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/*
* ASoC Audio Layer for Freescale MXS ADC/DAC
*
* Author: Vladislav Buzov <vbuzov@embeddedalley.com>
*
* Copyright 2008-2010 Freescale Semiconductor, Inc.
* Copyright 2008 Embedded Alley Solutions, Inc All Rights Reserved.
*/
/*
* The code contained herein is licensed under the GNU General Public
* License. You may obtain a copy of the GNU General Public License
* Version 2 or later at the following locations:
*
* http://www.opensource.org/licenses/gpl-license.html
* http://www.gnu.org/copyleft/gpl.html
*/
#include <linux/module.h>
#include <linux/init.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <linux/dma-mapping.h>
#include <linux/clk.h>
#include <linux/delay.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <mach/dma.h>
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <mach/regs-audioin.h>
#include <mach/regs-audioout.h>
#include "mxs-pcm.h"
#define MXS_ADC_RATES SNDRV_PCM_RATE_8000_192000
#define MXS_ADC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S32_LE)
struct mxs_pcm_dma_params mxs_audio_in = {
.name = "mxs-audio-in",
.dma_ch = MXS_DMA_CHANNEL_AHB_APBX_AUDIOADC,
.irq = IRQ_ADC_DMA,
};
struct mxs_pcm_dma_params mxs_audio_out = {
.name = "mxs-audio-out",
.dma_ch = MXS_DMA_CHANNEL_AHB_APBX_AUDIODAC,
.irq = IRQ_DAC_DMA,
};
static irqreturn_t mxs_err_irq(int irq, void *dev_id)
{
struct snd_pcm_substream *substream = dev_id;
int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
u32 ctrl_reg;
u32 overflow_mask;
u32 underflow_mask;
if (playback) {
ctrl_reg = __raw_readl(REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL);
underflow_mask = BM_AUDIOOUT_CTRL_FIFO_UNDERFLOW_IRQ;
overflow_mask = BM_AUDIOOUT_CTRL_FIFO_OVERFLOW_IRQ;
} else {
ctrl_reg = __raw_readl(REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL);
underflow_mask = BM_AUDIOIN_CTRL_FIFO_UNDERFLOW_IRQ;
overflow_mask = BM_AUDIOIN_CTRL_FIFO_OVERFLOW_IRQ;
}
if (ctrl_reg & underflow_mask) {
printk(KERN_DEBUG "%s underflow detected\n",
playback ? "DAC" : "ADC");
if (playback)
__raw_writel(
BM_AUDIOOUT_CTRL_FIFO_UNDERFLOW_IRQ,
REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_CLR);
else
__raw_writel(
BM_AUDIOIN_CTRL_FIFO_UNDERFLOW_IRQ,
REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_CLR);
} else if (ctrl_reg & overflow_mask) {
printk(KERN_DEBUG "%s overflow detected\n",
playback ? "DAC" : "ADC");
if (playback)
__raw_writel(
BM_AUDIOOUT_CTRL_FIFO_OVERFLOW_IRQ,
REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_CLR);
else
__raw_writel(BM_AUDIOIN_CTRL_FIFO_OVERFLOW_IRQ,
REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_CLR);
} else
printk(KERN_WARNING "Unknown DAC error interrupt\n");
return IRQ_HANDLED;
}
static int mxs_adc_trigger(struct snd_pcm_substream *substream,
int cmd,
struct snd_soc_dai *dai)
{
int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
int ret = 0;
u32 reg = 0;
u32 reg1 = 0;
u32 l, r;
u32 ll, rr;
int i;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
if (playback) {
reg = __raw_readl(REGS_AUDIOOUT_BASE + \
HW_AUDIOOUT_HPVOL);
reg1 = BM_AUDIOOUT_HPVOL_VOL_LEFT | \
BM_AUDIOOUT_HPVOL_VOL_RIGHT;
__raw_writel(reg1, REGS_AUDIOOUT_BASE + \
HW_AUDIOOUT_HPVOL);
__raw_writel(BM_AUDIOOUT_CTRL_RUN,
REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_SET);
__raw_writel(BM_AUDIOOUT_ANACTRL_HP_HOLD_GND,
REGS_AUDIOOUT_BASE + HW_AUDIOOUT_ANACTRL_CLR);
reg1 = reg & ~BM_AUDIOOUT_HPVOL_VOL_LEFT;
reg1 = reg1 & ~BM_AUDIOOUT_HPVOL_VOL_RIGHT;
l = (reg & BM_AUDIOOUT_HPVOL_VOL_LEFT) >>
BP_AUDIOOUT_HPVOL_VOL_LEFT;
r = (reg & BM_AUDIOOUT_HPVOL_VOL_RIGHT) >>
BP_AUDIOOUT_HPVOL_VOL_RIGHT;
for (i = 0x7f; i > 0 ; i -= 0x8) {
ll = i > l ? i : l;
rr = i > r ? i : r;
/* fade in hp vol */
reg = reg1 | BF_AUDIOOUT_HPVOL_VOL_LEFT(ll)
| BF_AUDIOOUT_HPVOL_VOL_RIGHT(rr);
__raw_writel(reg,
REGS_AUDIOOUT_BASE + HW_AUDIOOUT_HPVOL);
udelay(100);
}
__raw_writel(BM_AUDIOOUT_SPEAKERCTRL_MUTE,
REGS_AUDIOIN_BASE + HW_AUDIOOUT_SPEAKERCTRL_CLR);
}
else
__raw_writel(BM_AUDIOIN_CTRL_RUN,
REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_SET);
break;
case SNDRV_PCM_TRIGGER_STOP:
if (playback) {
__raw_writel(BM_AUDIOOUT_ANACTRL_HP_HOLD_GND,
REGS_AUDIOOUT_BASE + HW_AUDIOOUT_ANACTRL_SET);
__raw_writel(BM_AUDIOOUT_SPEAKERCTRL_MUTE,
REGS_AUDIOOUT_BASE + HW_AUDIOOUT_SPEAKERCTRL_SET);
__raw_writel(BM_AUDIOOUT_CTRL_RUN,
REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_CLR);
}
else
__raw_writel(BM_AUDIOIN_CTRL_RUN,
REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_CLR);
break;
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
break;
default:
ret = -EINVAL;
}
return ret;
}
static int mxs_adc_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
int irq;
int ret;
if (playback) {
irq = IRQ_DAC_ERROR;
cpu_dai->dma_data = &mxs_audio_out;
} else {
irq = IRQ_ADC_ERROR;
cpu_dai->dma_data = &mxs_audio_in;
}
ret = request_irq(irq, mxs_err_irq, 0, "MXS DAC/ADC Error",
substream);
if (ret) {
printk(KERN_ERR "%s: Unable to request ADC/DAC error irq %d\n",
__func__, IRQ_DAC_ERROR);
return ret;
}
/* Enable error interrupt */
if (playback) {
__raw_writel(BM_AUDIOOUT_CTRL_FIFO_OVERFLOW_IRQ,
REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_CLR);
__raw_writel(BM_AUDIOOUT_CTRL_FIFO_UNDERFLOW_IRQ,
REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_CLR);
__raw_writel(BM_AUDIOOUT_CTRL_FIFO_ERROR_IRQ_EN,
REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_SET);
} else {
__raw_writel(BM_AUDIOIN_CTRL_FIFO_OVERFLOW_IRQ,
REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_CLR);
__raw_writel(BM_AUDIOIN_CTRL_FIFO_UNDERFLOW_IRQ,
REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_CLR);
__raw_writel(BM_AUDIOIN_CTRL_FIFO_ERROR_IRQ_EN,
REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_SET);
}
return 0;
}
static void mxs_adc_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
/* Disable error interrupt */
if (playback) {
__raw_writel(BM_AUDIOOUT_CTRL_FIFO_ERROR_IRQ_EN,
REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_CLR);
free_irq(IRQ_DAC_ERROR, substream);
} else {
__raw_writel(BM_AUDIOIN_CTRL_FIFO_ERROR_IRQ_EN,
REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_CLR);
free_irq(IRQ_ADC_ERROR, substream);
}
}
#ifdef CONFIG_PM
static int mxs_adc_suspend(struct snd_soc_dai *cpu_dai)
{
return 0;
}
static int mxs_adc_resume(struct snd_soc_dai *cpu_dai)
{
return 0;
}
#else
#define mxs_adc_suspend NULL
#define mxs_adc_resume NULL
#endif /* CONFIG_PM */
struct snd_soc_dai_ops mxs_adc_dai_ops = {
.startup = mxs_adc_startup,
.shutdown = mxs_adc_shutdown,
.trigger = mxs_adc_trigger,
};
struct snd_soc_dai mxs_adc_dai = {
.name = "mxs adc/dac",
.id = 0,
.suspend = mxs_adc_suspend,
.resume = mxs_adc_resume,
.playback = {
.channels_min = 2,
.channels_max = 2,
.rates = MXS_ADC_RATES,
.formats = MXS_ADC_FORMATS,
},
.capture = {
.channels_min = 2,
.channels_max = 2,
.rates = MXS_ADC_RATES,
.formats = MXS_ADC_FORMATS,
},
.ops = &mxs_adc_dai_ops,
};
EXPORT_SYMBOL_GPL(mxs_adc_dai);
static int __init mxs_adc_dai_init(void)
{
return snd_soc_register_dai(&mxs_adc_dai);
}
static void __exit mxs_adc_dai_exit(void)
{
snd_soc_unregister_dai(&mxs_adc_dai);
}
module_init(mxs_adc_dai_init);
module_exit(mxs_adc_dai_exit);
MODULE_AUTHOR("Vladislav Buzov");
MODULE_DESCRIPTION("MXS ADC/DAC DAI");
MODULE_LICENSE("GPL");
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