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/*
* 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 <sound/soc-dapm.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->dai->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
int err;
/* Set codec DAI configuration */
err = snd_soc_dai_set_fmt(codec_dai,
SND_SOC_DAIFMT_I2S|
SND_SOC_DAIFMT_NB_IF |
SND_SOC_DAIFMT_CBM_CFM);
if (err < 0) {
printk(KERN_ERR "can't set codec DAI configuration\n");
return err;
}
/* Set cpu DAI configuration */
err = snd_soc_dai_set_fmt(cpu_dai,
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_IF |
SND_SOC_DAIFMT_CBM_CFM);
if (err < 0) {
printk(KERN_ERR "can't set cpu DAI configuration\n");
return 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 audio_map[] = {
{"Headphone Jack", NULL, "LHPOUT"},
{"Headphone Jack", NULL, "RHPOUT"},
{"LLINEIN", NULL, "Line In"},
{"RLINEIN", NULL, "Line In"},
{"MICIN", NULL, "Mic Jack"},
};
static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec)
{
/* Add afeb9260 specific widgets */
snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
ARRAY_SIZE(tlv320aic23_dapm_widgets));
/* Set up afeb9260 specific audio path audio_map */
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
snd_soc_dapm_enable_pin(codec, "Headphone Jack");
snd_soc_dapm_enable_pin(codec, "Line In");
snd_soc_dapm_enable_pin(codec, "Mic Jack");
snd_soc_dapm_sync(codec);
return 0;
}
/* Digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link afeb9260_dai = {
.name = "TLV320AIC23",
.stream_name = "AIC23",
.cpu_dai = &atmel_ssc_dai[0],
.codec_dai = &tlv320aic23_dai,
.init = afeb9260_tlv320aic23_init,
.ops = &afeb9260_ops,
};
/* Audio machine driver */
static struct snd_soc_card snd_soc_machine_afeb9260 = {
.name = "AFEB9260",
.platform = &atmel_soc_platform,
.dai_link = &afeb9260_dai,
.num_links = 1,
};
/* Audio subsystem */
static struct snd_soc_device afeb9260_snd_devdata = {
.card = &snd_soc_machine_afeb9260,
.codec_dev = &soc_codec_dev_tlv320aic23,
};
static struct platform_device *afeb9260_snd_device;
static int __init afeb9260_soc_init(void)
{
int err;
struct device *dev;
struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data;
struct ssc_device *ssc = NULL;
if (!(machine_is_afeb9260()))
return -ENODEV;
ssc = ssc_request(0);
if (IS_ERR(ssc)) {
printk(KERN_ERR "ASoC: Failed to request SSC 0\n");
err = PTR_ERR(ssc);
ssc = NULL;
goto err_ssc;
}
ssc_p->ssc = ssc;
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, &afeb9260_snd_devdata);
afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev;
err = platform_device_add(afeb9260_snd_device);
if (err)
goto err1;
dev = &afeb9260_snd_device->dev;
return 0;
err1:
platform_device_del(afeb9260_snd_device);
platform_device_put(afeb9260_snd_device);
err_ssc:
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");
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