blob: 9579799ace5469af7bebbc0c320716fb92d5b433 [file] [log] [blame]
/*
* afeb9260.c -- SoC audio for AFEB9260
*
* Copyright (C) 2009 Sergey Lapin <slapin@ossfans.org>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* version 2 as published by the Free Software Foundation.
*
* 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.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
* 02110-1301 USA
*
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/kernel.h>
#include <linux/clk.h>
#include <linux/platform_device.h>
#include <linux/atmel-ssc.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <linux/gpio.h>
#include "../codecs/tlv320aic23.h"
#include "atmel-pcm.h"
#include "atmel_ssc_dai.h"
#define CODEC_CLOCK 12000000
static int afeb9260_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;
int err;
/* Set the codec system clock for DAC and ADC */
err =
snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
if (err < 0) {
printk(KERN_ERR "can't set codec system clock\n");
return err;
}
return err;
}
static struct snd_soc_ops afeb9260_ops = {
.hw_params = afeb9260_hw_params,
};
static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_LINE("Line In", NULL),
SND_SOC_DAPM_MIC("Mic Jack", NULL),
};
static const struct snd_soc_dapm_route afeb9260_audio_map[] = {
{"Headphone Jack", NULL, "LHPOUT"},
{"Headphone Jack", NULL, "RHPOUT"},
{"LLINEIN", NULL, "Line In"},
{"RLINEIN", NULL, "Line In"},
{"MICIN", NULL, "Mic Jack"},
};
/* Digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link afeb9260_dai = {
.name = "TLV320AIC23",
.stream_name = "AIC23",
.cpu_dai_name = "atmel-ssc-dai.0",
.codec_dai_name = "tlv320aic23-hifi",
.platform_name = "atmel_pcm-audio",
.codec_name = "tlv320aic23-codec.0-001a",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |
SND_SOC_DAIFMT_CBM_CFM,
.ops = &afeb9260_ops,
};
/* Audio machine driver */
static struct snd_soc_card snd_soc_machine_afeb9260 = {
.name = "AFEB9260",
.owner = THIS_MODULE,
.dai_link = &afeb9260_dai,
.num_links = 1,
.dapm_widgets = tlv320aic23_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
.dapm_routes = afeb9260_audio_map,
.num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map),
};
static struct platform_device *afeb9260_snd_device;
static int __init afeb9260_soc_init(void)
{
int err;
struct device *dev;
if (!(machine_is_afeb9260()))
return -ENODEV;
afeb9260_snd_device = platform_device_alloc("soc-audio", -1);
if (!afeb9260_snd_device) {
printk(KERN_ERR "ASoC: Platform device allocation failed\n");
return -ENOMEM;
}
platform_set_drvdata(afeb9260_snd_device, &snd_soc_machine_afeb9260);
err = platform_device_add(afeb9260_snd_device);
if (err)
goto err1;
dev = &afeb9260_snd_device->dev;
return 0;
err1:
platform_device_put(afeb9260_snd_device);
return err;
}
static void __exit afeb9260_soc_exit(void)
{
platform_device_unregister(afeb9260_snd_device);
}
module_init(afeb9260_soc_init);
module_exit(afeb9260_soc_exit);
MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>");
MODULE_DESCRIPTION("ALSA SoC for AFEB9260");
MODULE_LICENSE("GPL");