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-rw-r--r--sound/soc/au1x/Kconfig32
-rw-r--r--sound/soc/au1x/Makefile13
-rw-r--r--sound/soc/au1x/dbdma2.c421
-rw-r--r--sound/soc/au1x/psc-ac97.c387
-rw-r--r--sound/soc/au1x/psc-i2s.c414
-rw-r--r--sound/soc/au1x/psc.h53
-rw-r--r--sound/soc/au1x/sample-ac97.c144
7 files changed, 1464 insertions, 0 deletions
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
new file mode 100644
index 000000000000..410a893aa66b
--- /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 000000000000..6c6950b8003a
--- /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 000000000000..1466d9328800
--- /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 000000000000..57facbad6825
--- /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 000000000000..ba4b5c199f21
--- /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 000000000000..8fdb1a04a07b
--- /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 000000000000..f75ae7f62c3d
--- /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>");
generated by cgit v1.2.3 (git 2.39.0) at 2024-12-21 03:14:34 +0300