diff options
39 files changed, 2947 insertions, 256 deletions
diff --git a/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt b/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt new file mode 100644 index 000000000000..2c3cd413f042 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt @@ -0,0 +1,11 @@ +* Freescale SGTL5000 Stereo Codec + +Required properties: +- compatible : "fsl,sgtl5000". + +Example: + +codec: sgtl5000@0a { + compatible = "fsl,sgtl5000"; + reg = <0x0a>; +}; diff --git a/Documentation/devicetree/bindings/sound/wm8731.txt b/Documentation/devicetree/bindings/sound/wm8731.txt new file mode 100644 index 000000000000..15f70048469b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8731.txt @@ -0,0 +1,18 @@ +WM8731 audio CODEC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8731" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Example: + +codec: wm8731@1a { + compatible = "wlf,wm8731"; + reg = <0x1a>; +}; diff --git a/arch/mips/alchemy/devboards/db1200/platform.c b/arch/mips/alchemy/devboards/db1200/platform.c index fbb55935b99e..dda090bf74e6 100644 --- a/arch/mips/alchemy/devboards/db1200/platform.c +++ b/arch/mips/alchemy/devboards/db1200/platform.c @@ -422,6 +422,7 @@ static struct resource au1200_psc1_res[] = { }, }; +/* AC97 or I2S device */ static struct platform_device db1200_audio_dev = { /* name assigned later based on switch setting */ .id = 1, /* PSC ID */ @@ -429,19 +430,32 @@ static struct platform_device db1200_audio_dev = { .resource = au1200_psc1_res, }; +/* DB1200 ASoC card device */ +static struct platform_device db1200_sound_dev = { + /* name assigned later based on switch setting */ + .id = 1, /* PSC ID */ +}; + static struct platform_device db1200_stac_dev = { .name = "ac97-codec", .id = 1, /* on PSC1 */ }; +static struct platform_device db1200_audiodma_dev = { + .name = "au1xpsc-pcm", + .id = 1, /* PSC ID */ +}; + static struct platform_device *db1200_devs[] __initdata = { NULL, /* PSC0, selected by S6.8 */ &db1200_ide_dev, &db1200_eth_dev, &db1200_rtc_dev, &db1200_nand_dev, + &db1200_audiodma_dev, &db1200_audio_dev, &db1200_stac_dev, + &db1200_sound_dev, }; static int __init db1200_dev_init(void) @@ -501,10 +515,12 @@ static int __init db1200_dev_init(void) if (sw == BCSR_SWITCHES_DIP_8) { bcsr_mod(BCSR_RESETS, 0, BCSR_RESETS_PSC1MUX); db1200_audio_dev.name = "au1xpsc_i2s"; + db1200_sound_dev.name = "db1200-i2s"; printk(KERN_INFO " S6.7 ON : PSC1 mode I2S\n"); } else { bcsr_mod(BCSR_RESETS, BCSR_RESETS_PSC1MUX, 0); db1200_audio_dev.name = "au1xpsc_ac97"; + db1200_sound_dev.name = "db1200-ac97"; printk(KERN_INFO " S6.7 OFF: PSC1 mode AC97\n"); } diff --git a/arch/mips/alchemy/devboards/db1x00/platform.c b/arch/mips/alchemy/devboards/db1x00/platform.c index 978d5ab3d678..7057d28f7301 100644 --- a/arch/mips/alchemy/devboards/db1x00/platform.c +++ b/arch/mips/alchemy/devboards/db1x00/platform.c @@ -19,8 +19,11 @@ */ #include <linux/init.h> +#include <linux/interrupt.h> #include <linux/platform_device.h> +#include <asm/mach-au1x00/au1000.h> +#include <asm/mach-au1x00/au1000_dma.h> #include <asm/mach-au1x00/au1xxx.h> #include <asm/mach-db1x00/bcsr.h> #include "../platform.h" @@ -85,6 +88,45 @@ #endif #endif +static struct resource alchemy_ac97c_res[] = { + [0] = { + .start = AU1000_AC97_PHYS_ADDR, + .end = AU1000_AC97_PHYS_ADDR + 0xfff, + .flags = IORESOURCE_MEM, + }, + [1] = { + .start = DMA_ID_AC97C_TX, + .end = DMA_ID_AC97C_TX, + .flags = IORESOURCE_DMA, + }, + [2] = { + .start = DMA_ID_AC97C_RX, + .end = DMA_ID_AC97C_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device alchemy_ac97c_dev = { + .name = "alchemy-ac97c", + .id = -1, + .resource = alchemy_ac97c_res, + .num_resources = ARRAY_SIZE(alchemy_ac97c_res), +}; + +static struct platform_device alchemy_ac97c_dma_dev = { + .name = "alchemy-pcm-dma", + .id = 0, +}; + +static struct platform_device db1x00_codec_dev = { + .name = "ac97-codec", + .id = -1, +}; + +static struct platform_device db1x00_audio_dev = { + .name = "db1000-audio", +}; + static int __init db1xxx_dev_init(void) { #ifdef DB1XXX_HAS_PCMCIA @@ -113,6 +155,12 @@ static int __init db1xxx_dev_init(void) 1); #endif db1x_register_norflash(BOARD_FLASH_SIZE, BOARD_FLASH_WIDTH, F_SWAPPED); + + platform_device_register(&db1x00_codec_dev); + platform_device_register(&alchemy_ac97c_dma_dev); + platform_device_register(&alchemy_ac97c_dev); + platform_device_register(&db1x00_audio_dev); + return 0; } device_initcall(db1xxx_dev_init); diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index f3ee84284670..61529143db57 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -72,6 +72,7 @@ #define WM8994_DC_SERVO_2 0x55 #define WM8994_DC_SERVO_4 0x57 #define WM8994_DC_SERVO_READBACK 0x58 +#define WM8994_DC_SERVO_4E 0x59 #define WM8994_ANALOGUE_HP_1 0x60 #define WM8958_MIC_DETECT_1 0xD0 #define WM8958_MIC_DETECT_2 0xD1 diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e0583b7769cb..350b1b395cac 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -524,6 +524,8 @@ struct snd_soc_dapm_context { enum snd_soc_bias_level target_bias_level; struct list_head list; + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index d02269437de3..3fe658eea28b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -637,6 +637,9 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); + /* codec stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; @@ -672,6 +675,9 @@ struct snd_soc_platform_driver { /* platform stream ops */ struct snd_pcm_ops *ops; + /* platform stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; 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 diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index f9054f7c1d52..1381db853ef0 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -53,6 +53,7 @@ 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/au1x/Kconfig b/sound/soc/au1x/Kconfig index 4b67140fdec3..6d592546e8fc 100644 --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -18,10 +18,38 @@ 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 ## +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 16873076e8c4..920710514ea0 100644 --- a/sound/soc/au1x/Makefile +++ b/sound/soc/au1x/Makefile @@ -3,11 +3,21 @@ 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-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/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 <manuel.lauss@googlemail.com> + * + * based on the old ALSA driver originally written by + * Charles Eidsness <charles@cooper-street.com> + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/mutex.h> +#include <linux/platform_device.h> +#include <linux/suspend.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/mach-au1x00/au1000.h> + +#include "psc.h" + +/* register offsets and bits */ +#define AC97_CONFIG 0x00 +#define AC97_STATUS 0x04 +#define AC97_DATA 0x08 +#define AC97_CMDRESP 0x0c +#define AC97_ENABLE 0x10 + +#define CFG_RC(x) (((x) & 0x3ff) << 13) /* valid rx slots mask */ +#define CFG_XS(x) (((x) & 0x3ff) << 3) /* valid tx slots mask */ +#define CFG_SG (1 << 2) /* sync gate */ +#define CFG_SN (1 << 1) /* sync control */ +#define CFG_RS (1 << 0) /* acrst# control */ +#define STAT_XU (1 << 11) /* tx underflow */ +#define STAT_XO (1 << 10) /* tx overflow */ +#define STAT_RU (1 << 9) /* rx underflow */ +#define STAT_RO (1 << 8) /* rx overflow */ +#define STAT_RD (1 << 7) /* codec ready */ +#define STAT_CP (1 << 6) /* command pending */ +#define STAT_TE (1 << 4) /* tx fifo empty */ +#define STAT_TF (1 << 3) /* tx fifo full */ +#define STAT_RE (1 << 1) /* rx fifo empty */ +#define STAT_RF (1 << 0) /* rx fifo full */ +#define CMD_SET_DATA(x) (((x) & 0xffff) << 16) +#define CMD_GET_DATA(x) ((x) & 0xffff) +#define CMD_READ (1 << 7) +#define CMD_WRITE (0 << 7) +#define CMD_IDX(x) ((x) & 0x7f) +#define EN_D (1 << 1) /* DISable bit */ +#define EN_CE (1 << 0) /* clock enable bit */ + +/* how often to retry failed codec register reads/writes */ +#define AC97_RW_RETRIES 5 + +#define AC97_RATES \ + SNDRV_PCM_RATE_CONTINUOUS + +#define AC97_FMTS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE) + +/* instance data. There can be only one, MacLeod!!!!, fortunately there IS only + * once AC97C on early Alchemy chips. The newer ones aren't so lucky. + */ +static struct au1xpsc_audio_data *ac97c_workdata; +#define ac97_to_ctx(x) ac97c_workdata + +static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg) +{ + return __raw_readl(ctx->mmio + reg); +} + +static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v) +{ + __raw_writel(v, ctx->mmio + reg); + wmb(); +} + +static unsigned short au1xac97c_ac97_read(struct snd_ac97 *ac97, + unsigned short r) +{ + struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97); + unsigned int tmo, retry; + unsigned long data; + + data = ~0; + retry = AC97_RW_RETRIES; + do { + mutex_lock(&ctx->lock); + + tmo = 5; + while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--) + udelay(21); /* wait an ac97 frame time */ + if (!tmo) { + pr_debug("ac97rd timeout #1\n"); + goto next; + } + + WR(ctx, AC97_CMDRESP, CMD_IDX(r) | CMD_READ); + + /* stupid errata: data is only valid for 21us, so + * poll, Forrest, poll... + */ + tmo = 0x10000; + while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--) + asm volatile ("nop"); + data = RD(ctx, AC97_CMDRESP); + + if (!tmo) + pr_debug("ac97rd timeout #2\n"); + +next: + mutex_unlock(&ctx->lock); + } while (--retry && !tmo); + + pr_debug("AC97RD %04x %04lx %d\n", r, data, retry); + + return retry ? data & 0xffff : 0xffff; +} + +static void au1xac97c_ac97_write(struct snd_ac97 *ac97, unsigned short r, + unsigned short v) +{ + struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97); + unsigned int tmo, retry; + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&ctx->lock); + + for (tmo = 5; (RD(ctx, AC97_STATUS) & STAT_CP) && tmo; tmo--) + udelay(21); + if (!tmo) { + pr_debug("ac97wr timeout #1\n"); + goto next; + } + + WR(ctx, AC97_CMDRESP, CMD_WRITE | CMD_IDX(r) | CMD_SET_DATA(v)); + + for (tmo = 10; (RD(ctx, AC97_STATUS) & STAT_CP) && tmo; tmo--) + udelay(21); + if (!tmo) + pr_debug("ac97wr timeout #2\n"); +next: + mutex_unlock(&ctx->lock); + } while (--retry && !tmo); + + pr_debug("AC97WR %04x %04x %d\n", r, v, retry); +} + +static void au1xac97c_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97); + + WR(ctx, AC97_CONFIG, ctx->cfg | CFG_SG | CFG_SN); + msleep(20); + WR(ctx, AC97_CONFIG, ctx->cfg | CFG_SG); + WR(ctx, AC97_CONFIG, ctx->cfg); +} + +static void au1xac97c_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97); + int i; + + WR(ctx, AC97_CONFIG, ctx->cfg | CFG_RS); + msleep(500); + WR(ctx, AC97_CONFIG, ctx->cfg); + + /* wait for codec ready */ + i = 50; + while (((RD(ctx, AC97_STATUS) & STAT_RD) == 0) && --i) + msleep(20); + if (!i) + printk(KERN_ERR "ac97c: codec not ready after cold reset\n"); +} + +/* AC97 controller operations */ +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = au1xac97c_ac97_read, + .write = au1xac97c_ac97_write, + .reset = au1xac97c_ac97_cold_reset, + .warm_reset = au1xac97c_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); /* globals be gone! */ + +static int alchemy_ac97c_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]); + return 0; +} + +static struct snd_soc_dai_ops alchemy_ac97c_ops = { + .startup = alchemy_ac97c_startup, +}; + +static int au1xac97c_dai_probe(struct snd_soc_dai *dai) +{ + return ac97c_workdata ? 0 : -ENODEV; +} + +static struct snd_soc_dai_driver au1xac97c_dai_driver = { + .name = "alchemy-ac97c", + .ac97_control = 1, + .probe = au1xac97c_dai_probe, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &alchemy_ac97c_ops, +}; + +static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) +{ + int ret; + struct resource *r; + struct au1xpsc_audio_data *ctx; + + ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + mutex_init(&ctx->lock); + + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r) { + ret = -ENODEV; + goto out0; + } + + ret = -EBUSY; + if (!request_mem_region(r->start, resource_size(r), pdev->name)) + goto out0; + + ctx->mmio = ioremap_nocache(r->start, resource_size(r)); + if (!ctx->mmio) + goto out1; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r) + goto out1; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) + goto out1; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + + /* switch it on */ + WR(ctx, AC97_ENABLE, EN_D | EN_CE); + WR(ctx, AC97_ENABLE, EN_CE); + + ctx->cfg = CFG_RC(3) | CFG_XS(3); + WR(ctx, AC97_CONFIG, ctx->cfg); + + platform_set_drvdata(pdev, ctx); + + ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver); + if (ret) + goto out1; + + ac97c_workdata = ctx; + return 0; + + + snd_soc_unregister_dai(&pdev->dev); +out1: + release_mem_region(r->start, resource_size(r)); +out0: + kfree(ctx); + return ret; +} + +static int __devexit au1xac97c_drvremove(struct platform_device *pdev) +{ + struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev); + struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + snd_soc_unregister_dai(&pdev->dev); + + WR(ctx, AC97_ENABLE, EN_D); /* clock off, disable */ + + iounmap(ctx->mmio); + release_mem_region(r->start, resource_size(r)); + kfree(ctx); + + ac97c_workdata = NULL; /* MDEV */ + + return 0; +} + +#ifdef CONFIG_PM +static int au1xac97c_drvsuspend(struct device *dev) +{ + struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev); + + WR(ctx, AC97_ENABLE, EN_D); /* clock off, disable */ + + return 0; +} + +static int au1xac97c_drvresume(struct device *dev) +{ + struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev); + + WR(ctx, AC97_ENABLE, EN_D | EN_CE); + WR(ctx, AC97_ENABLE, EN_CE); + WR(ctx, AC97_CONFIG, ctx->cfg); + + return 0; +} + +static const struct dev_pm_ops au1xpscac97_pmops = { + .suspend = au1xac97c_drvsuspend, + .resume = au1xac97c_drvresume, +}; + +#define AU1XPSCAC97_PMOPS (&au1xpscac97_pmops) + +#else + +#define AU1XPSCAC97_PMOPS NULL + +#endif + +static struct platform_driver au1xac97c_driver = { + .driver = { + .name = "alchemy-ac97c", + .owner = THIS_MODULE, + .pm = AU1XPSCAC97_PMOPS, + }, + .probe = au1xac97c_drvprobe, + .remove = __devexit_p(au1xac97c_drvremove), +}; + +static int __init au1xac97c_load(void) +{ + ac97c_workdata = NULL; + return platform_driver_register(&au1xac97c_driver); +} + +static void __exit au1xac97c_unload(void) +{ + platform_driver_unregister(&au1xac97c_driver); +} + +module_init(au1xac97c_load); +module_exit(au1xac97c_unload); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/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 <manuel.lauss@googlemail.com> + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <asm/mach-au1x00/au1000.h> +#include <asm/mach-db1x00/bcsr.h> + +#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"); 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 <manuel.lauss@gmail.com> + * (c) 2008-2011 Manuel Lauss <manuel.lauss@googlemail.com> * */ @@ -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); diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 20bb53a837b1..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 { @@ -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->stream, *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/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 <manuel.lauss@googlemail.com> + * + * copied almost verbatim from the old ALSA driver, written by + * Charles Eidsness <charles@cooper-street.com> + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <asm/mach-au1x00/au1000.h> +#include <asm/mach-au1x00/au1000_dma.h> + +#include "psc.h" + +#define ALCHEMY_PCM_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | \ + SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE | \ + 0) + +struct pcm_period { + u32 start; + u32 relative_end; /* relative to start of buffer */ + struct pcm_period *next; +}; + +struct audio_stream { + struct snd_pcm_substream *substream; + int dma; + struct pcm_period *buffer; + unsigned int period_size; + unsigned int periods; +}; + +struct alchemy_pcm_ctx { + struct audio_stream stream[2]; /* playback & capture */ +}; + +static void au1000_release_dma_link(struct audio_stream *stream) +{ + struct pcm_period *pointer; + struct pcm_period *pointer_next; + + stream->period_size = 0; + stream->periods = 0; + pointer = stream->buffer; + if (!pointer) + return; + do { + pointer_next = pointer->next; + kfree(pointer); + pointer = pointer_next; + } while (pointer != stream->buffer); + stream->buffer = NULL; +} + +static int au1000_setup_dma_link(struct audio_stream *stream, + unsigned int period_bytes, + unsigned int periods) +{ + struct snd_pcm_substream *substream = stream->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct pcm_period *pointer; + unsigned long dma_start; + int i; + + dma_start = virt_to_phys(runtime->dma_area); + + if (stream->period_size == period_bytes && + stream->periods == periods) + return 0; /* not changed */ + + au1000_release_dma_link(stream); + + stream->period_size = period_bytes; + stream->periods = periods; + + stream->buffer = kmalloc(sizeof(struct pcm_period), GFP_KERNEL); + if (!stream->buffer) + return -ENOMEM; + pointer = stream->buffer; + for (i = 0; i < periods; i++) { + pointer->start = (u32)(dma_start + (i * period_bytes)); + pointer->relative_end = (u32) (((i+1) * period_bytes) - 0x1); + if (i < periods - 1) { + pointer->next = kmalloc(sizeof(struct pcm_period), + GFP_KERNEL); + if (!pointer->next) { + au1000_release_dma_link(stream); + return -ENOMEM; + } + pointer = pointer->next; + } + } + pointer->next = stream->buffer; + return 0; +} + +static void au1000_dma_stop(struct audio_stream *stream) +{ + if (stream->buffer) + disable_dma(stream->dma); +} + +static void au1000_dma_start(struct audio_stream *stream) +{ + if (!stream->buffer) + return; + + init_dma(stream->dma); + if (get_dma_active_buffer(stream->dma) == 0) { + clear_dma_done0(stream->dma); + set_dma_addr0(stream->dma, stream->buffer->start); + set_dma_count0(stream->dma, stream->period_size >> 1); + set_dma_addr1(stream->dma, stream->buffer->next->start); + set_dma_count1(stream->dma, stream->period_size >> 1); + } else { + clear_dma_done1(stream->dma); + set_dma_addr1(stream->dma, stream->buffer->start); + set_dma_count1(stream->dma, stream->period_size >> 1); + set_dma_addr0(stream->dma, stream->buffer->next->start); + set_dma_count0(stream->dma, stream->period_size >> 1); + } + enable_dma_buffers(stream->dma); + start_dma(stream->dma); +} + +static irqreturn_t au1000_dma_interrupt(int irq, void *ptr) +{ + struct audio_stream *stream = (struct audio_stream *)ptr; + struct snd_pcm_substream *substream = stream->substream; + + switch (get_dma_buffer_done(stream->dma)) { + case DMA_D0: + stream->buffer = stream->buffer->next; + clear_dma_done0(stream->dma); + set_dma_addr0(stream->dma, stream->buffer->next->start); + set_dma_count0(stream->dma, stream->period_size >> 1); + enable_dma_buffer0(stream->dma); + break; + case DMA_D1: + stream->buffer = stream->buffer->next; + clear_dma_done1(stream->dma); + set_dma_addr1(stream->dma, stream->buffer->next->start); + set_dma_count1(stream->dma, stream->period_size >> 1); + enable_dma_buffer1(stream->dma); + break; + case (DMA_D0 | DMA_D1): + pr_debug("DMA %d missed interrupt.\n", stream->dma); + au1000_dma_stop(stream); + au1000_dma_start(stream); + break; + case (~DMA_D0 & ~DMA_D1): + pr_debug("DMA %d empty irq.\n", stream->dma); + } + snd_pcm_period_elapsed(substream); + return IRQ_HANDLED; +} + +static const struct snd_pcm_hardware alchemy_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH, + .formats = ALCHEMY_PCM_FMTS, + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = SNDRV_PCM_RATE_8000, + .rate_max = SNDRV_PCM_RATE_192000, + .channels_min = 2, + .channels_max = 2, + .period_bytes_min = 1024, + .period_bytes_max = 16 * 1024 - 1, + .periods_min = 4, + .periods_max = 255, + .buffer_bytes_max = 128 * 1024, + .fifo_size = 16, +}; + +static inline struct alchemy_pcm_ctx *ss_to_ctx(struct snd_pcm_substream *ss) +{ + struct snd_soc_pcm_runtime *rtd = ss->private_data; + return snd_soc_platform_get_drvdata(rtd->platform); +} + +static inline struct audio_stream *ss_to_as(struct snd_pcm_substream *ss) +{ + struct alchemy_pcm_ctx *ctx = ss_to_ctx(ss); + return &(ctx->stream[ss->stream]); +} + +static int alchemy_pcm_open(struct snd_pcm_substream *substream) +{ + struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int *dmaids, s = substream->stream; + char *name; + + dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dmaids) + return -ENODEV; /* whoa, has ordering changed? */ + + /* DMA setup */ + name = (s == SNDRV_PCM_STREAM_PLAYBACK) ? "audio-tx" : "audio-rx"; + ctx->stream[s].dma = request_au1000_dma(dmaids[s], name, + au1000_dma_interrupt, IRQF_DISABLED, + &ctx->stream[s]); + set_dma_mode(ctx->stream[s].dma, + get_dma_mode(ctx->stream[s].dma) & ~DMA_NC); + + ctx->stream[s].substream = substream; + ctx->stream[s].buffer = NULL; + snd_soc_set_runtime_hwparams(substream, &alchemy_pcm_hardware); + + return 0; +} + +static int alchemy_pcm_close(struct snd_pcm_substream *substream) +{ + struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream); + int stype = substream->stream; + + ctx->stream[stype].substream = NULL; + free_au1000_dma(ctx->stream[stype].dma); + + return 0; +} + +static int alchemy_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct audio_stream *stream = ss_to_as(substream); + int err; + + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + err = au1000_setup_dma_link(stream, + params_period_bytes(hw_params), + params_periods(hw_params)); + if (err) + snd_pcm_lib_free_pages(substream); + + return err; +} + +static int alchemy_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct audio_stream *stream = ss_to_as(substream); + au1000_release_dma_link(stream); + return snd_pcm_lib_free_pages(substream); +} + +static int alchemy_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct audio_stream *stream = ss_to_as(substream); + int err = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + au1000_dma_start(stream); + break; + case SNDRV_PCM_TRIGGER_STOP: + au1000_dma_stop(stream); + break; + default: + err = -EINVAL; + break; + } + return err; +} + +static snd_pcm_uframes_t alchemy_pcm_pointer(struct snd_pcm_substream *ss) +{ + struct audio_stream *stream = ss_to_as(ss); + long location; + + location = get_dma_residue(stream->dma); + location = stream->buffer->relative_end - location; + if (location == -1) + location = 0; + return bytes_to_frames(ss->runtime, location); +} + +static struct snd_pcm_ops alchemy_pcm_ops = { + .open = alchemy_pcm_open, + .close = alchemy_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alchemy_pcm_hw_params, + .hw_free = alchemy_pcm_hw_free, + .trigger = alchemy_pcm_trigger, + .pointer = alchemy_pcm_pointer, +}; + +static void alchemy_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int alchemy_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), 65536, (4096 * 1024) - 1); + + return 0; +} + +struct snd_soc_platform_driver alchemy_pcm_soc_platform = { + .ops = &alchemy_pcm_ops, + .pcm_new = alchemy_pcm_new, + .pcm_free = alchemy_pcm_free_dma_buffers, +}; + +static int __devinit alchemy_pcm_drvprobe(struct platform_device *pdev) +{ + struct alchemy_pcm_ctx *ctx; + int ret; + + ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + platform_set_drvdata(pdev, ctx); + + ret = snd_soc_register_platform(&pdev->dev, &alchemy_pcm_soc_platform); + if (ret) + kfree(ctx); + + return ret; +} + +static int __devexit alchemy_pcm_drvremove(struct platform_device *pdev) +{ + struct alchemy_pcm_ctx *ctx = platform_get_drvdata(pdev); + + snd_soc_unregister_platform(&pdev->dev); + kfree(ctx); + + return 0; +} + +static struct platform_driver alchemy_pcmdma_driver = { + .driver = { + .name = "alchemy-pcm-dma", + .owner = THIS_MODULE, + }, + .probe = alchemy_pcm_drvprobe, + .remove = __devexit_p(alchemy_pcm_drvremove), +}; + +static int __init alchemy_pcmdma_load(void) +{ + return platform_driver_register(&alchemy_pcmdma_driver); +} + +static void __exit alchemy_pcmdma_unload(void) +{ + platform_driver_unregister(&alchemy_pcmdma_driver); +} + +module_init(alchemy_pcmdma_load); +module_exit(alchemy_pcmdma_unload); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au1000/Au1500/Au1100 Audio DMA driver"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c new file mode 100644 index 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 <manuel.lauss@googlemail.com> + * + * Note: clock supplied to the I2S controller must be 256x samplerate. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/suspend.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/mach-au1x00/au1000.h> + +#include "psc.h" + +#define I2S_RXTX 0x00 +#define I2S_CFG 0x04 +#define I2S_ENABLE 0x08 + +#define CFG_XU (1 << 25) /* tx underflow */ +#define CFG_XO (1 << 24) +#define CFG_RU (1 << 23) +#define CFG_RO (1 << 22) +#define CFG_TR (1 << 21) +#define CFG_TE (1 << 20) +#define CFG_TF (1 << 19) +#define CFG_RR (1 << 18) +#define CFG_RF (1 << 17) +#define CFG_ICK (1 << 12) /* clock invert */ +#define CFG_PD (1 << 11) /* set to make I2SDIO INPUT */ +#define CFG_LB (1 << 10) /* loopback */ +#define CFG_IC (1 << 9) /* word select invert */ +#define CFG_FM_I2S (0 << 7) /* I2S format */ +#define CFG_FM_LJ (1 << 7) /* left-justified */ +#define CFG_FM_RJ (2 << 7) /* right-justified */ +#define CFG_FM_MASK (3 << 7) +#define CFG_TN (1 << 6) /* tx fifo en */ +#define CFG_RN (1 << 5) /* rx fifo en */ +#define CFG_SZ_8 (0x08) +#define CFG_SZ_16 (0x10) +#define CFG_SZ_18 (0x12) +#define CFG_SZ_20 (0x14) +#define CFG_SZ_24 (0x18) +#define CFG_SZ_MASK (0x1f) +#define EN_D (1 << 1) /* DISable */ +#define EN_CE (1 << 0) /* clock enable */ + +/* only limited by clock generator and board design */ +#define AU1XI2SC_RATES \ + SNDRV_PCM_RATE_CONTINUOUS + +#define AU1XI2SC_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | \ + SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_U18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S20_3BE | SNDRV_PCM_FMTBIT_U20_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE | \ + 0) + +static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg) +{ + return __raw_readl(ctx->mmio + reg); +} + +static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v) +{ + __raw_writel(v, ctx->mmio + reg); + wmb(); +} + +static int au1xi2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(cpu_dai); + unsigned long c; + int ret; + + ret = -EINVAL; + c = ctx->cfg; + + c &= ~CFG_FM_MASK; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + c |= CFG_FM_I2S; + break; + case SND_SOC_DAIFMT_MSB: + c |= CFG_FM_RJ; + break; + case SND_SOC_DAIFMT_LSB: + c |= CFG_FM_LJ; + break; + default: + goto out; + } + + c &= ~(CFG_IC | CFG_ICK); /* IB-IF */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + c |= CFG_IC | CFG_ICK; + break; + case SND_SOC_DAIFMT_NB_IF: + c |= CFG_IC; + break; + case SND_SOC_DAIFMT_IB_NF: + c |= CFG_ICK; + break; + case SND_SOC_DAIFMT_IB_IF: + break; + default: + goto out; + } + + /* I2S controller only supports master */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */ + break; + default: + goto out; + } + + ret = 0; + ctx->cfg = c; +out: + return ret; +} + +static int au1xi2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai); + int stype = SUBSTREAM_TYPE(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + /* power up */ + WR(ctx, I2S_ENABLE, EN_D | EN_CE); + WR(ctx, I2S_ENABLE, EN_CE); + ctx->cfg |= (stype == PCM_TX) ? CFG_TN : CFG_RN; + WR(ctx, I2S_CFG, ctx->cfg); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + ctx->cfg &= ~((stype == PCM_TX) ? CFG_TN : CFG_RN); + WR(ctx, I2S_CFG, ctx->cfg); + WR(ctx, I2S_ENABLE, EN_D); /* power off */ + break; + default: + return -EINVAL; + } + + return 0; +} + +static unsigned long msbits_to_reg(int msbits) +{ + switch (msbits) { + case 8: + return CFG_SZ_8; + case 16: + return CFG_SZ_16; + case 18: + return CFG_SZ_18; + case 20: + return CFG_SZ_20; + case 24: + return CFG_SZ_24; + } + return 0; +} + +static int au1xi2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai); + unsigned long v; + + v = msbits_to_reg(params->msbits); + if (!v) + return -EINVAL; + + ctx->cfg &= ~CFG_SZ_MASK; + ctx->cfg |= v; + return 0; +} + +static int au1xi2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]); + return 0; +} + +static const struct snd_soc_dai_ops au1xi2s_dai_ops = { + .startup = au1xi2s_startup, + .trigger = au1xi2s_trigger, + .hw_params = au1xi2s_hw_params, + .set_fmt = au1xi2s_set_fmt, +}; + +static struct snd_soc_dai_driver au1xi2s_dai_driver = { + .symmetric_rates = 1, + .playback = { + .rates = AU1XI2SC_RATES, + .formats = AU1XI2SC_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AU1XI2SC_RATES, + .formats = AU1XI2SC_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &au1xi2s_dai_ops, +}; + +static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) +{ + int ret; + struct resource *r; + struct au1xpsc_audio_data *ctx; + + ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r) { + ret = -ENODEV; + goto out0; + } + + ret = -EBUSY; + if (!request_mem_region(r->start, resource_size(r), pdev->name)) + goto out0; + + ctx->mmio = ioremap_nocache(r->start, resource_size(r)); + if (!ctx->mmio) + goto out1; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r) + goto out1; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) + goto out1; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + + platform_set_drvdata(pdev, ctx); + + ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver); + if (ret) + goto out1; + + return 0; + + snd_soc_unregister_dai(&pdev->dev); +out1: + release_mem_region(r->start, resource_size(r)); +out0: + kfree(ctx); + return ret; +} + +static int __devexit au1xi2s_drvremove(struct platform_device *pdev) +{ + struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev); + struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + snd_soc_unregister_dai(&pdev->dev); + + WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */ + + iounmap(ctx->mmio); + release_mem_region(r->start, resource_size(r)); + kfree(ctx); + + return 0; +} + +#ifdef CONFIG_PM +static int au1xi2s_drvsuspend(struct device *dev) +{ + struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev); + + WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */ + + return 0; +} + +static int au1xi2s_drvresume(struct device *dev) +{ + return 0; +} + +static const struct dev_pm_ops au1xi2sc_pmops = { + .suspend = au1xi2s_drvsuspend, + .resume = au1xi2s_drvresume, +}; + +#define AU1XI2SC_PMOPS (&au1xi2sc_pmops) + +#else + +#define AU1XI2SC_PMOPS NULL + +#endif + +static struct platform_driver au1xi2s_driver = { + .driver = { + .name = "alchemy-i2sc", + .owner = THIS_MODULE, + .pm = AU1XI2SC_PMOPS, + }, + .probe = au1xi2s_drvprobe, + .remove = __devexit_p(au1xi2s_drvremove), +}; + +static int __init au1xi2s_load(void) +{ + return platform_driver_register(&au1xi2s_driver); +} + +static void __exit au1xi2s_unload(void) +{ + platform_driver_unregister(&au1xi2s_driver); +} + +module_init(au1xi2s_load); +module_exit(au1xi2s_unload); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au1000/1500/1100 I2S ASoC driver"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index d0db66f24a00..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; @@ -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[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) + goto out2; + 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 | 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..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: @@ -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[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) + goto out2; + 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) */ @@ -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 b30eadd422a7..b16b2e02e0c9 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -1,7 +1,7 @@ /* - * Au12x0/Au1550 PSC ALSA ASoC audio support. + * Alchemy ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., + * (c) 2007-2011 MSC Vertriebsges.m.b.H., * Manuel Lauss <manuel.lauss@gmail.com> * * This program is free software; you can redistribute it and/or modify @@ -13,10 +13,6 @@ #ifndef _AU1X_PCM_H #define _AU1X_PCM_H -/* 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; @@ -27,15 +23,9 @@ struct au1xpsc_audio_data { unsigned long pm[2]; struct mutex lock; - struct platform_device *dmapd; + int dmaids[2]; }; -#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) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 76258f2a2ffb..666fae6e148d 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -20,6 +20,7 @@ #include <linux/regulator/driver.h> #include <linux/regulator/machine.h> #include <linux/regulator/consumer.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/tlv.h> #include <sound/pcm.h> @@ -33,73 +34,31 @@ #define SGTL5000_DAP_REG_OFFSET 0x0100 #define SGTL5000_MAX_REG_OFFSET 0x013A -/* default value of sgtl5000 registers except DAP */ -static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] = { - 0xa011, /* 0x0000, CHIP_ID. 11 stand for revison 17 */ - 0x0000, /* 0x0002, CHIP_DIG_POWER. */ - 0x0008, /* 0x0004, CHIP_CKL_CTRL */ - 0x0010, /* 0x0006, CHIP_I2S_CTRL */ - 0x0000, /* 0x0008, reserved */ - 0x0008, /* 0x000A, CHIP_SSS_CTRL */ - 0x0000, /* 0x000C, reserved */ - 0x020c, /* 0x000E, CHIP_ADCDAC_CTRL */ - 0x3c3c, /* 0x0010, CHIP_DAC_VOL */ - 0x0000, /* 0x0012, reserved */ - 0x015f, /* 0x0014, CHIP_PAD_STRENGTH */ - 0x0000, /* 0x0016, reserved */ - 0x0000, /* 0x0018, reserved */ - 0x0000, /* 0x001A, reserved */ - 0x0000, /* 0x001E, reserved */ - 0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */ - 0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */ - 0x0111, /* 0x0024, CHIP_ANN_CTRL */ - 0x0000, /* 0x0026, CHIP_LINREG_CTRL */ - 0x0000, /* 0x0028, CHIP_REF_CTRL */ - 0x0000, /* 0x002A, CHIP_MIC_CTRL */ - 0x0000, /* 0x002C, CHIP_LINE_OUT_CTRL */ - 0x0404, /* 0x002E, CHIP_LINE_OUT_VOL */ - 0x7060, /* 0x0030, CHIP_ANA_POWER */ - 0x5000, /* 0x0032, CHIP_PLL_CTRL */ - 0x0000, /* 0x0034, CHIP_CLK_TOP_CTRL */ - 0x0000, /* 0x0036, CHIP_ANA_STATUS */ - 0x0000, /* 0x0038, reserved */ - 0x0000, /* 0x003A, CHIP_ANA_TEST2 */ - 0x0000, /* 0x003C, CHIP_SHORT_CTRL */ - 0x0000, /* reserved */ -}; - -/* default value of dap registers */ -static const u16 sgtl5000_dap_regs[] = { - 0x0000, /* 0x0100, DAP_CONTROL */ - 0x0000, /* 0x0102, DAP_PEQ */ - 0x0040, /* 0x0104, DAP_BASS_ENHANCE */ - 0x051f, /* 0x0106, DAP_BASS_ENHANCE_CTRL */ - 0x0000, /* 0x0108, DAP_AUDIO_EQ */ - 0x0040, /* 0x010A, DAP_SGTL_SURROUND */ - 0x0000, /* 0x010C, DAP_FILTER_COEF_ACCESS */ - 0x0000, /* 0x010E, DAP_COEF_WR_B0_MSB */ - 0x0000, /* 0x0110, DAP_COEF_WR_B0_LSB */ - 0x0000, /* 0x0112, reserved */ - 0x0000, /* 0x0114, reserved */ - 0x002f, /* 0x0116, DAP_AUDIO_EQ_BASS_BAND0 */ - 0x002f, /* 0x0118, DAP_AUDIO_EQ_BAND0 */ - 0x002f, /* 0x011A, DAP_AUDIO_EQ_BAND2 */ - 0x002f, /* 0x011C, DAP_AUDIO_EQ_BAND3 */ - 0x002f, /* 0x011E, DAP_AUDIO_EQ_TREBLE_BAND4 */ - 0x8000, /* 0x0120, DAP_MAIN_CHAN */ - 0x0000, /* 0x0122, DAP_MIX_CHAN */ - 0x0510, /* 0x0124, DAP_AVC_CTRL */ - 0x1473, /* 0x0126, DAP_AVC_THRESHOLD */ - 0x0028, /* 0x0128, DAP_AVC_ATTACK */ - 0x0050, /* 0x012A, DAP_AVC_DECAY */ - 0x0000, /* 0x012C, DAP_COEF_WR_B1_MSB */ - 0x0000, /* 0x012E, DAP_COEF_WR_B1_LSB */ - 0x0000, /* 0x0130, DAP_COEF_WR_B2_MSB */ - 0x0000, /* 0x0132, DAP_COEF_WR_B2_LSB */ - 0x0000, /* 0x0134, DAP_COEF_WR_A1_MSB */ - 0x0000, /* 0x0136, DAP_COEF_WR_A1_LSB */ - 0x0000, /* 0x0138, DAP_COEF_WR_A2_MSB */ - 0x0000, /* 0x013A, DAP_COEF_WR_A2_LSB */ +/* default value of sgtl5000 registers */ +static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET] = { + [SGTL5000_CHIP_CLK_CTRL] = 0x0008, + [SGTL5000_CHIP_I2S_CTRL] = 0x0010, + [SGTL5000_CHIP_SSS_CTRL] = 0x0008, + [SGTL5000_CHIP_DAC_VOL] = 0x3c3c, + [SGTL5000_CHIP_PAD_STRENGTH] = 0x015f, + [SGTL5000_CHIP_ANA_HP_CTRL] = 0x1818, + [SGTL5000_CHIP_ANA_CTRL] = 0x0111, + [SGTL5000_CHIP_LINE_OUT_VOL] = 0x0404, + [SGTL5000_CHIP_ANA_POWER] = 0x7060, + [SGTL5000_CHIP_PLL_CTRL] = 0x5000, + [SGTL5000_DAP_BASS_ENHANCE] = 0x0040, + [SGTL5000_DAP_BASS_ENHANCE_CTRL] = 0x051f, + [SGTL5000_DAP_SURROUND] = 0x0040, + [SGTL5000_DAP_EQ_BASS_BAND0] = 0x002f, + [SGTL5000_DAP_EQ_BASS_BAND1] = 0x002f, + [SGTL5000_DAP_EQ_BASS_BAND2] = 0x002f, + [SGTL5000_DAP_EQ_BASS_BAND3] = 0x002f, + [SGTL5000_DAP_EQ_BASS_BAND4] = 0x002f, + [SGTL5000_DAP_MAIN_CHAN] = 0x8000, + [SGTL5000_DAP_AVC_CTRL] = 0x0510, + [SGTL5000_DAP_AVC_THRESHOLD] = 0x1473, + [SGTL5000_DAP_AVC_ATTACK] = 0x0028, + [SGTL5000_DAP_AVC_DECAY] = 0x0050, }; /* regulator supplies for sgtl5000, VDDD is an optional external supply */ @@ -1023,12 +982,10 @@ static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state) static int sgtl5000_restore_regs(struct snd_soc_codec *codec) { u16 *cache = codec->reg_cache; - int i; - int regular_regs = SGTL5000_CHIP_SHORT_CTRL >> 1; + u16 reg; /* restore regular registers */ - for (i = 0; i < regular_regs; i++) { - int reg = i << 1; + for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) { /* this regs depends on the others */ if (reg == SGTL5000_CHIP_ANA_POWER || @@ -1038,35 +995,31 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) reg == SGTL5000_CHIP_CLK_CTRL) continue; - snd_soc_write(codec, reg, cache[i]); + snd_soc_write(codec, reg, cache[reg]); } /* restore dap registers */ - for (i = SGTL5000_DAP_REG_OFFSET >> 1; - i < SGTL5000_MAX_REG_OFFSET >> 1; i++) { - int reg = i << 1; - - snd_soc_write(codec, reg, cache[i]); - } + for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2) + snd_soc_write(codec, reg, cache[reg]); /* * restore power and other regs according * to set_power() and set_clock() */ snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, - cache[SGTL5000_CHIP_LINREG_CTRL >> 1]); + cache[SGTL5000_CHIP_LINREG_CTRL]); snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, - cache[SGTL5000_CHIP_ANA_POWER >> 1]); + cache[SGTL5000_CHIP_ANA_POWER]); snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, - cache[SGTL5000_CHIP_CLK_CTRL >> 1]); + cache[SGTL5000_CHIP_CLK_CTRL]); snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, - cache[SGTL5000_CHIP_REF_CTRL >> 1]); + cache[SGTL5000_CHIP_REF_CTRL]); snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - cache[SGTL5000_CHIP_LINE_OUT_CTRL >> 1]); + cache[SGTL5000_CHIP_LINE_OUT_CTRL]); return 0; } @@ -1454,16 +1407,6 @@ static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, if (!sgtl5000) return -ENOMEM; - /* - * copy DAP default values to default value array. - * sgtl5000 register space has a big hole, merge it - * at init phase makes life easy. - * FIXME: should we drop 'const' of sgtl5000_regs? - */ - memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)), - sgtl5000_dap_regs, - SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET); - i2c_set_clientdata(client, sgtl5000); ret = snd_soc_register_codec(&client->dev, @@ -1494,10 +1437,17 @@ 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), 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); } 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, 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, 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 <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/spi/spi.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -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), 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 09e680ae88b2..fb5c96163610 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; @@ -2972,13 +2973,14 @@ 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; 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..017522e7cef9 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -18,6 +18,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/mfd/wm8994/registers.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -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; @@ -168,24 +179,25 @@ 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); /* 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); @@ -217,7 +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 */ @@ -440,9 +452,8 @@ static int hp_event(struct snd_soc_dapm_widget *w, reg |= WM8993_HPOUT1L_DLY | WM8993_HPOUT1R_DLY; snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg); - /* Smallest supported update interval */ snd_soc_update_bits(codec, WM8993_DC_SERVO_1, - WM8993_DCS_TIMER_PERIOD_01_MASK, 1); + WM8993_DCS_TIMER_PERIOD_01_MASK, 0); calibrate_dc_servo(codec); 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; 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 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 <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dmaengine.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/dma.h> +#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 <mach/dma.h> + +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 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 <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <mach/dma.h> +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/mxs.h> + +#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 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 <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/soc-dapm.h> +#include <asm/mach-types.h> + +#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"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83ad8ca27490..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'; @@ -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..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) { @@ -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); } /** |