linux/sound/soc/au1x/psc-i2s.c

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/*
* 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 cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo <tj@kernel.org> Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org> Cc: Ingo Molnar <mingo@redhat.com> Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
2010-03-24 08:04:11 +00:00
#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) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
#define I2SPCR_START(stype) \
((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
#define I2SPCR_STOP(stype) \
((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
#define I2SPCR_CLRFIFO(stype) \
((stype) == SNDRV_PCM_STREAM_PLAYBACK ? 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->stream;
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 int au1xpsc_i2s_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]);
return 0;
}
static const struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
.startup = au1xpsc_i2s_startup,
.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 *iores, *dmares;
unsigned long sel;
int ret;
struct au1xpsc_audio_data *wd;
wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data),
GFP_KERNEL);
if (!wd)
return -ENOMEM;
iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!iores)
return -ENODEV;
ret = -EBUSY;
if (!devm_request_mem_region(&pdev->dev, iores->start,
resource_size(iores),
pdev->name))
return -EBUSY;
wd->mmio = devm_ioremap(&pdev->dev, iores->start,
resource_size(iores));
if (!wd->mmio)
return -EBUSY;
dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0);
if (!dmares)
return -EBUSY;
wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start;
dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (!dmares)
return -EBUSY;
wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start;
/* 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);
return snd_soc_register_dai(&pdev->dev, &wd->dai_drv);
}
static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev)
{
struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
snd_soc_unregister_dai(&pdev->dev);
au_writel(0, I2S_CFG(wd));
au_sync();
au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
au_sync();
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),
};
module_platform_driver(au1xpsc_i2s_driver);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver");
MODULE_AUTHOR("Manuel Lauss");