blob: 517963ef484726676e4a0aa15e715ec6c7cd49f8 [file] [log] [blame]
/*
* SoC audio for EDB93xx
*
* Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
*
* 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.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* This driver support CS4271 codec being master or slave, working
* in control port mode, connected either via SPI or I2C.
* The data format accepted is I2S or left-justified.
* DAPM support not implemented.
*/
#include <linux/platform_device.h>
#include <linux/gpio.h>
#include <linux/module.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <asm/mach-types.h>
#include <mach/hardware.h>
static int edb93xx_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 *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int err;
unsigned int mclk_rate;
unsigned int rate = params_rate(params);
/*
* According to CS4271 datasheet we use MCLK/LRCK=256 for
* rates below 50kHz and 128 for higher sample rates
*/
if (rate < 50000)
mclk_rate = rate * 64 * 4;
else
mclk_rate = rate * 64 * 2;
err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate,
SND_SOC_CLOCK_IN);
if (err)
return err;
return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate,
SND_SOC_CLOCK_OUT);
}
static struct snd_soc_ops edb93xx_ops = {
.hw_params = edb93xx_hw_params,
};
static struct snd_soc_dai_link edb93xx_dai = {
.name = "CS4271",
.stream_name = "CS4271 HiFi",
.platform_name = "ep93xx-i2s",
.cpu_dai_name = "ep93xx-i2s",
.codec_name = "spi0.0",
.codec_dai_name = "cs4271-hifi",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
.ops = &edb93xx_ops,
};
static struct snd_soc_card snd_soc_edb93xx = {
.name = "EDB93XX",
.owner = THIS_MODULE,
.dai_link = &edb93xx_dai,
.num_links = 1,
};
static int edb93xx_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &snd_soc_edb93xx;
int ret;
ret = ep93xx_i2s_acquire();
if (ret)
return ret;
card->dev = &pdev->dev;
ret = snd_soc_register_card(card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
ep93xx_i2s_release();
}
return ret;
}
static int edb93xx_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
snd_soc_unregister_card(card);
ep93xx_i2s_release();
return 0;
}
static struct platform_driver edb93xx_driver = {
.driver = {
.name = "edb93xx-audio",
},
.probe = edb93xx_probe,
.remove = edb93xx_remove,
};
module_platform_driver(edb93xx_driver);
MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>");
MODULE_DESCRIPTION("ALSA SoC EDB93xx");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:edb93xx-audio");