ASoC: Alchemy AC97C/I2SC audio support
This patch adds ASoC support for the AC97 and I2S controllers
on the old Au1000/Au1500/Au1100 chips,
AC97 Tested on a Db1500. I2S untested since none of the boards
actually have an I2S codec wired up (just test pins).
Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
index 4b67140..0460b42 100644
--- a/sound/soc/au1x/Kconfig
+++ b/sound/soc/au1x/Kconfig
@@ -18,6 +18,25 @@
select SND_AC97_CODEC
select SND_SOC_AC97_BUS
+##
+## Au1000/1500/1100 DMA + AC97C/I2SC
+##
+config SND_SOC_AU1XAUDIO
+ tristate "SoC Audio for Au1000/Au1500/Au1100"
+ depends on MIPS_ALCHEMY
+ help
+ This is a driver set for the AC97 unit and the
+ old DMA controller as found on the Au1000/Au1500/Au1100 chips.
+
+config SND_SOC_AU1XAC97C
+ tristate
+ select AC97_BUS
+ select SND_AC97_CODEC
+ select SND_SOC_AC97_BUS
+
+config SND_SOC_AU1XI2SC
+ tristate
+
##
## Boards
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
index 1687307..ff5531e 100644
--- a/sound/soc/au1x/Makefile
+++ b/sound/soc/au1x/Makefile
@@ -3,9 +3,17 @@
snd-soc-au1xpsc-i2s-objs := psc-i2s.o
snd-soc-au1xpsc-ac97-objs := psc-ac97.o
+# Au1000/1500/1100 Audio units
+snd-soc-au1x-dma-objs := dma.o
+snd-soc-au1x-ac97c-objs := ac97c.o
+snd-soc-au1x-i2sc-objs := i2sc.o
+
obj-$(CONFIG_SND_SOC_AU1XPSC) += snd-soc-au1xpsc-dbdma.o
obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o
obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o
+obj-$(CONFIG_SND_SOC_AU1XAUDIO) += snd-soc-au1x-dma.o
+obj-$(CONFIG_SND_SOC_AU1XAC97C) += snd-soc-au1x-ac97c.o
+obj-$(CONFIG_SND_SOC_AU1XI2SC) += snd-soc-au1x-i2sc.o
# Boards
snd-soc-db1200-objs := db1200.o
diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c
new file mode 100644
index 0000000..9c05f38
--- /dev/null
+++ b/sound/soc/au1x/ac97c.c
@@ -0,0 +1,365 @@
+/*
+ * Au1000/Au1500/Au1100 AC97C controller driver for ASoC
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * based on the old ALSA driver originally written by
+ * Charles Eidsness <charles@cooper-street.com>
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/mutex.h>
+#include <linux/platform_device.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 "psc.h"
+
+/* register offsets and bits */
+#define AC97_CONFIG 0x00
+#define AC97_STATUS 0x04
+#define AC97_DATA 0x08
+#define AC97_CMDRESP 0x0c
+#define AC97_ENABLE 0x10
+
+#define CFG_RC(x) (((x) & 0x3ff) << 13) /* valid rx slots mask */
+#define CFG_XS(x) (((x) & 0x3ff) << 3) /* valid tx slots mask */
+#define CFG_SG (1 << 2) /* sync gate */
+#define CFG_SN (1 << 1) /* sync control */
+#define CFG_RS (1 << 0) /* acrst# control */
+#define STAT_XU (1 << 11) /* tx underflow */
+#define STAT_XO (1 << 10) /* tx overflow */
+#define STAT_RU (1 << 9) /* rx underflow */
+#define STAT_RO (1 << 8) /* rx overflow */
+#define STAT_RD (1 << 7) /* codec ready */
+#define STAT_CP (1 << 6) /* command pending */
+#define STAT_TE (1 << 4) /* tx fifo empty */
+#define STAT_TF (1 << 3) /* tx fifo full */
+#define STAT_RE (1 << 1) /* rx fifo empty */
+#define STAT_RF (1 << 0) /* rx fifo full */
+#define CMD_SET_DATA(x) (((x) & 0xffff) << 16)
+#define CMD_GET_DATA(x) ((x) & 0xffff)
+#define CMD_READ (1 << 7)
+#define CMD_WRITE (0 << 7)
+#define CMD_IDX(x) ((x) & 0x7f)
+#define EN_D (1 << 1) /* DISable bit */
+#define EN_CE (1 << 0) /* clock enable bit */
+
+/* how often to retry failed codec register reads/writes */
+#define AC97_RW_RETRIES 5
+
+#define AC97_RATES \
+ SNDRV_PCM_RATE_CONTINUOUS
+
+#define AC97_FMTS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE)
+
+/* instance data. There can be only one, MacLeod!!!!, fortunately there IS only
+ * once AC97C on early Alchemy chips. The newer ones aren't so lucky.
+ */
+static struct au1xpsc_audio_data *ac97c_workdata;
+#define ac97_to_ctx(x) ac97c_workdata
+
+static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg)
+{
+ return __raw_readl(ctx->mmio + reg);
+}
+
+static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v)
+{
+ __raw_writel(v, ctx->mmio + reg);
+ wmb();
+}
+
+static unsigned short au1xac97c_ac97_read(struct snd_ac97 *ac97,
+ unsigned short r)
+{
+ struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+ unsigned int tmo, retry;
+ unsigned long data;
+
+ data = ~0;
+ retry = AC97_RW_RETRIES;
+ do {
+ mutex_lock(&ctx->lock);
+
+ tmo = 5;
+ while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--)
+ udelay(21); /* wait an ac97 frame time */
+ if (!tmo) {
+ pr_debug("ac97rd timeout #1\n");
+ goto next;
+ }
+
+ WR(ctx, AC97_CMDRESP, CMD_IDX(r) | CMD_READ);
+
+ /* stupid errata: data is only valid for 21us, so
+ * poll, Forrest, poll...
+ */
+ tmo = 0x10000;
+ while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--)
+ asm volatile ("nop");
+ data = RD(ctx, AC97_CMDRESP);
+
+ if (!tmo)
+ pr_debug("ac97rd timeout #2\n");
+
+next:
+ mutex_unlock(&ctx->lock);
+ } while (--retry && !tmo);
+
+ pr_debug("AC97RD %04x %04lx %d\n", r, data, retry);
+
+ return retry ? data & 0xffff : 0xffff;
+}
+
+static void au1xac97c_ac97_write(struct snd_ac97 *ac97, unsigned short r,
+ unsigned short v)
+{
+ struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+ unsigned int tmo, retry;
+
+ retry = AC97_RW_RETRIES;
+ do {
+ mutex_lock(&ctx->lock);
+
+ for (tmo = 5; (RD(ctx, AC97_STATUS) & STAT_CP) && tmo; tmo--)
+ udelay(21);
+ if (!tmo) {
+ pr_debug("ac97wr timeout #1\n");
+ goto next;
+ }
+
+ WR(ctx, AC97_CMDRESP, CMD_WRITE | CMD_IDX(r) | CMD_SET_DATA(v));
+
+ for (tmo = 10; (RD(ctx, AC97_STATUS) & STAT_CP) && tmo; tmo--)
+ udelay(21);
+ if (!tmo)
+ pr_debug("ac97wr timeout #2\n");
+next:
+ mutex_unlock(&ctx->lock);
+ } while (--retry && !tmo);
+
+ pr_debug("AC97WR %04x %04x %d\n", r, v, retry);
+}
+
+static void au1xac97c_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+
+ WR(ctx, AC97_CONFIG, ctx->cfg | CFG_SG | CFG_SN);
+ msleep(20);
+ WR(ctx, AC97_CONFIG, ctx->cfg | CFG_SG);
+ WR(ctx, AC97_CONFIG, ctx->cfg);
+}
+
+static void au1xac97c_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+ struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+ int i;
+
+ WR(ctx, AC97_CONFIG, ctx->cfg | CFG_RS);
+ msleep(500);
+ WR(ctx, AC97_CONFIG, ctx->cfg);
+
+ /* wait for codec ready */
+ i = 50;
+ while (((RD(ctx, AC97_STATUS) & STAT_RD) == 0) && --i)
+ msleep(20);
+ if (!i)
+ printk(KERN_ERR "ac97c: codec not ready after cold reset\n");
+}
+
+/* AC97 controller operations */
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = au1xac97c_ac97_read,
+ .write = au1xac97c_ac97_write,
+ .reset = au1xac97c_ac97_cold_reset,
+ .warm_reset = au1xac97c_ac97_warm_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops); /* globals be gone! */
+
+static int alchemy_ac97c_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]);
+ return 0;
+}
+
+static struct snd_soc_dai_ops alchemy_ac97c_ops = {
+ .startup = alchemy_ac97c_startup,
+};
+
+static int au1xac97c_dai_probe(struct snd_soc_dai *dai)
+{
+ return ac97c_workdata ? 0 : -ENODEV;
+}
+
+static struct snd_soc_dai_driver au1xac97c_dai_driver = {
+ .name = "alchemy-ac97c",
+ .ac97_control = 1,
+ .probe = au1xac97c_dai_probe,
+ .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 = &alchemy_ac97c_ops,
+};
+
+static int __devinit au1xac97c_drvprobe(struct platform_device *pdev)
+{
+ int ret;
+ struct resource *r;
+ struct au1xpsc_audio_data *ctx;
+
+ ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+ if (!ctx)
+ return -ENOMEM;
+
+ mutex_init(&ctx->lock);
+
+ r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!r) {
+ ret = -ENODEV;
+ goto out0;
+ }
+
+ ret = -EBUSY;
+ if (!request_mem_region(r->start, resource_size(r), pdev->name))
+ goto out0;
+
+ ctx->mmio = ioremap_nocache(r->start, resource_size(r));
+ if (!ctx->mmio)
+ goto out1;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!r)
+ goto out1;
+ ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!r)
+ goto out1;
+ ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
+ /* switch it on */
+ WR(ctx, AC97_ENABLE, EN_D | EN_CE);
+ WR(ctx, AC97_ENABLE, EN_CE);
+
+ ctx->cfg = CFG_RC(3) | CFG_XS(3);
+ WR(ctx, AC97_CONFIG, ctx->cfg);
+
+ platform_set_drvdata(pdev, ctx);
+
+ ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver);
+ if (ret)
+ goto out1;
+
+ ac97c_workdata = ctx;
+ return 0;
+
+
+ snd_soc_unregister_dai(&pdev->dev);
+out1:
+ release_mem_region(r->start, resource_size(r));
+out0:
+ kfree(ctx);
+ return ret;
+}
+
+static int __devexit au1xac97c_drvremove(struct platform_device *pdev)
+{
+ struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
+ struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+
+ snd_soc_unregister_dai(&pdev->dev);
+
+ WR(ctx, AC97_ENABLE, EN_D); /* clock off, disable */
+
+ iounmap(ctx->mmio);
+ release_mem_region(r->start, resource_size(r));
+ kfree(ctx);
+
+ ac97c_workdata = NULL; /* MDEV */
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int au1xac97c_drvsuspend(struct device *dev)
+{
+ struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+ WR(ctx, AC97_ENABLE, EN_D); /* clock off, disable */
+
+ return 0;
+}
+
+static int au1xac97c_drvresume(struct device *dev)
+{
+ struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+ WR(ctx, AC97_ENABLE, EN_D | EN_CE);
+ WR(ctx, AC97_ENABLE, EN_CE);
+ WR(ctx, AC97_CONFIG, ctx->cfg);
+
+ return 0;
+}
+
+static const struct dev_pm_ops au1xpscac97_pmops = {
+ .suspend = au1xac97c_drvsuspend,
+ .resume = au1xac97c_drvresume,
+};
+
+#define AU1XPSCAC97_PMOPS (&au1xpscac97_pmops)
+
+#else
+
+#define AU1XPSCAC97_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xac97c_driver = {
+ .driver = {
+ .name = "alchemy-ac97c",
+ .owner = THIS_MODULE,
+ .pm = AU1XPSCAC97_PMOPS,
+ },
+ .probe = au1xac97c_drvprobe,
+ .remove = __devexit_p(au1xac97c_drvremove),
+};
+
+static int __init au1xac97c_load(void)
+{
+ ac97c_workdata = NULL;
+ return platform_driver_register(&au1xac97c_driver);
+}
+
+static void __exit au1xac97c_unload(void)
+{
+ platform_driver_unregister(&au1xac97c_driver);
+}
+
+module_init(au1xac97c_load);
+module_exit(au1xac97c_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c
new file mode 100644
index 0000000..7aa5b76
--- /dev/null
+++ b/sound/soc/au1x/dma.c
@@ -0,0 +1,377 @@
+/*
+ * Au1000/Au1500/Au1100 Audio DMA support.
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * copied almost verbatim from the old ALSA driver, written by
+ * Charles Eidsness <charles@cooper-street.com>
+ */
+
+#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/au1000_dma.h>
+
+#include "psc.h"
+
+#define ALCHEMY_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)
+
+struct pcm_period {
+ u32 start;
+ u32 relative_end; /* relative to start of buffer */
+ struct pcm_period *next;
+};
+
+struct audio_stream {
+ struct snd_pcm_substream *substream;
+ int dma;
+ struct pcm_period *buffer;
+ unsigned int period_size;
+ unsigned int periods;
+};
+
+struct alchemy_pcm_ctx {
+ struct audio_stream stream[2]; /* playback & capture */
+};
+
+static void au1000_release_dma_link(struct audio_stream *stream)
+{
+ struct pcm_period *pointer;
+ struct pcm_period *pointer_next;
+
+ stream->period_size = 0;
+ stream->periods = 0;
+ pointer = stream->buffer;
+ if (!pointer)
+ return;
+ do {
+ pointer_next = pointer->next;
+ kfree(pointer);
+ pointer = pointer_next;
+ } while (pointer != stream->buffer);
+ stream->buffer = NULL;
+}
+
+static int au1000_setup_dma_link(struct audio_stream *stream,
+ unsigned int period_bytes,
+ unsigned int periods)
+{
+ struct snd_pcm_substream *substream = stream->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_period *pointer;
+ unsigned long dma_start;
+ int i;
+
+ dma_start = virt_to_phys(runtime->dma_area);
+
+ if (stream->period_size == period_bytes &&
+ stream->periods == periods)
+ return 0; /* not changed */
+
+ au1000_release_dma_link(stream);
+
+ stream->period_size = period_bytes;
+ stream->periods = periods;
+
+ stream->buffer = kmalloc(sizeof(struct pcm_period), GFP_KERNEL);
+ if (!stream->buffer)
+ return -ENOMEM;
+ pointer = stream->buffer;
+ for (i = 0; i < periods; i++) {
+ pointer->start = (u32)(dma_start + (i * period_bytes));
+ pointer->relative_end = (u32) (((i+1) * period_bytes) - 0x1);
+ if (i < periods - 1) {
+ pointer->next = kmalloc(sizeof(struct pcm_period),
+ GFP_KERNEL);
+ if (!pointer->next) {
+ au1000_release_dma_link(stream);
+ return -ENOMEM;
+ }
+ pointer = pointer->next;
+ }
+ }
+ pointer->next = stream->buffer;
+ return 0;
+}
+
+static void au1000_dma_stop(struct audio_stream *stream)
+{
+ if (stream->buffer)
+ disable_dma(stream->dma);
+}
+
+static void au1000_dma_start(struct audio_stream *stream)
+{
+ if (!stream->buffer)
+ return;
+
+ init_dma(stream->dma);
+ if (get_dma_active_buffer(stream->dma) == 0) {
+ clear_dma_done0(stream->dma);
+ set_dma_addr0(stream->dma, stream->buffer->start);
+ set_dma_count0(stream->dma, stream->period_size >> 1);
+ set_dma_addr1(stream->dma, stream->buffer->next->start);
+ set_dma_count1(stream->dma, stream->period_size >> 1);
+ } else {
+ clear_dma_done1(stream->dma);
+ set_dma_addr1(stream->dma, stream->buffer->start);
+ set_dma_count1(stream->dma, stream->period_size >> 1);
+ set_dma_addr0(stream->dma, stream->buffer->next->start);
+ set_dma_count0(stream->dma, stream->period_size >> 1);
+ }
+ enable_dma_buffers(stream->dma);
+ start_dma(stream->dma);
+}
+
+static irqreturn_t au1000_dma_interrupt(int irq, void *ptr)
+{
+ struct audio_stream *stream = (struct audio_stream *)ptr;
+ struct snd_pcm_substream *substream = stream->substream;
+
+ switch (get_dma_buffer_done(stream->dma)) {
+ case DMA_D0:
+ stream->buffer = stream->buffer->next;
+ clear_dma_done0(stream->dma);
+ set_dma_addr0(stream->dma, stream->buffer->next->start);
+ set_dma_count0(stream->dma, stream->period_size >> 1);
+ enable_dma_buffer0(stream->dma);
+ break;
+ case DMA_D1:
+ stream->buffer = stream->buffer->next;
+ clear_dma_done1(stream->dma);
+ set_dma_addr1(stream->dma, stream->buffer->next->start);
+ set_dma_count1(stream->dma, stream->period_size >> 1);
+ enable_dma_buffer1(stream->dma);
+ break;
+ case (DMA_D0 | DMA_D1):
+ pr_debug("DMA %d missed interrupt.\n", stream->dma);
+ au1000_dma_stop(stream);
+ au1000_dma_start(stream);
+ break;
+ case (~DMA_D0 & ~DMA_D1):
+ pr_debug("DMA %d empty irq.\n", stream->dma);
+ }
+ snd_pcm_period_elapsed(substream);
+ return IRQ_HANDLED;
+}
+
+static const struct snd_pcm_hardware alchemy_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
+ .formats = ALCHEMY_PCM_FMTS,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .rate_min = SNDRV_PCM_RATE_8000,
+ .rate_max = SNDRV_PCM_RATE_192000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .period_bytes_min = 1024,
+ .period_bytes_max = 16 * 1024 - 1,
+ .periods_min = 4,
+ .periods_max = 255,
+ .buffer_bytes_max = 128 * 1024,
+ .fifo_size = 16,
+};
+
+static inline struct alchemy_pcm_ctx *ss_to_ctx(struct snd_pcm_substream *ss)
+{
+ struct snd_soc_pcm_runtime *rtd = ss->private_data;
+ return snd_soc_platform_get_drvdata(rtd->platform);
+}
+
+static inline struct audio_stream *ss_to_as(struct snd_pcm_substream *ss)
+{
+ struct alchemy_pcm_ctx *ctx = ss_to_ctx(ss);
+ return &(ctx->stream[ss->stream]);
+}
+
+static int alchemy_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream);
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int *dmaids, s = substream->stream;
+ char *name;
+
+ dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ if (!dmaids)
+ return -ENODEV; /* whoa, has ordering changed? */
+
+ /* DMA setup */
+ name = (s == SNDRV_PCM_STREAM_PLAYBACK) ? "audio-tx" : "audio-rx";
+ ctx->stream[s].dma = request_au1000_dma(dmaids[s], name,
+ au1000_dma_interrupt, IRQF_DISABLED,
+ &ctx->stream[s]);
+ set_dma_mode(ctx->stream[s].dma,
+ get_dma_mode(ctx->stream[s].dma) & ~DMA_NC);
+
+ ctx->stream[s].substream = substream;
+ ctx->stream[s].buffer = NULL;
+ snd_soc_set_runtime_hwparams(substream, &alchemy_pcm_hardware);
+
+ return 0;
+}
+
+static int alchemy_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream);
+ int stype = substream->stream;
+
+ ctx->stream[stype].substream = NULL;
+ free_au1000_dma(ctx->stream[stype].dma);
+
+ return 0;
+}
+
+static int alchemy_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct audio_stream *stream = ss_to_as(substream);
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ err = au1000_setup_dma_link(stream,
+ params_period_bytes(hw_params),
+ params_periods(hw_params));
+ if (err)
+ snd_pcm_lib_free_pages(substream);
+
+ return err;
+}
+
+static int alchemy_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct audio_stream *stream = ss_to_as(substream);
+ au1000_release_dma_link(stream);
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int alchemy_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct audio_stream *stream = ss_to_as(substream);
+ int err = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ au1000_dma_start(stream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ au1000_dma_stop(stream);
+ break;
+ default:
+ err = -EINVAL;
+ break;
+ }
+ return err;
+}
+
+static snd_pcm_uframes_t alchemy_pcm_pointer(struct snd_pcm_substream *ss)
+{
+ struct audio_stream *stream = ss_to_as(ss);
+ long location;
+
+ location = get_dma_residue(stream->dma);
+ location = stream->buffer->relative_end - location;
+ if (location == -1)
+ location = 0;
+ return bytes_to_frames(ss->runtime, location);
+}
+
+static struct snd_pcm_ops alchemy_pcm_ops = {
+ .open = alchemy_pcm_open,
+ .close = alchemy_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = alchemy_pcm_hw_params,
+ .hw_free = alchemy_pcm_hw_free,
+ .trigger = alchemy_pcm_trigger,
+ .pointer = alchemy_pcm_pointer,
+};
+
+static void alchemy_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int alchemy_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm *pcm = rtd->pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL), 65536, (4096 * 1024) - 1);
+
+ return 0;
+}
+
+struct snd_soc_platform_driver alchemy_pcm_soc_platform = {
+ .ops = &alchemy_pcm_ops,
+ .pcm_new = alchemy_pcm_new,
+ .pcm_free = alchemy_pcm_free_dma_buffers,
+};
+
+static int __devinit alchemy_pcm_drvprobe(struct platform_device *pdev)
+{
+ struct alchemy_pcm_ctx *ctx;
+ int ret;
+
+ ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+ if (!ctx)
+ return -ENOMEM;
+
+ platform_set_drvdata(pdev, ctx);
+
+ ret = snd_soc_register_platform(&pdev->dev, &alchemy_pcm_soc_platform);
+ if (ret)
+ kfree(ctx);
+
+ return ret;
+}
+
+static int __devexit alchemy_pcm_drvremove(struct platform_device *pdev)
+{
+ struct alchemy_pcm_ctx *ctx = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_platform(&pdev->dev);
+ kfree(ctx);
+
+ return 0;
+}
+
+static struct platform_driver alchemy_pcmdma_driver = {
+ .driver = {
+ .name = "alchemy-pcm-dma",
+ .owner = THIS_MODULE,
+ },
+ .probe = alchemy_pcm_drvprobe,
+ .remove = __devexit_p(alchemy_pcm_drvremove),
+};
+
+static int __init alchemy_pcmdma_load(void)
+{
+ return platform_driver_register(&alchemy_pcmdma_driver);
+}
+
+static void __exit alchemy_pcmdma_unload(void)
+{
+ platform_driver_unregister(&alchemy_pcmdma_driver);
+}
+
+module_init(alchemy_pcmdma_load);
+module_exit(alchemy_pcmdma_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/Au1500/Au1100 Audio DMA driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c
new file mode 100644
index 0000000..b4172fd
--- /dev/null
+++ b/sound/soc/au1x/i2sc.c
@@ -0,0 +1,347 @@
+/*
+ * Au1000/Au1500/Au1100 I2S controller driver for ASoC
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * Note: clock supplied to the I2S controller must be 256x samplerate.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+
+#include "psc.h"
+
+#define I2S_RXTX 0x00
+#define I2S_CFG 0x04
+#define I2S_ENABLE 0x08
+
+#define CFG_XU (1 << 25) /* tx underflow */
+#define CFG_XO (1 << 24)
+#define CFG_RU (1 << 23)
+#define CFG_RO (1 << 22)
+#define CFG_TR (1 << 21)
+#define CFG_TE (1 << 20)
+#define CFG_TF (1 << 19)
+#define CFG_RR (1 << 18)
+#define CFG_RF (1 << 17)
+#define CFG_ICK (1 << 12) /* clock invert */
+#define CFG_PD (1 << 11) /* set to make I2SDIO INPUT */
+#define CFG_LB (1 << 10) /* loopback */
+#define CFG_IC (1 << 9) /* word select invert */
+#define CFG_FM_I2S (0 << 7) /* I2S format */
+#define CFG_FM_LJ (1 << 7) /* left-justified */
+#define CFG_FM_RJ (2 << 7) /* right-justified */
+#define CFG_FM_MASK (3 << 7)
+#define CFG_TN (1 << 6) /* tx fifo en */
+#define CFG_RN (1 << 5) /* rx fifo en */
+#define CFG_SZ_8 (0x08)
+#define CFG_SZ_16 (0x10)
+#define CFG_SZ_18 (0x12)
+#define CFG_SZ_20 (0x14)
+#define CFG_SZ_24 (0x18)
+#define CFG_SZ_MASK (0x1f)
+#define EN_D (1 << 1) /* DISable */
+#define EN_CE (1 << 0) /* clock enable */
+
+/* only limited by clock generator and board design */
+#define AU1XI2SC_RATES \
+ SNDRV_PCM_RATE_CONTINUOUS
+
+#define AU1XI2SC_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_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | \
+ SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_U18_3BE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \
+ SNDRV_PCM_FMTBIT_S20_3BE | SNDRV_PCM_FMTBIT_U20_3BE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \
+ SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE | \
+ 0)
+
+static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg)
+{
+ return __raw_readl(ctx->mmio + reg);
+}
+
+static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v)
+{
+ __raw_writel(v, ctx->mmio + reg);
+ wmb();
+}
+
+static int au1xi2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(cpu_dai);
+ unsigned long c;
+ int ret;
+
+ ret = -EINVAL;
+ c = ctx->cfg;
+
+ c &= ~CFG_FM_MASK;
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ c |= CFG_FM_I2S;
+ break;
+ case SND_SOC_DAIFMT_MSB:
+ c |= CFG_FM_RJ;
+ break;
+ case SND_SOC_DAIFMT_LSB:
+ c |= CFG_FM_LJ;
+ break;
+ default:
+ goto out;
+ }
+
+ c &= ~(CFG_IC | CFG_ICK); /* IB-IF */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ c |= CFG_IC | CFG_ICK;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ c |= CFG_IC;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ c |= CFG_ICK;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ break;
+ default:
+ goto out;
+ }
+
+ /* I2S controller only supports master */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */
+ break;
+ default:
+ goto out;
+ }
+
+ ret = 0;
+ ctx->cfg = c;
+out:
+ return ret;
+}
+
+static int au1xi2s_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ int stype = SUBSTREAM_TYPE(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ /* power up */
+ WR(ctx, I2S_ENABLE, EN_D | EN_CE);
+ WR(ctx, I2S_ENABLE, EN_CE);
+ ctx->cfg |= (stype == PCM_TX) ? CFG_TN : CFG_RN;
+ WR(ctx, I2S_CFG, ctx->cfg);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ ctx->cfg &= ~((stype == PCM_TX) ? CFG_TN : CFG_RN);
+ WR(ctx, I2S_CFG, ctx->cfg);
+ WR(ctx, I2S_ENABLE, EN_D); /* power off */
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static unsigned long msbits_to_reg(int msbits)
+{
+ switch (msbits) {
+ case 8:
+ return CFG_SZ_8;
+ case 16:
+ return CFG_SZ_16;
+ case 18:
+ return CFG_SZ_18;
+ case 20:
+ return CFG_SZ_20;
+ case 24:
+ return CFG_SZ_24;
+ }
+ return 0;
+}
+
+static int au1xi2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ unsigned long v;
+
+ v = msbits_to_reg(params->msbits);
+ if (!v)
+ return -EINVAL;
+
+ ctx->cfg &= ~CFG_SZ_MASK;
+ ctx->cfg |= v;
+ return 0;
+}
+
+static int au1xi2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]);
+ return 0;
+}
+
+static const struct snd_soc_dai_ops au1xi2s_dai_ops = {
+ .startup = au1xi2s_startup,
+ .trigger = au1xi2s_trigger,
+ .hw_params = au1xi2s_hw_params,
+ .set_fmt = au1xi2s_set_fmt,
+};
+
+static struct snd_soc_dai_driver au1xi2s_dai_driver = {
+ .symmetric_rates = 1,
+ .playback = {
+ .rates = AU1XI2SC_RATES,
+ .formats = AU1XI2SC_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .capture = {
+ .rates = AU1XI2SC_RATES,
+ .formats = AU1XI2SC_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .ops = &au1xi2s_dai_ops,
+};
+
+static int __devinit au1xi2s_drvprobe(struct platform_device *pdev)
+{
+ int ret;
+ struct resource *r;
+ struct au1xpsc_audio_data *ctx;
+
+ ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+ if (!ctx)
+ return -ENOMEM;
+
+ r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!r) {
+ ret = -ENODEV;
+ goto out0;
+ }
+
+ ret = -EBUSY;
+ if (!request_mem_region(r->start, resource_size(r), pdev->name))
+ goto out0;
+
+ ctx->mmio = ioremap_nocache(r->start, resource_size(r));
+ if (!ctx->mmio)
+ goto out1;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!r)
+ goto out1;
+ ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!r)
+ goto out1;
+ ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
+ platform_set_drvdata(pdev, ctx);
+
+ ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver);
+ if (ret)
+ goto out1;
+
+ return 0;
+
+ snd_soc_unregister_dai(&pdev->dev);
+out1:
+ release_mem_region(r->start, resource_size(r));
+out0:
+ kfree(ctx);
+ return ret;
+}
+
+static int __devexit au1xi2s_drvremove(struct platform_device *pdev)
+{
+ struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
+ struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+
+ snd_soc_unregister_dai(&pdev->dev);
+
+ WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */
+
+ iounmap(ctx->mmio);
+ release_mem_region(r->start, resource_size(r));
+ kfree(ctx);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int au1xi2s_drvsuspend(struct device *dev)
+{
+ struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+ WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */
+
+ return 0;
+}
+
+static int au1xi2s_drvresume(struct device *dev)
+{
+ return 0;
+}
+
+static const struct dev_pm_ops au1xi2sc_pmops = {
+ .suspend = au1xi2s_drvsuspend,
+ .resume = au1xi2s_drvresume,
+};
+
+#define AU1XI2SC_PMOPS (&au1xi2sc_pmops)
+
+#else
+
+#define AU1XI2SC_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xi2s_driver = {
+ .driver = {
+ .name = "alchemy-i2sc",
+ .owner = THIS_MODULE,
+ .pm = AU1XI2SC_PMOPS,
+ },
+ .probe = au1xi2s_drvprobe,
+ .remove = __devexit_p(au1xi2s_drvremove),
+};
+
+static int __init au1xi2s_load(void)
+{
+ return platform_driver_register(&au1xi2s_driver);
+}
+
+static void __exit au1xi2s_unload(void)
+{
+ platform_driver_unregister(&au1xi2s_driver);
+}
+
+module_init(au1xi2s_load);
+module_exit(au1xi2s_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/1500/1100 I2S ASoC driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
index b30eadd..c59b9e5 100644
--- a/sound/soc/au1x/psc.h
+++ b/sound/soc/au1x/psc.h
@@ -1,7 +1,7 @@
/*
- * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ * Alchemy ALSA ASoC audio support.
*
- * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ * (c) 2007-2011 MSC Vertriebsges.m.b.H.,
* Manuel Lauss <manuel.lauss@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
@@ -13,7 +13,13 @@
#ifndef _AU1X_PCM_H
#define _AU1X_PCM_H
-/* DBDMA helpers */
+#define PCM_TX 0
+#define PCM_RX 1
+
+#define SUBSTREAM_TYPE(substream) \
+ ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX)
+
+/* PSC/DBDMA helpers */
extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev);
extern void au1xpsc_pcm_destroy(struct platform_device *dmapd);
@@ -27,15 +33,10 @@
unsigned long pm[2];
struct mutex lock;
+ int dmaids[2];
struct platform_device *dmapd;
};
-#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)