From 64a648c2204b0c750fe49828158751183d8b5f83 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 25 Jul 2011 11:15:15 +0100 Subject: ASoC: dapm - Add DAPM stream completion event. In preparation for Dynamic PCM (AKA DSP) support. This adds a callback function to be called at the completion of a DAPM stream event. This can be used by DSP components to perform calculations based on DAPM graphs after completion of stream events. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 ++ sound/soc/soc-dapm.c | 4 ++++ 2 files changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83ad8ca27490..9d3935bbbd0c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3141,6 +3141,7 @@ int snd_soc_register_platform(struct device *dev, platform->driver = platform_drv; platform->dapm.dev = dev; platform->dapm.platform = platform; + platform->dapm.stream_event = platform_drv->stream_event; mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); @@ -3253,6 +3254,7 @@ int snd_soc_register_codec(struct device *dev, codec->dapm.dev = dev; codec->dapm.codec = codec; codec->dapm.seq_notifier = codec_drv->seq_notifier; + codec->dapm.stream_event = codec_drv->stream_event; codec->dev = dev; codec->driver = codec_drv; codec->num_dai = num_dai; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e15914b3633..612a2a28979a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2604,6 +2604,10 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, } dapm_power_widgets(dapm, event); + + /* do we need to notify any clients that DAPM stream is complete */ + if (dapm->stream_event) + dapm->stream_event(dapm, event); } /** -- cgit v1.2.3 From ee47b364860bb21580cc105e6bb6e0dd76b75ad2 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 25 Jul 2011 11:15:50 +0100 Subject: ASoC: dapm - change stream event dbg to vdgb Stream event debug can be noisy on larger audio devices so improve the debug SNR by changing it to the verbose level. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 612a2a28979a..c26531132c66 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2584,7 +2584,7 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, { if (!w->sname || w->dapm != dapm) continue; - dev_dbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", + dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { switch(event) { -- cgit v1.2.3 From b3c70c9ea62a3ae6c63536e43fa28f965a56de91 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Mon, 25 Jul 2011 13:44:45 +0200 Subject: ASoC: Alchemy AC97C/I2SC audio support This patch adds ASoC support for the AC97 and I2S controllers on the old Au1000/Au1500/Au1100 chips, AC97 Tested on a Db1500. I2S untested since none of the boards actually have an I2S codec wired up (just test pins). Signed-off-by: Manuel Lauss Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- 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 (limited to 'sound') diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig index 4b67140fdec3..0460b428862c 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 16873076e8c4..ff5531eee613 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 000000000000..9c05f381d95e --- /dev/null +++ b/sound/soc/au1x/ac97c.c @@ -0,0 +1,365 @@ +/* + * Au1000/Au1500/Au1100 AC97C controller driver for ASoC + * + * (c) 2011 Manuel Lauss + * + * based on the old ALSA driver originally written by + * Charles Eidsness + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#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 000000000000..7aa5b7606777 --- /dev/null +++ b/sound/soc/au1x/dma.c @@ -0,0 +1,377 @@ +/* + * Au1000/Au1500/Au1100 Audio DMA support. + * + * (c) 2011 Manuel Lauss + * + * copied almost verbatim from the old ALSA driver, written by + * Charles Eidsness + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#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 000000000000..b4172fdd2c48 --- /dev/null +++ b/sound/soc/au1x/i2sc.c @@ -0,0 +1,347 @@ +/* + * Au1000/Au1500/Au1100 I2S controller driver for ASoC + * + * (c) 2011 Manuel Lauss + * + * Note: clock supplied to the I2S controller must be 256x samplerate. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#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 b30eadd422a7..c59b9e544e72 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 * * 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) -- cgit v1.2.3 From b2ce305dda483e59a78d5aa6e4211034c0cea38d Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Mon, 25 Jul 2011 13:44:46 +0200 Subject: ASoC: Add a DB1x00 AC97 machine driver Add a machine driver suitable for the AC97 part on the DB1000/DB1500/DB1100 boards. Run-tested on DB1500. Signed-off-by: Manuel Lauss Acked-by: Ralf Baechle Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/au1x/Kconfig | 9 ++++++ sound/soc/au1x/Makefile | 2 ++ sound/soc/au1x/db1000.c | 75 +++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 86 insertions(+) create mode 100644 sound/soc/au1x/db1000.c (limited to 'sound') diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig index 0460b428862c..6d592546e8fc 100644 --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -41,6 +41,15 @@ config SND_SOC_AU1XI2SC ## ## Boards ## +config SND_SOC_DB1000 + tristate "DB1000 Audio support" + depends on SND_SOC_AU1XAUDIO + select SND_SOC_AU1XAC97C + select SND_SOC_AC97_CODEC + help + Select this option to enable AC97 audio on the early DB1x00 series + of boards (DB1000/DB1500/DB1100). + config SND_SOC_DB1200 tristate "DB1200 AC97+I2S audio support" depends on SND_SOC_AU1XPSC diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile index ff5531eee613..920710514ea0 100644 --- a/sound/soc/au1x/Makefile +++ b/sound/soc/au1x/Makefile @@ -16,6 +16,8 @@ obj-$(CONFIG_SND_SOC_AU1XAC97C) += snd-soc-au1x-ac97c.o obj-$(CONFIG_SND_SOC_AU1XI2SC) += snd-soc-au1x-i2sc.o # Boards +snd-soc-db1000-objs := db1000.o snd-soc-db1200-objs := db1200.o +obj-$(CONFIG_SND_SOC_DB1000) += snd-soc-db1000.o obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c new file mode 100644 index 000000000000..127477a5e0c7 --- /dev/null +++ b/sound/soc/au1x/db1000.c @@ -0,0 +1,75 @@ +/* + * DB1000/DB1500/DB1100 ASoC audio fabric support code. + * + * (c) 2011 Manuel Lauss + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "psc.h" + +static struct snd_soc_dai_link db1000_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .codec_dai_name = "ac97-hifi", + .cpu_dai_name = "alchemy-ac97c", + .platform_name = "alchemy-pcm-dma.0", + .codec_name = "ac97-codec", +}; + +static struct snd_soc_card db1000_ac97 = { + .name = "DB1000_AC97", + .dai_link = &db1000_ac97_dai, + .num_links = 1, +}; + +static int __devinit db1000_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &db1000_ac97; + card->dev = &pdev->dev; + return snd_soc_register_card(card); +} + +static int __devexit db1000_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + snd_soc_unregister_card(card); + return 0; +} + +static struct platform_driver db1000_audio_driver = { + .driver = { + .name = "db1000-audio", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = db1000_audio_probe, + .remove = __devexit_p(db1000_audio_remove), +}; + +static int __init db1000_audio_load(void) +{ + return platform_driver_register(&db1000_audio_driver); +} + +static void __exit db1000_audio_unload(void) +{ + platform_driver_unregister(&db1000_audio_driver); +} + +module_init(db1000_audio_load); +module_exit(db1000_audio_unload); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("DB1000/DB1500/DB1100 ASoC audio"); +MODULE_AUTHOR("Manuel Lauss"); -- cgit v1.2.3 From 7137c6bcb7ff5d0e6f63f8a4175d5b77dc79abc0 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Mon, 25 Jul 2011 13:44:47 +0200 Subject: ALSA: deprecate MIPS AU1X00 AC97 driver Now that an ASoC variant is available, tell users that this driver is now living on borrowed time... Signed-off-by: Manuel Lauss Acked-by: Ralf Baechle Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/mips/Kconfig | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index a9823fad85c2..77dd0a13aecc 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -23,12 +23,15 @@ config SND_SGI_HAL2 config SND_AU1X00 - tristate "Au1x00 AC97 Port Driver" + tristate "Au1x00 AC97 Port Driver (DEPRECATED)" depends on SOC_AU1000 || SOC_AU1100 || SOC_AU1500 select SND_PCM select SND_AC97_CODEC help ALSA Sound driver for the Au1x00's AC97 port. + Newer drivers for ASoC are available, please do not use + this driver as it will be removed in the future. + endif # SND_MIPS -- cgit v1.2.3 From 5b0912be7a8ff1dbfe56358c5f933d65445bb8af Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Mon, 25 Jul 2011 13:45:02 +0200 Subject: ASoC: au1x: remove automatic DMA device registration from PSC drivers The PSC audio drivers (psc-ac97/psc-i2s) register the DMA platform_device on their own. This is frowned upon, from now on board code must register a simple pcm dma platform device for each PSC with sound duties. Signed-off-by: Manuel Lauss Acked-by: Ralf Baechle Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/au1x/dbdma2.c | 83 +++++++---------------------------------------- sound/soc/au1x/psc-ac97.c | 34 +++++++++++++------ sound/soc/au1x/psc-i2s.c | 32 ++++++++++++------ sound/soc/au1x/psc.h | 5 --- 4 files changed, 58 insertions(+), 96 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 20bb53a837b1..fd5378f7dece 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -293,6 +293,16 @@ au1xpsc_pcm_pointer(struct snd_pcm_substream *substream) static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) { + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int stype = SUBSTREAM_TYPE(substream), *dmaids; + + dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dmaids) + return -ENODEV; /* whoa, has ordering changed? */ + + pcd->ddma_id = dmaids[stype]; + snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware); return 0; } @@ -340,36 +350,18 @@ struct snd_soc_platform_driver au1xpsc_soc_platform = { static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev) { struct au1xpsc_audio_dmadata *dmadata; - struct resource *r; int ret; dmadata = kzalloc(2 * sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); if (!dmadata) return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) { - ret = -ENODEV; - goto out1; - } - dmadata[PCM_TX].ddma_id = r->start; - - /* RX DMA */ - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) { - ret = -ENODEV; - goto out1; - } - dmadata[PCM_RX].ddma_id = r->start; - platform_set_drvdata(pdev, dmadata); ret = snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform); - if (!ret) - return ret; + if (ret) + kfree(dmadata); -out1: - kfree(dmadata); return ret; } @@ -405,57 +397,6 @@ static void __exit au1xpsc_audio_dbdma_unload(void) module_init(au1xpsc_audio_dbdma_load); module_exit(au1xpsc_audio_dbdma_unload); - -struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev) -{ - struct resource *res, *r; - struct platform_device *pd; - int id[2]; - int ret; - - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) - return NULL; - id[0] = r->start; - - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) - return NULL; - id[1] = r->start; - - res = kzalloc(sizeof(struct resource) * 2, GFP_KERNEL); - if (!res) - return NULL; - - res[0].start = res[0].end = id[0]; - res[1].start = res[1].end = id[1]; - res[0].flags = res[1].flags = IORESOURCE_DMA; - - pd = platform_device_alloc("au1xpsc-pcm", pdev->id); - if (!pd) - goto out; - - pd->resource = res; - pd->num_resources = 2; - - ret = platform_device_add(pd); - if (!ret) - return pd; - - platform_device_put(pd); -out: - kfree(res); - return NULL; -} -EXPORT_SYMBOL_GPL(au1xpsc_pcm_add); - -void au1xpsc_pcm_destroy(struct platform_device *dmapd) -{ - if (dmapd) - platform_device_unregister(dmapd); -} -EXPORT_SYMBOL_GPL(au1xpsc_pcm_destroy); - MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver"); MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index d0db66f24a00..44296abfc38f 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -324,12 +324,21 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, return ret; } +static int au1xpsc_ac97_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]); + return 0; +} + static int au1xpsc_ac97_probe(struct snd_soc_dai *dai) { return au1xpsc_ac97_workdata ? 0 : -ENODEV; } static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { + .startup = au1xpsc_ac97_startup, .trigger = au1xpsc_ac97_trigger, .hw_params = au1xpsc_ac97_hw_params, }; @@ -379,6 +388,16 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) if (!wd->mmio) goto out1; + r = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r) + goto out2; + wd->dmaids[PCM_TX] = r->start; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) + goto out2; + wd->dmaids[PCM_RX] = r->start; + /* configuration: max dma trigger threshold, enable ac97 */ wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | PSC_AC97CFG_DE_ENABLE; @@ -401,15 +420,13 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); if (ret) - goto out1; + goto out2; - wd->dmapd = au1xpsc_pcm_add(pdev); - if (wd->dmapd) { - au1xpsc_ac97_workdata = wd; - return 0; - } + au1xpsc_ac97_workdata = wd; + return 0; - snd_soc_unregister_dai(&pdev->dev); +out2: + iounmap(wd->mmio); out1: release_mem_region(r->start, resource_size(r)); out0: @@ -422,9 +439,6 @@ static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev) struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (wd->dmapd) - au1xpsc_pcm_destroy(wd->dmapd); - snd_soc_unregister_dai(&pdev->dev); /* disable PSC completely */ diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index fca091276320..1b7ab5d422e0 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -257,7 +257,16 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int au1xpsc_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]); + return 0; +} + static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { + .startup = au1xpsc_i2s_startup, .trigger = au1xpsc_i2s_trigger, .hw_params = au1xpsc_i2s_hw_params, .set_fmt = au1xpsc_i2s_set_fmt, @@ -304,6 +313,16 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) if (!wd->mmio) goto out1; + r = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r) + goto out2; + wd->dmaids[PCM_TX] = r->start; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) + goto out2; + wd->dmaids[PCM_RX] = r->start; + /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) */ @@ -330,15 +349,11 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, wd); ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); - if (ret) - goto out1; - - /* finally add the DMA device for this PSC */ - wd->dmapd = au1xpsc_pcm_add(pdev); - if (wd->dmapd) + if (!ret) return 0; - snd_soc_unregister_dai(&pdev->dev); +out2: + iounmap(wd->mmio); out1: release_mem_region(r->start, resource_size(r)); out0: @@ -351,9 +366,6 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (wd->dmapd) - au1xpsc_pcm_destroy(wd->dmapd); - snd_soc_unregister_dai(&pdev->dev); au_writel(0, I2S_CFG(wd)); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index c59b9e544e72..1b21c4ffae12 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -19,10 +19,6 @@ #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); - struct au1xpsc_audio_data { void __iomem *mmio; @@ -34,7 +30,6 @@ struct au1xpsc_audio_data { unsigned long pm[2]; struct mutex lock; int dmaids[2]; - struct platform_device *dmapd; }; /* easy access macros */ -- cgit v1.2.3 From adbc7a5a61ee9225e0b80d3f5719e05a88db2b4c Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Mon, 25 Jul 2011 13:45:03 +0200 Subject: ASoC: au1x: update db1200 machine to the new way of things The use of the "soc-audio" platform device is no longer en vogue, update the code to the newer, simpler way of doing things. Signed-off-by: Manuel Lauss Acked-by: Ralf Baechle Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/au1x/db1200.c | 64 +++++++++++++++++++++++++++++++++---------------- 1 file changed, 43 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 1d3e258c9ea8..289312c14b99 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -1,7 +1,7 @@ /* * DB1200 ASoC audio fabric support code. * - * (c) 2008-9 Manuel Lauss + * (c) 2008-2011 Manuel Lauss * */ @@ -21,6 +21,17 @@ #include "../codecs/wm8731.h" #include "psc.h" +static struct platform_device_id db1200_pids[] = { + { + .name = "db1200-ac97", + .driver_data = 0, + }, { + .name = "db1200-i2s", + .driver_data = 1, + }, + {}, +}; + /*------------------------- AC97 PART ---------------------------*/ static struct snd_soc_dai_link db1200_ac97_dai = { @@ -89,36 +100,47 @@ static struct snd_soc_card db1200_i2s_machine = { /*------------------------- COMMON PART ---------------------------*/ -static struct platform_device *db1200_asoc_dev; +static struct snd_soc_card *db1200_cards[] __devinitdata = { + &db1200_ac97_machine, + &db1200_i2s_machine, +}; -static int __init db1200_audio_load(void) +static int __devinit db1200_audio_probe(struct platform_device *pdev) { - int ret; + const struct platform_device_id *pid = platform_get_device_id(pdev); + struct snd_soc_card *card; - ret = -ENOMEM; - db1200_asoc_dev = platform_device_alloc("soc-audio", 1); /* PSC1 */ - if (!db1200_asoc_dev) - goto out; + card = db1200_cards[pid->driver_data]; + card->dev = &pdev->dev; + return snd_soc_register_card(card); +} - /* DB1200 board setup set PSC1MUX to preferred audio device */ - if (bcsr_read(BCSR_RESETS) & BCSR_RESETS_PSC1MUX) - platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_machine); - else - platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_machine); +static int __devexit db1200_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + snd_soc_unregister_card(card); + return 0; +} - ret = platform_device_add(db1200_asoc_dev); +static struct platform_driver db1200_audio_driver = { + .driver = { + .name = "db1200-ac97", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .id_table = db1200_pids, + .probe = db1200_audio_probe, + .remove = __devexit_p(db1200_audio_remove), +}; - if (ret) { - platform_device_put(db1200_asoc_dev); - db1200_asoc_dev = NULL; - } -out: - return ret; +static int __init db1200_audio_load(void) +{ + return platform_driver_register(&db1200_audio_driver); } static void __exit db1200_audio_unload(void) { - platform_device_unregister(db1200_asoc_dev); + platform_driver_unregister(&db1200_audio_driver); } module_init(db1200_audio_load); -- cgit v1.2.3 From 25942fdc824a709c48190356ed058ef7be19fb6a Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Mon, 25 Jul 2011 13:45:04 +0200 Subject: ASoC: au1x: use substream stream info directly PCM_TX/RX are the same as SNDRV_PCM_STREAM_PLAYBACK/CAPTURE. Use them directly. Signed-off-by: Manuel Lauss Acked-by: Ralf Baechle Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/au1x/dbdma2.c | 10 +++++----- sound/soc/au1x/psc-ac97.c | 18 +++++++++--------- sound/soc/au1x/psc-i2s.c | 14 +++++++------- sound/soc/au1x/psc.h | 6 ------ 4 files changed, 21 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index fd5378f7dece..d7d04e26eee5 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -169,7 +169,7 @@ static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd, au1x_pcm_dbdma_free(pcd); - if (stype == PCM_RX) + if (stype == SNDRV_PCM_STREAM_CAPTURE) pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id, DSCR_CMD0_ALWAYS, au1x_pcm_dmarx_cb, (void *)pcd); @@ -198,7 +198,7 @@ static inline struct au1xpsc_audio_dmadata *to_dmadata(struct snd_pcm_substream struct snd_soc_pcm_runtime *rtd = ss->private_data; struct au1xpsc_audio_dmadata *pcd = snd_soc_platform_get_drvdata(rtd->platform); - return &pcd[SUBSTREAM_TYPE(ss)]; + return &pcd[ss->stream]; } static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, @@ -212,7 +212,7 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, if (ret < 0) goto out; - stype = SUBSTREAM_TYPE(substream); + stype = substream->stream; pcd = to_dmadata(substream); DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " @@ -255,7 +255,7 @@ static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) au1xxx_dbdma_reset(pcd->ddma_chan); - if (SUBSTREAM_TYPE(substream) == PCM_RX) { + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { au1x_pcm_queue_rx(pcd); au1x_pcm_queue_rx(pcd); } else { @@ -295,7 +295,7 @@ static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) { struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); struct snd_soc_pcm_runtime *rtd = substream->private_data; - int stype = SUBSTREAM_TYPE(substream), *dmaids; + int stype = substream->stream, *dmaids; dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if (!dmaids) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 44296abfc38f..172eefd38b2d 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -41,14 +41,14 @@ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE) #define AC97PCR_START(stype) \ - ((stype) == PCM_TX ? PSC_AC97PCR_TS : PSC_AC97PCR_RS) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TS : PSC_AC97PCR_RS) #define AC97PCR_STOP(stype) \ - ((stype) == PCM_TX ? PSC_AC97PCR_TP : PSC_AC97PCR_RP) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TP : PSC_AC97PCR_RP) #define AC97PCR_CLRFIFO(stype) \ - ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) #define AC97STAT_BUSY(stype) \ - ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; @@ -215,7 +215,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, { struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); unsigned long r, ro, stat; - int chans, t, stype = SUBSTREAM_TYPE(substream); + int chans, t, stype = substream->stream; chans = params_channels(params); @@ -235,7 +235,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_SET_LEN(params->msbits); /* channels: enable slots for front L/R channel */ - if (stype == PCM_TX) { + if (stype == SNDRV_PCM_STREAM_PLAYBACK) { r &= ~PSC_AC97CFG_TXSLOT_MASK; r |= PSC_AC97CFG_TXSLOT_ENA(3); r |= PSC_AC97CFG_TXSLOT_ENA(4); @@ -294,7 +294,7 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); - int ret, stype = SUBSTREAM_TYPE(substream); + int ret, stype = substream->stream; ret = 0; @@ -391,12 +391,12 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) r = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!r) goto out2; - wd->dmaids[PCM_TX] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; r = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!r) goto out2; - wd->dmaids[PCM_RX] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; /* configuration: max dma trigger threshold, enable ac97 */ wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 1b7ab5d422e0..7c5ae920544f 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -42,13 +42,13 @@ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) #define I2SSTAT_BUSY(stype) \ - ((stype) == PCM_TX ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB) #define I2SPCR_START(stype) \ - ((stype) == PCM_TX ? PSC_I2SPCR_TS : PSC_I2SPCR_RS) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TS : PSC_I2SPCR_RS) #define I2SPCR_STOP(stype) \ - ((stype) == PCM_TX ? PSC_I2SPCR_TP : PSC_I2SPCR_RP) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TP : PSC_I2SPCR_RP) #define I2SPCR_CLRFIFO(stype) \ - ((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TC : PSC_I2SPCR_RC) static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, @@ -240,7 +240,7 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); - int ret, stype = SUBSTREAM_TYPE(substream); + int ret, stype = substream->stream; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -316,12 +316,12 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) r = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!r) goto out2; - wd->dmaids[PCM_TX] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; r = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!r) goto out2; - wd->dmaids[PCM_RX] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 1b21c4ffae12..b16b2e02e0c9 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -13,12 +13,6 @@ #ifndef _AU1X_PCM_H #define _AU1X_PCM_H -#define PCM_TX 0 -#define PCM_RX 1 - -#define SUBSTREAM_TYPE(substream) \ - ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX) - struct au1xpsc_audio_data { void __iomem *mmio; -- cgit v1.2.3 From ed6e1d04c106f69882c055a72a63111ed9dadc01 Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Thu, 21 Jul 2011 12:36:55 +0800 Subject: ASoC: mxs: add mxs-pcm driver Signed-off-by: Dong Aisheng Acked-by: Liam Girdwood Tested-by: Wolfram Sang Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-pcm.c | 359 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/mxs/mxs-pcm.h | 43 ++++++ 2 files changed, 402 insertions(+) create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h (limited to 'sound') diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c new file mode 100644 index 000000000000..dea5aa4aa647 --- /dev/null +++ b/sound/soc/mxs/mxs-pcm.c @@ -0,0 +1,359 @@ +/* + * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * Based on sound/soc/imx/imx-pcm-dma-mx2.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include "mxs-pcm.h" + +static struct snd_pcm_hardware snd_mxs_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 2, + .channels_max = 2, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 52, + .buffer_bytes_max = 64 * 1024, + .fifo_size = 32, + +}; + +static void audio_dma_irq(void *data) +{ + struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + + iprtd->offset += iprtd->period_bytes; + iprtd->offset %= iprtd->period_bytes * iprtd->periods; + snd_pcm_period_elapsed(substream); +} + +static bool filter(struct dma_chan *chan, void *param) +{ + struct mxs_pcm_runtime_data *iprtd = param; + struct mxs_pcm_dma_params *dma_params = iprtd->dma_params; + + if (!mxs_dma_is_apbx(chan)) + return false; + + if (chan->chan_id != dma_params->chan_num) + return false; + + chan->private = &iprtd->dma_data; + + return true; +} + +static int mxs_dma_alloc(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + dma_cap_mask_t mask; + + iprtd->dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + iprtd->dma_data.chan_irq = iprtd->dma_params->chan_irq; + iprtd->dma_chan = dma_request_channel(mask, filter, iprtd); + if (!iprtd->dma_chan) + return -EINVAL; + + return 0; +} + +static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + unsigned long dma_addr; + struct dma_chan *chan; + int ret; + + ret = mxs_dma_alloc(substream, params); + if (ret) + return ret; + chan = iprtd->dma_chan; + + iprtd->size = params_buffer_bytes(params); + iprtd->periods = params_periods(params); + iprtd->period_bytes = params_period_bytes(params); + iprtd->offset = 0; + iprtd->period_time = HZ / (params_rate(params) / + params_period_size(params)); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + dma_addr = runtime->dma_addr; + + iprtd->buf = substream->dma_buffer.area; + + iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr, + iprtd->period_bytes * iprtd->periods, + iprtd->period_bytes, + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_TO_DEVICE : DMA_FROM_DEVICE); + if (!iprtd->desc) { + dev_err(&chan->dev->device, "cannot prepare slave dma\n"); + return -EINVAL; + } + + iprtd->desc->callback = audio_dma_irq; + iprtd->desc->callback_param = substream; + + return 0; +} + +static int snd_mxs_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + + if (iprtd->dma_chan) { + dma_release_channel(iprtd->dma_chan); + iprtd->dma_chan = NULL; + } + + return 0; +} + +static int snd_mxs_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dmaengine_submit(iprtd->desc); + + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dmaengine_terminate_all(iprtd->dma_chan); + + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_mxs_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + + return bytes_to_frames(substream->runtime, iprtd->offset); +} + +static int snd_mxs_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd; + int ret; + + iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + if (iprtd == NULL) + return -ENOMEM; + runtime->private_data = iprtd; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + kfree(iprtd); + return ret; + } + + snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware); + + return 0; +} + +static int snd_mxs_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + + kfree(iprtd); + + return 0; +} + +static int snd_mxs_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops mxs_pcm_ops = { + .open = snd_mxs_open, + .close = snd_mxs_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mxs_pcm_hw_params, + .hw_free = snd_mxs_pcm_hw_free, + .trigger = snd_mxs_pcm_trigger, + .pointer = snd_mxs_pcm_pointer, + .mmap = snd_mxs_pcm_mmap, +}; + +static int mxs_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = snd_mxs_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + + return 0; +} + +static u64 mxs_pcm_dmamask = DMA_BIT_MASK(32); +static int mxs_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &mxs_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = mxs_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = mxs_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + +out: + return ret; +} + +static void mxs_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static struct snd_soc_platform_driver mxs_soc_platform = { + .ops = &mxs_pcm_ops, + .pcm_new = mxs_pcm_new, + .pcm_free = mxs_pcm_free, +}; + +static int __devinit mxs_soc_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &mxs_soc_platform); +} + +static int __devexit mxs_soc_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver mxs_pcm_driver = { + .driver = { + .name = "mxs-pcm-audio", + .owner = THIS_MODULE, + }, + .probe = mxs_soc_platform_probe, + .remove = __devexit_p(mxs_soc_platform_remove), +}; + +static int __init snd_mxs_pcm_init(void) +{ + return platform_driver_register(&mxs_pcm_driver); +} +module_init(snd_mxs_pcm_init); + +static void __exit snd_mxs_pcm_exit(void) +{ + platform_driver_unregister(&mxs_pcm_driver); +} +module_exit(snd_mxs_pcm_exit); diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h new file mode 100644 index 000000000000..f55ac4f7a76a --- /dev/null +++ b/sound/soc/mxs/mxs-pcm.h @@ -0,0 +1,43 @@ +/* + * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef _MXS_PCM_H +#define _MXS_PCM_H + +#include + +struct mxs_pcm_dma_params { + int chan_irq; + int chan_num; +}; + +struct mxs_pcm_runtime_data { + int period_bytes; + int periods; + int dma; + unsigned long offset; + unsigned long size; + void *buf; + int period_time; + struct dma_async_tx_descriptor *desc; + struct dma_chan *dma_chan; + struct mxs_dma_data dma_data; + struct mxs_pcm_dma_params *dma_params; +}; + +#endif -- cgit v1.2.3 From 2a24f2ce89b6157192c10616492be8a981b0cce8 Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Thu, 21 Jul 2011 12:36:56 +0800 Subject: ASoC: mxs: add mxs-saif driver Signed-off-by: Dong Aisheng Acked-by: Liam Girdwood Tested-by: Wolfram Sang Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 677 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 130 +++++++++ 2 files changed, 807 insertions(+) create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c new file mode 100644 index 000000000000..0b3adaec9f4c --- /dev/null +++ b/sound/soc/mxs/mxs-saif.c @@ -0,0 +1,677 @@ +/* + * Copyright 2011 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "mxs-saif.h" + +static struct mxs_saif *mxs_saif[2]; + +static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + + switch (clk_id) { + case MXS_SAIF_MCLK: + saif->mclk = freq; + break; + default: + return -EINVAL; + } + return 0; +} + +/* + * Set SAIF clock and MCLK + */ +static int mxs_saif_set_clk(struct mxs_saif *saif, + unsigned int mclk, + unsigned int rate) +{ + u32 scr; + int ret; + + scr = __raw_readl(saif->base + SAIF_CTRL); + scr &= ~BM_SAIF_CTRL_BITCLK_MULT_RATE; + scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; + + /* + * Set SAIF clock + * + * The SAIF clock should be either 384*fs or 512*fs. + * If MCLK is used, the SAIF clk ratio need to match mclk ratio. + * For 32x mclk, set saif clk as 512*fs. + * For 48x mclk, set saif clk as 384*fs. + * + * If MCLK is not used, we just set saif clk to 512*fs. + */ + if (saif->mclk_in_use) { + if (mclk % 32 == 0) { + scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; + ret = clk_set_rate(saif->clk, 512 * rate); + } else if (mclk % 48 == 0) { + scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE; + ret = clk_set_rate(saif->clk, 384 * rate); + } else { + /* SAIF MCLK should be either 32x or 48x */ + return -EINVAL; + } + } else { + ret = clk_set_rate(saif->clk, 512 * rate); + scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; + } + + if (ret) + return ret; + + if (!saif->mclk_in_use) { + __raw_writel(scr, saif->base + SAIF_CTRL); + return 0; + } + + /* + * Program the over-sample rate for MCLK output + * + * The available MCLK range is 32x, 48x... 512x. The rate + * could be from 8kHz to 192kH. + */ + switch (mclk / rate) { + case 32: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(4); + break; + case 64: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(3); + break; + case 128: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(2); + break; + case 256: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(1); + break; + case 512: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(0); + break; + case 48: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(3); + break; + case 96: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(2); + break; + case 192: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(1); + break; + case 384: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(0); + break; + default: + return -EINVAL; + } + + __raw_writel(scr, saif->base + SAIF_CTRL); + + return 0; +} + +/* + * Put and disable MCLK. + */ +int mxs_saif_put_mclk(unsigned int saif_id) +{ + struct mxs_saif *saif = mxs_saif[saif_id]; + u32 stat; + + if (!saif) + return -EINVAL; + + stat = __raw_readl(saif->base + SAIF_STAT); + if (stat & BM_SAIF_STAT_BUSY) { + dev_err(saif->dev, "error: busy\n"); + return -EBUSY; + } + + clk_disable(saif->clk); + + /* disable MCLK output */ + __raw_writel(BM_SAIF_CTRL_CLKGATE, + saif->base + SAIF_CTRL + MXS_SET_ADDR); + __raw_writel(BM_SAIF_CTRL_RUN, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + + saif->mclk_in_use = 0; + return 0; +} + +/* + * Get MCLK and set clock rate, then enable it + * + * This interface is used for codecs who are using MCLK provided + * by saif. + */ +int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, + unsigned int rate) +{ + struct mxs_saif *saif = mxs_saif[saif_id]; + u32 stat; + int ret; + + if (!saif) + return -EINVAL; + + stat = __raw_readl(saif->base + SAIF_STAT); + if (stat & BM_SAIF_STAT_BUSY) { + dev_err(saif->dev, "error: busy\n"); + return -EBUSY; + } + + /* Clear Reset */ + __raw_writel(BM_SAIF_CTRL_SFTRST, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + + saif->mclk_in_use = 1; + ret = mxs_saif_set_clk(saif, mclk, rate); + if (ret) + return ret; + + ret = clk_enable(saif->clk); + if (ret) + return ret; + + /* enable MCLK output */ + __raw_writel(BM_SAIF_CTRL_CLKGATE, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + __raw_writel(BM_SAIF_CTRL_RUN, + saif->base + SAIF_CTRL + MXS_SET_ADDR); + + return 0; +} + +/* + * SAIF DAI format configuration. + * Should only be called when port is inactive. + */ +static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + u32 scr, stat; + u32 scr0; + struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + + stat = __raw_readl(saif->base + SAIF_STAT); + if (stat & BM_SAIF_STAT_BUSY) { + dev_err(cpu_dai->dev, "error: busy\n"); + return -EBUSY; + } + + scr0 = __raw_readl(saif->base + SAIF_CTRL); + scr0 = scr0 & ~BM_SAIF_CTRL_BITCLK_EDGE & ~BM_SAIF_CTRL_LRCLK_POLARITY \ + & ~BM_SAIF_CTRL_JUSTIFY & ~BM_SAIF_CTRL_DELAY; + scr = 0; + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* data frame low 1clk before data */ + scr |= BM_SAIF_CTRL_DELAY; + scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* data frame high with data */ + scr &= ~BM_SAIF_CTRL_DELAY; + scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY; + scr &= ~BM_SAIF_CTRL_JUSTIFY; + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + scr |= BM_SAIF_CTRL_BITCLK_EDGE; + scr |= BM_SAIF_CTRL_LRCLK_POLARITY; + break; + case SND_SOC_DAIFMT_IB_NF: + scr |= BM_SAIF_CTRL_BITCLK_EDGE; + scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY; + break; + case SND_SOC_DAIFMT_NB_IF: + scr &= ~BM_SAIF_CTRL_BITCLK_EDGE; + scr |= BM_SAIF_CTRL_LRCLK_POLARITY; + break; + case SND_SOC_DAIFMT_NB_NF: + scr &= ~BM_SAIF_CTRL_BITCLK_EDGE; + scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY; + break; + } + + /* + * Note: We simply just support master mode since SAIF TX can only + * work as master. + */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + __raw_writel(scr | scr0, saif->base + SAIF_CTRL); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int mxs_saif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + snd_soc_dai_set_dma_data(cpu_dai, substream, &saif->dma_param); + + /* clear error status to 0 for each re-open */ + saif->fifo_underrun = 0; + saif->fifo_overrun = 0; + + /* Clear Reset for normal operations */ + __raw_writel(BM_SAIF_CTRL_SFTRST, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + + return 0; +} + +/* + * Should only be called when port is inactive. + * although can be called multiple times by upper layers. + */ +static int mxs_saif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + u32 scr, stat; + int ret; + + /* mclk should already be set */ + if (!saif->mclk && saif->mclk_in_use) { + dev_err(cpu_dai->dev, "set mclk first\n"); + return -EINVAL; + } + + stat = __raw_readl(saif->base + SAIF_STAT); + if (stat & BM_SAIF_STAT_BUSY) { + dev_err(cpu_dai->dev, "error: busy\n"); + return -EBUSY; + } + + /* + * Set saif clk based on sample rate. + * If mclk is used, we also set mclk, if not, saif->mclk is + * default 0, means not used. + */ + ret = mxs_saif_set_clk(saif, saif->mclk, params_rate(params)); + if (ret) { + dev_err(cpu_dai->dev, "unable to get proper clk\n"); + return ret; + } + + scr = __raw_readl(saif->base + SAIF_CTRL); + + scr &= ~BM_SAIF_CTRL_WORD_LENGTH; + scr &= ~BM_SAIF_CTRL_BITCLK_48XFS_ENABLE; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + scr |= BF_SAIF_CTRL_WORD_LENGTH(0); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + scr |= BF_SAIF_CTRL_WORD_LENGTH(4); + scr |= BM_SAIF_CTRL_BITCLK_48XFS_ENABLE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + scr |= BF_SAIF_CTRL_WORD_LENGTH(8); + scr |= BM_SAIF_CTRL_BITCLK_48XFS_ENABLE; + break; + default: + return -EINVAL; + } + + /* Tx/Rx config */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* enable TX mode */ + scr &= ~BM_SAIF_CTRL_READ_MODE; + } else { + /* enable RX mode */ + scr |= BM_SAIF_CTRL_READ_MODE; + } + + __raw_writel(scr, saif->base + SAIF_CTRL); + return 0; +} + +static int mxs_saif_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + + /* clear clock gate */ + __raw_writel(BM_SAIF_CTRL_CLKGATE, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + + /* enable FIFO error irqs */ + __raw_writel(BM_SAIF_CTRL_FIFO_ERROR_IRQ_EN, + saif->base + SAIF_CTRL + MXS_SET_ADDR); + + return 0; +} + +static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *cpu_dai) +{ + struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dev_dbg(cpu_dai->dev, "start\n"); + + clk_enable(saif->clk); + if (!saif->mclk_in_use) + __raw_writel(BM_SAIF_CTRL_RUN, + saif->base + SAIF_CTRL + MXS_SET_ADDR); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* + * write a data to saif data register to trigger + * the transfer + */ + __raw_writel(0, saif->base + SAIF_DATA); + } else { + /* + * read a data from saif data register to trigger + * the receive + */ + __raw_readl(saif->base + SAIF_DATA); + } + + dev_dbg(cpu_dai->dev, "CTRL 0x%x STAT 0x%x\n", + __raw_readl(saif->base + SAIF_CTRL), + __raw_readl(saif->base + SAIF_STAT)); + + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dev_dbg(cpu_dai->dev, "stop\n"); + + clk_disable(saif->clk); + if (!saif->mclk_in_use) + __raw_writel(BM_SAIF_CTRL_RUN, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + + break; + default: + return -EINVAL; + } + + return 0; +} + +#define MXS_SAIF_RATES SNDRV_PCM_RATE_8000_192000 +#define MXS_SAIF_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops mxs_saif_dai_ops = { + .startup = mxs_saif_startup, + .trigger = mxs_saif_trigger, + .prepare = mxs_saif_prepare, + .hw_params = mxs_saif_hw_params, + .set_sysclk = mxs_saif_set_dai_sysclk, + .set_fmt = mxs_saif_set_dai_fmt, +}; + +static int mxs_saif_dai_probe(struct snd_soc_dai *dai) +{ + struct mxs_saif *saif = dev_get_drvdata(dai->dev); + + snd_soc_dai_set_drvdata(dai, saif); + + return 0; +} + +static struct snd_soc_dai_driver mxs_saif_dai = { + .name = "mxs-saif", + .probe = mxs_saif_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = MXS_SAIF_RATES, + .formats = MXS_SAIF_FORMATS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = MXS_SAIF_RATES, + .formats = MXS_SAIF_FORMATS, + }, + .ops = &mxs_saif_dai_ops, +}; + +static irqreturn_t mxs_saif_irq(int irq, void *dev_id) +{ + struct mxs_saif *saif = dev_id; + unsigned int stat; + + stat = __raw_readl(saif->base + SAIF_STAT); + if (!(stat & (BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ | + BM_SAIF_STAT_FIFO_OVERFLOW_IRQ))) + return IRQ_NONE; + + if (stat & BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ) { + dev_dbg(saif->dev, "underrun!!! %d\n", ++saif->fifo_underrun); + __raw_writel(BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ, + saif->base + SAIF_STAT + MXS_CLR_ADDR); + } + + if (stat & BM_SAIF_STAT_FIFO_OVERFLOW_IRQ) { + dev_dbg(saif->dev, "overrun!!! %d\n", ++saif->fifo_overrun); + __raw_writel(BM_SAIF_STAT_FIFO_OVERFLOW_IRQ, + saif->base + SAIF_STAT + MXS_CLR_ADDR); + } + + dev_dbg(saif->dev, "SAIF_CTRL %x SAIF_STAT %x\n", + __raw_readl(saif->base + SAIF_CTRL), + __raw_readl(saif->base + SAIF_STAT)); + + return IRQ_HANDLED; +} + +static int mxs_saif_probe(struct platform_device *pdev) +{ + struct resource *res; + struct mxs_saif *saif; + int ret = 0; + + saif = kzalloc(sizeof(*saif), GFP_KERNEL); + if (!saif) + return -ENOMEM; + + if (pdev->id >= ARRAY_SIZE(mxs_saif)) + return -EINVAL; + mxs_saif[pdev->id] = saif; + + saif->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(saif->clk)) { + ret = PTR_ERR(saif->clk); + dev_err(&pdev->dev, "Cannot get the clock: %d\n", + ret); + goto failed_clk; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + ret = -ENODEV; + dev_err(&pdev->dev, "failed to get io resource: %d\n", + ret); + goto failed_get_resource; + } + + if (!request_mem_region(res->start, resource_size(res), "mxs-saif")) { + dev_err(&pdev->dev, "request_mem_region failed\n"); + ret = -EBUSY; + goto failed_get_resource; + } + + saif->base = ioremap(res->start, resource_size(res)); + if (!saif->base) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENODEV; + goto failed_ioremap; + } + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + ret = -ENODEV; + dev_err(&pdev->dev, "failed to get dma resource: %d\n", + ret); + goto failed_ioremap; + } + saif->dma_param.chan_num = res->start; + + saif->irq = platform_get_irq(pdev, 0); + if (saif->irq < 0) { + ret = saif->irq; + dev_err(&pdev->dev, "failed to get irq resource: %d\n", + ret); + goto failed_get_irq1; + } + + saif->dev = &pdev->dev; + ret = request_irq(saif->irq, mxs_saif_irq, 0, "mxs-saif", saif); + if (ret) { + dev_err(&pdev->dev, "failed to request irq\n"); + goto failed_get_irq1; + } + + saif->dma_param.chan_irq = platform_get_irq(pdev, 1); + if (saif->dma_param.chan_irq < 0) { + ret = saif->dma_param.chan_irq; + dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", + ret); + goto failed_get_irq2; + } + + platform_set_drvdata(pdev, saif); + + ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai); + if (ret) { + dev_err(&pdev->dev, "register DAI failed\n"); + goto failed_register; + } + + saif->soc_platform_pdev = platform_device_alloc( + "mxs-pcm-audio", pdev->id); + if (!saif->soc_platform_pdev) { + ret = -ENOMEM; + goto failed_pdev_alloc; + } + + platform_set_drvdata(saif->soc_platform_pdev, saif); + ret = platform_device_add(saif->soc_platform_pdev); + if (ret) { + dev_err(&pdev->dev, "failed to add soc platform device\n"); + goto failed_pdev_add; + } + + return 0; + +failed_pdev_add: + platform_device_put(saif->soc_platform_pdev); +failed_pdev_alloc: + snd_soc_unregister_dai(&pdev->dev); +failed_register: +failed_get_irq2: + free_irq(saif->irq, saif); +failed_get_irq1: + iounmap(saif->base); +failed_ioremap: + release_mem_region(res->start, resource_size(res)); +failed_get_resource: + clk_put(saif->clk); +failed_clk: + kfree(saif); + + return ret; +} + +static int __devexit mxs_saif_remove(struct platform_device *pdev) +{ + struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + struct mxs_saif *saif = platform_get_drvdata(pdev); + + platform_device_unregister(saif->soc_platform_pdev); + + snd_soc_unregister_dai(&pdev->dev); + + iounmap(saif->base); + release_mem_region(res->start, resource_size(res)); + free_irq(saif->irq, saif); + + clk_put(saif->clk); + kfree(saif); + + return 0; +} + +static struct platform_driver mxs_saif_driver = { + .probe = mxs_saif_probe, + .remove = __devexit_p(mxs_saif_remove), + + .driver = { + .name = "mxs-saif", + .owner = THIS_MODULE, + }, +}; + +static int __init mxs_saif_init(void) +{ + return platform_driver_register(&mxs_saif_driver); +} + +static void __exit mxs_saif_exit(void) +{ + platform_driver_unregister(&mxs_saif_driver); +} + +module_init(mxs_saif_init); +module_exit(mxs_saif_exit); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("MXS ASoC SAIF driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h new file mode 100644 index 000000000000..0e2ff8cdbfee --- /dev/null +++ b/sound/soc/mxs/mxs-saif.h @@ -0,0 +1,130 @@ +/* + * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + + +#ifndef _MXS_SAIF_H +#define _MXS_SAIF_H + +#define SAIF_CTRL 0x0 +#define SAIF_STAT 0x10 +#define SAIF_DATA 0x20 +#define SAIF_VERSION 0X30 + +/* SAIF_CTRL */ +#define BM_SAIF_CTRL_SFTRST 0x80000000 +#define BM_SAIF_CTRL_CLKGATE 0x40000000 +#define BP_SAIF_CTRL_BITCLK_MULT_RATE 27 +#define BM_SAIF_CTRL_BITCLK_MULT_RATE 0x38000000 +#define BF_SAIF_CTRL_BITCLK_MULT_RATE(v) \ + (((v) << 27) & BM_SAIF_CTRL_BITCLK_MULT_RATE) +#define BM_SAIF_CTRL_BITCLK_BASE_RATE 0x04000000 +#define BM_SAIF_CTRL_FIFO_ERROR_IRQ_EN 0x02000000 +#define BM_SAIF_CTRL_FIFO_SERVICE_IRQ_EN 0x01000000 +#define BP_SAIF_CTRL_RSRVD2 21 +#define BM_SAIF_CTRL_RSRVD2 0x00E00000 + +#define BP_SAIF_CTRL_DMAWAIT_COUNT 16 +#define BM_SAIF_CTRL_DMAWAIT_COUNT 0x001F0000 +#define BF_SAIF_CTRL_DMAWAIT_COUNT(v) \ + (((v) << 16) & BM_SAIF_CTRL_DMAWAIT_COUNT) +#define BP_SAIF_CTRL_CHANNEL_NUM_SELECT 14 +#define BM_SAIF_CTRL_CHANNEL_NUM_SELECT 0x0000C000 +#define BF_SAIF_CTRL_CHANNEL_NUM_SELECT(v) \ + (((v) << 14) & BM_SAIF_CTRL_CHANNEL_NUM_SELECT) +#define BM_SAIF_CTRL_LRCLK_PULSE 0x00002000 +#define BM_SAIF_CTRL_BIT_ORDER 0x00001000 +#define BM_SAIF_CTRL_DELAY 0x00000800 +#define BM_SAIF_CTRL_JUSTIFY 0x00000400 +#define BM_SAIF_CTRL_LRCLK_POLARITY 0x00000200 +#define BM_SAIF_CTRL_BITCLK_EDGE 0x00000100 +#define BP_SAIF_CTRL_WORD_LENGTH 4 +#define BM_SAIF_CTRL_WORD_LENGTH 0x000000F0 +#define BF_SAIF_CTRL_WORD_LENGTH(v) \ + (((v) << 4) & BM_SAIF_CTRL_WORD_LENGTH) +#define BM_SAIF_CTRL_BITCLK_48XFS_ENABLE 0x00000008 +#define BM_SAIF_CTRL_SLAVE_MODE 0x00000004 +#define BM_SAIF_CTRL_READ_MODE 0x00000002 +#define BM_SAIF_CTRL_RUN 0x00000001 + +/* SAIF_STAT */ +#define BM_SAIF_STAT_PRESENT 0x80000000 +#define BP_SAIF_STAT_RSRVD2 17 +#define BM_SAIF_STAT_RSRVD2 0x7FFE0000 +#define BF_SAIF_STAT_RSRVD2(v) \ + (((v) << 17) & BM_SAIF_STAT_RSRVD2) +#define BM_SAIF_STAT_DMA_PREQ 0x00010000 +#define BP_SAIF_STAT_RSRVD1 7 +#define BM_SAIF_STAT_RSRVD1 0x0000FF80 +#define BF_SAIF_STAT_RSRVD1(v) \ + (((v) << 7) & BM_SAIF_STAT_RSRVD1) + +#define BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ 0x00000040 +#define BM_SAIF_STAT_FIFO_OVERFLOW_IRQ 0x00000020 +#define BM_SAIF_STAT_FIFO_SERVICE_IRQ 0x00000010 +#define BP_SAIF_STAT_RSRVD0 1 +#define BM_SAIF_STAT_RSRVD0 0x0000000E +#define BF_SAIF_STAT_RSRVD0(v) \ + (((v) << 1) & BM_SAIF_STAT_RSRVD0) +#define BM_SAIF_STAT_BUSY 0x00000001 + +/* SAFI_DATA */ +#define BP_SAIF_DATA_PCM_RIGHT 16 +#define BM_SAIF_DATA_PCM_RIGHT 0xFFFF0000 +#define BF_SAIF_DATA_PCM_RIGHT(v) \ + (((v) << 16) & BM_SAIF_DATA_PCM_RIGHT) +#define BP_SAIF_DATA_PCM_LEFT 0 +#define BM_SAIF_DATA_PCM_LEFT 0x0000FFFF +#define BF_SAIF_DATA_PCM_LEFT(v) \ + (((v) << 0) & BM_SAIF_DATA_PCM_LEFT) + +/* SAIF_VERSION */ +#define BP_SAIF_VERSION_MAJOR 24 +#define BM_SAIF_VERSION_MAJOR 0xFF000000 +#define BF_SAIF_VERSION_MAJOR(v) \ + (((v) << 24) & BM_SAIF_VERSION_MAJOR) +#define BP_SAIF_VERSION_MINOR 16 +#define BM_SAIF_VERSION_MINOR 0x00FF0000 +#define BF_SAIF_VERSION_MINOR(v) \ + (((v) << 16) & BM_SAIF_VERSION_MINOR) +#define BP_SAIF_VERSION_STEP 0 +#define BM_SAIF_VERSION_STEP 0x0000FFFF +#define BF_SAIF_VERSION_STEP(v) \ + (((v) << 0) & BM_SAIF_VERSION_STEP) + +#define MXS_SAIF_MCLK 0 + +#include "mxs-pcm.h" + +struct mxs_saif { + struct device *dev; + struct clk *clk; + unsigned int mclk; + unsigned int mclk_in_use; + void __iomem *base; + int irq; + struct mxs_pcm_dma_params dma_param; + + struct platform_device *soc_platform_pdev; + u32 fifo_underrun; + u32 fifo_overrun; +}; + +extern int mxs_saif_put_mclk(unsigned int saif_id); +extern int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, + unsigned int rate); +#endif -- cgit v1.2.3 From fcb5e47eff29a10e9cbc55a7660746695e303671 Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Thu, 21 Jul 2011 12:36:57 +0800 Subject: ASoC: mxs: add mxs-sgtl5000 machine driver The driver only supports playback firstly. For recording, as we have to use two saif instances to implement full duplex (playback & recording) due to hardware limitation, we need to figure out a good design to fit in ASoC. Signed-off-by: Dong Aisheng Acked-by: Liam Girdwood Tested-by: Wolfram Sang Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-sgtl5000.c | 165 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 165 insertions(+) create mode 100644 sound/soc/mxs/mxs-sgtl5000.c (limited to 'sound') diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c new file mode 100644 index 000000000000..a0d89c93df0f --- /dev/null +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -0,0 +1,165 @@ +/* + * Copyright 2011 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/sgtl5000.h" +#include "mxs-saif.h" + +static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int rate = params_rate(params); + u32 dai_format, mclk; + int ret; + + /* sgtl5000 does not support 512*rate when in 96000 fs */ + switch (rate) { + case 96000: + mclk = 256 * rate; + break; + default: + mclk = 512 * rate; + break; + } + + /* Sgtl5000 sysclk should be >= 8MHz and <= 27M */ + if (mclk < 8000000 || mclk > 27000000) + return -EINVAL; + + /* Set SGTL5000's SYSCLK (provided by SAIF MCLK) */ + ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk, 0); + if (ret) + return ret; + + /* The SAIF MCLK should be the same as SGTL5000_SYSCLK */ + ret = snd_soc_dai_set_sysclk(cpu_dai, MXS_SAIF_MCLK, mclk, 0); + if (ret) + return ret; + + /* set codec to slave mode */ + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, dai_format); + if (ret) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); + if (ret) + return ret; + + return 0; +} + +static struct snd_soc_ops mxs_sgtl5000_hifi_ops = { + .hw_params = mxs_sgtl5000_hw_params, +}; + +static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { + { + .name = "HiFi", + .stream_name = "HiFi Playback", + .codec_dai_name = "sgtl5000", + .codec_name = "sgtl5000.0-000a", + .cpu_dai_name = "mxs-saif.0", + .platform_name = "mxs-pcm-audio.0", + .ops = &mxs_sgtl5000_hifi_ops, + }, +}; + +static struct snd_soc_card mxs_sgtl5000 = { + .name = "mxs_sgtl5000", + .dai_link = mxs_sgtl5000_dai, + .num_links = ARRAY_SIZE(mxs_sgtl5000_dai), +}; + +static int __devinit mxs_sgtl5000_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &mxs_sgtl5000; + int ret; + + /* + * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w). + * The Sgtl5000 sysclk is derived from saif0 mclk and it's range + * should be >= 8MHz and <= 27M. + */ + ret = mxs_saif_get_mclk(0, 44100 * 256, 44100); + if (ret) + return ret; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + return ret; + } + + return 0; +} + +static int __devexit mxs_sgtl5000_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + mxs_saif_put_mclk(0); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver mxs_sgtl5000_audio_driver = { + .driver = { + .name = "mxs-sgtl5000", + .owner = THIS_MODULE, + }, + .probe = mxs_sgtl5000_probe, + .remove = __devexit_p(mxs_sgtl5000_remove), +}; + +static int __init mxs_sgtl5000_init(void) +{ + return platform_driver_register(&mxs_sgtl5000_audio_driver); +} +module_init(mxs_sgtl5000_init); + +static void __exit mxs_sgtl5000_exit(void) +{ + platform_driver_unregister(&mxs_sgtl5000_audio_driver); +} +module_exit(mxs_sgtl5000_exit); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("MXS ALSA SoC Machine driver"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 009ad054b71b77264157c70c58654543acc0c566 Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Thu, 21 Jul 2011 12:36:58 +0800 Subject: ASoC: mxs: add asoc configuration files Signed-off-by: Dong Aisheng Acked-by: Liam Girdwood Tested-by: Wolfram Sang Signed-off-by: Mark Brown --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mxs/Kconfig | 20 ++++++++++++++++++++ sound/soc/mxs/Makefile | 10 ++++++++++ 4 files changed, 32 insertions(+) create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile (limited to 'sound') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 8224db5f0434..47d07ce4e867 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -51,6 +51,7 @@ source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/kirkwood/Kconfig" source "sound/soc/mid-x86/Kconfig" +source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 4f913876f332..9ea8ac827adc 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -12,6 +12,7 @@ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += imx/ obj-$(CONFIG_SND_SOC) += jz4740/ obj-$(CONFIG_SND_SOC) += mid-x86/ +obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += kirkwood/ diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig new file mode 100644 index 000000000000..e4ba8d5f25fa --- /dev/null +++ b/sound/soc/mxs/Kconfig @@ -0,0 +1,20 @@ +menuconfig SND_MXS_SOC + tristate "SoC Audio for Freescale MXS CPUs" + depends on ARCH_MXS + select SND_PCM + help + Say Y or M if you want to add support for codecs attached to + the MXS SAIF interface. + + +if SND_MXS_SOC + +config SND_SOC_MXS_SGTL5000 + tristate "SoC Audio support for i.MX boards with sgtl5000" + depends on I2C + select SND_SOC_SGTL5000 + help + Say Y if you want to add support for SoC audio on an MXS board with + a sgtl5000 codec. + +endif # SND_MXS_SOC diff --git a/sound/soc/mxs/Makefile b/sound/soc/mxs/Makefile new file mode 100644 index 000000000000..565b5b51e8b7 --- /dev/null +++ b/sound/soc/mxs/Makefile @@ -0,0 +1,10 @@ +# MXS Platform Support +snd-soc-mxs-objs := mxs-saif.o +snd-soc-mxs-pcm-objs := mxs-pcm.o + +obj-$(CONFIG_SND_MXS_SOC) += snd-soc-mxs.o snd-soc-mxs-pcm.o + +# i.MX Machine Support +snd-soc-mxs-sgtl5000-objs := mxs-sgtl5000.o + +obj-$(CONFIG_SND_SOC_MXS_SGTL5000) += snd-soc-mxs-sgtl5000.o -- cgit v1.2.3 From a7f96e4dc6bb5f45d0612782419e6a63032a2ac0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 26 Jul 2011 21:00:13 +0100 Subject: ASoC: Add device tree binding for WM8731 Tested with the famous "hey, look! this compiles" test plan. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Acked by: Grant Likely --- sound/soc/codecs/wm8731.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 76b4361e9b80..f76b6fc6766a 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include #include @@ -607,6 +608,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { .num_dapm_routes = ARRAY_SIZE(wm8731_intercon), }; +static const struct of_device_id wm8731_of_match[] = { + { .compatible = "wlf,wm8731", }, + { } +}; + +MODULE_DEVICE_TABLE(of, wm8731_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8731_spi_probe(struct spi_device *spi) { @@ -638,6 +646,7 @@ static struct spi_driver wm8731_spi_driver = { .driver = { .name = "wm8731", .owner = THIS_MODULE, + .of_match_table = wm8731_of_match, }, .probe = wm8731_spi_probe, .remove = __devexit_p(wm8731_spi_remove), @@ -682,6 +691,7 @@ static struct i2c_driver wm8731_i2c_driver = { .driver = { .name = "wm8731", .owner = THIS_MODULE, + .of_match_table = wm8731_of_match, }, .probe = wm8731_i2c_probe, .remove = __devexit_p(wm8731_i2c_remove), -- cgit v1.2.3 From 58e494247a9f09f0ae8d9867fcfb672a9bcdd6ae Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 22 Jul 2011 00:28:51 +0800 Subject: ASoC: sgtl5000: add device tree probe support It adds device tree probe support for sgtl5000 driver. Signed-off-by: Shawn Guo Acked-by: Grant Likely Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 76258f2a2ffb..cf6eea8b458e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -1494,10 +1495,17 @@ static const struct i2c_device_id sgtl5000_id[] = { MODULE_DEVICE_TABLE(i2c, sgtl5000_id); +static const struct of_device_id sgtl5000_dt_ids[] = { + { .compatible = "fsl,sgtl5000", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(i2c, sgtl5000_dt_ids); + static struct i2c_driver sgtl5000_i2c_driver = { .driver = { .name = "sgtl5000", .owner = THIS_MODULE, + .of_match_table = sgtl5000_dt_ids, }, .probe = sgtl5000_i2c_probe, .remove = __devexit_p(sgtl5000_i2c_remove), -- cgit v1.2.3 From 25032c119e5f43725b624ab30e2ccb8c23b9ebd3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Aug 2011 13:52:48 +0900 Subject: ASoC: Trivial formatting fix in soc-core.c Utterly trivial but it annoys me. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9d3935bbbd0c..ae93aa81244c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -105,7 +105,7 @@ static int format_register_str(struct snd_soc_codec *codec, if (wordsize + regsize + 2 + 1 != len) return -EINVAL; - ret = snd_soc_read(codec , reg); + ret = snd_soc_read(codec, reg); if (ret < 0) { memset(regbuf, 'X', regsize); regbuf[regsize] = '\0'; -- cgit v1.2.3 From 79ef0abcd85842bc12ffb3297b958565f060464c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Aug 2011 13:02:17 +0900 Subject: ASoC: Implement new DC servo readback mode for late WM8994 revisions Later WM8994 devices implement a new DC servo readback mode with the register used to access the offset moved to register 0x59. Implement support for this and enable it on the appropriate devices. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 3 ++- sound/soc/codecs/wm_hubs.c | 19 +++++++++++++++---- 2 files changed, 17 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 09e680ae88b2..c0956899d5b5 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -107,6 +107,7 @@ static int wm8994_volatile(struct snd_soc_codec *codec, unsigned int reg) case WM8994_LDO_2: case WM8958_DSP2_EXECCONTROL: case WM8958_MIC_DETECT_3: + case WM8994_DC_SERVO_4E: return 1; default: return 0; @@ -2978,7 +2979,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.series_startup = 1; break; default: - wm8994->hubs.dcs_readback_mode = 1; + wm8994->hubs.dcs_readback_mode = 2; break; } diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 4cc2d567f22f..84a84f4eed95 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -116,14 +117,23 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); s8 offset; - u16 reg, reg_l, reg_r, dcs_cfg; + u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg; + + switch (hubs->dcs_readback_mode) { + case 2: + dcs_reg = WM8994_DC_SERVO_4E; + break; + default: + dcs_reg = WM8993_DC_SERVO_3; + break; + } /* If we're using a digital only path and have a previously * callibrated DC servo offset stored then use that. */ if (hubs->class_w && hubs->class_w_dcs) { dev_dbg(codec->dev, "Using cached DC servo offset %x\n", hubs->class_w_dcs); - snd_soc_write(codec, WM8993_DC_SERVO_3, hubs->class_w_dcs); + snd_soc_write(codec, dcs_reg, hubs->class_w_dcs); wait_for_dc_servo(codec, WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); @@ -154,8 +164,9 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) & WM8993_DCS_INTEG_CHAN_1_MASK; break; + case 2: case 1: - reg = snd_soc_read(codec, WM8993_DC_SERVO_3); + reg = snd_soc_read(codec, dcs_reg); reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; @@ -185,7 +196,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg); /* Do it */ - snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg); + snd_soc_write(codec, dcs_reg, dcs_cfg); wait_for_dc_servo(codec, WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); -- cgit v1.2.3 From 4537c4e7618d05c77e9f8c5259c977f927a37e2f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Aug 2011 13:10:16 +0900 Subject: ASoC: Support separate left and right channel dcs_codes values Some devices can have performance optimized by setting different offsets for left and right channels. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8993.c | 3 ++- sound/soc/codecs/wm8994.c | 3 ++- sound/soc/codecs/wm_hubs.c | 13 +++++++------ sound/soc/codecs/wm_hubs.h | 3 ++- 4 files changed, 13 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 6e85b8869af7..f014e5676d20 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1433,7 +1433,8 @@ static int wm8993_probe(struct snd_soc_codec *codec) int ret, i, val; wm8993->hubs_data.hp_startup_mode = 1; - wm8993->hubs_data.dcs_codes = -2; + wm8993->hubs_data.dcs_codes_l = -2; + wm8993->hubs_data.dcs_codes_r = -2; wm8993->hubs_data.series_startup = 1; ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c0956899d5b5..fb5c96163610 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2973,7 +2973,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (wm8994->revision) { case 2: case 3: - wm8994->hubs.dcs_codes = -5; + wm8994->hubs.dcs_codes_l = -5; + wm8994->hubs.dcs_codes_r = -5; wm8994->hubs.hp_startup_mode = 1; wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.series_startup = 1; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 84a84f4eed95..26e21d01e137 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -179,18 +179,19 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r); /* Apply correction to DC servo result */ - if (hubs->dcs_codes) { - dev_dbg(codec->dev, "Applying %d code DC servo correction\n", - hubs->dcs_codes); + if (hubs->dcs_codes_l || hubs->dcs_codes_r) { + dev_dbg(codec->dev, + "Applying %d/%d code DC servo correction\n", + hubs->dcs_codes_l, hubs->dcs_codes_r); /* HPOUT1R */ offset = reg_r; - offset += hubs->dcs_codes; + offset += hubs->dcs_codes_r; dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1L */ offset = reg_l; - offset += hubs->dcs_codes; + offset += hubs->dcs_codes_l; dcs_cfg |= (u8)offset; dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg); @@ -228,7 +229,7 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, /* If we're applying an offset correction then updating the * callibration would be likely to introduce further offsets. */ - if (hubs->dcs_codes || hubs->no_series_update) + if (hubs->dcs_codes_l || hubs->dcs_codes_r || hubs->no_series_update) return ret; /* Only need to do this if the outputs are active */ diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 676b1252ab91..c674c7a502a6 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -23,7 +23,8 @@ extern const unsigned int wm_hubs_spkmix_tlv[]; /* This *must* be the first element of the codec->private_data struct */ struct wm_hubs_data { - int dcs_codes; + int dcs_codes_l; + int dcs_codes_r; int dcs_readback_mode; int hp_startup_mode; int series_startup; -- cgit v1.2.3 From c56c5d08e121d103adc026df112ed11ee3a8d1d1 Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Mon, 1 Aug 2011 19:41:18 +0800 Subject: ASoC: sgtl5000: add one missed cache reg Signed-off-by: Dong Aisheng Signed-off-by: Zeng Zhaoming Acked-by: Wolfram Sang Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index cf6eea8b458e..d9f8becafbf6 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -50,6 +50,7 @@ static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] = { 0x0000, /* 0x0016, reserved */ 0x0000, /* 0x0018, reserved */ 0x0000, /* 0x001A, reserved */ + 0x0000, /* 0x001C, reserved */ 0x0000, /* 0x001E, reserved */ 0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */ 0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */ -- cgit v1.2.3 From eaefb38f344d12321cd5372d1c8ad35d264d1b35 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 29 Jul 2011 16:27:18 +0100 Subject: ASoC: Parse board ID/revision information from WM1250-EV1 board The WM1250-EV1 board has an ID chip on it, check the board ID and display the board revision during startup. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm1250-ev1.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index bcc208967917..bbcf9ec34759 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -56,8 +56,26 @@ static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { }; static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) + const struct i2c_device_id *i2c_id) { + int ret, id, board, rev; + + board = i2c_smbus_read_byte_data(i2c, 0); + if (board < 0) { + dev_err(&i2c->dev, "Failed to read ID: %d\n", ret); + return ret; + } + + id = (board & 0xfe) >> 2; + rev = board & 0x3; + + if (id != 1) { + dev_err(&i2c->dev, "Unknown board ID %d\n", id); + return -ENODEV; + } + + dev_info(&i2c->dev, "revision %d\n", rev); + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, &wm1250_ev1_dai, 1); } -- cgit v1.2.3 From 9665408eac564374f95cc8a216e9db0aecb17ef5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 2 Aug 2011 13:04:14 +0900 Subject: ASoC: Remove -codec from WM8523 driver name It's redundant to specify it. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8523.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 4fd4d8dca0fc..131200917c56 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -551,7 +551,7 @@ MODULE_DEVICE_TABLE(i2c, wm8523_i2c_id); static struct i2c_driver wm8523_i2c_driver = { .driver = { - .name = "wm8523-codec", + .name = "wm8523", .owner = THIS_MODULE, }, .probe = wm8523_i2c_probe, -- cgit v1.2.3 From 722d0daf2b607a32dad1357bf797e3803484af0a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 2 Aug 2011 13:21:53 +0900 Subject: ASoC: Remove redundant -codec from WM8580 driver name Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8580.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 4bbc0a79f01e..95ac6651094f 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -943,7 +943,7 @@ MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id); static struct i2c_driver wm8580_i2c_driver = { .driver = { - .name = "wm8580-codec", + .name = "wm8580", .owner = THIS_MODULE, }, .probe = wm8580_i2c_probe, -- cgit v1.2.3 From be3ea3b9e8df64acb3606055c01291f0b58876a6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Jun 2011 19:35:29 +0100 Subject: ASoC: Use new register map API for ASoC generic physical I/O Remove all the ASoC specific physical I/O code and replace it with calls into the regmap API. The bulk write code can only be used safely if all regmap calls are locked with the CODEC lock, we need to add bulk support to the regmap API or replace the code with an open coded loop (though currently it has no users...). Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/Kconfig | 2 + sound/soc/soc-io.c | 319 ++++++----------------------------------------------- 2 files changed, 36 insertions(+), 285 deletions(-) (limited to 'sound') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 8224db5f0434..f9054f7c1d52 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -7,6 +7,8 @@ menuconfig SND_SOC select SND_PCM select AC97_BUS if SND_SOC_AC97_BUS select SND_JACK if INPUT=y || INPUT=SND + select REGMAP_I2C if I2C + select REGMAP_SPI if SPI_MASTER ---help--- If you want ASoC support, you should say Y here and also to the diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index cca490c80589..b56e1c4bb9e6 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -13,26 +13,13 @@ #include #include +#include #include #include -#ifdef CONFIG_SPI_MASTER -static int do_spi_write(void *control, const char *data, int len) -{ - struct spi_device *spi = control; - int ret; - - ret = spi_write(spi, data, len); - if (ret < 0) - return ret; - - return len; -} -#endif - -static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value, const void *data, int len) +static int hw_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) { int ret; @@ -49,13 +36,7 @@ static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } - ret = codec->hw_write(codec->control_data, data, len); - if (ret == len) - return 0; - if (ret < 0) - return ret; - else - return -EIO; + return regmap_write(codec->control_data, reg, value); } static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) @@ -69,8 +50,11 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) if (codec->cache_only) return -1; - BUG_ON(!codec->hw_read); - return codec->hw_read(codec, reg); + ret = regmap_read(codec->control_data, reg, &val); + if (ret == 0) + return val; + else + return ret; } ret = snd_soc_cache_read(codec, reg, &val); @@ -79,183 +63,18 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) return val; } -static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data; - - data = cpu_to_be16((reg << 12) | (value & 0xffffff)); - - return do_hw_write(codec, reg, value, &data, 2); -} - -static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data; - - data = cpu_to_be16((reg << 9) | (value & 0x1ff)); - - return do_hw_write(codec, reg, value, &data, 2); -} - -static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - reg &= 0xff; - data[0] = reg; - data[1] = value & 0xff; - - return do_hw_write(codec, reg, value, data, 2); -} - -static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - u16 val = cpu_to_be16(value); - - data[0] = reg; - memcpy(&data[1], &val, sizeof(val)); - - return do_hw_write(codec, reg, value, data, 3); -} - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int do_i2c_read(struct snd_soc_codec *codec, - void *reg, int reglen, - void *data, int datalen) -{ - struct i2c_msg xfer[2]; - int ret; - struct i2c_client *client = codec->control_data; - - /* Write register */ - xfer[0].addr = client->addr; - xfer[0].flags = 0; - xfer[0].len = reglen; - xfer[0].buf = reg; - - /* Read data */ - xfer[1].addr = client->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = datalen; - xfer[1].buf = data; - - ret = i2c_transfer(client->adapter, xfer, 2); - if (ret == 2) - return 0; - else if (ret < 0) - return ret; - else - return -EIO; -} -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u8 reg = r; - u8 data; - int ret; - - ret = do_i2c_read(codec, ®, 1, &data, 1); - if (ret < 0) - return 0; - return data; -} -#else -#define snd_soc_8_8_read_i2c NULL -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u8 reg = r; - u16 data; - int ret; - - ret = do_i2c_read(codec, ®, 1, &data, 2); - if (ret < 0) - return 0; - return (data >> 8) | ((data & 0xff) << 8); -} -#else -#define snd_soc_8_16_read_i2c NULL -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u16 reg = r; - u8 data; - int ret; - - ret = do_i2c_read(codec, ®, 2, &data, 1); - if (ret < 0) - return 0; - return data; -} -#else -#define snd_soc_16_8_read_i2c NULL -#endif - -static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - u16 rval = cpu_to_be16(reg); - - memcpy(data, &rval, sizeof(rval)); - data[2] = value; - - return do_hw_write(codec, reg, value, data, 3); -} - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u16 reg = cpu_to_be16(r); - u16 data; - int ret; - - ret = do_i2c_read(codec, ®, 2, &data, 2); - if (ret < 0) - return 0; - return be16_to_cpu(data); -} -#else -#define snd_soc_16_16_read_i2c NULL -#endif - -static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data[2]; - - data[0] = cpu_to_be16(reg); - data[1] = cpu_to_be16(value); - - return do_hw_write(codec, reg, value, data, sizeof(data)); -} - /* Primitive bulk write support for soc-cache. The data pointed to by - * `data' needs to already be in the form the hardware expects - * including any leading register specific data. Any data written - * through this function will not go through the cache as it only - * handles writing to volatile or out of bounds registers. + * `data' needs to already be in the form the hardware expects. Any + * data written through this function will not go through the cache as + * it only handles writing to volatile or out of bounds registers. + * + * This is currently only supported for devices using the regmap API + * wrappers. */ -static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg, +static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, + unsigned int reg, const void *data, size_t len) { - int ret; - /* To ensure that we don't get out of sync with the cache, check * whether the base register is volatile or if we've directly asked * to bypass the cache. Out of bounds registers are considered @@ -266,66 +85,9 @@ static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int r && reg < codec->driver->reg_cache_size) return -EINVAL; - switch (codec->control_type) { -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) - case SND_SOC_I2C: - ret = i2c_master_send(to_i2c_client(codec->dev), data, len); - break; -#endif -#if defined(CONFIG_SPI_MASTER) - case SND_SOC_SPI: - ret = spi_write(to_spi_device(codec->dev), data, len); - break; -#endif - default: - BUG(); - } - - if (ret == len) - return 0; - if (ret < 0) - return ret; - else - return -EIO; + return regmap_raw_write(codec->control_data, reg, data, len); } -static struct { - int addr_bits; - int data_bits; - int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); -} io_types[] = { - { - .addr_bits = 4, .data_bits = 12, - .write = snd_soc_4_12_write, - }, - { - .addr_bits = 7, .data_bits = 9, - .write = snd_soc_7_9_write, - }, - { - .addr_bits = 8, .data_bits = 8, - .write = snd_soc_8_8_write, - .i2c_read = snd_soc_8_8_read_i2c, - }, - { - .addr_bits = 8, .data_bits = 16, - .write = snd_soc_8_16_write, - .i2c_read = snd_soc_8_16_read_i2c, - }, - { - .addr_bits = 16, .data_bits = 8, - .write = snd_soc_16_8_write, - .i2c_read = snd_soc_16_8_read_i2c, - }, - { - .addr_bits = 16, .data_bits = 16, - .write = snd_soc_16_16_write, - .i2c_read = snd_soc_16_16_read_i2c, - }, -}; - /** * snd_soc_codec_set_cache_io: Set up standard I/O functions. * @@ -349,47 +111,34 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int addr_bits, int data_bits, enum snd_soc_control_type control) { - int i; - - for (i = 0; i < ARRAY_SIZE(io_types); i++) - if (io_types[i].addr_bits == addr_bits && - io_types[i].data_bits == data_bits) - break; - if (i == ARRAY_SIZE(io_types)) { - printk(KERN_ERR - "No I/O functions for %d bit address %d bit data\n", - addr_bits, data_bits); - return -EINVAL; - } + struct regmap_config config; - codec->write = io_types[i].write; + memset(&config, 0, sizeof(config)); + codec->write = hw_write; codec->read = hw_read; codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; + config.reg_bits = addr_bits; + config.val_bits = data_bits; + switch (control) { case SND_SOC_I2C: -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) - codec->hw_write = (hw_write_t)i2c_master_send; -#endif - if (io_types[i].i2c_read) - codec->hw_read = io_types[i].i2c_read; - - codec->control_data = container_of(codec->dev, - struct i2c_client, - dev); + codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev), + &config); break; case SND_SOC_SPI: -#ifdef CONFIG_SPI_MASTER - codec->hw_write = do_spi_write; -#endif - - codec->control_data = container_of(codec->dev, - struct spi_device, - dev); + codec->control_data = regmap_init_spi(to_spi_device(codec->dev), + &config); break; + + default: + return -EINVAL; } + if (IS_ERR(codec->control_data)) + return PTR_ERR(codec->control_data); + return 0; } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); -- cgit v1.2.3 From 0671da189c1d75eec5f6aba786d57d25209dd2bc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 24 Jul 2011 12:23:37 +0100 Subject: ASoC: Add regmap as a control type Allow drivers to set up their own regmap API structures. This is mainly useful with MFDs where the core driver will have set up regmap at the minute, though it may make sense to push the existing regmap setup out of the core into the drivers. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-io.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index b56e1c4bb9e6..e471ed667fe9 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -132,6 +132,10 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, &config); break; + case SND_SOC_REGMAP: + /* Device has made its own regmap arrangements */ + break; + default: return -EINVAL; } -- cgit v1.2.3 From 630106342e459904f7be8bf25a2493908dabe40b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 8 Aug 2011 12:44:02 +0900 Subject: ASoC: Remove unneeded -codec from WM8753 driver name Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8753.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index ffa2ffe5ec11..a7025505a7c7 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1519,7 +1519,7 @@ static int __devexit wm8753_spi_remove(struct spi_device *spi) static struct spi_driver wm8753_spi_driver = { .driver = { - .name = "wm8753-codec", + .name = "wm8753", .owner = THIS_MODULE, }, .probe = wm8753_spi_probe, @@ -1563,7 +1563,7 @@ MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id); static struct i2c_driver wm8753_i2c_driver = { .driver = { - .name = "wm8753-codec", + .name = "wm8753", .owner = THIS_MODULE, }, .probe = wm8753_i2c_probe, -- cgit v1.2.3 From 5d5d09b2fe8f00778576021d91c27b749a936420 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Aug 2011 17:34:41 +0900 Subject: ASoC: Update SMDKs for WM8580 -codec removal Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/samsung/smdk_wm8580.c | 6 +++--- sound/soc/samsung/smdk_wm8580pcm.c | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 3d26f6607aa4..20deecf3b243 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -210,7 +210,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .init = smdk_wm8580_init_paifrx, .ops = &smdk_ops, }, @@ -220,7 +220,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8580-hifi-capture", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .init = smdk_wm8580_init_paiftx, .ops = &smdk_ops, }, @@ -230,7 +230,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-i2s.x", .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .init = smdk_wm8580_init_paifrx, .ops = &smdk_ops, }, diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index 0d12092df164..4b9c73477ce0 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -127,7 +127,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-pcm.0", .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .ops = &smdk_wm8580_pcm_ops, }, { .name = "WM8580 PAIF PCM TX", @@ -135,7 +135,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-pcm.0", .codec_dai_name = "wm8580-hifi-capture", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .ops = &smdk_wm8580_pcm_ops, }, }; -- cgit v1.2.3 From 0473e61b9aeb92e167516a90bf045aa925aa3782 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Aug 2011 16:52:10 +0900 Subject: ASoC: Remove some more redundant -codecs from driver names Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8711.c | 4 ++-- sound/soc/codecs/wm8728.c | 4 ++-- sound/soc/codecs/wm8741.c | 2 +- 3 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index a537e4af6ae7..e1db7e416675 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -443,7 +443,7 @@ static int __devexit wm8711_spi_remove(struct spi_device *spi) static struct spi_driver wm8711_spi_driver = { .driver = { - .name = "wm8711-codec", + .name = "wm8711", .owner = THIS_MODULE, }, .probe = wm8711_spi_probe, @@ -487,7 +487,7 @@ MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id); static struct i2c_driver wm8711_i2c_driver = { .driver = { - .name = "wm8711-codec", + .name = "wm8711", .owner = THIS_MODULE, }, .probe = wm8711_i2c_probe, diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 86d4718d3a76..c8564f7a59a9 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -298,7 +298,7 @@ static int __devexit wm8728_spi_remove(struct spi_device *spi) static struct spi_driver wm8728_spi_driver = { .driver = { - .name = "wm8728-codec", + .name = "wm8728", .owner = THIS_MODULE, }, .probe = wm8728_spi_probe, @@ -342,7 +342,7 @@ MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id); static struct i2c_driver wm8728_i2c_driver = { .driver = { - .name = "wm8728-codec", + .name = "wm8728", .owner = THIS_MODULE, }, .probe = wm8728_i2c_probe, diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 25af901fe813..3def27ce9a65 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -526,7 +526,7 @@ MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id); static struct i2c_driver wm8741_i2c_driver = { .driver = { - .name = "wm8741-codec", + .name = "wm8741", .owner = THIS_MODULE, }, .probe = wm8741_i2c_probe, -- cgit v1.2.3 From 398575db00c17a2068a7e0c1c36d340445bc7d65 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Aug 2011 17:16:11 +0900 Subject: ASoC: Refactor WM8741 regulator handling into CODEC generic code No meaningful runtime impact but is more in line with other CODECs and will support further work. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8741.c | 76 +++++++++++++++++++++++++++-------------------- 1 file changed, 44 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 3def27ce9a65..00f80f7655cb 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -422,17 +422,35 @@ static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); int ret = 0; + int i; + + for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) + wm8741->supplies[i].supply = wm8741_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8741->control_type); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; + goto err_enable; } ret = wm8741_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); - return ret; + goto err_enable; } /* Change some default settings - latch VU */ @@ -451,10 +469,28 @@ static int wm8741_probe(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Successful registration\n"); return ret; + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); +err: + return ret; +} + +static int wm8741_remove(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + + return 0; } static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .probe = wm8741_probe, + .remove = wm8741_remove, .resume = wm8741_resume, .reg_cache_size = ARRAY_SIZE(wm8741_reg_defaults), .reg_word_size = sizeof(u16), @@ -466,43 +502,22 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8741_priv *wm8741; - int ret, i; + int ret; wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL); if (wm8741 == NULL) return -ENOMEM; - for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) - wm8741->supplies[i].supply = wm8741_supply_names[i]; - - ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8741->supplies), - wm8741->supplies); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); - goto err; - } - - ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies), - wm8741->supplies); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; - } - i2c_set_clientdata(i2c, wm8741); wm8741->control_type = SND_SOC_I2C; - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8741, &wm8741_dai, 1); - if (ret < 0) - goto err_enable; - return ret; + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_wm8741, &wm8741_dai, 1); + if (ret != 0) + goto err; -err_enable: - regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + return ret; -err_get: - regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); err: kfree(wm8741); return ret; @@ -510,10 +525,7 @@ err: static int wm8741_i2c_remove(struct i2c_client *client) { - struct wm8741_priv *wm8741 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.3 From 39e9b8d25d1c3a11e41e0044e010034a883f02ef Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Aug 2011 17:30:57 +0900 Subject: ASoC: Add SPI support for WM8741 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8741.c | 48 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 48 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 00f80f7655cb..9f6e952da8ec 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -547,6 +548,43 @@ static struct i2c_driver wm8741_i2c_driver = { }; #endif +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8741_spi_probe(struct spi_device *spi) +{ + struct wm8741_priv *wm8741; + int ret; + + wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL); + if (wm8741 == NULL) + return -ENOMEM; + + wm8741->control_type = SND_SOC_SPI; + spi_set_drvdata(spi, wm8741); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_wm8741, &wm8741_dai, 1); + if (ret < 0) + kfree(wm8741); + return ret; +} + +static int __devexit wm8741_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + kfree(spi_get_drvdata(spi)); + return 0; +} + +static struct spi_driver wm8741_spi_driver = { + .driver = { + .name = "wm8741", + .owner = THIS_MODULE, + }, + .probe = wm8741_spi_probe, + .remove = __devexit_p(wm8741_spi_remove), +}; +#endif /* CONFIG_SPI_MASTER */ + static int __init wm8741_modinit(void) { int ret = 0; @@ -556,6 +594,13 @@ static int __init wm8741_modinit(void) if (ret != 0) pr_err("Failed to register WM8741 I2C driver: %d\n", ret); #endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8741_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register wm8741 SPI driver: %d\n", + ret); + } +#endif return ret; } @@ -563,6 +608,9 @@ module_init(wm8741_modinit); static void __exit wm8741_exit(void) { +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8741_spi_driver); +#endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8741_i2c_driver); #endif -- cgit v1.2.3 From c38071c0ca00562f3008726c8f802797ad561fa7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 29 Jul 2011 16:27:18 +0100 Subject: ASoC: Fix warning in WM1250-EV1 driver Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm1250-ev1.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index bbcf9ec34759..a98a3ff1ee73 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -58,12 +58,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, const struct i2c_device_id *i2c_id) { - int ret, id, board, rev; + int id, board, rev; board = i2c_smbus_read_byte_data(i2c, 0); if (board < 0) { - dev_err(&i2c->dev, "Failed to read ID: %d\n", ret); - return ret; + dev_err(&i2c->dev, "Failed to read ID: %d\n", board); + return board; } id = (board & 0xfe) >> 2; -- cgit v1.2.3 From 53b2bb3a417606953584527b9fbf3feafad376c4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Aug 2011 14:20:09 +0900 Subject: ASoC: Specify register defaults for WM8958 MICBIAS1 and MICBIAS2 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994-tables.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index a87adbd05ee1..13e5a0186eb3 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -1635,8 +1635,8 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R58 - MICBIAS */ 0x000D, /* R59 - LDO 1 */ 0x0003, /* R60 - LDO 2 */ - 0x0000, /* R61 */ - 0x0000, /* R62 */ + 0x0039, /* R61 - MICBIAS1 */ + 0x0039, /* R62 - MICBIAS2 */ 0x0000, /* R63 */ 0x0000, /* R64 */ 0x0000, /* R65 */ -- cgit v1.2.3 From f024d9a0854cb3f2d09603d1ed3a52f04778330d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 10 Aug 2011 16:24:12 +0800 Subject: ASoC: soc-io: Add CONFIG_REGMAP_I2C/CONFIG_REGMAP_SPI guards for regmap_init_i2c/regmap_init_spi In the case of "make da8xx_omapl_defconfig;make", the SPI support is disabled. Thus calling regmap_init_spi in soc-io.c has below build error. ERROR: "regmap_init_spi" [sound/soc/snd-soc-core.ko] undefined! make[1]: *** [__modpost] Error 1 make: *** [modules] Error 2 This patch fixes the build error by adding CONFIG_REGMAP_I2C/CONFIG_REGMAP_SPI guards for regmap_init_i2c/regmap_init_spi. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/soc-io.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index e471ed667fe9..be5aac3d7a1b 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -122,15 +122,19 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, config.val_bits = data_bits; switch (control) { +#ifdef CONFIG_REGMAP_I2C case SND_SOC_I2C: codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev), &config); break; +#endif +#ifdef CONFIG_REGMAP_SPI case SND_SOC_SPI: codec->control_data = regmap_init_spi(to_spi_device(codec->dev), &config); break; +#endif case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ -- cgit v1.2.3 From 4c54c6de1aaad76092a1bc3194b351956d071b84 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 11 Aug 2011 22:19:16 +0800 Subject: ASoC: sgtl5000: fix module device table type for sgtl5000_dt_ids The module device table for of_device_id should use "of" type. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 666fae6e148d..91130fbc6913 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1441,7 +1441,7 @@ static const struct of_device_id sgtl5000_dt_ids[] = { { .compatible = "fsl,sgtl5000", }, { /* sentinel */ } }; -MODULE_DEVICE_TABLE(i2c, sgtl5000_dt_ids); +MODULE_DEVICE_TABLE(of, sgtl5000_dt_ids); static struct i2c_driver sgtl5000_i2c_driver = { .driver = { -- cgit v1.2.3 From 68d5a59e0ceb69fe8e4123666d9398c3c1331d8a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 4 Aug 2011 18:13:45 +0900 Subject: ASoC: Allow userspace control of Speyside headphone output In order to facilitate the widest range of use cases (especially things like speakerphone) allow the headphone output to be enabled and disabled by the application layer. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/samsung/speyside.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 590e9274b062..bfed1ff7093f 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -125,10 +125,6 @@ static struct snd_soc_jack_pin speyside_headset_pins[] = { .pin = "Headset Mic", .mask = SND_JACK_MICROPHONE, }, - { - .pin = "Headphone", - .mask = SND_JACK_HEADPHONE, - }, }; /* Default the headphone selection to active high */ @@ -252,6 +248,7 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Main AMIC"), SOC_DAPM_PIN_SWITCH("WM1250 Input"), SOC_DAPM_PIN_SWITCH("WM1250 Output"), + SOC_DAPM_PIN_SWITCH("Headphone"), }; static struct snd_soc_dapm_widget widgets[] = { -- cgit v1.2.3 From 81bca7624db4720db686fa38435c39ea95b7be8f Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 11 Aug 2011 11:59:11 -0600 Subject: ASoC: soc-io: Fix CONFIG_REGMAP_I2C/SPI guards to support regmap modules When CONFIG_REGMAP_I2C/SPI are m, CONFIG_REGMAP_I2C_MODULE is set in the pre-processor instead of CONFIG_REGMAP_I2C. This removes SND_SOC_I2C as a valid option for snd_soc_codec_set_cache_io()'s control parameter, and causes any ASoC regmap-using codec built as a module to fail to initialize. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/soc-io.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index be5aac3d7a1b..22b64317182b 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -122,14 +122,14 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, config.val_bits = data_bits; switch (control) { -#ifdef CONFIG_REGMAP_I2C +#if defined(CONFIG_REGMAP_I2C) || defined(CONFIG_REGMAP_I2C_MODULE) case SND_SOC_I2C: codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev), &config); break; #endif -#ifdef CONFIG_REGMAP_SPI +#if defined(CONFIG_REGMAP_SPI) || defined(CONFIG_REGMAP_SPI_MODULE) case SND_SOC_SPI: codec->control_data = regmap_init_spi(to_spi_device(codec->dev), &config); -- cgit v1.2.3 From 0a422e1a19e1583913d6f762882f115605109107 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 2 Aug 2011 13:03:04 +0900 Subject: ASoC: Add device tree binding for WM8510 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8510.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index db0dced74843..55a4c830e111 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -598,6 +599,11 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8510 = { .reg_cache_default =wm8510_reg, }; +static const struct of_device_id wm8510_of_match[] = { + { .compatible = "wlf,wm8510" }, + { }, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8510_spi_probe(struct spi_device *spi) { @@ -628,6 +634,7 @@ static struct spi_driver wm8510_spi_driver = { .driver = { .name = "wm8510", .owner = THIS_MODULE, + .of_match_table = wm8510_of_match, }, .probe = wm8510_spi_probe, .remove = __devexit_p(wm8510_spi_remove), @@ -671,6 +678,7 @@ static struct i2c_driver wm8510_i2c_driver = { .driver = { .name = "wm8510-codec", .owner = THIS_MODULE, + .of_match_table = wm8510_of_match, }, .probe = wm8510_i2c_probe, .remove = __devexit_p(wm8510_i2c_remove), -- cgit v1.2.3 From bf5a85be9739ea26f05874992cb05a9c774f359f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 2 Aug 2011 13:08:13 +0900 Subject: ASoC: Add device tree binding for WM8523 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8523.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 131200917c56..52812d1a90e4 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -514,6 +515,11 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8523 = { .volatile_register = wm8523_volatile_register, }; +static const struct of_device_id wm8523_of_match[] = { + { .compatible = "wlf,wm8523" }, + { }, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -553,6 +559,7 @@ static struct i2c_driver wm8523_i2c_driver = { .driver = { .name = "wm8523", .owner = THIS_MODULE, + .of_match_table = wm8523_of_match, }, .probe = wm8523_i2c_probe, .remove = __devexit_p(wm8523_i2c_remove), -- cgit v1.2.3 From 733eef3e96726350888bd0ec1526f2561bec44ed Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 2 Aug 2011 13:22:36 +0900 Subject: ASoC: Add device tree support for WM8580 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8580.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 95ac6651094f..4664c3a76c78 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include @@ -907,6 +908,11 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8580 = { .reg_cache_default = wm8580_reg, }; +static const struct of_device_id wm8580_of_match[] = { + { .compatible = "wlf,wm8580" }, + { }, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -945,6 +951,7 @@ static struct i2c_driver wm8580_i2c_driver = { .driver = { .name = "wm8580", .owner = THIS_MODULE, + .of_match_table = wm8580_of_match, }, .probe = wm8580_i2c_probe, .remove = wm8580_i2c_remove, -- cgit v1.2.3 From 1552c8f67ea70803b18e2c7c525a8da5f90384c1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Aug 2011 16:52:47 +0900 Subject: ASoC: Add device tree binding for WM8711 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8711.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index e1db7e416675..8457d3cb5962 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include @@ -414,6 +415,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8711 = { .num_dapm_routes = ARRAY_SIZE(wm8711_intercon), }; +static const struct of_device_id wm8711_of_match[] = { + { .compatible = "wlf,wm8711", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8711_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8711_spi_probe(struct spi_device *spi) { @@ -445,6 +452,7 @@ static struct spi_driver wm8711_spi_driver = { .driver = { .name = "wm8711", .owner = THIS_MODULE, + .of_match_table = wm8711_of_match, }, .probe = wm8711_spi_probe, .remove = __devexit_p(wm8711_spi_remove), @@ -489,6 +497,7 @@ static struct i2c_driver wm8711_i2c_driver = { .driver = { .name = "wm8711", .owner = THIS_MODULE, + .of_match_table = wm8711_of_match, }, .probe = wm8711_i2c_probe, .remove = __devexit_p(wm8711_i2c_remove), -- cgit v1.2.3 From 45b4d043da571c71500add0fa3ea17c9b8f1f648 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Aug 2011 16:53:02 +0900 Subject: ASoC: Add device tree binding for WM8728 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8728.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index c8564f7a59a9..04b027efd5c0 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -269,6 +270,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8728 = { .num_dapm_routes = ARRAY_SIZE(wm8728_intercon), }; +static const struct of_device_id wm8728_of_match[] = { + { .compatible = "wlf,wm8728", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8728_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8728_spi_probe(struct spi_device *spi) { @@ -300,6 +307,7 @@ static struct spi_driver wm8728_spi_driver = { .driver = { .name = "wm8728", .owner = THIS_MODULE, + .of_match_table = wm8728_of_match, }, .probe = wm8728_spi_probe, .remove = __devexit_p(wm8728_spi_remove), @@ -344,6 +352,7 @@ static struct i2c_driver wm8728_i2c_driver = { .driver = { .name = "wm8728", .owner = THIS_MODULE, + .of_match_table = wm8728_of_match, }, .probe = wm8728_i2c_probe, .remove = __devexit_p(wm8728_i2c_remove), -- cgit v1.2.3 From 53a5a83d24d8ee9567bfcbaf3b37ca5a00ab16a9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Aug 2011 16:53:22 +0900 Subject: ASoC: Add device tree binding for WM8737 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8737.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 30c67d06a904..f6aef58845c2 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -634,6 +635,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8737 = { .reg_cache_default = wm8737_reg, }; +static const struct of_device_id wm8737_of_match[] = { + { .compatible = "wlf,wm8737", }, + { } +}; + +MODULE_DEVICE_TABLE(of, wm8737_of_match); + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8737_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -673,6 +681,7 @@ static struct i2c_driver wm8737_i2c_driver = { .driver = { .name = "wm8737", .owner = THIS_MODULE, + .of_match_table = wm8737_of_match, }, .probe = wm8737_i2c_probe, .remove = __devexit_p(wm8737_i2c_remove), @@ -711,6 +720,7 @@ static struct spi_driver wm8737_spi_driver = { .driver = { .name = "wm8737", .owner = THIS_MODULE, + .of_match_table = wm8737_of_match, }, .probe = wm8737_spi_probe, .remove = __devexit_p(wm8737_spi_remove), -- cgit v1.2.3 From ce31a0f5a6d2c73b61d04f7d886a1f8101eed29f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Aug 2011 17:34:59 +0900 Subject: ASoC: Add device tree binding for WM8750 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8750.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 82ac5fcaa2b2..bbb697470d82 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include @@ -751,6 +752,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8750 = { .reg_cache_default = wm8750_reg, }; +static const struct of_device_id wm8750_of_match[] = { + { .compatible = "wlf,wm8750", }, + { .compatible = "wlf,wm8987", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8750_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8750_spi_probe(struct spi_device *spi) { @@ -789,6 +797,7 @@ static struct spi_driver wm8750_spi_driver = { .driver = { .name = "wm8750-codec", .owner = THIS_MODULE, + .of_match_table = wm8750_of_match, }, .id_table = wm8750_spi_ids, .probe = wm8750_spi_probe, @@ -835,6 +844,7 @@ static struct i2c_driver wm8750_i2c_driver = { .driver = { .name = "wm8750-codec", .owner = THIS_MODULE, + .of_match_table = wm8750_of_match, }, .probe = wm8750_i2c_probe, .remove = __devexit_p(wm8750_i2c_remove), -- cgit v1.2.3 From 70e141228a24a538dfcd5ba641f92c1bdc239eb0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 8 Aug 2011 12:44:27 +0900 Subject: ASoC: Add device tree binding for WM8753 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8753.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a7025505a7c7..fe04a101d657 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -38,6 +38,7 @@ #include #include #include +#include #include #include #include @@ -1490,6 +1491,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8753 = { .reg_cache_default = wm8753_reg, }; +static const struct of_device_id wm8753_of_match[] = { + { .compatible = "wlf,wm8753", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8753_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8753_spi_probe(struct spi_device *spi) { @@ -1521,6 +1528,7 @@ static struct spi_driver wm8753_spi_driver = { .driver = { .name = "wm8753", .owner = THIS_MODULE, + .of_match_table = wm8753_of_match, }, .probe = wm8753_spi_probe, .remove = __devexit_p(wm8753_spi_remove), @@ -1565,6 +1573,7 @@ static struct i2c_driver wm8753_i2c_driver = { .driver = { .name = "wm8753", .owner = THIS_MODULE, + .of_match_table = wm8753_of_match, }, .probe = wm8753_i2c_probe, .remove = __devexit_p(wm8753_i2c_remove), -- cgit v1.2.3 From 4e04adaf87c678425b8009c5f208d9acfc1530ab Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 15 Jul 2011 15:12:31 +0900 Subject: ASoC: Add VMID widget for wm_hubs devices Currently this does not actually do anything, it is being introduced in order to facilitate additional power optimisations for current generation devices. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8993.c | 4 ++++ sound/soc/codecs/wm8994.c | 3 +++ sound/soc/codecs/wm_hubs.c | 16 ++++++++++++++++ 3 files changed, 23 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index f014e5676d20..eec8e1435116 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -847,6 +847,7 @@ SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8993_BUS_CONTROL_1, 1, 0, clk_sys_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("TOCLK", WM8993_CLOCKING_1, 14, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8993_CLOCKING_3, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_ADC("ADCL", NULL, WM8993_POWER_MANAGEMENT_2, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, WM8993_POWER_MANAGEMENT_2, 0, 0), @@ -880,6 +881,9 @@ SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), }; static const struct snd_soc_dapm_route routes[] = { + { "MICBIAS1", NULL, "VMID" }, + { "MICBIAS2", NULL, "VMID" }, + { "ADCL", NULL, "CLK_SYS" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCR", NULL, "CLK_SYS" }, diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 4487c5e6ad89..f57e01344adb 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1209,6 +1209,7 @@ SND_SOC_DAPM_INPUT("Clock"), SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1526,6 +1527,8 @@ static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { static const struct snd_soc_dapm_route wm8994_intercon[] = { { "AIF2DACL", NULL, "AIF2DAC Mux" }, { "AIF2DACR", NULL, "AIF2DAC Mux" }, + { "MICBIAS1", NULL, "VMID" }, + { "MICBIAS2", NULL, "VMID" }, }; static const struct snd_soc_dapm_route wm8958_intercon[] = { diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 017522e7cef9..ca8ce03510f4 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -711,6 +711,11 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "IN1L PGA", "IN1LP Switch", "IN1LP" }, { "IN1L PGA", "IN1LN Switch", "IN1LN" }, + { "IN1L PGA", NULL, "VMID" }, + { "IN1R PGA", NULL, "VMID" }, + { "IN2L PGA", NULL, "VMID" }, + { "IN2R PGA", NULL, "VMID" }, + { "IN1R PGA", "IN1RP Switch", "IN1RP" }, { "IN1R PGA", "IN1RN Switch", "IN1RN" }, @@ -728,12 +733,14 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "MIXINL", NULL, "Direct Voice" }, { "MIXINL", NULL, "IN1LP" }, { "MIXINL", NULL, "Left Output Mixer" }, + { "MIXINL", NULL, "VMID" }, { "MIXINR", "IN1R Switch", "IN1R PGA" }, { "MIXINR", "IN2R Switch", "IN2R PGA" }, { "MIXINR", NULL, "Direct Voice" }, { "MIXINR", NULL, "IN1RP" }, { "MIXINR", NULL, "Right Output Mixer" }, + { "MIXINR", NULL, "VMID" }, { "ADCL", NULL, "MIXINL" }, { "ADCR", NULL, "MIXINR" }, @@ -764,6 +771,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Earpiece Mixer", "Left Output Switch", "Left Output PGA" }, { "Earpiece Mixer", "Right Output Switch", "Right Output PGA" }, + { "Earpiece Driver", NULL, "VMID" }, { "Earpiece Driver", NULL, "Earpiece Mixer" }, { "HPOUT2N", NULL, "Earpiece Driver" }, { "HPOUT2P", NULL, "Earpiece Driver" }, @@ -786,9 +794,11 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "SPKR Boost", "SPKR Switch", "SPKR" }, { "SPKR Boost", "SPKL Switch", "SPKL" }, + { "SPKL Driver", NULL, "VMID" }, { "SPKL Driver", NULL, "SPKL Boost" }, { "SPKL Driver", NULL, "CLK_SYS" }, + { "SPKR Driver", NULL, "VMID" }, { "SPKR Driver", NULL, "SPKR Boost" }, { "SPKR Driver", NULL, "CLK_SYS" }, @@ -802,12 +812,18 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Headphone PGA", NULL, "Left Headphone Mux" }, { "Headphone PGA", NULL, "Right Headphone Mux" }, + { "Headphone PGA", NULL, "VMID" }, { "Headphone PGA", NULL, "CLK_SYS" }, { "Headphone PGA", NULL, "Headphone Supply" }, { "HPOUT1L", NULL, "Headphone PGA" }, { "HPOUT1R", NULL, "Headphone PGA" }, + { "LINEOUT1N Driver", NULL, "VMID" }, + { "LINEOUT1P Driver", NULL, "VMID" }, + { "LINEOUT2N Driver", NULL, "VMID" }, + { "LINEOUT2P Driver", NULL, "VMID" }, + { "LINEOUT1N", NULL, "LINEOUT1N Driver" }, { "LINEOUT1P", NULL, "LINEOUT1P Driver" }, { "LINEOUT2N", NULL, "LINEOUT2N Driver" }, -- cgit v1.2.3 From 4b7ed83aa3c7f4b9fe363875440836e0f2aabbdf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Aug 2011 17:47:33 +0900 Subject: ASoC: Disable WM8994 VMID for digital only paths On WM8994 class devices only the analogue portions of the CODEC require VMID so when running digital only paths we can leave VMID disabled. On some earlier devices the FLL uses VMID so we don't use DAPM reference counting alone, we maintain an internal reference count which is also enabled and disabled by the FLL startup. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 183 ++++++++++++++++++++++++++++++++-------------- sound/soc/codecs/wm8994.h | 2 + 2 files changed, 129 insertions(+), 56 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index f57e01344adb..e5691ad8a2d3 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -682,6 +682,97 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, return 0; } +static void vmid_reference(struct snd_soc_codec *codec) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + wm8994->vmid_refcount++; + + dev_dbg(codec->dev, "Referencing VMID, refcount is now %d\n", + wm8994->vmid_refcount); + + if (wm8994->vmid_refcount == 1) { + /* Startup bias, VMID ramp & buffer */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (0x11 << WM8994_VMID_RAMP_SHIFT)); + + /* Main bias enable, VMID=2x40k */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | + WM8994_VMID_SEL_MASK, + WM8994_BIAS_ENA | 0x2); + + msleep(20); + } +} + +static void vmid_dereference(struct snd_soc_codec *codec) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + wm8994->vmid_refcount--; + + dev_dbg(codec->dev, "Dereferencing VMID, refcount is now %d\n", + wm8994->vmid_refcount); + + if (wm8994->vmid_refcount == 0) { + /* Switch over to startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (1 << WM8994_VMID_RAMP_SHIFT)); + + /* Disable main biases */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | + WM8994_VMID_SEL_MASK, 0); + + /* Discharge line */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH); + + msleep(5); + + /* Switch off startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, 0); + } +} + +static int vmid_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + vmid_reference(codec); + break; + + case SND_SOC_DAPM_POST_PMD: + vmid_dereference(codec); + break; + } + + return 0; +} + static void wm8994_update_class_w(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -1209,7 +1300,8 @@ SND_SOC_DAPM_INPUT("Clock"), SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev, SND_SOC_DAPM_PRE_PMU), -SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, vmid_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1633,10 +1725,12 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, unsigned int freq_in, unsigned int freq_out) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; int reg_offset, ret; struct fll_div fll; u16 reg, aif1, aif2; unsigned long timeout; + bool was_enabled; aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) & WM8994_AIF1CLK_ENA; @@ -1657,6 +1751,9 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, return -EINVAL; } + reg = snd_soc_read(codec, WM8994_FLL1_CONTROL_1 + reg_offset); + was_enabled = reg & WM8994_FLL1_ENA; + switch (src) { case 0: /* Allow no source specification when stopping */ @@ -1723,6 +1820,21 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, /* Enable (with fractional mode if required) */ if (freq_out) { + /* Enable VMID if we need it */ + if (!was_enabled) { + switch (control->type) { + case WM8994: + vmid_reference(codec); + break; + case WM8958: + if (wm8994->revision < 1) + vmid_reference(codec); + break; + default: + break; + } + } + if (fll.k) reg = WM8994_FLL1_ENA | WM8994_FLL1_FRAC; else @@ -1740,6 +1852,20 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, } else { msleep(5); } + } else { + if (was_enabled) { + switch (control->type) { + case WM8994: + vmid_dereference(codec); + break; + case WM8958: + if (wm8994->revision < 1) + vmid_dereference(codec); + break; + default: + break; + } + } } wm8994->fll[id].in = freq_in; @@ -1856,9 +1982,6 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_PREPARE: - /* VMID=2x40k */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_VMID_SEL_MASK, 0x2); break; case SND_SOC_BIAS_STANDBY: @@ -1900,65 +2023,13 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, WM8994_LINEOUT2_DISCH, WM8994_LINEOUT1_DISCH | WM8994_LINEOUT2_DISCH); - - /* Startup bias, VMID ramp & buffer */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - (0x11 << WM8994_VMID_RAMP_SHIFT)); - - /* Main bias enable, VMID=2x40k */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_BIAS_ENA | - WM8994_VMID_SEL_MASK, - WM8994_BIAS_ENA | 0x2); - - msleep(20); } - /* VMID=2x500k */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_VMID_SEL_MASK, 0x4); break; case SND_SOC_BIAS_OFF: if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { - /* Switch over to startup biases */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_BIAS_SRC | - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, - WM8994_BIAS_SRC | - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - (1 << WM8994_VMID_RAMP_SHIFT)); - - /* Disable main biases */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_BIAS_ENA | - WM8994_VMID_SEL_MASK, 0); - - /* Discharge line */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_1, - WM8994_LINEOUT1_DISCH | - WM8994_LINEOUT2_DISCH, - WM8994_LINEOUT1_DISCH | - WM8994_LINEOUT2_DISCH); - - msleep(5); - - /* Switch off startup biases */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_BIAS_SRC | - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, 0); - wm8994->cur_fw = NULL; pm_runtime_put(codec->dev); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 1ab2266039f7..f4f1355efc82 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -83,6 +83,8 @@ struct wm8994_priv { struct completion fll_locked[2]; bool fll_locked_irq; + int vmid_refcount; + int dac_rates[2]; int lrclk_shared[2]; -- cgit v1.2.3 From dc5de62be6138dd141e78a454e9af0fd88d1eec8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Aug 2011 18:22:28 +0900 Subject: ASoC: Remove redundant -codec from WM8750 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8750.c | 4 ++-- sound/soc/pxa/spitz.c | 2 +- sound/soc/pxa/z2.c | 2 +- sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/smartq_wm8987.c | 2 +- 5 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index bbb697470d82..3f2aeec309a7 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -795,7 +795,7 @@ MODULE_DEVICE_TABLE(spi, wm8750_spi_ids); static struct spi_driver wm8750_spi_driver = { .driver = { - .name = "wm8750-codec", + .name = "wm8750", .owner = THIS_MODULE, .of_match_table = wm8750_of_match, }, @@ -842,7 +842,7 @@ MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); static struct i2c_driver wm8750_i2c_driver = { .driver = { - .name = "wm8750-codec", + .name = "wm8750", .owner = THIS_MODULE, .of_match_table = wm8750_of_match, }, diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index b253d864868a..ce920e3cfea1 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -312,7 +312,7 @@ static struct snd_soc_dai_link spitz_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8750-codec.0-001b", + .codec_name = "wm8750.0-001b", .init = spitz_wm8750_init, .ops = &spitz_ops, }; diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index d69d9fc32233..4b81ffd87566 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -198,7 +198,7 @@ static struct snd_soc_dai_link z2_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8750-codec.0-001b", + .codec_name = "wm8750.0-001b", .init = z2_wm8750_init, .ops = &z2_ops, }; diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 14eb6ea69e7c..ed8f13a29c85 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -131,7 +131,7 @@ static struct snd_soc_dai_link jive_dai = { .cpu_dai_name = "s3c2412-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-001a", + .codec_name = "wm8750.0-001a", .init = jive_wm8750_init, .ops = &jive_ops, }; diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 0a2c4f223038..bbd14768ecd3 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -207,7 +207,7 @@ static struct snd_soc_dai_link smartq_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-0x1a", + .codec_name = "wm8750.0-0x1a", .init = smartq_wm8987_init, .ops = &smartq_hifi_ops, }, -- cgit v1.2.3 From 4835ff9aca639107ca0233c10aa854d460c8797d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 13 Aug 2011 11:50:48 +0900 Subject: ASoC: Support !CONFIG_REGMAP builds Since we changed regmap to be selected and register per bus rather than via the core only we can't rely on it being enabled by the ASoC core. Support compiling it out. Signed-off-by: Mark Brown Reported-by: Axel Lin --- sound/soc/soc-io.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 22b64317182b..66fcccd79efe 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -18,6 +18,7 @@ #include +#ifdef CONFIG_REGMAP static int hw_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -150,4 +151,12 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, return 0; } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); - +#else +int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, + int addr_bits, int data_bits, + enum snd_soc_control_type control) +{ + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); +#endif -- cgit v1.2.3 From ea19f494d6944ade02085d894a17c92d8e3057f0 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 13 Aug 2011 11:32:56 +0800 Subject: ASoC: s6000-pcm: remove unused variable 'dai' MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Signed-off-by: Axel Lin Acked-by: Daniel Glöckner Signed-off-by: Mark Brown --- sound/soc/s6000/s6000-pcm.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 80c85fd64e1a..55efc2bdf0bd 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -446,7 +446,6 @@ static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32); static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime) { struct snd_card *card = runtime->card->snd_card; - struct snd_soc_dai *dai = runtime->cpu_dai; struct snd_pcm *pcm = runtime->pcm; struct s6000_pcm_dma_params *params; int res; -- cgit v1.2.3 From 31e12dd377f463eba21609486ee809fd1418d400 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 13 Aug 2011 21:12:01 +0800 Subject: ASoC: soc-cache: Remove unneeded codec_drv pointer variable in snd_soc_flat_cache_init Since commit d779fce5d79525d66269c8f6e430e1515d697f3d "ASoC: soc-cache: Ensure flat compression uses a copy of the defaults cache", the codec_drv pointer variable is not used any more. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d9f8aded51f3..c17d01211835 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -868,10 +868,6 @@ static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec) static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) { - const struct snd_soc_codec_driver *codec_drv; - - codec_drv = codec->driver; - if (codec->reg_def_copy) codec->reg_cache = kmemdup(codec->reg_def_copy, codec->reg_size, GFP_KERNEL); -- cgit v1.2.3 From 3a52f19ee6b5cbe36b9a47ab01ee7ca6d2e559e8 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 13 Aug 2011 21:27:16 +0800 Subject: ASoC: soc-cache: Remove unneeded codec_drv pointer variable in snd_soc_lzo_get_blksize Since commit aea170a099793abcd0e6de46b947458073204241 "ASoC: soc-cache: Add reg_size as a member to snd_soc_codec", the codec_drv pointer variable is not used in snd_soc_lzo_get_blksize. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index c17d01211835..fdfd4881c9d1 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -548,9 +548,6 @@ static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec, static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec) { - const struct snd_soc_codec_driver *codec_drv; - - codec_drv = codec->driver; return DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); } -- cgit v1.2.3 From 6fc562e49cb42792ff347f253fe9d026e8e16f34 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 14 Aug 2011 08:00:12 +0800 Subject: ASoC: soc-pcm: Remove unused global mutex Since commit b8c0dab9bf3373010e857a8d3f1b594c60a348dd "ASoC: core - PCM mutex per rtd", the global pcm_mutex is not being used any more. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b5759397afa3..1347584d64df 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -27,8 +27,6 @@ #include #include -static DEFINE_MUTEX(pcm_mutex); - static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; -- cgit v1.2.3 From 1f9099b41723c90bd2ff8238482e8598ef21a621 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 15 Aug 2011 16:38:34 +0800 Subject: ASoC: nuc900-pcm: remove unused variable 'dai' Remove unused variable 'dai' to eliminate below warning. CC sound/soc/nuc900/nuc900-pcm.o sound/soc/nuc900/nuc900-pcm.c: In function 'nuc900_dma_new': sound/soc/nuc900/nuc900-pcm.c:321: warning: unused variable 'dai' Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-pcm.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index d589ef14e917..e46d5516e000 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -318,7 +318,6 @@ static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32); static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; if (!card->dev->dma_mask) -- cgit v1.2.3 From a595238bad3d11b26d00bbda4ccbd38fe107cd1e Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 14 Aug 2011 11:31:04 +0800 Subject: ASoC: sta32x: shortcut the for loop to get ir and mcs There is exactly one match or no match at all during the for loop iteration, thus we can break from the for loop once a match is found. Signed-off-by: Axel Lin Acked-by: Johannes Stezenbach Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index fbd7eb9e61ce..3d155f526672 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -524,13 +524,17 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); pr_debug("rate: %u\n", rate); for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) - if (interpolation_ratios[i].fs == rate) + if (interpolation_ratios[i].fs == rate) { ir = interpolation_ratios[i].ir; + break; + } if (ir < 0) return -EINVAL; for (i = 0; mclk_ratios[ir][i].ratio; i++) - if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) + if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) { mcs = mclk_ratios[ir][i].mcs; + break; + } if (mcs < 0) return -EINVAL; -- cgit v1.2.3 From 80080ec5399acb4e83f1216b24fd07e93c5c4b2c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Aug 2011 17:31:26 +0900 Subject: ASoC: Add device tree binding for WM8741 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8741.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 9f6e952da8ec..78c9e5ab3fa5 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include @@ -498,6 +499,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .reg_cache_default = wm8741_reg_defaults, }; +static const struct of_device_id wm8741_of_match[] = { + { .compatible = "wlf,wm8741", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8741_of_match); + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -541,6 +548,7 @@ static struct i2c_driver wm8741_i2c_driver = { .driver = { .name = "wm8741", .owner = THIS_MODULE, + .of_match_table = wm8741_of_match, }, .probe = wm8741_i2c_probe, .remove = wm8741_i2c_remove, @@ -579,6 +587,7 @@ static struct spi_driver wm8741_spi_driver = { .driver = { .name = "wm8741", .owner = THIS_MODULE, + .of_match_table = wm8741_of_match, }, .probe = wm8741_spi_probe, .remove = __devexit_p(wm8741_spi_remove), -- cgit v1.2.3 From 5c58b739c3a1cc41a80991c37b17e181dd1bb51d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 14 Aug 2011 19:07:33 +0900 Subject: ASoC: Correct revision display for WM1250-EV1 module The hardware documentation uses revision numbers starting at 1. Signed-off-by: Mark Brown --- sound/soc/codecs/wm1250-ev1.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index a98a3ff1ee73..4523c4cec02b 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -74,7 +74,7 @@ static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, return -ENODEV; } - dev_info(&i2c->dev, "revision %d\n", rev); + dev_info(&i2c->dev, "revision %d\n", rev + 1); return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, &wm1250_ev1_dai, 1); -- cgit v1.2.3 From d09f3ecf1a7ba658934fa3c45f67ed2620a50950 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 15 Aug 2011 11:01:02 +0900 Subject: ASoC: Disable pulls on WM8994 AIF2 when starting it Pull control is availalbe for WM8994 AIF2, generally disabled as part of the GPIO configuration in order to save power after system startup. As on newer devices in the series there is no GPIO functionality on these pins this will happen less naturally so have the driver disable the pulls as the AIF is probed. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e5691ad8a2d3..0f36eeeb5fae 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2459,6 +2459,21 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) return snd_soc_update_bits(codec, reg, mask, val); } +static int wm8994_aif2_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* Disable the pulls on the AIF if we're using it to save power. */ + snd_soc_update_bits(codec, WM8994_GPIO_3, + WM8994_GPN_PU | WM8994_GPN_PD, 0); + snd_soc_update_bits(codec, WM8994_GPIO_4, + WM8994_GPN_PU | WM8994_GPN_PD, 0); + snd_soc_update_bits(codec, WM8994_GPIO_5, + WM8994_GPN_PU | WM8994_GPN_PD, 0); + + return 0; +} + #define WM8994_RATES SNDRV_PCM_RATE_8000_96000 #define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ @@ -2526,6 +2541,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, + .probe = wm8994_aif2_probe, .ops = &wm8994_aif2_dai_ops, }, { -- cgit v1.2.3 From 82cd87643be7f133ad9a85865f67a0dcadcb76c7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 15 Aug 2011 20:15:21 +0200 Subject: ASoC: DAPM: Allow multiple mixer sources to be routed via the same switch Currently it is only possible to route one source per switch into a mixer. This patch modifies the code, so that it is possible to route multiple sources into a mixer via the same switch. One use-case for this is routing a stereo channel pair into a mono-mixer via the same switch. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c26531132c66..170c4ffa609f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -443,6 +443,11 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) if (path->name != (char *)w->kcontrol_news[i].name) continue; + if (w->kcontrols[i]) { + path->kcontrol = w->kcontrols[i]; + continue; + } + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + sizeof(struct snd_soc_dapm_widget *), wlist = kzalloc(wlistsize, GFP_KERNEL); @@ -1556,7 +1561,6 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, /* found, now check type */ found = 1; path->connect = connect; - break; } if (found) -- cgit v1.2.3 From 70ce6aee664a3e61ca5b4278d61db6da0996cade Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 16 Aug 2011 21:36:38 +0900 Subject: ASoC: Run Speyside WM8962 at 512fs Ensure we have access to all the advanced DSP functinality offered by the WM8962 by running the system clock at 512fs. Signed-off-by: Mark Brown --- sound/soc/samsung/speyside_wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 0b9eb5f7ec4c..753e1c2702b0 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -28,13 +28,13 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, 32768, - 44100 * 256); + 44100 * 512); if (ret < 0) pr_err("Failed to start FLL: %d\n", ret); ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, - 44100 * 256, + 44100 * 512, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("Failed to set SYSCLK: %d\n", ret); -- cgit v1.2.3 From 1ddc07d0f13a753f8e345e0538562e1899d2bc26 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 16 Aug 2011 10:08:48 +0900 Subject: ASoC: Add WM8958 noise gate support Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994-tables.c | 12 ++++++------ sound/soc/codecs/wm8994.c | 38 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 44 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index 13e5a0186eb3..df5a8b9a250f 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -1073,8 +1073,8 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x0000, 0x0000 }, /* R1069 */ { 0x0000, 0x0000 }, /* R1070 */ { 0x0000, 0x0000 }, /* R1071 */ - { 0x0000, 0x0000 }, /* R1072 */ - { 0x0000, 0x0000 }, /* R1073 */ + { 0x006F, 0x006F }, /* R1072 - AIF1 DAC1 Noise Gate */ + { 0x006F, 0x006F }, /* R1073 - AIF1 DAC2 Noise Gate */ { 0x0000, 0x0000 }, /* R1074 */ { 0x0000, 0x0000 }, /* R1075 */ { 0x0000, 0x0000 }, /* R1076 */ @@ -1329,7 +1329,7 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x0000, 0x0000 }, /* R1325 */ { 0x0000, 0x0000 }, /* R1326 */ { 0x0000, 0x0000 }, /* R1327 */ - { 0x0000, 0x0000 }, /* R1328 */ + { 0x006F, 0x006F }, /* R1328 - AIF2 DAC Noise Gate */ { 0x0000, 0x0000 }, /* R1329 */ { 0x0000, 0x0000 }, /* R1330 */ { 0x0000, 0x0000 }, /* R1331 */ @@ -2646,8 +2646,8 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R1069 */ 0x0000, /* R1070 */ 0x0000, /* R1071 */ - 0x0000, /* R1072 */ - 0x0000, /* R1073 */ + 0x0068, /* R1072 - AIF1 DAC1 Noise Gate */ + 0x0068, /* R1073 - AIF1 DAC2 Noise Gate */ 0x0000, /* R1074 */ 0x0000, /* R1075 */ 0x0000, /* R1076 */ @@ -2902,7 +2902,7 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R1325 */ 0x0000, /* R1326 */ 0x0000, /* R1327 */ - 0x0000, /* R1328 */ + 0x0068, /* R1328 - AIF2 DAC Noise Gate */ 0x0000, /* R1329 */ 0x0000, /* R1330 */ 0x0000, /* R1331 */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0f36eeeb5fae..a0d6274ec280 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -282,6 +282,7 @@ static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); static const DECLARE_TLV_DB_SCALE(st_tlv, -3600, 300, 0); static const DECLARE_TLV_DB_SCALE(wm8994_3d_tlv, -1600, 183, 0); static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); #define WM8994_DRC_SWITCH(xname, reg, shift) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -661,8 +662,45 @@ SOC_SINGLE_TLV("AIF2 EQ5 Volume", WM8994_AIF2_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static const char *wm8958_ng_text[] = { + "30ms", "125ms", "250ms", "500ms", +}; + +static const struct soc_enum wm8958_aif1dac1_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF1_DAC1_NOISE_GATE, + WM8958_AIF1DAC1_NG_THR_SHIFT, 4, wm8958_ng_text); + +static const struct soc_enum wm8958_aif1dac2_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF1_DAC2_NOISE_GATE, + WM8958_AIF1DAC2_NG_THR_SHIFT, 4, wm8958_ng_text); + +static const struct soc_enum wm8958_aif2dac_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF2_DAC_NOISE_GATE, + WM8958_AIF2DAC_NG_THR_SHIFT, 4, wm8958_ng_text); + static const struct snd_kcontrol_new wm8958_snd_controls[] = { SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv), + +SOC_SINGLE("AIF1DAC1 Noise Gate Switch", WM8958_AIF1_DAC1_NOISE_GATE, + WM8958_AIF1DAC1_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF1DAC1 Noise Gate Hold Time", wm8958_aif1dac1_ng_hold), +SOC_SINGLE_TLV("AIF1DAC1 Noise Gate Threshold Volume", + WM8958_AIF1_DAC1_NOISE_GATE, WM8958_AIF1DAC1_NG_THR_SHIFT, + 7, 1, ng_tlv), + +SOC_SINGLE("AIF1DAC2 Noise Gate Switch", WM8958_AIF1_DAC2_NOISE_GATE, + WM8958_AIF1DAC2_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF1DAC2 Noise Gate Hold Time", wm8958_aif1dac2_ng_hold), +SOC_SINGLE_TLV("AIF1DAC2 Noise Gate Threshold Volume", + WM8958_AIF1_DAC2_NOISE_GATE, WM8958_AIF1DAC2_NG_THR_SHIFT, + 7, 1, ng_tlv), + +SOC_SINGLE("AIF2DAC Noise Gate Switch", WM8958_AIF2_DAC_NOISE_GATE, + WM8958_AIF2DAC_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF2DAC Noise Gate Hold Time", wm8958_aif2dac_ng_hold), +SOC_SINGLE_TLV("AIF2DAC Noise Gate Threshold Volume", + WM8958_AIF2_DAC_NOISE_GATE, WM8958_AIF2DAC_NG_THR_SHIFT, + 7, 1, ng_tlv), }; static int clk_sys_event(struct snd_soc_dapm_widget *w, -- cgit v1.2.3 From f0b182b003e22320efac5a33cacc460e0079c135 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 16 Aug 2011 12:01:27 +0900 Subject: ASoC: Implement WM8994 thermal warning and shutdown interrupt support ALSA doesn't really have good mechanisms for dealing with these so we just log them - the hardware already has automatic shutdown support. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a0d6274ec280..94124913bb3e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3045,6 +3045,24 @@ static irqreturn_t wm8994_fifo_error(int irq, void *data) return IRQ_HANDLED; } +static irqreturn_t wm8994_temp_warn(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_err(codec->dev, "Thermal warning\n"); + + return IRQ_HANDLED; +} + +static irqreturn_t wm8994_temp_shut(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_crit(codec->dev, "Thermal shutdown\n"); + + return IRQ_HANDLED; +} + static int wm8994_codec_probe(struct snd_soc_codec *codec) { struct wm8994 *control; @@ -3123,6 +3141,10 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, wm8994_fifo_error, "FIFO error", codec); + wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, + wm8994_temp_warn, "Thermal warning", codec); + wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, + wm8994_temp_shut, "Thermal shutdown", codec); ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, wm_hubs_dcs_done, "DC servo done", @@ -3387,6 +3409,8 @@ err_irq: wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec); err: kfree(wm8994); return ret; @@ -3409,6 +3433,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec); switch (control->type) { case WM8994: -- cgit v1.2.3 From ddd7a26094c93a950f4b2e6b4d5865c93976372e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 15 Aug 2011 20:15:22 +0200 Subject: ASoC: Add ADAU1373 codec support This patch adds support for the Analog Devices ADAU1373 audio codec. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adau1373.c | 1414 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/adau1373.h | 29 + 4 files changed, 1449 insertions(+) create mode 100644 sound/soc/codecs/adau1373.c create mode 100644 sound/soc/codecs/adau1373.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 665d9240c4ae..71b46c8f70d7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 + select SND_SOC_ADAU1373 if I2C select SND_SOC_ADAV80X select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER @@ -139,6 +140,9 @@ config SND_SOC_ADAU1701 select SIGMA tristate +config SND_SOC_ADAU1373 + tristate + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5119a7e2c1a8..70c1769acd15 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -5,6 +5,7 @@ snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o +snd-soc-adau1373-objs := adau1373.o snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o @@ -100,6 +101,7 @@ obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c new file mode 100644 index 000000000000..2aa40c3731d0 --- /dev/null +++ b/sound/soc/codecs/adau1373.c @@ -0,0 +1,1414 @@ +/* + * Analog Devices ADAU1373 Audio Codec drive + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2 or later. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#include "adau1373.h" + +struct adau1373_dai { + unsigned int clk_src; + unsigned int sysclk; + bool enable_src; + bool master; +}; + +struct adau1373 { + struct adau1373_dai dais[3]; +}; + +#define ADAU1373_INPUT_MODE 0x00 +#define ADAU1373_AINL_CTRL(x) (0x01 + (x) * 2) +#define ADAU1373_AINR_CTRL(x) (0x02 + (x) * 2) +#define ADAU1373_LLINE_OUT(x) (0x9 + (x) * 2) +#define ADAU1373_RLINE_OUT(x) (0xa + (x) * 2) +#define ADAU1373_LSPK_OUT 0x0d +#define ADAU1373_RSPK_OUT 0x0e +#define ADAU1373_LHP_OUT 0x0f +#define ADAU1373_RHP_OUT 0x10 +#define ADAU1373_ADC_GAIN 0x11 +#define ADAU1373_LADC_MIXER 0x12 +#define ADAU1373_RADC_MIXER 0x13 +#define ADAU1373_LLINE1_MIX 0x14 +#define ADAU1373_RLINE1_MIX 0x15 +#define ADAU1373_LLINE2_MIX 0x16 +#define ADAU1373_RLINE2_MIX 0x17 +#define ADAU1373_LSPK_MIX 0x18 +#define ADAU1373_RSPK_MIX 0x19 +#define ADAU1373_LHP_MIX 0x1a +#define ADAU1373_RHP_MIX 0x1b +#define ADAU1373_EP_MIX 0x1c +#define ADAU1373_HP_CTRL 0x1d +#define ADAU1373_HP_CTRL2 0x1e +#define ADAU1373_LS_CTRL 0x1f +#define ADAU1373_EP_CTRL 0x21 +#define ADAU1373_MICBIAS_CTRL1 0x22 +#define ADAU1373_MICBIAS_CTRL2 0x23 +#define ADAU1373_OUTPUT_CTRL 0x24 +#define ADAU1373_PWDN_CTRL1 0x25 +#define ADAU1373_PWDN_CTRL2 0x26 +#define ADAU1373_PWDN_CTRL3 0x27 +#define ADAU1373_DPLL_CTRL(x) (0x28 + (x) * 7) +#define ADAU1373_PLL_CTRL1(x) (0x29 + (x) * 7) +#define ADAU1373_PLL_CTRL2(x) (0x2a + (x) * 7) +#define ADAU1373_PLL_CTRL3(x) (0x2b + (x) * 7) +#define ADAU1373_PLL_CTRL4(x) (0x2c + (x) * 7) +#define ADAU1373_PLL_CTRL5(x) (0x2d + (x) * 7) +#define ADAU1373_PLL_CTRL6(x) (0x2e + (x) * 7) +#define ADAU1373_PLL_CTRL7(x) (0x2f + (x) * 7) +#define ADAU1373_HEADDECT 0x36 +#define ADAU1373_ADC_DAC_STATUS 0x37 +#define ADAU1373_ADC_CTRL 0x3c +#define ADAU1373_DAI(x) (0x44 + (x)) +#define ADAU1373_CLK_SRC_DIV(x) (0x40 + (x) * 2) +#define ADAU1373_BCLKDIV(x) (0x47 + (x)) +#define ADAU1373_SRC_RATIOA(x) (0x4a + (x) * 2) +#define ADAU1373_SRC_RATIOB(x) (0x4b + (x) * 2) +#define ADAU1373_DEEMP_CTRL 0x50 +#define ADAU1373_SRC_DAI_CTRL(x) (0x51 + (x)) +#define ADAU1373_DIN_MIX_CTRL(x) (0x56 + (x)) +#define ADAU1373_DOUT_MIX_CTRL(x) (0x5b + (x)) +#define ADAU1373_DAI_PBL_VOL(x) (0x62 + (x) * 2) +#define ADAU1373_DAI_PBR_VOL(x) (0x63 + (x) * 2) +#define ADAU1373_DAI_RECL_VOL(x) (0x68 + (x) * 2) +#define ADAU1373_DAI_RECR_VOL(x) (0x69 + (x) * 2) +#define ADAU1373_DAC1_PBL_VOL 0x6e +#define ADAU1373_DAC1_PBR_VOL 0x6f +#define ADAU1373_DAC2_PBL_VOL 0x70 +#define ADAU1373_DAC2_PBR_VOL 0x71 +#define ADAU1373_ADC_RECL_VOL 0x72 +#define ADAU1373_ADC_RECR_VOL 0x73 +#define ADAU1373_DMIC_RECL_VOL 0x74 +#define ADAU1373_DMIC_RECR_VOL 0x75 +#define ADAU1373_VOL_GAIN1 0x76 +#define ADAU1373_VOL_GAIN2 0x77 +#define ADAU1373_VOL_GAIN3 0x78 +#define ADAU1373_HPF_CTRL 0x7d +#define ADAU1373_BASS1 0x7e +#define ADAU1373_BASS2 0x7f +#define ADAU1373_DRC(x) (0x80 + (x) * 0x10) +#define ADAU1373_3D_CTRL1 0xc0 +#define ADAU1373_3D_CTRL2 0xc1 +#define ADAU1373_FDSP_SEL1 0xdc +#define ADAU1373_FDSP_SEL2 0xdd +#define ADAU1373_FDSP_SEL3 0xde +#define ADAU1373_FDSP_SEL4 0xdf +#define ADAU1373_DIGMICCTRL 0xe2 +#define ADAU1373_DIGEN 0xeb +#define ADAU1373_SOFT_RESET 0xff + + +#define ADAU1373_PLL_CTRL6_DPLL_BYPASS BIT(1) +#define ADAU1373_PLL_CTRL6_PLL_EN BIT(0) + +#define ADAU1373_DAI_INVERT_BCLK BIT(7) +#define ADAU1373_DAI_MASTER BIT(6) +#define ADAU1373_DAI_INVERT_LRCLK BIT(4) +#define ADAU1373_DAI_WLEN_16 0x0 +#define ADAU1373_DAI_WLEN_20 0x4 +#define ADAU1373_DAI_WLEN_24 0x8 +#define ADAU1373_DAI_WLEN_32 0xc +#define ADAU1373_DAI_WLEN_MASK 0xc +#define ADAU1373_DAI_FORMAT_RIGHT_J 0x0 +#define ADAU1373_DAI_FORMAT_LEFT_J 0x1 +#define ADAU1373_DAI_FORMAT_I2S 0x2 +#define ADAU1373_DAI_FORMAT_DSP 0x3 + +#define ADAU1373_BCLKDIV_SOURCE BIT(5) +#define ADAU1373_BCLKDIV_32 0x03 +#define ADAU1373_BCLKDIV_64 0x02 +#define ADAU1373_BCLKDIV_128 0x01 +#define ADAU1373_BCLKDIV_256 0x00 + +#define ADAU1373_ADC_CTRL_PEAK_DETECT BIT(0) +#define ADAU1373_ADC_CTRL_RESET BIT(1) +#define ADAU1373_ADC_CTRL_RESET_FORCE BIT(2) + +#define ADAU1373_OUTPUT_CTRL_LDIFF BIT(3) +#define ADAU1373_OUTPUT_CTRL_LNFBEN BIT(2) + +#define ADAU1373_PWDN_CTRL3_PWR_EN BIT(0) + +#define ADAU1373_EP_CTRL_MICBIAS1_OFFSET 4 +#define ADAU1373_EP_CTRL_MICBIAS2_OFFSET 2 + +static const uint8_t adau1373_default_regs[] = { + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x00 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x10 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x20 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, /* 0x30 */ + 0x00, 0x00, 0x00, 0x80, 0x00, 0x01, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x0a, 0x0a, 0x0a, 0x00, /* 0x40 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x08, 0x08, 0x08, 0x00, 0x00, 0x00, 0x00, /* 0x50 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x60 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x70 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x80 */ + 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, + 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x90 */ + 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, + 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0xa0 */ + 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, + 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff, 0xff, /* 0xb0 */ + 0xff, 0xff, 0xff, 0xff, 0xff, 0x1f, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xc0 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xd0 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 0xe0 */ + 0x00, 0x1f, 0x0f, 0x00, 0x00, +}; + +static const unsigned int adau1373_out_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 7, TLV_DB_SCALE_ITEM(-7900, 400, 1), + 8, 15, TLV_DB_SCALE_ITEM(-4700, 300, 0), + 16, 23, TLV_DB_SCALE_ITEM(-2300, 200, 0), + 24, 31, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const DECLARE_TLV_DB_MINMAX(adau1373_digital_tlv, -9563, 0); +static const DECLARE_TLV_DB_SCALE(adau1373_in_pga_tlv, -1300, 100, 1); +static const DECLARE_TLV_DB_SCALE(adau1373_ep_tlv, -600, 600, 1); + +static const DECLARE_TLV_DB_SCALE(adau1373_input_boost_tlv, 0, 2000, 0); +static const DECLARE_TLV_DB_SCALE(adau1373_gain_boost_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adau1373_speaker_boost_tlv, 1200, 600, 0); + +static const char *adau1373_fdsp_sel_text[] = { + "None", + "Channel 1", + "Channel 2", + "Channel 3", + "Channel 4", + "Channel 5", +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_drc1_channel_enum, + ADAU1373_FDSP_SEL1, 4, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_drc2_channel_enum, + ADAU1373_FDSP_SEL1, 0, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_drc3_channel_enum, + ADAU1373_FDSP_SEL2, 0, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_channel_enum, + ADAU1373_FDSP_SEL3, 0, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_bass_channel_enum, + ADAU1373_FDSP_SEL4, 4, adau1373_fdsp_sel_text); + +static const char *adau1373_hpf_cutoff_text[] = { + "3.7Hz", "50Hz", "100Hz", "150Hz", "200Hz", "250Hz", "300Hz", "350Hz", + "400Hz", "450Hz", "500Hz", "550Hz", "600Hz", "650Hz", "700Hz", "750Hz", + "800Hz", +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_cutoff_enum, + ADAU1373_HPF_CTRL, 3, adau1373_hpf_cutoff_text); + +static const char *adau1373_bass_lpf_cutoff_text[] = { + "801Hz", "1001Hz", +}; + +static const char *adau1373_bass_clip_level_text[] = { + "0.125", "0.250", "0.370", "0.500", "0.625", "0.750", "0.875", +}; + +static const unsigned int adau1373_bass_clip_level_values[] = { + 1, 2, 3, 4, 5, 6, 7, +}; + +static const char *adau1373_bass_hpf_cutoff_text[] = { + "158Hz", "232Hz", "347Hz", "520Hz", +}; + +static const unsigned int adau1373_bass_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 2, TLV_DB_SCALE_ITEM(-600, 600, 1), + 3, 4, TLV_DB_SCALE_ITEM(950, 250, 0), + 5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0), +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_bass_lpf_cutoff_enum, + ADAU1373_BASS1, 5, adau1373_bass_lpf_cutoff_text); + +static const SOC_VALUE_ENUM_SINGLE_DECL(adau1373_bass_clip_level_enum, + ADAU1373_BASS1, 2, 7, adau1373_bass_clip_level_text, + adau1373_bass_clip_level_values); + +static const SOC_ENUM_SINGLE_DECL(adau1373_bass_hpf_cutoff_enum, + ADAU1373_BASS1, 0, adau1373_bass_hpf_cutoff_text); + +static const char *adau1373_3d_level_text[] = { + "0%", "6.67%", "13.33%", "20%", "26.67%", "33.33%", + "40%", "46.67%", "53.33%", "60%", "66.67%", "73.33%", + "80%", "86.67", "99.33%", "100%" +}; + +static const char *adau1373_3d_cutoff_text[] = { + "No 3D", "0.03125 fs", "0.04583 fs", "0.075 fs", "0.11458 fs", + "0.16875 fs", "0.27083 fs" +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_3d_level_enum, + ADAU1373_3D_CTRL1, 4, adau1373_3d_level_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_3d_cutoff_enum, + ADAU1373_3D_CTRL1, 0, adau1373_3d_cutoff_text); + +static const unsigned int adau1373_3d_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_LINEAR_ITEM(-1800, -120), +}; + +static const char *adau1373_lr_mux_text[] = { + "Mute", + "Right Channel (L+R)", + "Left Channel (L+R)", + "Stereo", +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_lineout1_lr_mux_enum, + ADAU1373_OUTPUT_CTRL, 4, adau1373_lr_mux_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_lineout2_lr_mux_enum, + ADAU1373_OUTPUT_CTRL, 6, adau1373_lr_mux_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_speaker_lr_mux_enum, + ADAU1373_LS_CTRL, 4, adau1373_lr_mux_text); + +static const struct snd_kcontrol_new adau1373_controls[] = { + SOC_DOUBLE_R_TLV("AIF1 Capture Volume", ADAU1373_DAI_RECL_VOL(0), + ADAU1373_DAI_RECR_VOL(0), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF2 Capture Volume", ADAU1373_DAI_RECL_VOL(1), + ADAU1373_DAI_RECR_VOL(1), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF3 Capture Volume", ADAU1373_DAI_RECL_VOL(2), + ADAU1373_DAI_RECR_VOL(2), 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ADAU1373_ADC_RECL_VOL, + ADAU1373_ADC_RECR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("DMIC Capture Volume", ADAU1373_DMIC_RECL_VOL, + ADAU1373_DMIC_RECR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("AIF1 Playback Volume", ADAU1373_DAI_PBL_VOL(0), + ADAU1373_DAI_PBR_VOL(0), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF2 Playback Volume", ADAU1373_DAI_PBL_VOL(1), + ADAU1373_DAI_PBR_VOL(1), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF3 Playback Volume", ADAU1373_DAI_PBL_VOL(2), + ADAU1373_DAI_PBR_VOL(2), 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", ADAU1373_DAC1_PBL_VOL, + ADAU1373_DAC1_PBR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", ADAU1373_DAC2_PBL_VOL, + ADAU1373_DAC2_PBR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("Lineout1 Playback Volume", ADAU1373_LLINE_OUT(0), + ADAU1373_RLINE_OUT(0), 0, 0x1f, 0, adau1373_out_tlv), + SOC_DOUBLE_R_TLV("Speaker Playback Volume", ADAU1373_LSPK_OUT, + ADAU1373_RSPK_OUT, 0, 0x1f, 0, adau1373_out_tlv), + SOC_DOUBLE_R_TLV("Headphone Playback Volume", ADAU1373_LHP_OUT, + ADAU1373_RHP_OUT, 0, 0x1f, 0, adau1373_out_tlv), + + SOC_DOUBLE_R_TLV("Input 1 Capture Volume", ADAU1373_AINL_CTRL(0), + ADAU1373_AINR_CTRL(0), 0, 0x1f, 0, adau1373_in_pga_tlv), + SOC_DOUBLE_R_TLV("Input 2 Capture Volume", ADAU1373_AINL_CTRL(1), + ADAU1373_AINR_CTRL(1), 0, 0x1f, 0, adau1373_in_pga_tlv), + SOC_DOUBLE_R_TLV("Input 3 Capture Volume", ADAU1373_AINL_CTRL(2), + ADAU1373_AINR_CTRL(2), 0, 0x1f, 0, adau1373_in_pga_tlv), + SOC_DOUBLE_R_TLV("Input 4 Capture Volume", ADAU1373_AINL_CTRL(3), + ADAU1373_AINR_CTRL(3), 0, 0x1f, 0, adau1373_in_pga_tlv), + + SOC_SINGLE_TLV("Earpiece Playback Volume", ADAU1373_EP_CTRL, 0, 3, 0, + adau1373_ep_tlv), + + SOC_DOUBLE_TLV("AIF3 Boost Playback Volume", ADAU1373_VOL_GAIN1, 4, 5, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF2 Boost Playback Volume", ADAU1373_VOL_GAIN1, 2, 3, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF1 Boost Playback Volume", ADAU1373_VOL_GAIN1, 0, 1, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF3 Boost Capture Volume", ADAU1373_VOL_GAIN2, 4, 5, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF2 Boost Capture Volume", ADAU1373_VOL_GAIN2, 2, 3, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF1 Boost Capture Volume", ADAU1373_VOL_GAIN2, 0, 1, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("DMIC Boost Capture Volume", ADAU1373_VOL_GAIN3, 6, 7, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("ADC Boost Capture Volume", ADAU1373_VOL_GAIN3, 4, 5, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("DAC2 Boost Playback Volume", ADAU1373_VOL_GAIN3, 2, 3, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("DAC1 Boost Playback Volume", ADAU1373_VOL_GAIN3, 0, 1, + 1, 0, adau1373_gain_boost_tlv), + + SOC_DOUBLE_TLV("Input 1 Boost Capture Volume", ADAU1373_ADC_GAIN, 0, 4, + 1, 0, adau1373_input_boost_tlv), + SOC_DOUBLE_TLV("Input 2 Boost Capture Volume", ADAU1373_ADC_GAIN, 1, 5, + 1, 0, adau1373_input_boost_tlv), + SOC_DOUBLE_TLV("Input 3 Boost Capture Volume", ADAU1373_ADC_GAIN, 2, 6, + 1, 0, adau1373_input_boost_tlv), + SOC_DOUBLE_TLV("Input 4 Boost Capture Volume", ADAU1373_ADC_GAIN, 3, 7, + 1, 0, adau1373_input_boost_tlv), + + SOC_DOUBLE_TLV("Speaker Boost Playback Volume", ADAU1373_LS_CTRL, 2, 3, + 1, 0, adau1373_speaker_boost_tlv), + + SOC_ENUM("Lineout1 LR Mux", adau1373_lineout1_lr_mux_enum), + SOC_ENUM("Speaker LR Mux", adau1373_speaker_lr_mux_enum), + + SOC_ENUM("HPF Cutoff", adau1373_hpf_cutoff_enum), + SOC_DOUBLE("HPF Switch", ADAU1373_HPF_CTRL, 1, 0, 1, 0), + SOC_ENUM("HPF Channel", adau1373_hpf_channel_enum), + + SOC_ENUM("Bass HPF Cutoff", adau1373_bass_hpf_cutoff_enum), + SOC_VALUE_ENUM("Bass Clip Level Threshold", + adau1373_bass_clip_level_enum), + SOC_ENUM("Bass LPF Cutoff", adau1373_bass_lpf_cutoff_enum), + SOC_DOUBLE("Bass Playback Switch", ADAU1373_BASS2, 0, 1, 1, 0), + SOC_SINGLE_TLV("Bass Playback Volume", ADAU1373_BASS2, 2, 7, 0, + adau1373_bass_tlv), + SOC_ENUM("Bass Channel", adau1373_bass_channel_enum), + + SOC_ENUM("3D Freq", adau1373_3d_cutoff_enum), + SOC_ENUM("3D Level", adau1373_3d_level_enum), + SOC_SINGLE("3D Playback Switch", ADAU1373_3D_CTRL2, 0, 1, 0), + SOC_SINGLE_TLV("3D Playback Volume", ADAU1373_3D_CTRL2, 2, 7, 0, + adau1373_3d_tlv), + SOC_ENUM("3D Channel", adau1373_bass_channel_enum), + + SOC_SINGLE("Zero Cross Switch", ADAU1373_PWDN_CTRL3, 7, 1, 0), +}; + +static const struct snd_kcontrol_new adau1373_lineout2_controls[] = { + SOC_DOUBLE_R_TLV("Lineout2 Playback Volume", ADAU1373_LLINE_OUT(1), + ADAU1373_RLINE_OUT(1), 0, 0x1f, 0, adau1373_out_tlv), + SOC_ENUM("Lineout2 LR Mux", adau1373_lineout2_lr_mux_enum), +}; + +static const struct snd_kcontrol_new adau1373_drc_controls[] = { + SOC_ENUM("DRC1 Channel", adau1373_drc1_channel_enum), + SOC_ENUM("DRC2 Channel", adau1373_drc2_channel_enum), + SOC_ENUM("DRC3 Channel", adau1373_drc3_channel_enum), +}; + +static int adau1373_pll_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int pll_id = w->name[3] - '1'; + unsigned int val; + + if (SND_SOC_DAPM_EVENT_ON(event)) + val = ADAU1373_PLL_CTRL6_PLL_EN; + else + val = 0; + + snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + ADAU1373_PLL_CTRL6_PLL_EN, val); + + if (SND_SOC_DAPM_EVENT_ON(event)) + mdelay(5); + + return 0; +} + +static const char *adau1373_decimator_text[] = { + "ADC", + "DMIC1", +}; + +static const struct soc_enum adau1373_decimator_enum = + SOC_ENUM_SINGLE(0, 0, 2, adau1373_decimator_text); + +static const struct snd_kcontrol_new adau1373_decimator_mux = + SOC_DAPM_ENUM_VIRT("Decimator Mux", adau1373_decimator_enum); + +static const struct snd_kcontrol_new adau1373_left_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC1 Switch", ADAU1373_LADC_MIXER, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Switch", ADAU1373_LADC_MIXER, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Switch", ADAU1373_LADC_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Switch", ADAU1373_LADC_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Switch", ADAU1373_LADC_MIXER, 0, 1, 0), +}; + +static const struct snd_kcontrol_new adau1373_right_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC1 Switch", ADAU1373_RADC_MIXER, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Switch", ADAU1373_RADC_MIXER, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Switch", ADAU1373_RADC_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Switch", ADAU1373_RADC_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Switch", ADAU1373_RADC_MIXER, 0, 1, 0), +}; + +#define DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(_name, _reg) \ +const struct snd_kcontrol_new _name[] = { \ + SOC_DAPM_SINGLE("Left DAC2 Switch", _reg, 7, 1, 0), \ + SOC_DAPM_SINGLE("Right DAC2 Switch", _reg, 6, 1, 0), \ + SOC_DAPM_SINGLE("Left DAC1 Switch", _reg, 5, 1, 0), \ + SOC_DAPM_SINGLE("Right DAC1 Switch", _reg, 4, 1, 0), \ + SOC_DAPM_SINGLE("Input 4 Bypass Switch", _reg, 3, 1, 0), \ + SOC_DAPM_SINGLE("Input 3 Bypass Switch", _reg, 2, 1, 0), \ + SOC_DAPM_SINGLE("Input 2 Bypass Switch", _reg, 1, 1, 0), \ + SOC_DAPM_SINGLE("Input 1 Bypass Switch", _reg, 0, 1, 0), \ +} + +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_line1_mixer_controls, + ADAU1373_LLINE1_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_line1_mixer_controls, + ADAU1373_RLINE1_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_line2_mixer_controls, + ADAU1373_LLINE2_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_line2_mixer_controls, + ADAU1373_RLINE2_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_spk_mixer_controls, + ADAU1373_LSPK_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_spk_mixer_controls, + ADAU1373_RSPK_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_ep_mixer_controls, + ADAU1373_EP_MIX); + +static const struct snd_kcontrol_new adau1373_left_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", ADAU1373_LHP_MIX, 5, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", ADAU1373_LHP_MIX, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Bypass Switch", ADAU1373_LHP_MIX, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Bypass Switch", ADAU1373_LHP_MIX, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Bypass Switch", ADAU1373_LHP_MIX, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Bypass Switch", ADAU1373_LHP_MIX, 0, 1, 0), +}; + +static const struct snd_kcontrol_new adau1373_right_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Right DAC1 Switch", ADAU1373_RHP_MIX, 5, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", ADAU1373_RHP_MIX, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Bypass Switch", ADAU1373_RHP_MIX, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Bypass Switch", ADAU1373_RHP_MIX, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Bypass Switch", ADAU1373_RHP_MIX, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Bypass Switch", ADAU1373_RHP_MIX, 0, 1, 0), +}; + +#define DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(_name, _reg) \ +const struct snd_kcontrol_new _name[] = { \ + SOC_DAPM_SINGLE("DMIC2 Swapped Switch", _reg, 6, 1, 0), \ + SOC_DAPM_SINGLE("DMIC2 Switch", _reg, 5, 1, 0), \ + SOC_DAPM_SINGLE("ADC/DMIC1 Swapped Switch", _reg, 4, 1, 0), \ + SOC_DAPM_SINGLE("ADC/DMIC1 Switch", _reg, 3, 1, 0), \ + SOC_DAPM_SINGLE("AIF3 Switch", _reg, 2, 1, 0), \ + SOC_DAPM_SINGLE("AIF2 Switch", _reg, 1, 1, 0), \ + SOC_DAPM_SINGLE("AIF1 Switch", _reg, 0, 1, 0), \ +} + +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel1_mixer_controls, + ADAU1373_DIN_MIX_CTRL(0)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel2_mixer_controls, + ADAU1373_DIN_MIX_CTRL(1)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel3_mixer_controls, + ADAU1373_DIN_MIX_CTRL(2)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel4_mixer_controls, + ADAU1373_DIN_MIX_CTRL(3)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel5_mixer_controls, + ADAU1373_DIN_MIX_CTRL(4)); + +#define DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(_name, _reg) \ +const struct snd_kcontrol_new _name[] = { \ + SOC_DAPM_SINGLE("DSP Channel5 Switch", _reg, 4, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel4 Switch", _reg, 3, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel3 Switch", _reg, 2, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel2 Switch", _reg, 1, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel1 Switch", _reg, 0, 1, 0), \ +} + +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif1_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(0)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif2_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(1)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif3_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(2)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_dac1_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(3)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_dac2_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(4)); + +static const struct snd_soc_dapm_widget adau1373_dapm_widgets[] = { + /* Datasheet claims Left ADC is bit 6 and Right ADC is bit 7, but that + * doesn't seem to be the case. */ + SND_SOC_DAPM_ADC("Left ADC", NULL, ADAU1373_PWDN_CTRL1, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", NULL, ADAU1373_PWDN_CTRL1, 6, 0), + + SND_SOC_DAPM_ADC("DMIC1", NULL, ADAU1373_DIGMICCTRL, 0, 0), + SND_SOC_DAPM_ADC("DMIC2", NULL, ADAU1373_DIGMICCTRL, 2, 0), + + SND_SOC_DAPM_VIRT_MUX("Decimator Mux", SND_SOC_NOPM, 0, 0, + &adau1373_decimator_mux), + + SND_SOC_DAPM_SUPPLY("MICBIAS2", ADAU1373_PWDN_CTRL1, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", ADAU1373_PWDN_CTRL1, 4, 0, NULL, 0), + + SND_SOC_DAPM_PGA("IN4PGA", ADAU1373_PWDN_CTRL1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN3PGA", ADAU1373_PWDN_CTRL1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN2PGA", ADAU1373_PWDN_CTRL1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN1PGA", ADAU1373_PWDN_CTRL1, 0, 0, NULL, 0), + + SND_SOC_DAPM_DAC("Left DAC2", NULL, ADAU1373_PWDN_CTRL2, 7, 0), + SND_SOC_DAPM_DAC("Right DAC2", NULL, ADAU1373_PWDN_CTRL2, 6, 0), + SND_SOC_DAPM_DAC("Left DAC1", NULL, ADAU1373_PWDN_CTRL2, 5, 0), + SND_SOC_DAPM_DAC("Right DAC1", NULL, ADAU1373_PWDN_CTRL2, 4, 0), + + SOC_MIXER_ARRAY("Left ADC Mixer", SND_SOC_NOPM, 0, 0, + adau1373_left_adc_mixer_controls), + SOC_MIXER_ARRAY("Right ADC Mixer", SND_SOC_NOPM, 0, 0, + adau1373_right_adc_mixer_controls), + + SOC_MIXER_ARRAY("Left Lineout2 Mixer", ADAU1373_PWDN_CTRL2, 3, 0, + adau1373_left_line2_mixer_controls), + SOC_MIXER_ARRAY("Right Lineout2 Mixer", ADAU1373_PWDN_CTRL2, 2, 0, + adau1373_right_line2_mixer_controls), + SOC_MIXER_ARRAY("Left Lineout1 Mixer", ADAU1373_PWDN_CTRL2, 1, 0, + adau1373_left_line1_mixer_controls), + SOC_MIXER_ARRAY("Right Lineout1 Mixer", ADAU1373_PWDN_CTRL2, 0, 0, + adau1373_right_line1_mixer_controls), + + SOC_MIXER_ARRAY("Earpiece Mixer", ADAU1373_PWDN_CTRL3, 4, 0, + adau1373_ep_mixer_controls), + SOC_MIXER_ARRAY("Left Speaker Mixer", ADAU1373_PWDN_CTRL3, 3, 0, + adau1373_left_spk_mixer_controls), + SOC_MIXER_ARRAY("Right Speaker Mixer", ADAU1373_PWDN_CTRL3, 2, 0, + adau1373_right_spk_mixer_controls), + SOC_MIXER_ARRAY("Left Headphone Mixer", SND_SOC_NOPM, 0, 0, + adau1373_left_hp_mixer_controls), + SOC_MIXER_ARRAY("Right Headphone Mixer", SND_SOC_NOPM, 0, 0, + adau1373_right_hp_mixer_controls), + SND_SOC_DAPM_SUPPLY("Headphone Enable", ADAU1373_PWDN_CTRL3, 1, 0, + NULL, 0), + + SND_SOC_DAPM_SUPPLY("AIF1 CLK", ADAU1373_SRC_DAI_CTRL(0), 0, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF2 CLK", ADAU1373_SRC_DAI_CTRL(1), 0, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF3 CLK", ADAU1373_SRC_DAI_CTRL(2), 0, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF1 IN SRC", ADAU1373_SRC_DAI_CTRL(0), 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF1 OUT SRC", ADAU1373_SRC_DAI_CTRL(0), 1, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF2 IN SRC", ADAU1373_SRC_DAI_CTRL(1), 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF2 OUT SRC", ADAU1373_SRC_DAI_CTRL(1), 1, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF3 IN SRC", ADAU1373_SRC_DAI_CTRL(2), 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF3 OUT SRC", ADAU1373_SRC_DAI_CTRL(2), 1, 0, + NULL, 0), + + SND_SOC_DAPM_AIF_IN("AIF1 IN", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1 OUT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2 IN", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2 OUT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3 IN", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3 OUT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), + + SOC_MIXER_ARRAY("DSP Channel1 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel1_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel2 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel2_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel3 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel3_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel4 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel4_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel5 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel5_mixer_controls), + + SOC_MIXER_ARRAY("AIF1 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_aif1_mixer_controls), + SOC_MIXER_ARRAY("AIF2 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_aif2_mixer_controls), + SOC_MIXER_ARRAY("AIF3 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_aif3_mixer_controls), + SOC_MIXER_ARRAY("DAC1 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dac1_mixer_controls), + SOC_MIXER_ARRAY("DAC2 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dac2_mixer_controls), + + SND_SOC_DAPM_SUPPLY("DSP", ADAU1373_DIGEN, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Recording Engine B", ADAU1373_DIGEN, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Recording Engine A", ADAU1373_DIGEN, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Playback Engine B", ADAU1373_DIGEN, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Playback Engine A", ADAU1373_DIGEN, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("PLL1", SND_SOC_NOPM, 0, 0, adau1373_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("PLL2", SND_SOC_NOPM, 0, 0, adau1373_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("SYSCLK1", ADAU1373_CLK_SRC_DIV(0), 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SYSCLK2", ADAU1373_CLK_SRC_DIV(1), 7, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + SND_SOC_DAPM_INPUT("AIN3L"), + SND_SOC_DAPM_INPUT("AIN3R"), + SND_SOC_DAPM_INPUT("AIN4L"), + SND_SOC_DAPM_INPUT("AIN4R"), + + SND_SOC_DAPM_INPUT("DMIC1DAT"), + SND_SOC_DAPM_INPUT("DMIC2DAT"), + + SND_SOC_DAPM_OUTPUT("LOUT1L"), + SND_SOC_DAPM_OUTPUT("LOUT1R"), + SND_SOC_DAPM_OUTPUT("LOUT2L"), + SND_SOC_DAPM_OUTPUT("LOUT2R"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPKL"), + SND_SOC_DAPM_OUTPUT("SPKR"), + SND_SOC_DAPM_OUTPUT("EP"), +}; + +static int adau1373_check_aif_clk(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + unsigned int dai; + const char *clk; + + dai = sink->name[3] - '1'; + + if (!adau1373->dais[dai].master) + return 0; + + if (adau1373->dais[dai].clk_src == ADAU1373_CLK_SRC_PLL1) + clk = "SYSCLK1"; + else + clk = "SYSCLK2"; + + return strcmp(source->name, clk) == 0; +} + +static int adau1373_check_src(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + unsigned int dai; + + dai = sink->name[3] - '1'; + + return adau1373->dais[dai].enable_src; +} + +#define DSP_CHANNEL_MIXER_ROUTES(_sink) \ + { _sink, "DMIC2 Swapped Switch", "DMIC2" }, \ + { _sink, "DMIC2 Switch", "DMIC2" }, \ + { _sink, "ADC/DMIC1 Swapped Switch", "Decimator Mux" }, \ + { _sink, "ADC/DMIC1 Switch", "Decimator Mux" }, \ + { _sink, "AIF1 Switch", "AIF1 IN" }, \ + { _sink, "AIF2 Switch", "AIF2 IN" }, \ + { _sink, "AIF3 Switch", "AIF3 IN" } + +#define DSP_OUTPUT_MIXER_ROUTES(_sink) \ + { _sink, "DSP Channel1 Switch", "DSP Channel1 Mixer" }, \ + { _sink, "DSP Channel2 Switch", "DSP Channel2 Mixer" }, \ + { _sink, "DSP Channel3 Switch", "DSP Channel3 Mixer" }, \ + { _sink, "DSP Channel4 Switch", "DSP Channel4 Mixer" }, \ + { _sink, "DSP Channel5 Switch", "DSP Channel5 Mixer" } + +#define LEFT_OUTPUT_MIXER_ROUTES(_sink) \ + { _sink, "Right DAC2 Switch", "Right DAC2" }, \ + { _sink, "Left DAC2 Switch", "Left DAC2" }, \ + { _sink, "Right DAC1 Switch", "Right DAC1" }, \ + { _sink, "Left DAC1 Switch", "Left DAC1" }, \ + { _sink, "Input 1 Bypass Switch", "IN1PGA" }, \ + { _sink, "Input 2 Bypass Switch", "IN2PGA" }, \ + { _sink, "Input 3 Bypass Switch", "IN3PGA" }, \ + { _sink, "Input 4 Bypass Switch", "IN4PGA" } + +#define RIGHT_OUTPUT_MIXER_ROUTES(_sink) \ + { _sink, "Right DAC2 Switch", "Right DAC2" }, \ + { _sink, "Left DAC2 Switch", "Left DAC2" }, \ + { _sink, "Right DAC1 Switch", "Right DAC1" }, \ + { _sink, "Left DAC1 Switch", "Left DAC1" }, \ + { _sink, "Input 1 Bypass Switch", "IN1PGA" }, \ + { _sink, "Input 2 Bypass Switch", "IN2PGA" }, \ + { _sink, "Input 3 Bypass Switch", "IN3PGA" }, \ + { _sink, "Input 4 Bypass Switch", "IN4PGA" } + +static const struct snd_soc_dapm_route adau1373_dapm_routes[] = { + { "Left ADC Mixer", "DAC1 Switch", "Left DAC1" }, + { "Left ADC Mixer", "Input 1 Switch", "IN1PGA" }, + { "Left ADC Mixer", "Input 2 Switch", "IN2PGA" }, + { "Left ADC Mixer", "Input 3 Switch", "IN3PGA" }, + { "Left ADC Mixer", "Input 4 Switch", "IN4PGA" }, + + { "Right ADC Mixer", "DAC1 Switch", "Right DAC1" }, + { "Right ADC Mixer", "Input 1 Switch", "IN1PGA" }, + { "Right ADC Mixer", "Input 2 Switch", "IN2PGA" }, + { "Right ADC Mixer", "Input 3 Switch", "IN3PGA" }, + { "Right ADC Mixer", "Input 4 Switch", "IN4PGA" }, + + { "Left ADC", NULL, "Left ADC Mixer" }, + { "Right ADC", NULL, "Right ADC Mixer" }, + + { "Decimator Mux", "ADC", "Left ADC" }, + { "Decimator Mux", "ADC", "Right ADC" }, + { "Decimator Mux", "DMIC1", "DMIC1" }, + + DSP_CHANNEL_MIXER_ROUTES("DSP Channel1 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel2 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel3 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel4 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel5 Mixer"), + + DSP_OUTPUT_MIXER_ROUTES("AIF1 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("AIF2 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("AIF3 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("DAC1 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("DAC2 Mixer"), + + { "AIF1 OUT", NULL, "AIF1 Mixer" }, + { "AIF2 OUT", NULL, "AIF2 Mixer" }, + { "AIF3 OUT", NULL, "AIF3 Mixer" }, + { "Left DAC1", NULL, "DAC1 Mixer" }, + { "Right DAC1", NULL, "DAC1 Mixer" }, + { "Left DAC2", NULL, "DAC2 Mixer" }, + { "Right DAC2", NULL, "DAC2 Mixer" }, + + LEFT_OUTPUT_MIXER_ROUTES("Left Lineout1 Mixer"), + RIGHT_OUTPUT_MIXER_ROUTES("Right Lineout1 Mixer"), + LEFT_OUTPUT_MIXER_ROUTES("Left Lineout2 Mixer"), + RIGHT_OUTPUT_MIXER_ROUTES("Right Lineout2 Mixer"), + LEFT_OUTPUT_MIXER_ROUTES("Left Speaker Mixer"), + RIGHT_OUTPUT_MIXER_ROUTES("Right Speaker Mixer"), + + { "Left Headphone Mixer", "Left DAC2 Switch", "Left DAC2" }, + { "Left Headphone Mixer", "Left DAC1 Switch", "Left DAC1" }, + { "Left Headphone Mixer", "Input 1 Bypass Switch", "IN1PGA" }, + { "Left Headphone Mixer", "Input 2 Bypass Switch", "IN2PGA" }, + { "Left Headphone Mixer", "Input 3 Bypass Switch", "IN3PGA" }, + { "Left Headphone Mixer", "Input 4 Bypass Switch", "IN4PGA" }, + { "Right Headphone Mixer", "Right DAC2 Switch", "Right DAC2" }, + { "Right Headphone Mixer", "Right DAC1 Switch", "Right DAC1" }, + { "Right Headphone Mixer", "Input 1 Bypass Switch", "IN1PGA" }, + { "Right Headphone Mixer", "Input 2 Bypass Switch", "IN2PGA" }, + { "Right Headphone Mixer", "Input 3 Bypass Switch", "IN3PGA" }, + { "Right Headphone Mixer", "Input 4 Bypass Switch", "IN4PGA" }, + + { "Left Headphone Mixer", NULL, "Headphone Enable" }, + { "Right Headphone Mixer", NULL, "Headphone Enable" }, + + { "Earpiece Mixer", "Right DAC2 Switch", "Right DAC2" }, + { "Earpiece Mixer", "Left DAC2 Switch", "Left DAC2" }, + { "Earpiece Mixer", "Right DAC1 Switch", "Right DAC1" }, + { "Earpiece Mixer", "Left DAC1 Switch", "Left DAC1" }, + { "Earpiece Mixer", "Input 1 Bypass Switch", "IN1PGA" }, + { "Earpiece Mixer", "Input 2 Bypass Switch", "IN2PGA" }, + { "Earpiece Mixer", "Input 3 Bypass Switch", "IN3PGA" }, + { "Earpiece Mixer", "Input 4 Bypass Switch", "IN4PGA" }, + + { "LOUT1L", NULL, "Left Lineout1 Mixer" }, + { "LOUT1R", NULL, "Right Lineout1 Mixer" }, + { "LOUT2L", NULL, "Left Lineout2 Mixer" }, + { "LOUT2R", NULL, "Right Lineout2 Mixer" }, + { "SPKL", NULL, "Left Speaker Mixer" }, + { "SPKR", NULL, "Right Speaker Mixer" }, + { "HPL", NULL, "Left Headphone Mixer" }, + { "HPR", NULL, "Right Headphone Mixer" }, + { "EP", NULL, "Earpiece Mixer" }, + + { "IN1PGA", NULL, "AIN1L" }, + { "IN2PGA", NULL, "AIN2L" }, + { "IN3PGA", NULL, "AIN3L" }, + { "IN4PGA", NULL, "AIN4L" }, + { "IN1PGA", NULL, "AIN1R" }, + { "IN2PGA", NULL, "AIN2R" }, + { "IN3PGA", NULL, "AIN3R" }, + { "IN4PGA", NULL, "AIN4R" }, + + { "SYSCLK1", NULL, "PLL1" }, + { "SYSCLK2", NULL, "PLL2" }, + + { "Left DAC1", NULL, "SYSCLK1" }, + { "Right DAC1", NULL, "SYSCLK1" }, + { "Left DAC2", NULL, "SYSCLK1" }, + { "Right DAC2", NULL, "SYSCLK1" }, + { "Left ADC", NULL, "SYSCLK1" }, + { "Right ADC", NULL, "SYSCLK1" }, + + { "DSP", NULL, "SYSCLK1" }, + + { "AIF1 Mixer", NULL, "DSP" }, + { "AIF2 Mixer", NULL, "DSP" }, + { "AIF3 Mixer", NULL, "DSP" }, + { "DAC1 Mixer", NULL, "DSP" }, + { "DAC2 Mixer", NULL, "DSP" }, + { "DAC1 Mixer", NULL, "Playback Engine A" }, + { "DAC2 Mixer", NULL, "Playback Engine B" }, + { "Left ADC Mixer", NULL, "Recording Engine A" }, + { "Right ADC Mixer", NULL, "Recording Engine A" }, + + { "AIF1 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk }, + { "AIF2 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk }, + { "AIF3 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk }, + { "AIF1 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk }, + { "AIF2 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk }, + { "AIF3 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk }, + + { "AIF1 IN", NULL, "AIF1 CLK" }, + { "AIF1 OUT", NULL, "AIF1 CLK" }, + { "AIF2 IN", NULL, "AIF2 CLK" }, + { "AIF2 OUT", NULL, "AIF2 CLK" }, + { "AIF3 IN", NULL, "AIF3 CLK" }, + { "AIF3 OUT", NULL, "AIF3 CLK" }, + { "AIF1 IN", NULL, "AIF1 IN SRC", adau1373_check_src }, + { "AIF1 OUT", NULL, "AIF1 OUT SRC", adau1373_check_src }, + { "AIF2 IN", NULL, "AIF2 IN SRC", adau1373_check_src }, + { "AIF2 OUT", NULL, "AIF2 OUT SRC", adau1373_check_src }, + { "AIF3 IN", NULL, "AIF3 IN SRC", adau1373_check_src }, + { "AIF3 OUT", NULL, "AIF3 OUT SRC", adau1373_check_src }, + + { "DMIC1", NULL, "DMIC1DAT" }, + { "DMIC1", NULL, "SYSCLK1" }, + { "DMIC1", NULL, "Recording Engine A" }, + { "DMIC2", NULL, "DMIC2DAT" }, + { "DMIC2", NULL, "SYSCLK1" }, + { "DMIC2", NULL, "Recording Engine B" }, +}; + +static int adau1373_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id]; + unsigned int div; + unsigned int freq; + unsigned int ctrl; + + freq = adau1373_dai->sysclk; + + if (freq % params_rate(params) != 0) + return -EINVAL; + + switch (freq / params_rate(params)) { + case 1024: /* sysclk / 256 */ + div = 0; + break; + case 1536: /* 2/3 sysclk / 256 */ + div = 1; + break; + case 2048: /* 1/2 sysclk / 256 */ + div = 2; + break; + case 3072: /* 1/3 sysclk / 256 */ + div = 3; + break; + case 4096: /* 1/4 sysclk / 256 */ + div = 4; + break; + case 6144: /* 1/6 sysclk / 256 */ + div = 5; + break; + case 5632: /* 2/11 sysclk / 256 */ + div = 6; + break; + default: + return -EINVAL; + } + + adau1373_dai->enable_src = (div != 0); + + snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), + ~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + ctrl = ADAU1373_DAI_WLEN_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + ctrl = ADAU1373_DAI_WLEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ctrl = ADAU1373_DAI_WLEN_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + ctrl = ADAU1373_DAI_WLEN_32; + break; + default: + return -EINVAL; + } + + return snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + ADAU1373_DAI_WLEN_MASK, ctrl); +} + +static int adau1373_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id]; + unsigned int ctrl; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl = ADAU1373_DAI_MASTER; + adau1373_dai->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl = 0; + adau1373_dai->master = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl |= ADAU1373_DAI_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl |= ADAU1373_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl |= ADAU1373_DAI_FORMAT_RIGHT_J; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl |= ADAU1373_DAI_FORMAT_DSP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl |= ADAU1373_DAI_INVERT_BCLK; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl |= ADAU1373_DAI_INVERT_LRCLK; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl |= ADAU1373_DAI_INVERT_LRCLK | ADAU1373_DAI_INVERT_BCLK; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + ~ADAU1373_DAI_WLEN_MASK, ctrl); + + return 0; +} + +static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(dai->codec); + struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id]; + + switch (clk_id) { + case ADAU1373_CLK_SRC_PLL1: + case ADAU1373_CLK_SRC_PLL2: + break; + default: + return -EINVAL; + } + + adau1373_dai->sysclk = freq; + adau1373_dai->clk_src = clk_id; + + snd_soc_update_bits(dai->codec, ADAU1373_BCLKDIV(dai->id), + ADAU1373_BCLKDIV_SOURCE, clk_id << 5); + + return 0; +} + +static const struct snd_soc_dai_ops adau1373_dai_ops = { + .hw_params = adau1373_hw_params, + .set_sysclk = adau1373_set_dai_sysclk, + .set_fmt = adau1373_set_dai_fmt, +}; + +#define ADAU1373_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver adau1373_dai_driver[] = { + { + .id = 0, + .name = "adau1373-aif1", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .ops = &adau1373_dai_ops, + .symmetric_rates = 1, + }, + { + .id = 1, + .name = "adau1373-aif2", + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .ops = &adau1373_dai_ops, + .symmetric_rates = 1, + }, + { + .id = 2, + .name = "adau1373-aif3", + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .ops = &adau1373_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + unsigned int dpll_div = 0; + unsigned int x, r, n, m, i, j, mode; + + switch (pll_id) { + case ADAU1373_PLL1: + case ADAU1373_PLL2: + break; + default: + return -EINVAL; + } + + switch (source) { + case ADAU1373_PLL_SRC_BCLK1: + case ADAU1373_PLL_SRC_BCLK2: + case ADAU1373_PLL_SRC_BCLK3: + case ADAU1373_PLL_SRC_LRCLK1: + case ADAU1373_PLL_SRC_LRCLK2: + case ADAU1373_PLL_SRC_LRCLK3: + case ADAU1373_PLL_SRC_MCLK1: + case ADAU1373_PLL_SRC_MCLK2: + case ADAU1373_PLL_SRC_GPIO1: + case ADAU1373_PLL_SRC_GPIO2: + case ADAU1373_PLL_SRC_GPIO3: + case ADAU1373_PLL_SRC_GPIO4: + break; + default: + return -EINVAL; + } + + if (freq_in < 7813 || freq_in > 27000000) + return -EINVAL; + + if (freq_out < 45158000 || freq_out > 49152000) + return -EINVAL; + + /* APLL input needs to be >= 8Mhz, so in case freq_in is less we use the + * DPLL to get it there. DPLL_out = (DPLL_in / div) * 1024 */ + while (freq_in < 8000000) { + freq_in *= 2; + dpll_div++; + } + + if (freq_out % freq_in != 0) { + /* fout = fin * (r + (n/m)) / x */ + x = DIV_ROUND_UP(freq_in, 13500000); + freq_in /= x; + r = freq_out / freq_in; + i = freq_out % freq_in; + j = gcd(i, freq_in); + n = i / j; + m = freq_in / j; + x--; + mode = 1; + } else { + /* fout = fin / r */ + r = freq_out / freq_in; + n = 0; + m = 0; + x = 0; + mode = 0; + } + + if (r < 2 || r > 8 || x > 3 || m > 0xffff || n > 0xffff) + return -EINVAL; + + if (dpll_div) { + dpll_div = 11 - dpll_div; + snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + ADAU1373_PLL_CTRL6_DPLL_BYPASS, 0); + } else { + snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + ADAU1373_PLL_CTRL6_DPLL_BYPASS, + ADAU1373_PLL_CTRL6_DPLL_BYPASS); + } + + snd_soc_write(codec, ADAU1373_DPLL_CTRL(pll_id), + (source << 4) | dpll_div); + snd_soc_write(codec, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL5(pll_id), + (r << 3) | (x << 1) | mode); + + /* Set sysclk to pll_rate / 4 */ + snd_soc_update_bits(codec, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); + + return 0; +} + +static void adau1373_load_drc_settings(struct snd_soc_codec *codec, + unsigned int nr, uint8_t *drc) +{ + unsigned int i; + + for (i = 0; i < ADAU1373_DRC_SIZE; ++i) + snd_soc_write(codec, ADAU1373_DRC(nr) + i, drc[i]); +} + +static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias) +{ + switch (micbias) { + case ADAU1373_MICBIAS_2_9V: + case ADAU1373_MICBIAS_2_2V: + case ADAU1373_MICBIAS_2_6V: + case ADAU1373_MICBIAS_1_8V: + return true; + default: + break; + } + return false; +} + +static int adau1373_probe(struct snd_soc_codec *codec) +{ + struct adau1373_platform_data *pdata = codec->dev->platform_data; + bool lineout_differential = false; + unsigned int val; + int ret; + int i; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); + return ret; + } + + codec->dapm.idle_bias_off = true; + + if (pdata) { + if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting)) + return -EINVAL; + + if (!adau1373_valid_micbias(pdata->micbias1) || + !adau1373_valid_micbias(pdata->micbias2)) + return -EINVAL; + + for (i = 0; i < pdata->num_drc; ++i) { + adau1373_load_drc_settings(codec, i, + pdata->drc_setting[i]); + } + + snd_soc_add_controls(codec, adau1373_drc_controls, + pdata->num_drc); + + val = 0; + for (i = 0; i < 4; ++i) { + if (pdata->input_differential[i]) + val |= BIT(i); + } + snd_soc_write(codec, ADAU1373_INPUT_MODE, val); + + val = 0; + if (pdata->lineout_differential) + val |= ADAU1373_OUTPUT_CTRL_LDIFF; + if (pdata->lineout_ground_sense) + val |= ADAU1373_OUTPUT_CTRL_LNFBEN; + snd_soc_write(codec, ADAU1373_OUTPUT_CTRL, val); + + lineout_differential = pdata->lineout_differential; + + snd_soc_write(codec, ADAU1373_EP_CTRL, + (pdata->micbias1 << ADAU1373_EP_CTRL_MICBIAS1_OFFSET) | + (pdata->micbias2 << ADAU1373_EP_CTRL_MICBIAS2_OFFSET)); + } + + if (!lineout_differential) { + snd_soc_add_controls(codec, adau1373_lineout2_controls, + ARRAY_SIZE(adau1373_lineout2_controls)); + } + + snd_soc_write(codec, ADAU1373_ADC_CTRL, + ADAU1373_ADC_CTRL_RESET_FORCE | ADAU1373_ADC_CTRL_PEAK_DETECT); + + return 0; +} + +static int adau1373_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + ADAU1373_PWDN_CTRL3_PWR_EN, ADAU1373_PWDN_CTRL3_PWR_EN); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + ADAU1373_PWDN_CTRL3_PWR_EN, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int adau1373_remove(struct snd_soc_codec *codec) +{ + adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int adau1373_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int adau1373_resume(struct snd_soc_codec *codec) +{ + adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + snd_soc_cache_sync(codec); + + return 0; +} + +static struct snd_soc_codec_driver adau1373_codec_driver = { + .probe = adau1373_probe, + .remove = adau1373_remove, + .suspend = adau1373_suspend, + .resume = adau1373_resume, + .set_bias_level = adau1373_set_bias_level, + .reg_cache_size = ARRAY_SIZE(adau1373_default_regs), + .reg_cache_default = adau1373_default_regs, + .reg_word_size = sizeof(uint8_t), + + .set_pll = adau1373_set_pll, + + .controls = adau1373_controls, + .num_controls = ARRAY_SIZE(adau1373_controls), + .dapm_widgets = adau1373_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau1373_dapm_widgets), + .dapm_routes = adau1373_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adau1373_dapm_routes), +}; + +static int __devinit adau1373_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct adau1373 *adau1373; + int ret; + + adau1373 = kzalloc(sizeof(*adau1373), GFP_KERNEL); + if (!adau1373) + return -ENOMEM; + + dev_set_drvdata(&client->dev, adau1373); + + ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver, + adau1373_dai_driver, ARRAY_SIZE(adau1373_dai_driver)); + if (ret < 0) + kfree(adau1373); + + return ret; +} + +static int __devexit adau1373_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(dev_get_drvdata(&client->dev)); + return 0; +} + +static const struct i2c_device_id adau1373_i2c_id[] = { + { "adau1373", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adau1373_i2c_id); + +static struct i2c_driver adau1373_i2c_driver = { + .driver = { + .name = "adau1373", + .owner = THIS_MODULE, + }, + .probe = adau1373_i2c_probe, + .remove = __devexit_p(adau1373_i2c_remove), + .id_table = adau1373_i2c_id, +}; + +static int __init adau1373_init(void) +{ + return i2c_add_driver(&adau1373_i2c_driver); +} +module_init(adau1373_init); + +static void __exit adau1373_exit(void) +{ + i2c_del_driver(&adau1373_i2c_driver); +} +module_exit(adau1373_exit); + +MODULE_DESCRIPTION("ASoC ADAU1373 driver"); +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1373.h b/sound/soc/codecs/adau1373.h new file mode 100644 index 000000000000..c6ab5530760c --- /dev/null +++ b/sound/soc/codecs/adau1373.h @@ -0,0 +1,29 @@ +#ifndef __ADAU1373_H__ +#define __ADAU1373_H__ + +enum adau1373_pll_src { + ADAU1373_PLL_SRC_MCLK1 = 0, + ADAU1373_PLL_SRC_BCLK1 = 1, + ADAU1373_PLL_SRC_BCLK2 = 2, + ADAU1373_PLL_SRC_BCLK3 = 3, + ADAU1373_PLL_SRC_LRCLK1 = 4, + ADAU1373_PLL_SRC_LRCLK2 = 5, + ADAU1373_PLL_SRC_LRCLK3 = 6, + ADAU1373_PLL_SRC_GPIO1 = 7, + ADAU1373_PLL_SRC_GPIO2 = 8, + ADAU1373_PLL_SRC_GPIO3 = 9, + ADAU1373_PLL_SRC_GPIO4 = 10, + ADAU1373_PLL_SRC_MCLK2 = 11, +}; + +enum adau1373_pll { + ADAU1373_PLL1 = 0, + ADAU1373_PLL2 = 1, +}; + +enum adau1373_clk_src { + ADAU1373_CLK_SRC_PLL1 = 0, + ADAU1373_CLK_SRC_PLL2 = 1, +}; + +#endif -- cgit v1.2.3 From f049ffb3f8cf682df405f029914938f95d667695 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 15 Aug 2011 20:15:23 +0200 Subject: ASoC: Blackfin: ADAU1373 eval board support Add a machine driver to support the EVAL-ADAU1373 board connected to a Analog Devices BF5XX evaluation board. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/Kconfig | 13 ++ sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bfin-eval-adau1373.c | 202 ++++++++++++++++++++++++++++++++ 3 files changed, 217 insertions(+) create mode 100644 sound/soc/blackfin/bfin-eval-adau1373.c (limited to 'sound') diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index fe9d548a6837..9f6bc55fc399 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -27,6 +27,19 @@ config SND_SOC_BFIN_EVAL_ADAU1701 board connected to one of the Blackfin evaluation boards like the BF5XX-STAMP or BF5XX-EZKIT. +config SND_SOC_BFIN_EVAL_ADAU1373 + tristate "Support for the EVAL-ADAU1373 board on Blackfin eval boards" + depends on SND_BF5XX_I2S && I2C + select SND_BF5XX_SOC_I2S + select SND_SOC_ADAU1373 + help + Say Y if you want to add support for the Analog Devices EVAL-ADAU1373 + board connected to one of the Blackfin evaluation boards like the + BF5XX-STAMP or BF5XX-EZKIT. + + Note: This driver assumes that first ADAU1373 DAI is connected to the + first SPORT port on the BF5XX board. + config SND_SOC_BFIN_EVAL_ADAV80X tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards" depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 6018bf52a234..1bf86ccaa8de 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -21,6 +21,7 @@ snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o snd-ad73311-objs := bf5xx-ad73311.o snd-ad193x-objs := bf5xx-ad193x.o +snd-soc-bfin-eval-adau1373-objs := bfin-eval-adau1373.o snd-soc-bfin-eval-adau1701-objs := bfin-eval-adau1701.o snd-soc-bfin-eval-adav80x-objs := bfin-eval-adav80x.o @@ -29,5 +30,6 @@ obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o +obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1373) += snd-soc-bfin-eval-adau1373.o obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701) += snd-soc-bfin-eval-adau1701.o obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) += snd-soc-bfin-eval-adav80x.o diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c new file mode 100644 index 000000000000..8df2a3b0cb36 --- /dev/null +++ b/sound/soc/blackfin/bfin-eval-adau1373.c @@ -0,0 +1,202 @@ +/* + * Machine driver for EVAL-ADAU1373 on Analog Devices bfin + * evaluation boards. + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2 or later. + */ + +#include +#include +#include +#include +#include +#include + +#include "../codecs/adau1373.h" + +static const struct snd_soc_dapm_widget bfin_eval_adau1373_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Line In1", NULL), + SND_SOC_DAPM_LINE("Line In2", NULL), + SND_SOC_DAPM_LINE("Line In3", NULL), + SND_SOC_DAPM_LINE("Line In4", NULL), + + SND_SOC_DAPM_LINE("Line Out1", NULL), + SND_SOC_DAPM_LINE("Line Out2", NULL), + SND_SOC_DAPM_LINE("Stereo Out", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_HP("Earpiece", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route bfin_eval_adau1373_dapm_routes[] = { + { "AIN1L", NULL, "Line In1" }, + { "AIN1R", NULL, "Line In1" }, + { "AIN2L", NULL, "Line In2" }, + { "AIN2R", NULL, "Line In2" }, + { "AIN3L", NULL, "Line In3" }, + { "AIN3R", NULL, "Line In3" }, + { "AIN4L", NULL, "Line In4" }, + { "AIN4R", NULL, "Line In4" }, + + /* MICBIAS can be connected via a jumper to the line-in jack, since w + don't know which one is going to be used, just power both. */ + { "Line In1", NULL, "MICBIAS1" }, + { "Line In2", NULL, "MICBIAS1" }, + { "Line In3", NULL, "MICBIAS1" }, + { "Line In4", NULL, "MICBIAS1" }, + { "Line In1", NULL, "MICBIAS2" }, + { "Line In2", NULL, "MICBIAS2" }, + { "Line In3", NULL, "MICBIAS2" }, + { "Line In4", NULL, "MICBIAS2" }, + + { "Line Out1", NULL, "LOUT1L" }, + { "Line Out1", NULL, "LOUT1R" }, + { "Line Out2", NULL, "LOUT2L" }, + { "Line Out2", NULL, "LOUT2R" }, + { "Headphone", NULL, "HPL" }, + { "Headphone", NULL, "HPR" }, + { "Earpiece", NULL, "EP" }, + { "Speaker", NULL, "SPKL" }, + { "Stereo Out", NULL, "SPKR" }, +}; + +static int bfin_eval_adau1373_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + int pll_rate; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret) + return ret; + + switch (params_rate(params)) { + case 48000: + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + pll_rate = 48000 * 1024; + break; + case 44100: + case 7350: + case 11025: + case 14700: + case 22050: + case 29400: + pll_rate = 44100 * 1024; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1, + ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate, + SND_SOC_CLOCK_IN); + + return ret; +} + +static int bfin_eval_adau1373_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int pll_rate = 48000 * 1024; + int ret; + + ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1, + ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate, + SND_SOC_CLOCK_IN); + + return ret; +} +static struct snd_soc_ops bfin_eval_adau1373_ops = { + .hw_params = bfin_eval_adau1373_hw_params, +}; + +static struct snd_soc_dai_link bfin_eval_adau1373_dai = { + .name = "adau1373", + .stream_name = "adau1373", + .cpu_dai_name = "bfin-i2s.0", + .codec_dai_name = "adau1373-aif1", + .platform_name = "bfin-i2s-pcm-audio", + .codec_name = "adau1373.0-001a", + .ops = &bfin_eval_adau1373_ops, + .init = bfin_eval_adau1373_codec_init, +}; + +static struct snd_soc_card bfin_eval_adau1373 = { + .name = "bfin-eval-adau1373", + .dai_link = &bfin_eval_adau1373_dai, + .num_links = 1, + + .dapm_widgets = bfin_eval_adau1373_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1373_dapm_widgets), + .dapm_routes = bfin_eval_adau1373_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1373_dapm_routes), +}; + +static int bfin_eval_adau1373_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &bfin_eval_adau1373; + + card->dev = &pdev->dev; + + return snd_soc_register_card(&bfin_eval_adau1373); +} + +static int __devexit bfin_eval_adau1373_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver bfin_eval_adau1373_driver = { + .driver = { + .name = "bfin-eval-adau1373", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = bfin_eval_adau1373_probe, + .remove = __devexit_p(bfin_eval_adau1373_remove), +}; + +static int __init bfin_eval_adau1373_init(void) +{ + return platform_driver_register(&bfin_eval_adau1373_driver); +} +module_init(bfin_eval_adau1373_init); + +static void __exit bfin_eval_adau1373_exit(void) +{ + platform_driver_unregister(&bfin_eval_adau1373_driver); +} +module_exit(bfin_eval_adau1373_exit); + +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_DESCRIPTION("ALSA SoC bfin adau1373 driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:bfin-eval-adau1373"); -- cgit v1.2.3 From 1fab6cafc798c987caa6e98ee8e04991e9171cd0 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Tue, 16 Aug 2011 18:47:45 -0400 Subject: ASoC: claim the IRQ when the fsl_ssi device is probed, not opened The PowerPC Freescale SSI driver is claiming the IRQ when the IRQ when the device is opened, which means that the /proc/interrupts entry for the SSI exists only during playback or capture. This also meant that the user won't know that the IRQ number is wrong until he tries to use the device. Instead, we should claim the IRQ when the device is probed. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 61 ++++++++++++++++++++++++++++++------------------- 1 file changed, 37 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index d48afea5d93d..06ac2b92faf3 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -289,16 +289,6 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, */ if (!ssi_private->playback && !ssi_private->capture) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - int ret; - - /* The 'name' should not have any slashes in it. */ - ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, - ssi_private->name, ssi_private); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not claim irq %u\n", ssi_private->irq); - return ret; - } /* * Section 16.5 of the MPC8610 reference manual says that the @@ -522,15 +512,12 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, ssi_private->second_stream = NULL; /* - * If this is the last active substream, disable the SSI and release - * the IRQ. + * If this is the last active substream, disable the SSI. */ if (!ssi_private->playback && !ssi_private->capture) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); - - free_irq(ssi_private->irq, ssi_private); } } @@ -675,17 +662,30 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) ret = of_address_to_resource(np, 0, &res); if (ret) { dev_err(&pdev->dev, "could not determine device resources\n"); - kfree(ssi_private); - return ret; + goto error_kmalloc; } ssi_private->ssi = of_iomap(np, 0); if (!ssi_private->ssi) { dev_err(&pdev->dev, "could not map device resources\n"); - kfree(ssi_private); - return -ENOMEM; + ret = -ENOMEM; + goto error_kmalloc; } ssi_private->ssi_phys = res.start; + ssi_private->irq = irq_of_parse_and_map(np, 0); + if (ssi_private->irq == NO_IRQ) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + ret = -ENXIO; + goto error_iomap; + } + + /* The 'name' should not have any slashes in it. */ + ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name, + ssi_private); + if (ret < 0) { + dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); + goto error_irqmap; + } /* Are the RX and the TX clocks locked? */ if (of_find_property(np, "fsl,ssi-asynchronous", NULL)) @@ -711,7 +711,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "could not create sysfs %s file\n", ssi_private->dev_attr.attr.name); - goto error; + goto error_irq; } /* Register with ASoC */ @@ -720,7 +720,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &ssi_private->cpu_dai_drv); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); - goto error; + goto error_dev; } /* Trigger the machine driver's probe function. The platform driver @@ -741,18 +741,28 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (IS_ERR(ssi_private->pdev)) { ret = PTR_ERR(ssi_private->pdev); dev_err(&pdev->dev, "failed to register platform: %d\n", ret); - goto error; + goto error_dai; } return 0; -error: +error_dai: snd_soc_unregister_dai(&pdev->dev); + +error_dev: dev_set_drvdata(&pdev->dev, NULL); - if (dev_attr) - device_remove_file(&pdev->dev, dev_attr); + device_remove_file(&pdev->dev, dev_attr); + +error_irq: + free_irq(ssi_private->irq, ssi_private); + +error_irqmap: irq_dispose_mapping(ssi_private->irq); + +error_iomap: iounmap(ssi_private->ssi); + +error_kmalloc: kfree(ssi_private); return ret; @@ -766,6 +776,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) snd_soc_unregister_dai(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); + free_irq(ssi_private->irq, ssi_private); + irq_dispose_mapping(ssi_private->irq); + kfree(ssi_private); dev_set_drvdata(&pdev->dev, NULL); -- cgit v1.2.3 From 60e3ee62af12e7c5d91153ce724956254a857c2e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 17 Aug 2011 15:14:17 +0900 Subject: ASoC: Fix backport of WM8994 thermal warning Signed-off-by: Mark Brown Reported-by: Stephen Rothwell --- sound/soc/codecs/wm8994.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 94124913bb3e..e5372675123d 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3141,9 +3141,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, wm8994_fifo_error, "FIFO error", codec); - wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, + wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_WARN, wm8994_temp_warn, "Thermal warning", codec); - wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, + wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_SHUT, wm8994_temp_shut, "Thermal shutdown", codec); ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, -- cgit v1.2.3 From 96af5c6a8266003a2212d9d0b383603f1af9b109 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Tue, 16 Aug 2011 21:58:29 -0400 Subject: ASoC: fsl: fix build warning in fsl_dma The previous patch to fsl_dma.c ("fix initialization of DMA buffers") left behind an unused local variable that causes a build warning. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 732208c8c0b4..0efc04af8f15 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -297,7 +297,6 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; static u64 fsl_dma_dmamask = DMA_BIT_MASK(36); int ret; -- cgit v1.2.3 From 4f7e7954a7f66735b0ee4b304c075c24ffae091a Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 17 Aug 2011 15:14:33 +0800 Subject: ASoC: Remove unreachable code in au1xac97c_drvprobe and au1xi2s_drvprobe Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/au1x/ac97c.c | 2 -- sound/soc/au1x/i2sc.c | 1 - 2 files changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 9c05f381d95e..13802ff7cf05 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -275,8 +275,6 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) ac97c_workdata = ctx; return 0; - - snd_soc_unregister_dai(&pdev->dev); out1: release_mem_region(r->start, resource_size(r)); out0: diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index b4172fdd2c48..19e0d2a9c828 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -267,7 +267,6 @@ static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) return 0; - snd_soc_unregister_dai(&pdev->dev); out1: release_mem_region(r->start, resource_size(r)); out0: -- cgit v1.2.3 From e3d73c1bbf08b9abd3f56293796ba7b5c15008f5 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 18 Aug 2011 15:31:04 +0800 Subject: ASoC: sta32x: Move resource allocation and release to the corresponding callback functions This patch includes below small fixes: 1. Move sta32x_set_bias_level() from sta32x_i2c_remove() to sta32x_remove(). 2. Remove a redundant regulator_bulk_free() call in sta32x_i2c_remove(), as we will call regulator_bulk_free() in sta32x_remove(). 3. Remove unneeded snd_soc_codec_set_drvdata(codec, NULL) in sta32x_i2c_remove. The i2c core will set the clientdata to NULL. Signed-off-by: Axel Lin Johannes Stezenbach Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 3d155f526672..5c7def3979c0 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -812,6 +812,7 @@ static int sta32x_remove(struct snd_soc_codec *codec) { struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); @@ -871,18 +872,8 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, static __devexit int sta32x_i2c_remove(struct i2c_client *client) { struct sta32x_priv *sta32x = i2c_get_clientdata(client); - struct snd_soc_codec *codec = sta32x->codec; - - if (codec) - sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); - - regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); - - if (codec) { - snd_soc_unregister_codec(&client->dev); - snd_soc_codec_set_drvdata(codec, NULL); - } + snd_soc_unregister_codec(&client->dev); kfree(sta32x); return 0; } -- cgit v1.2.3 From 889c85c550ebdf8af69f5c08387fde3f6f48d10f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 20 Aug 2011 19:00:50 +0100 Subject: ASoC: Automatically manage WM8996 MICBIAS regulating mode For non-audio uses like accessory detection we can use a lower quality, unregulated microphone bias, saving a little power. As the hardware can manually enable and disable the biases we can select regulating mode automatically with supply widgets connected to the biases. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8996.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index ab8e9d1aaff0..2a0a612a4edb 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -982,6 +982,8 @@ SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICB1 Audio", WM8996_MICBIAS_1, 4, 1, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICB2 Audio", WM8996_MICBIAS_2, 4, 1, NULL, 0), SND_SOC_DAPM_MICBIAS("MICB2", WM8996_POWER_MANAGEMENT_1, 9, 0), SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0), @@ -1142,7 +1144,9 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "Charge Pump", NULL, "SYSCLK" }, { "MICB1", NULL, "LDO2" }, + { "MICB1", NULL, "MICB1 Audio" }, { "MICB2", NULL, "LDO2" }, + { "MICB2", NULL, "MICB2 Audio" }, { "IN1L PGA", NULL, "IN2LN" }, { "IN1L PGA", NULL, "IN2LP" }, -- cgit v1.2.3 From 2fde6e80dd5460a54651c74b5e5d9a22e4f82af2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 20 Aug 2011 19:28:59 +0100 Subject: ASoC: Optimise WM8996 no interrupt path This occurs frequently if we are in edge triggered mode as we must poll the interrupt status register until we get no more interrupts so it's worth the effort - it means we skip writing null acknowledgements to the chip. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8996.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 2a0a612a4edb..acbad5be69bc 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2408,6 +2408,9 @@ static irqreturn_t wm8996_irq(int irq, void *data) } irq_val &= ~snd_soc_read(codec, WM8996_INTERRUPT_STATUS_2_MASK); + if (!irq_val) + return IRQ_NONE; + snd_soc_write(codec, WM8996_INTERRUPT_STATUS_2, irq_val); if (irq_val & (WM8996_DCS_DONE_01_EINT | WM8996_DCS_DONE_23_EINT)) { @@ -2426,10 +2429,7 @@ static irqreturn_t wm8996_irq(int irq, void *data) if (irq_val & WM8996_MICD_EINT) wm8996_micd(codec); - if (irq_val) - return IRQ_HANDLED; - else - return IRQ_NONE; + return IRQ_HANDLED; } static irqreturn_t wm8996_edge_irq(int irq, void *data) -- cgit v1.2.3 From 1ab63da7212d4422cbc40d4ead5cff97f6050a50 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 21 Aug 2011 10:54:38 +0100 Subject: ASoC: Add basic WM8962 capture low/high pass filter control Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8962.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 28650edfdebb..5538737c88ce 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2049,6 +2049,14 @@ static const char *cap_hpf_mode_text[] = { static const struct soc_enum cap_hpf_mode = SOC_ENUM_SINGLE(WM8962_ADC_DAC_CONTROL_2, 10, 2, cap_hpf_mode_text); + +static const char *cap_lhpf_mode_text[] = { + "LPF", "HPF" +}; + +static const struct soc_enum cap_lhpf_mode = + SOC_ENUM_SINGLE(WM8962_LHPF1, 1, 2, cap_lhpf_mode_text); + static const struct snd_kcontrol_new wm8962_snd_controls[] = { SOC_DOUBLE("Input Mixer Switch", WM8962_INPUT_MIXER_CONTROL_1, 3, 2, 1, 1), @@ -2077,6 +2085,8 @@ SOC_DOUBLE_R("Capture ZC Switch", WM8962_LEFT_INPUT_VOLUME, SOC_SINGLE("Capture HPF Switch", WM8962_ADC_DAC_CONTROL_1, 0, 1, 1), SOC_ENUM("Capture HPF Mode", cap_hpf_mode), SOC_SINGLE("Capture HPF Cutoff", WM8962_ADC_DAC_CONTROL_2, 7, 7, 0), +SOC_SINGLE("Capture LHPF Switch", WM8962_LHPF1, 0, 1, 0), +SOC_ENUM("Capture LHPF Mode", cap_lhpf_mode), SOC_DOUBLE_R_TLV("Sidetone Volume", WM8962_DAC_DSP_MIXING_1, WM8962_DAC_DSP_MIXING_2, 4, 12, 0, st_tlv), -- cgit v1.2.3 From 6f88a4e5785fbf4db9a2c7e16670e1f19e6566d2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 17 Aug 2011 10:03:51 +0900 Subject: ASoC: Initial WM8962 DSP2 support The WM8962 features a DSP providing a number of signal processing features including HD Bass and Virtual Surround Sound (VSS). Enable initial support for this, allowing users to enable and disable the algorithms using the default coefficient sets. Further patches will add support for runtime configuration of the DSP coefficients. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8962.c | 157 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 156 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 5538737c88ce..75e784053603 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -63,6 +63,8 @@ struct wm8962_priv { int fll_fref; int fll_fout; + u16 dsp2_ena; + struct delayed_work mic_work; struct snd_soc_jack *jack; @@ -965,7 +967,7 @@ static const struct wm8962_reg_access { [584] = { 0x002D, 0x002D, 0x0000 }, /* R584 - IRQ Debounce */ [586] = { 0xC000, 0xC000, 0x0000 }, /* R586 - MICINT Source Pol */ [768] = { 0x0001, 0x0001, 0x0000 }, /* R768 - DSP2 Power Management */ - [1037] = { 0x0000, 0x003F, 0x0000 }, /* R1037 - DSP2_ExecControl */ + [1037] = { 0x0000, 0x003F, 0xFFFF }, /* R1037 - DSP2_ExecControl */ [4096] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4096 - Write Sequencer 0 */ [4097] = { 0x00FF, 0x00FF, 0x0000 }, /* R4097 - Write Sequencer 1 */ [4098] = { 0x070F, 0x070F, 0x0000 }, /* R4098 - Write Sequencer 2 */ @@ -1986,6 +1988,122 @@ static const unsigned int classd_tlv[] = { }; static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static int wm8962_dsp2_write_config(struct snd_soc_codec *codec) +{ + return 0; +} + +static int wm8962_dsp2_set_enable(struct snd_soc_codec *codec, u16 val) +{ + u16 adcl = snd_soc_read(codec, WM8962_LEFT_ADC_VOLUME); + u16 adcr = snd_soc_read(codec, WM8962_RIGHT_ADC_VOLUME); + u16 dac = snd_soc_read(codec, WM8962_ADC_DAC_CONTROL_1); + + /* Mute the ADCs and DACs */ + snd_soc_write(codec, WM8962_LEFT_ADC_VOLUME, 0); + snd_soc_write(codec, WM8962_RIGHT_ADC_VOLUME, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1, + WM8962_DAC_MUTE, WM8962_DAC_MUTE); + + snd_soc_write(codec, WM8962_SOUNDSTAGE_ENABLES_0, val); + + /* Restore the ADCs and DACs */ + snd_soc_write(codec, WM8962_LEFT_ADC_VOLUME, adcl); + snd_soc_write(codec, WM8962_RIGHT_ADC_VOLUME, adcr); + snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1, + WM8962_DAC_MUTE, dac); + + return 0; +} + +static int wm8962_dsp2_start(struct snd_soc_codec *codec) +{ + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + wm8962_dsp2_write_config(codec); + + snd_soc_write(codec, WM8962_DSP2_EXECCONTROL, WM8962_DSP2_RUNR); + + wm8962_dsp2_set_enable(codec, wm8962->dsp2_ena); + + return 0; +} + +static int wm8962_dsp2_stop(struct snd_soc_codec *codec) +{ + wm8962_dsp2_set_enable(codec, 0); + + snd_soc_write(codec, WM8962_DSP2_EXECCONTROL, WM8962_DSP2_STOP); + + return 0; +} + +#define WM8962_DSP2_ENABLE(xname, xshift) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = wm8962_dsp2_ena_info, \ + .get = wm8962_dsp2_ena_get, .put = wm8962_dsp2_ena_put, \ + .private_value = xshift } + +static int wm8962_dsp2_ena_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} + +static int wm8962_dsp2_ena_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int shift = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !!(wm8962->dsp2_ena & 1 << shift); + + return 0; +} + +static int wm8962_dsp2_ena_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int shift = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + int old = wm8962->dsp2_ena; + int ret = 0; + int dsp2_running = snd_soc_read(codec, WM8962_DSP2_POWER_MANAGEMENT) & + WM8962_DSP2_ENA; + + mutex_lock(&codec->mutex); + + if (ucontrol->value.integer.value[0]) + wm8962->dsp2_ena |= 1 << shift; + else + wm8962->dsp2_ena &= ~(1 << shift); + + if (wm8962->dsp2_ena == old) + goto out; + + ret = 1; + + if (dsp2_running) { + if (wm8962->dsp2_ena) + wm8962_dsp2_set_enable(codec, wm8962->dsp2_ena); + else + wm8962_dsp2_stop(codec); + } + +out: + mutex_unlock(&codec->mutex); + + return ret; +} + /* The VU bits for the headphones are in a different register to the mute * bits and only take effect on the PGA if it is actually powered. */ @@ -2144,6 +2262,11 @@ SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B4_GAIN_SHIFT, 31, 0, eq_tlv), SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv), + +WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT), +WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT), +WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT), +WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT), }; static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = { @@ -2403,6 +2526,31 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, } } +static int dsp2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (wm8962->dsp2_ena) + wm8962_dsp2_start(codec); + break; + + case SND_SOC_DAPM_PRE_PMD: + if (wm8962->dsp2_ena) + wm8962_dsp2_stop(codec); + break; + + default: + BUG(); + return -EINVAL; + } + + return 0; +} + static const char *st_text[] = { "None", "Right", "Left" }; static const struct soc_enum str_enum = @@ -2525,6 +2673,9 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, sysclk_event, SND_SOC_DAPM_SUPPLY("Charge Pump", WM8962_CHARGE_PUMP_1, 0, 0, cp_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY_S("DSP2", 1, WM8962_DSP2_POWER_MANAGEMENT, + WM8962_DSP2_ENA_SHIFT, 0, dsp2_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_MIXER("INPGAL", WM8962_LEFT_INPUT_PGA_CONTROL, 4, 0, inpgal, ARRAY_SIZE(inpgal)), @@ -2620,11 +2771,13 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "ADCL", NULL, "TOCLK" }, { "ADCL", NULL, "MIXINL" }, { "ADCL", NULL, "DMIC" }, + { "ADCL", NULL, "DSP2" }, { "ADCR", NULL, "SYSCLK" }, { "ADCR", NULL, "TOCLK" }, { "ADCR", NULL, "MIXINR" }, { "ADCR", NULL, "DMIC" }, + { "ADCR", NULL, "DSP2" }, { "STL", "Left", "ADCL" }, { "STL", "Right", "ADCR" }, @@ -2636,11 +2789,13 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "DACL", NULL, "TOCLK" }, { "DACL", NULL, "Beep" }, { "DACL", NULL, "STL" }, + { "DACL", NULL, "DSP2" }, { "DACR", NULL, "SYSCLK" }, { "DACR", NULL, "TOCLK" }, { "DACR", NULL, "Beep" }, { "DACR", NULL, "STR" }, + { "DACR", NULL, "DSP2" }, { "HPMIXL", "IN4L Switch", "IN4L" }, { "HPMIXL", "IN4R Switch", "IN4R" }, -- cgit v1.2.3 From e6ef58700a8afba46f2aa98a0de12c35e4b1f295 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 21 Aug 2011 11:47:14 +0100 Subject: ASoC: Report IRQ_NONE when we don't see an interrupt from WM8962 This should never happen with level triggered IRQs. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8962.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 75e784053603..add07fff4495 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3570,6 +3570,9 @@ static irqreturn_t wm8962_irq(int irq, void *data) active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); active &= ~mask; + if (!active) + return IRQ_NONE; + /* Acknowledge the interrupts */ snd_soc_write(codec, WM8962_INTERRUPT_STATUS_2, active); -- cgit v1.2.3 From fbf04076ef9b704ab27dbd1b2f97569227775bb4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 21 Aug 2011 18:07:44 +0100 Subject: ASoC: Provide more detail on WM8962 thermal shutdown status Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8962.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index add07fff4495..382c8779e605 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -839,7 +839,7 @@ static const struct wm8962_reg_access { [40] = { 0x00FF, 0x01FF, 0x0000 }, /* R40 - SPKOUTL volume */ [41] = { 0x00FF, 0x01FF, 0x0000 }, /* R41 - SPKOUTR volume */ - [47] = { 0x000F, 0x0000, 0x0000 }, /* R47 - Thermal Shutdown Status */ + [47] = { 0x000F, 0x0000, 0xFFFF }, /* R47 - Thermal Shutdown Status */ [48] = { 0x7EC7, 0x7E07, 0xFFFF }, /* R48 - Additional Control (4) */ [49] = { 0x00D3, 0x00D7, 0xFFFF }, /* R49 - Class D Control 1 */ [51] = { 0x0047, 0x0047, 0x0000 }, /* R51 - Class D Control 2 */ @@ -3564,6 +3564,7 @@ static irqreturn_t wm8962_irq(int irq, void *data) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int mask; int active; + int reg; mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK); @@ -3584,9 +3585,21 @@ static irqreturn_t wm8962_irq(int irq, void *data) if (active & WM8962_FIFOS_ERR_EINT) dev_err(codec->dev, "FIFO error\n"); - if (active & WM8962_TEMP_SHUT_EINT) + if (active & WM8962_TEMP_SHUT_EINT) { dev_crit(codec->dev, "Thermal shutdown\n"); + reg = snd_soc_read(codec, WM8962_THERMAL_SHUTDOWN_STATUS); + + if (reg & WM8962_TEMP_ERR_HP) + dev_crit(codec->dev, "Headphone thermal error\n"); + if (reg & WM8962_TEMP_WARN_HP) + dev_crit(codec->dev, "Headphone thermal warning\n"); + if (reg & WM8962_TEMP_ERR_SPK) + dev_crit(codec->dev, "Speaker thermal error\n"); + if (reg & WM8962_TEMP_WARN_SPK) + dev_crit(codec->dev, "Speaker thermal warning\n"); + } + if (active & (WM8962_MICSCD_EINT | WM8962_MICD_EINT)) { dev_dbg(codec->dev, "Microphone event detected\n"); -- cgit v1.2.3 From 1661699aaa64e6024770ea7adff4fc6216cb25ca Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 22 Aug 2011 16:02:43 +0100 Subject: ASoC: Convert WM8523 to table based control and DAPM initialization Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8523.c | 26 +++++++++----------------- 1 file changed, 9 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 52812d1a90e4..5355a7a944f7 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -85,7 +85,7 @@ static const char *wm8523_zd_count_text[] = { static const struct soc_enum wm8523_zc_count = SOC_ENUM_SINGLE(WM8523_ZERO_DETECT, 0, 2, wm8523_zd_count_text); -static const struct snd_kcontrol_new wm8523_snd_controls[] = { +static const struct snd_kcontrol_new wm8523_controls[] = { SOC_DOUBLE_R_TLV("Playback Volume", WM8523_DAC_GAINL, WM8523_DAC_GAINR, 0, 448, 0, dac_tlv), SOC_SINGLE("ZC Switch", WM8523_DAC_CTRL3, 4, 1, 0), @@ -102,22 +102,11 @@ SND_SOC_DAPM_OUTPUT("LINEVOUTL"), SND_SOC_DAPM_OUTPUT("LINEVOUTR"), }; -static const struct snd_soc_dapm_route intercon[] = { +static const struct snd_soc_dapm_route wm8523_dapm_routes[] = { { "LINEVOUTL", NULL, "DAC" }, { "LINEVOUTR", NULL, "DAC" }, }; -static int wm8523_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8523_dapm_widgets, - ARRAY_SIZE(wm8523_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - static struct { int value; int ratio; @@ -480,10 +469,6 @@ static int wm8523_probe(struct snd_soc_codec *codec) /* Bias level configuration will have done an extra enable */ regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); - snd_soc_add_controls(codec, wm8523_snd_controls, - ARRAY_SIZE(wm8523_snd_controls)); - wm8523_add_widgets(codec); - return 0; err_enable: @@ -513,6 +498,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8523 = { .reg_word_size = sizeof(u16), .reg_cache_default = wm8523_reg, .volatile_register = wm8523_volatile_register, + + .controls = wm8523_controls, + .num_controls = ARRAY_SIZE(wm8523_controls), + .dapm_widgets = wm8523_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8523_dapm_widgets), + .dapm_routes = wm8523_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8523_dapm_routes), }; static const struct of_device_id wm8523_of_match[] = { -- cgit v1.2.3 From 33c5f969b969c277e96cd9e9bf8472c4b8709c25 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 22 Aug 2011 18:40:30 +0100 Subject: ASoC: Allow idle_bias_off to be specified in CODEC drivers If devices can unconditionally support idle_bias_off let them flag it in their driver structure. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ae93aa81244c..f8f985a4f2a8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -956,6 +956,8 @@ static int soc_probe_codec(struct snd_soc_card *card, snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, driver->num_dapm_widgets); + codec->dapm.idle_bias_off = driver->idle_bias_off; + if (driver->probe) { ret = driver->probe(codec); if (ret < 0) { -- cgit v1.2.3 From bbe8ff5e25afd4d06c8a8bad009aca5f0d0c22ef Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Sun, 21 Aug 2011 23:45:40 +0800 Subject: ASoC: mxs-saif: clear clk gate first before register setting Saif needs clear clk gate first before writing registers or the write will not success. The original xx_get_mclk function clear clk gate after mclk setting that may cause the former mclk setting unwork, then the real output mclk maybe inaccurate. Placing the clear before setting mclk to avoid such an issue. We also have to clear clk gate in startup instead of in prepare function. Signed-off-by: Dong Aisheng Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 0b3adaec9f4c..530017f7d14a 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -187,16 +187,20 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, if (!saif) return -EINVAL; + /* Clear Reset */ + __raw_writel(BM_SAIF_CTRL_SFTRST, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + + /* FIXME: need clear clk gate for register r/w */ + __raw_writel(BM_SAIF_CTRL_CLKGATE, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + stat = __raw_readl(saif->base + SAIF_STAT); if (stat & BM_SAIF_STAT_BUSY) { dev_err(saif->dev, "error: busy\n"); return -EBUSY; } - /* Clear Reset */ - __raw_writel(BM_SAIF_CTRL_SFTRST, - saif->base + SAIF_CTRL + MXS_CLR_ADDR); - saif->mclk_in_use = 1; ret = mxs_saif_set_clk(saif, mclk, rate); if (ret) @@ -207,8 +211,6 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, return ret; /* enable MCLK output */ - __raw_writel(BM_SAIF_CTRL_CLKGATE, - saif->base + SAIF_CTRL + MXS_CLR_ADDR); __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_SET_ADDR); @@ -303,6 +305,10 @@ static int mxs_saif_startup(struct snd_pcm_substream *substream, __raw_writel(BM_SAIF_CTRL_SFTRST, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + /* clear clock gate */ + __raw_writel(BM_SAIF_CTRL_CLKGATE, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + return 0; } @@ -379,10 +385,6 @@ static int mxs_saif_prepare(struct snd_pcm_substream *substream, { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); - /* clear clock gate */ - __raw_writel(BM_SAIF_CTRL_CLKGATE, - saif->base + SAIF_CTRL + MXS_CLR_ADDR); - /* enable FIFO error irqs */ __raw_writel(BM_SAIF_CTRL_FIFO_ERROR_IRQ_EN, saif->base + SAIF_CTRL + MXS_SET_ADDR); -- cgit v1.2.3 From 0bb98ba2b045e53b4724f34509455b7653c329d3 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 21 Aug 2011 13:18:45 +0200 Subject: sound/soc/mxs/mxs-saif.c: add missing kfree Move the test on pdev->id before the kzalloc to avoid requiring kfree when the test fails. This fix was suggested by Wolfram Sang. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @@ identifier x; expression E1!=0,E2,E3,E4; statement S; iterator I; @@ ( if (...) { ... when != kfree(x) when != x = E3 when != E3 = x * return ...; } ... when != x = E2 when != I(...,x,...) S if (...) { ... when != x = E4 kfree(x); ... return ...; } ) // Signed-off-by: Julia Lawall Acked-by: Dong Aisheng Reviewed-by: Wolfram Sang Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 530017f7d14a..af5734f6dab7 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -521,12 +521,13 @@ static int mxs_saif_probe(struct platform_device *pdev) struct mxs_saif *saif; int ret = 0; + if (pdev->id >= ARRAY_SIZE(mxs_saif)) + return -EINVAL; + saif = kzalloc(sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; - if (pdev->id >= ARRAY_SIZE(mxs_saif)) - return -EINVAL; mxs_saif[pdev->id] = saif; saif->clk = clk_get(&pdev->dev, NULL); -- cgit v1.2.3 From 78a262c87157bc049a1b08faf4762c606b24fed9 Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Mon, 22 Aug 2011 00:02:46 +0800 Subject: ASoC: mxs-sgtl5000: add record function Signed-off-by: Dong Aisheng Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-sgtl5000.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index a0d89c93df0f..7fbeaec06eb4 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -85,13 +85,21 @@ static struct snd_soc_ops mxs_sgtl5000_hifi_ops = { static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { { - .name = "HiFi", + .name = "HiFi Tx", .stream_name = "HiFi Playback", .codec_dai_name = "sgtl5000", .codec_name = "sgtl5000.0-000a", .cpu_dai_name = "mxs-saif.0", .platform_name = "mxs-pcm-audio.0", .ops = &mxs_sgtl5000_hifi_ops, + }, { + .name = "HiFi Rx", + .stream_name = "HiFi Capture", + .codec_dai_name = "sgtl5000", + .codec_name = "sgtl5000.0-000a", + .cpu_dai_name = "mxs-saif.1", + .platform_name = "mxs-pcm-audio.1", + .ops = &mxs_sgtl5000_hifi_ops, }, }; -- cgit v1.2.3 From dff2836707a40868766ec37c0869ff60adfc5706 Mon Sep 17 00:00:00 2001 From: Sangbeom Kim Date: Tue, 23 Aug 2011 18:59:08 +0900 Subject: ASoC: SAMSUNG: Add Kconfig to support SMDK4212 This patch adds Kconfig to support SMDK4212. SMDK4212 is based on samsung exynos4212 SoC. And WM8994 is used for audio codec. Signed-off-by: Sangbeom Kim Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 65f980ef2870..dd3b3eac0805 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -63,7 +63,7 @@ config SND_SOC_SAMSUNG_SMDK_WM8580 config SND_SOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210 || MACH_SMDK4212) select SND_SOC_WM8994 select SND_SAMSUNG_I2S help @@ -158,7 +158,7 @@ config SND_SOC_GONI_AQUILA_WM8994 config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310) + depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310 || MACH_SMDK4212) select SND_SAMSUNG_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. @@ -173,7 +173,7 @@ config SND_SOC_SMDK_WM8580_PCM config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310) + depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310 || MACH_SMDK4212) select SND_SOC_WM8994 select SND_SAMSUNG_PCM help -- cgit v1.2.3 From 0a9d1385282841ba33d5815f06ed5b62fde7ff8c Mon Sep 17 00:00:00 2001 From: Ben Gardiner Date: Fri, 26 Aug 2011 12:02:44 -0400 Subject: ASoC: davinci-mcasp: add support for unsigned PCM formats Although the McASP supports sign-extending samples in RX or TX [1]; the davinci-mcasp driver does not touch the {R,X}PBIT or {R,X}PAD field of the {R,X}FMT registers meaning that the McASP will serialize the bytes it is given regardless of their signedness. So supporting unsigned formats is as simple as adding them to the metadata of the davinci-mcasp driver. Update the FMTBITs reported in the snd_soc_dai_driver and also update the case statements in davinci-mcasp's hw_params() function so that the McASP can be connected to CODECs that use unsigned values. [1] http://www.ti.com/lit/ug/sprufm1/sprufm1.pdf Signed-off-by: Ben Gardiner Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 8566238db2a5..7173df254a91 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -732,16 +732,19 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, davinci_hw_param(dev, substream->stream); switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_S8: dma_params->data_type = 1; word_length = DAVINCI_AUDIO_WORD_8; break; + case SNDRV_PCM_FORMAT_U16_LE: case SNDRV_PCM_FORMAT_S16_LE: dma_params->data_type = 2; word_length = DAVINCI_AUDIO_WORD_16; break; + case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; word_length = DAVINCI_AUDIO_WORD_32; @@ -818,6 +821,13 @@ static struct snd_soc_dai_ops davinci_mcasp_dai_ops = { }; +#define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + static struct snd_soc_dai_driver davinci_mcasp_dai[] = { { .name = "davinci-mcasp.0", @@ -825,17 +835,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .channels_min = 2, .channels_max = 2, .rates = DAVINCI_MCASP_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, + .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, .channels_max = 2, .rates = DAVINCI_MCASP_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, + .formats = DAVINCI_MCASP_PCM_FMTS, }, .ops = &davinci_mcasp_dai_ops, @@ -846,7 +852,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .channels_min = 1, .channels_max = 384, .rates = DAVINCI_MCASP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = DAVINCI_MCASP_PCM_FMTS, }, .ops = &davinci_mcasp_dai_ops, }, -- cgit v1.2.3 From 18a4eef3d5a7ba0f96e5a7a84a5ab4827a52dffd Mon Sep 17 00:00:00 2001 From: susan gao Date: Fri, 26 Aug 2011 12:14:14 -0700 Subject: ASoC: Add 3D stereo support for wm8996 My first patch to ASoC ever! If I did something wrong, blame Ian. Signed-off-by: Susan Gao Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 3c9080f78266..e5e46075c365 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -414,6 +414,7 @@ static const DECLARE_TLV_DB_SCALE(out_digital_tlv, -1200, 150, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -900, 75, 0); static const DECLARE_TLV_DB_SCALE(spk_tlv, -900, 150, 0); static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(threedstereo_tlv, -1600, 183, 1); static const char *sidetone_hpf_text[] = { "2.9kHz", "1.5kHz", "735Hz", "403Hz", "196Hz", "98Hz", "49Hz" @@ -608,6 +609,14 @@ SOC_SINGLE("DAC High Performance Switch", WM8996_OVERSAMPLING, 0, 1, 0), SOC_SINGLE("DAC Soft Mute Switch", WM8996_DAC_SOFTMUTE, 1, 1, 0), SOC_SINGLE("DAC Slow Soft Mute Switch", WM8996_DAC_SOFTMUTE, 0, 1, 0), +SOC_SINGLE("DSP1 3D Stereo Switch", WM8996_DSP1_RX_FILTERS_2, 8, 1, 0), +SOC_SINGLE("DSP2 3D Stereo Switch", WM8996_DSP2_RX_FILTERS_2, 8, 1, 0), + +SOC_SINGLE_TLV("DSP1 3D Stereo Volume", WM8996_DSP1_RX_FILTERS_2, 10, 15, + 0, threedstereo_tlv), +SOC_SINGLE_TLV("DSP2 3D Stereo Volume", WM8996_DSP2_RX_FILTERS_2, 10, 15, + 0, threedstereo_tlv), + SOC_DOUBLE_TLV("Digital Output 1 Volume", WM8996_DAC1_HPOUT1_VOLUME, 0, 4, 8, 0, out_digital_tlv), SOC_DOUBLE_TLV("Digital Output 2 Volume", WM8996_DAC2_HPOUT2_VOLUME, 0, 4, -- cgit v1.2.3 From d4ba7854c9ea388e83731ee4b3c6546076f70f9d Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Thu, 25 Aug 2011 15:54:55 +0200 Subject: ASoC: imx-ssi: use dma_writecombine consistently If the channel is allocated as writecombine, then mmaping it should also use writecombine. Also, add a proper device for the call. Ported from a similar fix for mach-mxs. Signed-off-by: Wolfram Sang Acked-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 10a8e2783751..3b8d5cd2516a 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -357,8 +357,8 @@ int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; int ret; - ret = dma_mmap_coherent(NULL, vma, runtime->dma_area, - runtime->dma_addr, runtime->dma_bytes); + ret = dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, runtime->dma_area, -- cgit v1.2.3 From 35dcf58634cf696966cdec69f62b14a7f49a8c42 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Thu, 25 Aug 2011 15:54:56 +0200 Subject: ASoC: imx: use more robust checking of available streams Replace the channels_min check with a check for the relevant substream being present. Suggested here [1] when mxs implemented the audio-support. [1] http://www.spinics.net/lists/arm-kernel/msg133010.html Signed-off-by: Wolfram Sang Acked-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-fiq.c | 11 +++++------ sound/soc/imx/imx-ssi.c | 4 ++-- 2 files changed, 7 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 309c59e6fb6c..ac790e87e231 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -243,23 +243,22 @@ static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; + struct snd_pcm_substream *substream; int ret; ret = imx_pcm_new(rtd); if (ret) return ret; - if (dai->driver->playback.channels_min) { - struct snd_pcm_substream *substream = - pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (substream) { struct snd_dma_buffer *buf = &substream->dma_buffer; imx_ssi_fiq_tx_buffer = (unsigned long)buf->area; } - if (dai->driver->capture.channels_min) { - struct snd_pcm_substream *substream = - pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + if (substream) { struct snd_dma_buffer *buf = &substream->dma_buffer; imx_ssi_fiq_rx_buffer = (unsigned long)buf->area; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 3b8d5cd2516a..4297cb6af42e 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -399,14 +399,14 @@ int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) card->dev->dma_mask = &imx_pcm_dmamask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = imx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = imx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) -- cgit v1.2.3 From b92d150baeb876cbf0fd4cc8d997f005cc57e3f4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 27 Aug 2011 18:24:14 +0200 Subject: ASoC: soc_codec_reg_show use snd_soc_codec_readable_register Use snd_soc_codec_readable_register instead of open-coding it. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4065d4e84ea3..fc7fff3604f7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -143,7 +143,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf, step = codec->driver->reg_cache_step; for (i = 0; i < codec->driver->reg_cache_size; i += step) { - if (codec->readable_register && !codec->readable_register(codec, i)) + if (!snd_soc_codec_readable_register(codec, i)) continue; if (codec->driver->display_register) { count += codec->driver->display_register(codec, buf + count, -- cgit v1.2.3 From 9a810e959bbff538dea30f27b261a5c4346b4cc7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 30 Aug 2011 15:33:28 +0100 Subject: ASoC: Remove unused mutex from WM9090 driver Signed-off-by: Mark Brown --- sound/soc/codecs/wm9090.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 4de12203e611..f2f3077928da 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -139,7 +139,6 @@ static const u16 wm9090_reg_defaults[] = { /* This struct is used to save the context */ struct wm9090_priv { - struct mutex mutex; struct wm9090_platform_data pdata; void *control_data; }; @@ -663,7 +662,6 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm9090); wm9090->control_data = i2c; - mutex_init(&wm9090->mutex); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9090, NULL, 0); -- cgit v1.2.3 From 1e3ad571d56ab96e5fab87cea71c0e657d4708cb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Aug 2011 14:10:42 +0100 Subject: ASoC: Remove redundant -codec from WM8776 driver name Signed-off-by: Mark Brown Acked-by: Timur Tabi --- sound/soc/codecs/wm8776.c | 4 ++-- sound/soc/fsl/p1022_ds.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 8e7953b1b790..367a990e6cc4 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -481,7 +481,7 @@ static int __devexit wm8776_spi_remove(struct spi_device *spi) static struct spi_driver wm8776_spi_driver = { .driver = { - .name = "wm8776-codec", + .name = "wm8776", .owner = THIS_MODULE, }, .probe = wm8776_spi_probe, @@ -525,7 +525,7 @@ MODULE_DEVICE_TABLE(i2c, wm8776_i2c_id); static struct i2c_driver wm8776_i2c_driver = { .driver = { - .name = "wm8776-codec", + .name = "wm8776", .owner = THIS_MODULE, }, .probe = wm8776_i2c_probe, diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index fcb862eb0c73..e8849ed36cbd 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -267,7 +267,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) if (bus < 0) return bus; - snprintf(buf, len, "%s-codec.%u-%04x", temp, bus, addr); + snprintf(buf, len, "%s.%u-%04x", temp, bus, addr); return 0; } -- cgit v1.2.3 From 13c7d08f54cc83c1cd9884c5e142e485b748de18 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Aug 2011 14:12:15 +0100 Subject: ASoC: Add device tree binding for WM8770 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8770.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 19b92baa9e8c..aa05e6507f84 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include @@ -684,6 +685,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8770 = { .reg_cache_default = wm8770_reg_defs }; +static const struct of_device_id wm8770_of_match[] = { + { .compatible = "wlf,wm8770", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8770_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8770_spi_probe(struct spi_device *spi) { @@ -715,6 +722,7 @@ static struct spi_driver wm8770_spi_driver = { .driver = { .name = "wm8770", .owner = THIS_MODULE, + .of_match_table = wm8770_of_match, }, .probe = wm8770_spi_probe, .remove = __devexit_p(wm8770_spi_remove) -- cgit v1.2.3 From b6de431556023a6ed901a27284f15fff2e043598 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Aug 2011 14:14:45 +0100 Subject: ASoC: Add device tree binding for WM8776 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8776.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 367a990e6cc4..0cfbfc1dc093 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -452,6 +453,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8776 = { .reg_cache_default = wm8776_reg, }; +static const struct of_device_id wm8776_of_match[] = { + { .compatible = "wlf,wm8776", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8776_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8776_spi_probe(struct spi_device *spi) { @@ -483,6 +490,7 @@ static struct spi_driver wm8776_spi_driver = { .driver = { .name = "wm8776", .owner = THIS_MODULE, + .of_match_table = wm8776_of_match, }, .probe = wm8776_spi_probe, .remove = __devexit_p(wm8776_spi_remove), @@ -527,6 +535,7 @@ static struct i2c_driver wm8776_i2c_driver = { .driver = { .name = "wm8776", .owner = THIS_MODULE, + .of_match_table = wm8776_of_match, }, .probe = wm8776_i2c_probe, .remove = __devexit_p(wm8776_i2c_remove), -- cgit v1.2.3 From d2dd0540c1dab1ebe4192e69d8dbfcf018ff02b2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Aug 2011 14:23:05 +0100 Subject: ASoC: Add device tree binding for WM8804 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 9a5e67c5a6bd..9ee072b85975 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -717,6 +718,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .volatile_register = wm8804_volatile }; +static const struct of_device_id wm8804_of_match[] = { + { .compatible = "wlf,wm8804", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8804_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8804_spi_probe(struct spi_device *spi) { @@ -748,6 +755,7 @@ static struct spi_driver wm8804_spi_driver = { .driver = { .name = "wm8804", .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, }, .probe = wm8804_spi_probe, .remove = __devexit_p(wm8804_spi_remove) @@ -792,6 +800,7 @@ static struct i2c_driver wm8804_i2c_driver = { .driver = { .name = "wm8804", .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, }, .probe = wm8804_i2c_probe, .remove = __devexit_p(wm8804_i2c_remove), -- cgit v1.2.3 From da1c6ea6cf85544292c30295c70a89e8555358bc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 24 Aug 2011 20:09:01 +0100 Subject: ASoC: Allow source specification for CODEC level sysclk Similarly to PLLs/FLLs some modern CODECs provide selectable system clock sources. When the clock is the clock for a DAI we do not usually need to identify which clock is being configured so can use clk_id for the source clock but with CODEC wide system clocks we will need to specify both the clock being configured and the source. Add a source argument to the CODEC driver set_sysclk() operation to reflect this. As this operation is not as widely used as the DAI set_sysclk() operation the change is not very invasive. We probably ought to go and make the same alternation for DAIs at some point. Signed-off-by: Mark Brown --- sound/soc/codecs/adav80x.c | 3 ++- sound/soc/codecs/wm9081.c | 4 ++-- sound/soc/samsung/speyside.c | 2 +- sound/soc/soc-core.c | 8 +++++--- 4 files changed, 10 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 300c04b70e71..f9f08948e5e8 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -523,7 +523,8 @@ static int adav80x_hw_params(struct snd_pcm_substream *substream, } static int adav80x_set_sysclk(struct snd_soc_codec *codec, - int clk_id, unsigned int freq, int dir) + int clk_id, int source, + unsigned int freq, int dir) { struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index a4691321f9b3..f32ab1ee9647 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1120,8 +1120,8 @@ static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } -static int wm9081_set_sysclk(struct snd_soc_codec *codec, - int clk_id, unsigned int freq, int dir) +static int wm9081_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) { struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index bfed1ff7093f..09df8afbb447 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -223,7 +223,7 @@ static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) snd_soc_dapm_nc_pin(dapm, "LINEOUT"); /* At any time the WM9081 is active it will have this clock */ - return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, + return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, 48000 * 256, 0); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fc7fff3604f7..4ec93d1df047 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2670,7 +2670,7 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, if (dai->driver && dai->driver->ops->set_sysclk) return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); else if (dai->codec && dai->codec->driver->set_sysclk) - return dai->codec->driver->set_sysclk(dai->codec, clk_id, + return dai->codec->driver->set_sysclk(dai->codec, clk_id, 0, freq, dir); else return -EINVAL; @@ -2681,16 +2681,18 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); * snd_soc_codec_set_sysclk - configure CODEC system or master clock. * @codec: CODEC * @clk_id: DAI specific clock ID + * @source: Source for the clock * @freq: new clock frequency in Hz * @dir: new clock direction - input/output. * * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. */ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, - unsigned int freq, int dir) + int source, unsigned int freq, int dir) { if (codec->driver->set_sysclk) - return codec->driver->set_sysclk(codec, clk_id, freq, dir); + return codec->driver->set_sysclk(codec, clk_id, source, + freq, dir); else return -EINVAL; } -- cgit v1.2.3 From 7b4615ba8108649ed30804450eb054925e347ad3 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 3 Sep 2011 13:41:47 +0800 Subject: ASoC: sn95031: Fix the logic to find free channel In the case of no free channel available, current implementation returns 0 instead of negative errno. This patch fixes the logic to return -EINVAL if no free channel available. Signed-off-by: Axel Lin Acked-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 84ffdebb8a8b..b4f1cb494ffc 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -79,7 +79,7 @@ static void configure_adc(struct snd_soc_codec *sn95031_codec, int val) */ static int find_free_channel(struct snd_soc_codec *sn95031_codec) { - int ret = 0, i, value; + int i, value; /* check whether ADC is enabled */ value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); @@ -91,12 +91,10 @@ static int find_free_channel(struct snd_soc_codec *sn95031_codec) for (i = 0; i < SN95031_ADC_CHANLS_MAX; i++) { value = snd_soc_read(sn95031_codec, SN95031_ADC_CHNL_START_ADDR + i); - if (value & SN95031_STOPBIT_MASK) { - ret = i; + if (value & SN95031_STOPBIT_MASK) break; - } } - return (ret > SN95031_ADC_LOOP_MAX) ? (-EINVAL) : ret; + return (i == SN95031_ADC_CHANLS_MAX) ? (-EINVAL) : i; } /* Initialize the ADC for reading micbias values. Can sleep. */ -- cgit v1.2.3 From 4ed0d012c945af332ede1f9853db5184b6c24da1 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 2 Sep 2011 21:47:41 +0800 Subject: ASoC: Add missing platform_device_put in raumfeld_audio_init error path Signed-off-by: Axel Lin Acked-by: Daniel Mack Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/pxa/raumfeld.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 1a591f1ebfbd..b899a3bc8f42 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -306,8 +306,10 @@ static int __init raumfeld_audio_init(void) &snd_soc_raumfeld_connector); ret = platform_device_add(raumfeld_audio_device); - if (ret < 0) + if (ret < 0) { + platform_device_put(raumfeld_audio_device); return ret; + } raumfeld_enable_audio(true); return 0; -- cgit v1.2.3 From 27b6d92a2434219caad8e3a0bc7b5ea5e44c3702 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 4 Sep 2011 09:35:47 -0700 Subject: ASoC: Check that WM8996 FLL started even if we don't have the IRQ We can directly read the FLL lock status on WM8996 so even if we don't have an interrupt wired up we can still verify that the FLL started successfully. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8996.c | 26 +++++++++++++++++++------- 1 file changed, 19 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index e5e46075c365..e386d25aba82 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2054,7 +2054,7 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, struct i2c_client *i2c = to_i2c_client(codec->dev); struct _fll_div fll_div; unsigned long timeout; - int ret, reg; + int ret, reg, retry; /* Any change? */ if (source == wm8996->fll_src && Fref == wm8996->fll_fref && @@ -2141,17 +2141,29 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, else timeout = msecs_to_jiffies(2); - /* Allow substantially longer if we've actually got the IRQ */ + /* Allow substantially longer if we've actually got the IRQ, poll + * at a slightly higher rate if we don't. + */ if (i2c->irq) - timeout *= 1000; + timeout *= 10; + else + timeout /= 2; - ret = wait_for_completion_timeout(&wm8996->fll_lock, timeout); + for (retry = 0; retry < 10; retry++) { + ret = wait_for_completion_timeout(&wm8996->fll_lock, + timeout); + if (ret != 0) { + WARN_ON(!i2c->irq); + break; + } - if (ret == 0 && i2c->irq) { + ret = snd_soc_read(codec, WM8996_INTERRUPT_RAW_STATUS_2); + if (ret & WM8996_FLL_LOCK_STS) + break; + } + if (retry == 10) { dev_err(codec->dev, "Timed out waiting for FLL\n"); ret = -ETIMEDOUT; - } else { - ret = 0; } dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); -- cgit v1.2.3 From c8f4b7fd681b236a1878dffaebc47f4f18c66d80 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 4 Sep 2011 22:48:27 +0800 Subject: ASoC: alc5623: Remove unused mutex Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/alc5623.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index eecffb548947..05173159507e 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -41,7 +41,6 @@ MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); struct alc5623_priv { enum snd_soc_control_type control_type; void *control_data; - struct mutex mutex; u8 id; unsigned int sysclk; u16 reg_cache[ALC5623_VENDOR_ID2+2]; @@ -1052,7 +1051,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, alc5623); alc5623->control_data = client; alc5623->control_type = SND_SOC_I2C; - mutex_init(&alc5623->mutex); ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5623, &alc5623_dai, 1); -- cgit v1.2.3 From 3ed464659a1d83a87f4ef79fab4d85a8dcf677c9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 6 Sep 2011 16:39:17 -0700 Subject: ASoC: Remove unused step size from debugfs CODEC write function We don't use the step size so there's no need to work it out. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4ec93d1df047..10e5cdeeb18e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -244,7 +244,6 @@ static ssize_t codec_reg_write_file(struct file *file, size_t buf_size; char *start = buf; unsigned long reg, value; - int step = 1; struct snd_soc_codec *codec = file->private_data; buf_size = min(count, (sizeof(buf)-1)); @@ -252,9 +251,6 @@ static ssize_t codec_reg_write_file(struct file *file, return -EFAULT; buf[buf_size] = 0; - if (codec->driver->reg_cache_step) - step = codec->driver->reg_cache_step; - while (*start == ' ') start++; reg = simple_strtoul(start, &start, 16); -- cgit v1.2.3 From 0f73644f372281f2f9c33a0459dfdfc8bc77fbda Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 6 Sep 2011 10:37:48 +0800 Subject: ASoC: ad1980: Return proper error if vendor id mismatch Return -ENODEV instead of 0 if vendor id mismatch. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ad1980.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 923b364a3e41..4c0fc30a4ccb 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -200,18 +200,22 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) } /* Read out vendor ID to make sure it is ad1980 */ - if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144) + if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144) { + ret = -ENODEV; goto reset_err; + } vendor_id2 = ac97_read(codec, AC97_VENDOR_ID2); if (vendor_id2 != 0x5370) { - if (vendor_id2 != 0x5374) + if (vendor_id2 != 0x5374) { + ret = -ENODEV; goto reset_err; - else + } else { printk(KERN_WARNING "ad1980: " "Found AD1981 - only 2/2 IN/OUT Channels " "supported\n"); + } } /* unmute captures and playbacks volume */ -- cgit v1.2.3 From c2f6fce33ed6146d0442b6ad8c8b827f507d3aec Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Tue, 6 Sep 2011 15:21:34 +0800 Subject: ASoC: sst_platform: trivial coding style fix Signed-off-by: Lu Guanqun Acked-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 3e7826058efe..d99f2535dc4c 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -469,7 +469,7 @@ static struct platform_driver sst_platform_driver = { static int __init sst_soc_platform_init(void) { pr_debug("sst_soc_platform_init called\n"); - return platform_driver_register(&sst_platform_driver); + return platform_driver_register(&sst_platform_driver); } module_init(sst_soc_platform_init); -- cgit v1.2.3 From 22be504aaa4a3133d81e3fb0c4287960aea19c37 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Tue, 6 Sep 2011 15:21:38 +0800 Subject: ASoC: sst_platform: using builtin function Use the builtin snd_soc_set_runtime_hwparams() instead of assigning it by myself. Signed-off-by: Lu Guanqun Acked-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index d99f2535dc4c..af666ae671ae 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -226,13 +226,14 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) static int sst_platform_open(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime; + struct snd_pcm_runtime *runtime = substream->runtime; struct sst_runtime_stream *stream; int ret_val = 0; pr_debug("sst_platform_open called\n"); - runtime = substream->runtime; - runtime->hw = sst_platform_pcm_hw; + + snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); + stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; -- cgit v1.2.3 From 283e42e0114aba331b0055839f6277a4a7cfbc64 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Tue, 6 Sep 2011 15:21:43 +0800 Subject: ASoC: sst_platform: fix memory leak snd_pcm_hw_constraint_integer() could return -1, in this case, sst platform is not opened successfully. However the corresponding close callback isn't able to be called later on to release these two allocated memories, thus resulting in memory leak. This patch moves the check for hardware contraints earlier, thus resolving this issue. Signed-off-by: Lu Guanqun Acked-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index af666ae671ae..9925d20ab0a3 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -233,6 +233,10 @@ static int sst_platform_open(struct snd_pcm_substream *substream) pr_debug("sst_platform_open called\n"); snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); + ret_val = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret_val < 0) + return ret_val; stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) @@ -260,8 +264,8 @@ static int sst_platform_open(struct snd_pcm_substream *substream) return ret_val; } runtime->private_data = stream; - return snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); + + return 0; } static int sst_platform_close(struct snd_pcm_substream *substream) -- cgit v1.2.3 From 694741471b8df3734e01ecae7650be60ec111c2c Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 9 Sep 2011 10:15:37 +0800 Subject: ASoC: playpaq_wm8510: Return proper error if clk_get fails Return proper error instead of 0 if clk_get fails. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/atmel/playpaq_wm8510.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 1aac2f4dbcf6..2909bfaed265 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -383,14 +383,17 @@ static int __init playpaq_asoc_init(void) _gclk0 = clk_get(NULL, "gclk0"); if (IS_ERR(_gclk0)) { _gclk0 = NULL; + ret = PTR_ERR(_gclk0); goto err_gclk0; } _pll0 = clk_get(NULL, "pll0"); if (IS_ERR(_pll0)) { _pll0 = NULL; + ret = PTR_ERR(_pll0); goto err_pll0; } - if (clk_set_parent(_gclk0, _pll0)) { + ret = clk_set_parent(_gclk0, _pll0); + if (ret) { pr_warning("snd-soc-playpaq: " "Failed to set PLL0 as parent for DAC clock\n"); goto err_set_clk; -- cgit v1.2.3 From 30ab1e78864ca5781de5b1fb501bed9df2c215f1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 5 Sep 2011 20:46:33 +0200 Subject: ASoC: ad193x: Setup regmap read and write flag masks for SPI Currently register read-back for the ad193x is broken, because it expects bit 0 of the upper byte to be set to indicate a read operation, while the regmap default for SPI is to use bit 7. This patch also addresses another oddity of the device. There are SPI and I2C versions of this codec. In both cases the registers are 8-bit wide and numbered from 0x0 to 0x10, but in the SPI case there is also a so called 'global address' which is prefixed in-front of the register address. The global address mimics I2C behaviour and includes a static device address the and the read/write flag. This basically extends the register address to an 16-bit value numbered from 0x800 to 0x810. These are the register numbers which are currently used by the driver. This works, because I2C will ignore the upper 8 bits of the register, but it is still a bit confusing, as there are no such register numbers in the I2C case. The approach taken by this patch is to number the registers from 0x00 to 0x10 and encode the global address for SPI mode into the read and write flag masks. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 65 +++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/ad193x.h | 34 ++++++++++++------------- 2 files changed, 71 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index eedb6f5e5823..f934670199a5 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -23,7 +23,7 @@ /* codec private data */ struct ad193x_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk; }; @@ -349,10 +349,8 @@ static int ad193x_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - if (ad193x->control_type == SND_SOC_I2C) - ret = snd_soc_codec_set_cache_io(codec, 8, 8, ad193x->control_type); - else - ret = snd_soc_codec_set_cache_io(codec, 16, 8, ad193x->control_type); + codec->control_data = ad193x->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; @@ -388,6 +386,14 @@ static struct snd_soc_codec_driver soc_codec_dev_ad193x = { }; #if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config ad193x_spi_regmap_config = { + .val_bits = 8, + .reg_bits = 16, + .read_flag_mask = 0x09, + .write_flag_mask = 0x08, +}; + static int __devinit ad193x_spi_probe(struct spi_device *spi) { struct ad193x_priv *ad193x; @@ -397,20 +403,36 @@ static int __devinit ad193x_spi_probe(struct spi_device *spi) if (ad193x == NULL) return -ENOMEM; + ad193x->regmap = regmap_init_spi(spi, &ad193x_spi_regmap_config); + if (IS_ERR(ad193x->regmap)) { + ret = PTR_ERR(ad193x->regmap); + goto err_free; + } + spi_set_drvdata(spi, ad193x); - ad193x->control_type = SND_SOC_SPI; ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_ad193x, &ad193x_dai, 1); if (ret < 0) - kfree(ad193x); + goto err_regmap_exit; + + return 0; + +err_regmap_exit: + regmap_exit(ad193x->regmap); +err_free: + kfree(ad193x); + return ret; } static int __devexit ad193x_spi_remove(struct spi_device *spi) { + struct ad193x_priv *ad193x = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); + regmap_exit(ad193x->regmap); + kfree(ad193x); return 0; } @@ -425,6 +447,12 @@ static struct spi_driver ad193x_spi_driver = { #endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +static const struct regmap_config ad193x_i2c_regmap_config = { + .val_bits = 8, + .reg_bits = 8, +}; + static const struct i2c_device_id ad193x_id[] = { { "ad1936", 0 }, { "ad1937", 0 }, @@ -442,20 +470,35 @@ static int __devinit ad193x_i2c_probe(struct i2c_client *client, if (ad193x == NULL) return -ENOMEM; + ad193x->regmap = regmap_init_i2c(client, &ad193x_i2c_regmap_config); + if (IS_ERR(ad193x->regmap)) { + ret = PTR_ERR(ad193x->regmap); + goto err_free; + } + i2c_set_clientdata(client, ad193x); - ad193x->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_ad193x, &ad193x_dai, 1); if (ret < 0) - kfree(ad193x); + goto err_regmap_exit; + + return 0; + +err_regmap_exit: + regmap_exit(ad193x->regmap); +err_free: + kfree(ad193x); return ret; } static int __devexit ad193x_i2c_remove(struct i2c_client *client) { + struct ad193x_priv *ad193x = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + regmap_exit(ad193x->regmap); + kfree(ad193x); return 0; } diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index cccc2e8e5fbd..536e5f2b136e 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -9,20 +9,20 @@ #ifndef __AD193X_H__ #define __AD193X_H__ -#define AD193X_PLL_CLK_CTRL0 0x800 +#define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 #define AD193X_PLL_INPUT_MASK (~0x6) #define AD193X_PLL_INPUT_256 (0 << 1) #define AD193X_PLL_INPUT_384 (1 << 1) #define AD193X_PLL_INPUT_512 (2 << 1) #define AD193X_PLL_INPUT_768 (3 << 1) -#define AD193X_PLL_CLK_CTRL1 0x801 -#define AD193X_DAC_CTRL0 0x802 +#define AD193X_PLL_CLK_CTRL1 0x01 +#define AD193X_DAC_CTRL0 0x02 #define AD193X_DAC_POWERDOWN 0x01 #define AD193X_DAC_SERFMT_MASK 0xC0 #define AD193X_DAC_SERFMT_STEREO (0 << 6) #define AD193X_DAC_SERFMT_TDM (1 << 6) -#define AD193X_DAC_CTRL1 0x803 +#define AD193X_DAC_CTRL1 0x03 #define AD193X_DAC_2_CHANNELS 0 #define AD193X_DAC_4_CHANNELS 1 #define AD193X_DAC_8_CHANNELS 2 @@ -33,11 +33,11 @@ #define AD193X_DAC_BCLK_MASTER (1 << 5) #define AD193X_DAC_LEFT_HIGH (1 << 3) #define AD193X_DAC_BCLK_INV (1 << 7) -#define AD193X_DAC_CTRL2 0x804 +#define AD193X_DAC_CTRL2 0x04 #define AD193X_DAC_WORD_LEN_SHFT 3 #define AD193X_DAC_WORD_LEN_MASK 0x18 #define AD193X_DAC_MASTER_MUTE 1 -#define AD193X_DAC_CHNL_MUTE 0x805 +#define AD193X_DAC_CHNL_MUTE 0x05 #define AD193X_DACL1_MUTE 0 #define AD193X_DACR1_MUTE 1 #define AD193X_DACL2_MUTE 2 @@ -46,28 +46,28 @@ #define AD193X_DACR3_MUTE 5 #define AD193X_DACL4_MUTE 6 #define AD193X_DACR4_MUTE 7 -#define AD193X_DAC_L1_VOL 0x806 -#define AD193X_DAC_R1_VOL 0x807 -#define AD193X_DAC_L2_VOL 0x808 -#define AD193X_DAC_R2_VOL 0x809 -#define AD193X_DAC_L3_VOL 0x80a -#define AD193X_DAC_R3_VOL 0x80b -#define AD193X_DAC_L4_VOL 0x80c -#define AD193X_DAC_R4_VOL 0x80d -#define AD193X_ADC_CTRL0 0x80e +#define AD193X_DAC_L1_VOL 0x06 +#define AD193X_DAC_R1_VOL 0x07 +#define AD193X_DAC_L2_VOL 0x08 +#define AD193X_DAC_R2_VOL 0x09 +#define AD193X_DAC_L3_VOL 0x0a +#define AD193X_DAC_R3_VOL 0x0b +#define AD193X_DAC_L4_VOL 0x0c +#define AD193X_DAC_R4_VOL 0x0d +#define AD193X_ADC_CTRL0 0x0e #define AD193X_ADC_POWERDOWN 0x01 #define AD193X_ADC_HIGHPASS_FILTER 1 #define AD193X_ADCL1_MUTE 2 #define AD193X_ADCR1_MUTE 3 #define AD193X_ADCL2_MUTE 4 #define AD193X_ADCR2_MUTE 5 -#define AD193X_ADC_CTRL1 0x80f +#define AD193X_ADC_CTRL1 0x0f #define AD193X_ADC_SERFMT_MASK 0x60 #define AD193X_ADC_SERFMT_STEREO (0 << 5) #define AD193X_ADC_SERFMT_TDM (1 << 5) #define AD193X_ADC_SERFMT_AUX (2 << 5) #define AD193X_ADC_WORD_LEN_MASK 0x3 -#define AD193X_ADC_CTRL2 0x810 +#define AD193X_ADC_CTRL2 0x10 #define AD193X_ADC_2_CHANNELS 0 #define AD193X_ADC_4_CHANNELS 1 #define AD193X_ADC_8_CHANNELS 2 -- cgit v1.2.3 From c83495af6395446b81da54b17a479557ad0b2fc8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Sep 2011 10:05:18 +0100 Subject: ASoC: Disable WM8996 CPVDD supply when not in use The WM8996 only requires CPVDD when the charge pump is active so control it separately to the other supplies, only enabling it when the charge pump is active. This will result in a small power saving on systems which are able to provide independent software control of the supply. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8996.c | 38 ++++++++++++++++++++++++++++++-------- 1 file changed, 30 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index e386d25aba82..9d0ab87bad96 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -41,12 +41,11 @@ #define HPOUT2L 4 #define HPOUT2R 8 -#define WM8996_NUM_SUPPLIES 4 +#define WM8996_NUM_SUPPLIES 3 static const char *wm8996_supply_names[WM8996_NUM_SUPPLIES] = { "DBVDD", "AVDD1", "AVDD2", - "CPVDD", }; struct wm8996_priv { @@ -71,6 +70,7 @@ struct wm8996_priv { struct regulator_bulk_data supplies[WM8996_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8996_NUM_SUPPLIES]; + struct regulator *cpvdd; struct wm8996_pdata pdata; @@ -112,7 +112,6 @@ static int wm8996_regulator_event_##n(struct notifier_block *nb, \ WM8996_REGULATOR_EVENT(0) WM8996_REGULATOR_EVENT(1) WM8996_REGULATOR_EVENT(2) -WM8996_REGULATOR_EVENT(3) static const u16 wm8996_reg[WM8996_MAX_REGISTER] = { [WM8996_SOFTWARE_RESET] = 0x8996, @@ -670,16 +669,29 @@ SOC_SINGLE_TLV("DSP2 EQ B5 Volume", WM8996_DSP2_RX_EQ_GAINS_2, 6, 31, 0, static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = w->codec; + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(wm8996->cpvdd); + if (ret != 0) + dev_err(codec->dev, "Failed to enable CPVDD: %d\n", + ret); + break; case SND_SOC_DAPM_POST_PMU: msleep(5); break; + case SND_SOC_DAPM_POST_PMD: + regulator_disable_deferred(wm8996->cpvdd, 20); + break; default: BUG(); - return -EINVAL; + ret = -EINVAL; } - return 0; + return ret; } static int rmv_short_event(struct snd_soc_dapm_widget *w, @@ -988,7 +1000,8 @@ SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event, - SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICB1 Audio", WM8996_MICBIAS_1, 4, 1, NULL, 0), @@ -2573,7 +2586,13 @@ static int wm8996_probe(struct snd_soc_codec *codec) wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0; wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1; wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2; - wm8996->disable_nb[3].notifier_call = wm8996_regulator_event_3; + + wm8996->cpvdd = regulator_get(&i2c->dev, "CPVDD"); + if (IS_ERR(wm8996->cpvdd)) { + ret = PTR_ERR(wm8996->cpvdd); + dev_err(&i2c->dev, "Failed to get CPVDD: %d\n", ret); + goto err_get; + } /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) { @@ -2590,7 +2609,7 @@ static int wm8996_probe(struct snd_soc_codec *codec) wm8996->supplies); if (ret != 0) { dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; + goto err_cpvdd; } if (wm8996->pdata.ldo_ena >= 0) { @@ -2833,6 +2852,8 @@ err_enable: gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); +err_cpvdd: + regulator_put(wm8996->cpvdd); err_get: regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); err: @@ -2856,6 +2877,7 @@ static int wm8996_remove(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) regulator_unregister_notifier(wm8996->supplies[i].consumer, &wm8996->disable_nb[i]); + regulator_put(wm8996->cpvdd); regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); return 0; -- cgit v1.2.3 From 32d2a0c17d81016215381d337dad876dc972ee83 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 10 Sep 2011 22:36:17 -0700 Subject: ASoC: Correct channel numbers for WM8996 AIF2 The AIF1 channels are numbered from zero than one; do the same thing for AIF2 too. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8996.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 9d0ab87bad96..7280a10d5fe7 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1059,14 +1059,14 @@ SND_SOC_DAPM_DAC("DAC2R", NULL, WM8996_POWER_MANAGEMENT_5, 2, 0), SND_SOC_DAPM_DAC("DAC1L", NULL, WM8996_POWER_MANAGEMENT_5, 1, 0), SND_SOC_DAPM_DAC("DAC1R", NULL, WM8996_POWER_MANAGEMENT_5, 0, 0), -SND_SOC_DAPM_AIF_IN("AIF2RX1", "AIF2 Playback", 1, +SND_SOC_DAPM_AIF_IN("AIF2RX1", "AIF2 Playback", 0, WM8996_POWER_MANAGEMENT_4, 9, 0), -SND_SOC_DAPM_AIF_IN("AIF2RX0", "AIF2 Playback", 2, +SND_SOC_DAPM_AIF_IN("AIF2RX0", "AIF2 Playback", 1, WM8996_POWER_MANAGEMENT_4, 8, 0), -SND_SOC_DAPM_AIF_IN("AIF2TX1", "AIF2 Capture", 1, +SND_SOC_DAPM_AIF_IN("AIF2TX1", "AIF2 Capture", 0, WM8996_POWER_MANAGEMENT_6, 9, 0), -SND_SOC_DAPM_AIF_IN("AIF2TX0", "AIF2 Capture", 2, +SND_SOC_DAPM_AIF_IN("AIF2TX0", "AIF2 Capture", 1, WM8996_POWER_MANAGEMENT_6, 8, 0), SND_SOC_DAPM_AIF_IN("AIF1RX5", "AIF1 Playback", 5, -- cgit v1.2.3 From 5e538ecade22a5ec4c8e18d494db0ecf924254eb Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Tue, 13 Sep 2011 12:59:37 -0500 Subject: ASoC: improve asynchronous mode support in the fsl_ssi driver The Freescale SSI audio controller supports "synchronous" and "asynchronous" modes. In synchronous mode, playback and capture use the same input clock, so sample rates must be the same during simultaneous playback and capture. Unfortunately, the code which supports asynchronous mode is just broken in various ways. In particular, it was constraining sample sizes as well as the sample rate. The fix also allows us to simplify the code by eliminating the 'asynchronous', 'playback', and 'capture' variables that were used to keep track of playback and capture streams. Unfortunately, it turns out that simulataneous playback and record does not actually work on the only platform that supports asynchronous mode: the Freescale P1022DS reference board. If a second stream is started, the SSI grinds to halt for both streams. This is true even if the P1022 is configured for synchronous mode, so it's likely a hardware problem that needs to be worked around. Signed-off-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 145 +++++++++++++++++++++++------------------------- 1 file changed, 68 insertions(+), 77 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 06ac2b92faf3..0268cf989736 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -78,7 +78,6 @@ * @second_stream: pointer to second stream * @playback: the number of playback streams opened * @capture: the number of capture streams opened - * @asynchronous: 0=synchronous mode, 1=asynchronous mode * @cpu_dai: the CPU DAI for this device * @dev_attr: the sysfs device attribute structure * @stats: SSI statistics @@ -90,9 +89,6 @@ struct fsl_ssi_private { unsigned int irq; struct snd_pcm_substream *first_stream; struct snd_pcm_substream *second_stream; - unsigned int playback; - unsigned int capture; - int asynchronous; unsigned int fifo_depth; struct snd_soc_dai_driver cpu_dai_drv; struct device_attribute dev_attr; @@ -281,15 +277,19 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_ssi_private *ssi_private = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; /* * If this is the first stream opened, then request the IRQ * and initialize the SSI registers. */ - if (!ssi_private->playback && !ssi_private->capture) { + if (!ssi_private->first_stream) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + ssi_private->first_stream = substream; + /* * Section 16.5 of the MPC8610 reference manual says that the * SSI needs to be disabled before updating the registers we set @@ -306,7 +306,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE - | (ssi_private->asynchronous ? 0 : CCSR_SSI_SCR_SYN)); + | (synchronous ? CCSR_SSI_SCR_SYN : 0)); out_be32(&ssi->stcr, CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -323,7 +323,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * master. */ - /* 4. Enable the interrupts and DMA requests */ + /* Enable the interrupts and DMA requests */ out_be32(&ssi->sier, SIER_FLAGS); /* @@ -352,58 +352,47 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * this is bad is because at this point, the PCM driver has not * finished initializing the DMA controller. */ - } - - if (!ssi_private->first_stream) - ssi_private->first_stream = substream; - else { - /* This is the second stream open, so we need to impose sample - * rate and maybe sample size constraints. Note that this can - * cause a race condition if the second stream is opened before - * the first stream is fully initialized. - * - * We provide some protection by checking to make sure the first - * stream is initialized, but it's not perfect. ALSA sometimes - * re-initializes the driver with a different sample rate or - * size. If the second stream is opened before the first stream - * has received its final parameters, then the second stream may - * be constrained to the wrong sample rate or size. - * - * FIXME: This code does not handle opening and closing streams - * repeatedly. If you open two streams and then close the first - * one, you may not be able to open another stream until you - * close the second one as well. - */ - struct snd_pcm_runtime *first_runtime = - ssi_private->first_stream->runtime; - - if (!first_runtime->sample_bits) { - dev_err(substream->pcm->card->dev, - "set sample size in %s stream first\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? "capture" : "playback"); - return -EAGAIN; - } + } else { + if (synchronous) { + struct snd_pcm_runtime *first_runtime = + ssi_private->first_stream->runtime; + /* + * This is the second stream open, and we're in + * synchronous mode, so we need to impose sample + * sample size constraints. This is because STCCR is + * used for playback and capture in synchronous mode, + * so there's no way to specify different word + * lengths. + * + * Note that this can cause a race condition if the + * second stream is opened before the first stream is + * fully initialized. We provide some protection by + * checking to make sure the first stream is + * initialized, but it's not perfect. ALSA sometimes + * re-initializes the driver with a different sample + * rate or size. If the second stream is opened + * before the first stream has received its final + * parameters, then the second stream may be + * constrained to the wrong sample rate or size. + */ + if (!first_runtime->sample_bits) { + dev_err(substream->pcm->card->dev, + "set sample size in %s stream first\n", + substream->stream == + SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + return -EAGAIN; + } - /* If we're in synchronous mode, then we need to constrain - * the sample size as well. We don't support independent sample - * rates in asynchronous mode. - */ - if (!ssi_private->asynchronous) snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, first_runtime->sample_bits, first_runtime->sample_bits); + } ssi_private->second_stream = substream; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ssi_private->playback++; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ssi_private->capture++; - return 0; } @@ -424,24 +413,35 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int sample_size = + snd_pcm_format_width(params_format(hw_params)); + u32 wl = CCSR_SSI_SxCCR_WL(sample_size); + int enabled = in_be32(&ssi->scr) & CCSR_SSI_SCR_SSIEN; - if (substream == ssi_private->first_stream) { - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - unsigned int sample_size = - snd_pcm_format_width(params_format(hw_params)); - u32 wl = CCSR_SSI_SxCCR_WL(sample_size); + /* + * If we're in synchronous mode, and the SSI is already enabled, + * then STCCR is already set properly. + */ + if (enabled && ssi_private->cpu_dai_drv.symmetric_rates) + return 0; - /* The SSI should always be disabled at this points (SSIEN=0) */ + /* + * FIXME: The documentation says that SxCCR[WL] should not be + * modified while the SSI is enabled. The only time this can + * happen is if we're trying to do simultaneous playback and + * capture in asynchronous mode. Unfortunately, I have been enable + * to get that to work at all on the P1022DS. Therefore, we don't + * bother to disable/enable the SSI when setting SxCCR[WL], because + * the SSI will stop anyway. Maybe one day, this will get fixed. + */ - /* In synchronous mode, the SSI uses STCCR for capture */ - if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || - !ssi_private->asynchronous) - clrsetbits_be32(&ssi->stccr, - CCSR_SSI_SxCCR_WL_MASK, wl); - else - clrsetbits_be32(&ssi->srccr, - CCSR_SSI_SxCCR_WL_MASK, wl); - } + /* In synchronous mode, the SSI uses STCCR for capture */ + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || + ssi_private->cpu_dai_drv.symmetric_rates) + clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); + else + clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); return 0; } @@ -464,7 +464,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) setbits32(&ssi->scr, @@ -500,12 +499,6 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ssi_private->playback--; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ssi_private->capture--; - if (ssi_private->first_stream == substream) ssi_private->first_stream = ssi_private->second_stream; @@ -514,7 +507,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, /* * If this is the last active substream, disable the SSI. */ - if (!ssi_private->playback && !ssi_private->capture) { + if (!ssi_private->first_stream) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); @@ -688,9 +681,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) } /* Are the RX and the TX clocks locked? */ - if (of_find_property(np, "fsl,ssi-asynchronous", NULL)) - ssi_private->asynchronous = 1; - else + if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) ssi_private->cpu_dai_drv.symmetric_rates = 1; /* Determine the FIFO depth. */ -- cgit v1.2.3 From 7803e329bb8357afe94e8e5c3f78478d6a98d1b5 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 15 Sep 2011 10:36:54 +0800 Subject: ASoC: samsung: Fix checking return value of clk_get clk_get() returns a pointer to the struct clk or an ERR_PTR(). This patch also use PTR_ERR() for return value. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/samsung/s3c2412-i2s.c | 4 ++-- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- sound/soc/samsung/s3c24xx_uda134x.c | 8 ++++---- 3 files changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 841ab14c1100..7ab8e2c29216 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -69,10 +69,10 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; s3c2412_i2s.iis_cclk = clk_get(dai->dev, "i2sclk"); - if (s3c2412_i2s.iis_cclk == NULL) { + if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); iounmap(s3c2412_i2s.regs); - return -ENODEV; + return PTR_ERR(s3c2412_i2s.iis_cclk); } /* Set MPLL as the source for IIS CLK */ diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 63d8849d80bd..21c92e2e3007 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -383,10 +383,10 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) return -ENXIO; s3c24xx_i2s.iis_clk = clk_get(dai->dev, "iis"); - if (s3c24xx_i2s.iis_clk == NULL) { + if (IS_ERR(s3c24xx_i2s.iis_clk)) { pr_err("failed to get iis_clock\n"); iounmap(s3c24xx_i2s.regs); - return -ENODEV; + return PTR_ERR(s3c24xx_i2s.iis_clk); } clk_enable(s3c24xx_i2s.iis_clk); diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index dc9d551f6788..65c1cfd47d8a 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -66,17 +66,17 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) pr_debug("%s %d\n", __func__, clk_users); if (clk_users == 0) { xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); - if (!xtal) { + if (IS_ERR(xtal)) { printk(KERN_ERR "%s cannot get xtal\n", __func__); - ret = -EBUSY; + ret = PTR_ERR(xtal); } else { pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, "pclk"); - if (!pclk) { + if (IS_ERR(pclk)) { printk(KERN_ERR "%s cannot get pclk\n", __func__); clk_put(xtal); - ret = -EBUSY; + ret = PTR_ERR(pclk); } } if (!ret) { -- cgit v1.2.3 From f998f257c92216df314610dd5aebc5f5d23e6ec0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 15 Sep 2011 10:52:11 +0100 Subject: ASoC: Fix WM8996 DC servo operation without IRQ We need to count the timeout down. Reported-by: Axel Lin Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8996.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 7280a10d5fe7..5174874a5f7b 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -719,7 +719,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) { struct i2c_client *i2c = to_i2c_client(codec->dev); struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); - int i, ret; + int ret; unsigned long timeout = 200; snd_soc_write(codec, WM8996_DC_SERVO_2, mask); @@ -734,15 +734,12 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) } else { msleep(1); - if (--i) { - timeout = 0; - break; - } + timeout--; } ret = snd_soc_read(codec, WM8996_DC_SERVO_2); dev_dbg(codec->dev, "DC servo state: %x\n", ret); - } while (ret & mask); + } while (timeout && ret & mask); if (timeout == 0) dev_err(codec->dev, "DC servo timed out for %x\n", mask); -- cgit v1.2.3 From d1dc698a54259cb454284456483b45f67c865cf8 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Tue, 13 Sep 2011 12:59:35 -0500 Subject: ASoC: support sample sizes properly in the WM8776 codec driver Use snd_pcm_format_width() to determine the sample size, instead of checking specify sample formats and assuming that those are the only valid format. This change adds support for big-endian architectures (which use the _BE formats) and the packed 24-bit format (SNDRV_PCM_FORMAT_S24_3xE). [Fixed single letter variable name legibility problem -- broonie] Signed-off-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8776.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 0cfbfc1dc093..5b17627aab0f 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -216,8 +216,6 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream, int ratio_shift, master; int i; - iface = 0; - switch (dai->driver->id) { case WM8776_DAI_DAC: iface_reg = WM8776_DACIFCTRL; @@ -233,20 +231,23 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Set word length */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(params_format(params))) { + case 16: + iface = 0; + case 20: + iface = 0x10; break; - case SNDRV_PCM_FORMAT_S20_3LE: - iface |= 0x10; + case 24: + iface = 0x20; break; - case SNDRV_PCM_FORMAT_S24_LE: - iface |= 0x20; - break; - case SNDRV_PCM_FORMAT_S32_LE: - iface |= 0x30; + case 32: + iface = 0x30; break; + default: + dev_err(codec->dev, "Unsupported sample size: %i\n", + snd_pcm_format_width(params_format(params))); + return -EINVAL; } /* Only need to set MCLK/LRCLK ratio if we're master */ -- cgit v1.2.3 From 62e4f7d1386f3e9c126fc7febc719d504b3e344b Mon Sep 17 00:00:00 2001 From: Mika Westerberg Date: Sun, 11 Sep 2011 12:28:52 +0300 Subject: ASoC: snappercl15: convert to use snd_soc_register_card() Current method for machine driver to register with the ASoC core is to use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Mika Westerberg Reviewed-by: Ryan Mallon Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/ep93xx/snappercl15.c | 53 ++++++++++++++++++++++++++++-------------- 1 file changed, 36 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c index c8aa8a5003ca..f74ac54c285a 100644 --- a/sound/soc/ep93xx/snappercl15.c +++ b/sound/soc/ep93xx/snappercl15.c @@ -104,37 +104,56 @@ static struct snd_soc_card snd_soc_snappercl15 = { .num_links = 1, }; -static struct platform_device *snappercl15_snd_device; - -static int __init snappercl15_init(void) +static int __devinit snappercl15_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_snappercl15; int ret; - if (!machine_is_snapper_cl15()) - return -ENODEV; - ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, EP93XX_SYSCON_I2SCLKDIV_ORIDE | EP93XX_SYSCON_I2SCLKDIV_SPOL); if (ret) return ret; - snappercl15_snd_device = platform_device_alloc("soc-audio", -1); - if (!snappercl15_snd_device) - return -ENOMEM; - - platform_set_drvdata(snappercl15_snd_device, &snd_soc_snappercl15); - ret = platform_device_add(snappercl15_snd_device); - if (ret) - platform_device_put(snappercl15_snd_device); + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + ep93xx_i2s_release(); + } return ret; } -static void __exit snappercl15_exit(void) +static int __devexit snappercl15_remove(struct platform_device *pdev) { - platform_device_unregister(snappercl15_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); ep93xx_i2s_release(); + + return 0; +} + +static struct platform_driver snappercl15_driver = { + .driver = { + .name = "snappercl15-audio", + .owner = THIS_MODULE, + }, + .probe = snappercl15_probe, + .remove = __devexit_p(snappercl15_remove), +}; + +static int __init snappercl15_init(void) +{ + return platform_driver_register(&snappercl15_driver); +} + +static void __exit snappercl15_exit(void) +{ + platform_driver_unregister(&snappercl15_driver); } module_init(snappercl15_init); @@ -143,4 +162,4 @@ module_exit(snappercl15_exit); MODULE_AUTHOR("Ryan Mallon"); MODULE_DESCRIPTION("ALSA SoC Snapper CL15"); MODULE_LICENSE("GPL"); - +MODULE_ALIAS("platform:snappercl15-audio"); -- cgit v1.2.3 From 9306816954e6e7c3986495f9141a04d5393f998a Mon Sep 17 00:00:00 2001 From: Mika Westerberg Date: Sun, 11 Sep 2011 12:28:49 +0300 Subject: ASoC: ep93xx-pcm: add MODULE_ALIAS To get the PCM module loaded automatically by udev et al. we need to add a proper MODULE_ALIAS. Signed-off-by: Mika Westerberg Reviewed-by: Ryan Mallon Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/ep93xx/ep93xx-pcm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 8dfd3ad84b19..d00230a591b1 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -355,3 +355,4 @@ module_exit(ep93xx_soc_platform_exit); MODULE_AUTHOR("Ryan Mallon"); MODULE_DESCRIPTION("EP93xx ALSA PCM interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:ep93xx-pcm-audio"); -- cgit v1.2.3 From 5a0a03c5ef79cc14336c789c183822902519d8da Mon Sep 17 00:00:00 2001 From: Mika Westerberg Date: Sun, 11 Sep 2011 12:28:50 +0300 Subject: ASoC: simone: convert to use snd_soc_register_card() Current method for machine driver to register with the ASoC core is to use snd_soc_register_card() instead of creating a "soc-audio" platform device. In addition we use platform_device_register_simple() to create a platform device for the codec. This function will handle putting and deleting the device automatically which simplifies the error handling in the machine driver. Signed-off-by: Mika Westerberg Reviewed-by: Ryan Mallon Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/ep93xx/simone.c | 64 ++++++++++++++++++++++++++--------------------- 1 file changed, 36 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/ep93xx/simone.c b/sound/soc/ep93xx/simone.c index 286817946c56..968cb316d511 100644 --- a/sound/soc/ep93xx/simone.c +++ b/sound/soc/ep93xx/simone.c @@ -39,53 +39,61 @@ static struct snd_soc_card snd_soc_simone = { }; static struct platform_device *simone_snd_ac97_device; -static struct platform_device *simone_snd_device; -static int __init simone_init(void) +static int __devinit simone_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_simone; int ret; - if (!machine_is_sim_one()) - return -ENODEV; - - simone_snd_ac97_device = platform_device_alloc("ac97-codec", -1); - if (!simone_snd_ac97_device) - return -ENOMEM; + simone_snd_ac97_device = platform_device_register_simple("ac97-codec", + -1, NULL, 0); + if (IS_ERR(simone_snd_ac97_device)) + return PTR_ERR(simone_snd_ac97_device); - ret = platform_device_add(simone_snd_ac97_device); - if (ret) - goto fail1; + card->dev = &pdev->dev; - simone_snd_device = platform_device_alloc("soc-audio", -1); - if (!simone_snd_device) { - ret = -ENOMEM; - goto fail2; + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + platform_device_unregister(simone_snd_ac97_device); } - platform_set_drvdata(simone_snd_device, &snd_soc_simone); - ret = platform_device_add(simone_snd_device); - if (ret) - goto fail3; + return ret; +} + +static int __devexit simone_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + platform_device_unregister(simone_snd_ac97_device); return 0; +} -fail3: - platform_device_put(simone_snd_device); -fail2: - platform_device_del(simone_snd_ac97_device); -fail1: - platform_device_put(simone_snd_ac97_device); - return ret; +static struct platform_driver simone_driver = { + .driver = { + .name = "simone-audio", + .owner = THIS_MODULE, + }, + .probe = simone_probe, + .remove = __devexit_p(simone_remove), +}; + +static int __init simone_init(void) +{ + return platform_driver_register(&simone_driver); } module_init(simone_init); static void __exit simone_exit(void) { - platform_device_unregister(simone_snd_device); - platform_device_unregister(simone_snd_ac97_device); + platform_driver_unregister(&simone_driver); } module_exit(simone_exit); MODULE_DESCRIPTION("ALSA SoC Simplemachines Sim.One"); MODULE_AUTHOR("Mika Westerberg "); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:simone-audio"); -- cgit v1.2.3 From 8a386ca26d51d477729f2b54e9d81bd97da4467e Mon Sep 17 00:00:00 2001 From: Mika Westerberg Date: Sun, 11 Sep 2011 12:28:51 +0300 Subject: ASoC: edb93xx: convert to use snd_soc_register_card() Current method for machine driver to register with the ASoC core is to use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Mika Westerberg Reviewed-by: Ryan Mallon Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/ep93xx/edb93xx.c | 60 +++++++++++++++++++++++++--------------------- 1 file changed, 33 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index d3aa15119d26..0134d4e9131c 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -28,12 +28,6 @@ #include #include "ep93xx-pcm.h" -#define edb93xx_has_audio() (machine_is_edb9301() || \ - machine_is_edb9302() || \ - machine_is_edb9302a() || \ - machine_is_edb9307a() || \ - machine_is_edb9315a()) - static int edb93xx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -94,49 +88,61 @@ static struct snd_soc_card snd_soc_edb93xx = { .num_links = 1, }; -static struct platform_device *edb93xx_snd_device; - -static int __init edb93xx_init(void) +static int __devinit edb93xx_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_edb93xx; int ret; - if (!edb93xx_has_audio()) - return -ENODEV; - ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, EP93XX_SYSCON_I2SCLKDIV_ORIDE | EP93XX_SYSCON_I2SCLKDIV_SPOL); if (ret) return ret; - edb93xx_snd_device = platform_device_alloc("soc-audio", -1); - if (!edb93xx_snd_device) { - ret = -ENOMEM; - goto free_i2s; + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + ep93xx_i2s_release(); } - platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx); - ret = platform_device_add(edb93xx_snd_device); - if (ret) - goto device_put; + return ret; +} - return 0; +static int __devexit edb93xx_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); -device_put: - platform_device_put(edb93xx_snd_device); -free_i2s: + snd_soc_unregister_card(card); ep93xx_i2s_release(); - return ret; + + return 0; +} + +static struct platform_driver edb93xx_driver = { + .driver = { + .name = "edb93xx-audio", + .owner = THIS_MODULE, + }, + .probe = edb93xx_probe, + .remove = __devexit_p(edb93xx_remove), +}; + +static int __init edb93xx_init(void) +{ + return platform_driver_register(&edb93xx_driver); } module_init(edb93xx_init); static void __exit edb93xx_exit(void) { - platform_device_unregister(edb93xx_snd_device); - ep93xx_i2s_release(); + platform_driver_unregister(&edb93xx_driver); } module_exit(edb93xx_exit); MODULE_AUTHOR("Alexander Sverdlin "); MODULE_DESCRIPTION("ALSA SoC EDB93xx"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:edb93xx-audio"); -- cgit v1.2.3 From be4ff9612271ac63e16bb2a8e6666e62538b60ea Mon Sep 17 00:00:00 2001 From: Ben Gardiner Date: Fri, 9 Sep 2011 17:06:05 -0400 Subject: ASoC: davinci-pcm: trivial: replace link with actual chan/link The ambiguously named variable 'link' is used as a temporary throughout davinci-pcm -- its presence makes grepping (and groking) the code difficult. Replace link with the value of link in almost all sites. The exception is a couple places where the last-assigned link/chan needs to be returned by a function -- in these cases, rename to last_link. Signed-off-by: Ben Gardiner Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-pcm.c | 123 +++++++++++++++++++--------------------- 1 file changed, 59 insertions(+), 64 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index a49e667373bc..d5fe08cc5db7 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -180,7 +180,6 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - int link = prtd->asp_link[0]; unsigned int period_size; unsigned int dma_offset; dma_addr_t dma_pos; @@ -198,7 +197,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) fifo_level = prtd->params->fifo_level; pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " - "dma_ptr = %x period_size=%x\n", link, dma_pos, period_size); + "dma_ptr = %x period_size=%x\n", prtd->asp_link[0], dma_pos, + period_size); data_type = prtd->params->data_type; count = period_size / data_type; @@ -222,17 +222,19 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) } acnt = prtd->params->acnt; - edma_set_src(link, src, INCR, W8BIT); - edma_set_dest(link, dst, INCR, W8BIT); + edma_set_src(prtd->asp_link[0], src, INCR, W8BIT); + edma_set_dest(prtd->asp_link[0], dst, INCR, W8BIT); - edma_set_src_index(link, src_bidx, src_cidx); - edma_set_dest_index(link, dst_bidx, dst_cidx); + edma_set_src_index(prtd->asp_link[0], src_bidx, src_cidx); + edma_set_dest_index(prtd->asp_link[0], dst_bidx, dst_cidx); if (!fifo_level) - edma_set_transfer_params(link, acnt, count, 1, 0, ASYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, + ASYNC); else - edma_set_transfer_params(link, acnt, fifo_level, count, - fifo_level, ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, + count, fifo_level, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) @@ -305,7 +307,6 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) unsigned int acnt = params->acnt; /* divide by 2 for ping/pong */ unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1; - int link = prtd->asp_link[1]; unsigned int fifo_level = prtd->params->fifo_level; unsigned int count; if ((data_type == 0) || (data_type > 4)) { @@ -316,28 +317,26 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) dma_addr_t asp_src_pong = iram_dma->addr + ping_size; ram_src_cidx = ping_size; ram_dst_cidx = -ping_size; - edma_set_src(link, asp_src_pong, INCR, W8BIT); + edma_set_src(prtd->asp_link[1], asp_src_pong, INCR, W8BIT); - link = prtd->asp_link[0]; - edma_set_src_index(link, data_type, data_type * fifo_level); - link = prtd->asp_link[1]; - edma_set_src_index(link, data_type, data_type * fifo_level); + edma_set_src_index(prtd->asp_link[0], data_type, + data_type * fifo_level); + edma_set_src_index(prtd->asp_link[1], data_type, + data_type * fifo_level); - link = prtd->ram_link; - edma_set_src(link, runtime->dma_addr, INCR, W32BIT); + edma_set_src(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); } else { dma_addr_t asp_dst_pong = iram_dma->addr + ping_size; ram_src_cidx = -ping_size; ram_dst_cidx = ping_size; - edma_set_dest(link, asp_dst_pong, INCR, W8BIT); + edma_set_dest(prtd->asp_link[1], asp_dst_pong, INCR, W8BIT); - link = prtd->asp_link[0]; - edma_set_dest_index(link, data_type, data_type * fifo_level); - link = prtd->asp_link[1]; - edma_set_dest_index(link, data_type, data_type * fifo_level); + edma_set_dest_index(prtd->asp_link[0], data_type, + data_type * fifo_level); + edma_set_dest_index(prtd->asp_link[1], data_type, + data_type * fifo_level); - link = prtd->ram_link; - edma_set_dest(link, runtime->dma_addr, INCR, W32BIT); + edma_set_dest(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); } if (!fifo_level) { @@ -354,10 +353,9 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) count, fifo_level, ABSYNC); } - link = prtd->ram_link; - edma_set_src_index(link, ping_size, ram_src_cidx); - edma_set_dest_index(link, ping_size, ram_dst_cidx); - edma_set_transfer_params(link, ping_size, 2, + edma_set_src_index(prtd->ram_link, ping_size, ram_src_cidx); + edma_set_dest_index(prtd->ram_link, ping_size, ram_dst_cidx); + edma_set_transfer_params(prtd->ram_link, ping_size, 2, runtime->periods, 2, ASYNC); /* init master params */ @@ -406,32 +404,32 @@ static int request_ping_pong(struct snd_pcm_substream *substream, { dma_addr_t asp_src_ping; dma_addr_t asp_dst_ping; - int link; + int ret; struct davinci_pcm_dma_params *params = prtd->params; /* Request ram master channel */ - link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, + ret = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, davinci_pcm_dma_irq, substream, prtd->params->ram_chan_q); - if (link < 0) + if (ret < 0) goto exit1; /* Request ram link channel */ - link = prtd->ram_link = edma_alloc_slot( + ret = prtd->ram_link = edma_alloc_slot( EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit2; - link = prtd->asp_link[1] = edma_alloc_slot( + ret = prtd->asp_link[1] = edma_alloc_slot( EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit3; prtd->ram_link2 = -1; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - link = prtd->ram_link2 = edma_alloc_slot( + ret = prtd->ram_link2 = edma_alloc_slot( EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit4; } /* circle ping-pong buffers */ @@ -448,36 +446,33 @@ static int request_ping_pong(struct snd_pcm_substream *substream, asp_dst_ping = iram_dma->addr; } /* ping */ - link = prtd->asp_link[0]; - edma_set_src(link, asp_src_ping, INCR, W16BIT); - edma_set_dest(link, asp_dst_ping, INCR, W16BIT); - edma_set_src_index(link, 0, 0); - edma_set_dest_index(link, 0, 0); + edma_set_src(prtd->asp_link[0], asp_src_ping, INCR, W16BIT); + edma_set_dest(prtd->asp_link[0], asp_dst_ping, INCR, W16BIT); + edma_set_src_index(prtd->asp_link[0], 0, 0); + edma_set_dest_index(prtd->asp_link[0], 0, 0); - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); prtd->asp_params.opt |= TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(link, &prtd->asp_params); + edma_write_slot(prtd->asp_link[0], &prtd->asp_params); /* pong */ - link = prtd->asp_link[1]; - edma_set_src(link, asp_src_ping, INCR, W16BIT); - edma_set_dest(link, asp_dst_ping, INCR, W16BIT); - edma_set_src_index(link, 0, 0); - edma_set_dest_index(link, 0, 0); + edma_set_src(prtd->asp_link[1], asp_src_ping, INCR, W16BIT); + edma_set_dest(prtd->asp_link[1], asp_dst_ping, INCR, W16BIT); + edma_set_src_index(prtd->asp_link[1], 0, 0); + edma_set_dest_index(prtd->asp_link[1], 0, 0); - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[1], &prtd->asp_params); prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); /* interrupt after every pong completion */ prtd->asp_params.opt |= TCINTEN | TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(link, &prtd->asp_params); + edma_write_slot(prtd->asp_link[1], &prtd->asp_params); /* ram */ - link = prtd->ram_link; - edma_set_src(link, iram_dma->addr, INCR, W32BIT); - edma_set_dest(link, iram_dma->addr, INCR, W32BIT); + edma_set_src(prtd->ram_link, iram_dma->addr, INCR, W32BIT); + edma_set_dest(prtd->ram_link, iram_dma->addr, INCR, W32BIT); pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u," "for asp:%u %u %u\n", __func__, prtd->ram_channel, prtd->ram_link, prtd->ram_link2, @@ -494,7 +489,7 @@ exit2: edma_free_channel(prtd->ram_channel); prtd->ram_channel = -1; exit1: - return link; + return ret; } static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) @@ -502,22 +497,22 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) struct snd_dma_buffer *iram_dma; struct davinci_runtime_data *prtd = substream->runtime->private_data; struct davinci_pcm_dma_params *params = prtd->params; - int link; + int ret; if (!params) return -ENODEV; /* Request asp master DMA channel */ - link = prtd->asp_channel = edma_alloc_channel(params->channel, + ret = prtd->asp_channel = edma_alloc_channel(params->channel, davinci_pcm_dma_irq, substream, prtd->params->asp_chan_q); - if (link < 0) + if (ret < 0) goto exit1; /* Request asp link channels */ - link = prtd->asp_link[0] = edma_alloc_slot( + ret = prtd->asp_link[0] = edma_alloc_slot( EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit2; iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; @@ -537,17 +532,17 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) * the buffer and its length (ccnt) ... use it as a template * so davinci_pcm_enqueue_dma() takes less time in IRQ. */ - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); prtd->asp_params.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); - prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(link) << 5; - edma_write_slot(link, &prtd->asp_params); + prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5; + edma_write_slot(prtd->asp_link[0], &prtd->asp_params); return 0; exit2: edma_free_channel(prtd->asp_channel); prtd->asp_channel = -1; exit1: - return link; + return ret; } static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -- cgit v1.2.3 From 275708f88d3dce0728e2d099b5de8ebc0f15c69a Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 9 Sep 2011 16:47:00 +0800 Subject: ASoC: tpa6130a2: Remove obsolete cleanup for clientdata The i2c core will clear the clientdata pointer automatically, we don't have to set the `data' field to NULL in remove() or if probe() failed anymore. Signed-off-by: Axel Lin Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 239e0c461068..b2572c451c35 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -446,7 +446,6 @@ err_regulator: gpio_free(data->power_gpio); err_gpio: kfree(data); - i2c_set_clientdata(tpa6130a2_client, NULL); tpa6130a2_client = NULL; return ret; -- cgit v1.2.3 From 6fa0c25bf450fa9b2325e1eff9c58627330026ab Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 16 Sep 2011 10:46:33 +0800 Subject: ASoC: wm8995: Return -EINVAL if device ID mismatch Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 5ad873fda814..05f779532b1b 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1642,6 +1642,7 @@ static int wm8995_probe(struct snd_soc_codec *codec) if (ret != 0x8995) { dev_err(codec->dev, "Invalid device ID: %#x\n", ret); + ret = -EINVAL; goto err_reg_enable; } -- cgit v1.2.3 From 0547d0f3dadfd9a3eb8523630fef52612ab14de4 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 16 Sep 2011 10:47:37 +0800 Subject: ASoC: wm8995: Remove unused i2c variable in wm8995_remove() Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 05f779532b1b..74ae5995a786 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1573,9 +1573,7 @@ static int wm8995_resume(struct snd_soc_codec *codec) static int wm8995_remove(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; - struct i2c_client *i2c; - i2c = container_of(codec->dev, struct i2c_client, dev); wm8995 = snd_soc_codec_get_drvdata(codec); wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; -- cgit v1.2.3 From 0016226d03fa8d695fb7b933ea1810503d774820 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Fri, 16 Sep 2011 09:16:54 -0500 Subject: ASoC: support all possible sample rates in the WM8776 driver The WM8776 supports a continuous range of sample rates rather than discrete values and supports a wider range of sample rates on the playback path than is currently supported. Update the constraints on the DAIs to reflect this. Signed-off-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8776.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 5b17627aab0f..00d8846fae8a 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -322,11 +322,6 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define WM8776_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) - - #define WM8776_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -351,7 +346,9 @@ static struct snd_soc_dai_driver wm8776_dai[] = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = WM8776_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 32000, + .rate_max = 192000, .formats = WM8776_FORMATS, }, .ops = &wm8776_dac_ops, @@ -363,7 +360,9 @@ static struct snd_soc_dai_driver wm8776_dai[] = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = WM8776_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 32000, + .rate_max = 96000, .formats = WM8776_FORMATS, }, .ops = &wm8776_adc_ops, -- cgit v1.2.3 From 53daf20893b18000768aaa617a60b987fa39f875 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 5 Sep 2011 10:51:05 -0700 Subject: ASoC: Display the error code when we fail to add a DAPM control Useful for diagnostics. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4859ad77eac7..4a440b52dd7a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -584,8 +584,8 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) name + prefix_len, prefix); ret = snd_ctl_add(card, kcontrol); if (ret < 0) { - dev_err(dapm->dev, - "asoc: failed to add kcontrol %s\n", w->name); + dev_err(dapm->dev, "failed to add kcontrol %s: %d\n", + w->name, ret); kfree(wlist); return ret; } -- cgit v1.2.3 From 8259df12fd3f3429648411bfff37dfbb34a2d9b2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Sep 2011 17:55:06 +0100 Subject: ASoC: WM8996 only needs bandgap for analogue functionality Rather than managing the bandgap in the bias level control use a supply widget as we only actually need to enable it for analogue paths, not fully digital ones. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8996.c | 40 +++++++++++++++++++++++++++++----------- 1 file changed, 29 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 5174874a5f7b..c584e3e6a6fe 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -666,6 +666,25 @@ SOC_SINGLE_TLV("DSP2 EQ B5 Volume", WM8996_DSP2_RX_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static int bg_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(2); + break; + default: + BUG(); + ret = -EINVAL; + } + + return ret; +} + static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -999,7 +1018,8 @@ SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - +SND_SOC_DAPM_SUPPLY("Bandgap", WM8996_POWER_MANAGEMENT_1, WM8996_BG_ENA_SHIFT, + 0, bg_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICB1 Audio", WM8996_MICBIAS_1, 4, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("MICB2 Audio", WM8996_MICBIAS_2, 4, 1, NULL, 0), @@ -1159,18 +1179,22 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "MICB1", NULL, "LDO2" }, { "MICB1", NULL, "MICB1 Audio" }, + { "MICB1", NULL, "Bandgap" }, { "MICB2", NULL, "LDO2" }, { "MICB2", NULL, "MICB2 Audio" }, + { "MICB2", NULL, "Bandgap" }, { "IN1L PGA", NULL, "IN2LN" }, { "IN1L PGA", NULL, "IN2LP" }, { "IN1L PGA", NULL, "IN1LN" }, { "IN1L PGA", NULL, "IN1LP" }, + { "IN1L PGA", NULL, "Bandgap" }, { "IN1R PGA", NULL, "IN2RN" }, { "IN1R PGA", NULL, "IN2RP" }, { "IN1R PGA", NULL, "IN1RN" }, { "IN1R PGA", NULL, "IN1RP" }, + { "IN1R PGA", NULL, "Bandgap" }, { "ADCL", NULL, "IN1L PGA" }, @@ -1304,6 +1328,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "DAC2R", NULL, "DAC2R Mixer" }, { "HPOUT2L PGA", NULL, "Charge Pump" }, + { "HPOUT2L PGA", NULL, "Bandgap" }, { "HPOUT2L PGA", NULL, "DAC2L" }, { "HPOUT2L_DLY", NULL, "HPOUT2L PGA" }, { "HPOUT2L_DCS", NULL, "HPOUT2L_DLY" }, @@ -1311,6 +1336,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "HPOUT2L_RMV_SHORT", NULL, "HPOUT2L_OUTP" }, { "HPOUT2R PGA", NULL, "Charge Pump" }, + { "HPOUT2R PGA", NULL, "Bandgap" }, { "HPOUT2R PGA", NULL, "DAC2R" }, { "HPOUT2R_DLY", NULL, "HPOUT2R PGA" }, { "HPOUT2R_DCS", NULL, "HPOUT2R_DLY" }, @@ -1318,6 +1344,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "HPOUT2R_RMV_SHORT", NULL, "HPOUT2R_OUTP" }, { "HPOUT1L PGA", NULL, "Charge Pump" }, + { "HPOUT1L PGA", NULL, "Bandgap" }, { "HPOUT1L PGA", NULL, "DAC1L" }, { "HPOUT1L_DLY", NULL, "HPOUT1L PGA" }, { "HPOUT1L_DCS", NULL, "HPOUT1L_DLY" }, @@ -1325,6 +1352,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "HPOUT1L_RMV_SHORT", NULL, "HPOUT1L_OUTP" }, { "HPOUT1R PGA", NULL, "Charge Pump" }, + { "HPOUT1R PGA", NULL, "Bandgap" }, { "HPOUT1R PGA", NULL, "DAC1R" }, { "HPOUT1R_DLY", NULL, "HPOUT1R PGA" }, { "HPOUT1R_DCS", NULL, "HPOUT1R_DLY" }, @@ -1643,14 +1671,7 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - break; - case SND_SOC_BIAS_PREPARE: - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { - snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, - WM8996_BG_ENA, WM8996_BG_ENA); - msleep(2); - } break; case SND_SOC_BIAS_STANDBY: @@ -1673,9 +1694,6 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, codec->cache_only = false; snd_soc_cache_sync(codec); } - - snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, - WM8996_BG_ENA, 0); break; case SND_SOC_BIAS_OFF: -- cgit v1.2.3 From 0b684cc14a791accdd6d97cb68242ab5009ece3e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 4 Sep 2011 07:50:31 -0700 Subject: ASoC: Initial WM8996 headphone impedance measurement support The WM8996 can measure the impedance of accessories connected to the headphone output. Implement initial support for this, measuring the left channel impedance when an accessory is detected and using this to distinguish between a line load and a headphone load. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8996.c | 137 +++++++++++++++++++++++++++++++++++++--------- 1 file changed, 112 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index c584e3e6a6fe..cd1ba9637c01 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2350,12 +2350,94 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, /* Enable interrupts and we're off */ snd_soc_update_bits(codec, WM8996_INTERRUPT_STATUS_2_MASK, - WM8996_IM_MICD_EINT, 0); + WM8996_IM_MICD_EINT | WM8996_HP_DONE_EINT, 0); return 0; } EXPORT_SYMBOL_GPL(wm8996_detect); +static void wm8996_hpdet_irq(struct snd_soc_codec *codec) +{ + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + int val, reg, report; + + /* Assume headphone in error conditions; we need to report + * something or we stall our state machine. + */ + report = SND_JACK_HEADPHONE; + + reg = snd_soc_read(codec, WM8996_HEADPHONE_DETECT_2); + if (reg < 0) { + dev_err(codec->dev, "Failed to read HPDET status\n"); + goto out; + } + + if (!(reg & WM8996_HP_DONE)) { + dev_err(codec->dev, "Got HPDET IRQ but HPDET is busy\n"); + goto out; + } + + val = reg & WM8996_HP_LVL_MASK; + + dev_dbg(codec->dev, "HPDET measured %d ohms\n", val); + + /* If we've got high enough impedence then report as line, + * otherwise assume headphone. + */ + if (val >= 126) + report = SND_JACK_LINEOUT; + else + report = SND_JACK_HEADPHONE; + +out: + if (wm8996->jack_mic) + report |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(wm8996->jack, report, + SND_JACK_LINEOUT | SND_JACK_HEADSET); + + wm8996->detecting = false; + + /* If the output isn't running re-clamp it */ + if (!(snd_soc_read(codec, WM8996_POWER_MANAGEMENT_1) & + (WM8996_HPOUT1L_ENA | WM8996_HPOUT1R_RMV_SHORT))) + snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT, 0); + + /* Go back to looking at the microphone */ + snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_1, + WM8996_JD_MODE_MASK, 0); + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA, + WM8996_MICD_ENA); + + snd_soc_dapm_disable_pin(&codec->dapm, "Bandgap"); + snd_soc_dapm_sync(&codec->dapm); +} + +static void wm8996_hpdet_start(struct snd_soc_codec *codec) +{ + /* Unclamp the output, we can't measure while we're shorting it */ + snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT); + + /* We need bandgap for HPDET */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "Bandgap"); + snd_soc_dapm_sync(&codec->dapm); + + /* Go into headphone detect left mode */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA, 0); + snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_1, + WM8996_JD_MODE_MASK, 1); + + /* Trigger a measurement */ + snd_soc_update_bits(codec, WM8996_HEADPHONE_DETECT_1, + WM8996_HP_POLL, WM8996_HP_POLL); +} + static void wm8996_micd(struct snd_soc_codec *codec) { struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); @@ -2376,28 +2458,36 @@ static void wm8996_micd(struct snd_soc_codec *codec) wm8996->jack_mic = false; wm8996->detecting = true; snd_soc_jack_report(wm8996->jack, 0, - SND_JACK_HEADSET | SND_JACK_BTN_0); + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0); + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_RATE_MASK, WM8996_MICD_RATE_MASK); return; } - /* If the measurement is very high we've got a microphone but - * do a little debounce to account for mechanical issues. + /* If the measurement is very high we've got a microphone, + * either we just detected one or if we already reported then + * we've got a button release event. */ if (val & 0x400) { - dev_dbg(codec->dev, "Microphone detected\n"); - snd_soc_jack_report(wm8996->jack, SND_JACK_HEADSET, - SND_JACK_HEADSET | SND_JACK_BTN_0); - wm8996->jack_mic = true; - wm8996->detecting = false; - - /* Increase poll rate to give better responsiveness - * for buttons */ - snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - 5 << WM8996_MICD_RATE_SHIFT); + if (wm8996->detecting) { + dev_dbg(codec->dev, "Microphone detected\n"); + wm8996->jack_mic = true; + wm8996_hpdet_start(codec); + + /* Increase poll rate to give better responsiveness + * for buttons */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, + WM8996_MICD_RATE_MASK, + 5 << WM8996_MICD_RATE_SHIFT); + } else { + dev_dbg(codec->dev, "Mic button up\n"); + snd_soc_jack_report(wm8996->jack, 0, SND_JACK_BTN_0); + } + + return; } /* If we detected a lower impedence during initial startup @@ -2429,15 +2519,11 @@ static void wm8996_micd(struct snd_soc_codec *codec) if (val & 0x3fc) { if (wm8996->jack_mic) { dev_dbg(codec->dev, "Mic button detected\n"); - snd_soc_jack_report(wm8996->jack, - SND_JACK_HEADSET | SND_JACK_BTN_0, - SND_JACK_HEADSET | SND_JACK_BTN_0); - } else { - dev_dbg(codec->dev, "Headphone detected\n"); - snd_soc_jack_report(wm8996->jack, - SND_JACK_HEADPHONE, - SND_JACK_HEADSET | + snd_soc_jack_report(wm8996->jack, SND_JACK_BTN_0, SND_JACK_BTN_0); + } else if (wm8996->detecting) { + dev_dbg(codec->dev, "Headphone detected\n"); + wm8996_hpdet_start(codec); /* Increase the detection rate a bit for * responsiveness. @@ -2445,8 +2531,6 @@ static void wm8996_micd(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_RATE_MASK, 7 << WM8996_MICD_RATE_SHIFT); - - wm8996->detecting = false; } } } @@ -2486,6 +2570,9 @@ static irqreturn_t wm8996_irq(int irq, void *data) if (irq_val & WM8996_MICD_EINT) wm8996_micd(codec); + if (irq_val & WM8996_HP_DONE_EINT) + wm8996_hpdet_irq(codec); + return IRQ_HANDLED; } -- cgit v1.2.3 From 45cf367e8058ff3f1f6f9c3e9f617bfd7cff65b1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 4 Sep 2011 07:54:55 -0700 Subject: ASoC: Add line loads to the list of supported detections for Speyside Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/samsung/speyside.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 09df8afbb447..b9e213f6cc06 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -167,7 +167,8 @@ static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd) gpio_direction_output(WM8996_HPSEL_GPIO, speyside_jack_polarity); ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0, &speyside_headset); if (ret) return ret; -- cgit v1.2.3 From 5d42940c25ac69c4f5240392cf5e26bf08029e7a Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 19 Sep 2011 16:34:28 +0800 Subject: ASoC: sn95031: Staticize sn95031_pcm_hw_params Signed-off-by: Axel Lin Acked-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index b4f1cb494ffc..29945b004135 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -658,7 +658,7 @@ static int sn95031_pcm_spkr_mute(struct snd_soc_dai *dai, int mute) return 0; } -int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, +static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { unsigned int format, rate; -- cgit v1.2.3 From 76067540c642b1a14679ab74bd027a074c23e63b Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Wed, 7 Sep 2011 20:51:50 +0800 Subject: ASoC: mxs-saif: add record function 1. add different clkmux mode handling SAIF can use two instances to implement full duplex (playback & recording) and record saif may work on EXTMASTER mode which is using other saif's BITCLK&LRCLK. The clkmux mode could be set in pdata->init() in mach-specific code. For generic saif driver, it only needs to know who is his master and the master id is also provided in mach-specific code. 2. support playback and capture simutaneously however the sample rates can not be different due to hw limitation. Signed-off-by: Dong Aisheng Acked-by: Wolfram Sang Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 145 ++++++++++++++++++++++++++++++++++++++++++----- sound/soc/mxs/mxs-saif.h | 4 ++ 2 files changed, 135 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index af5734f6dab7..401944cf4560 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -23,10 +23,12 @@ #include #include #include +#include #include #include #include #include +#include #include #include #include @@ -36,6 +38,24 @@ static struct mxs_saif *mxs_saif[2]; +/* + * SAIF is a little different with other normal SOC DAIs on clock using. + * + * For MXS, two SAIF modules are instantiated on-chip. + * Each SAIF has a set of clock pins and can be operating in master + * mode simultaneously if they are connected to different off-chip codecs. + * Also, one of the two SAIFs can master or drive the clock pins while the + * other SAIF, in slave mode, receives clocking from the master SAIF. + * This also means that both SAIFs must operate at the same sample rate. + * + * We abstract this as each saif has a master, the master could be + * himself or other saifs. In the generic saif driver, saif does not need + * to know the different clkmux. Saif only needs to know who is his master + * and operating his master to generate the proper clock rate for him. + * The master id is provided in mach-specific layer according to different + * clkmux setting. + */ + static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -51,6 +71,17 @@ static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return 0; } +/* + * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK + * is provided by other SAIF, we provide a interface here to get its master + * from its master_id. + * Note that the master could be himself. + */ +static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif) +{ + return mxs_saif[saif->master_id]; +} + /* * Set SAIF clock and MCLK */ @@ -60,8 +91,26 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, { u32 scr; int ret; + struct mxs_saif *master_saif; - scr = __raw_readl(saif->base + SAIF_CTRL); + dev_dbg(saif->dev, "mclk %d rate %d\n", mclk, rate); + + /* Set master saif to generate proper clock */ + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + + dev_dbg(saif->dev, "master saif%d\n", master_saif->id); + + /* Checking if can playback and capture simutaneously */ + if (master_saif->ongoing && rate != master_saif->cur_rate) { + dev_err(saif->dev, + "can not change clock, master saif%d(rate %d) is ongoing\n", + master_saif->id, master_saif->cur_rate); + return -EINVAL; + } + + scr = __raw_readl(master_saif->base + SAIF_CTRL); scr &= ~BM_SAIF_CTRL_BITCLK_MULT_RATE; scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; @@ -75,27 +124,29 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, * * If MCLK is not used, we just set saif clk to 512*fs. */ - if (saif->mclk_in_use) { + if (master_saif->mclk_in_use) { if (mclk % 32 == 0) { scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; - ret = clk_set_rate(saif->clk, 512 * rate); + ret = clk_set_rate(master_saif->clk, 512 * rate); } else if (mclk % 48 == 0) { scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE; - ret = clk_set_rate(saif->clk, 384 * rate); + ret = clk_set_rate(master_saif->clk, 384 * rate); } else { /* SAIF MCLK should be either 32x or 48x */ return -EINVAL; } } else { - ret = clk_set_rate(saif->clk, 512 * rate); + ret = clk_set_rate(master_saif->clk, 512 * rate); scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; } if (ret) return ret; - if (!saif->mclk_in_use) { - __raw_writel(scr, saif->base + SAIF_CTRL); + master_saif->cur_rate = rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(scr, master_saif->base + SAIF_CTRL); return 0; } @@ -137,7 +188,7 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, return -EINVAL; } - __raw_writel(scr, saif->base + SAIF_CTRL); + __raw_writel(scr, master_saif->base + SAIF_CTRL); return 0; } @@ -183,6 +234,7 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, struct mxs_saif *saif = mxs_saif[saif_id]; u32 stat; int ret; + struct mxs_saif *master_saif; if (!saif) return -EINVAL; @@ -195,6 +247,12 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, __raw_writel(BM_SAIF_CTRL_CLKGATE, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + master_saif = mxs_saif_get_master(saif); + if (saif != master_saif) { + dev_err(saif->dev, "can not get mclk from a non-master saif\n"); + return -EINVAL; + } + stat = __raw_readl(saif->base + SAIF_STAT); if (stat & BM_SAIF_STAT_BUSY) { dev_err(saif->dev, "error: busy\n"); @@ -278,10 +336,17 @@ static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) /* * Note: We simply just support master mode since SAIF TX can only * work as master. + * Here the master is relative to codec side. + * Saif internally could be slave when working on EXTMASTER mode. + * We just hide this to machine driver. */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + if (saif->id == saif->master_id) + scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + else + scr |= BM_SAIF_CTRL_SLAVE_MODE; + __raw_writel(scr | scr0, saif->base + SAIF_CTRL); break; default: @@ -396,6 +461,12 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + struct mxs_saif *master_saif; + u32 delay; + + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -403,10 +474,20 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: dev_dbg(cpu_dai->dev, "start\n"); - clk_enable(saif->clk); - if (!saif->mclk_in_use) + clk_enable(master_saif->clk); + if (!master_saif->mclk_in_use) + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_SET_ADDR); + + /* + * If the saif's master is not himself, we also need to enable + * itself clk for its internal basic logic to work. + */ + if (saif != master_saif) { + clk_enable(saif->clk); __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_SET_ADDR); + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* @@ -422,20 +503,39 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, __raw_readl(saif->base + SAIF_DATA); } - dev_dbg(cpu_dai->dev, "CTRL 0x%x STAT 0x%x\n", + master_saif->ongoing = 1; + + dev_dbg(saif->dev, "CTRL 0x%x STAT 0x%x\n", __raw_readl(saif->base + SAIF_CTRL), __raw_readl(saif->base + SAIF_STAT)); + dev_dbg(master_saif->dev, "CTRL 0x%x STAT 0x%x\n", + __raw_readl(master_saif->base + SAIF_CTRL), + __raw_readl(master_saif->base + SAIF_STAT)); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: dev_dbg(cpu_dai->dev, "stop\n"); - clk_disable(saif->clk); - if (!saif->mclk_in_use) + /* wait a while for the current sample to complete */ + delay = USEC_PER_SEC / master_saif->cur_rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(delay); + } + clk_disable(master_saif->clk); + + if (saif != master_saif) { __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(delay); + clk_disable(saif->clk); + } + + master_saif->ongoing = 0; break; default: @@ -519,16 +619,33 @@ static int mxs_saif_probe(struct platform_device *pdev) { struct resource *res; struct mxs_saif *saif; + struct mxs_saif_platform_data *pdata; int ret = 0; if (pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; + pdata = pdev->dev.platform_data; + if (pdata && pdata->init) { + ret = pdata->init(); + if (ret) + return ret; + } + saif = kzalloc(sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; mxs_saif[pdev->id] = saif; + saif->id = pdev->id; + + saif->master_id = saif->id; + if (pdata && pdata->get_master_id) { + saif->master_id = pdata->get_master_id(saif->id); + if (saif->master_id < 0 || + saif->master_id >= ARRAY_SIZE(mxs_saif)) + return -EINVAL; + } saif->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 0e2ff8cdbfee..12c91e4eb941 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -118,6 +118,10 @@ struct mxs_saif { void __iomem *base; int irq; struct mxs_pcm_dma_params dma_param; + unsigned int id; + unsigned int master_id; + unsigned int cur_rate; + unsigned int ongoing; struct platform_device *soc_platform_pdev; u32 fifo_underrun; -- cgit v1.2.3 From a52762eee97d42344691c190cf8786dd9edde4d7 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 15 Sep 2011 15:39:27 +0300 Subject: ASoC: twl6040: Chip initialization cleanup There is no need to write to the vio registers at probe time, since most them either read only, or shared with MFD or not used. On the other hand it is a good idea to updated the ASICREV register in the cache at this time. After power up we need to restore some registers. Clean up the list to contain only the registers we are going to restore. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 100 ++++++--------------------------------------- 1 file changed, 13 insertions(+), 87 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 443032b3b329..8bbd46a9bfd5 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -155,41 +155,8 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { 0x00, /* TWL6040_STATUS (ro) 0x2E */ }; -/* - * twl6040 vio/gnd registers: - * registers under vio/gnd supply can be accessed - * before the power-up sequence, after NRESPWRON goes high - */ -static const int twl6040_vio_reg[TWL6040_VIOREGNUM] = { - TWL6040_REG_ASICID, - TWL6040_REG_ASICREV, - TWL6040_REG_INTID, - TWL6040_REG_INTMR, - TWL6040_REG_NCPCTL, - TWL6040_REG_LDOCTL, - TWL6040_REG_AMICBCTL, - TWL6040_REG_DMICBCTL, - TWL6040_REG_HKCTL1, - TWL6040_REG_HKCTL2, - TWL6040_REG_GPOCTL, - TWL6040_REG_TRIM1, - TWL6040_REG_TRIM2, - TWL6040_REG_TRIM3, - TWL6040_REG_HSOTRIM, - TWL6040_REG_HFOTRIM, - TWL6040_REG_ACCCTL, - TWL6040_REG_STATUS, -}; - -/* - * twl6040 vdd/vss registers: - * registers under vdd/vss supplies can only be accessed - * after the power-up sequence - */ -static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = { - TWL6040_REG_HPPLLCTL, - TWL6040_REG_LPPLLCTL, - TWL6040_REG_LPPLLDIV, +/* List of registers to be restored after power up */ +static const int twl6040_restore_list[] = { TWL6040_REG_MICLCTL, TWL6040_REG_MICRCTL, TWL6040_REG_MICGAIN, @@ -202,12 +169,6 @@ static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = { TWL6040_REG_HFLGAIN, TWL6040_REG_HFRCTL, TWL6040_REG_HFRGAIN, - TWL6040_REG_VIBCTLL, - TWL6040_REG_VIBDATL, - TWL6040_REG_VIBCTLR, - TWL6040_REG_VIBDATR, - TWL6040_REG_ALB, - TWL6040_REG_DLB, }; /* set of rates for each pll: low-power and high-performance */ @@ -296,56 +257,23 @@ static int twl6040_write(struct snd_soc_codec *codec, return twl6040_reg_write(twl6040, reg, value); } -static void twl6040_init_vio_regs(struct snd_soc_codec *codec) +static void twl6040_init_chip(struct snd_soc_codec *codec) { - u8 *cache = codec->reg_cache; - int reg, i; + struct twl6040 *twl6040 = codec->control_data; + u8 val; + + val = twl6040_get_revid(twl6040); + twl6040_write_reg_cache(codec, TWL6040_REG_ASICREV, val); - for (i = 0; i < TWL6040_VIOREGNUM; i++) { - reg = twl6040_vio_reg[i]; - /* - * skip read-only registers (ASICID, ASICREV, STATUS) - * and registers shared among MFD children - */ - switch (reg) { - case TWL6040_REG_ASICID: - case TWL6040_REG_ASICREV: - case TWL6040_REG_INTID: - case TWL6040_REG_INTMR: - case TWL6040_REG_NCPCTL: - case TWL6040_REG_LDOCTL: - case TWL6040_REG_GPOCTL: - case TWL6040_REG_ACCCTL: - case TWL6040_REG_STATUS: - continue; - default: - break; - } - twl6040_write(codec, reg, cache[reg]); - } } -static void twl6040_init_vdd_regs(struct snd_soc_codec *codec) +static void twl6040_restore_regs(struct snd_soc_codec *codec) { u8 *cache = codec->reg_cache; int reg, i; - for (i = 0; i < TWL6040_VDDREGNUM; i++) { - reg = twl6040_vdd_reg[i]; - /* skip vibra and PLL registers */ - switch (reg) { - case TWL6040_REG_VIBCTLL: - case TWL6040_REG_VIBDATL: - case TWL6040_REG_VIBCTLR: - case TWL6040_REG_VIBDATR: - case TWL6040_REG_HPPLLCTL: - case TWL6040_REG_LPPLLCTL: - case TWL6040_REG_LPPLLDIV: - continue; - default: - break; - } - + for (i = 0; i < ARRAY_SIZE(twl6040_restore_list); i++) { + reg = twl6040_restore_list[i]; twl6040_write(codec, reg, cache[reg]); } } @@ -1325,8 +1253,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, priv->codec_powered = 1; - /* initialize vdd/vss registers with reg_cache */ - twl6040_init_vdd_regs(codec); + twl6040_restore_regs(codec); /* Set external boost GPO */ twl6040_write(codec, TWL6040_REG_GPOCTL, 0x02); @@ -1620,8 +1547,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) goto plugirq_err; } - /* init vio registers */ - twl6040_init_vio_regs(codec); + twl6040_init_chip(codec); /* power on device */ ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- cgit v1.2.3 From 2c27ff41d8f81fa4967936151ece9fc16db96dce Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 15 Sep 2011 15:39:28 +0300 Subject: ASoC: twl6040: Use chip defaults in the initial reg_cache Reset the twl6040_reg array to hold the chip default values. The only changed values were for the microphone input selection. Select no input for the microphones in the twl6040_init_chip function. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 8bbd46a9bfd5..987d9c9c9dfd 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -118,8 +118,8 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { 0x4A, /* TWL6040_LPPLLDIV 0x09 */ 0x00, /* TWL6040_AMICBCTL 0x0A */ 0x00, /* TWL6040_DMICBCTL 0x0B */ - 0x18, /* TWL6040_MICLCTL 0x0C - No input selected on Left Mic */ - 0x18, /* TWL6040_MICRCTL 0x0D - No input selected on Right Mic */ + 0x00, /* TWL6040_MICLCTL 0x0C */ + 0x00, /* TWL6040_MICRCTL 0x0D */ 0x00, /* TWL6040_MICGAIN 0x0E */ 0x1B, /* TWL6040_LINEGAIN 0x0F */ 0x00, /* TWL6040_HSLCTL 0x10 */ @@ -265,6 +265,10 @@ static void twl6040_init_chip(struct snd_soc_codec *codec) val = twl6040_get_revid(twl6040); twl6040_write_reg_cache(codec, TWL6040_REG_ASICREV, val); + /* Change chip defaults */ + /* No imput selected for microphone amplifiers */ + twl6040_write_reg_cache(codec, TWL6040_REG_MICLCTL, 0x18); + twl6040_write_reg_cache(codec, TWL6040_REG_MICRCTL, 0x18); } static void twl6040_restore_regs(struct snd_soc_codec *codec) -- cgit v1.2.3 From d8dd032d533719cbaae2de6ca6dcc5553af3034e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 15 Sep 2011 15:59:18 +0300 Subject: ASoC: twl6040: Fix the number of channels for vibra Only mono audio can be used for vibra (DL4 channel). Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 987d9c9c9dfd..97f3e374fc67 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1449,8 +1449,8 @@ static struct snd_soc_dai_driver twl6040_dai[] = { .name = "twl6040-vib", .playback = { .stream_name = "Vibra Playback", - .channels_min = 2, - .channels_max = 2, + .channels_min = 1, + .channels_max = 1, .rates = SNDRV_PCM_RATE_CONTINUOUS, .formats = TWL6040_FORMATS, }, -- cgit v1.2.3 From cdd5054c3edcf556f67b629798d4dab64959c7cb Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 15 Sep 2011 15:59:19 +0300 Subject: ASoC: twl6040: Correct supported number of playback channels twl6040 supports 5 playback, and 2 capture channels Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 97f3e374fc67..81645c632447 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1399,7 +1399,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { .playback = { .stream_name = "Playback", .channels_min = 1, - .channels_max = 2, + .channels_max = 5, .rates = TWL6040_RATES, .formats = TWL6040_FORMATS, }, -- cgit v1.2.3 From c9d023adb62e17569f714669c93d3da07be49e3f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Sep 2011 16:16:08 +0100 Subject: ASoC: Fix unused variable warning in WM8996 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8996.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index cd1ba9637c01..1fe676c7aead 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -669,8 +669,6 @@ SOC_SINGLE_TLV("DSP2 EQ B5 Volume", WM8996_DSP2_RX_EQ_GAINS_2, 6, 31, 0, static int bg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; - struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); int ret = 0; switch (event) { -- cgit v1.2.3 From ded71dcb77ae0dee71fdb9c4e2d2b3dc3d1b7693 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Sep 2011 18:50:05 +0100 Subject: ASoC: Refcount WM8996 bandgap from FLL too For digital only paths we need to make sure the bandgap is enabled prior to starting the FLL which isn't tied into DAPM. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8996.c | 45 +++++++++++++++++++++++++++++++++++++++------ 1 file changed, 39 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1fe676c7aead..833df74c5584 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -71,6 +71,7 @@ struct wm8996_priv { struct regulator_bulk_data supplies[WM8996_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8996_NUM_SUPPLIES]; struct regulator *cpvdd; + int bg_ena; struct wm8996_pdata pdata; @@ -666,14 +667,40 @@ SOC_SINGLE_TLV("DSP2 EQ B5 Volume", WM8996_DSP2_RX_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static void wm8996_bg_enable(struct snd_soc_codec *codec) +{ + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + + wm8996->bg_ena++; + if (wm8996->bg_ena == 1) { + snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, + WM8996_BG_ENA, WM8996_BG_ENA); + msleep(2); + } +} + +static void wm8996_bg_disable(struct snd_soc_codec *codec) +{ + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + + wm8996->bg_ena--; + if (!wm8996->bg_ena) + snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, + WM8996_BG_ENA, 0); +} + static int bg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = w->codec; int ret = 0; switch (event) { - case SND_SOC_DAPM_POST_PMU: - msleep(2); + case SND_SOC_DAPM_PRE_PMU: + wm8996_bg_enable(codec); + break; + case SND_SOC_DAPM_POST_PMD: + wm8996_bg_disable(codec); break; default: BUG(); @@ -1014,10 +1041,9 @@ SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_SUPPLY("Bandgap", WM8996_POWER_MANAGEMENT_1, WM8996_BG_ENA_SHIFT, - 0, bg_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("Bandgap", SND_SOC_NOPM, 0, 0, bg_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICB1 Audio", WM8996_MICBIAS_1, 4, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("MICB2 Audio", WM8996_MICBIAS_2, 4, 1, NULL, 0), @@ -2096,6 +2122,8 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1, WM8996_FLL_ENA, 0); + wm8996_bg_disable(codec); + return 0; } @@ -2150,6 +2178,11 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda); + /* Enable the bandgap if it's not already enabled */ + ret = snd_soc_read(codec, WM8996_FLL_CONTROL_1); + if (!(ret & WM8996_FLL_ENA)) + wm8996_bg_enable(codec); + /* Clear any pending completions (eg, from failed startups) */ try_wait_for_completion(&wm8996->fll_lock); -- cgit v1.2.3 From d890a1a42dff2e6987f04f18fc9e467b10e99cc9 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 20 Sep 2011 15:09:00 +0800 Subject: ASoC: fsl: Fix error handling if platform_device_add fails Call platform_device_put() instead of platform_device_unregister() if platform_device_add() fails. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 358f0baaf71b..31af405bda84 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -505,7 +505,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) return 0; error_sound: - platform_device_unregister(sound_device); + platform_device_put(sound_device); error: kfree(machine_data); error_alloc: diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index e8849ed36cbd..2c064a9824ad 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -506,7 +506,7 @@ static int p1022_ds_probe(struct platform_device *pdev) error: if (sound_device) - platform_device_unregister(sound_device); + platform_device_put(sound_device); kfree(mdata); error_put: -- cgit v1.2.3 From 26806a4266c5d3301d3858317e67b1cca7ccfebb Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 20 Sep 2011 08:19:58 +0200 Subject: ASoC: ssm2602: Do not dereference codec->control_data The driver assumes that control_data points to the drivers i2c_client struct, but this is no longer the case since the ASoC core has switched to regmap. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 84f4ad568556..cceb0022f02c 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -294,7 +294,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c = codec->control_data; struct snd_pcm_runtime *master_runtime; /* The DAI has shared clocks so if we already have a playback or @@ -303,7 +302,7 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, */ if (ssm2602->master_substream) { master_runtime = ssm2602->master_substream->runtime; - dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", + dev_dbg(codec->dev, "Constraining to %d bits at %dHz\n", master_runtime->sample_bits, master_runtime->rate); -- cgit v1.2.3 From 88e24c3a4b30a6bd361f2b5ce602667a8161b2e8 Mon Sep 17 00:00:00 2001 From: Yong Zhang Date: Thu, 22 Sep 2011 16:59:20 +0800 Subject: sound: irq: Remove IRQF_DISABLED Since commit [e58aa3d2: genirq: Run irq handlers with interrupts disabled], We run all interrupt handlers with interrupts disabled and we even check and yell when an interrupt handler returns with interrupts enabled (see commit [b738a50a: genirq: Warn when handler enables interrupts]). So now this flag is a NOOP and can be removed. Signed-off-by: Yong Zhang Acked-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 2 +- sound/arm/pxa2xx-ac97-lib.c | 2 +- sound/drivers/ml403-ac97cr.c | 4 ++-- sound/drivers/mpu401/mpu401_uart.c | 2 +- sound/drivers/mtpav.c | 2 +- sound/drivers/serial-u16550.c | 2 +- sound/isa/ad1816a/ad1816a_lib.c | 2 +- sound/isa/es1688/es1688_lib.c | 2 +- sound/isa/es18xx.c | 2 +- sound/isa/gus/gus_main.c | 2 +- sound/isa/gus/gusmax.c | 2 +- sound/isa/gus/interwave.c | 2 +- sound/isa/opl3sa2.c | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 2 +- sound/isa/sb/sb_common.c | 2 +- sound/isa/wavefront/wavefront.c | 2 +- sound/isa/wss/wss_lib.c | 2 +- sound/mips/au1x00.c | 4 ++-- sound/pci/sis7019.c | 4 ++-- sound/ppc/snd_ps3.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/nuc900/nuc900-pcm.c | 2 +- sound/soc/samsung/ac97.c | 2 +- sound/soc/sh/fsi.c | 2 +- sound/soc/txx9/txx9aclc-ac97.c | 2 +- sound/sparc/amd7930.c | 2 +- 27 files changed, 30 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index d0cead38d5fb..e518d38b1c74 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -443,7 +443,7 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream) mutex_lock(&aaci->irq_lock); if (!aaci->users++) { ret = request_irq(aaci->dev->irq[0], aaci_irq, - IRQF_SHARED | IRQF_DISABLED, DRIVER_NAME, aaci); + IRQF_SHARED, DRIVER_NAME, aaci); if (ret != 0) aaci->users--; } diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 88eec3847df2..8ad65352bf91 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -359,7 +359,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (ret) goto err_clk2; - ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL); + ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, 0, "AC97", NULL); if (ret < 0) goto err_irq; diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 5cfcb908c430..2c7a7636f472 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1153,7 +1153,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, "0x%x done\n", (unsigned int)ml403_ac97cr->port); /* get irq */ irq = platform_get_irq(pfdev, 0); - if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, + if (request_irq(irq, snd_ml403_ac97cr_irq, 0, dev_name(&pfdev->dev), (void *)ml403_ac97cr)) { snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " "unable to grab IRQ %d\n", @@ -1166,7 +1166,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, "request (playback) irq %d done\n", ml403_ac97cr->irq); irq = platform_get_irq(pfdev, 1); - if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, + if (request_irq(irq, snd_ml403_ac97cr_irq, 0, dev_name(&pfdev->dev), (void *)ml403_ac97cr)) { snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " "unable to grab IRQ %d\n", diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 9d01c181feca..e91698a634b2 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -577,7 +577,7 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, else mpu->cport = port + 1; if (irq >= 0) { - if (request_irq(irq, snd_mpu401_uart_interrupt, IRQF_DISABLED, + if (request_irq(irq, snd_mpu401_uart_interrupt, 0, "MPU401 UART", (void *) mpu)) { snd_printk(KERN_ERR "mpu401_uart: " "unable to grab IRQ %d\n", irq); diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 5c426df87678..1eef4ccebe4b 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -589,7 +589,7 @@ static int __devinit snd_mtpav_get_ISA(struct mtpav * mcard) return -EBUSY; } mcard->port = port; - if (request_irq(irq, snd_mtpav_irqh, IRQF_DISABLED, "MOTU MTPAV", mcard)) { + if (request_irq(irq, snd_mtpav_irqh, 0, "MOTU MTPAV", mcard)) { snd_printk(KERN_ERR "MTVAP IRQ %d busy\n", irq); return -EBUSY; } diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index a25fb7b1f441..fc1d822802c3 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -816,7 +816,7 @@ static int __devinit snd_uart16550_create(struct snd_card *card, if (irq >= 0 && irq != SNDRV_AUTO_IRQ) { if (request_irq(irq, snd_uart16550_interrupt, - IRQF_DISABLED, "Serial MIDI", uart)) { + 0, "Serial MIDI", uart)) { snd_printk(KERN_WARNING "irq %d busy. Using Polling.\n", irq); } else { diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index 05aef8b97e96..177eed3271bc 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -595,7 +595,7 @@ int __devinit snd_ad1816a_create(struct snd_card *card, snd_ad1816a_free(chip); return -EBUSY; } - if (request_irq(irq, snd_ad1816a_interrupt, IRQF_DISABLED, "AD1816A", (void *) chip)) { + if (request_irq(irq, snd_ad1816a_interrupt, 0, "AD1816A", (void *) chip)) { snd_printk(KERN_ERR "ad1816a: can't grab IRQ %d\n", irq); snd_ad1816a_free(chip); return -EBUSY; diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index 07676200496a..d3eab6fb0866 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -661,7 +661,7 @@ int snd_es1688_create(struct snd_card *card, snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4); return -EBUSY; } - if (request_irq(irq, snd_es1688_interrupt, IRQF_DISABLED, "ES1688", (void *) chip)) { + if (request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip)) { snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq); return -EBUSY; } diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index aeee8f8bf5e9..bf6ad0bf51c6 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1805,7 +1805,7 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, return -EBUSY; } - if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx", + if (request_irq(irq, snd_es18xx_interrupt, 0, "ES18xx", (void *) card)) { snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap IRQ %d\n", irq); diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index 12eb98f2f931..3167e5ac3699 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -180,7 +180,7 @@ int snd_gus_create(struct snd_card *card, snd_gus_free(gus); return -EBUSY; } - if (irq >= 0 && request_irq(irq, snd_gus_interrupt, IRQF_DISABLED, "GUS GF1", (void *) gus)) { + if (irq >= 0 && request_irq(irq, snd_gus_interrupt, 0, "GUS GF1", (void *) gus)) { snd_printk(KERN_ERR "gus: can't grab irq %d\n", irq); snd_gus_free(gus); return -EBUSY; diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index 3e4a58b72913..c43faa057ff6 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -291,7 +291,7 @@ static int __devinit snd_gusmax_probe(struct device *pdev, unsigned int dev) goto _err; } - if (request_irq(xirq, snd_gusmax_interrupt, IRQF_DISABLED, "GUS MAX", (void *)maxcard)) { + if (request_irq(xirq, snd_gusmax_interrupt, 0, "GUS MAX", (void *)maxcard)) { snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq); err = -EBUSY; goto _err; diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index c7b80e4730fc..5f869a32b48c 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -684,7 +684,7 @@ static int __devinit snd_interwave_probe(struct snd_card *card, int dev) if ((err = snd_gus_initialize(gus)) < 0) return err; - if (request_irq(xirq, snd_interwave_interrupt, IRQF_DISABLED, + if (request_irq(xirq, snd_interwave_interrupt, 0, "InterWave", iwcard)) { snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq); return -EBUSY; diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index de99f47770bf..bbafb0b543ea 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -667,7 +667,7 @@ static int __devinit snd_opl3sa2_probe(struct snd_card *card, int dev) err = snd_opl3sa2_detect(card); if (err < 0) return err; - err = request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED, + err = request_irq(xirq, snd_opl3sa2_interrupt, 0, "OPL3-SA2", card); if (err) { snd_printk(KERN_ERR PFX "can't grab IRQ %d\n", xirq); diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 346e12baa98e..6dbbfa76b440 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -892,7 +892,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) #endif #ifdef OPTi93X error = request_irq(irq, snd_opti93x_interrupt, - IRQF_DISABLED, DEV_NAME" - WSS", chip); + 0, DEV_NAME" - WSS", chip); if (error < 0) { snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", irq); return error; diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index eae6c1c0eff9..d2e19215813e 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -240,7 +240,7 @@ int snd_sbdsp_create(struct snd_card *card, if (request_irq(irq, irq_handler, (hardware == SB_HW_ALS4000 || hardware == SB_HW_CS5530) ? - IRQF_SHARED : IRQF_DISABLED, + IRQF_SHARED : 0, "SoundBlaster", (void *) chip)) { snd_printk(KERN_ERR "sb: can't grab irq %d\n", irq); snd_sbdsp_free(chip); diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 83f291d89a95..87142977335a 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -418,7 +418,7 @@ snd_wavefront_probe (struct snd_card *card, int dev) return -EBUSY; } if (request_irq(ics2115_irq[dev], snd_wavefront_ics2115_interrupt, - IRQF_DISABLED, "ICS2115", acard)) { + 0, "ICS2115", acard)) { snd_printk(KERN_ERR "unable to use ICS2115 IRQ %d\n", ics2115_irq[dev]); return -EBUSY; } diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 2a42cc377957..7277c5b7df6c 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1833,7 +1833,7 @@ int snd_wss_create(struct snd_card *card, } chip->cport = cport; if (!(hwshare & WSS_HWSHARE_IRQ)) - if (request_irq(irq, snd_wss_interrupt, IRQF_DISABLED, + if (request_irq(irq, snd_wss_interrupt, 0, "WSS", (void *) chip)) { snd_printk(KERN_ERR "wss: can't grab IRQ %d\n", irq); snd_wss_free(chip); diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index 446cf9748664..7567ebd71913 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -465,13 +465,13 @@ snd_au1000_pcm_new(struct snd_au1000 *au1000) flags = claim_dma_lock(); if ((au1000->stream[PLAYBACK]->dma = request_au1000_dma(DMA_ID_AC97C_TX, - "AC97 TX", au1000_dma_interrupt, IRQF_DISABLED, + "AC97 TX", au1000_dma_interrupt, 0, au1000->stream[PLAYBACK])) < 0) { release_dma_lock(flags); return -EBUSY; } if ((au1000->stream[CAPTURE]->dma = request_au1000_dma(DMA_ID_AC97C_RX, - "AC97 RX", au1000_dma_interrupt, IRQF_DISABLED, + "AC97 RX", au1000_dma_interrupt, 0, au1000->stream[CAPTURE])) < 0){ release_dma_lock(flags); return -EBUSY; diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index bcf61524a13f..5ffb20b18786 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1234,7 +1234,7 @@ static int sis_resume(struct pci_dev *pci) goto error; } - if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, sis)) { printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq); goto error; @@ -1340,7 +1340,7 @@ static int __devinit sis_chip_create(struct snd_card *card, if (rc) goto error_out_cleanup; - if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, sis)) { printk(KERN_ERR "unable to allocate irq %d\n", sis->irq); goto error_out_cleanup; diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index bc823a547550..775bd95d4be6 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -845,7 +845,7 @@ static int __devinit snd_ps3_allocate_irq(void) return ret; } - ret = request_irq(the_card.irq_no, snd_ps3_interrupt, IRQF_DISABLED, + ret = request_irq(the_card.irq_no, snd_ps3_interrupt, 0, SND_PS3_DRIVER_NAME, &the_card); if (ret) { pr_info("%s: request_irq failed (%d)\n", __func__, ret); diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index 7aa5b7606777..177f7137a9c8 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -211,7 +211,7 @@ static int alchemy_pcm_open(struct snd_pcm_substream *substream) /* 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, + au1000_dma_interrupt, 0, &ctx->stream[s]); set_dma_mode(ctx->stream[s].dma, get_dma_mode(ctx->stream[s].dma) & ~DMA_NC); diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index faa5e9fb1471..243d17711211 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1431,7 +1431,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) /* Check if the IRQ number is valid and request it */ if (dac33->irq >= 0) { ret = request_irq(dac33->irq, dac33_interrupt_handler, - IRQF_TRIGGER_RISING | IRQF_DISABLED, + IRQF_TRIGGER_RISING, codec->name, codec); if (ret < 0) { dev_err(codec->dev, "Could not request IRQ%d (%d)\n", diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index e46d5516e000..865b288bd748 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -268,7 +268,7 @@ static int nuc900_dma_open(struct snd_pcm_substream *substream) nuc900_audio = nuc900_ac97_data; if (request_irq(nuc900_audio->irq_num, nuc900_dma_interrupt, - IRQF_DISABLED, "nuc900-dma", substream)) + 0, "nuc900-dma", substream)) return -EBUSY; runtime->private_data = nuc900_audio; diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index f97110e72e85..884c8a107bf9 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -444,7 +444,7 @@ static __devinit int s3c_ac97_probe(struct platform_device *pdev) } ret = request_irq(irq_res->start, s3c_ac97_irq, - IRQF_DISABLED, "AC97", NULL); + 0, "AC97", NULL); if (ret < 0) { dev_err(&pdev->dev, "ac97: interrupt request failed.\n"); goto err4; diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 8e112ccffb13..1493ebf4d943 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1285,7 +1285,7 @@ static int fsi_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); dev_set_drvdata(&pdev->dev, master); - ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, + ret = request_irq(irq, &fsi_interrupt, 0, id_entry->name, master); if (ret) { dev_err(&pdev->dev, "irq request err\n"); diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 743d07b82c06..a4e3f5501847 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -201,7 +201,7 @@ static int __devinit txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (!drvdata->base) return -EBUSY; err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq, - IRQF_DISABLED, dev_name(&pdev->dev), drvdata); + 0, dev_name(&pdev->dev), drvdata); if (err < 0) return err; diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index ad7d4d7d9237..f036776380b5 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -962,7 +962,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card, amd7930_idle(amd); if (request_irq(irq, snd_amd7930_interrupt, - IRQF_DISABLED | IRQF_SHARED, "amd7930", amd)) { + IRQF_SHARED, "amd7930", amd)) { snd_printk(KERN_ERR "amd7930-%d: Unable to grab IRQ %d\n", dev, irq); snd_amd7930_free(amd); -- cgit v1.2.3