#include <linux/types.h>
#include <linux/module.h>

#include <sound/soc.h>

#include <mc_pcm.h>

static int mc_tlv320_init(struct snd_soc_pcm_runtime *rtd);
static int mc_tlv320_startup(struct snd_pcm_substream *substream);
static int mc_tlv320_hw_params(struct snd_pcm_substream *substream,
                struct snd_pcm_hw_params *params);

static unsigned int rates[] = {
    5512,
    8000,
    11025,
    16000,
    22050,
    32000,
    44100,
    48000,
    64000,
    88200,
    96000,
    176400,
    192000
};

static struct snd_pcm_hw_constraint_list hw_rates = {
    .count = ARRAY_SIZE(rates),
    .list = rates,
    .mask = 0,
};

static struct snd_soc_ops mc_tlv320_ops = {
    .startup    = mc_tlv320_startup,
    .hw_params  = mc_tlv320_hw_params,
};

/* Multicore digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link mc_tlv320_dai[] = {
    {
        .name           = "TLV320AIC3X",
        .stream_name    = "AIC3X",
        .cpu_dai_name   = "mc_i2s.3",
        .codec_dai_name = "tlv320aic3x-hifi",
        .init           = mc_tlv320_init,
        .platform_name  = "multicore-audio.3",
        .codec_name     = "tlv320aic3x-codec.0-0018",
        .ops            = &mc_tlv320_ops,
    },
};

static struct snd_soc_card mc_asoc = {
    .name = "mc_audio",
    .owner = THIS_MODULE,
    .dai_link = mc_tlv320_dai,
    .num_links = ARRAY_SIZE(mc_tlv320_dai),
};

static struct platform_device *mc_snd_device;

static int mc_tlv320_startup(struct snd_pcm_substream *substream)
{
    struct snd_pcm_runtime *runtime = substream->runtime;

    runtime->hw.rate_min = hw_rates.list[0];
    runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
    runtime->hw.rates = MC_PCM_RATES;

    return snd_pcm_hw_constraint_list(runtime, 0,
                    SNDRV_PCM_HW_PARAM_RATE,
                    &hw_rates);
}

static int mc_tlv320_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;
    int ret;

    /* set codec DAI configuration */
    ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
        SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
    if (ret < 0)
        return ret;

    /* set cpu DAI configuration */
    ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
        SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS |
        SND_SOC_DAIFMT_CONT);
    if (ret < 0)
        return ret;
        
	/* set the codec system clock */
	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 10000000, SND_SOC_CLOCK_OUT);
	if (ret < 0)
		return ret;
        
    /*
    ret = snd_soc_dapm_disable_pin(codec_dai, "MONO_LOUT");
	if (ret < 0)
		return ret;
    */

    return 0;
}

/* davinci-evm machine dapm widgets */
static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
	SND_SOC_DAPM_HP("Headphone Jack", NULL),
	SND_SOC_DAPM_LINE("Line Out", NULL),
	SND_SOC_DAPM_MIC("Mic Jack", NULL),
	SND_SOC_DAPM_LINE("Line In", NULL),
};

/* davinci-evm machine audio_mapnections to the codec pins */
static const struct snd_soc_dapm_route audio_map[] = {
	/* Headphone connected to HPLOUT, HPROUT */
	{"Headphone Jack", NULL, "HPLOUT"},
	{"Headphone Jack", NULL, "HPROUT"},

	/* Line Out connected to LLOUT, RLOUT */
	{"Line Out", NULL, "LLOUT"},
	{"Line Out", NULL, "RLOUT"},

	/* Mic connected to (MIC3L | MIC3R) */
	{"MIC3L", NULL, "Mic Bias 2V"},
	{"MIC3R", NULL, "Mic Bias 2V"},
	{"Mic Bias 2V", NULL, "Mic Jack"},

	/* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */
	{"LINE1L", NULL, "Line In"},
	{"LINE2L", NULL, "Line In"},
	{"LINE1R", NULL, "Line In"},
	{"LINE2R", NULL, "Line In"},
};

static int mc_tlv320_init(struct snd_soc_pcm_runtime *rtd)
{
	struct snd_soc_codec *codec = rtd->codec;
	struct snd_soc_dapm_context *dapm = &codec->dapm;

	/* Add davinci-evm specific widgets */
	snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets,
				  ARRAY_SIZE(aic3x_dapm_widgets));

	/* Set up davinci-evm specific audio path audio_map */
	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));

	/* not connected */
	snd_soc_dapm_disable_pin(dapm, "MONO_LOUT");
	snd_soc_dapm_disable_pin(dapm, "HPLCOM");
	snd_soc_dapm_disable_pin(dapm, "HPRCOM");

	/* always connected */
	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
	snd_soc_dapm_enable_pin(dapm, "Line Out");
	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
	snd_soc_dapm_enable_pin(dapm, "Line In");
    
    return 0;
}

static int __init mc_audio_init(void)
{
    int ret;

    mc_snd_device = platform_device_alloc("soc-audio", -1);
    if (!mc_snd_device)
        return -ENOMEM;

    platform_set_drvdata(mc_snd_device, &mc_asoc);
    ret = platform_device_add(mc_snd_device);

    if (ret) {
        platform_device_put(mc_snd_device);
        return ret;
    }

    return 0;
}

static void __exit mc_audio_exit(void)
{
    platform_device_unregister(mc_snd_device);
}

late_initcall(mc_audio_init);
module_exit(mc_audio_exit);

MODULE_AUTHOR("Dmitry Podkhvatilin");
MODULE_DESCRIPTION("ALSA SoC MULTICORE");
MODULE_LICENSE("GPL");