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 --- V5: incorporate feedback from Liam Girdwood 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.
sound/soc/au1x/Kconfig | 19 +++ sound/soc/au1x/Makefile | 8 + sound/soc/au1x/ac97c.c | 365 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/au1x/dma.c | 377 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/au1x/i2sc.c | 347 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/au1x/psc.h | 19 ++- 6 files changed, 1126 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..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 @@ 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)