[alsa-devel] [PATCH V4 1/3] ASoC: Alchemy AC97C/I2SC audio support
Liam Girdwood
lrg at ti.com
Mon Jul 25 11:32:21 CEST 2011
On 24/07/11 11:11, Manuel Lauss wrote:
> 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 at googlemail.com>
> ---
> V4: dropped hunk which removed I2S constants in au1000.h header to avoid merge
> conflicts with other patches, use the context structure in psc.h since it
> fits really well.
> V3: implemented feedback from Lars-Peter Clausen: src tidying, no more
> automatic dma device registration, split off db1000 board code.
> V2: added untested I2S controller driver for completeness, removed the audio
> defines from the au1000 header as well.
>
Looks mostly OK, I just have some questions below:-
> sound/soc/au1x/Kconfig | 19 +++
> sound/soc/au1x/Makefile | 8 +
> sound/soc/au1x/ac97c.c | 365 +++++++++++++++++++++++++++++++++++++++++++++
> sound/soc/au1x/dma.c | 374 +++++++++++++++++++++++++++++++++++++++++++++++
> sound/soc/au1x/i2sc.c | 342 +++++++++++++++++++++++++++++++++++++++++++
> sound/soc/au1x/psc.h | 19 ++-
> 6 files changed, 1118 insertions(+), 9 deletions(-)
> create mode 100644 sound/soc/au1x/ac97c.c
> create mode 100644 sound/soc/au1x/dma.c
> create mode 100644 sound/soc/au1x/i2sc.c
>
> 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 @@ config SND_SOC_AU1XPSC_AC97
> 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-dbdma-objs := dbdma2.o
> 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..35884ae
> --- /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 at googlemail.com>
> + *
> + * based on the old ALSA driver originally written by
> + * Charles Eidsness <charles at 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_8000_44100
Just curious, is there any reason this doesn't support 48kHz ?
> +
> +#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[PCM_TX] = r->start;
> +
> + r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
> + if (!r)
> + goto out1;
> + ctx->dmaids[PCM_RX] = 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..20fedbd
> --- /dev/null
> +++ b/sound/soc/au1x/dma.c
> @@ -0,0 +1,374 @@
> +/*
> + * Au1000/Au1500/Au1100 Audio DMA support.
> + *
> + * (c) 2011 Manuel Lauss <manuel.lauss at googlemail.com>
> + *
> + * copied almost verbatim from the old ALSA driver, written by
> + * Charles Eidsness <charles at 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[SUBSTREAM_TYPE(ss)]);
> +}
> +
> +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 stype = SUBSTREAM_TYPE(substream);
> + int *dmaids;
> + char *name;
> +
> + dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
> + if (!dmaids)
> + return -ENODEV; /* whoa, has ordering changed? */
> +
> + /* DMA setup */
> + name = (stype == PCM_TX) ? "audio-tx" : "audio-rx";
> + ctx->stream[stype].dma = request_au1000_dma(dmaids[stype], name,
> + au1000_dma_interrupt, IRQF_DISABLED,
> + &ctx->stream[stype]);
> + set_dma_mode(ctx->stream[stype].dma,
> + get_dma_mode(ctx->stream[stype].dma) & ~DMA_NC);
> +
> + ctx->stream[stype].substream = substream;
> + ctx->stream[stype].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_TYPE(substream);
> +
> + ctx->stream[SUBSTREAM_TYPE(substream)].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;
> + return au1000_setup_dma_link(stream,
> + params_period_bytes(hw_params),
> + params_periods(hw_params));
What happens if this fails ? You already have malloc'ed some pages.
> +}
> +
> +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_card *card,
> + struct snd_soc_dai *dai,
> + struct snd_pcm *pcm)
This API call has been updated to only pass the rtd *
> +{
> + 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..e3964a2
> --- /dev/null
> +++ b/sound/soc/au1x/i2sc.c
> @@ -0,0 +1,342 @@
> +/*
> + * Au1000/Au1500/Au1100 I2S controller driver for ASoC
> + *
> + * (c) 2011 Manuel Lauss <manuel.lauss at 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();
> +}
Btw, just wondering if arch/mips already supplies a suitable RD()/WR() for you ?
> +
> +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;
It's best to format all the switch statements consistently throughout your code.
> + }
> + 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[PCM_TX] = r->start;
> +
> + r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
> + if (!r)
> + goto out1;
> + ctx->dmaids[PCM_RX] = 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)
> +{
Should we not enalbe the clock here (i.e. in order to balance the clock off in suspend) ?
> + 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 at 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
Is there any need for these macros, SNDRV_PCM_STREAM_PLAYBACK and SNDRV_PCMP_STREAM_CAPTURE should be used for this type of logic.
> +
> +#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 @@ struct au1xpsc_audio_data {
>
> 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)
> --
> 1.7.6
>
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