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/*
* 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/types.h>
#include <linux/gpio.h>
#include <linux/module.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include "regs-iis.h"
#include <asm/mach-types.h>
#include <mach/gpio-samsung.h>
#include "s3c24xx-i2s.h"
static const unsigned int rates[] = {
11025,
22050,
44100,
};
static const struct snd_pcm_hw_constraint_list hw_rates = {
.count = ARRAY_SIZE(rates),
.list = rates,
};
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;
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;
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;
}
/* 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(S3C_GPIO_END + 9, 1);
else
gpio_set_value(S3C_GPIO_END + 9, 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)
{
snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE,
&hp_jack, hp_jack_pins, ARRAY_SIZE(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 = "s3c24xx-iis",
.codec_name = "uda1380-codec.0-001a",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
.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(S3C_GPIO_END + 9, "speaker-power");
if (ret)
goto err_out;
ret = gpio_direction_output(S3C_GPIO_END + 9, 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(S3C_GPIO_END + 9);
err_out:
return ret;
}
static void __exit h1940_exit(void)
{
platform_device_unregister(s3c24xx_snd_device);
gpio_free(S3C_GPIO_END + 9);
}
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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