h1940_uda1380.c

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/*
 * h1940-uda1380.c  --  ALSA Soc Audio Layer
 *
 * Copyright (c) 2010 Arnaud Patard <[email protected]>
 * Copyright (c) 2010 Vasily Khoruzhick <[email protected]>
 *
 * Based on version from Arnaud Patard <[email protected]>
 *
 * This program is free software; you can redistribute  it and/or modify it
 * under  the terms of  the GNU General  Public License as published by the
 * Free Software Foundation;  either version 2 of the  License, or (at your
 * option) any later version.
 *
 */

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

#include <sound/soc.h>
#include <sound/jack.h>

#include <plat/regs-iis.h>
#include <mach/h1940-latch.h>
#include <asm/mach-types.h>

#include "s3c24xx-i2s.h"

static unsigned int rates[] = {
    11025,
    22050,
    44100,
};

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

static struct snd_soc_jack hp_jack;

static struct snd_soc_jack_pin hp_jack_pins[] = {
    {
        .pin    = "Headphone Jack",
        .mask   = SND_JACK_HEADPHONE,
    },
    {
        .pin    = "Speaker",
        .mask   = SND_JACK_HEADPHONE,
        .invert = 1,
    },
};

static struct snd_soc_jack_gpio hp_jack_gpios[] = {
    {
        .gpio           = S3C2410_GPG(4),
        .name           = "hp-gpio",
        .report         = SND_JACK_HEADPHONE,
        .invert         = 1,
        .debounce_time      = 200,
    },
};

static int h1940_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 = SNDRV_PCM_RATE_KNOT;

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

static int h1940_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 div;
    int ret;
    unsigned int rate = params_rate(params);

    switch (rate) {
    case 11025:
    case 22050:
    case 44100:
        div = s3c24xx_i2s_get_clockrate() / (384 * rate);
        if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate))
            div++;
        break;
    default:
        dev_err(&rtd->dev, "%s: rate %d is not supported\n",
            __func__, rate);
        return -EINVAL;
    }

    /* 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);
    if (ret < 0)
        return ret;

    /* select clock source */
    ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
            SND_SOC_CLOCK_OUT);
    if (ret < 0)
        return ret;

    /* set MCLK division for sample rate */
    ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
        S3C2410_IISMOD_384FS);
    if (ret < 0)
        return ret;

    /* set BCLK division for sample rate */
    ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
        S3C2410_IISMOD_32FS);
    if (ret < 0)
        return ret;

    /* set prescaler division for sample rate */
    ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
        S3C24XX_PRESCALE(div, div));
    if (ret < 0)
        return ret;

    return 0;
}

static struct snd_soc_ops h1940_ops = {
    .startup    = h1940_startup,
    .hw_params  = h1940_hw_params,
};

static int h1940_spk_power(struct snd_soc_dapm_widget *w,
                struct snd_kcontrol *kcontrol, int event)
{
    if (SND_SOC_DAPM_EVENT_ON(event))
        gpio_set_value(H1940_LATCH_AUDIO_POWER, 1);
    else
        gpio_set_value(H1940_LATCH_AUDIO_POWER, 0);

    return 0;
}

/* h1940 machine dapm widgets */
static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
    SND_SOC_DAPM_HP("Headphone Jack", NULL),
    SND_SOC_DAPM_MIC("Mic Jack", NULL),
    SND_SOC_DAPM_SPK("Speaker", h1940_spk_power),
};

/* h1940 machine audio_map */
static const struct snd_soc_dapm_route audio_map[] = {
    /* headphone connected to VOUTLHP, VOUTRHP */
    {"Headphone Jack", NULL, "VOUTLHP"},
    {"Headphone Jack", NULL, "VOUTRHP"},

    /* ext speaker connected to VOUTL, VOUTR  */
    {"Speaker", NULL, "VOUTL"},
    {"Speaker", NULL, "VOUTR"},

    /* mic is connected to VINM */
    {"VINM", NULL, "Mic Jack"},
};

static struct platform_device *s3c24xx_snd_device;

static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
{
    struct snd_soc_codec *codec = rtd->codec;
    struct snd_soc_dapm_context *dapm = &codec->dapm;
    int err;

    snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
    snd_soc_dapm_enable_pin(dapm, "Speaker");
    snd_soc_dapm_enable_pin(dapm, "Mic Jack");

    snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
        &hp_jack);

    snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
        hp_jack_pins);

    snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
        hp_jack_gpios);

    return 0;
}

/* s3c24xx digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link h1940_uda1380_dai[] = {
    {
        .name       = "uda1380",
        .stream_name    = "UDA1380 Duplex",
        .cpu_dai_name   = "s3c24xx-iis",
        .codec_dai_name = "uda1380-hifi",
        .init       = h1940_uda1380_init,
        .platform_name  = "samsung-audio",
        .codec_name = "uda1380-codec.0-001a",
        .ops        = &h1940_ops,
    },
};

static struct snd_soc_card h1940_asoc = {
    .name = "h1940",
    .owner = THIS_MODULE,
    .dai_link = h1940_uda1380_dai,
    .num_links = ARRAY_SIZE(h1940_uda1380_dai),

    .dapm_widgets = uda1380_dapm_widgets,
    .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
    .dapm_routes = audio_map,
    .num_dapm_routes = ARRAY_SIZE(audio_map),
};

static int __init h1940_init(void)
{
    int ret;

    if (!machine_is_h1940())
        return -ENODEV;

    /* configure some gpios */
    ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power");
    if (ret)
        goto err_out;

    ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0);
    if (ret)
        goto err_gpio;

    s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
    if (!s3c24xx_snd_device) {
        ret = -ENOMEM;
        goto err_gpio;
    }

    platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc);
    ret = platform_device_add(s3c24xx_snd_device);

    if (ret)
        goto err_plat;

    return 0;

err_plat:
    platform_device_put(s3c24xx_snd_device);
err_gpio:
    gpio_free(H1940_LATCH_AUDIO_POWER);

err_out:
    return ret;
}

static void __exit h1940_exit(void)
{
    platform_device_unregister(s3c24xx_snd_device);
    snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
        hp_jack_gpios);
    gpio_free(H1940_LATCH_AUDIO_POWER);
}

module_init(h1940_init);
module_exit(h1940_exit);

/* Module information */
MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
MODULE_DESCRIPTION("ALSA SoC H1940");
MODULE_LICENSE("GPL");