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h1940_uda1380.c

/*
 * h1940-uda1380.c  --  ALSA Soc Audio Layer
 *
 * Copyright (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
 * Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com>
 *
 * Based on version from Arnaud Patard <arnaud.patard@rtp-net.org>
 *
 * 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/gpio.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;

      /* Add h1940 specific widgets */
      err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
                          ARRAY_SIZE(uda1380_dapm_widgets));
      if (err)
            return err;

      /* Set up h1940 specific audio path audio_mapnects */
      err = snd_soc_dapm_add_routes(dapm, audio_map,
                              ARRAY_SIZE(audio_map));
      if (err)
            return 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_dapm_sync(dapm);

      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",
      .dai_link = h1940_uda1380_dai,
      .num_links = ARRAY_SIZE(h1940_uda1380_dai),
};

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");

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