6];
        priv->dai_link[1].num_platforms = 1;

        priv->dai_link[2].cpus = &dlc[7];
        priv->dai_link[2].num_cpus = 1;
        priv->dai_link[2].codecs = &dlc[8];
        priv->dai_link[2].num_codecs = 1;

        priv->card.dapm_routes = audio_map;
        priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
        priv->card.driver_name = DRIVER_NAME;

        for (codec_idx = 0; codec_idx < 2; codec_idx++) {
                priv->codec_priv[codec_idx].fll_id = -1;
                priv->codec_priv[codec_idx].pll_id = -1;
        }

        /* Diversify the card configurations */
        if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
                codec_dai_name[0] = "cs42888";
                priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv[0].mclk_freq;
                priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv[0].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_CBC_CFC;
        } else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) {
                codec_dai_name[0] = "cs4271-hifi";
                priv->codec_priv[0].mclk_id = CS427x_SYSCLK_MCLK;
                priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
        } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
                codec_dai_name[0] = "sgtl5000";
                priv->codec_priv[0].mclk_id = SGTL5000_SYSCLK;
                priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
        } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic32x4")) {
                codec_dai_name[0] = "tlv320aic32x4-hifi";
                priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
        } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic31xx")) {
                codec_dai_name[0] = "tlv320dac31xx-hifi";
                priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
                priv->dai_link[1].playback_only = 1;
                priv->dai_link[2].playback_only = 1;
                priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
                priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
                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-wm8962")) {
                codec_dai_name[0] = "wm8962";
                priv->codec_priv[0].mclk_id = WM8962_SYSCLK_MCLK;
                priv->codec_priv[0].fll_id = WM8962_SYSCLK_FLL;
                priv->codec_priv[0].pll_id = WM8962_FLL;
                priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
        } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
                codec_dai_name[0] = "wm8960-hifi";
                priv->codec_priv[0].fll_id = WM8960_SYSCLK_AUTO;
                priv->codec_priv[0].pll_id = WM8960_SYSCLK_AUTO;
                priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
        } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
                codec_dai_name[0] = "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[0] = "fsl-mqs-dai";
                priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J |
                                SND_SOC_DAIFMT_CBC_CFC |
                                SND_SOC_DAIFMT_NB_NF;
                priv->dai_link[1].playback_only = 1;
                priv->dai_link[2].playback_only = 1;
                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[0] = "wm8524-hifi";
                priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
                priv->dai_link[1].playback_only = 1;
                priv->dai_link[2].playback_only = 1;
                priv->cpu_priv.slot_width = 32;
                priv->card.dapm_routes = audio_map_tx;
                priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
                priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
                priv->cpu_priv.sysclk_ratio[TX] = 256;
        } else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) {
                codec_dai_name[0] = "si476x-codec";
                priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
                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[0] = "wm8994-aif1";
                priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
                priv->codec_priv[0].mclk_id = WM8994_FLL_SRC_MCLK1;
                priv->codec_priv[0].fll_id = WM8994_SYSCLK_FLL1;
                priv->codec_priv[0].pll_id = WM8994_FLL1;
                priv->codec_priv[0].free_freq = priv->codec_priv[0].mclk_freq;
                priv->card.dapm_routes = NULL;
                priv->card.num_dapm_routes = 0;
        } else if (of_device_is_compatible(np, "fsl,imx-audio-nau8822")) {
                codec_dai_name[0] = "nau8822-hifi";
                priv->codec_priv[0].mclk_id = NAU8822_CLK_MCLK;
                priv->codec_priv[0].fll_id = NAU8822_CLK_PLL;
                priv->codec_priv[0].pll_id = NAU8822_CLK_PLL;
                priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
                if (codec_dev[0])
                        priv->codec_priv[0].mclk = devm_clk_get(codec_dev[0], NULL);
        } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8904")) {
                codec_dai_name[0] = "wm8904-hifi";
                priv->codec_priv[0].mclk_id = WM8904_FLL_MCLK;
                priv->codec_priv[0].fll_id = WM8904_CLK_FLL;
                priv->codec_priv[0].pll_id = WM8904_FLL_MCLK;
                priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
        } else if (of_device_is_compatible(np, "fsl,imx-audio-spdif")) {
                ret = fsl_asoc_card_spdif_init(codec_np, cpu_np, codec_dai_name, priv);
                if (ret)
                        goto asrc_fail;
        } else {
                dev_err(&pdev->dev, "unknown Device Tree compatible\n");
                ret = -EINVAL;
                goto asrc_fail;
        }

        /*
         * Allow setting mclk-id from the device-tree node. Otherwise, the
         * default value for each card configuration is used.
         */
        for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) {
                of_property_read_u32_index(np, "mclk-id", codec_idx,
                                        &priv->codec_priv[codec_idx].mclk_id);
        }

        /* Format info from DT is optional. */
        snd_soc_daifmt_parse_clock_provider_as_phandle(np, NULL, &bitclkprovider, &frameprovider);
        if (bitclkprovider || frameprovider) {
                unsigned int daifmt = snd_soc_daifmt_parse_format(np, NULL);
                bool codec_bitclkprovider = false;
                bool codec_frameprovider = false;

                for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) {
                        if (bitclkprovider && codec_np[codec_idx] == bitclkprovider)
                                codec_bitclkprovider = true;
                        if (frameprovider && codec_np[codec_idx] == frameprovider)
                                codec_frameprovider = true;
                }

                if (codec_bitclkprovider)
                        daifmt |= (codec_frameprovider) ?
                                SND_SOC_DAIFMT_CBP_CFP : SND_SOC_DAIFMT_CBP_CFC;
                else
                        daifmt |= (codec_frameprovider) ?
                                SND_SOC_DAIFMT_CBC_CFP : SND_SOC_DAIFMT_CBC_CFC;

                /* Override dai_fmt with value from DT */
                priv->dai_fmt = daifmt;
        }

        /* Change direction according to format */
        if (priv->dai_fmt & SND_SOC_DAIFMT_CBP_CFP) {
                priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN;
                priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN;
        }

        of_node_put(bitclkprovider);
        of_node_put(frameprovider);

        if (!fsl_asoc_card_is_ac97(priv) && !codec_dev[0]
            && codec_dai_name[0] != snd_soc_dummy_dlc.dai_name) {
                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->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[0]);
                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_present(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;
        for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) {
                codec_comp->dai_name = codec_dai_name[codec_idx];
        }

        // Old SPDIF DT binding support
        if (codec_dai_name[0] == snd_soc_dummy_dlc.dai_name)
                priv->dai_link[0].codecs[0].name = snd_soc_dummy_dlc.name;

        if (!fsl_asoc_card_is_ac97(priv)) {
                for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) {
                        codec_comp->of_node = codec_np[codec_idx];
                }
        } 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[0].name =
                                devm_kasprintf(&pdev->dev, GFP_KERNEL,
                                               "ac97-codec.%u",
                                               (unsigned int)idx);
                if (!priv->dai_link[0].codecs[0].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 exists */
                priv->dai_link[1].cpus->of_node = asrc_np;
                priv->dai_link[1].platforms->of_node = asrc_np;
                for_each_link_codecs((&(priv->dai_link[2])), codec_idx, codec_comp) {
                        codec_comp->dai_name = priv->dai_link[0].codecs[codec_idx].dai_name;
                        codec_comp->of_node = priv->dai_link[0].codecs[codec_idx].of_node;
                        codec_comp->name = priv->dai_link[0].codecs[codec_idx].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", &asrc_fmt);
                priv->asrc_format = (__force snd_pcm_format_t)asrc_fmt;
                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) {
                dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n");
                goto asrc_fail;
        }

        /*
         * Properties "hp-det-gpios" and "mic-det-gpios" are optional, and
         * simple_util_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_present(np, "hp-det-gpios") ||
            of_property_present(np, "hp-det-gpio") /* deprecated */) {
                ret = simple_util_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_present(np, "mic-det-gpios") ||
            of_property_present(np, "mic-det-gpio") /* deprecated */) {
                ret = simple_util_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[0]);
        of_node_put(codec_np[1]);
        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-tlv320aic31xx", },
        { .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", },
        { .compatible = "fsl,imx-audio-nau8822", },
        { .compatible = "fsl,imx-audio-wm8904", },
        { .compatible = "fsl,imx-audio-spdif", },
        {}
};
MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);

static struct platform_driver fsl_asoc_card_driver = {
        .probe = fsl_asoc_card_probe,
        .driver = {
                .name = DRIVER_NAME,
                .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:" DRIVER_NAME);
MODULE_LICENSE("GPL");