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/*
* osk5912.c -- SoC audio for OSK 5912
*
* Copyright (C) 2008 Mistral Solutions
*
* Contact: Arun KS <arunks@mistralsolutions.com>
*
* 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/clk.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <linux/gpio.h>
#include <mach/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
#include "../codecs/tlv320aic23.h"
#define CODEC_CLOCK 12000000
static struct clk *tlv320aic23_mclk;
static int osk_startup(struct snd_pcm_substream *substream)
{
return clk_enable(tlv320aic23_mclk);
}
static void osk_shutdown(struct snd_pcm_substream *substream)
{
clk_disable(tlv320aic23_mclk);
}
static int osk_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_DSP_A |
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_DSP_A |
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 osk_ops = {
.startup = osk_startup,
.hw_params = osk_hw_params,
.shutdown = osk_shutdown,
};
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 osk_tlv320aic23_init(struct snd_soc_codec *codec)
{
/* Add osk5912 specific widgets */
snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
ARRAY_SIZE(tlv320aic23_dapm_widgets));
/* Set up osk5912 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 osk_dai = {
.name = "TLV320AIC23",
.stream_name = "AIC23",
.cpu_dai = &omap_mcbsp_dai[0],
.codec_dai = &tlv320aic23_dai,
.init = osk_tlv320aic23_init,
.ops = &osk_ops,
};
/* Audio machine driver */
static struct snd_soc_machine snd_soc_machine_osk = {
.name = "OSK5912",
.dai_link = &osk_dai,
.num_links = 1,
};
/* Audio subsystem */
static struct snd_soc_device osk_snd_devdata = {
.machine = &snd_soc_machine_osk,
.platform = &omap_soc_platform,
.codec_dev = &soc_codec_dev_tlv320aic23,
};
static struct platform_device *osk_snd_device;
static int __init osk_soc_init(void)
{
int err;
u32 curRate;
struct device *dev;
if (!(machine_is_omap_osk()))
return -ENODEV;
osk_snd_device = platform_device_alloc("soc-audio", -1);
if (!osk_snd_device)
return -ENOMEM;
platform_set_drvdata(osk_snd_device, &osk_snd_devdata);
osk_snd_devdata.dev = &osk_snd_device->dev;
*(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */
err = platform_device_add(osk_snd_device);
if (err)
goto err1;
dev = &osk_snd_device->dev;
tlv320aic23_mclk = clk_get(dev, "mclk");
if (IS_ERR(tlv320aic23_mclk)) {
printk(KERN_ERR "Could not get mclk clock\n");
return -ENODEV;
}
if (clk_get_usecount(tlv320aic23_mclk) > 0) {
/* MCLK is already in use */
printk(KERN_WARNING
"MCLK in use at %d Hz. We change it to %d Hz\n",
(uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
}
/*
* Configure 12 MHz output on MCLK.
*/
curRate = (uint) clk_get_rate(tlv320aic23_mclk);
if (curRate != CODEC_CLOCK) {
if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
err = -ECANCELED;
goto err1;
}
}
printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n",
(uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK,
clk_get_usecount(tlv320aic23_mclk));
return 0;
err1:
clk_put(tlv320aic23_mclk);
platform_device_del(osk_snd_device);
platform_device_put(osk_snd_device);
return err;
}
static void __exit osk_soc_exit(void)
{
platform_device_unregister(osk_snd_device);
}
module_init(osk_soc_init);
module_exit(osk_soc_exit);
MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
MODULE_DESCRIPTION("ALSA SoC OSK 5912");
MODULE_LICENSE("GPL");
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