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/*
* 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 <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include "ep93xx-pcm.h"
#define edb93xx_has_audio() (machine_is_edb9301() || \
machine_is_edb9302() || \
machine_is_edb9302a() || \
machine_is_edb9307a() || \
machine_is_edb9315a())
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 rate = params_rate(params);
/*
* We set LRCLK equal to `rate' and SCLK = LRCLK * 64,
* because our sample size is 32 bit * 2 channels.
* I2S standard permits us to transmit more bits than
* the codec uses.
* MCLK = SCLK * 4 is the best recommended value,
* but we have to fall back to ratio 2 for higher
* sample rates.
*/
unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2);
err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_IF |
SND_SOC_DAIFMT_CBS_CFS);
if (err)
return err;
err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_IF |
SND_SOC_DAIFMT_CBS_CFS);
if (err)
return err;
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-pcm-audio",
.cpu_dai_name = "ep93xx-i2s",
.codec_name = "spi0.0",
.codec_dai_name = "cs4271-hifi",
.ops = &edb93xx_ops,
};
static struct snd_soc_card snd_soc_edb93xx = {
.name = "EDB93XX",
.dai_link = &edb93xx_dai,
.num_links = 1,
};
static struct platform_device *edb93xx_snd_device;
static int __init edb93xx_init(void)
{
int ret;
if (!edb93xx_has_audio())
return -ENODEV;
ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97,
EP93XX_SYSCON_I2SCLKDIV_ORIDE |
EP93XX_SYSCON_I2SCLKDIV_SPOL);
if (ret)
return ret;
edb93xx_snd_device = platform_device_alloc("soc-audio", -1);
if (!edb93xx_snd_device) {
ret = -ENOMEM;
goto free_i2s;
}
platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx);
ret = platform_device_add(edb93xx_snd_device);
if (ret)
goto device_put;
return 0;
device_put:
platform_device_put(edb93xx_snd_device);
free_i2s:
ep93xx_i2s_release();
return ret;
}
module_init(edb93xx_init);
static void __exit edb93xx_exit(void)
{
platform_device_unregister(edb93xx_snd_device);
ep93xx_i2s_release();
}
module_exit(edb93xx_exit);
MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>");
MODULE_DESCRIPTION("ALSA SoC EDB93xx");
MODULE_LICENSE("GPL");
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