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cyb4_linux/sound/soc/s3c/smdk6400_wm8990.c

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/*
* smdk6400_wm8990.c -- SoC audio for Neo1973
*
* Copyright 2007, 2008 Wolfson Microelectronics PLC.
* Author: Liam Girdwood
* lg@opensource.wolfsonmicro.com or linux@wolfsonmicro.com
*
* Copyright (C) 2007, Ryu Euiyoul <ryu.real@gmail.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
* Revision history
* 28th Feb 2008 Initial version.
*
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <linux/i2c.h>
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
#include <asm/hardware/scoop.h>
#include <asm/arch/regs-iis.h>
#include <asm/arch/regs-gpio.h>
#include <asm/hardware.h>
#include <asm/arch/audio.h>
#include <asm/io.h>
#include <asm/arch/spi-gpio.h>
#ifndef CONFIG_CPU_S3C6400
#include <asm/arch/regs-clock.h>
#else
#include <asm/arch/regs-s3c6400-clock.h>
#endif
#include "../codecs/wm8990.h"
#include "s3c-pcm.h"
#include "s3c-i2s.h"
/* define the scenarios */
#define SMDK6400_AUDIO_OFF 0
#define SMDK6400_CAPTURE_MIC1 3
#define SMDK6400_STEREO_TO_HEADPHONES 2
#define SMDK6400_CAPTURE_LINE_IN 1
#ifdef CONFIG_SND_DEBUG
#define s3cdbg(x...) printk(x)
#else
#define s3cdbg(x...)
#endif
/*
* TODO: - We need to work out PLL values for 256FS for every rate.
*/
static int smdk6400_hifi_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_codec_dai *codec_dai = rtd->dai->codec_dai;
struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
unsigned int pll_out = 0, bclk = 0;
int ret = 0;
unsigned int iispsr, iismod;
unsigned int prescaler = 4;
s3cdbg("Entered %s, rate = %d\n", __FUNCTION__, params_rate(params));
/*PCLK & SCLK gating enable*/
writel(readl(S3C_PCLK_GATE)|S3C_CLKCON_PCLK_IIS0, S3C_PCLK_GATE);
writel(readl(S3C_SCLK_GATE)|S3C_CLKCON_SCLK_AUDIO0, S3C_SCLK_GATE);
iismod = readl(S3C_IIS0MOD);
iismod &=~(0x3<<3);
/*Clear I2S prescaler value [13:8] and disable prescaler*/
iispsr = readl(S3C_IIS0PSR);
iispsr &=~((0x3f<<8)|(1<<15));
writel(iispsr, S3C_IIS0PSR);
s3cdbg("%s: %d , params = %d \n", __FUNCTION__, __LINE__, params_rate(params));
switch (params_rate(params)) {
case 8000:
case 16000:
case 32000:
case 64000:
writel(50332, S3C_EPLL_CON1);
writel((1<<31)|(32<<16)|(1<<8)|(3<<0) ,S3C_EPLL_CON0);
break;
case 11025:
case 22050:
case 44100:
case 88200:
writel(10398, S3C_EPLL_CON1);
writel((1<<31)|(45<<16)|(1<<8)|(3<<0) ,S3C_EPLL_CON0);
break;
case 48000:
case 96000:
writel(9961, S3C_EPLL_CON1);
writel((1<<31)|(49<<16)|(1<<8)|(3<<0) ,S3C_EPLL_CON0);
break;
default:
writel(0, S3C_EPLL_CON1);
writel((1<<31)|(128<<16)|(25<<8)|(0<<0) ,S3C_EPLL_CON0);
break;
}
s3cdbg("%s, IISCON: %x IISMOD: %x,IISFIC: %x,IISPSR: %x",
__FUNCTION__ , readl(S3C_IIS0CON), readl(S3C_IIS0MOD),
readl(S3C_IIS0FIC), readl(S3C_IIS0PSR));
while(!(__raw_readl(S3C_EPLL_CON0)&(1<<30)));
/* MUXepll : FOUTepll */
writel(readl(S3C_CLK_SRC)|S3C_CLKSRC_EPLL_CLKSEL, S3C_CLK_SRC);
/* AUDIO0 sel : FOUTepll */
writel((readl(S3C_CLK_SRC)&~(0x7<<7))|(0<<7), S3C_CLK_SRC);
/* CLK_DIV2 setting */
writel(0x0,S3C_CLK_DIV2);
iismod |= S3C_IIS0MOD_256FS;
/* WM8990 can use 256 FS */
switch (params_rate(params)) {
case 8000:
bclk = WM8990_BCLK_DIV_8;
pll_out = 2048000;
prescaler = 24;
break;
case 11025:
bclk = WM8990_BCLK_DIV_8;
pll_out = 2822400;
prescaler = 24;
break;
case 16000:
bclk = WM8990_BCLK_DIV_8;
pll_out = 4096000;
prescaler = 12;
break;
case 22050:
bclk = WM8990_BCLK_DIV_8;
pll_out = 5644800;
prescaler = 12;
break;
case 32000:
bclk = WM8990_BCLK_DIV_8;
pll_out = 8192000;
prescaler = 6;
break;
case 44100:
bclk = WM8990_BCLK_DIV_8;
pll_out = 11289600;
prescaler = 6;
break;
case 48000:
bclk = WM8990_BCLK_DIV_8;
pll_out = 12288000;
prescaler = 6;
break;
case 88200:
bclk = WM8990_BCLK_DIV_8;
pll_out = 22579200;
prescaler = 3;
break;
case 96000:
bclk = WM8990_BCLK_DIV_8;
pll_out = 24576000;
prescaler = 3;
break;
default:
/* somtimes 32000 rate comes to 96000
default values are same as 32000 */
iismod |= S3C_IIS0MOD_384FS;
bclk = WM8990_BCLK_DIV_2;
pll_out = 12288000;
break;
}
prescaler = prescaler - 1;
writel(iismod , S3C_IIS0MOD);
/* set codec DAI configuration */
ret = codec_dai->dai_ops.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 = cpu_dai->dai_ops.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 = codec_dai->dai_ops.set_sysclk(codec_dai, WM8990_MCLK, pll_out,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
/* set MCLK division for sample rate */
ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
S3C2410_IISMOD_32FS );
if (ret < 0)
return ret;
#if 0 /* not needed for slave */
/* set codec BCLK division for sample rate */
ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8990_BCLK_DIV, bclk);
if (ret < 0)
return ret;
#endif
/* set prescaler division for sample rate */
ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
(prescaler << 0x8));
if (ret < 0)
return ret;
return 0;
}
/*
* Neo1973 WM8990 HiFi DAI opserations.
*/
static struct snd_soc_ops smdk6400_hifi_ops = {
.hw_params = smdk6400_hifi_hw_params,
};
static int smdk6400_scenario = 0;
static int smdk6400_get_scenario(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = smdk6400_scenario;
return 0;
}
static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
{
smdk6400_scenario = scenario;
switch (smdk6400_scenario) {
case SMDK6400_AUDIO_OFF:
snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
snd_soc_dapm_set_endpoint(codec, "Mic1 Jack", 0);
snd_soc_dapm_set_endpoint(codec, "Line In Jack", 0);
break;
case SMDK6400_STEREO_TO_HEADPHONES:
snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
snd_soc_dapm_set_endpoint(codec, "Mic1 Jack", 0);
snd_soc_dapm_set_endpoint(codec, "Line In Jack", 0);
break;
case SMDK6400_CAPTURE_MIC1:
snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
snd_soc_dapm_set_endpoint(codec, "Mic1 Jack", 1);
snd_soc_dapm_set_endpoint(codec, "Line In Jack", 0);
break;
case SMDK6400_CAPTURE_LINE_IN:
snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
snd_soc_dapm_set_endpoint(codec, "Mic1 Jack", 0);
snd_soc_dapm_set_endpoint(codec, "Line In Jack", 1);
break;
default:
snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
snd_soc_dapm_set_endpoint(codec, "Mic1 Jack", 1);
snd_soc_dapm_set_endpoint(codec, "Line In Jack", 1);
break;
}
snd_soc_dapm_sync_endpoints(codec);
return 0;
}
static int smdk6400_set_scenario(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
if (smdk6400_scenario == ucontrol->value.integer.value[0])
return 0;
set_scenario_endpoints(codec, ucontrol->value.integer.value[0]);
return 1;
}
static const struct snd_soc_dapm_widget wm8990_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Mic1 Jack", NULL),
SND_SOC_DAPM_LINE("Line In Jack", NULL),
};
/* example machine audio_mapnections */
static const char* audio_map[][3] = {
/* no irq jack detect */
{"Headphone Jack", NULL, "LOUT"},
{"Headphone Jack", NULL, "ROUT"},
/* mic is connected to LIN1 and LIN2 - with bias */
{"LIN1", NULL, "Mic1 Jack"},
{"LIN2", NULL, "Mic1 Jack"},
{"RIN2", NULL, "Line In Jack"},
{NULL, NULL, NULL},
};
static const char *smdk_scenarios[] = {
"Off",
"Capture Line In",
"Headphones",
"Capture Mic1",
};
static const struct soc_enum smdk_scenario_enum[] = {
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(smdk_scenarios),smdk_scenarios),
};
static const struct snd_kcontrol_new wm8990_smdk6400_controls[] = {
SOC_ENUM_EXT("SMDK Mode", smdk_scenario_enum[0],
smdk6400_get_scenario, smdk6400_set_scenario),
};
/*
* This is an example machine initialisation for a wm8990 connected to a
* smdk6400. It is missing logic to detect hp/mic insertions and logic
* to re-route the audio in such an event.
*/
static int smdk6400_wm8990_init(struct snd_soc_codec *codec)
{
int i, err;
/* Add smdk6400 specific widgets */
for (i = 0; i < ARRAY_SIZE(wm8990_dapm_widgets); i++)
snd_soc_dapm_new_control(codec, &wm8990_dapm_widgets[i]);
/* add smdk6400 specific controls */
for (i = 0; i < ARRAY_SIZE(wm8990_smdk6400_controls); i++) {
err = snd_ctl_add(codec->card,
snd_soc_cnew(&wm8990_smdk6400_controls[i],
codec, NULL));
if (err < 0)
return err;
}
/* set up smdk6400 specific audio paths */
for (i = 0; audio_map[i][0] != NULL; i++) {
snd_soc_dapm_connect_input(codec, audio_map[i][0],
audio_map[i][1], audio_map[i][2]);
}
/* not connected */
snd_soc_dapm_set_endpoint(codec, "RIN1", 0);
snd_soc_dapm_set_endpoint(codec, "LIN3", 0);
snd_soc_dapm_set_endpoint(codec, "LIN4", 0);
snd_soc_dapm_set_endpoint(codec, "RIN3", 0);
snd_soc_dapm_set_endpoint(codec, "RIN4", 0);
snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
snd_soc_dapm_set_endpoint(codec, "OUT4", 0);
snd_soc_dapm_set_endpoint(codec, "SPKP", 0);
snd_soc_dapm_set_endpoint(codec, "SPKN", 0);
snd_soc_dapm_set_endpoint(codec, "ROP", 0);
snd_soc_dapm_set_endpoint(codec, "RON", 0);
snd_soc_dapm_set_endpoint(codec, "LOP", 0);
snd_soc_dapm_set_endpoint(codec, "LON", 0);
/* set endpoints to default mode & sync with DAPM */
set_scenario_endpoints(codec, SMDK6400_STEREO_TO_HEADPHONES);
return 0;
}
static struct snd_soc_dai_link smdk6400_dai[] = {
{ /* Hifi Playback - for similatious use with voice below */
.name = "WM8990",
.stream_name = "WM8990 HiFi",
.cpu_dai = &s3c_i2s_dai,
.codec_dai = &wm8990_dai,
.init = smdk6400_wm8990_init,
.ops = &smdk6400_hifi_ops,
},
};
static struct snd_soc_machine smdk6400 = {
.name = "smdk6400",
.dai_link = smdk6400_dai,
.num_links = ARRAY_SIZE(smdk6400_dai),
};
static struct wm8990_setup_data smdk6400_wm8990_setup = {
.i2c_address = 0x1a,
};
static struct snd_soc_device smdk6400_snd_devdata = {
.machine = &smdk6400,
.platform = &s3c24xx_soc_platform,
.codec_dev = &soc_codec_dev_wm8990,
.codec_data = &smdk6400_wm8990_setup,
};
static struct platform_device *smdk6400_snd_device;
static int __init smdk6400_init(void)
{
int ret;
smdk6400_snd_device = platform_device_alloc("soc-audio", -1);
if (!smdk6400_snd_device)
return -ENOMEM;
platform_set_drvdata(smdk6400_snd_device, &smdk6400_snd_devdata);
smdk6400_snd_devdata.dev = &smdk6400_snd_device->dev;
ret = platform_device_add(smdk6400_snd_device);
if (ret)
platform_device_put(smdk6400_snd_device);
return ret;
}
static void __exit smdk6400_exit(void)
{
platform_device_unregister(smdk6400_snd_device);
}
module_init(smdk6400_init);
module_exit(smdk6400_exit);
/* Module information */
MODULE_AUTHOR("Liam Girdwood");
MODULE_DESCRIPTION("ALSA SoC WM8990 SMDK6400");
MODULE_LICENSE("GPL");
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