mirror of
https://github.com/AuxXxilium/linux_dsm_epyc7002.git
synced 2024-12-05 07:06:59 +07:00
bc6c117ef0
Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
164 lines
4.1 KiB
C
164 lines
4.1 KiB
C
/*
|
|
* 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"},
|
|
};
|
|
|
|
static int afeb9260_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
|
|
{
|
|
struct snd_soc_codec *codec = rtd->codec;
|
|
struct snd_soc_dapm_context *dapm = &codec->dapm;
|
|
|
|
snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
|
|
snd_soc_dapm_enable_pin(dapm, "Line In");
|
|
snd_soc_dapm_enable_pin(dapm, "Mic Jack");
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* 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",
|
|
.init = afeb9260_tlv320aic23_init,
|
|
.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");
|
|
|