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@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_ALCHEMYhelpThis is a driver set for the AC97 unit and theold DMA controller as found on the Au1000/Au1500/Au1100 chips.+config SND_SOC_AU1XAC97C
tristateselect AC97_BUSselect SND_AC97_CODECselect 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@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_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@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[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@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@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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