* ASoC: AC97 codec PM
@ 2008-07-03 7:33 Manuel Lauss
2008-07-03 9:29 ` Liam Girdwood
0 siblings, 1 reply; 20+ messages in thread
From: Manuel Lauss @ 2008-07-03 7:33 UTC (permalink / raw)
To: lg; +Cc: alsa-devel
Hi Liam,
This simple patch adds suspend/resume support to the ASoC AC97 codec.
Tested on Au1200, works fine for me(TM).
Thanks!
Manuel Lauss
---
From: Manuel Lauss <mano@roarinelk.homelinux.net>
Simple suspend/resume for AC97 ASoC codec.
Signed-off-by: Manuel Lauss <mano@roarinelk.homelinux.net>
---
sound/soc/codecs/ac97.c | 25 +++++++++++++++++++++++++
1 files changed, 25 insertions(+), 0 deletions(-)
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 2a1ffe3..fb9ac9f 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -146,9 +146,34 @@ static int ac97_soc_remove(struct platform_device *pdev)
return 0;
}
+#ifdef CONFIG_PM
+static int ac97_soc_suspend(struct platform_device *pdev, pm_message_t msg)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_ac97_suspend(socdev->codec->ac97);
+
+ return 0;
+}
+
+static int ac97_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_ac97_resume(socdev->codec->ac97);
+
+ return 0;
+}
+#else
+#define ac97_soc_suspend NULL
+#define ac97_soc_resume NULL
+#endif
+
struct snd_soc_codec_device soc_codec_dev_ac97 = {
.probe = ac97_soc_probe,
.remove = ac97_soc_remove,
+ .suspend = ac97_soc_suspend,
+ .resume = ac97_soc_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_ac97);
--
1.5.6.1
^ permalink raw reply related [flat|nested] 20+ messages in thread
* Re: ASoC: AC97 codec PM
2008-07-03 7:33 ASoC: AC97 codec PM Manuel Lauss
@ 2008-07-03 9:29 ` Liam Girdwood
2008-07-03 9:53 ` Manuel Lauss
0 siblings, 1 reply; 20+ messages in thread
From: Liam Girdwood @ 2008-07-03 9:29 UTC (permalink / raw)
To: Manuel Lauss, Takashi Iwai; +Cc: alsa-devel
On Thu, 2008-07-03 at 09:33 +0200, Manuel Lauss wrote:
> Hi Liam,
>
> This simple patch adds suspend/resume support to the ASoC AC97 codec.
> Tested on Au1200, works fine for me(TM).
>
> Thanks!
> Manuel Lauss
>
> ---
> From: Manuel Lauss <mano@roarinelk.homelinux.net>
>
> Simple suspend/resume for AC97 ASoC codec.
>
> Signed-off-by: Manuel Lauss <mano@roarinelk.homelinux.net>
Thanks.
Acked-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Btw, any plans to fix or remove some of the 'Fixmes' in your other ASoC
code. We could upstream the Au1x000 stuff after a little cleanup and
resolution of the fixmes.
Liam
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: ASoC: AC97 codec PM
2008-07-03 9:29 ` Liam Girdwood
@ 2008-07-03 9:53 ` Manuel Lauss
2008-07-03 9:55 ` Liam Girdwood
0 siblings, 1 reply; 20+ messages in thread
From: Manuel Lauss @ 2008-07-03 9:53 UTC (permalink / raw)
To: Liam Girdwood; +Cc: Takashi Iwai, alsa-devel
On Thu, Jul 03, 2008 at 10:29:19AM +0100, Liam Girdwood wrote:
> On Thu, 2008-07-03 at 09:33 +0200, Manuel Lauss wrote:
> > Hi Liam,
> >
> > This simple patch adds suspend/resume support to the ASoC AC97 codec.
> > Tested on Au1200, works fine for me(TM).
> >
> > Thanks!
> > Manuel Lauss
> >
> > ---
> > From: Manuel Lauss <mano@roarinelk.homelinux.net>
> >
> > Simple suspend/resume for AC97 ASoC codec.
> >
> > Signed-off-by: Manuel Lauss <mano@roarinelk.homelinux.net>
>
> Thanks.
>
> Acked-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
>
>
> Btw, any plans to fix or remove some of the 'Fixmes' in your other ASoC
> code. We could upstream the Au1x000 stuff after a little cleanup and
> resolution of the fixmes.
I've updated the code considerably since the last year. It's still ASoC v1,
and lots of the FIXMEs in the ac97 part won't disappear until it's moved to
ASoC v2 (I hope, didn't investigate).
If you're interested I can send you a patch (against current linus' HEAD)
with the newest, most awesome au1xxx psc asoc code ;-)
Thanks!
Manuel Lauss
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: ASoC: AC97 codec PM
2008-07-03 9:53 ` Manuel Lauss
@ 2008-07-03 9:55 ` Liam Girdwood
2008-07-03 10:09 ` ASoC: Au12x0/Au1550 PSC Audio support Manuel Lauss
0 siblings, 1 reply; 20+ messages in thread
From: Liam Girdwood @ 2008-07-03 9:55 UTC (permalink / raw)
To: Manuel Lauss; +Cc: Takashi Iwai, alsa-devel
On Thu, 2008-07-03 at 11:53 +0200, Manuel Lauss wrote:
> On Thu, Jul 03, 2008 at 10:29:19AM +0100, Liam Girdwood wrote:
> >
> > Btw, any plans to fix or remove some of the 'Fixmes' in your other ASoC
> > code. We could upstream the Au1x000 stuff after a little cleanup and
> > resolution of the fixmes.
>
> I've updated the code considerably since the last year. It's still ASoC v1,
> and lots of the FIXMEs in the ac97 part won't disappear until it's moved to
> ASoC v2 (I hope, didn't investigate).
>
> If you're interested I can send you a patch (against current linus' HEAD)
> with the newest, most awesome au1xxx psc asoc code ;-)
>
Yes please.
Liam
^ permalink raw reply [flat|nested] 20+ messages in thread
* ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-03 9:55 ` Liam Girdwood
@ 2008-07-03 10:09 ` Manuel Lauss
2008-07-04 15:13 ` Takashi Iwai
0 siblings, 1 reply; 20+ messages in thread
From: Manuel Lauss @ 2008-07-03 10:09 UTC (permalink / raw)
To: Liam Girdwood; +Cc: Takashi Iwai, alsa-devel
On Thu, Jul 03, 2008 at 10:55:47AM +0100, Liam Girdwood wrote:
> On Thu, 2008-07-03 at 11:53 +0200, Manuel Lauss wrote:
> > On Thu, Jul 03, 2008 at 10:29:19AM +0100, Liam Girdwood wrote:
> > >
> > > Btw, any plans to fix or remove some of the 'Fixmes' in your other ASoC
> > > code. We could upstream the Au1x000 stuff after a little cleanup and
> > > resolution of the fixmes.
> >
> > I've updated the code considerably since the last year. It's still ASoC v1,
> > and lots of the FIXMEs in the ac97 part won't disappear until it's moved to
> > ASoC v2 (I hope, didn't investigate).
> >
> > If you're interested I can send you a patch (against current linus' HEAD)
> > with the newest, most awesome au1xxx psc asoc code ;-)
> >
>
> Yes please.
Here you go, apply on top of linus' head + the ac97 PM patch.
---
From: Manuel Lauss <mano@roarinelk.homelinux.net>
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>
---
include/asm-mips/mach-au1x00/au1xxx_psc.h | 51 ++---
sound/soc/Kconfig | 1 +
sound/soc/Makefile | 2 +-
sound/soc/au1x/Kconfig | 36 +++
sound/soc/au1x/Makefile | 13 +
sound/soc/au1x/dbdma2.c | 435 +++++++++++++++++++++++++++++
sound/soc/au1x/psc-ac97.c | 378 +++++++++++++++++++++++++
sound/soc/au1x/psc-i2s.c | 426 ++++++++++++++++++++++++++++
sound/soc/au1x/psc.h | 48 ++++
sound/soc/au1x/sample-ac97.c | 144 ++++++++++
10 files changed, 1499 insertions(+), 35 deletions(-)
create mode 100644 sound/soc/au1x/Kconfig
create mode 100644 sound/soc/au1x/Makefile
create mode 100644 sound/soc/au1x/dbdma2.c
create mode 100644 sound/soc/au1x/psc-ac97.c
create mode 100644 sound/soc/au1x/psc-i2s.c
create mode 100644 sound/soc/au1x/psc.h
create mode 100644 sound/soc/au1x/sample-ac97.c
diff --git a/include/asm-mips/mach-au1x00/au1xxx_psc.h b/include/asm-mips/mach-au1x00/au1xxx_psc.h
index dae4eca..912768d 100644
--- a/include/asm-mips/mach-au1x00/au1xxx_psc.h
+++ b/include/asm-mips/mach-au1x00/au1xxx_psc.h
@@ -69,29 +69,16 @@
#define PSC_CTRL_ENABLE 3
/* AC97 Registers. */
-#define PSC_AC97CFG_OFFSET 0x00000008
-#define PSC_AC97MSK_OFFSET 0x0000000c
-#define PSC_AC97PCR_OFFSET 0x00000010
-#define PSC_AC97STAT_OFFSET 0x00000014
-#define PSC_AC97EVNT_OFFSET 0x00000018
-#define PSC_AC97TXRX_OFFSET 0x0000001c
-#define PSC_AC97CDC_OFFSET 0x00000020
-#define PSC_AC97RST_OFFSET 0x00000024
-#define PSC_AC97GPO_OFFSET 0x00000028
-#define PSC_AC97GPI_OFFSET 0x0000002c
-
-#define AC97_PSC_SEL (AC97_PSC_BASE + PSC_SEL_OFFSET)
-#define AC97_PSC_CTRL (AC97_PSC_BASE + PSC_CTRL_OFFSET)
-#define PSC_AC97CFG (AC97_PSC_BASE + PSC_AC97CFG_OFFSET)
-#define PSC_AC97MSK (AC97_PSC_BASE + PSC_AC97MSK_OFFSET)
-#define PSC_AC97PCR (AC97_PSC_BASE + PSC_AC97PCR_OFFSET)
-#define PSC_AC97STAT (AC97_PSC_BASE + PSC_AC97STAT_OFFSET)
-#define PSC_AC97EVNT (AC97_PSC_BASE + PSC_AC97EVNT_OFFSET)
-#define PSC_AC97TXRX (AC97_PSC_BASE + PSC_AC97TXRX_OFFSET)
-#define PSC_AC97CDC (AC97_PSC_BASE + PSC_AC97CDC_OFFSET)
-#define PSC_AC97RST (AC97_PSC_BASE + PSC_AC97RST_OFFSET)
-#define PSC_AC97GPO (AC97_PSC_BASE + PSC_AC97GPO_OFFSET)
-#define PSC_AC97GPI (AC97_PSC_BASE + PSC_AC97GPI_OFFSET)
+#define PSC_AC97CFG 0x00000008
+#define PSC_AC97MSK 0x0000000c
+#define PSC_AC97PCR 0x00000010
+#define PSC_AC97STAT 0x00000014
+#define PSC_AC97EVNT 0x00000018
+#define PSC_AC97TXRX 0x0000001c
+#define PSC_AC97CDC 0x00000020
+#define PSC_AC97RST 0x00000024
+#define PSC_AC97GPO 0x00000028
+#define PSC_AC97GPI 0x0000002c
/* AC97 Config Register. */
#define PSC_AC97CFG_RT_MASK (3 << 30)
@@ -192,17 +179,13 @@
#define PSC_AC97RST_SNC (1 << 0)
/* PSC in I2S Mode. */
-typedef struct psc_i2s {
- u32 psc_sel;
- u32 psc_ctrl;
- u32 psc_i2scfg;
- u32 psc_i2smsk;
- u32 psc_i2spcr;
- u32 psc_i2sstat;
- u32 psc_i2sevent;
- u32 psc_i2stxrx;
- u32 psc_i2sudf;
-} psc_i2s_t;
+#define PSC_I2SCFG 0x08
+#define PSC_I2SMASK 0x0C
+#define PSC_I2SPCR 0x10
+#define PSC_I2SSTAT 0x14
+#define PSC_I2SEVENT 0x18
+#define PSC_I2SRXTX 0x1C
+#define PSC_I2SUDF 0x20
/* I2S Config Register. */
#define PSC_I2SCFG_RT_MASK (3 << 30)
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 18f28ac..caacf95 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -25,6 +25,7 @@ config SND_SOC
# All the supported Soc's
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"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 782db21..5dafddd 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,4 @@
snd-soc-core-objs := soc-core.o soc-dapm.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
-obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
+obj-$(CONFIG_SND_SOC) += codecs/ at91/ au1x/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
new file mode 100644
index 0000000..8ef9015
--- /dev/null
+++ b/sound/soc/au1x/Kconfig
@@ -0,0 +1,36 @@
+menu "SoC Audio for the Alchemy/AMD/RMI Au1xxx"
+ depends on SOC_AU1200 || SOC_AU1550
+
+##
+## Au1200/Au1550 PSC + DBDMA
+##
+config SND_SOC_AU1XPSC
+ tristate "SoC Audio for Au1200/Au1250/Au1550"
+ depends on SND_SOC && (SOC_AU1200 || SOC_AU1550)
+ help
+ This option enables support for the Programmable Serial
+ Controllers in AC97 and I2S mode, and the Descriptor-Based DMA
+ Controller (DBDMA) as found on the Au1200/Au1250/Au1550 SoC.
+
+config SND_SOC_AU1XPSC_I2S
+ tristate
+
+config SND_SOC_AU1XPSC_AC97
+ tristate
+ select AC97_BUS
+ select SND_AC97_CODEC
+ select SND_SOC_AC97_BUS
+
+
+##
+## Boards
+##
+config SND_SOC_SAMPLE_PSC_AC97
+ tristate "Sample Au12x0/Au1550 PSC AC97 sound machine"
+ select SND_SOC_AU1XPSC_AC97
+ select SND_SOC_AC97_CODEC
+ help
+ This is a sample AC97 sound machine for use in Au12x0/Au1550
+ based systems which have audio on PSC1 (e.g. Db1200 demoboard).
+
+endmenu
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
new file mode 100644
index 0000000..6c6950b
--- /dev/null
+++ b/sound/soc/au1x/Makefile
@@ -0,0 +1,13 @@
+# Au1200/Au1550 PSC audio
+snd-soc-au1xpsc-dbdma-objs := dbdma2.o
+snd-soc-au1xpsc-i2s-objs := psc-i2s.o
+snd-soc-au1xpsc-ac97-objs := psc-ac97.o
+
+obj-$(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
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
new file mode 100644
index 0000000..f924c54
--- /dev/null
+++ b/sound/soc/au1x/dbdma2.c
@@ -0,0 +1,435 @@
+/*
+ * 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 "au1x-dbdma: " 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 DMA area (phyical area) */
+ unsigned long dma_area_s; /* start address of DMA area (phyical 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;
+};
+
+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 is_rx,
+ 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 (is_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;
+}
+
+/*
+ * Called by ALSA when the hardware params are set by application. This
+ * function can also be called multiple times and can allocate buffers
+ * (using snd_pcm_lib_* ). It's non-atomic.
+ */
+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 is_rx, ret;
+
+ ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ if (ret < 0)
+ goto out;
+
+ is_rx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+ pcd = au1xpsc_audio_pcmdma[is_rx];
+
+ 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), is_rx);
+
+ ret = au1x_pcm_dbdma_realloc(pcd, is_rx, 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;
+ int is_rx;
+
+ is_rx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+ pcd = au1xpsc_audio_pcmdma[is_rx];
+
+ au1xxx_dbdma_reset(pcd->ddma_chan);
+
+ if (is_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)
+{
+ int is_rx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+ u32 chan = au1xpsc_audio_pcmdma[is_rx]->ddma_chan;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ au1xxx_dbdma_start(chan);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ au1xxx_dbdma_stop(chan);
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static snd_pcm_uframes_t
+au1xpsc_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ int is_rx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+
+ return bytes_to_frames(substream->runtime,
+ au1xpsc_audio_pcmdma[is_rx]->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)
+{
+ int is_rx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+
+ au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[is_rx]);
+
+ 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_codec_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[0])
+ return -EBUSY;
+
+ /* TX DMA */
+ au1xpsc_audio_pcmdma[0]
+ = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
+ if (!au1xpsc_audio_pcmdma[0])
+ return -ENOMEM;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!r) {
+ ret = -ENODEV;
+ goto out1;
+ }
+ (au1xpsc_audio_pcmdma[0])->ddma_id = r->start;
+
+ /* RX DMA */
+ au1xpsc_audio_pcmdma[1]
+ = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
+ if (!au1xpsc_audio_pcmdma[1])
+ return -ENOMEM;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!r) {
+ ret = -ENODEV;
+ goto out2;
+ }
+ (au1xpsc_audio_pcmdma[1])->ddma_id = r->start;
+
+ return 0;
+
+out2:
+ kfree(au1xpsc_audio_pcmdma[1]);
+ au1xpsc_audio_pcmdma[1] = NULL;
+out1:
+ kfree(au1xpsc_audio_pcmdma[0]);
+ au1xpsc_audio_pcmdma[0] = 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[0] = NULL;
+ au1xpsc_audio_pcmdma[1] = 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>");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
new file mode 100644
index 0000000..c9cf72b
--- /dev/null
+++ b/sound/soc/au1x/psc-ac97.c
@@ -0,0 +1,378 @@
+/*
+ * 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_RD (1<<25)
+
+#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
+
+/* 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(AC97_RD | ((reg & 127) << 16), AC97_CDC(pscdata));
+ au_sync();
+
+ tmo = 1000;
+ while ((!(au_readl(AC97_EVNT(pscdata)) & (1<<24))) && --tmo)
+ udelay(2);
+
+ if (!tmo)
+ data = 0xffff;
+ else
+ data = au_readl(AC97_CDC(pscdata)) & 0xffff;
+
+ au_writel(1<<24, 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(((reg & 127) << 16) | (val & 0xffff), AC97_CDC(pscdata));
+ au_sync();
+ tmo = 1000;
+ while ((!(au_readl(AC97_EVNT(pscdata)) & (1 << 24))) && --tmo)
+ au_sync();
+
+ au_writel(1 << 24, 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(1, 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, PSC_CTRL(pscdata));
+
+ /* issue cold reset */
+ au_writel(2, AC97_RST(pscdata));
+ au_sync();
+ msleep(500);
+ au_writel(0, AC97_RST(pscdata));
+ au_sync();
+
+ /* enable PSC */
+ au_writel(3, PSC_CTRL(pscdata));
+ au_sync();
+
+ /* wait for PSC to indicate it's ready */
+ i = 100000;
+ while (((au_readl(AC97_STAT(pscdata)) & 1) == 0) && (--i))
+ au_sync();
+
+ if (i == 0) {
+ printk(KERN_ALERT "psc-ac97: PSC not ready!\n");
+ return;
+ }
+
+ /* enable the ac97 function */
+ au_writel(pscdata->cfg | 0x04000000, AC97_CFG(pscdata));
+ au_sync();
+
+ /* wait for AC97 core to become ready */
+ i = 100000;
+ while (((au_readl(AC97_STAT(pscdata)) & 2) == 0) && (--i))
+ au_sync();
+ if (i == 0)
+ printk(KERN_ALERT "psc-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)
+{
+ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
+ unsigned long r;
+ int chans, recv;
+
+ chans = params_channels(params);
+ recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+
+ /* need to disable the controller before changing any other
+ * AC97CFG reg contents
+ */
+ r = au_readl(AC97_CFG(pscdata));
+ au_writel(r & ~(1<<26), AC97_CFG(pscdata));
+ au_sync();
+
+ /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */
+ r &= ~(0xf << 21);
+ r |= (((params->msbits-2)>>1) & 0xf) << 21;
+
+ /* channels */
+ r |= (3 << (1 + (recv ? 0 : 10))); /* stereo pair */
+
+ /* set FIFO params: max fifo threshold, 8 slots TX/RX */
+ r |= (3<<30) | (3<<28);
+
+ /* finally enable the AC97 controller again */
+ au_writel(r | (1<<26), AC97_CFG(pscdata));
+ au_sync();
+
+ return 0;
+}
+
+static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
+ int ret, rcv;
+
+ rcv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+ ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ au_writel(1 << (rcv ? 4 : 0), AC97_PCR(pscdata));
+ au_sync();
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ au_writel(1 << (rcv ? 5 : 1), AC97_PCR(pscdata));
+ au_sync();
+ break;
+ default:
+ ret = -EINVAL;
+ }
+ return ret;
+}
+
+static int au1xpsc_ac97_probe(struct platform_device *pdev)
+{
+ 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;
+
+ /* 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();
+ /* enable PSC */
+ au_writel(PSC_CTRL_ENABLE, PSC_CTRL(au1xpsc_ac97_workdata));
+ au_sync();
+
+ 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)
+{
+ /* 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_cpu_dai *cpu_dai)
+{
+ /* save interesting registers and disable PSC */
+ au1xpsc_ac97_workdata->pm[0] =
+ au_readl(PSC_SEL(au1xpsc_ac97_workdata));
+ au1xpsc_ac97_workdata->pm[1] =
+ au_readl(AC97_CFG(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_cpu_dai *cpu_dai)
+{
+ int i;
+
+ au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata));
+ au_sync();
+ au_writel(0, PSC_SEL(au1xpsc_ac97_workdata));
+ au_sync();
+
+ au_writel(au1xpsc_ac97_workdata->pm[0],
+ PSC_SEL(au1xpsc_ac97_workdata));
+ au_sync();
+ au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
+ au_sync();
+
+ /* enable PSC */
+ au_writel(PSC_CTRL_ENABLE, PSC_CTRL(au1xpsc_ac97_workdata));
+ au_sync();
+
+ /* wait for PSC to indicate it's ready */
+ i = 100000;
+ while ((!(au_readl(AC97_STAT(au1xpsc_ac97_workdata)) & 1)) && (--i))
+ au_sync();
+
+ /* after this point the ac97 core will cold-reset the codec.
+ * During cold-reset the code will write pre-defined data to
+ * the config register.
+ */
+ au1xpsc_ac97_workdata->cfg = au1xpsc_ac97_workdata->pm[1];
+
+ return 0;
+}
+
+struct snd_soc_cpu_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>");
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
new file mode 100644
index 0000000..42d4488
--- /dev/null
+++ b/sound/soc/au1x/psc-i2s.c
@@ -0,0 +1,426 @@
+/*
+ * 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.
+ */
+
+#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*/)
+
+
+/* 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_cpu_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 &= ~((1<<9)|(1<<10)); /* MSB (left-) justified*/
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ ct |= (1<<9); /* enable I2S mode */
+ break;
+ case SND_SOC_DAIFMT_MSB:
+ break;
+ case SND_SOC_DAIFMT_LSB:
+ ct |= (1<<10); /* LSB (right-) justified */
+ break;
+ default:
+ goto out;
+ }
+
+ ct &= ~((1 << 12) | (1 << 15)); /* IB-IF */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ ct |= (1<<12) | (1<<15); /* NF: left = low */
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ ct |= (1<<12);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ ct |= (1<<15); /* IB-NF */
+ 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 |= (1<<0); /* PSC I2S slave mode */
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */
+ ct &= ~(1<<0); /* 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 */
+ /* FIXME: should probably ONLY check if pscdata->rate is != 0 */
+ stat = au_readl(I2S_STAT(pscdata));
+ if (stat & (3<<4)) {
+ /* already active, check settings (don't trust pscdata->cfg) */
+ cfgbits = au_readl(I2S_CFG(pscdata));
+ cfgbits = ((cfgbits >> 4) & 0x1f) + 1;
+ if (cfgbits != params->msbits)
+ return -EINVAL;
+
+ /* FIXME: does ALSA/ASoC already check? */
+ if (params_rate(params) != pscdata->rate)
+ return -EINVAL;
+
+ } else {
+ /* set sample bitdepth */
+ pscdata->cfg &= ~(0x1f << 4);
+ pscdata->cfg |= (((params->msbits - 1) & 0x1f) << 4);
+ /* 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;
+
+ au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata));
+ au_sync();
+
+ /* wait for PSC unit to become ready */
+ tmo = 1000000;
+ while (!(au_readl(I2S_STAT(pscdata)) & 1) && tmo)
+ tmo--;
+
+ if (!tmo)
+ return -ETIMEDOUT;
+
+ /* configure the I2S controller; need to disable it first. */
+ au_writel(0, I2S_CFG(pscdata));
+ au_sync();
+
+ /* start I2S controller: config | max_tx_thresh | max_rx_thresh | enable */
+ au_writel(pscdata->cfg | (1<<26), I2S_CFG(pscdata));
+ au_sync();
+
+ /* wait for I2S controller to become ready */
+ tmo = 1000000;
+ while (!(au_readl(I2S_STAT(pscdata)) & 2) && tmo)
+ tmo--;
+
+ return (tmo == 0) ? -ETIMEDOUT : 0;
+}
+
+static int au1xpsc_i2s_start(struct au1xpsc_audio_data *pscdata, int play)
+{
+ unsigned long tmo;
+ int ret;
+
+ ret = 0;
+
+ /* if both TX and RX are idle, configure the PSC */
+ if ((au_readl(I2S_STAT(pscdata)) & ((1<<4)|(1<<5))) == 0) {
+ ret = au1xpsc_i2s_configure(pscdata);
+ if (ret)
+ goto out;
+ }
+
+ /* clear fifo */
+ au_writel(play ? (1<<2) : (1<<6), I2S_PCR(pscdata));
+ au_sync();
+
+ /* and start */
+ au_writel(play ? (1<<0) : (1<<4), I2S_PCR(pscdata));
+ au_sync();
+
+ /* wait for start confirmation */
+ tmo = 1000000;
+ while ((0 == (au_readl(I2S_STAT(pscdata)) & (play ? (1<<4) : (1<<5)))) && tmo)
+ tmo--;
+
+ if (!tmo) {
+ au_writel(play ? (1<<1) : (1<<5), I2S_PCR(pscdata));
+ au_sync();
+ ret = -ETIMEDOUT;
+ }
+out:
+ return ret;
+}
+
+static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int play)
+{
+ unsigned long tmo, stat;
+
+ au_writel(play ? (1<<1) : (1<<5), I2S_PCR(pscdata));
+ au_sync();
+ /* wait for stop confirmation */
+ tmo = 1000000;
+ do {
+ stat = au_readl(I2S_STAT(pscdata));
+ tmo--;
+ } while ((stat & (play ? (1<<4) : (1<<5))) && tmo);
+
+ /* if both TX and RX are idle, disable the I2S and PSC */
+ stat = au_readl(I2S_STAT(pscdata)) & (3<<4);
+ if (!stat) {
+ /* disable I2S controller */
+ au_writel(0, I2S_CFG(pscdata));
+ au_sync();
+
+ /* suspend PSC */
+ au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
+ au_sync();
+
+ pscdata->rate = 0;
+ /* don't change pscdata->cfg! PM depends on it! */
+ }
+ return 0;
+}
+
+static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
+ int ret, play;
+
+ play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ ret = au1xpsc_i2s_start(pscdata, play);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ ret = au1xpsc_i2s_stop(pscdata, play);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+ return ret;
+}
+
+static int au1xpsc_i2s_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct resource *r;
+ unsigned long sel;
+
+ 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();
+ au_writel(PSC_CTRL_ENABLE, PSC_CTRL(au1xpsc_i2s_workdata));
+ au_sync();
+
+ /* preconfigure: set max rx/tx fifo depths */
+ au1xpsc_i2s_workdata->cfg |= (3<<30) | (3<<28);
+
+ /* controller might not become ready if it is clocked by the codec;
+ * codec is initialized later on and parameters are set even later
+ */
+
+ 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)
+{
+ 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_cpu_dai *cpu_dai)
+{
+ /* save interesting registers and disable PSC */
+ au1xpsc_i2s_workdata->pm[0] =
+ au_readl(PSC_SEL(au1xpsc_i2s_workdata));
+ au1xpsc_i2s_workdata->pm[1] =
+ au_readl(I2S_CFG(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_cpu_dai *cpu_dai)
+{
+ 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();
+
+ /* enable PSC */
+ au_writel(PSC_CTRL_ENABLE, PSC_CTRL(au1xpsc_i2s_workdata));
+ au_sync();
+
+ /* same comment as in probe() callback also applies here */
+
+ /* write back saved config */
+ au_writel(au1xpsc_i2s_workdata->pm[1],
+ I2S_CFG(au1xpsc_i2s_workdata));
+ au_sync();
+
+ return 0;
+}
+
+struct snd_soc_cpu_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,},
+ .capture = {
+ .rates = AU1XPSC_I2S_RATES,
+ .formats = AU1XPSC_I2S_FMTS,
+ .channels_min = 2,
+ .channels_max = 8,},
+ .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>");
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
new file mode 100644
index 0000000..98e29eb
--- /dev/null
+++ b/sound/soc/au1x/psc.h
@@ -0,0 +1,48 @@
+/*
+ * 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_cpu_dai au1xpsc_ac97_dai;
+extern struct snd_soc_cpu_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;
+ int irq;
+ struct resource *ioarea;
+
+ unsigned long cfg;
+ unsigned long rate;
+
+ unsigned long pm[2];
+};
+
+/* 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)
+#define I2S_CFG(x) ((unsigned long)((x)->mmio) + PSC_I2SCFG)
+#define I2S_PCR(x) ((unsigned long)((x)->mmio) + PSC_I2SPCR)
+#define AC97_CFG(x) ((unsigned long)((x)->mmio) + PSC_AC97CFG)
+#define AC97_CDC(x) ((unsigned long)((x)->mmio) + PSC_AC97CDC)
+#define AC97_EVNT(x) ((unsigned long)((x)->mmio) + PSC_AC97EVNT)
+#define AC97_PCR(x) ((unsigned long)((x)->mmio) + PSC_AC97PCR)
+#define AC97_RST(x) ((unsigned long)((x)->mmio) + PSC_AC97RST)
+#define AC97_STAT(x) ((unsigned long)((x)->mmio) + PSC_AC97STAT)
+
+#endif
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
new file mode 100644
index 0000000..fce81da
--- /dev/null
+++ b/sound/soc/au1x/sample-ac97.c
@@ -0,0 +1,144 @@
+/*
+ * 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_endpoints(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 = NULL;
+
+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>");
--
1.5.6.1
^ permalink raw reply related [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-03 10:09 ` ASoC: Au12x0/Au1550 PSC Audio support Manuel Lauss
@ 2008-07-04 15:13 ` Takashi Iwai
2008-07-04 16:24 ` Manuel Lauss
0 siblings, 1 reply; 20+ messages in thread
From: Takashi Iwai @ 2008-07-04 15:13 UTC (permalink / raw)
To: Mark Brown; +Cc: Manuel Lauss, alsa-devel
At Thu, 3 Jul 2008 12:09:25 +0200,
Manuel Lauss wrote:
>
> On Thu, Jul 03, 2008 at 10:55:47AM +0100, Liam Girdwood wrote:
> > On Thu, 2008-07-03 at 11:53 +0200, Manuel Lauss wrote:
> > > On Thu, Jul 03, 2008 at 10:29:19AM +0100, Liam Girdwood wrote:
> > > >
> > > > Btw, any plans to fix or remove some of the 'Fixmes' in your other ASoC
> > > > code. We could upstream the Au1x000 stuff after a little cleanup and
> > > > resolution of the fixmes.
> > >
> > > I've updated the code considerably since the last year. It's still ASoC v1,
> > > and lots of the FIXMEs in the ac97 part won't disappear until it's moved to
> > > ASoC v2 (I hope, didn't investigate).
> > >
> > > If you're interested I can send you a patch (against current linus' HEAD)
> > > with the newest, most awesome au1xxx psc asoc code ;-)
> > >
> >
> > Yes please.
>
> Here you go, apply on top of linus' head + the ac97 PM patch.
>
> ---
>
> From: Manuel Lauss <mano@roarinelk.homelinux.net>
>
> 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>
Shall I apply this one? At a quick glance, the patch looks OK.
thanks,
Takashi
> ---
> include/asm-mips/mach-au1x00/au1xxx_psc.h | 51 ++---
> sound/soc/Kconfig | 1 +
> sound/soc/Makefile | 2 +-
> sound/soc/au1x/Kconfig | 36 +++
> sound/soc/au1x/Makefile | 13 +
> sound/soc/au1x/dbdma2.c | 435 +++++++++++++++++++++++++++++
> sound/soc/au1x/psc-ac97.c | 378 +++++++++++++++++++++++++
> sound/soc/au1x/psc-i2s.c | 426 ++++++++++++++++++++++++++++
> sound/soc/au1x/psc.h | 48 ++++
> sound/soc/au1x/sample-ac97.c | 144 ++++++++++
> 10 files changed, 1499 insertions(+), 35 deletions(-)
> create mode 100644 sound/soc/au1x/Kconfig
> create mode 100644 sound/soc/au1x/Makefile
> create mode 100644 sound/soc/au1x/dbdma2.c
> create mode 100644 sound/soc/au1x/psc-ac97.c
> create mode 100644 sound/soc/au1x/psc-i2s.c
> create mode 100644 sound/soc/au1x/psc.h
> create mode 100644 sound/soc/au1x/sample-ac97.c
>
> diff --git a/include/asm-mips/mach-au1x00/au1xxx_psc.h b/include/asm-mips/mach-au1x00/au1xxx_psc.h
> index dae4eca..912768d 100644
> --- a/include/asm-mips/mach-au1x00/au1xxx_psc.h
> +++ b/include/asm-mips/mach-au1x00/au1xxx_psc.h
> @@ -69,29 +69,16 @@
> #define PSC_CTRL_ENABLE 3
>
> /* AC97 Registers. */
> -#define PSC_AC97CFG_OFFSET 0x00000008
> -#define PSC_AC97MSK_OFFSET 0x0000000c
> -#define PSC_AC97PCR_OFFSET 0x00000010
> -#define PSC_AC97STAT_OFFSET 0x00000014
> -#define PSC_AC97EVNT_OFFSET 0x00000018
> -#define PSC_AC97TXRX_OFFSET 0x0000001c
> -#define PSC_AC97CDC_OFFSET 0x00000020
> -#define PSC_AC97RST_OFFSET 0x00000024
> -#define PSC_AC97GPO_OFFSET 0x00000028
> -#define PSC_AC97GPI_OFFSET 0x0000002c
> -
> -#define AC97_PSC_SEL (AC97_PSC_BASE + PSC_SEL_OFFSET)
> -#define AC97_PSC_CTRL (AC97_PSC_BASE + PSC_CTRL_OFFSET)
> -#define PSC_AC97CFG (AC97_PSC_BASE + PSC_AC97CFG_OFFSET)
> -#define PSC_AC97MSK (AC97_PSC_BASE + PSC_AC97MSK_OFFSET)
> -#define PSC_AC97PCR (AC97_PSC_BASE + PSC_AC97PCR_OFFSET)
> -#define PSC_AC97STAT (AC97_PSC_BASE + PSC_AC97STAT_OFFSET)
> -#define PSC_AC97EVNT (AC97_PSC_BASE + PSC_AC97EVNT_OFFSET)
> -#define PSC_AC97TXRX (AC97_PSC_BASE + PSC_AC97TXRX_OFFSET)
> -#define PSC_AC97CDC (AC97_PSC_BASE + PSC_AC97CDC_OFFSET)
> -#define PSC_AC97RST (AC97_PSC_BASE + PSC_AC97RST_OFFSET)
> -#define PSC_AC97GPO (AC97_PSC_BASE + PSC_AC97GPO_OFFSET)
> -#define PSC_AC97GPI (AC97_PSC_BASE + PSC_AC97GPI_OFFSET)
> +#define PSC_AC97CFG 0x00000008
> +#define PSC_AC97MSK 0x0000000c
> +#define PSC_AC97PCR 0x00000010
> +#define PSC_AC97STAT 0x00000014
> +#define PSC_AC97EVNT 0x00000018
> +#define PSC_AC97TXRX 0x0000001c
> +#define PSC_AC97CDC 0x00000020
> +#define PSC_AC97RST 0x00000024
> +#define PSC_AC97GPO 0x00000028
> +#define PSC_AC97GPI 0x0000002c
>
> /* AC97 Config Register. */
> #define PSC_AC97CFG_RT_MASK (3 << 30)
> @@ -192,17 +179,13 @@
> #define PSC_AC97RST_SNC (1 << 0)
>
> /* PSC in I2S Mode. */
> -typedef struct psc_i2s {
> - u32 psc_sel;
> - u32 psc_ctrl;
> - u32 psc_i2scfg;
> - u32 psc_i2smsk;
> - u32 psc_i2spcr;
> - u32 psc_i2sstat;
> - u32 psc_i2sevent;
> - u32 psc_i2stxrx;
> - u32 psc_i2sudf;
> -} psc_i2s_t;
> +#define PSC_I2SCFG 0x08
> +#define PSC_I2SMASK 0x0C
> +#define PSC_I2SPCR 0x10
> +#define PSC_I2SSTAT 0x14
> +#define PSC_I2SEVENT 0x18
> +#define PSC_I2SRXTX 0x1C
> +#define PSC_I2SUDF 0x20
>
> /* I2S Config Register. */
> #define PSC_I2SCFG_RT_MASK (3 << 30)
> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
> index 18f28ac..caacf95 100644
> --- a/sound/soc/Kconfig
> +++ b/sound/soc/Kconfig
> @@ -25,6 +25,7 @@ config SND_SOC
>
> # All the supported Soc's
> 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"
> diff --git a/sound/soc/Makefile b/sound/soc/Makefile
> index 782db21..5dafddd 100644
> --- a/sound/soc/Makefile
> +++ b/sound/soc/Makefile
> @@ -1,4 +1,4 @@
> snd-soc-core-objs := soc-core.o soc-dapm.o
>
> obj-$(CONFIG_SND_SOC) += snd-soc-core.o
> -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
> +obj-$(CONFIG_SND_SOC) += codecs/ at91/ au1x/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
> diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
> new file mode 100644
> index 0000000..8ef9015
> --- /dev/null
> +++ b/sound/soc/au1x/Kconfig
> @@ -0,0 +1,36 @@
> +menu "SoC Audio for the Alchemy/AMD/RMI Au1xxx"
> + depends on SOC_AU1200 || SOC_AU1550
> +
> +##
> +## Au1200/Au1550 PSC + DBDMA
> +##
> +config SND_SOC_AU1XPSC
> + tristate "SoC Audio for Au1200/Au1250/Au1550"
> + depends on SND_SOC && (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"
> + 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).
> +
> +endmenu
> diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
> new file mode 100644
> index 0000000..6c6950b
> --- /dev/null
> +++ b/sound/soc/au1x/Makefile
> @@ -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
> diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
> new file mode 100644
> index 0000000..f924c54
> --- /dev/null
> +++ b/sound/soc/au1x/dbdma2.c
> @@ -0,0 +1,435 @@
> +/*
> + * 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 "au1x-dbdma: " 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 DMA area (phyical area) */
> + unsigned long dma_area_s; /* start address of DMA area (phyical 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;
> +};
> +
> +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 is_rx,
> + 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 (is_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;
> +}
> +
> +/*
> + * Called by ALSA when the hardware params are set by application. This
> + * function can also be called multiple times and can allocate buffers
> + * (using snd_pcm_lib_* ). It's non-atomic.
> + */
> +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 is_rx, ret;
> +
> + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
> + if (ret < 0)
> + goto out;
> +
> + is_rx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
> + pcd = au1xpsc_audio_pcmdma[is_rx];
> +
> + 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), is_rx);
> +
> + ret = au1x_pcm_dbdma_realloc(pcd, is_rx, 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;
> + int is_rx;
> +
> + is_rx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
> + pcd = au1xpsc_audio_pcmdma[is_rx];
> +
> + au1xxx_dbdma_reset(pcd->ddma_chan);
> +
> + if (is_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)
> +{
> + int is_rx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
> + u32 chan = au1xpsc_audio_pcmdma[is_rx]->ddma_chan;
> +
> + switch (cmd) {
> + case SNDRV_PCM_TRIGGER_START:
> + case SNDRV_PCM_TRIGGER_RESUME:
> + au1xxx_dbdma_start(chan);
> + break;
> + case SNDRV_PCM_TRIGGER_STOP:
> + case SNDRV_PCM_TRIGGER_SUSPEND:
> + au1xxx_dbdma_stop(chan);
> + break;
> + default:
> + return -EINVAL;
> + }
> + return 0;
> +}
> +
> +static snd_pcm_uframes_t
> +au1xpsc_pcm_pointer(struct snd_pcm_substream *substream)
> +{
> + int is_rx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
> +
> + return bytes_to_frames(substream->runtime,
> + au1xpsc_audio_pcmdma[is_rx]->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)
> +{
> + int is_rx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
> +
> + au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[is_rx]);
> +
> + 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_codec_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[0])
> + return -EBUSY;
> +
> + /* TX DMA */
> + au1xpsc_audio_pcmdma[0]
> + = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
> + if (!au1xpsc_audio_pcmdma[0])
> + return -ENOMEM;
> +
> + r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
> + if (!r) {
> + ret = -ENODEV;
> + goto out1;
> + }
> + (au1xpsc_audio_pcmdma[0])->ddma_id = r->start;
> +
> + /* RX DMA */
> + au1xpsc_audio_pcmdma[1]
> + = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
> + if (!au1xpsc_audio_pcmdma[1])
> + return -ENOMEM;
> +
> + r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
> + if (!r) {
> + ret = -ENODEV;
> + goto out2;
> + }
> + (au1xpsc_audio_pcmdma[1])->ddma_id = r->start;
> +
> + return 0;
> +
> +out2:
> + kfree(au1xpsc_audio_pcmdma[1]);
> + au1xpsc_audio_pcmdma[1] = NULL;
> +out1:
> + kfree(au1xpsc_audio_pcmdma[0]);
> + au1xpsc_audio_pcmdma[0] = 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[0] = NULL;
> + au1xpsc_audio_pcmdma[1] = 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>");
> diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
> new file mode 100644
> index 0000000..c9cf72b
> --- /dev/null
> +++ b/sound/soc/au1x/psc-ac97.c
> @@ -0,0 +1,378 @@
> +/*
> + * 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_RD (1<<25)
> +
> +#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
> +
> +/* 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(AC97_RD | ((reg & 127) << 16), AC97_CDC(pscdata));
> + au_sync();
> +
> + tmo = 1000;
> + while ((!(au_readl(AC97_EVNT(pscdata)) & (1<<24))) && --tmo)
> + udelay(2);
> +
> + if (!tmo)
> + data = 0xffff;
> + else
> + data = au_readl(AC97_CDC(pscdata)) & 0xffff;
> +
> + au_writel(1<<24, 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(((reg & 127) << 16) | (val & 0xffff), AC97_CDC(pscdata));
> + au_sync();
> + tmo = 1000;
> + while ((!(au_readl(AC97_EVNT(pscdata)) & (1 << 24))) && --tmo)
> + au_sync();
> +
> + au_writel(1 << 24, 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(1, 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, PSC_CTRL(pscdata));
> +
> + /* issue cold reset */
> + au_writel(2, AC97_RST(pscdata));
> + au_sync();
> + msleep(500);
> + au_writel(0, AC97_RST(pscdata));
> + au_sync();
> +
> + /* enable PSC */
> + au_writel(3, PSC_CTRL(pscdata));
> + au_sync();
> +
> + /* wait for PSC to indicate it's ready */
> + i = 100000;
> + while (((au_readl(AC97_STAT(pscdata)) & 1) == 0) && (--i))
> + au_sync();
> +
> + if (i == 0) {
> + printk(KERN_ALERT "psc-ac97: PSC not ready!\n");
> + return;
> + }
> +
> + /* enable the ac97 function */
> + au_writel(pscdata->cfg | 0x04000000, AC97_CFG(pscdata));
> + au_sync();
> +
> + /* wait for AC97 core to become ready */
> + i = 100000;
> + while (((au_readl(AC97_STAT(pscdata)) & 2) == 0) && (--i))
> + au_sync();
> + if (i == 0)
> + printk(KERN_ALERT "psc-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)
> +{
> + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
> + unsigned long r;
> + int chans, recv;
> +
> + chans = params_channels(params);
> + recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
> +
> + /* need to disable the controller before changing any other
> + * AC97CFG reg contents
> + */
> + r = au_readl(AC97_CFG(pscdata));
> + au_writel(r & ~(1<<26), AC97_CFG(pscdata));
> + au_sync();
> +
> + /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */
> + r &= ~(0xf << 21);
> + r |= (((params->msbits-2)>>1) & 0xf) << 21;
> +
> + /* channels */
> + r |= (3 << (1 + (recv ? 0 : 10))); /* stereo pair */
> +
> + /* set FIFO params: max fifo threshold, 8 slots TX/RX */
> + r |= (3<<30) | (3<<28);
> +
> + /* finally enable the AC97 controller again */
> + au_writel(r | (1<<26), AC97_CFG(pscdata));
> + au_sync();
> +
> + return 0;
> +}
> +
> +static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
> + int cmd)
> +{
> + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
> + int ret, rcv;
> +
> + rcv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
> + ret = 0;
> +
> + switch (cmd) {
> + case SNDRV_PCM_TRIGGER_START:
> + case SNDRV_PCM_TRIGGER_RESUME:
> + au_writel(1 << (rcv ? 4 : 0), AC97_PCR(pscdata));
> + au_sync();
> + break;
> + case SNDRV_PCM_TRIGGER_STOP:
> + case SNDRV_PCM_TRIGGER_SUSPEND:
> + au_writel(1 << (rcv ? 5 : 1), AC97_PCR(pscdata));
> + au_sync();
> + break;
> + default:
> + ret = -EINVAL;
> + }
> + return ret;
> +}
> +
> +static int au1xpsc_ac97_probe(struct platform_device *pdev)
> +{
> + 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;
> +
> + /* 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();
> + /* enable PSC */
> + au_writel(PSC_CTRL_ENABLE, PSC_CTRL(au1xpsc_ac97_workdata));
> + au_sync();
> +
> + 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)
> +{
> + /* 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_cpu_dai *cpu_dai)
> +{
> + /* save interesting registers and disable PSC */
> + au1xpsc_ac97_workdata->pm[0] =
> + au_readl(PSC_SEL(au1xpsc_ac97_workdata));
> + au1xpsc_ac97_workdata->pm[1] =
> + au_readl(AC97_CFG(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_cpu_dai *cpu_dai)
> +{
> + int i;
> +
> + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata));
> + au_sync();
> + au_writel(0, PSC_SEL(au1xpsc_ac97_workdata));
> + au_sync();
> +
> + au_writel(au1xpsc_ac97_workdata->pm[0],
> + PSC_SEL(au1xpsc_ac97_workdata));
> + au_sync();
> + au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
> + au_sync();
> +
> + /* enable PSC */
> + au_writel(PSC_CTRL_ENABLE, PSC_CTRL(au1xpsc_ac97_workdata));
> + au_sync();
> +
> + /* wait for PSC to indicate it's ready */
> + i = 100000;
> + while ((!(au_readl(AC97_STAT(au1xpsc_ac97_workdata)) & 1)) && (--i))
> + au_sync();
> +
> + /* after this point the ac97 core will cold-reset the codec.
> + * During cold-reset the code will write pre-defined data to
> + * the config register.
> + */
> + au1xpsc_ac97_workdata->cfg = au1xpsc_ac97_workdata->pm[1];
> +
> + return 0;
> +}
> +
> +struct snd_soc_cpu_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>");
> diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
> new file mode 100644
> index 0000000..42d4488
> --- /dev/null
> +++ b/sound/soc/au1x/psc-i2s.c
> @@ -0,0 +1,426 @@
> +/*
> + * 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.
> + */
> +
> +#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*/)
> +
> +
> +/* 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_cpu_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 &= ~((1<<9)|(1<<10)); /* MSB (left-) justified*/
> + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
> + case SND_SOC_DAIFMT_I2S:
> + ct |= (1<<9); /* enable I2S mode */
> + break;
> + case SND_SOC_DAIFMT_MSB:
> + break;
> + case SND_SOC_DAIFMT_LSB:
> + ct |= (1<<10); /* LSB (right-) justified */
> + break;
> + default:
> + goto out;
> + }
> +
> + ct &= ~((1 << 12) | (1 << 15)); /* IB-IF */
> + switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
> + case SND_SOC_DAIFMT_NB_NF:
> + ct |= (1<<12) | (1<<15); /* NF: left = low */
> + break;
> + case SND_SOC_DAIFMT_NB_IF:
> + ct |= (1<<12);
> + break;
> + case SND_SOC_DAIFMT_IB_NF:
> + ct |= (1<<15); /* IB-NF */
> + 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 |= (1<<0); /* PSC I2S slave mode */
> + break;
> + case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */
> + ct &= ~(1<<0); /* 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 */
> + /* FIXME: should probably ONLY check if pscdata->rate is != 0 */
> + stat = au_readl(I2S_STAT(pscdata));
> + if (stat & (3<<4)) {
> + /* already active, check settings (don't trust pscdata->cfg) */
> + cfgbits = au_readl(I2S_CFG(pscdata));
> + cfgbits = ((cfgbits >> 4) & 0x1f) + 1;
> + if (cfgbits != params->msbits)
> + return -EINVAL;
> +
> + /* FIXME: does ALSA/ASoC already check? */
> + if (params_rate(params) != pscdata->rate)
> + return -EINVAL;
> +
> + } else {
> + /* set sample bitdepth */
> + pscdata->cfg &= ~(0x1f << 4);
> + pscdata->cfg |= (((params->msbits - 1) & 0x1f) << 4);
> + /* 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;
> +
> + au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata));
> + au_sync();
> +
> + /* wait for PSC unit to become ready */
> + tmo = 1000000;
> + while (!(au_readl(I2S_STAT(pscdata)) & 1) && tmo)
> + tmo--;
> +
> + if (!tmo)
> + return -ETIMEDOUT;
> +
> + /* configure the I2S controller; need to disable it first. */
> + au_writel(0, I2S_CFG(pscdata));
> + au_sync();
> +
> + /* start I2S controller: config | max_tx_thresh | max_rx_thresh | enable */
> + au_writel(pscdata->cfg | (1<<26), I2S_CFG(pscdata));
> + au_sync();
> +
> + /* wait for I2S controller to become ready */
> + tmo = 1000000;
> + while (!(au_readl(I2S_STAT(pscdata)) & 2) && tmo)
> + tmo--;
> +
> + return (tmo == 0) ? -ETIMEDOUT : 0;
> +}
> +
> +static int au1xpsc_i2s_start(struct au1xpsc_audio_data *pscdata, int play)
> +{
> + unsigned long tmo;
> + int ret;
> +
> + ret = 0;
> +
> + /* if both TX and RX are idle, configure the PSC */
> + if ((au_readl(I2S_STAT(pscdata)) & ((1<<4)|(1<<5))) == 0) {
> + ret = au1xpsc_i2s_configure(pscdata);
> + if (ret)
> + goto out;
> + }
> +
> + /* clear fifo */
> + au_writel(play ? (1<<2) : (1<<6), I2S_PCR(pscdata));
> + au_sync();
> +
> + /* and start */
> + au_writel(play ? (1<<0) : (1<<4), I2S_PCR(pscdata));
> + au_sync();
> +
> + /* wait for start confirmation */
> + tmo = 1000000;
> + while ((0 == (au_readl(I2S_STAT(pscdata)) & (play ? (1<<4) : (1<<5)))) && tmo)
> + tmo--;
> +
> + if (!tmo) {
> + au_writel(play ? (1<<1) : (1<<5), I2S_PCR(pscdata));
> + au_sync();
> + ret = -ETIMEDOUT;
> + }
> +out:
> + return ret;
> +}
> +
> +static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int play)
> +{
> + unsigned long tmo, stat;
> +
> + au_writel(play ? (1<<1) : (1<<5), I2S_PCR(pscdata));
> + au_sync();
> + /* wait for stop confirmation */
> + tmo = 1000000;
> + do {
> + stat = au_readl(I2S_STAT(pscdata));
> + tmo--;
> + } while ((stat & (play ? (1<<4) : (1<<5))) && tmo);
> +
> + /* if both TX and RX are idle, disable the I2S and PSC */
> + stat = au_readl(I2S_STAT(pscdata)) & (3<<4);
> + if (!stat) {
> + /* disable I2S controller */
> + au_writel(0, I2S_CFG(pscdata));
> + au_sync();
> +
> + /* suspend PSC */
> + au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
> + au_sync();
> +
> + pscdata->rate = 0;
> + /* don't change pscdata->cfg! PM depends on it! */
> + }
> + return 0;
> +}
> +
> +static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
> +{
> + struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
> + int ret, play;
> +
> + play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
> +
> + switch (cmd) {
> + case SNDRV_PCM_TRIGGER_START:
> + case SNDRV_PCM_TRIGGER_RESUME:
> + ret = au1xpsc_i2s_start(pscdata, play);
> + break;
> + case SNDRV_PCM_TRIGGER_STOP:
> + case SNDRV_PCM_TRIGGER_SUSPEND:
> + ret = au1xpsc_i2s_stop(pscdata, play);
> + break;
> + default:
> + ret = -EINVAL;
> + }
> + return ret;
> +}
> +
> +static int au1xpsc_i2s_probe(struct platform_device *pdev)
> +{
> + int ret;
> + struct resource *r;
> + unsigned long sel;
> +
> + 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();
> + au_writel(PSC_CTRL_ENABLE, PSC_CTRL(au1xpsc_i2s_workdata));
> + au_sync();
> +
> + /* preconfigure: set max rx/tx fifo depths */
> + au1xpsc_i2s_workdata->cfg |= (3<<30) | (3<<28);
> +
> + /* controller might not become ready if it is clocked by the codec;
> + * codec is initialized later on and parameters are set even later
> + */
> +
> + 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)
> +{
> + 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_cpu_dai *cpu_dai)
> +{
> + /* save interesting registers and disable PSC */
> + au1xpsc_i2s_workdata->pm[0] =
> + au_readl(PSC_SEL(au1xpsc_i2s_workdata));
> + au1xpsc_i2s_workdata->pm[1] =
> + au_readl(I2S_CFG(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_cpu_dai *cpu_dai)
> +{
> + 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();
> +
> + /* enable PSC */
> + au_writel(PSC_CTRL_ENABLE, PSC_CTRL(au1xpsc_i2s_workdata));
> + au_sync();
> +
> + /* same comment as in probe() callback also applies here */
> +
> + /* write back saved config */
> + au_writel(au1xpsc_i2s_workdata->pm[1],
> + I2S_CFG(au1xpsc_i2s_workdata));
> + au_sync();
> +
> + return 0;
> +}
> +
> +struct snd_soc_cpu_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,},
> + .capture = {
> + .rates = AU1XPSC_I2S_RATES,
> + .formats = AU1XPSC_I2S_FMTS,
> + .channels_min = 2,
> + .channels_max = 8,},
> + .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>");
> diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
> new file mode 100644
> index 0000000..98e29eb
> --- /dev/null
> +++ b/sound/soc/au1x/psc.h
> @@ -0,0 +1,48 @@
> +/*
> + * 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_cpu_dai au1xpsc_ac97_dai;
> +extern struct snd_soc_cpu_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;
> + int irq;
> + struct resource *ioarea;
> +
> + unsigned long cfg;
> + unsigned long rate;
> +
> + unsigned long pm[2];
> +};
> +
> +/* 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)
> +#define I2S_CFG(x) ((unsigned long)((x)->mmio) + PSC_I2SCFG)
> +#define I2S_PCR(x) ((unsigned long)((x)->mmio) + PSC_I2SPCR)
> +#define AC97_CFG(x) ((unsigned long)((x)->mmio) + PSC_AC97CFG)
> +#define AC97_CDC(x) ((unsigned long)((x)->mmio) + PSC_AC97CDC)
> +#define AC97_EVNT(x) ((unsigned long)((x)->mmio) + PSC_AC97EVNT)
> +#define AC97_PCR(x) ((unsigned long)((x)->mmio) + PSC_AC97PCR)
> +#define AC97_RST(x) ((unsigned long)((x)->mmio) + PSC_AC97RST)
> +#define AC97_STAT(x) ((unsigned long)((x)->mmio) + PSC_AC97STAT)
> +
> +#endif
> diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
> new file mode 100644
> index 0000000..fce81da
> --- /dev/null
> +++ b/sound/soc/au1x/sample-ac97.c
> @@ -0,0 +1,144 @@
> +/*
> + * 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_endpoints(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 = NULL;
> +
> +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>");
> --
> 1.5.6.1
>
> _______________________________________________
> Alsa-devel mailing list
> Alsa-devel@alsa-project.org
> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-04 15:13 ` Takashi Iwai
@ 2008-07-04 16:24 ` Manuel Lauss
2008-07-04 17:36 ` Liam Girdwood
0 siblings, 1 reply; 20+ messages in thread
From: Manuel Lauss @ 2008-07-04 16:24 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown
Hello Takashi,
Takashi Iwai wrote:
> At Thu, 3 Jul 2008 12:09:25 +0200,
> Manuel Lauss wrote:
>> On Thu, Jul 03, 2008 at 10:55:47AM +0100, Liam Girdwood wrote:
>>> On Thu, 2008-07-03 at 11:53 +0200, Manuel Lauss wrote:
>>>> On Thu, Jul 03, 2008 at 10:29:19AM +0100, Liam Girdwood wrote:
>>>>> Btw, any plans to fix or remove some of the 'Fixmes' in your other ASoC
>>>>> code. We could upstream the Au1x000 stuff after a little cleanup and
>>>>> resolution of the fixmes.
>>>> I've updated the code considerably since the last year. It's still ASoC v1,
>>>> and lots of the FIXMEs in the ac97 part won't disappear until it's moved to
>>>> ASoC v2 (I hope, didn't investigate).
>>>>
>>>> If you're interested I can send you a patch (against current linus' HEAD)
>>>> with the newest, most awesome au1xxx psc asoc code ;-)
>>>>
>>> Yes please.
>> Here you go, apply on top of linus' head + the ac97 PM patch.
>>
>> ---
>>
>> From: Manuel Lauss <mano@roarinelk.homelinux.net>
>>
>> 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>
>
> Shall I apply this one? At a quick glance, the patch looks OK.
I certainly have no objections, but I intended to wait for ASoCv2 to hit
mainline, convert the drivers to it, then (re)submit. There's still a lot
to be done IMO (it has never been tested with preemption, i2s bitclk master
mode is untested, ...)
Best regards,
Manuel Lauss
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-04 16:24 ` Manuel Lauss
@ 2008-07-04 17:36 ` Liam Girdwood
2008-07-07 16:05 ` Takashi Iwai
0 siblings, 1 reply; 20+ messages in thread
From: Liam Girdwood @ 2008-07-04 17:36 UTC (permalink / raw)
To: Manuel Lauss; +Cc: Takashi Iwai, alsa-devel, Mark Brown
On Fri, 2008-07-04 at 18:24 +0200, Manuel Lauss wrote:
> Hello Takashi,
>
> Takashi Iwai wrote:
> >
> > Shall I apply this one? At a quick glance, the patch looks OK.
>
> I certainly have no objections, but I intended to wait for ASoCv2 to hit
> mainline, convert the drivers to it, then (re)submit. There's still a lot
> to be done IMO (it has never been tested with preemption, i2s bitclk master
> mode is untested, ...)
>
Please apply.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Btw, I've now started sending patches for v2 (although I seem to have
sent a little extra in the last patch ) ;)
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-04 17:36 ` Liam Girdwood
@ 2008-07-07 16:05 ` Takashi Iwai
2008-07-07 16:09 ` Liam Girdwood
0 siblings, 1 reply; 20+ messages in thread
From: Takashi Iwai @ 2008-07-07 16:05 UTC (permalink / raw)
To: Liam Girdwood; +Cc: Manuel Lauss, Mark Brown, alsa-devel
At Fri, 04 Jul 2008 18:36:56 +0100,
Liam Girdwood wrote:
>
> On Fri, 2008-07-04 at 18:24 +0200, Manuel Lauss wrote:
> > Hello Takashi,
> >
> > Takashi Iwai wrote:
> > >
> > > Shall I apply this one? At a quick glance, the patch looks OK.
> >
> > I certainly have no objections, but I intended to wait for ASoCv2 to hit
> > mainline, convert the drivers to it, then (re)submit. There's still a lot
> > to be done IMO (it has never been tested with preemption, i2s bitclk master
> > mode is untested, ...)
> >
>
> Please apply.
>
> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
>
> Btw, I've now started sending patches for v2 (although I seem to have
> sent a little extra in the last patch ) ;)
Manuel's patch can't be applied cleanly to the latest ALSA tree.
And I applied Liam's patches right now, so there must be more conflicts
(e.g. snd_soc_cpu_dai replacement).
Manuel, could you regenerate the patch against the following tree?
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git
thanks,
Takashi
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-07 16:05 ` Takashi Iwai
@ 2008-07-07 16:09 ` Liam Girdwood
2008-07-07 16:11 ` Takashi Iwai
` (2 more replies)
0 siblings, 3 replies; 20+ messages in thread
From: Liam Girdwood @ 2008-07-07 16:09 UTC (permalink / raw)
To: Takashi Iwai; +Cc: Manuel Lauss, Mark Brown, alsa-devel
On Mon, 2008-07-07 at 18:05 +0200, Takashi Iwai wrote:
> At Fri, 04 Jul 2008 18:36:56 +0100,
> Liam Girdwood wrote:
> >
> > On Fri, 2008-07-04 at 18:24 +0200, Manuel Lauss wrote:
> > > Hello Takashi,
> > >
> > > Takashi Iwai wrote:
> > > >
> > > > Shall I apply this one? At a quick glance, the patch looks OK.
> > >
> > > I certainly have no objections, but I intended to wait for ASoCv2 to hit
> > > mainline, convert the drivers to it, then (re)submit. There's still a lot
> > > to be done IMO (it has never been tested with preemption, i2s bitclk master
> > > mode is untested, ...)
> > >
> >
> > Please apply.
> >
> > Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
> >
> > Btw, I've now started sending patches for v2 (although I seem to have
> > sent a little extra in the last patch ) ;)
>
> Manuel's patch can't be applied cleanly to the latest ALSA tree.
> And I applied Liam's patches right now, so there must be more conflicts
> (e.g. snd_soc_cpu_dai replacement).
>
My bad ;)
> Manuel, could you regenerate the patch against the following tree?
> git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git
Manuel, I'll fix the Au1xx and resubmit with the DAI changes. I have
another pending change wrt DAI and will queue the Aux1xx after this.
Liam
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-07 16:09 ` Liam Girdwood
@ 2008-07-07 16:11 ` Takashi Iwai
2008-07-07 17:15 ` Manuel Lauss
2008-07-07 17:38 ` Manuel Lauss
2 siblings, 0 replies; 20+ messages in thread
From: Takashi Iwai @ 2008-07-07 16:11 UTC (permalink / raw)
To: Liam Girdwood; +Cc: Manuel Lauss, Mark Brown, alsa-devel
At Mon, 07 Jul 2008 17:09:56 +0100,
Liam Girdwood wrote:
>
> On Mon, 2008-07-07 at 18:05 +0200, Takashi Iwai wrote:
> > At Fri, 04 Jul 2008 18:36:56 +0100,
> > Liam Girdwood wrote:
> > >
> > > On Fri, 2008-07-04 at 18:24 +0200, Manuel Lauss wrote:
> > > > Hello Takashi,
> > > >
> > > > Takashi Iwai wrote:
> > > > >
> > > > > Shall I apply this one? At a quick glance, the patch looks OK.
> > > >
> > > > I certainly have no objections, but I intended to wait for ASoCv2 to hit
> > > > mainline, convert the drivers to it, then (re)submit. There's still a lot
> > > > to be done IMO (it has never been tested with preemption, i2s bitclk master
> > > > mode is untested, ...)
> > > >
> > >
> > > Please apply.
> > >
> > > Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
> > >
> > > Btw, I've now started sending patches for v2 (although I seem to have
> > > sent a little extra in the last patch ) ;)
> >
> > Manuel's patch can't be applied cleanly to the latest ALSA tree.
> > And I applied Liam's patches right now, so there must be more conflicts
> > (e.g. snd_soc_cpu_dai replacement).
> >
>
> My bad ;)
>
> > Manuel, could you regenerate the patch against the following tree?
> > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git
>
>
> Manuel, I'll fix the Au1xx and resubmit with the DAI changes. I have
> another pending change wrt DAI and will queue the Aux1xx after this.
That's also fine, of course :)
thanks,
Takashi
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-07 16:09 ` Liam Girdwood
2008-07-07 16:11 ` Takashi Iwai
@ 2008-07-07 17:15 ` Manuel Lauss
2008-07-07 17:38 ` Manuel Lauss
2 siblings, 0 replies; 20+ messages in thread
From: Manuel Lauss @ 2008-07-07 17:15 UTC (permalink / raw)
To: Liam Girdwood; +Cc: Takashi Iwai, alsa-devel, Mark Brown
Hi Liam, Takashi,
On Mon, Jul 07, 2008 at 05:09:56PM +0100, Liam Girdwood wrote:
> On Mon, 2008-07-07 at 18:05 +0200, Takashi Iwai wrote:
> > At Fri, 04 Jul 2008 18:36:56 +0100,
> > Liam Girdwood wrote:
> > >
> > > On Fri, 2008-07-04 at 18:24 +0200, Manuel Lauss wrote:
> > > > Hello Takashi,
> > > >
> > > > Takashi Iwai wrote:
> > > > >
> > > > > Shall I apply this one? At a quick glance, the patch looks OK.
> > > >
> > > > I certainly have no objections, but I intended to wait for ASoCv2 to hit
> > > > mainline, convert the drivers to it, then (re)submit. There's still a lot
> > > > to be done IMO (it has never been tested with preemption, i2s bitclk master
> > > > mode is untested, ...)
> > > >
> > >
> > > Please apply.
> > >
> > > Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
> > >
> > > Btw, I've now started sending patches for v2 (although I seem to have
> > > sent a little extra in the last patch ) ;)
> >
> > Manuel's patch can't be applied cleanly to the latest ALSA tree.
> > And I applied Liam's patches right now, so there must be more conflicts
> > (e.g. snd_soc_cpu_dai replacement).
> >
>
> My bad ;)
>
> > Manuel, could you regenerate the patch against the following tree?
> > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git
>
>
> Manuel, I'll fix the Au1xx and resubmit with the DAI changes. I have
> another pending change wrt DAI and will queue the Aux1xx after this.
Hold on a sec, I've prepared a new patch which fixes a few smaller issues
(most notably it no longer breaks the OSS Au1550_ac97 driver [not that it
matters as it is broken anyway], it is now checkpatch clean and fixes an
AC97 issue wrt. simultaneous playback and capture).
Find it below,
Thanks!
Manuel Lauss
---
From: Manuel Lauss <mano@roarinelk.homelinux.net>
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>
---
include/asm-mips/mach-au1x00/au1xxx_psc.h | 18 +-
sound/soc/Kconfig | 1 +
sound/soc/Makefile | 2 +-
sound/soc/au1x/Kconfig | 36 +++
sound/soc/au1x/Makefile | 13 +
sound/soc/au1x/dbdma2.c | 421 +++++++++++++++++++++++++++++
sound/soc/au1x/psc-ac97.c | 385 ++++++++++++++++++++++++++
sound/soc/au1x/psc-i2s.c | 413 ++++++++++++++++++++++++++++
sound/soc/au1x/psc.h | 54 ++++
sound/soc/au1x/sample-ac97.c | 144 ++++++++++
10 files changed, 1475 insertions(+), 12 deletions(-)
create mode 100644 sound/soc/au1x/Kconfig
create mode 100644 sound/soc/au1x/Makefile
create mode 100644 sound/soc/au1x/dbdma2.c
create mode 100644 sound/soc/au1x/psc-ac97.c
create mode 100644 sound/soc/au1x/psc-i2s.c
create mode 100644 sound/soc/au1x/psc.h
create mode 100644 sound/soc/au1x/sample-ac97.c
diff --git a/include/asm-mips/mach-au1x00/au1xxx_psc.h b/include/asm-mips/mach-au1x00/au1xxx_psc.h
index dae4eca..449ab9e 100644
--- a/include/asm-mips/mach-au1x00/au1xxx_psc.h
+++ b/include/asm-mips/mach-au1x00/au1xxx_psc.h
@@ -192,17 +192,13 @@
#define PSC_AC97RST_SNC (1 << 0)
/* PSC in I2S Mode. */
-typedef struct psc_i2s {
- u32 psc_sel;
- u32 psc_ctrl;
- u32 psc_i2scfg;
- u32 psc_i2smsk;
- u32 psc_i2spcr;
- u32 psc_i2sstat;
- u32 psc_i2sevent;
- u32 psc_i2stxrx;
- 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)
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 18f28ac..caacf95 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -25,6 +25,7 @@ config SND_SOC
# All the supported Soc's
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"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 782db21..5dafddd 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,4 @@
snd-soc-core-objs := soc-core.o soc-dapm.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
-obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
+obj-$(CONFIG_SND_SOC) += codecs/ at91/ au1x/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
new file mode 100644
index 0000000..8ef9015
--- /dev/null
+++ b/sound/soc/au1x/Kconfig
@@ -0,0 +1,36 @@
+menu "SoC Audio for the Alchemy/AMD/RMI Au1xxx"
+ depends on SOC_AU1200 || SOC_AU1550
+
+##
+## Au1200/Au1550 PSC + DBDMA
+##
+config SND_SOC_AU1XPSC
+ tristate "SoC Audio for Au1200/Au1250/Au1550"
+ depends on SND_SOC && (SOC_AU1200 || SOC_AU1550)
+ help
+ This option enables support for the Programmable Serial
+ Controllers in AC97 and I2S mode, and the Descriptor-Based DMA
+ Controller (DBDMA) as found on the Au1200/Au1250/Au1550 SoC.
+
+config SND_SOC_AU1XPSC_I2S
+ tristate
+
+config SND_SOC_AU1XPSC_AC97
+ tristate
+ select AC97_BUS
+ select SND_AC97_CODEC
+ select SND_SOC_AC97_BUS
+
+
+##
+## Boards
+##
+config SND_SOC_SAMPLE_PSC_AC97
+ tristate "Sample Au12x0/Au1550 PSC AC97 sound machine"
+ select SND_SOC_AU1XPSC_AC97
+ select SND_SOC_AC97_CODEC
+ help
+ This is a sample AC97 sound machine for use in Au12x0/Au1550
+ based systems which have audio on PSC1 (e.g. Db1200 demoboard).
+
+endmenu
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
new file mode 100644
index 0000000..6c6950b
--- /dev/null
+++ b/sound/soc/au1x/Makefile
@@ -0,0 +1,13 @@
+# Au1200/Au1550 PSC audio
+snd-soc-au1xpsc-dbdma-objs := dbdma2.o
+snd-soc-au1xpsc-i2s-objs := psc-i2s.o
+snd-soc-au1xpsc-ac97-objs := psc-ac97.o
+
+obj-$(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
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
new file mode 100644
index 0000000..30d9e01
--- /dev/null
+++ b/sound/soc/au1x/dbdma2.c
@@ -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_codec_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>");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
new file mode 100644
index 0000000..dc7c437
--- /dev/null
+++ b/sound/soc/au1x/psc-ac97.c
@@ -0,0 +1,385 @@
+/*
+ * 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)
+{
+ 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)
+{
+ /* 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_cpu_dai *cpu_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_cpu_dai *cpu_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_cpu_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>");
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
new file mode 100644
index 0000000..a2f4b71
--- /dev/null
+++ b/sound/soc/au1x/psc-i2s.c
@@ -0,0 +1,413 @@
+/*
+ * 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_cpu_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 - 1);
+ /* 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 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)
+{
+ 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_cpu_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_cpu_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_cpu_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>");
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
new file mode 100644
index 0000000..2232972
--- /dev/null
+++ b/sound/soc/au1x/psc.h
@@ -0,0 +1,54 @@
+/*
+ * 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_cpu_dai au1xpsc_ac97_dai;
+extern struct snd_soc_cpu_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;
+ int irq;
+ struct resource *ioarea;
+
+ unsigned long cfg;
+ unsigned long rate;
+
+ unsigned long pm[2];
+};
+
+#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
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
new file mode 100644
index 0000000..31d7fe8
--- /dev/null
+++ b/sound/soc/au1x/sample-ac97.c
@@ -0,0 +1,144 @@
+/*
+ * 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_endpoints(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>");
--
1.5.6.1
^ permalink raw reply related [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-07 16:09 ` Liam Girdwood
2008-07-07 16:11 ` Takashi Iwai
2008-07-07 17:15 ` Manuel Lauss
@ 2008-07-07 17:38 ` Manuel Lauss
2008-07-08 15:13 ` Liam Girdwood
2 siblings, 1 reply; 20+ messages in thread
From: Manuel Lauss @ 2008-07-07 17:38 UTC (permalink / raw)
To: Liam Girdwood; +Cc: Takashi Iwai, alsa-devel, Mark Brown
Hi again,
[...]
the previous patch contained a bug in the I2S part, here's a new one.
Thanks!
Manuel Lauss
---
From: Manuel Lauss <mano@roarinelk.homelinux.net>
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>
---
include/asm-mips/mach-au1x00/au1xxx_psc.h | 18 +-
sound/soc/Kconfig | 1 +
sound/soc/Makefile | 2 +-
sound/soc/au1x/Kconfig | 36 +++
sound/soc/au1x/Makefile | 13 +
sound/soc/au1x/dbdma2.c | 421 +++++++++++++++++++++++++++++
sound/soc/au1x/psc-ac97.c | 385 ++++++++++++++++++++++++++
sound/soc/au1x/psc-i2s.c | 413 ++++++++++++++++++++++++++++
sound/soc/au1x/psc.h | 53 ++++
sound/soc/au1x/sample-ac97.c | 144 ++++++++++
10 files changed, 1474 insertions(+), 12 deletions(-)
create mode 100644 sound/soc/au1x/Kconfig
create mode 100644 sound/soc/au1x/Makefile
create mode 100644 sound/soc/au1x/dbdma2.c
create mode 100644 sound/soc/au1x/psc-ac97.c
create mode 100644 sound/soc/au1x/psc-i2s.c
create mode 100644 sound/soc/au1x/psc.h
create mode 100644 sound/soc/au1x/sample-ac97.c
diff --git a/include/asm-mips/mach-au1x00/au1xxx_psc.h b/include/asm-mips/mach-au1x00/au1xxx_psc.h
index dae4eca..449ab9e 100644
--- a/include/asm-mips/mach-au1x00/au1xxx_psc.h
+++ b/include/asm-mips/mach-au1x00/au1xxx_psc.h
@@ -192,17 +192,13 @@
#define PSC_AC97RST_SNC (1 << 0)
/* PSC in I2S Mode. */
-typedef struct psc_i2s {
- u32 psc_sel;
- u32 psc_ctrl;
- u32 psc_i2scfg;
- u32 psc_i2smsk;
- u32 psc_i2spcr;
- u32 psc_i2sstat;
- u32 psc_i2sevent;
- u32 psc_i2stxrx;
- 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)
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 18f28ac..caacf95 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -25,6 +25,7 @@ config SND_SOC
# All the supported Soc's
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"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 782db21..5dafddd 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,4 @@
snd-soc-core-objs := soc-core.o soc-dapm.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
-obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
+obj-$(CONFIG_SND_SOC) += codecs/ at91/ au1x/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
new file mode 100644
index 0000000..8ef9015
--- /dev/null
+++ b/sound/soc/au1x/Kconfig
@@ -0,0 +1,36 @@
+menu "SoC Audio for the Alchemy/AMD/RMI Au1xxx"
+ depends on SOC_AU1200 || SOC_AU1550
+
+##
+## Au1200/Au1550 PSC + DBDMA
+##
+config SND_SOC_AU1XPSC
+ tristate "SoC Audio for Au1200/Au1250/Au1550"
+ depends on SND_SOC && (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"
+ 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).
+
+endmenu
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
new file mode 100644
index 0000000..6c6950b
--- /dev/null
+++ b/sound/soc/au1x/Makefile
@@ -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
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
new file mode 100644
index 0000000..30d9e01
--- /dev/null
+++ b/sound/soc/au1x/dbdma2.c
@@ -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_codec_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>");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
new file mode 100644
index 0000000..dc7c437
--- /dev/null
+++ b/sound/soc/au1x/psc-ac97.c
@@ -0,0 +1,385 @@
+/*
+ * 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)
+{
+ 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)
+{
+ /* 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_cpu_dai *cpu_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_cpu_dai *cpu_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_cpu_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>");
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
new file mode 100644
index 0000000..0af9683
--- /dev/null
+++ b/sound/soc/au1x/psc-i2s.c
@@ -0,0 +1,413 @@
+/*
+ * 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_cpu_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 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)
+{
+ 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_cpu_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_cpu_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_cpu_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>");
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
new file mode 100644
index 0000000..85a0540
--- /dev/null
+++ b/sound/soc/au1x/psc.h
@@ -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_cpu_dai au1xpsc_ac97_dai;
+extern struct snd_soc_cpu_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
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
new file mode 100644
index 0000000..31d7fe8
--- /dev/null
+++ b/sound/soc/au1x/sample-ac97.c
@@ -0,0 +1,144 @@
+/*
+ * 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_endpoints(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>");
--
1.5.6.1
^ permalink raw reply related [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-07 17:38 ` Manuel Lauss
@ 2008-07-08 15:13 ` Liam Girdwood
2008-07-08 16:36 ` Manuel Lauss
0 siblings, 1 reply; 20+ messages in thread
From: Liam Girdwood @ 2008-07-08 15:13 UTC (permalink / raw)
To: Manuel Lauss; +Cc: Takashi Iwai, alsa-devel, Mark Brown
On Mon, 2008-07-07 at 19:38 +0200, Manuel Lauss wrote:
> Hi again,
>
> [...]
>
> the previous patch contained a bug in the I2S part, here's a new one.
>
I've now updated with the new API changes.
Manuel, there are changes to au1xxx_psc.h (I assume dependencies).
Should this not go to the MIPS maintainer first ? Would you also give
this a quick check too as I'm not able to build for this target.
Thanks
From: Manuel Lauss <mano@roarinelk.homelinux.net>
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>
---
include/asm-mips/mach-au1x00/au1xxx_psc.h | 18 +-
sound/soc/Kconfig | 1 +
sound/soc/Makefile | 2 +-
sound/soc/au1x/Kconfig | 36 +++
sound/soc/au1x/Makefile | 13 +
sound/soc/au1x/dbdma2.c | 421 +++++++++++++++++++++++++++++
sound/soc/au1x/psc-ac97.c | 385 ++++++++++++++++++++++++++
sound/soc/au1x/psc-i2s.c | 413 ++++++++++++++++++++++++++++
sound/soc/au1x/psc.h | 53 ++++
sound/soc/au1x/sample-ac97.c | 144 ++++++++++
10 files changed, 1474 insertions(+), 12 deletions(-)
create mode 100644 sound/soc/au1x/Kconfig
create mode 100644 sound/soc/au1x/Makefile
create mode 100644 sound/soc/au1x/dbdma2.c
create mode 100644 sound/soc/au1x/psc-ac97.c
create mode 100644 sound/soc/au1x/psc-i2s.c
create mode 100644 sound/soc/au1x/psc.h
create mode 100644 sound/soc/au1x/sample-ac97.c
diff --git a/include/asm-mips/mach-au1x00/au1xxx_psc.h b/include/asm-mips/mach-au1x00/au1xxx_psc.h
index dae4eca..449ab9e 100644
--- a/include/asm-mips/mach-au1x00/au1xxx_psc.h
+++ b/include/asm-mips/mach-au1x00/au1xxx_psc.h
@@ -192,17 +192,13 @@
#define PSC_AC97RST_SNC (1 << 0)
/* PSC in I2S Mode. */
-typedef struct psc_i2s {
- u32 psc_sel;
- u32 psc_ctrl;
- u32 psc_i2scfg;
- u32 psc_i2smsk;
- u32 psc_i2spcr;
- u32 psc_i2sstat;
- u32 psc_i2sevent;
- u32 psc_i2stxrx;
- 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)
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index b939e22..f743530 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -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"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 3645f95..933a66d 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -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/
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
new file mode 100644
index 0000000..8ef9015
--- /dev/null
+++ b/sound/soc/au1x/Kconfig
@@ -0,0 +1,36 @@
+menu "SoC Audio for the Alchemy/AMD/RMI Au1xxx"
+ depends on SOC_AU1200 || SOC_AU1550
+
+##
+## Au1200/Au1550 PSC + DBDMA
+##
+config SND_SOC_AU1XPSC
+ tristate "SoC Audio for Au1200/Au1250/Au1550"
+ depends on SND_SOC && (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"
+ 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).
+
+endmenu
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
new file mode 100644
index 0000000..6c6950b
--- /dev/null
+++ b/sound/soc/au1x/Makefile
@@ -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
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
new file mode 100644
index 0000000..30d9e01
--- /dev/null
+++ b/sound/soc/au1x/dbdma2.c
@@ -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_codec_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>");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
new file mode 100644
index 0000000..3dac05f
--- /dev/null
+++ b/sound/soc/au1x/psc-ac97.c
@@ -0,0 +1,385 @@
+/*
+ * 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)
+{
+ 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)
+{
+ /* 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 *cpu_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 *cpu_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>");
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
new file mode 100644
index 0000000..44d074e
--- /dev/null
+++ b/sound/soc/au1x/psc-i2s.c
@@ -0,0 +1,413 @@
+/*
+ * 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 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)
+{
+ 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>");
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
new file mode 100644
index 0000000..8fdb1a0
--- /dev/null
+++ b/sound/soc/au1x/psc.h
@@ -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
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
new file mode 100644
index 0000000..f75ae7f
--- /dev/null
+++ b/sound/soc/au1x/sample-ac97.c
@@ -0,0 +1,144 @@
+/*
+ * 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>");
--
1.5.4.2
^ permalink raw reply related [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-08 15:13 ` Liam Girdwood
@ 2008-07-08 16:36 ` Manuel Lauss
2008-07-09 5:34 ` Manuel Lauss
2008-07-09 17:48 ` Takashi Iwai
0 siblings, 2 replies; 20+ messages in thread
From: Manuel Lauss @ 2008-07-08 16:36 UTC (permalink / raw)
To: Liam Girdwood; +Cc: Takashi Iwai, alsa-devel, Mark Brown
Hi Liam,
On Tue, Jul 08, 2008 at 04:13:27PM +0100, Liam Girdwood wrote:
> On Mon, 2008-07-07 at 19:38 +0200, Manuel Lauss wrote:
> > Hi again,
> >
> > [...]
> >
> > the previous patch contained a bug in the I2S part, here's a new one.
> >
>
> I've now updated with the new API changes.
>
> Manuel, there are changes to au1xxx_psc.h (I assume dependencies).
I've changed this hunk to only add the constants I need. There are no
current in-tree users of this part anyway.
> Should this not go to the MIPS maintainer first ? Would you also give
> this a quick check too as I'm not able to build for this target.
It needed a few more touches to compile. I can't test until tomorrow
morning, but I see no reason why it should not.
---
Subject: [PATCH] ASoC: Au12x0/Au1550 PSC Audio support.
Audio for Au12x0/Au1550 PSCs in AC97 and I2S mode, for ASoC v1 framework.
- DBDMA, AC97 and I2S drivers
- sample AC97 machine code (e.g. Db1200)
Signed-off-by: Manuel Lauss <mano@roarinelk.homelinux.net>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
---
include/asm-mips/mach-au1x00/au1xxx_psc.h | 8 +
sound/soc/Kconfig | 1 +
sound/soc/Makefile | 2 +-
sound/soc/au1x/Kconfig | 36 +++
sound/soc/au1x/Makefile | 13 +
sound/soc/au1x/dbdma2.c | 421 +++++++++++++++++++++++++++++
sound/soc/au1x/psc-ac97.c | 387 ++++++++++++++++++++++++++
sound/soc/au1x/psc-i2s.c | 414 ++++++++++++++++++++++++++++
sound/soc/au1x/psc.h | 53 ++++
sound/soc/au1x/sample-ac97.c | 144 ++++++++++
10 files changed, 1478 insertions(+), 1 deletions(-)
create mode 100644 sound/soc/au1x/Kconfig
create mode 100644 sound/soc/au1x/Makefile
create mode 100644 sound/soc/au1x/dbdma2.c
create mode 100644 sound/soc/au1x/psc-ac97.c
create mode 100644 sound/soc/au1x/psc-i2s.c
create mode 100644 sound/soc/au1x/psc.h
create mode 100644 sound/soc/au1x/sample-ac97.c
diff --git a/include/asm-mips/mach-au1x00/au1xxx_psc.h b/include/asm-mips/mach-au1x00/au1xxx_psc.h
index dae4eca..892b7f1 100644
--- a/include/asm-mips/mach-au1x00/au1xxx_psc.h
+++ b/include/asm-mips/mach-au1x00/au1xxx_psc.h
@@ -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)
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index b939e22..f743530 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -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"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 3645f95..933a66d 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -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/
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
new file mode 100644
index 0000000..8ef9015
--- /dev/null
+++ b/sound/soc/au1x/Kconfig
@@ -0,0 +1,36 @@
+menu "SoC Audio for the Alchemy/AMD/RMI Au1xxx"
+ depends on SOC_AU1200 || SOC_AU1550
+
+##
+## Au1200/Au1550 PSC + DBDMA
+##
+config SND_SOC_AU1XPSC
+ tristate "SoC Audio for Au1200/Au1250/Au1550"
+ depends on SND_SOC && (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"
+ 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).
+
+endmenu
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
new file mode 100644
index 0000000..6c6950b
--- /dev/null
+++ b/sound/soc/au1x/Makefile
@@ -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
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
new file mode 100644
index 0000000..1466d93
--- /dev/null
+++ b/sound/soc/au1x/dbdma2.c
@@ -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>");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
new file mode 100644
index 0000000..57facba
--- /dev/null
+++ b/sound/soc/au1x/psc-ac97.c
@@ -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>");
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
new file mode 100644
index 0000000..ba4b5c1
--- /dev/null
+++ b/sound/soc/au1x/psc-i2s.c
@@ -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>");
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
new file mode 100644
index 0000000..8fdb1a0
--- /dev/null
+++ b/sound/soc/au1x/psc.h
@@ -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
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
new file mode 100644
index 0000000..f75ae7f
--- /dev/null
+++ b/sound/soc/au1x/sample-ac97.c
@@ -0,0 +1,144 @@
+/*
+ * 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>");
--
1.5.6.1
^ permalink raw reply related [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-08 16:36 ` Manuel Lauss
@ 2008-07-09 5:34 ` Manuel Lauss
2008-07-09 17:48 ` Takashi Iwai
1 sibling, 0 replies; 20+ messages in thread
From: Manuel Lauss @ 2008-07-09 5:34 UTC (permalink / raw)
To: alsa-devel
I wrote:
> It needed a few more touches to compile. I can't test until tomorrow
> morning, but I see no reason why it should not.
Well, it works on real hardware too. So please apply.
Thanks!
Manuel Lauss
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-09 17:48 ` Takashi Iwai
@ 2008-07-09 14:27 ` Manuel Lauss
2008-07-09 14:55 ` Liam Girdwood
2008-07-09 19:27 ` Takashi Iwai
0 siblings, 2 replies; 20+ messages in thread
From: Manuel Lauss @ 2008-07-09 14:27 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel, Mark Brown
Hi Takashi,
On Wed, Jul 09, 2008 at 07:48:00PM +0200, Takashi Iwai wrote:
> Hi,
>
> there are still some minor Kconfig issues:
>
> > --- /dev/null
> > +++ b/sound/soc/au1x/Kconfig
> > @@ -0,0 +1,36 @@
> > +menu "SoC Audio for the Alchemy/AMD/RMI Au1xxx"
> > + depends on SOC_AU1200 || SOC_AU1550
>
> Avoid menu. If really needed, use rather menuconfig.
Done. (Someone should do that for all others. It's annoying
that the OMAP and Freescale SOC options are visible on Au1xxx).
> > +
> > +##
> > +## Au1200/Au1550 PSC + DBDMA
> > +##
> > +config SND_SOC_AU1XPSC
> > + tristate "SoC Audio for Au1200/Au1250/Au1550"
> > + depends on SND_SOC && (SOC_AU1200 || SOC_AU1550)
>
> As mentioned earlier, dependency on SND_SOC is superfluous here.
Also done.
Thanks!
Manuel Lauss
---
Subject: [PATCH] ASoC: Au12x0/Au1550 PSC Audio support.
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>
---
include/asm-mips/mach-au1x00/au1xxx_psc.h | 8 +
sound/soc/Kconfig | 1 +
sound/soc/Makefile | 2 +-
sound/soc/au1x/Kconfig | 32 +++
sound/soc/au1x/Makefile | 13 +
sound/soc/au1x/dbdma2.c | 421 +++++++++++++++++++++++++++++
sound/soc/au1x/psc-ac97.c | 387 ++++++++++++++++++++++++++
sound/soc/au1x/psc-i2s.c | 414 ++++++++++++++++++++++++++++
sound/soc/au1x/psc.h | 53 ++++
sound/soc/au1x/sample-ac97.c | 144 ++++++++++
10 files changed, 1474 insertions(+), 1 deletions(-)
create mode 100644 sound/soc/au1x/Kconfig
create mode 100644 sound/soc/au1x/Makefile
create mode 100644 sound/soc/au1x/dbdma2.c
create mode 100644 sound/soc/au1x/psc-ac97.c
create mode 100644 sound/soc/au1x/psc-i2s.c
create mode 100644 sound/soc/au1x/psc.h
create mode 100644 sound/soc/au1x/sample-ac97.c
diff --git a/include/asm-mips/mach-au1x00/au1xxx_psc.h b/include/asm-mips/mach-au1x00/au1xxx_psc.h
index dae4eca..892b7f1 100644
--- a/include/asm-mips/mach-au1x00/au1xxx_psc.h
+++ b/include/asm-mips/mach-au1x00/au1xxx_psc.h
@@ -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)
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index b939e22..f743530 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -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"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 3645f95..933a66d 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -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/
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
new file mode 100644
index 0000000..410a893
--- /dev/null
+++ b/sound/soc/au1x/Kconfig
@@ -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).
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
new file mode 100644
index 0000000..6c6950b
--- /dev/null
+++ b/sound/soc/au1x/Makefile
@@ -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
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
new file mode 100644
index 0000000..1466d93
--- /dev/null
+++ b/sound/soc/au1x/dbdma2.c
@@ -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>");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
new file mode 100644
index 0000000..57facba
--- /dev/null
+++ b/sound/soc/au1x/psc-ac97.c
@@ -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>");
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
new file mode 100644
index 0000000..ba4b5c1
--- /dev/null
+++ b/sound/soc/au1x/psc-i2s.c
@@ -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>");
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
new file mode 100644
index 0000000..8fdb1a0
--- /dev/null
+++ b/sound/soc/au1x/psc.h
@@ -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
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
new file mode 100644
index 0000000..f75ae7f
--- /dev/null
+++ b/sound/soc/au1x/sample-ac97.c
@@ -0,0 +1,144 @@
+/*
+ * 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>");
--
1.5.6.1
^ permalink raw reply related [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-09 14:27 ` Manuel Lauss
@ 2008-07-09 14:55 ` Liam Girdwood
2008-07-09 19:27 ` Takashi Iwai
1 sibling, 0 replies; 20+ messages in thread
From: Liam Girdwood @ 2008-07-09 14:55 UTC (permalink / raw)
To: Manuel Lauss; +Cc: Takashi Iwai, alsa-devel, Mark Brown
On Wed, 2008-07-09 at 16:27 +0200, Manuel Lauss wrote:
> Hi Takashi,
>
> On Wed, Jul 09, 2008 at 07:48:00PM +0200, Takashi Iwai wrote:
> > Hi,
> >
> > there are still some minor Kconfig issues:
> >
> > > --- /dev/null
> > > +++ b/sound/soc/au1x/Kconfig
> > > @@ -0,0 +1,36 @@
> > > +menu "SoC Audio for the Alchemy/AMD/RMI Au1xxx"
> > > + depends on SOC_AU1200 || SOC_AU1550
> >
> > Avoid menu. If really needed, use rather menuconfig.
>
> Done. (Someone should do that for all others. It's annoying
> that the OMAP and Freescale SOC options are visible on Au1xxx).
I'll fix this.
Liam
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-08 16:36 ` Manuel Lauss
2008-07-09 5:34 ` Manuel Lauss
@ 2008-07-09 17:48 ` Takashi Iwai
2008-07-09 14:27 ` Manuel Lauss
1 sibling, 1 reply; 20+ messages in thread
From: Takashi Iwai @ 2008-07-09 17:48 UTC (permalink / raw)
To: Manuel Lauss; +Cc: alsa-devel, Mark Brown
Hi,
there are still some minor Kconfig issues:
> --- /dev/null
> +++ b/sound/soc/au1x/Kconfig
> @@ -0,0 +1,36 @@
> +menu "SoC Audio for the Alchemy/AMD/RMI Au1xxx"
> + depends on SOC_AU1200 || SOC_AU1550
Avoid menu. If really needed, use rather menuconfig.
> +
> +##
> +## Au1200/Au1550 PSC + DBDMA
> +##
> +config SND_SOC_AU1XPSC
> + tristate "SoC Audio for Au1200/Au1250/Au1550"
> + depends on SND_SOC && (SOC_AU1200 || SOC_AU1550)
As mentioned earlier, dependency on SND_SOC is superfluous here.
Could you fix and repost?
thanks,
Takashi
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: ASoC: Au12x0/Au1550 PSC Audio support.
2008-07-09 14:27 ` Manuel Lauss
2008-07-09 14:55 ` Liam Girdwood
@ 2008-07-09 19:27 ` Takashi Iwai
1 sibling, 0 replies; 20+ messages in thread
From: Takashi Iwai @ 2008-07-09 19:27 UTC (permalink / raw)
To: Manuel Lauss; +Cc: alsa-devel, Mark Brown
At Wed, 9 Jul 2008 16:27:56 +0200,
Manuel Lauss wrote:
>
> Hi Takashi,
>
> On Wed, Jul 09, 2008 at 07:48:00PM +0200, Takashi Iwai wrote:
> > Hi,
> >
> > there are still some minor Kconfig issues:
> >
> > > --- /dev/null
> > > +++ b/sound/soc/au1x/Kconfig
> > > @@ -0,0 +1,36 @@
> > > +menu "SoC Audio for the Alchemy/AMD/RMI Au1xxx"
> > > + depends on SOC_AU1200 || SOC_AU1550
> >
> > Avoid menu. If really needed, use rather menuconfig.
>
> Done. (Someone should do that for all others. It's annoying
> that the OMAP and Freescale SOC options are visible on Au1xxx).
>
>
> > > +
> > > +##
> > > +## Au1200/Au1550 PSC + DBDMA
> > > +##
> > > +config SND_SOC_AU1XPSC
> > > + tristate "SoC Audio for Au1200/Au1250/Au1550"
> > > + depends on SND_SOC && (SOC_AU1200 || SOC_AU1550)
> >
> > As mentioned earlier, dependency on SND_SOC is superfluous here.
>
> Also done.
Thanks, applied now.
Takashi
^ permalink raw reply [flat|nested] 20+ messages in thread
end of thread, other threads:[~2008-07-09 15:27 UTC | newest]
Thread overview: 20+ messages (download: mbox.gz follow: Atom feed
-- links below jump to the message on this page --
2008-07-03 7:33 ASoC: AC97 codec PM Manuel Lauss
2008-07-03 9:29 ` Liam Girdwood
2008-07-03 9:53 ` Manuel Lauss
2008-07-03 9:55 ` Liam Girdwood
2008-07-03 10:09 ` ASoC: Au12x0/Au1550 PSC Audio support Manuel Lauss
2008-07-04 15:13 ` Takashi Iwai
2008-07-04 16:24 ` Manuel Lauss
2008-07-04 17:36 ` Liam Girdwood
2008-07-07 16:05 ` Takashi Iwai
2008-07-07 16:09 ` Liam Girdwood
2008-07-07 16:11 ` Takashi Iwai
2008-07-07 17:15 ` Manuel Lauss
2008-07-07 17:38 ` Manuel Lauss
2008-07-08 15:13 ` Liam Girdwood
2008-07-08 16:36 ` Manuel Lauss
2008-07-09 5:34 ` Manuel Lauss
2008-07-09 17:48 ` Takashi Iwai
2008-07-09 14:27 ` Manuel Lauss
2008-07-09 14:55 ` Liam Girdwood
2008-07-09 19:27 ` Takashi Iwai
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