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/*
* ASoC Audio Layer for Freescale STMP37XX/STMP378X ADC/DAC
*
* Author: Vladislav Buzov <vbuzov@embeddedalley.com>
*
* Copyright 2008-2009 Freescale Semiconductor, Inc. All Rights Reserved.
* 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/init.h>
#include <linux/module.h>
#include <linux/device.h>
#include <linux/delay.h>
#include <linux/interrupt.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include <asm/dma.h>
#include <mach/regs-audioin.h>
#include <mach/regs-audioout.h>
#include "stmp3xxx_pcm.h"
#define STMP3XXX_ADC_RATES SNDRV_PCM_RATE_8000_192000
#define STMP3XXX_ADC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S32_LE)
struct stmp3xxx_pcm_dma_params stmp3xxx_audio_in = {
.name = "stmp3xxx adc",
.dma_bus = STMP3XXX_BUS_APBX,
.dma_ch = 0,
.irq = IRQ_ADC_DMA,
};
struct stmp3xxx_pcm_dma_params stmp3xxx_audio_out = {
.name = "stmp3xxx dac",
.dma_bus = STMP3XXX_BUS_APBX,
.dma_ch = 1,
.irq = IRQ_DAC_DMA,
};
static irqreturn_t stmp3xxx_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 = HW_AUDIOOUT_CTRL_RD();
underflow_mask = BM_AUDIOOUT_CTRL_FIFO_UNDERFLOW_IRQ;
overflow_mask = BM_AUDIOOUT_CTRL_FIFO_OVERFLOW_IRQ;
} else {
ctrl_reg = HW_AUDIOIN_CTRL_RD();
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)
HW_AUDIOOUT_CTRL_CLR(
BM_AUDIOOUT_CTRL_FIFO_UNDERFLOW_IRQ);
else
HW_AUDIOIN_CTRL_CLR(
BM_AUDIOIN_CTRL_FIFO_UNDERFLOW_IRQ);
} else if (ctrl_reg & overflow_mask) {
printk(KERN_DEBUG "%s overflow detected\n",
playback ? "DAC" : "ADC");
if (playback)
HW_AUDIOOUT_CTRL_CLR(
BM_AUDIOOUT_CTRL_FIFO_OVERFLOW_IRQ);
else
HW_AUDIOIN_CTRL_CLR(BM_AUDIOIN_CTRL_FIFO_OVERFLOW_IRQ);
} else
printk(KERN_WARNING "Unknown DAC error interrupt\n");
return IRQ_HANDLED;
}
static int stmp3xxx_adc_trigger(struct snd_pcm_substream *substream, int cmd)
{
int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
int ret = 0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
if (playback)
HW_AUDIOOUT_CTRL_SET(BM_AUDIOOUT_CTRL_RUN);
else
HW_AUDIOIN_CTRL_SET(BM_AUDIOIN_CTRL_RUN);
break;
case SNDRV_PCM_TRIGGER_STOP:
if (playback)
HW_AUDIOOUT_CTRL_CLR(BM_AUDIOOUT_CTRL_RUN);
else
HW_AUDIOIN_CTRL_CLR(BM_AUDIOIN_CTRL_RUN);
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 stmp3xxx_adc_startup(struct snd_pcm_substream *substream)
{
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 = &stmp3xxx_audio_out;
} else {
irq = IRQ_ADC_ERROR;
cpu_dai->dma_data = &stmp3xxx_audio_in;
}
ret = request_irq(irq, stmp3xxx_err_irq, 0, "STMP3xxx 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) {
HW_AUDIOOUT_CTRL_CLR(BM_AUDIOOUT_CTRL_FIFO_OVERFLOW_IRQ);
HW_AUDIOOUT_CTRL_CLR(BM_AUDIOOUT_CTRL_FIFO_UNDERFLOW_IRQ);
HW_AUDIOOUT_CTRL_SET(BM_AUDIOOUT_CTRL_FIFO_ERROR_IRQ_EN);
} else {
HW_AUDIOIN_CTRL_CLR(BM_AUDIOIN_CTRL_FIFO_OVERFLOW_IRQ);
HW_AUDIOIN_CTRL_CLR(BM_AUDIOIN_CTRL_FIFO_UNDERFLOW_IRQ);
HW_AUDIOIN_CTRL_SET(BM_AUDIOIN_CTRL_FIFO_ERROR_IRQ_EN);
}
return 0;
}
static void stmp3xxx_adc_shutdown(struct snd_pcm_substream *substream)
{
int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
/* Disable error interrupt */
if (playback) {
HW_AUDIOOUT_CTRL_CLR(BM_AUDIOOUT_CTRL_FIFO_ERROR_IRQ_EN);
free_irq(IRQ_DAC_ERROR, substream);
} else {
HW_AUDIOIN_CTRL_CLR(BM_AUDIOIN_CTRL_FIFO_ERROR_IRQ_EN);
free_irq(IRQ_ADC_ERROR, substream);
}
}
#ifdef CONFIG_PM
static int stmp3xxx_adc_suspend(struct platform_device *pdev,
struct snd_soc_dai *cpu_dai)
{
return 0;
}
static int stmp3xxx_adc_resume(struct platform_device *pdev,
struct snd_soc_dai *cpu_dai)
{
return 0;
}
#else
#define stmp3xxx_adc_suspend NULL
#define stmp3xxx_adc_resume NULL
#endif /* CONFIG_PM */
struct snd_soc_dai stmp3xxx_adc_dai = {
.name = "stmp3xxx adc/dac",
.id = 0,
.type = SND_SOC_DAI_PCM,
.suspend = stmp3xxx_adc_suspend,
.resume = stmp3xxx_adc_resume,
.playback = {
.channels_min = 2,
.channels_max = 2,
.rates = STMP3XXX_ADC_RATES,
.formats = STMP3XXX_ADC_FORMATS,
},
.capture = {
.channels_min = 2,
.channels_max = 2,
.rates = STMP3XXX_ADC_RATES,
.formats = STMP3XXX_ADC_FORMATS,
},
.ops = {
.startup = stmp3xxx_adc_startup,
.shutdown = stmp3xxx_adc_shutdown,
.trigger = stmp3xxx_adc_trigger,
},
};
EXPORT_SYMBOL_GPL(stmp3xxx_adc_dai);
MODULE_AUTHOR("Vladislav Buzov");
MODULE_DESCRIPTION("stmp3xxx dac/adc DAI");
MODULE_LICENSE("GPL");
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