blob: fca091276320930e7f8abd230556dabbc392e2d7 [file] [log] [blame]
/*
* Au12x0/Au1550 PSC ALSA ASoC audio support.
*
* (c) 2007-2008 MSC Vertriebsges.m.b.H.,
* Manuel Lauss <manuel.lauss@gmail.com>
*
* 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.
*
* Au1xxx-PSC I2S glue.
*
* NOTE: so far only PSC slave mode (bit- and frameclock) is supported.
*/
#include <linux/init.h>
#include <linux/module.h>
#include <linux/slab.h>
#include <linux/suspend.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include <asm/mach-au1x00/au1000.h>
#include <asm/mach-au1x00/au1xxx_psc.h>
#include "psc.h"
/* supported I2S DAI hardware formats */
#define AU1XPSC_I2S_DAIFMT \
(SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | \
SND_SOC_DAIFMT_NB_NF)
/* supported I2S direction */
#define AU1XPSC_I2S_DIR \
(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
#define AU1XPSC_I2S_RATES \
SNDRV_PCM_RATE_8000_192000
#define AU1XPSC_I2S_FMTS \
(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
#define I2SSTAT_BUSY(stype) \
((stype) == PCM_TX ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
#define I2SPCR_START(stype) \
((stype) == PCM_TX ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
#define I2SPCR_STOP(stype) \
((stype) == PCM_TX ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
#define I2SPCR_CLRFIFO(stype) \
((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(cpu_dai);
unsigned long ct;
int ret;
ret = -EINVAL;
ct = pscdata->cfg;
ct &= ~(PSC_I2SCFG_XM | PSC_I2SCFG_MLJ); /* left-justified */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
ct |= PSC_I2SCFG_XM; /* enable I2S mode */
break;
case SND_SOC_DAIFMT_MSB:
break;
case SND_SOC_DAIFMT_LSB:
ct |= PSC_I2SCFG_MLJ; /* LSB (right-) justified */
break;
default:
goto out;
}
ct &= ~(PSC_I2SCFG_BI | PSC_I2SCFG_WI); /* IB-IF */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
ct |= PSC_I2SCFG_BI | PSC_I2SCFG_WI;
break;
case SND_SOC_DAIFMT_NB_IF:
ct |= PSC_I2SCFG_BI;
break;
case SND_SOC_DAIFMT_IB_NF:
ct |= PSC_I2SCFG_WI;
break;
case SND_SOC_DAIFMT_IB_IF:
break;
default:
goto out;
}
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM: /* CODEC master */
ct |= PSC_I2SCFG_MS; /* PSC I2S slave mode */
break;
case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */
ct &= ~PSC_I2SCFG_MS; /* PSC I2S Master mode */
break;
default:
goto out;
}
pscdata->cfg = ct;
ret = 0;
out:
return ret;
}
static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
int cfgbits;
unsigned long stat;
/* check if the PSC is already streaming data */
stat = au_readl(I2S_STAT(pscdata));
if (stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB)) {
/* reject parameters not currently set up in hardware */
cfgbits = au_readl(I2S_CFG(pscdata));
if ((PSC_I2SCFG_GET_LEN(cfgbits) != params->msbits) ||
(params_rate(params) != pscdata->rate))
return -EINVAL;
} else {
/* set sample bitdepth */
pscdata->cfg &= ~(0x1f << 4);
pscdata->cfg |= PSC_I2SCFG_SET_LEN(params->msbits);
/* remember current rate for other stream */
pscdata->rate = params_rate(params);
}
return 0;
}
/* Configure PSC late: on my devel systems the codec is I2S master and
* supplies the i2sbitclock __AND__ i2sMclk (!) to the PSC unit. ASoC
* uses aggressive PM and switches the codec off when it is not in use
* which also means the PSC unit doesn't get any clocks and is therefore
* dead. That's why this chunk here gets called from the trigger callback
* because I can be reasonably certain the codec is driving the clocks.
*/
static int au1xpsc_i2s_configure(struct au1xpsc_audio_data *pscdata)
{
unsigned long tmo;
/* bring PSC out of sleep, and configure I2S unit */
au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata));
au_sync();
tmo = 1000000;
while (!(au_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_SR) && tmo)
tmo--;
if (!tmo)
goto psc_err;
au_writel(0, I2S_CFG(pscdata));
au_sync();
au_writel(pscdata->cfg | PSC_I2SCFG_DE_ENABLE, I2S_CFG(pscdata));
au_sync();
/* wait for I2S controller to become ready */
tmo = 1000000;
while (!(au_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_DR) && tmo)
tmo--;
if (tmo)
return 0;
psc_err:
au_writel(0, I2S_CFG(pscdata));
au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
au_sync();
return -ETIMEDOUT;
}
static int au1xpsc_i2s_start(struct au1xpsc_audio_data *pscdata, int stype)
{
unsigned long tmo, stat;
int ret;
ret = 0;
/* if both TX and RX are idle, configure the PSC */
stat = au_readl(I2S_STAT(pscdata));
if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) {
ret = au1xpsc_i2s_configure(pscdata);
if (ret)
goto out;
}
au_writel(I2SPCR_CLRFIFO(stype), I2S_PCR(pscdata));
au_sync();
au_writel(I2SPCR_START(stype), I2S_PCR(pscdata));
au_sync();
/* wait for start confirmation */
tmo = 1000000;
while (!(au_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo)
tmo--;
if (!tmo) {
au_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata));
au_sync();
ret = -ETIMEDOUT;
}
out:
return ret;
}
static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype)
{
unsigned long tmo, stat;
au_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata));
au_sync();
/* wait for stop confirmation */
tmo = 1000000;
while ((au_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo)
tmo--;
/* if both TX and RX are idle, disable PSC */
stat = au_readl(I2S_STAT(pscdata));
if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) {
au_writel(0, I2S_CFG(pscdata));
au_sync();
au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
au_sync();
}
return 0;
}
static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
int ret, stype = SUBSTREAM_TYPE(substream);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
ret = au1xpsc_i2s_start(pscdata, stype);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
ret = au1xpsc_i2s_stop(pscdata, stype);
break;
default:
ret = -EINVAL;
}
return ret;
}
static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
.trigger = au1xpsc_i2s_trigger,
.hw_params = au1xpsc_i2s_hw_params,
.set_fmt = au1xpsc_i2s_set_fmt,
};
static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = {
.playback = {
.rates = AU1XPSC_I2S_RATES,
.formats = AU1XPSC_I2S_FMTS,
.channels_min = 2,
.channels_max = 8, /* 2 without external help */
},
.capture = {
.rates = AU1XPSC_I2S_RATES,
.formats = AU1XPSC_I2S_FMTS,
.channels_min = 2,
.channels_max = 8, /* 2 without external help */
},
.ops = &au1xpsc_i2s_dai_ops,
};
static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
{
struct resource *r;
unsigned long sel;
int ret;
struct au1xpsc_audio_data *wd;
wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL);
if (!wd)
return -ENOMEM;
r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!r) {
ret = -ENODEV;
goto out0;
}
ret = -EBUSY;
if (!request_mem_region(r->start, resource_size(r), pdev->name))
goto out0;
wd->mmio = ioremap(r->start, resource_size(r));
if (!wd->mmio)
goto out1;
/* preserve PSC clock source set up by platform (dev.platform_data
* is already occupied by soc layer)
*/
sel = au_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK;
au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
au_sync();
au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(wd));
au_writel(0, I2S_CFG(wd));
au_sync();
/* preconfigure: set max rx/tx fifo depths */
wd->cfg |= PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8;
/* don't wait for I2S core to become ready now; clocks may not
* be running yet; depending on clock input for PSC a wait might
* time out.
*/
/* name the DAI like this device instance ("au1xpsc-i2s.PSCINDEX") */
memcpy(&wd->dai_drv, &au1xpsc_i2s_dai_template,
sizeof(struct snd_soc_dai_driver));
wd->dai_drv.name = dev_name(&pdev->dev);
platform_set_drvdata(pdev, wd);
ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv);
if (ret)
goto out1;
/* finally add the DMA device for this PSC */
wd->dmapd = au1xpsc_pcm_add(pdev);
if (wd->dmapd)
return 0;
snd_soc_unregister_dai(&pdev->dev);
out1:
release_mem_region(r->start, resource_size(r));
out0:
kfree(wd);
return ret;
}
static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev)
{
struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (wd->dmapd)
au1xpsc_pcm_destroy(wd->dmapd);
snd_soc_unregister_dai(&pdev->dev);
au_writel(0, I2S_CFG(wd));
au_sync();
au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
au_sync();
iounmap(wd->mmio);
release_mem_region(r->start, resource_size(r));
kfree(wd);
return 0;
}
#ifdef CONFIG_PM
static int au1xpsc_i2s_drvsuspend(struct device *dev)
{
struct au1xpsc_audio_data *wd = dev_get_drvdata(dev);
/* save interesting register and disable PSC */
wd->pm[0] = au_readl(PSC_SEL(wd));
au_writel(0, I2S_CFG(wd));
au_sync();
au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
au_sync();
return 0;
}
static int au1xpsc_i2s_drvresume(struct device *dev)
{
struct au1xpsc_audio_data *wd = dev_get_drvdata(dev);
/* select I2S mode and PSC clock */
au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
au_sync();
au_writel(0, PSC_SEL(wd));
au_sync();
au_writel(wd->pm[0], PSC_SEL(wd));
au_sync();
return 0;
}
static struct dev_pm_ops au1xpsci2s_pmops = {
.suspend = au1xpsc_i2s_drvsuspend,
.resume = au1xpsc_i2s_drvresume,
};
#define AU1XPSCI2S_PMOPS &au1xpsci2s_pmops
#else
#define AU1XPSCI2S_PMOPS NULL
#endif
static struct platform_driver au1xpsc_i2s_driver = {
.driver = {
.name = "au1xpsc_i2s",
.owner = THIS_MODULE,
.pm = AU1XPSCI2S_PMOPS,
},
.probe = au1xpsc_i2s_drvprobe,
.remove = __devexit_p(au1xpsc_i2s_drvremove),
};
static int __init au1xpsc_i2s_load(void)
{
return platform_driver_register(&au1xpsc_i2s_driver);
}
static void __exit au1xpsc_i2s_unload(void)
{
platform_driver_unregister(&au1xpsc_i2s_driver);
}
module_init(au1xpsc_i2s_load);
module_exit(au1xpsc_i2s_unload);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver");
MODULE_AUTHOR("Manuel Lauss");