android_kernel_motorola_sm6225/sound/soc/s3c24xx/jive_wm8750.c
Ben Dooks dc85447b19 ASoC: Split s3c2412-i2s.c into core and SoC specific parts
The S3C2412 I2S (IIS) interface is replicated on further Samsung SoC
parts in a broadly compatible way, so split the common code out into
a core called s3c-i2s-v2.[ch] so that the newer SoCs such as the
S3C6410 can make use of it.

As such, all the original s3c2412 functions are currently being left
with their original names, and will be renamed later in the series.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-06 13:37:08 +00:00

216 lines
5.1 KiB
C

/* sound/soc/s3c24xx/jive_wm8750.c
*
* Copyright 2007,2008 Simtec Electronics
*
* Based on sound/soc/pxa/spitz.c
* Copyright 2005 Wolfson Microelectronics PLC.
* Copyright 2005 Openedhand Ltd.
*
* 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.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <linux/clk.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
#include "s3c24xx-pcm.h"
#include "s3c2412-i2s.h"
#include "../codecs/wm8750.h"
static const struct snd_soc_dapm_route audio_map[] = {
{ "Headphone Jack", NULL, "LOUT1" },
{ "Headphone Jack", NULL, "ROUT1" },
{ "Internal Speaker", NULL, "LOUT2" },
{ "Internal Speaker", NULL, "ROUT2" },
{ "LINPUT1", NULL, "Line Input" },
{ "RINPUT1", NULL, "Line Input" },
};
static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_SPK("Internal Speaker", NULL),
SND_SOC_DAPM_LINE("Line In", NULL),
};
static int jive_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->socdev->codec;
snd_soc_dapm_enable_pin(codec, "Headphone Jack");
snd_soc_dapm_enable_pin(codec, "Internal Speaker");
snd_soc_dapm_enable_pin(codec, "Line In");
snd_soc_dapm_sync(codec);
return 0;
}
static int jive_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;
struct s3c_i2sv2_rate_calc div;
unsigned int clk = 0;
int ret = 0;
switch (params_rate(params)) {
case 8000:
case 16000:
case 48000:
case 96000:
clk = 12288000;
break;
case 11025:
case 22050:
case 44100:
clk = 11289600;
break;
}
s3c_i2sv2_calc_rate(&div, NULL, params_rate(params),
s3c2412_get_iisclk());
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set the codec system clock for DAC and ADC */
ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER,
div.clk_div - 1);
if (ret < 0)
return ret;
return 0;
}
static struct snd_soc_ops jive_ops = {
.startup = jive_startup,
.hw_params = jive_hw_params,
};
static int jive_wm8750_init(struct snd_soc_codec *codec)
{
int err;
/* These endpoints are not being used. */
snd_soc_dapm_disable_pin(codec, "LINPUT2");
snd_soc_dapm_disable_pin(codec, "RINPUT2");
snd_soc_dapm_disable_pin(codec, "LINPUT3");
snd_soc_dapm_disable_pin(codec, "RINPUT3");
snd_soc_dapm_disable_pin(codec, "OUT3");
snd_soc_dapm_disable_pin(codec, "MONO");
/* Add jive specific widgets */
err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
ARRAY_SIZE(wm8750_dapm_widgets));
if (err) {
printk(KERN_ERR "%s: failed to add widgets (%d)\n",
__func__, err);
return err;
}
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
snd_soc_dapm_sync(codec);
return 0;
}
static struct snd_soc_dai_link jive_dai = {
.name = "wm8750",
.stream_name = "WM8750",
.cpu_dai = &s3c2412_i2s_dai,
.codec_dai = &wm8750_dai,
.init = jive_wm8750_init,
.ops = &jive_ops,
};
/* jive audio machine driver */
static struct snd_soc_machine snd_soc_machine_jive = {
.name = "Jive",
.dai_link = &jive_dai,
.num_links = 1,
};
/* jive audio private data */
static struct wm8750_setup_data jive_wm8750_setup = {
};
/* jive audio subsystem */
static struct snd_soc_device jive_snd_devdata = {
.machine = &snd_soc_machine_jive,
.platform = &s3c24xx_soc_platform,
.codec_dev = &soc_codec_dev_wm8750_spi,
.codec_data = &jive_wm8750_setup,
};
static struct platform_device *jive_snd_device;
static int __init jive_init(void)
{
int ret;
if (!machine_is_jive())
return 0;
printk("JIVE WM8750 Audio support\n");
jive_snd_device = platform_device_alloc("soc-audio", -1);
if (!jive_snd_device)
return -ENOMEM;
platform_set_drvdata(jive_snd_device, &jive_snd_devdata);
jive_snd_devdata.dev = &jive_snd_device->dev;
ret = platform_device_add(jive_snd_device);
if (ret)
platform_device_put(jive_snd_device);
return ret;
}
static void __exit jive_exit(void)
{
platform_device_unregister(jive_snd_device);
}
module_init(jive_init);
module_exit(jive_exit);
MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
MODULE_DESCRIPTION("ALSA SoC Jive Audio support");
MODULE_LICENSE("GPL");