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ALSA: ASoC: Au12x0/Au1550 PSC Audio support

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Audio for Au12x0/Au1550 PSCs in AC97 and I2S mode, for ASoC v1 framework.

- DBDMA, AC97 and I2S drivers
- sample AC97 machine code (Db1200)

Signed-off-by: Manuel Lauss <mano@roarinelk.homelinux.net>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This commit is contained in:
Manuel Lauss 2008-07-09 16:27:56 +02:00 committed by Jaroslav Kysela
parent bf41534506
commit 4a161d235b
10 changed files with 1474 additions and 1 deletions
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8
include/asm-mips/mach-au1x00/au1xxx_psc.h
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@ -204,6 +204,14 @@ typedef struct psc_i2s {
u32 psc_i2sudf;
} psc_i2s_t;
#define PSC_I2SCFG_OFFSET 0x08
#define PSC_I2SMASK_OFFSET 0x0C
#define PSC_I2SPCR_OFFSET 0x10
#define PSC_I2SSTAT_OFFSET 0x14
#define PSC_I2SEVENT_OFFSET 0x18
#define PSC_I2SRXTX_OFFSET 0x1C
#define PSC_I2SUDF_OFFSET 0x20
/* I2S Config Register. */
#define PSC_I2SCFG_RT_MASK (3 << 30)
#define PSC_I2SCFG_RT_FIFO1 (0 << 30)

1
sound/soc/Kconfig
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@ -24,6 +24,7 @@ config SND_SOC_AC97_BUS
# All the supported Soc's
source "sound/soc/at32/Kconfig"
source "sound/soc/at91/Kconfig"
source "sound/soc/au1x/Kconfig"
source "sound/soc/pxa/Kconfig"
source "sound/soc/s3c24xx/Kconfig"
source "sound/soc/sh/Kconfig"

2
sound/soc/Makefile
View File

@ -2,4 +2,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/
obj-$(CONFIG_SND_SOC) += omap/
obj-$(CONFIG_SND_SOC) += omap/ au1x/

32
sound/soc/au1x/Kconfig Normal file
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@ -0,0 +1,32 @@
##
## Au1200/Au1550 PSC + DBDMA
##
config SND_SOC_AU1XPSC
tristate "SoC Audio for Au1200/Au1250/Au1550"
depends on SOC_AU1200 || SOC_AU1550
help
This option enables support for the Programmable Serial
Controllers in AC97 and I2S mode, and the Descriptor-Based DMA
Controller (DBDMA) as found on the Au1200/Au1250/Au1550 SoC.
config SND_SOC_AU1XPSC_I2S
tristate
config SND_SOC_AU1XPSC_AC97
tristate
select AC97_BUS
select SND_AC97_CODEC
select SND_SOC_AC97_BUS
##
## Boards
##
config SND_SOC_SAMPLE_PSC_AC97
tristate "Sample Au12x0/Au1550 PSC AC97 sound machine"
depends on SND_SOC_AU1XPSC
select SND_SOC_AU1XPSC_AC97
select SND_SOC_AC97_CODEC
help
This is a sample AC97 sound machine for use in Au12x0/Au1550
based systems which have audio on PSC1 (e.g. Db1200 demoboard).

13
sound/soc/au1x/Makefile Normal file
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@ -0,0 +1,13 @@
# Au1200/Au1550 PSC audio
snd-soc-au1xpsc-dbdma-objs := dbdma2.o
snd-soc-au1xpsc-i2s-objs := psc-i2s.o
snd-soc-au1xpsc-ac97-objs := psc-ac97.o
obj-$(CONFIG_SND_SOC_AU1XPSC) += snd-soc-au1xpsc-dbdma.o
obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o
obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o
# Boards
snd-soc-sample-ac97-objs := sample-ac97.o
obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o

421
sound/soc/au1x/dbdma2.c Normal file
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@ -0,0 +1,421 @@
/*
* Au12x0/Au1550 PSC ALSA ASoC audio support.
*
* (c) 2007-2008 MSC Vertriebsges.m.b.H.,
* Manuel Lauss <mano@roarinelk.homelinux.net>
*
* 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.
*
* DMA glue for Au1x-PSC audio.
*
* NOTE: all of these drivers can only work with a SINGLE instance
* of a PSC. Multiple independent audio devices are impossible
* with ASoC v1.
*/
#include <linux/module.h>
#include <linux/init.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <linux/dma-mapping.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <asm/mach-au1x00/au1000.h>
#include <asm/mach-au1x00/au1xxx_dbdma.h>
#include <asm/mach-au1x00/au1xxx_psc.h>
#include "psc.h"
/*#define PCM_DEBUG*/
#define MSG(x...) printk(KERN_INFO "au1xpsc_pcm: " x)
#ifdef PCM_DEBUG
#define DBG MSG
#else
#define DBG(x...) do {} while (0)
#endif
struct au1xpsc_audio_dmadata {
/* DDMA control data */
unsigned int ddma_id; /* DDMA direction ID for this PSC */
u32 ddma_chan; /* DDMA context */
/* PCM context (for irq handlers) */
struct snd_pcm_substream *substream;
unsigned long curr_period; /* current segment DDMA is working on */
unsigned long q_period; /* queue period(s) */
unsigned long dma_area; /* address of queued DMA area */
unsigned long dma_area_s; /* start address of DMA area */
unsigned long pos; /* current byte position being played */
unsigned long periods; /* number of SG segments in total */
unsigned long period_bytes; /* size in bytes of one SG segment */
/* runtime data */
int msbits;
};
/* instance data. There can be only one, MacLeod!!!! */
static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
/*
* These settings are somewhat okay, at least on my machine audio plays
* almost skip-free. Especially the 64kB buffer seems to help a LOT.
*/
#define AU1XPSC_PERIOD_MIN_BYTES 1024
#define AU1XPSC_BUFFER_MIN_BYTES 65536
#define AU1XPSC_PCM_FMTS \
(SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \
SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \
SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \
SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | \
SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE | \
0)
/* PCM hardware DMA capabilities - platform specific */
static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED,
.formats = AU1XPSC_PCM_FMTS,
.period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
.period_bytes_max = 4096 * 1024 - 1,
.periods_min = 2,
.periods_max = 4096, /* 2 to as-much-as-you-like */
.buffer_bytes_max = 4096 * 1024 - 1,
.fifo_size = 16, /* fifo entries of AC97/I2S PSC */
};
static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd)
{
au1xxx_dbdma_put_source_flags(cd->ddma_chan,
(void *)phys_to_virt(cd->dma_area),
cd->period_bytes, DDMA_FLAGS_IE);
/* update next-to-queue period */
++cd->q_period;
cd->dma_area += cd->period_bytes;
if (cd->q_period >= cd->periods) {
cd->q_period = 0;
cd->dma_area = cd->dma_area_s;
}
}
static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd)
{
au1xxx_dbdma_put_dest_flags(cd->ddma_chan,
(void *)phys_to_virt(cd->dma_area),
cd->period_bytes, DDMA_FLAGS_IE);
/* update next-to-queue period */
++cd->q_period;
cd->dma_area += cd->period_bytes;
if (cd->q_period >= cd->periods) {
cd->q_period = 0;
cd->dma_area = cd->dma_area_s;
}
}
static void au1x_pcm_dmatx_cb(int irq, void *dev_id)
{
struct au1xpsc_audio_dmadata *cd = dev_id;
cd->pos += cd->period_bytes;
if (++cd->curr_period >= cd->periods) {
cd->pos = 0;
cd->curr_period = 0;
}
snd_pcm_period_elapsed(cd->substream);
au1x_pcm_queue_tx(cd);
}
static void au1x_pcm_dmarx_cb(int irq, void *dev_id)
{
struct au1xpsc_audio_dmadata *cd = dev_id;
cd->pos += cd->period_bytes;
if (++cd->curr_period >= cd->periods) {
cd->pos = 0;
cd->curr_period = 0;
}
snd_pcm_period_elapsed(cd->substream);
au1x_pcm_queue_rx(cd);
}
static void au1x_pcm_dbdma_free(struct au1xpsc_audio_dmadata *pcd)
{
if (pcd->ddma_chan) {
au1xxx_dbdma_stop(pcd->ddma_chan);
au1xxx_dbdma_reset(pcd->ddma_chan);
au1xxx_dbdma_chan_free(pcd->ddma_chan);
pcd->ddma_chan = 0;
pcd->msbits = 0;
}
}
/* in case of missing DMA ring or changed TX-source / RX-dest bit widths,
* allocate (or reallocate) a 2-descriptor DMA ring with bit depth according
* to ALSA-supplied sample depth. This is due to limitations in the dbdma api
* (cannot adjust source/dest widths of already allocated descriptor ring).
*/
static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd,
int stype, int msbits)
{
/* DMA only in 8/16/32 bit widths */
if (msbits == 24)
msbits = 32;
/* check current config: correct bits and descriptors allocated? */
if ((pcd->ddma_chan) && (msbits == pcd->msbits))
goto out; /* all ok! */
au1x_pcm_dbdma_free(pcd);
if (stype == PCM_RX)
pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id,
DSCR_CMD0_ALWAYS,
au1x_pcm_dmarx_cb, (void *)pcd);
else
pcd->ddma_chan = au1xxx_dbdma_chan_alloc(DSCR_CMD0_ALWAYS,
pcd->ddma_id,
au1x_pcm_dmatx_cb, (void *)pcd);
if (!pcd->ddma_chan)
return -ENOMEM;;
au1xxx_dbdma_set_devwidth(pcd->ddma_chan, msbits);
au1xxx_dbdma_ring_alloc(pcd->ddma_chan, 2);
pcd->msbits = msbits;
au1xxx_dbdma_stop(pcd->ddma_chan);
au1xxx_dbdma_reset(pcd->ddma_chan);
out:
return 0;
}
static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct au1xpsc_audio_dmadata *pcd;
int stype, ret;
ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
if (ret < 0)
goto out;
stype = SUBSTREAM_TYPE(substream);
pcd = au1xpsc_audio_pcmdma[stype];
DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d "
"runtime->min_align %d\n",
(unsigned long)runtime->dma_area,
(unsigned long)runtime->dma_addr, runtime->dma_bytes,
runtime->min_align);
DBG("bits %d frags %d frag_bytes %d is_rx %d\n", params->msbits,
params_periods(params), params_period_bytes(params), stype);
ret = au1x_pcm_dbdma_realloc(pcd, stype, params->msbits);
if (ret) {
MSG("DDMA channel (re)alloc failed!\n");
goto out;
}
pcd->substream = substream;
pcd->period_bytes = params_period_bytes(params);
pcd->periods = params_periods(params);
pcd->dma_area_s = pcd->dma_area = (unsigned long)runtime->dma_addr;
pcd->q_period = 0;
pcd->curr_period = 0;
pcd->pos = 0;
ret = 0;
out:
return ret;
}
static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream)
{
snd_pcm_lib_free_pages(substream);
return 0;
}
static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream)
{
struct au1xpsc_audio_dmadata *pcd =
au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)];
au1xxx_dbdma_reset(pcd->ddma_chan);
if (SUBSTREAM_TYPE(substream) == PCM_RX) {
au1x_pcm_queue_rx(pcd);
au1x_pcm_queue_rx(pcd);
} else {
au1x_pcm_queue_tx(pcd);
au1x_pcm_queue_tx(pcd);
}
return 0;
}
static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
u32 c = au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->ddma_chan;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
au1xxx_dbdma_start(c);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
au1xxx_dbdma_stop(c);
break;
default:
return -EINVAL;
}
return 0;
}
static snd_pcm_uframes_t
au1xpsc_pcm_pointer(struct snd_pcm_substream *substream)
{
return bytes_to_frames(substream->runtime,
au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->pos);
}
static int au1xpsc_pcm_open(struct snd_pcm_substream *substream)
{
snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware);
return 0;
}
static int au1xpsc_pcm_close(struct snd_pcm_substream *substream)
{
au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]);
return 0;
}
struct snd_pcm_ops au1xpsc_pcm_ops = {
.open = au1xpsc_pcm_open,
.close = au1xpsc_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = au1xpsc_pcm_hw_params,
.hw_free = au1xpsc_pcm_hw_free,
.prepare = au1xpsc_pcm_prepare,
.trigger = au1xpsc_pcm_trigger,
.pointer = au1xpsc_pcm_pointer,
};
static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm)
{
snd_pcm_lib_preallocate_free_for_all(pcm);
}
static int au1xpsc_pcm_new(struct snd_card *card,
struct snd_soc_dai *dai,
struct snd_pcm *pcm)
{
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1);
return 0;
}
static int au1xpsc_pcm_probe(struct platform_device *pdev)
{
struct resource *r;
int ret;
if (au1xpsc_audio_pcmdma[PCM_TX] || au1xpsc_audio_pcmdma[PCM_RX])
return -EBUSY;
/* TX DMA */
au1xpsc_audio_pcmdma[PCM_TX]
= kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
if (!au1xpsc_audio_pcmdma[PCM_TX])
return -ENOMEM;
r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
if (!r) {
ret = -ENODEV;
goto out1;
}
(au1xpsc_audio_pcmdma[PCM_TX])->ddma_id = r->start;
/* RX DMA */
au1xpsc_audio_pcmdma[PCM_RX]
= kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
if (!au1xpsc_audio_pcmdma[PCM_RX])
return -ENOMEM;
r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (!r) {
ret = -ENODEV;
goto out2;
}
(au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start;
return 0;
out2:
kfree(au1xpsc_audio_pcmdma[PCM_RX]);
au1xpsc_audio_pcmdma[PCM_RX] = NULL;
out1:
kfree(au1xpsc_audio_pcmdma[PCM_TX]);
au1xpsc_audio_pcmdma[PCM_TX] = NULL;
return ret;
}
static int au1xpsc_pcm_remove(struct platform_device *pdev)
{
int i;
for (i = 0; i < 2; i++) {
if (au1xpsc_audio_pcmdma[i]) {
au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]);
kfree(au1xpsc_audio_pcmdma[i]);
au1xpsc_audio_pcmdma[i] = NULL;
}
}
return 0;
}
/* au1xpsc audio platform */
struct snd_soc_platform au1xpsc_soc_platform = {
.name = "au1xpsc-pcm-dbdma",
.probe = au1xpsc_pcm_probe,
.remove = au1xpsc_pcm_remove,
.pcm_ops = &au1xpsc_pcm_ops,
.pcm_new = au1xpsc_pcm_new,
.pcm_free = au1xpsc_pcm_free_dma_buffers,
};
EXPORT_SYMBOL_GPL(au1xpsc_soc_platform);
static int __init au1xpsc_audio_dbdma_init(void)
{
au1xpsc_audio_pcmdma[PCM_TX] = NULL;
au1xpsc_audio_pcmdma[PCM_RX] = NULL;
return 0;
}
static void __exit au1xpsc_audio_dbdma_exit(void)
{
}
module_init(au1xpsc_audio_dbdma_init);
module_exit(au1xpsc_audio_dbdma_exit);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver");
MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");

387
sound/soc/au1x/psc-ac97.c Normal file
View File

@ -0,0 +1,387 @@
/*
* Au12x0/Au1550 PSC ALSA ASoC audio support.
*
* (c) 2007-2008 MSC Vertriebsges.m.b.H.,
* Manuel Lauss <mano@roarinelk.homelinux.net>
*
* 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 AC97 glue.
*
* NOTE: all of these drivers can only work with a SINGLE instance
* of a PSC. Multiple independent audio devices are impossible
* with ASoC v1.
*/
#include <linux/init.h>
#include <linux/module.h>
#include <linux/device.h>
#include <linux/delay.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"
#define AC97_DIR \
(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
#define AC97_RATES \
SNDRV_PCM_RATE_8000_48000
#define AC97_FMTS \
(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE)
#define AC97PCR_START(stype) \
((stype) == PCM_TX ? PSC_AC97PCR_TS : PSC_AC97PCR_RS)
#define AC97PCR_STOP(stype) \
((stype) == PCM_TX ? PSC_AC97PCR_TP : PSC_AC97PCR_RP)
#define AC97PCR_CLRFIFO(stype) \
((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
/* instance data. There can be only one, MacLeod!!!! */
static struct au1xpsc_audio_data *au1xpsc_ac97_workdata;
/* AC97 controller reads codec register */
static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97,
unsigned short reg)
{
/* FIXME */
struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
unsigned short data, tmo;
au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata));
au_sync();
tmo = 1000;
while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo)
udelay(2);
if (!tmo)
data = 0xffff;
else
data = au_readl(AC97_CDC(pscdata)) & 0xffff;
au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
au_sync();
return data;
}
/* AC97 controller writes to codec register */
static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
unsigned short val)
{
/* FIXME */
struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
unsigned int tmo;
au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata));
au_sync();
tmo = 1000;
while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo)
au_sync();
au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
au_sync();
}
/* AC97 controller asserts a warm reset */
static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97)
{
/* FIXME */
struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
au_writel(PSC_AC97RST_SNC, AC97_RST(pscdata));
au_sync();
msleep(10);
au_writel(0, AC97_RST(pscdata));
au_sync();
}
static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97)
{
/* FIXME */
struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
int i;
/* disable PSC during cold reset */
au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
au_sync();
au_writel(PSC_CTRL_DISABLE, PSC_CTRL(pscdata));
au_sync();
/* issue cold reset */
au_writel(PSC_AC97RST_RST, AC97_RST(pscdata));
au_sync();
msleep(500);
au_writel(0, AC97_RST(pscdata));
au_sync();
/* enable PSC */
au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata));
au_sync();
/* wait for PSC to indicate it's ready */
i = 100000;
while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i))
au_sync();
if (i == 0) {
printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n");
return;
}
/* enable the ac97 function */
au_writel(pscdata->cfg | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
au_sync();
/* wait for AC97 core to become ready */
i = 100000;
while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i))
au_sync();
if (i == 0)
printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n");
}
/* AC97 controller operations */
struct snd_ac97_bus_ops soc_ac97_ops = {
.read = au1xpsc_ac97_read,
.write = au1xpsc_ac97_write,
.reset = au1xpsc_ac97_cold_reset,
.warm_reset = au1xpsc_ac97_warm_reset,
};
EXPORT_SYMBOL_GPL(soc_ac97_ops);
static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
/* FIXME */
struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
unsigned long r, stat;
int chans, stype = SUBSTREAM_TYPE(substream);
chans = params_channels(params);
r = au_readl(AC97_CFG(pscdata));
stat = au_readl(AC97_STAT(pscdata));
/* already active? */
if (stat & (PSC_AC97STAT_TB | PSC_AC97STAT_RB)) {
/* reject parameters not currently set up */
if ((PSC_AC97CFG_GET_LEN(r) != params->msbits) ||
(pscdata->rate != params_rate(params)))
return -EINVAL;
} else {
/* disable AC97 device controller first */
au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
au_sync();
/* set sample bitdepth: REG[24:21]=(BITS-2)/2 */
r &= ~PSC_AC97CFG_LEN_MASK;
r |= PSC_AC97CFG_SET_LEN(params->msbits);
/* channels: enable slots for front L/R channel */
if (stype == PCM_TX) {
r &= ~PSC_AC97CFG_TXSLOT_MASK;
r |= PSC_AC97CFG_TXSLOT_ENA(3);
r |= PSC_AC97CFG_TXSLOT_ENA(4);
} else {
r &= ~PSC_AC97CFG_RXSLOT_MASK;
r |= PSC_AC97CFG_RXSLOT_ENA(3);
r |= PSC_AC97CFG_RXSLOT_ENA(4);
}
/* finally enable the AC97 controller again */
au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
au_sync();
pscdata->cfg = r;
pscdata->rate = params_rate(params);
}
return 0;
}
static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
int cmd)
{
/* FIXME */
struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
int ret, stype = SUBSTREAM_TYPE(substream);
ret = 0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
au_writel(AC97PCR_START(stype), AC97_PCR(pscdata));
au_sync();
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata));
au_sync();
break;
default:
ret = -EINVAL;
}
return ret;
}
static int au1xpsc_ac97_probe(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
int ret;
struct resource *r;
unsigned long sel;
if (au1xpsc_ac97_workdata)
return -EBUSY;
au1xpsc_ac97_workdata =
kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL);
if (!au1xpsc_ac97_workdata)
return -ENOMEM;
r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!r) {
ret = -ENODEV;
goto out0;
}
ret = -EBUSY;
au1xpsc_ac97_workdata->ioarea =
request_mem_region(r->start, r->end - r->start + 1,
"au1xpsc_ac97");
if (!au1xpsc_ac97_workdata->ioarea)
goto out0;
au1xpsc_ac97_workdata->mmio = ioremap(r->start, 0xffff);
if (!au1xpsc_ac97_workdata->mmio)
goto out1;
/* configuration: max dma trigger threshold, enable ac97 */
au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 |
PSC_AC97CFG_TT_FIFO8 |
PSC_AC97CFG_DE_ENABLE;
/* preserve PSC clock source set up by platform (dev.platform_data
* is already occupied by soc layer)
*/
sel = au_readl(PSC_SEL(au1xpsc_ac97_workdata)) & PSC_SEL_CLK_MASK;
au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata));
au_sync();
au_writel(0, PSC_SEL(au1xpsc_ac97_workdata));
au_sync();
au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(au1xpsc_ac97_workdata));
au_sync();
/* next up: cold reset. Dont check for PSC-ready now since
* there may not be any codec clock yet.
*/
return 0;
out1:
release_resource(au1xpsc_ac97_workdata->ioarea);
kfree(au1xpsc_ac97_workdata->ioarea);
out0:
kfree(au1xpsc_ac97_workdata);
au1xpsc_ac97_workdata = NULL;
return ret;
}
static void au1xpsc_ac97_remove(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
/* disable PSC completely */
au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
au_sync();
au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata));
au_sync();
iounmap(au1xpsc_ac97_workdata->mmio);
release_resource(au1xpsc_ac97_workdata->ioarea);
kfree(au1xpsc_ac97_workdata->ioarea);
kfree(au1xpsc_ac97_workdata);
au1xpsc_ac97_workdata = NULL;
}
static int au1xpsc_ac97_suspend(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
/* save interesting registers and disable PSC */
au1xpsc_ac97_workdata->pm[0] =
au_readl(PSC_SEL(au1xpsc_ac97_workdata));
au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
au_sync();
au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata));
au_sync();
return 0;
}
static int au1xpsc_ac97_resume(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
/* restore PSC clock config */
au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE,
PSC_SEL(au1xpsc_ac97_workdata));
au_sync();
/* after this point the ac97 core will cold-reset the codec.
* During cold-reset the PSC is reinitialized and the last
* configuration set up in hw_params() is restored.
*/
return 0;
}
struct snd_soc_dai au1xpsc_ac97_dai = {
.name = "au1xpsc_ac97",
.type = SND_SOC_DAI_AC97,
.probe = au1xpsc_ac97_probe,
.remove = au1xpsc_ac97_remove,
.suspend = au1xpsc_ac97_suspend,
.resume = au1xpsc_ac97_resume,
.playback = {
.rates = AC97_RATES,
.formats = AC97_FMTS,
.channels_min = 2,
.channels_max = 2,
},
.capture = {
.rates = AC97_RATES,
.formats = AC97_FMTS,
.channels_min = 2,
.channels_max = 2,
},
.ops = {
.trigger = au1xpsc_ac97_trigger,
.hw_params = au1xpsc_ac97_hw_params,
},
};
EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
static int __init au1xpsc_ac97_init(void)
{
au1xpsc_ac97_workdata = NULL;
return 0;
}
static void __exit au1xpsc_ac97_exit(void)
{
}
module_init(au1xpsc_ac97_init);
module_exit(au1xpsc_ac97_exit);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver");
MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");

414
sound/soc/au1x/psc-i2s.c Normal file
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@ -0,0 +1,414 @@
/*
* Au12x0/Au1550 PSC ALSA ASoC audio support.
*
* (c) 2007-2008 MSC Vertriebsges.m.b.H.,
* Manuel Lauss <mano@roarinelk.homelinux.net>
*
* 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: all of these drivers can only work with a SINGLE instance
* of a PSC. Multiple independent audio devices are impossible
* with ASoC v1.
* NOTE: so far only PSC slave mode (bit- and frameclock) is supported.
*/
#include <linux/init.h>
#include <linux/module.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)
/* instance data. There can be only one, MacLeod!!!! */
static struct au1xpsc_audio_data *au1xpsc_i2s_workdata;
static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
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 au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
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_RB | 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 au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
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 int au1xpsc_i2s_probe(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
struct resource *r;
unsigned long sel;
int ret;
if (au1xpsc_i2s_workdata)
return -EBUSY;
au1xpsc_i2s_workdata =
kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL);
if (!au1xpsc_i2s_workdata)
return -ENOMEM;
r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!r) {
ret = -ENODEV;
goto out0;
}
ret = -EBUSY;
au1xpsc_i2s_workdata->ioarea =
request_mem_region(r->start, r->end - r->start + 1,
"au1xpsc_i2s");
if (!au1xpsc_i2s_workdata->ioarea)
goto out0;
au1xpsc_i2s_workdata->mmio = ioremap(r->start, 0xffff);
if (!au1xpsc_i2s_workdata->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(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK;
au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
au_sync();
au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(au1xpsc_i2s_workdata));
au_writel(0, I2S_CFG(au1xpsc_i2s_workdata));
au_sync();
/* preconfigure: set max rx/tx fifo depths */
au1xpsc_i2s_workdata->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.
*/
return 0;
out1:
release_resource(au1xpsc_i2s_workdata->ioarea);
kfree(au1xpsc_i2s_workdata->ioarea);
out0:
kfree(au1xpsc_i2s_workdata);
au1xpsc_i2s_workdata = NULL;
return ret;
}
static void au1xpsc_i2s_remove(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
au_writel(0, I2S_CFG(au1xpsc_i2s_workdata));
au_sync();
au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
au_sync();
iounmap(au1xpsc_i2s_workdata->mmio);
release_resource(au1xpsc_i2s_workdata->ioarea);
kfree(au1xpsc_i2s_workdata->ioarea);
kfree(au1xpsc_i2s_workdata);
au1xpsc_i2s_workdata = NULL;
}
static int au1xpsc_i2s_suspend(struct platform_device *pdev,
struct snd_soc_dai *cpu_dai)
{
/* save interesting register and disable PSC */
au1xpsc_i2s_workdata->pm[0] =
au_readl(PSC_SEL(au1xpsc_i2s_workdata));
au_writel(0, I2S_CFG(au1xpsc_i2s_workdata));
au_sync();
au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
au_sync();
return 0;
}
static int au1xpsc_i2s_resume(struct platform_device *pdev,
struct snd_soc_dai *cpu_dai)
{
/* select I2S mode and PSC clock */
au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
au_sync();
au_writel(0, PSC_SEL(au1xpsc_i2s_workdata));
au_sync();
au_writel(au1xpsc_i2s_workdata->pm[0],
PSC_SEL(au1xpsc_i2s_workdata));
au_sync();
return 0;
}
struct snd_soc_dai au1xpsc_i2s_dai = {
.name = "au1xpsc_i2s",
.type = SND_SOC_DAI_I2S,
.probe = au1xpsc_i2s_probe,
.remove = au1xpsc_i2s_remove,
.suspend = au1xpsc_i2s_suspend,
.resume = au1xpsc_i2s_resume,
.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 = {
.trigger = au1xpsc_i2s_trigger,
.hw_params = au1xpsc_i2s_hw_params,
},
.dai_ops = {
.set_fmt = au1xpsc_i2s_set_fmt,
},
};
EXPORT_SYMBOL(au1xpsc_i2s_dai);
static int __init au1xpsc_i2s_init(void)
{
au1xpsc_i2s_workdata = NULL;
return 0;
}
static void __exit au1xpsc_i2s_exit(void)
{
}
module_init(au1xpsc_i2s_init);
module_exit(au1xpsc_i2s_exit);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver");
MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");

53
sound/soc/au1x/psc.h Normal file
View File

@ -0,0 +1,53 @@
/*
* Au12x0/Au1550 PSC ALSA ASoC audio support.
*
* (c) 2007-2008 MSC Vertriebsges.m.b.H.,
* Manuel Lauss <mano@roarinelk.homelinux.net>
*
* 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.
*
* NOTE: all of these drivers can only work with a SINGLE instance
* of a PSC. Multiple independent audio devices are impossible
* with ASoC v1.
*/
#ifndef _AU1X_PCM_H
#define _AU1X_PCM_H
extern struct snd_soc_dai au1xpsc_ac97_dai;
extern struct snd_soc_dai au1xpsc_i2s_dai;
extern struct snd_soc_platform au1xpsc_soc_platform;
extern struct snd_ac97_bus_ops soc_ac97_ops;
struct au1xpsc_audio_data {
void __iomem *mmio;
unsigned long cfg;
unsigned long rate;
unsigned long pm[2];
struct resource *ioarea;
};
#define PCM_TX 0
#define PCM_RX 1
#define SUBSTREAM_TYPE(substream) \
((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX)
/* easy access macros */
#define PSC_CTRL(x) ((unsigned long)((x)->mmio) + PSC_CTRL_OFFSET)
#define PSC_SEL(x) ((unsigned long)((x)->mmio) + PSC_SEL_OFFSET)
#define I2S_STAT(x) ((unsigned long)((x)->mmio) + PSC_I2SSTAT_OFFSET)
#define I2S_CFG(x) ((unsigned long)((x)->mmio) + PSC_I2SCFG_OFFSET)
#define I2S_PCR(x) ((unsigned long)((x)->mmio) + PSC_I2SPCR_OFFSET)
#define AC97_CFG(x) ((unsigned long)((x)->mmio) + PSC_AC97CFG_OFFSET)
#define AC97_CDC(x) ((unsigned long)((x)->mmio) + PSC_AC97CDC_OFFSET)
#define AC97_EVNT(x) ((unsigned long)((x)->mmio) + PSC_AC97EVNT_OFFSET)
#define AC97_PCR(x) ((unsigned long)((x)->mmio) + PSC_AC97PCR_OFFSET)
#define AC97_RST(x) ((unsigned long)((x)->mmio) + PSC_AC97RST_OFFSET)
#define AC97_STAT(x) ((unsigned long)((x)->mmio) + PSC_AC97STAT_OFFSET)
#endif

144
sound/soc/au1x/sample-ac97.c Normal file
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/*
* Sample Au12x0/Au1550 PSC AC97 sound machine.
*
* Copyright (c) 2007-2008 Manuel Lauss <mano@roarinelk.homelinux.net>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms outlined in the file COPYING at the root of this
* source archive.
*
* This is a very generic AC97 sound machine driver for boards which
* have (AC97) audio at PSC1 (e.g. DB1200 demoboards).
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/mach-au1x00/au1000.h>
#include <asm/mach-au1x00/au1xxx_psc.h>
#include <asm/mach-au1x00/au1xxx_dbdma.h>
#include "../codecs/ac97.h"
#include "psc.h"
static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec)
{
snd_soc_dapm_sync(codec);
return 0;
}
static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = {
.name = "AC97",
.stream_name = "AC97 HiFi",
.cpu_dai = &au1xpsc_ac97_dai, /* see psc-ac97.c */
.codec_dai = &ac97_dai, /* see codecs/ac97.c */
.init = au1xpsc_sample_ac97_init,
.ops = NULL,
};
static struct snd_soc_machine au1xpsc_sample_ac97_machine = {
.name = "Au1xxx PSC AC97 Audio",
.dai_link = &au1xpsc_sample_ac97_dai,
.num_links = 1,
};
static struct snd_soc_device au1xpsc_sample_ac97_devdata = {
.machine = &au1xpsc_sample_ac97_machine,
.platform = &au1xpsc_soc_platform, /* see dbdma2.c */
.codec_dev = &soc_codec_dev_ac97,
};
static struct resource au1xpsc_psc1_res[] = {
[0] = {
.start = CPHYSADDR(PSC1_BASE_ADDR),
.end = CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff,
.flags = IORESOURCE_MEM,
},
[1] = {
#ifdef CONFIG_SOC_AU1200
.start = AU1200_PSC1_INT,
.end = AU1200_PSC1_INT,
#elif defined(CONFIG_SOC_AU1550)
.start = AU1550_PSC1_INT,
.end = AU1550_PSC1_INT,
#endif
.flags = IORESOURCE_IRQ,
},
[2] = {
.start = DSCR_CMD0_PSC1_TX,
.end = DSCR_CMD0_PSC1_TX,
.flags = IORESOURCE_DMA,
},
[3] = {
.start = DSCR_CMD0_PSC1_RX,
.end = DSCR_CMD0_PSC1_RX,
.flags = IORESOURCE_DMA,
},
};
static struct platform_device *au1xpsc_sample_ac97_dev;
static int __init au1xpsc_sample_ac97_load(void)
{
int ret;
#ifdef CONFIG_SOC_AU1200
unsigned long io;
/* modify sys_pinfunc for AC97 on PSC1 */
io = au_readl(SYS_PINFUNC);
io |= SYS_PINFUNC_P1C;
io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B);
au_writel(io, SYS_PINFUNC);
au_sync();
#endif
ret = -ENOMEM;
/* setup PSC clock source for AC97 part: external clock provided
* by codec. The psc-ac97.c driver depends on this setting!
*/
au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET);
au_sync();
au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1);
if (!au1xpsc_sample_ac97_dev)
goto out;
au1xpsc_sample_ac97_dev->resource =
kmemdup(au1xpsc_psc1_res, sizeof(struct resource) *
ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL);
au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res);
au1xpsc_sample_ac97_dev->id = 1;
platform_set_drvdata(au1xpsc_sample_ac97_dev,
&au1xpsc_sample_ac97_devdata);
au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev;
ret = platform_device_add(au1xpsc_sample_ac97_dev);
if (ret) {
platform_device_put(au1xpsc_sample_ac97_dev);
au1xpsc_sample_ac97_dev = NULL;
}
out:
return ret;
}
static void __exit au1xpsc_sample_ac97_exit(void)
{
platform_device_unregister(au1xpsc_sample_ac97_dev);
}
module_init(au1xpsc_sample_ac97_load);
module_exit(au1xpsc_sample_ac97_exit);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine");
MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");