diff options
Diffstat (limited to 'sound')
99 files changed, 6032 insertions, 1175 deletions
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 8224db5f0434..1381db853ef0 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -7,6 +7,8 @@ menuconfig SND_SOC select SND_PCM select AC97_BUS if SND_SOC_AC97_BUS select SND_JACK if INPUT=y || INPUT=SND + select REGMAP_I2C if I2C + select REGMAP_SPI if SPI_MASTER ---help--- If you want ASoC support, you should say Y here and also to the @@ -51,6 +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/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 1aac2f4dbcf6..2909bfaed265 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -383,14 +383,17 @@ static int __init playpaq_asoc_init(void) _gclk0 = clk_get(NULL, "gclk0"); if (IS_ERR(_gclk0)) { _gclk0 = NULL; + ret = PTR_ERR(_gclk0); goto err_gclk0; } _pll0 = clk_get(NULL, "pll0"); if (IS_ERR(_pll0)) { _pll0 = NULL; + ret = PTR_ERR(_pll0); goto err_pll0; } - if (clk_set_parent(_gclk0, _pll0)) { + ret = clk_set_parent(_gclk0, _pll0); + if (ret) { pr_warning("snd-soc-playpaq: " "Failed to set PLL0 as parent for DAC clock\n"); goto err_set_clk; 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..13802ff7cf05 --- /dev/null +++ b/sound/soc/au1x/ac97c.c @@ -0,0 +1,363 @@ +/* + * 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; + +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..19e0d2a9c828 --- /dev/null +++ b/sound/soc/au1x/i2sc.c @@ -0,0 +1,346 @@ +/* + * 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; + +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/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index fe9d548a6837..9f6bc55fc399 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -27,6 +27,19 @@ config SND_SOC_BFIN_EVAL_ADAU1701 board connected to one of the Blackfin evaluation boards like the BF5XX-STAMP or BF5XX-EZKIT. +config SND_SOC_BFIN_EVAL_ADAU1373 + tristate "Support for the EVAL-ADAU1373 board on Blackfin eval boards" + depends on SND_BF5XX_I2S && I2C + select SND_BF5XX_SOC_I2S + select SND_SOC_ADAU1373 + help + Say Y if you want to add support for the Analog Devices EVAL-ADAU1373 + board connected to one of the Blackfin evaluation boards like the + BF5XX-STAMP or BF5XX-EZKIT. + + Note: This driver assumes that first ADAU1373 DAI is connected to the + first SPORT port on the BF5XX board. + config SND_SOC_BFIN_EVAL_ADAV80X tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards" depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 6018bf52a234..1bf86ccaa8de 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -21,6 +21,7 @@ snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o snd-ad73311-objs := bf5xx-ad73311.o snd-ad193x-objs := bf5xx-ad193x.o +snd-soc-bfin-eval-adau1373-objs := bfin-eval-adau1373.o snd-soc-bfin-eval-adau1701-objs := bfin-eval-adau1701.o snd-soc-bfin-eval-adav80x-objs := bfin-eval-adav80x.o @@ -29,5 +30,6 @@ obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o +obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1373) += snd-soc-bfin-eval-adau1373.o obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701) += snd-soc-bfin-eval-adau1701.o obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) += snd-soc-bfin-eval-adav80x.o diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 732a247f2527..b94eb7ef7d16 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -128,7 +128,7 @@ static int snd_ad73311_configure(void) return 0; } -static int bf5xx_probe(struct platform_device *pdev) +static int bf5xx_probe(struct snd_soc_card *card) { int err; if (gpio_request(GPIO_SE, "AD73311_SE")) { diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c new file mode 100644 index 000000000000..8df2a3b0cb36 --- /dev/null +++ b/sound/soc/blackfin/bfin-eval-adau1373.c @@ -0,0 +1,202 @@ +/* + * Machine driver for EVAL-ADAU1373 on Analog Devices bfin + * evaluation boards. + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars@metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include "../codecs/adau1373.h" + +static const struct snd_soc_dapm_widget bfin_eval_adau1373_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Line In1", NULL), + SND_SOC_DAPM_LINE("Line In2", NULL), + SND_SOC_DAPM_LINE("Line In3", NULL), + SND_SOC_DAPM_LINE("Line In4", NULL), + + SND_SOC_DAPM_LINE("Line Out1", NULL), + SND_SOC_DAPM_LINE("Line Out2", NULL), + SND_SOC_DAPM_LINE("Stereo Out", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_HP("Earpiece", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route bfin_eval_adau1373_dapm_routes[] = { + { "AIN1L", NULL, "Line In1" }, + { "AIN1R", NULL, "Line In1" }, + { "AIN2L", NULL, "Line In2" }, + { "AIN2R", NULL, "Line In2" }, + { "AIN3L", NULL, "Line In3" }, + { "AIN3R", NULL, "Line In3" }, + { "AIN4L", NULL, "Line In4" }, + { "AIN4R", NULL, "Line In4" }, + + /* MICBIAS can be connected via a jumper to the line-in jack, since w + don't know which one is going to be used, just power both. */ + { "Line In1", NULL, "MICBIAS1" }, + { "Line In2", NULL, "MICBIAS1" }, + { "Line In3", NULL, "MICBIAS1" }, + { "Line In4", NULL, "MICBIAS1" }, + { "Line In1", NULL, "MICBIAS2" }, + { "Line In2", NULL, "MICBIAS2" }, + { "Line In3", NULL, "MICBIAS2" }, + { "Line In4", NULL, "MICBIAS2" }, + + { "Line Out1", NULL, "LOUT1L" }, + { "Line Out1", NULL, "LOUT1R" }, + { "Line Out2", NULL, "LOUT2L" }, + { "Line Out2", NULL, "LOUT2R" }, + { "Headphone", NULL, "HPL" }, + { "Headphone", NULL, "HPR" }, + { "Earpiece", NULL, "EP" }, + { "Speaker", NULL, "SPKL" }, + { "Stereo Out", NULL, "SPKR" }, +}; + +static int bfin_eval_adau1373_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + int pll_rate; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret) + return ret; + + switch (params_rate(params)) { + case 48000: + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + pll_rate = 48000 * 1024; + break; + case 44100: + case 7350: + case 11025: + case 14700: + case 22050: + case 29400: + pll_rate = 44100 * 1024; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1, + ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate, + SND_SOC_CLOCK_IN); + + return ret; +} + +static int bfin_eval_adau1373_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int pll_rate = 48000 * 1024; + int ret; + + ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1, + ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate, + SND_SOC_CLOCK_IN); + + return ret; +} +static struct snd_soc_ops bfin_eval_adau1373_ops = { + .hw_params = bfin_eval_adau1373_hw_params, +}; + +static struct snd_soc_dai_link bfin_eval_adau1373_dai = { + .name = "adau1373", + .stream_name = "adau1373", + .cpu_dai_name = "bfin-i2s.0", + .codec_dai_name = "adau1373-aif1", + .platform_name = "bfin-i2s-pcm-audio", + .codec_name = "adau1373.0-001a", + .ops = &bfin_eval_adau1373_ops, + .init = bfin_eval_adau1373_codec_init, +}; + +static struct snd_soc_card bfin_eval_adau1373 = { + .name = "bfin-eval-adau1373", + .dai_link = &bfin_eval_adau1373_dai, + .num_links = 1, + + .dapm_widgets = bfin_eval_adau1373_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1373_dapm_widgets), + .dapm_routes = bfin_eval_adau1373_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1373_dapm_routes), +}; + +static int bfin_eval_adau1373_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &bfin_eval_adau1373; + + card->dev = &pdev->dev; + + return snd_soc_register_card(&bfin_eval_adau1373); +} + +static int __devexit bfin_eval_adau1373_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver bfin_eval_adau1373_driver = { + .driver = { + .name = "bfin-eval-adau1373", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = bfin_eval_adau1373_probe, + .remove = __devexit_p(bfin_eval_adau1373_remove), +}; + +static int __init bfin_eval_adau1373_init(void) +{ + return platform_driver_register(&bfin_eval_adau1373_driver); +} +module_init(bfin_eval_adau1373_init); + +static void __exit bfin_eval_adau1373_exit(void) +{ + platform_driver_unregister(&bfin_eval_adau1373_driver); +} +module_exit(bfin_eval_adau1373_exit); + +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_DESCRIPTION("ALSA SoC bfin adau1373 driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:bfin-eval-adau1373"); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 665d9240c4ae..71b46c8f70d7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 + select SND_SOC_ADAU1373 if I2C select SND_SOC_ADAV80X select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER @@ -139,6 +140,9 @@ config SND_SOC_ADAU1701 select SIGMA tristate +config SND_SOC_ADAU1373 + tristate + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5119a7e2c1a8..70c1769acd15 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -5,6 +5,7 @@ snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o +snd-soc-adau1373-objs := adau1373.o snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o @@ -100,6 +101,7 @@ obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index eedb6f5e5823..f934670199a5 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -23,7 +23,7 @@ /* codec private data */ struct ad193x_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk; }; @@ -349,10 +349,8 @@ static int ad193x_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - if (ad193x->control_type == SND_SOC_I2C) - ret = snd_soc_codec_set_cache_io(codec, 8, 8, ad193x->control_type); - else - ret = snd_soc_codec_set_cache_io(codec, 16, 8, ad193x->control_type); + codec->control_data = ad193x->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; @@ -388,6 +386,14 @@ static struct snd_soc_codec_driver soc_codec_dev_ad193x = { }; #if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config ad193x_spi_regmap_config = { + .val_bits = 8, + .reg_bits = 16, + .read_flag_mask = 0x09, + .write_flag_mask = 0x08, +}; + static int __devinit ad193x_spi_probe(struct spi_device *spi) { struct ad193x_priv *ad193x; @@ -397,20 +403,36 @@ static int __devinit ad193x_spi_probe(struct spi_device *spi) if (ad193x == NULL) return -ENOMEM; + ad193x->regmap = regmap_init_spi(spi, &ad193x_spi_regmap_config); + if (IS_ERR(ad193x->regmap)) { + ret = PTR_ERR(ad193x->regmap); + goto err_free; + } + spi_set_drvdata(spi, ad193x); - ad193x->control_type = SND_SOC_SPI; ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_ad193x, &ad193x_dai, 1); if (ret < 0) - kfree(ad193x); + goto err_regmap_exit; + + return 0; + +err_regmap_exit: + regmap_exit(ad193x->regmap); +err_free: + kfree(ad193x); + return ret; } static int __devexit ad193x_spi_remove(struct spi_device *spi) { + struct ad193x_priv *ad193x = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); + regmap_exit(ad193x->regmap); + kfree(ad193x); return 0; } @@ -425,6 +447,12 @@ static struct spi_driver ad193x_spi_driver = { #endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +static const struct regmap_config ad193x_i2c_regmap_config = { + .val_bits = 8, + .reg_bits = 8, +}; + static const struct i2c_device_id ad193x_id[] = { { "ad1936", 0 }, { "ad1937", 0 }, @@ -442,20 +470,35 @@ static int __devinit ad193x_i2c_probe(struct i2c_client *client, if (ad193x == NULL) return -ENOMEM; + ad193x->regmap = regmap_init_i2c(client, &ad193x_i2c_regmap_config); + if (IS_ERR(ad193x->regmap)) { + ret = PTR_ERR(ad193x->regmap); + goto err_free; + } + i2c_set_clientdata(client, ad193x); - ad193x->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_ad193x, &ad193x_dai, 1); if (ret < 0) - kfree(ad193x); + goto err_regmap_exit; + + return 0; + +err_regmap_exit: + regmap_exit(ad193x->regmap); +err_free: + kfree(ad193x); return ret; } static int __devexit ad193x_i2c_remove(struct i2c_client *client) { + struct ad193x_priv *ad193x = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + regmap_exit(ad193x->regmap); + kfree(ad193x); return 0; } diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index cccc2e8e5fbd..536e5f2b136e 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -9,20 +9,20 @@ #ifndef __AD193X_H__ #define __AD193X_H__ -#define AD193X_PLL_CLK_CTRL0 0x800 +#define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 #define AD193X_PLL_INPUT_MASK (~0x6) #define AD193X_PLL_INPUT_256 (0 << 1) #define AD193X_PLL_INPUT_384 (1 << 1) #define AD193X_PLL_INPUT_512 (2 << 1) #define AD193X_PLL_INPUT_768 (3 << 1) -#define AD193X_PLL_CLK_CTRL1 0x801 -#define AD193X_DAC_CTRL0 0x802 +#define AD193X_PLL_CLK_CTRL1 0x01 +#define AD193X_DAC_CTRL0 0x02 #define AD193X_DAC_POWERDOWN 0x01 #define AD193X_DAC_SERFMT_MASK 0xC0 #define AD193X_DAC_SERFMT_STEREO (0 << 6) #define AD193X_DAC_SERFMT_TDM (1 << 6) -#define AD193X_DAC_CTRL1 0x803 +#define AD193X_DAC_CTRL1 0x03 #define AD193X_DAC_2_CHANNELS 0 #define AD193X_DAC_4_CHANNELS 1 #define AD193X_DAC_8_CHANNELS 2 @@ -33,11 +33,11 @@ #define AD193X_DAC_BCLK_MASTER (1 << 5) #define AD193X_DAC_LEFT_HIGH (1 << 3) #define AD193X_DAC_BCLK_INV (1 << 7) -#define AD193X_DAC_CTRL2 0x804 +#define AD193X_DAC_CTRL2 0x04 #define AD193X_DAC_WORD_LEN_SHFT 3 #define AD193X_DAC_WORD_LEN_MASK 0x18 #define AD193X_DAC_MASTER_MUTE 1 -#define AD193X_DAC_CHNL_MUTE 0x805 +#define AD193X_DAC_CHNL_MUTE 0x05 #define AD193X_DACL1_MUTE 0 #define AD193X_DACR1_MUTE 1 #define AD193X_DACL2_MUTE 2 @@ -46,28 +46,28 @@ #define AD193X_DACR3_MUTE 5 #define AD193X_DACL4_MUTE 6 #define AD193X_DACR4_MUTE 7 -#define AD193X_DAC_L1_VOL 0x806 -#define AD193X_DAC_R1_VOL 0x807 -#define AD193X_DAC_L2_VOL 0x808 -#define AD193X_DAC_R2_VOL 0x809 -#define AD193X_DAC_L3_VOL 0x80a -#define AD193X_DAC_R3_VOL 0x80b -#define AD193X_DAC_L4_VOL 0x80c -#define AD193X_DAC_R4_VOL 0x80d -#define AD193X_ADC_CTRL0 0x80e +#define AD193X_DAC_L1_VOL 0x06 +#define AD193X_DAC_R1_VOL 0x07 +#define AD193X_DAC_L2_VOL 0x08 +#define AD193X_DAC_R2_VOL 0x09 +#define AD193X_DAC_L3_VOL 0x0a +#define AD193X_DAC_R3_VOL 0x0b +#define AD193X_DAC_L4_VOL 0x0c +#define AD193X_DAC_R4_VOL 0x0d +#define AD193X_ADC_CTRL0 0x0e #define AD193X_ADC_POWERDOWN 0x01 #define AD193X_ADC_HIGHPASS_FILTER 1 #define AD193X_ADCL1_MUTE 2 #define AD193X_ADCR1_MUTE 3 #define AD193X_ADCL2_MUTE 4 #define AD193X_ADCR2_MUTE 5 -#define AD193X_ADC_CTRL1 0x80f +#define AD193X_ADC_CTRL1 0x0f #define AD193X_ADC_SERFMT_MASK 0x60 #define AD193X_ADC_SERFMT_STEREO (0 << 5) #define AD193X_ADC_SERFMT_TDM (1 << 5) #define AD193X_ADC_SERFMT_AUX (2 << 5) #define AD193X_ADC_WORD_LEN_MASK 0x3 -#define AD193X_ADC_CTRL2 0x810 +#define AD193X_ADC_CTRL2 0x10 #define AD193X_ADC_2_CHANNELS 0 #define AD193X_ADC_4_CHANNELS 1 #define AD193X_ADC_8_CHANNELS 2 diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 923b364a3e41..4c0fc30a4ccb 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -200,18 +200,22 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) } /* Read out vendor ID to make sure it is ad1980 */ - if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144) + if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144) { + ret = -ENODEV; goto reset_err; + } vendor_id2 = ac97_read(codec, AC97_VENDOR_ID2); if (vendor_id2 != 0x5370) { - if (vendor_id2 != 0x5374) + if (vendor_id2 != 0x5374) { + ret = -ENODEV; goto reset_err; - else + } else { printk(KERN_WARNING "ad1980: " "Found AD1981 - only 2/2 IN/OUT Channels " "supported\n"); + } } /* unmute captures and playbacks volume */ diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c new file mode 100644 index 000000000000..2aa40c3731d0 --- /dev/null +++ b/sound/soc/codecs/adau1373.c @@ -0,0 +1,1414 @@ +/* + * Analog Devices ADAU1373 Audio Codec drive + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars@metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/gcd.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/adau1373.h> + +#include "adau1373.h" + +struct adau1373_dai { + unsigned int clk_src; + unsigned int sysclk; + bool enable_src; + bool master; +}; + +struct adau1373 { + struct adau1373_dai dais[3]; +}; + +#define ADAU1373_INPUT_MODE 0x00 +#define ADAU1373_AINL_CTRL(x) (0x01 + (x) * 2) +#define ADAU1373_AINR_CTRL(x) (0x02 + (x) * 2) +#define ADAU1373_LLINE_OUT(x) (0x9 + (x) * 2) +#define ADAU1373_RLINE_OUT(x) (0xa + (x) * 2) +#define ADAU1373_LSPK_OUT 0x0d +#define ADAU1373_RSPK_OUT 0x0e +#define ADAU1373_LHP_OUT 0x0f +#define ADAU1373_RHP_OUT 0x10 +#define ADAU1373_ADC_GAIN 0x11 +#define ADAU1373_LADC_MIXER 0x12 +#define ADAU1373_RADC_MIXER 0x13 +#define ADAU1373_LLINE1_MIX 0x14 +#define ADAU1373_RLINE1_MIX 0x15 +#define ADAU1373_LLINE2_MIX 0x16 +#define ADAU1373_RLINE2_MIX 0x17 +#define ADAU1373_LSPK_MIX 0x18 +#define ADAU1373_RSPK_MIX 0x19 +#define ADAU1373_LHP_MIX 0x1a +#define ADAU1373_RHP_MIX 0x1b +#define ADAU1373_EP_MIX 0x1c +#define ADAU1373_HP_CTRL 0x1d +#define ADAU1373_HP_CTRL2 0x1e +#define ADAU1373_LS_CTRL 0x1f +#define ADAU1373_EP_CTRL 0x21 +#define ADAU1373_MICBIAS_CTRL1 0x22 +#define ADAU1373_MICBIAS_CTRL2 0x23 +#define ADAU1373_OUTPUT_CTRL 0x24 +#define ADAU1373_PWDN_CTRL1 0x25 +#define ADAU1373_PWDN_CTRL2 0x26 +#define ADAU1373_PWDN_CTRL3 0x27 +#define ADAU1373_DPLL_CTRL(x) (0x28 + (x) * 7) +#define ADAU1373_PLL_CTRL1(x) (0x29 + (x) * 7) +#define ADAU1373_PLL_CTRL2(x) (0x2a + (x) * 7) +#define ADAU1373_PLL_CTRL3(x) (0x2b + (x) * 7) +#define ADAU1373_PLL_CTRL4(x) (0x2c + (x) * 7) +#define ADAU1373_PLL_CTRL5(x) (0x2d + (x) * 7) +#define ADAU1373_PLL_CTRL6(x) (0x2e + (x) * 7) +#define ADAU1373_PLL_CTRL7(x) (0x2f + (x) * 7) +#define ADAU1373_HEADDECT 0x36 +#define ADAU1373_ADC_DAC_STATUS 0x37 +#define ADAU1373_ADC_CTRL 0x3c +#define ADAU1373_DAI(x) (0x44 + (x)) +#define ADAU1373_CLK_SRC_DIV(x) (0x40 + (x) * 2) +#define ADAU1373_BCLKDIV(x) (0x47 + (x)) +#define ADAU1373_SRC_RATIOA(x) (0x4a + (x) * 2) +#define ADAU1373_SRC_RATIOB(x) (0x4b + (x) * 2) +#define ADAU1373_DEEMP_CTRL 0x50 +#define ADAU1373_SRC_DAI_CTRL(x) (0x51 + (x)) +#define ADAU1373_DIN_MIX_CTRL(x) (0x56 + (x)) +#define ADAU1373_DOUT_MIX_CTRL(x) (0x5b + (x)) +#define ADAU1373_DAI_PBL_VOL(x) (0x62 + (x) * 2) +#define ADAU1373_DAI_PBR_VOL(x) (0x63 + (x) * 2) +#define ADAU1373_DAI_RECL_VOL(x) (0x68 + (x) * 2) +#define ADAU1373_DAI_RECR_VOL(x) (0x69 + (x) * 2) +#define ADAU1373_DAC1_PBL_VOL 0x6e +#define ADAU1373_DAC1_PBR_VOL 0x6f +#define ADAU1373_DAC2_PBL_VOL 0x70 +#define ADAU1373_DAC2_PBR_VOL 0x71 +#define ADAU1373_ADC_RECL_VOL 0x72 +#define ADAU1373_ADC_RECR_VOL 0x73 +#define ADAU1373_DMIC_RECL_VOL 0x74 +#define ADAU1373_DMIC_RECR_VOL 0x75 +#define ADAU1373_VOL_GAIN1 0x76 +#define ADAU1373_VOL_GAIN2 0x77 +#define ADAU1373_VOL_GAIN3 0x78 +#define ADAU1373_HPF_CTRL 0x7d +#define ADAU1373_BASS1 0x7e +#define ADAU1373_BASS2 0x7f +#define ADAU1373_DRC(x) (0x80 + (x) * 0x10) +#define ADAU1373_3D_CTRL1 0xc0 +#define ADAU1373_3D_CTRL2 0xc1 +#define ADAU1373_FDSP_SEL1 0xdc +#define ADAU1373_FDSP_SEL2 0xdd +#define ADAU1373_FDSP_SEL3 0xde +#define ADAU1373_FDSP_SEL4 0xdf +#define ADAU1373_DIGMICCTRL 0xe2 +#define ADAU1373_DIGEN 0xeb +#define ADAU1373_SOFT_RESET 0xff + + +#define ADAU1373_PLL_CTRL6_DPLL_BYPASS BIT(1) +#define ADAU1373_PLL_CTRL6_PLL_EN BIT(0) + +#define ADAU1373_DAI_INVERT_BCLK BIT(7) +#define ADAU1373_DAI_MASTER BIT(6) +#define ADAU1373_DAI_INVERT_LRCLK BIT(4) +#define ADAU1373_DAI_WLEN_16 0x0 +#define ADAU1373_DAI_WLEN_20 0x4 +#define ADAU1373_DAI_WLEN_24 0x8 +#define ADAU1373_DAI_WLEN_32 0xc +#define ADAU1373_DAI_WLEN_MASK 0xc +#define ADAU1373_DAI_FORMAT_RIGHT_J 0x0 +#define ADAU1373_DAI_FORMAT_LEFT_J 0x1 +#define ADAU1373_DAI_FORMAT_I2S 0x2 +#define ADAU1373_DAI_FORMAT_DSP 0x3 + +#define ADAU1373_BCLKDIV_SOURCE BIT(5) +#define ADAU1373_BCLKDIV_32 0x03 +#define ADAU1373_BCLKDIV_64 0x02 +#define ADAU1373_BCLKDIV_128 0x01 +#define ADAU1373_BCLKDIV_256 0x00 + +#define ADAU1373_ADC_CTRL_PEAK_DETECT BIT(0) +#define ADAU1373_ADC_CTRL_RESET BIT(1) +#define ADAU1373_ADC_CTRL_RESET_FORCE BIT(2) + +#define ADAU1373_OUTPUT_CTRL_LDIFF BIT(3) +#define ADAU1373_OUTPUT_CTRL_LNFBEN BIT(2) + +#define ADAU1373_PWDN_CTRL3_PWR_EN BIT(0) + +#define ADAU1373_EP_CTRL_MICBIAS1_OFFSET 4 +#define ADAU1373_EP_CTRL_MICBIAS2_OFFSET 2 + +static const uint8_t adau1373_default_regs[] = { + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x00 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x10 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x20 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, /* 0x30 */ + 0x00, 0x00, 0x00, 0x80, 0x00, 0x01, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x0a, 0x0a, 0x0a, 0x00, /* 0x40 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x08, 0x08, 0x08, 0x00, 0x00, 0x00, 0x00, /* 0x50 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x60 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x70 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x80 */ + 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, + 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x90 */ + 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, + 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0xa0 */ + 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, + 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff, 0xff, /* 0xb0 */ + 0xff, 0xff, 0xff, 0xff, 0xff, 0x1f, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xc0 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xd0 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 0xe0 */ + 0x00, 0x1f, 0x0f, 0x00, 0x00, +}; + +static const unsigned int adau1373_out_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 7, TLV_DB_SCALE_ITEM(-7900, 400, 1), + 8, 15, TLV_DB_SCALE_ITEM(-4700, 300, 0), + 16, 23, TLV_DB_SCALE_ITEM(-2300, 200, 0), + 24, 31, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const DECLARE_TLV_DB_MINMAX(adau1373_digital_tlv, -9563, 0); +static const DECLARE_TLV_DB_SCALE(adau1373_in_pga_tlv, -1300, 100, 1); +static const DECLARE_TLV_DB_SCALE(adau1373_ep_tlv, -600, 600, 1); + +static const DECLARE_TLV_DB_SCALE(adau1373_input_boost_tlv, 0, 2000, 0); +static const DECLARE_TLV_DB_SCALE(adau1373_gain_boost_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adau1373_speaker_boost_tlv, 1200, 600, 0); + +static const char *adau1373_fdsp_sel_text[] = { + "None", + "Channel 1", + "Channel 2", + "Channel 3", + "Channel 4", + "Channel 5", +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_drc1_channel_enum, + ADAU1373_FDSP_SEL1, 4, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_drc2_channel_enum, + ADAU1373_FDSP_SEL1, 0, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_drc3_channel_enum, + ADAU1373_FDSP_SEL2, 0, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_channel_enum, + ADAU1373_FDSP_SEL3, 0, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_bass_channel_enum, + ADAU1373_FDSP_SEL4, 4, adau1373_fdsp_sel_text); + +static const char *adau1373_hpf_cutoff_text[] = { + "3.7Hz", "50Hz", "100Hz", "150Hz", "200Hz", "250Hz", "300Hz", "350Hz", + "400Hz", "450Hz", "500Hz", "550Hz", "600Hz", "650Hz", "700Hz", "750Hz", + "800Hz", +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_cutoff_enum, + ADAU1373_HPF_CTRL, 3, adau1373_hpf_cutoff_text); + +static const char *adau1373_bass_lpf_cutoff_text[] = { + "801Hz", "1001Hz", +}; + +static const char *adau1373_bass_clip_level_text[] = { + "0.125", "0.250", "0.370", "0.500", "0.625", "0.750", "0.875", +}; + +static const unsigned int adau1373_bass_clip_level_values[] = { + 1, 2, 3, 4, 5, 6, 7, +}; + +static const char *adau1373_bass_hpf_cutoff_text[] = { + "158Hz", "232Hz", "347Hz", "520Hz", +}; + +static const unsigned int adau1373_bass_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 2, TLV_DB_SCALE_ITEM(-600, 600, 1), + 3, 4, TLV_DB_SCALE_ITEM(950, 250, 0), + 5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0), +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_bass_lpf_cutoff_enum, + ADAU1373_BASS1, 5, adau1373_bass_lpf_cutoff_text); + +static const SOC_VALUE_ENUM_SINGLE_DECL(adau1373_bass_clip_level_enum, + ADAU1373_BASS1, 2, 7, adau1373_bass_clip_level_text, + adau1373_bass_clip_level_values); + +static const SOC_ENUM_SINGLE_DECL(adau1373_bass_hpf_cutoff_enum, + ADAU1373_BASS1, 0, adau1373_bass_hpf_cutoff_text); + +static const char *adau1373_3d_level_text[] = { + "0%", "6.67%", "13.33%", "20%", "26.67%", "33.33%", + "40%", "46.67%", "53.33%", "60%", "66.67%", "73.33%", + "80%", "86.67", "99.33%", "100%" +}; + +static const char *adau1373_3d_cutoff_text[] = { + "No 3D", "0.03125 fs", "0.04583 fs", "0.075 fs", "0.11458 fs", + "0.16875 fs", "0.27083 fs" +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_3d_level_enum, + ADAU1373_3D_CTRL1, 4, adau1373_3d_level_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_3d_cutoff_enum, + ADAU1373_3D_CTRL1, 0, adau1373_3d_cutoff_text); + +static const unsigned int adau1373_3d_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_LINEAR_ITEM(-1800, -120), +}; + +static const char *adau1373_lr_mux_text[] = { + "Mute", + "Right Channel (L+R)", + "Left Channel (L+R)", + "Stereo", +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_lineout1_lr_mux_enum, + ADAU1373_OUTPUT_CTRL, 4, adau1373_lr_mux_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_lineout2_lr_mux_enum, + ADAU1373_OUTPUT_CTRL, 6, adau1373_lr_mux_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_speaker_lr_mux_enum, + ADAU1373_LS_CTRL, 4, adau1373_lr_mux_text); + +static const struct snd_kcontrol_new adau1373_controls[] = { + SOC_DOUBLE_R_TLV("AIF1 Capture Volume", ADAU1373_DAI_RECL_VOL(0), + ADAU1373_DAI_RECR_VOL(0), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF2 Capture Volume", ADAU1373_DAI_RECL_VOL(1), + ADAU1373_DAI_RECR_VOL(1), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF3 Capture Volume", ADAU1373_DAI_RECL_VOL(2), + ADAU1373_DAI_RECR_VOL(2), 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ADAU1373_ADC_RECL_VOL, + ADAU1373_ADC_RECR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("DMIC Capture Volume", ADAU1373_DMIC_RECL_VOL, + ADAU1373_DMIC_RECR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("AIF1 Playback Volume", ADAU1373_DAI_PBL_VOL(0), + ADAU1373_DAI_PBR_VOL(0), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF2 Playback Volume", ADAU1373_DAI_PBL_VOL(1), + ADAU1373_DAI_PBR_VOL(1), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF3 Playback Volume", ADAU1373_DAI_PBL_VOL(2), + ADAU1373_DAI_PBR_VOL(2), 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", ADAU1373_DAC1_PBL_VOL, + ADAU1373_DAC1_PBR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", ADAU1373_DAC2_PBL_VOL, + ADAU1373_DAC2_PBR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("Lineout1 Playback Volume", ADAU1373_LLINE_OUT(0), + ADAU1373_RLINE_OUT(0), 0, 0x1f, 0, adau1373_out_tlv), + SOC_DOUBLE_R_TLV("Speaker Playback Volume", ADAU1373_LSPK_OUT, + ADAU1373_RSPK_OUT, 0, 0x1f, 0, adau1373_out_tlv), + SOC_DOUBLE_R_TLV("Headphone Playback Volume", ADAU1373_LHP_OUT, + ADAU1373_RHP_OUT, 0, 0x1f, 0, adau1373_out_tlv), + + SOC_DOUBLE_R_TLV("Input 1 Capture Volume", ADAU1373_AINL_CTRL(0), + ADAU1373_AINR_CTRL(0), 0, 0x1f, 0, adau1373_in_pga_tlv), + SOC_DOUBLE_R_TLV("Input 2 Capture Volume", ADAU1373_AINL_CTRL(1), + ADAU1373_AINR_CTRL(1), 0, 0x1f, 0, adau1373_in_pga_tlv), + SOC_DOUBLE_R_TLV("Input 3 Capture Volume", ADAU1373_AINL_CTRL(2), + ADAU1373_AINR_CTRL(2), 0, 0x1f, 0, adau1373_in_pga_tlv), + SOC_DOUBLE_R_TLV("Input 4 Capture Volume", ADAU1373_AINL_CTRL(3), + ADAU1373_AINR_CTRL(3), 0, 0x1f, 0, adau1373_in_pga_tlv), + + SOC_SINGLE_TLV("Earpiece Playback Volume", ADAU1373_EP_CTRL, 0, 3, 0, + adau1373_ep_tlv), + + SOC_DOUBLE_TLV("AIF3 Boost Playback Volume", ADAU1373_VOL_GAIN1, 4, 5, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF2 Boost Playback Volume", ADAU1373_VOL_GAIN1, 2, 3, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF1 Boost Playback Volume", ADAU1373_VOL_GAIN1, 0, 1, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF3 Boost Capture Volume", ADAU1373_VOL_GAIN2, 4, 5, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF2 Boost Capture Volume", ADAU1373_VOL_GAIN2, 2, 3, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF1 Boost Capture Volume", ADAU1373_VOL_GAIN2, 0, 1, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("DMIC Boost Capture Volume", ADAU1373_VOL_GAIN3, 6, 7, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("ADC Boost Capture Volume", ADAU1373_VOL_GAIN3, 4, 5, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("DAC2 Boost Playback Volume", ADAU1373_VOL_GAIN3, 2, 3, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("DAC1 Boost Playback Volume", ADAU1373_VOL_GAIN3, 0, 1, + 1, 0, adau1373_gain_boost_tlv), + + SOC_DOUBLE_TLV("Input 1 Boost Capture Volume", ADAU1373_ADC_GAIN, 0, 4, + 1, 0, adau1373_input_boost_tlv), + SOC_DOUBLE_TLV("Input 2 Boost Capture Volume", ADAU1373_ADC_GAIN, 1, 5, + 1, 0, adau1373_input_boost_tlv), + SOC_DOUBLE_TLV("Input 3 Boost Capture Volume", ADAU1373_ADC_GAIN, 2, 6, + 1, 0, adau1373_input_boost_tlv), + SOC_DOUBLE_TLV("Input 4 Boost Capture Volume", ADAU1373_ADC_GAIN, 3, 7, + 1, 0, adau1373_input_boost_tlv), + + SOC_DOUBLE_TLV("Speaker Boost Playback Volume", ADAU1373_LS_CTRL, 2, 3, + 1, 0, adau1373_speaker_boost_tlv), + + SOC_ENUM("Lineout1 LR Mux", adau1373_lineout1_lr_mux_enum), + SOC_ENUM("Speaker LR Mux", adau1373_speaker_lr_mux_enum), + + SOC_ENUM("HPF Cutoff", adau1373_hpf_cutoff_enum), + SOC_DOUBLE("HPF Switch", ADAU1373_HPF_CTRL, 1, 0, 1, 0), + SOC_ENUM("HPF Channel", adau1373_hpf_channel_enum), + + SOC_ENUM("Bass HPF Cutoff", adau1373_bass_hpf_cutoff_enum), + SOC_VALUE_ENUM("Bass Clip Level Threshold", + adau1373_bass_clip_level_enum), + SOC_ENUM("Bass LPF Cutoff", adau1373_bass_lpf_cutoff_enum), + SOC_DOUBLE("Bass Playback Switch", ADAU1373_BASS2, 0, 1, 1, 0), + SOC_SINGLE_TLV("Bass Playback Volume", ADAU1373_BASS2, 2, 7, 0, + adau1373_bass_tlv), + SOC_ENUM("Bass Channel", adau1373_bass_channel_enum), + + SOC_ENUM("3D Freq", adau1373_3d_cutoff_enum), + SOC_ENUM("3D Level", adau1373_3d_level_enum), + SOC_SINGLE("3D Playback Switch", ADAU1373_3D_CTRL2, 0, 1, 0), + SOC_SINGLE_TLV("3D Playback Volume", ADAU1373_3D_CTRL2, 2, 7, 0, + adau1373_3d_tlv), + SOC_ENUM("3D Channel", adau1373_bass_channel_enum), + + SOC_SINGLE("Zero Cross Switch", ADAU1373_PWDN_CTRL3, 7, 1, 0), +}; + +static const struct snd_kcontrol_new adau1373_lineout2_controls[] = { + SOC_DOUBLE_R_TLV("Lineout2 Playback Volume", ADAU1373_LLINE_OUT(1), + ADAU1373_RLINE_OUT(1), 0, 0x1f, 0, adau1373_out_tlv), + SOC_ENUM("Lineout2 LR Mux", adau1373_lineout2_lr_mux_enum), +}; + +static const struct snd_kcontrol_new adau1373_drc_controls[] = { + SOC_ENUM("DRC1 Channel", adau1373_drc1_channel_enum), + SOC_ENUM("DRC2 Channel", adau1373_drc2_channel_enum), + SOC_ENUM("DRC3 Channel", adau1373_drc3_channel_enum), +}; + +static int adau1373_pll_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int pll_id = w->name[3] - '1'; + unsigned int val; + + if (SND_SOC_DAPM_EVENT_ON(event)) + val = ADAU1373_PLL_CTRL6_PLL_EN; + else + val = 0; + + snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + ADAU1373_PLL_CTRL6_PLL_EN, val); + + if (SND_SOC_DAPM_EVENT_ON(event)) + mdelay(5); + + return 0; +} + +static const char *adau1373_decimator_text[] = { + "ADC", + "DMIC1", +}; + +static const struct soc_enum adau1373_decimator_enum = + SOC_ENUM_SINGLE(0, 0, 2, adau1373_decimator_text); + +static const struct snd_kcontrol_new adau1373_decimator_mux = + SOC_DAPM_ENUM_VIRT("Decimator Mux", adau1373_decimator_enum); + +static const struct snd_kcontrol_new adau1373_left_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC1 Switch", ADAU1373_LADC_MIXER, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Switch", ADAU1373_LADC_MIXER, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Switch", ADAU1373_LADC_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Switch", ADAU1373_LADC_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Switch", ADAU1373_LADC_MIXER, 0, 1, 0), +}; + +static const struct snd_kcontrol_new adau1373_right_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC1 Switch", ADAU1373_RADC_MIXER, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Switch", ADAU1373_RADC_MIXER, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Switch", ADAU1373_RADC_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Switch", ADAU1373_RADC_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Switch", ADAU1373_RADC_MIXER, 0, 1, 0), +}; + +#define DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(_name, _reg) \ +const struct snd_kcontrol_new _name[] = { \ + SOC_DAPM_SINGLE("Left DAC2 Switch", _reg, 7, 1, 0), \ + SOC_DAPM_SINGLE("Right DAC2 Switch", _reg, 6, 1, 0), \ + SOC_DAPM_SINGLE("Left DAC1 Switch", _reg, 5, 1, 0), \ + SOC_DAPM_SINGLE("Right DAC1 Switch", _reg, 4, 1, 0), \ + SOC_DAPM_SINGLE("Input 4 Bypass Switch", _reg, 3, 1, 0), \ + SOC_DAPM_SINGLE("Input 3 Bypass Switch", _reg, 2, 1, 0), \ + SOC_DAPM_SINGLE("Input 2 Bypass Switch", _reg, 1, 1, 0), \ + SOC_DAPM_SINGLE("Input 1 Bypass Switch", _reg, 0, 1, 0), \ +} + +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_line1_mixer_controls, + ADAU1373_LLINE1_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_line1_mixer_controls, + ADAU1373_RLINE1_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_line2_mixer_controls, + ADAU1373_LLINE2_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_line2_mixer_controls, + ADAU1373_RLINE2_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_spk_mixer_controls, + ADAU1373_LSPK_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_spk_mixer_controls, + ADAU1373_RSPK_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_ep_mixer_controls, + ADAU1373_EP_MIX); + +static const struct snd_kcontrol_new adau1373_left_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", ADAU1373_LHP_MIX, 5, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", ADAU1373_LHP_MIX, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Bypass Switch", ADAU1373_LHP_MIX, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Bypass Switch", ADAU1373_LHP_MIX, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Bypass Switch", ADAU1373_LHP_MIX, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Bypass Switch", ADAU1373_LHP_MIX, 0, 1, 0), +}; + +static const struct snd_kcontrol_new adau1373_right_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Right DAC1 Switch", ADAU1373_RHP_MIX, 5, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", ADAU1373_RHP_MIX, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Bypass Switch", ADAU1373_RHP_MIX, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Bypass Switch", ADAU1373_RHP_MIX, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Bypass Switch", ADAU1373_RHP_MIX, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Bypass Switch", ADAU1373_RHP_MIX, 0, 1, 0), +}; + +#define DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(_name, _reg) \ +const struct snd_kcontrol_new _name[] = { \ + SOC_DAPM_SINGLE("DMIC2 Swapped Switch", _reg, 6, 1, 0), \ + SOC_DAPM_SINGLE("DMIC2 Switch", _reg, 5, 1, 0), \ + SOC_DAPM_SINGLE("ADC/DMIC1 Swapped Switch", _reg, 4, 1, 0), \ + SOC_DAPM_SINGLE("ADC/DMIC1 Switch", _reg, 3, 1, 0), \ + SOC_DAPM_SINGLE("AIF3 Switch", _reg, 2, 1, 0), \ + SOC_DAPM_SINGLE("AIF2 Switch", _reg, 1, 1, 0), \ + SOC_DAPM_SINGLE("AIF1 Switch", _reg, 0, 1, 0), \ +} + +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel1_mixer_controls, + ADAU1373_DIN_MIX_CTRL(0)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel2_mixer_controls, + ADAU1373_DIN_MIX_CTRL(1)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel3_mixer_controls, + ADAU1373_DIN_MIX_CTRL(2)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel4_mixer_controls, + ADAU1373_DIN_MIX_CTRL(3)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel5_mixer_controls, + ADAU1373_DIN_MIX_CTRL(4)); + +#define DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(_name, _reg) \ +const struct snd_kcontrol_new _name[] = { \ + SOC_DAPM_SINGLE("DSP Channel5 Switch", _reg, 4, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel4 Switch", _reg, 3, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel3 Switch", _reg, 2, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel2 Switch", _reg, 1, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel1 Switch", _reg, 0, 1, 0), \ +} + +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif1_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(0)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif2_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(1)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif3_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(2)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_dac1_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(3)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_dac2_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(4)); + +static const struct snd_soc_dapm_widget adau1373_dapm_widgets[] = { + /* Datasheet claims Left ADC is bit 6 and Right ADC is bit 7, but that + * doesn't seem to be the case. */ + SND_SOC_DAPM_ADC("Left ADC", NULL, ADAU1373_PWDN_CTRL1, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", NULL, ADAU1373_PWDN_CTRL1, 6, 0), + + SND_SOC_DAPM_ADC("DMIC1", NULL, ADAU1373_DIGMICCTRL, 0, 0), + SND_SOC_DAPM_ADC("DMIC2", NULL, ADAU1373_DIGMICCTRL, 2, 0), + + SND_SOC_DAPM_VIRT_MUX("Decimator Mux", SND_SOC_NOPM, 0, 0, + &adau1373_decimator_mux), + + SND_SOC_DAPM_SUPPLY("MICBIAS2", ADAU1373_PWDN_CTRL1, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", ADAU1373_PWDN_CTRL1, 4, 0, NULL, 0), + + SND_SOC_DAPM_PGA("IN4PGA", ADAU1373_PWDN_CTRL1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN3PGA", ADAU1373_PWDN_CTRL1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN2PGA", ADAU1373_PWDN_CTRL1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN1PGA", ADAU1373_PWDN_CTRL1, 0, 0, NULL, 0), + + SND_SOC_DAPM_DAC("Left DAC2", NULL, ADAU1373_PWDN_CTRL2, 7, 0), + SND_SOC_DAPM_DAC("Right DAC2", NULL, ADAU1373_PWDN_CTRL2, 6, 0), + SND_SOC_DAPM_DAC("Left DAC1", NULL, ADAU1373_PWDN_CTRL2, 5, 0), + SND_SOC_DAPM_DAC("Right DAC1", NULL, ADAU1373_PWDN_CTRL2, 4, 0), + + SOC_MIXER_ARRAY("Left ADC Mixer", SND_SOC_NOPM, 0, 0, + adau1373_left_adc_mixer_controls), + SOC_MIXER_ARRAY("Right ADC Mixer", SND_SOC_NOPM, 0, 0, + adau1373_right_adc_mixer_controls), + + SOC_MIXER_ARRAY("Left Lineout2 Mixer", ADAU1373_PWDN_CTRL2, 3, 0, + adau1373_left_line2_mixer_controls), + SOC_MIXER_ARRAY("Right Lineout2 Mixer", ADAU1373_PWDN_CTRL2, 2, 0, + adau1373_right_line2_mixer_controls), + SOC_MIXER_ARRAY("Left Lineout1 Mixer", ADAU1373_PWDN_CTRL2, 1, 0, + adau1373_left_line1_mixer_controls), + SOC_MIXER_ARRAY("Right Lineout1 Mixer", ADAU1373_PWDN_CTRL2, 0, 0, + adau1373_right_line1_mixer_controls), + + SOC_MIXER_ARRAY("Earpiece Mixer", ADAU1373_PWDN_CTRL3, 4, 0, + adau1373_ep_mixer_controls), + SOC_MIXER_ARRAY("Left Speaker Mixer", ADAU1373_PWDN_CTRL3, 3, 0, + adau1373_left_spk_mixer_controls), + SOC_MIXER_ARRAY("Right Speaker Mixer", ADAU1373_PWDN_CTRL3, 2, 0, + adau1373_right_spk_mixer_controls), + SOC_MIXER_ARRAY("Left Headphone Mixer", SND_SOC_NOPM, 0, 0, + adau1373_left_hp_mixer_controls), + SOC_MIXER_ARRAY("Right Headphone Mixer", SND_SOC_NOPM, 0, 0, + adau1373_right_hp_mixer_controls), + SND_SOC_DAPM_SUPPLY("Headphone Enable", ADAU1373_PWDN_CTRL3, 1, 0, + NULL, 0), + + SND_SOC_DAPM_SUPPLY("AIF1 CLK", ADAU1373_SRC_DAI_CTRL(0), 0, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF2 CLK", ADAU1373_SRC_DAI_CTRL(1), 0, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF3 CLK", ADAU1373_SRC_DAI_CTRL(2), 0, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF1 IN SRC", ADAU1373_SRC_DAI_CTRL(0), 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF1 OUT SRC", ADAU1373_SRC_DAI_CTRL(0), 1, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF2 IN SRC", ADAU1373_SRC_DAI_CTRL(1), 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF2 OUT SRC", ADAU1373_SRC_DAI_CTRL(1), 1, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF3 IN SRC", ADAU1373_SRC_DAI_CTRL(2), 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF3 OUT SRC", ADAU1373_SRC_DAI_CTRL(2), 1, 0, + NULL, 0), + + SND_SOC_DAPM_AIF_IN("AIF1 IN", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1 OUT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2 IN", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2 OUT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3 IN", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3 OUT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), + + SOC_MIXER_ARRAY("DSP Channel1 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel1_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel2 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel2_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel3 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel3_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel4 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel4_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel5 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel5_mixer_controls), + + SOC_MIXER_ARRAY("AIF1 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_aif1_mixer_controls), + SOC_MIXER_ARRAY("AIF2 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_aif2_mixer_controls), + SOC_MIXER_ARRAY("AIF3 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_aif3_mixer_controls), + SOC_MIXER_ARRAY("DAC1 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dac1_mixer_controls), + SOC_MIXER_ARRAY("DAC2 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dac2_mixer_controls), + + SND_SOC_DAPM_SUPPLY("DSP", ADAU1373_DIGEN, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Recording Engine B", ADAU1373_DIGEN, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Recording Engine A", ADAU1373_DIGEN, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Playback Engine B", ADAU1373_DIGEN, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Playback Engine A", ADAU1373_DIGEN, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("PLL1", SND_SOC_NOPM, 0, 0, adau1373_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("PLL2", SND_SOC_NOPM, 0, 0, adau1373_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("SYSCLK1", ADAU1373_CLK_SRC_DIV(0), 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SYSCLK2", ADAU1373_CLK_SRC_DIV(1), 7, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + SND_SOC_DAPM_INPUT("AIN3L"), + SND_SOC_DAPM_INPUT("AIN3R"), + SND_SOC_DAPM_INPUT("AIN4L"), + SND_SOC_DAPM_INPUT("AIN4R"), + + SND_SOC_DAPM_INPUT("DMIC1DAT"), + SND_SOC_DAPM_INPUT("DMIC2DAT"), + + SND_SOC_DAPM_OUTPUT("LOUT1L"), + SND_SOC_DAPM_OUTPUT("LOUT1R"), + SND_SOC_DAPM_OUTPUT("LOUT2L"), + SND_SOC_DAPM_OUTPUT("LOUT2R"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPKL"), + SND_SOC_DAPM_OUTPUT("SPKR"), + SND_SOC_DAPM_OUTPUT("EP"), +}; + +static int adau1373_check_aif_clk(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + unsigned int dai; + const char *clk; + + dai = sink->name[3] - '1'; + + if (!adau1373->dais[dai].master) + return 0; + + if (adau1373->dais[dai].clk_src == ADAU1373_CLK_SRC_PLL1) + clk = "SYSCLK1"; + else + clk = "SYSCLK2"; + + return strcmp(source->name, clk) == 0; +} + +static int adau1373_check_src(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + unsigned int dai; + + dai = sink->name[3] - '1'; + + return adau1373->dais[dai].enable_src; +} + +#define DSP_CHANNEL_MIXER_ROUTES(_sink) \ + { _sink, "DMIC2 Swapped Switch", "DMIC2" }, \ + { _sink, "DMIC2 Switch", "DMIC2" }, \ + { _sink, "ADC/DMIC1 Swapped Switch", "Decimator Mux" }, \ + { _sink, "ADC/DMIC1 Switch", "Decimator Mux" }, \ + { _sink, "AIF1 Switch", "AIF1 IN" }, \ + { _sink, "AIF2 Switch", "AIF2 IN" }, \ + { _sink, "AIF3 Switch", "AIF3 IN" } + +#define DSP_OUTPUT_MIXER_ROUTES(_sink) \ + { _sink, "DSP Channel1 Switch", "DSP Channel1 Mixer" }, \ + { _sink, "DSP Channel2 Switch", "DSP Channel2 Mixer" }, \ + { _sink, "DSP Channel3 Switch", "DSP Channel3 Mixer" }, \ + { _sink, "DSP Channel4 Switch", "DSP Channel4 Mixer" }, \ + { _sink, "DSP Channel5 Switch", "DSP Channel5 Mixer" } + +#define LEFT_OUTPUT_MIXER_ROUTES(_sink) \ + { _sink, "Right DAC2 Switch", "Right DAC2" }, \ + { _sink, "Left DAC2 Switch", "Left DAC2" }, \ + { _sink, "Right DAC1 Switch", "Right DAC1" }, \ + { _sink, "Left DAC1 Switch", "Left DAC1" }, \ + { _sink, "Input 1 Bypass Switch", "IN1PGA" }, \ + { _sink, "Input 2 Bypass Switch", "IN2PGA" }, \ + { _sink, "Input 3 Bypass Switch", "IN3PGA" }, \ + { _sink, "Input 4 Bypass Switch", "IN4PGA" } + +#define RIGHT_OUTPUT_MIXER_ROUTES(_sink) \ + { _sink, "Right DAC2 Switch", "Right DAC2" }, \ + { _sink, "Left DAC2 Switch", "Left DAC2" }, \ + { _sink, "Right DAC1 Switch", "Right DAC1" }, \ + { _sink, "Left DAC1 Switch", "Left DAC1" }, \ + { _sink, "Input 1 Bypass Switch", "IN1PGA" }, \ + { _sink, "Input 2 Bypass Switch", "IN2PGA" }, \ + { _sink, "Input 3 Bypass Switch", "IN3PGA" }, \ + { _sink, "Input 4 Bypass Switch", "IN4PGA" } + +static const struct snd_soc_dapm_route adau1373_dapm_routes[] = { + { "Left ADC Mixer", "DAC1 Switch", "Left DAC1" }, + { "Left ADC Mixer", "Input 1 Switch", "IN1PGA" }, + { "Left ADC Mixer", "Input 2 Switch", "IN2PGA" }, + { "Left ADC Mixer", "Input 3 Switch", "IN3PGA" }, + { "Left ADC Mixer", "Input 4 Switch", "IN4PGA" }, + + { "Right ADC Mixer", "DAC1 Switch", "Right DAC1" }, + { "Right ADC Mixer", "Input 1 Switch", "IN1PGA" }, + { "Right ADC Mixer", "Input 2 Switch", "IN2PGA" }, + { "Right ADC Mixer", "Input 3 Switch", "IN3PGA" }, + { "Right ADC Mixer", "Input 4 Switch", "IN4PGA" }, + + { "Left ADC", NULL, "Left ADC Mixer" }, + { "Right ADC", NULL, "Right ADC Mixer" }, + + { "Decimator Mux", "ADC", "Left ADC" }, + { "Decimator Mux", "ADC", "Right ADC" }, + { "Decimator Mux", "DMIC1", "DMIC1" }, + + DSP_CHANNEL_MIXER_ROUTES("DSP Channel1 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel2 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel3 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel4 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel5 Mixer"), + + DSP_OUTPUT_MIXER_ROUTES("AIF1 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("AIF2 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("AIF3 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("DAC1 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("DAC2 Mixer"), + + { "AIF1 OUT", NULL, "AIF1 Mixer" }, + { "AIF2 OUT", NULL, "AIF2 Mixer" }, + { "AIF3 OUT", NULL, "AIF3 Mixer" }, + { "Left DAC1", NULL, "DAC1 Mixer" }, + { "Right DAC1", NULL, "DAC1 Mixer" }, + { "Left DAC2", NULL, "DAC2 Mixer" }, + { "Right DAC2", NULL, "DAC2 Mixer" }, + + LEFT_OUTPUT_MIXER_ROUTES("Left Lineout1 Mixer"), + RIGHT_OUTPUT_MIXER_ROUTES("Right Lineout1 Mixer"), + LEFT_OUTPUT_MIXER_ROUTES("Left Lineout2 Mixer"), + RIGHT_OUTPUT_MIXER_ROUTES("Right Lineout2 Mixer"), + LEFT_OUTPUT_MIXER_ROUTES("Left Speaker Mixer"), + RIGHT_OUTPUT_MIXER_ROUTES("Right Speaker Mixer"), + + { "Left Headphone Mixer", "Left DAC2 Switch", "Left DAC2" }, + { "Left Headphone Mixer", "Left DAC1 Switch", "Left DAC1" }, + { "Left Headphone Mixer", "Input 1 Bypass Switch", "IN1PGA" }, + { "Left Headphone Mixer", "Input 2 Bypass Switch", "IN2PGA" }, + { "Left Headphone Mixer", "Input 3 Bypass Switch", "IN3PGA" }, + { "Left Headphone Mixer", "Input 4 Bypass Switch", "IN4PGA" }, + { "Right Headphone Mixer", "Right DAC2 Switch", "Right DAC2" }, + { "Right Headphone Mixer", "Right DAC1 Switch", "Right DAC1" }, + { "Right Headphone Mixer", "Input 1 Bypass Switch", "IN1PGA" }, + { "Right Headphone Mixer", "Input 2 Bypass Switch", "IN2PGA" }, + { "Right Headphone Mixer", "Input 3 Bypass Switch", "IN3PGA" }, + { "Right Headphone Mixer", "Input 4 Bypass Switch", "IN4PGA" }, + + { "Left Headphone Mixer", NULL, "Headphone Enable" }, + { "Right Headphone Mixer", NULL, "Headphone Enable" }, + + { "Earpiece Mixer", "Right DAC2 Switch", "Right DAC2" }, + { "Earpiece Mixer", "Left DAC2 Switch", "Left DAC2" }, + { "Earpiece Mixer", "Right DAC1 Switch", "Right DAC1" }, + { "Earpiece Mixer", "Left DAC1 Switch", "Left DAC1" }, + { "Earpiece Mixer", "Input 1 Bypass Switch", "IN1PGA" }, + { "Earpiece Mixer", "Input 2 Bypass Switch", "IN2PGA" }, + { "Earpiece Mixer", "Input 3 Bypass Switch", "IN3PGA" }, + { "Earpiece Mixer", "Input 4 Bypass Switch", "IN4PGA" }, + + { "LOUT1L", NULL, "Left Lineout1 Mixer" }, + { "LOUT1R", NULL, "Right Lineout1 Mixer" }, + { "LOUT2L", NULL, "Left Lineout2 Mixer" }, + { "LOUT2R", NULL, "Right Lineout2 Mixer" }, + { "SPKL", NULL, "Left Speaker Mixer" }, + { "SPKR", NULL, "Right Speaker Mixer" }, + { "HPL", NULL, "Left Headphone Mixer" }, + { "HPR", NULL, "Right Headphone Mixer" }, + { "EP", NULL, "Earpiece Mixer" }, + + { "IN1PGA", NULL, "AIN1L" }, + { "IN2PGA", NULL, "AIN2L" }, + { "IN3PGA", NULL, "AIN3L" }, + { "IN4PGA", NULL, "AIN4L" }, + { "IN1PGA", NULL, "AIN1R" }, + { "IN2PGA", NULL, "AIN2R" }, + { "IN3PGA", NULL, "AIN3R" }, + { "IN4PGA", NULL, "AIN4R" }, + + { "SYSCLK1", NULL, "PLL1" }, + { "SYSCLK2", NULL, "PLL2" }, + + { "Left DAC1", NULL, "SYSCLK1" }, + { "Right DAC1", NULL, "SYSCLK1" }, + { "Left DAC2", NULL, "SYSCLK1" }, + { "Right DAC2", NULL, "SYSCLK1" }, + { "Left ADC", NULL, "SYSCLK1" }, + { "Right ADC", NULL, "SYSCLK1" }, + + { "DSP", NULL, "SYSCLK1" }, + + { "AIF1 Mixer", NULL, "DSP" }, + { "AIF2 Mixer", NULL, "DSP" }, + { "AIF3 Mixer", NULL, "DSP" }, + { "DAC1 Mixer", NULL, "DSP" }, + { "DAC2 Mixer", NULL, "DSP" }, + { "DAC1 Mixer", NULL, "Playback Engine A" }, + { "DAC2 Mixer", NULL, "Playback Engine B" }, + { "Left ADC Mixer", NULL, "Recording Engine A" }, + { "Right ADC Mixer", NULL, "Recording Engine A" }, + + { "AIF1 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk }, + { "AIF2 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk }, + { "AIF3 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk }, + { "AIF1 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk }, + { "AIF2 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk }, + { "AIF3 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk }, + + { "AIF1 IN", NULL, "AIF1 CLK" }, + { "AIF1 OUT", NULL, "AIF1 CLK" }, + { "AIF2 IN", NULL, "AIF2 CLK" }, + { "AIF2 OUT", NULL, "AIF2 CLK" }, + { "AIF3 IN", NULL, "AIF3 CLK" }, + { "AIF3 OUT", NULL, "AIF3 CLK" }, + { "AIF1 IN", NULL, "AIF1 IN SRC", adau1373_check_src }, + { "AIF1 OUT", NULL, "AIF1 OUT SRC", adau1373_check_src }, + { "AIF2 IN", NULL, "AIF2 IN SRC", adau1373_check_src }, + { "AIF2 OUT", NULL, "AIF2 OUT SRC", adau1373_check_src }, + { "AIF3 IN", NULL, "AIF3 IN SRC", adau1373_check_src }, + { "AIF3 OUT", NULL, "AIF3 OUT SRC", adau1373_check_src }, + + { "DMIC1", NULL, "DMIC1DAT" }, + { "DMIC1", NULL, "SYSCLK1" }, + { "DMIC1", NULL, "Recording Engine A" }, + { "DMIC2", NULL, "DMIC2DAT" }, + { "DMIC2", NULL, "SYSCLK1" }, + { "DMIC2", NULL, "Recording Engine B" }, +}; + +static int adau1373_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id]; + unsigned int div; + unsigned int freq; + unsigned int ctrl; + + freq = adau1373_dai->sysclk; + + if (freq % params_rate(params) != 0) + return -EINVAL; + + switch (freq / params_rate(params)) { + case 1024: /* sysclk / 256 */ + div = 0; + break; + case 1536: /* 2/3 sysclk / 256 */ + div = 1; + break; + case 2048: /* 1/2 sysclk / 256 */ + div = 2; + break; + case 3072: /* 1/3 sysclk / 256 */ + div = 3; + break; + case 4096: /* 1/4 sysclk / 256 */ + div = 4; + break; + case 6144: /* 1/6 sysclk / 256 */ + div = 5; + break; + case 5632: /* 2/11 sysclk / 256 */ + div = 6; + break; + default: + return -EINVAL; + } + + adau1373_dai->enable_src = (div != 0); + + snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), + ~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + ctrl = ADAU1373_DAI_WLEN_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + ctrl = ADAU1373_DAI_WLEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ctrl = ADAU1373_DAI_WLEN_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + ctrl = ADAU1373_DAI_WLEN_32; + break; + default: + return -EINVAL; + } + + return snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + ADAU1373_DAI_WLEN_MASK, ctrl); +} + +static int adau1373_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id]; + unsigned int ctrl; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl = ADAU1373_DAI_MASTER; + adau1373_dai->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl = 0; + adau1373_dai->master = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl |= ADAU1373_DAI_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl |= ADAU1373_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl |= ADAU1373_DAI_FORMAT_RIGHT_J; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl |= ADAU1373_DAI_FORMAT_DSP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl |= ADAU1373_DAI_INVERT_BCLK; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl |= ADAU1373_DAI_INVERT_LRCLK; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl |= ADAU1373_DAI_INVERT_LRCLK | ADAU1373_DAI_INVERT_BCLK; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + ~ADAU1373_DAI_WLEN_MASK, ctrl); + + return 0; +} + +static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(dai->codec); + struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id]; + + switch (clk_id) { + case ADAU1373_CLK_SRC_PLL1: + case ADAU1373_CLK_SRC_PLL2: + break; + default: + return -EINVAL; + } + + adau1373_dai->sysclk = freq; + adau1373_dai->clk_src = clk_id; + + snd_soc_update_bits(dai->codec, ADAU1373_BCLKDIV(dai->id), + ADAU1373_BCLKDIV_SOURCE, clk_id << 5); + + return 0; +} + +static const struct snd_soc_dai_ops adau1373_dai_ops = { + .hw_params = adau1373_hw_params, + .set_sysclk = adau1373_set_dai_sysclk, + .set_fmt = adau1373_set_dai_fmt, +}; + +#define ADAU1373_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver adau1373_dai_driver[] = { + { + .id = 0, + .name = "adau1373-aif1", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .ops = &adau1373_dai_ops, + .symmetric_rates = 1, + }, + { + .id = 1, + .name = "adau1373-aif2", + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .ops = &adau1373_dai_ops, + .symmetric_rates = 1, + }, + { + .id = 2, + .name = "adau1373-aif3", + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .ops = &adau1373_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + unsigned int dpll_div = 0; + unsigned int x, r, n, m, i, j, mode; + + switch (pll_id) { + case ADAU1373_PLL1: + case ADAU1373_PLL2: + break; + default: + return -EINVAL; + } + + switch (source) { + case ADAU1373_PLL_SRC_BCLK1: + case ADAU1373_PLL_SRC_BCLK2: + case ADAU1373_PLL_SRC_BCLK3: + case ADAU1373_PLL_SRC_LRCLK1: + case ADAU1373_PLL_SRC_LRCLK2: + case ADAU1373_PLL_SRC_LRCLK3: + case ADAU1373_PLL_SRC_MCLK1: + case ADAU1373_PLL_SRC_MCLK2: + case ADAU1373_PLL_SRC_GPIO1: + case ADAU1373_PLL_SRC_GPIO2: + case ADAU1373_PLL_SRC_GPIO3: + case ADAU1373_PLL_SRC_GPIO4: + break; + default: + return -EINVAL; + } + + if (freq_in < 7813 || freq_in > 27000000) + return -EINVAL; + + if (freq_out < 45158000 || freq_out > 49152000) + return -EINVAL; + + /* APLL input needs to be >= 8Mhz, so in case freq_in is less we use the + * DPLL to get it there. DPLL_out = (DPLL_in / div) * 1024 */ + while (freq_in < 8000000) { + freq_in *= 2; + dpll_div++; + } + + if (freq_out % freq_in != 0) { + /* fout = fin * (r + (n/m)) / x */ + x = DIV_ROUND_UP(freq_in, 13500000); + freq_in /= x; + r = freq_out / freq_in; + i = freq_out % freq_in; + j = gcd(i, freq_in); + n = i / j; + m = freq_in / j; + x--; + mode = 1; + } else { + /* fout = fin / r */ + r = freq_out / freq_in; + n = 0; + m = 0; + x = 0; + mode = 0; + } + + if (r < 2 || r > 8 || x > 3 || m > 0xffff || n > 0xffff) + return -EINVAL; + + if (dpll_div) { + dpll_div = 11 - dpll_div; + snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + ADAU1373_PLL_CTRL6_DPLL_BYPASS, 0); + } else { + snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + ADAU1373_PLL_CTRL6_DPLL_BYPASS, + ADAU1373_PLL_CTRL6_DPLL_BYPASS); + } + + snd_soc_write(codec, ADAU1373_DPLL_CTRL(pll_id), + (source << 4) | dpll_div); + snd_soc_write(codec, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL5(pll_id), + (r << 3) | (x << 1) | mode); + + /* Set sysclk to pll_rate / 4 */ + snd_soc_update_bits(codec, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); + + return 0; +} + +static void adau1373_load_drc_settings(struct snd_soc_codec *codec, + unsigned int nr, uint8_t *drc) +{ + unsigned int i; + + for (i = 0; i < ADAU1373_DRC_SIZE; ++i) + snd_soc_write(codec, ADAU1373_DRC(nr) + i, drc[i]); +} + +static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias) +{ + switch (micbias) { + case ADAU1373_MICBIAS_2_9V: + case ADAU1373_MICBIAS_2_2V: + case ADAU1373_MICBIAS_2_6V: + case ADAU1373_MICBIAS_1_8V: + return true; + default: + break; + } + return false; +} + +static int adau1373_probe(struct snd_soc_codec *codec) +{ + struct adau1373_platform_data *pdata = codec->dev->platform_data; + bool lineout_differential = false; + unsigned int val; + int ret; + int i; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); + return ret; + } + + codec->dapm.idle_bias_off = true; + + if (pdata) { + if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting)) + return -EINVAL; + + if (!adau1373_valid_micbias(pdata->micbias1) || + !adau1373_valid_micbias(pdata->micbias2)) + return -EINVAL; + + for (i = 0; i < pdata->num_drc; ++i) { + adau1373_load_drc_settings(codec, i, + pdata->drc_setting[i]); + } + + snd_soc_add_controls(codec, adau1373_drc_controls, + pdata->num_drc); + + val = 0; + for (i = 0; i < 4; ++i) { + if (pdata->input_differential[i]) + val |= BIT(i); + } + snd_soc_write(codec, ADAU1373_INPUT_MODE, val); + + val = 0; + if (pdata->lineout_differential) + val |= ADAU1373_OUTPUT_CTRL_LDIFF; + if (pdata->lineout_ground_sense) + val |= ADAU1373_OUTPUT_CTRL_LNFBEN; + snd_soc_write(codec, ADAU1373_OUTPUT_CTRL, val); + + lineout_differential = pdata->lineout_differential; + + snd_soc_write(codec, ADAU1373_EP_CTRL, + (pdata->micbias1 << ADAU1373_EP_CTRL_MICBIAS1_OFFSET) | + (pdata->micbias2 << ADAU1373_EP_CTRL_MICBIAS2_OFFSET)); + } + + if (!lineout_differential) { + snd_soc_add_controls(codec, adau1373_lineout2_controls, + ARRAY_SIZE(adau1373_lineout2_controls)); + } + + snd_soc_write(codec, ADAU1373_ADC_CTRL, + ADAU1373_ADC_CTRL_RESET_FORCE | ADAU1373_ADC_CTRL_PEAK_DETECT); + + return 0; +} + +static int adau1373_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + ADAU1373_PWDN_CTRL3_PWR_EN, ADAU1373_PWDN_CTRL3_PWR_EN); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + ADAU1373_PWDN_CTRL3_PWR_EN, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int adau1373_remove(struct snd_soc_codec *codec) +{ + adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int adau1373_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int adau1373_resume(struct snd_soc_codec *codec) +{ + adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + snd_soc_cache_sync(codec); + + return 0; +} + +static struct snd_soc_codec_driver adau1373_codec_driver = { + .probe = adau1373_probe, + .remove = adau1373_remove, + .suspend = adau1373_suspend, + .resume = adau1373_resume, + .set_bias_level = adau1373_set_bias_level, + .reg_cache_size = ARRAY_SIZE(adau1373_default_regs), + .reg_cache_default = adau1373_default_regs, + .reg_word_size = sizeof(uint8_t), + + .set_pll = adau1373_set_pll, + + .controls = adau1373_controls, + .num_controls = ARRAY_SIZE(adau1373_controls), + .dapm_widgets = adau1373_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau1373_dapm_widgets), + .dapm_routes = adau1373_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adau1373_dapm_routes), +}; + +static int __devinit adau1373_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct adau1373 *adau1373; + int ret; + + adau1373 = kzalloc(sizeof(*adau1373), GFP_KERNEL); + if (!adau1373) + return -ENOMEM; + + dev_set_drvdata(&client->dev, adau1373); + + ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver, + adau1373_dai_driver, ARRAY_SIZE(adau1373_dai_driver)); + if (ret < 0) + kfree(adau1373); + + return ret; +} + +static int __devexit adau1373_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(dev_get_drvdata(&client->dev)); + return 0; +} + +static const struct i2c_device_id adau1373_i2c_id[] = { + { "adau1373", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adau1373_i2c_id); + +static struct i2c_driver adau1373_i2c_driver = { + .driver = { + .name = "adau1373", + .owner = THIS_MODULE, + }, + .probe = adau1373_i2c_probe, + .remove = __devexit_p(adau1373_i2c_remove), + .id_table = adau1373_i2c_id, +}; + +static int __init adau1373_init(void) +{ + return i2c_add_driver(&adau1373_i2c_driver); +} +module_init(adau1373_init); + +static void __exit adau1373_exit(void) +{ + i2c_del_driver(&adau1373_i2c_driver); +} +module_exit(adau1373_exit); + +MODULE_DESCRIPTION("ASoC ADAU1373 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1373.h b/sound/soc/codecs/adau1373.h new file mode 100644 index 000000000000..c6ab5530760c --- /dev/null +++ b/sound/soc/codecs/adau1373.h @@ -0,0 +1,29 @@ +#ifndef __ADAU1373_H__ +#define __ADAU1373_H__ + +enum adau1373_pll_src { + ADAU1373_PLL_SRC_MCLK1 = 0, + ADAU1373_PLL_SRC_BCLK1 = 1, + ADAU1373_PLL_SRC_BCLK2 = 2, + ADAU1373_PLL_SRC_BCLK3 = 3, + ADAU1373_PLL_SRC_LRCLK1 = 4, + ADAU1373_PLL_SRC_LRCLK2 = 5, + ADAU1373_PLL_SRC_LRCLK3 = 6, + ADAU1373_PLL_SRC_GPIO1 = 7, + ADAU1373_PLL_SRC_GPIO2 = 8, + ADAU1373_PLL_SRC_GPIO3 = 9, + ADAU1373_PLL_SRC_GPIO4 = 10, + ADAU1373_PLL_SRC_MCLK2 = 11, +}; + +enum adau1373_pll { + ADAU1373_PLL1 = 0, + ADAU1373_PLL2 = 1, +}; + +enum adau1373_clk_src { + ADAU1373_CLK_SRC_PLL1 = 0, + ADAU1373_CLK_SRC_PLL2 = 1, +}; + +#endif diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 300c04b70e71..f9f08948e5e8 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -523,7 +523,8 @@ static int adav80x_hw_params(struct snd_pcm_substream *substream, } static int adav80x_set_sysclk(struct snd_soc_codec *codec, - int clk_id, unsigned int freq, int dir) + int clk_id, int source, + unsigned int freq, int dir) { struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index eecffb548947..05173159507e 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -41,7 +41,6 @@ MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); struct alc5623_priv { enum snd_soc_control_type control_type; void *control_data; - struct mutex mutex; u8 id; unsigned int sysclk; u16 reg_cache[ALC5623_VENDOR_ID2+2]; @@ -1052,7 +1051,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, alc5623); alc5623->control_data = client; alc5623->control_type = SND_SOC_I2C; - mutex_init(&alc5623->mutex); ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5623, &alc5623_dai, 1); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 7e4066e131e6..91130fbc6913 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> @@ -1436,10 +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(of, 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/sn95031.c b/sound/soc/codecs/sn95031.c index 84ffdebb8a8b..29945b004135 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -79,7 +79,7 @@ static void configure_adc(struct snd_soc_codec *sn95031_codec, int val) */ static int find_free_channel(struct snd_soc_codec *sn95031_codec) { - int ret = 0, i, value; + int i, value; /* check whether ADC is enabled */ value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); @@ -91,12 +91,10 @@ static int find_free_channel(struct snd_soc_codec *sn95031_codec) for (i = 0; i < SN95031_ADC_CHANLS_MAX; i++) { value = snd_soc_read(sn95031_codec, SN95031_ADC_CHNL_START_ADDR + i); - if (value & SN95031_STOPBIT_MASK) { - ret = i; + if (value & SN95031_STOPBIT_MASK) break; - } } - return (ret > SN95031_ADC_LOOP_MAX) ? (-EINVAL) : ret; + return (i == SN95031_ADC_CHANLS_MAX) ? (-EINVAL) : i; } /* Initialize the ADC for reading micbias values. Can sleep. */ @@ -660,7 +658,7 @@ static int sn95031_pcm_spkr_mute(struct snd_soc_dai *dai, int mute) return 0; } -int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, +static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { unsigned int format, rate; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 84f4ad568556..cceb0022f02c 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -294,7 +294,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c = codec->control_data; struct snd_pcm_runtime *master_runtime; /* The DAI has shared clocks so if we already have a playback or @@ -303,7 +302,7 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, */ if (ssm2602->master_substream) { master_runtime = ssm2602->master_substream->runtime; - dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", + dev_dbg(codec->dev, "Constraining to %d bits at %dHz\n", master_runtime->sample_bits, master_runtime->rate); diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index fbd7eb9e61ce..5c7def3979c0 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -524,13 +524,17 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); pr_debug("rate: %u\n", rate); for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) - if (interpolation_ratios[i].fs == rate) + if (interpolation_ratios[i].fs == rate) { ir = interpolation_ratios[i].ir; + break; + } if (ir < 0) return -EINVAL; for (i = 0; mclk_ratios[ir][i].ratio; i++) - if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) + if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) { mcs = mclk_ratios[ir][i].mcs; + break; + } if (mcs < 0) return -EINVAL; @@ -808,6 +812,7 @@ static int sta32x_remove(struct snd_soc_codec *codec) { struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); @@ -867,18 +872,8 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, static __devexit int sta32x_i2c_remove(struct i2c_client *client) { struct sta32x_priv *sta32x = i2c_get_clientdata(client); - struct snd_soc_codec *codec = sta32x->codec; - - if (codec) - sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); - - regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); - - if (codec) { - snd_soc_unregister_codec(&client->dev); - snd_soc_codec_set_drvdata(codec, NULL); - } + snd_soc_unregister_codec(&client->dev); kfree(sta32x); return 0; } diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 239e0c461068..b2572c451c35 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -446,7 +446,6 @@ err_regulator: gpio_free(data->power_gpio); err_gpio: kfree(data); - i2c_set_clientdata(tpa6130a2_client, NULL); tpa6130a2_client = NULL; return ret; diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 443032b3b329..81645c632447 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -118,8 +118,8 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { 0x4A, /* TWL6040_LPPLLDIV 0x09 */ 0x00, /* TWL6040_AMICBCTL 0x0A */ 0x00, /* TWL6040_DMICBCTL 0x0B */ - 0x18, /* TWL6040_MICLCTL 0x0C - No input selected on Left Mic */ - 0x18, /* TWL6040_MICRCTL 0x0D - No input selected on Right Mic */ + 0x00, /* TWL6040_MICLCTL 0x0C */ + 0x00, /* TWL6040_MICRCTL 0x0D */ 0x00, /* TWL6040_MICGAIN 0x0E */ 0x1B, /* TWL6040_LINEGAIN 0x0F */ 0x00, /* TWL6040_HSLCTL 0x10 */ @@ -155,41 +155,8 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { 0x00, /* TWL6040_STATUS (ro) 0x2E */ }; -/* - * twl6040 vio/gnd registers: - * registers under vio/gnd supply can be accessed - * before the power-up sequence, after NRESPWRON goes high - */ -static const int twl6040_vio_reg[TWL6040_VIOREGNUM] = { - TWL6040_REG_ASICID, - TWL6040_REG_ASICREV, - TWL6040_REG_INTID, - TWL6040_REG_INTMR, - TWL6040_REG_NCPCTL, - TWL6040_REG_LDOCTL, - TWL6040_REG_AMICBCTL, - TWL6040_REG_DMICBCTL, - TWL6040_REG_HKCTL1, - TWL6040_REG_HKCTL2, - TWL6040_REG_GPOCTL, - TWL6040_REG_TRIM1, - TWL6040_REG_TRIM2, - TWL6040_REG_TRIM3, - TWL6040_REG_HSOTRIM, - TWL6040_REG_HFOTRIM, - TWL6040_REG_ACCCTL, - TWL6040_REG_STATUS, -}; - -/* - * twl6040 vdd/vss registers: - * registers under vdd/vss supplies can only be accessed - * after the power-up sequence - */ -static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = { - TWL6040_REG_HPPLLCTL, - TWL6040_REG_LPPLLCTL, - TWL6040_REG_LPPLLDIV, +/* List of registers to be restored after power up */ +static const int twl6040_restore_list[] = { TWL6040_REG_MICLCTL, TWL6040_REG_MICRCTL, TWL6040_REG_MICGAIN, @@ -202,12 +169,6 @@ static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = { TWL6040_REG_HFLGAIN, TWL6040_REG_HFRCTL, TWL6040_REG_HFRGAIN, - TWL6040_REG_VIBCTLL, - TWL6040_REG_VIBDATL, - TWL6040_REG_VIBCTLR, - TWL6040_REG_VIBDATR, - TWL6040_REG_ALB, - TWL6040_REG_DLB, }; /* set of rates for each pll: low-power and high-performance */ @@ -296,56 +257,27 @@ static int twl6040_write(struct snd_soc_codec *codec, return twl6040_reg_write(twl6040, reg, value); } -static void twl6040_init_vio_regs(struct snd_soc_codec *codec) +static void twl6040_init_chip(struct snd_soc_codec *codec) { - u8 *cache = codec->reg_cache; - int reg, i; + struct twl6040 *twl6040 = codec->control_data; + u8 val; - for (i = 0; i < TWL6040_VIOREGNUM; i++) { - reg = twl6040_vio_reg[i]; - /* - * skip read-only registers (ASICID, ASICREV, STATUS) - * and registers shared among MFD children - */ - switch (reg) { - case TWL6040_REG_ASICID: - case TWL6040_REG_ASICREV: - case TWL6040_REG_INTID: - case TWL6040_REG_INTMR: - case TWL6040_REG_NCPCTL: - case TWL6040_REG_LDOCTL: - case TWL6040_REG_GPOCTL: - case TWL6040_REG_ACCCTL: - case TWL6040_REG_STATUS: - continue; - default: - break; - } - twl6040_write(codec, reg, cache[reg]); - } + val = twl6040_get_revid(twl6040); + twl6040_write_reg_cache(codec, TWL6040_REG_ASICREV, val); + + /* Change chip defaults */ + /* No imput selected for microphone amplifiers */ + twl6040_write_reg_cache(codec, TWL6040_REG_MICLCTL, 0x18); + twl6040_write_reg_cache(codec, TWL6040_REG_MICRCTL, 0x18); } -static void twl6040_init_vdd_regs(struct snd_soc_codec *codec) +static void twl6040_restore_regs(struct snd_soc_codec *codec) { u8 *cache = codec->reg_cache; int reg, i; - for (i = 0; i < TWL6040_VDDREGNUM; i++) { - reg = twl6040_vdd_reg[i]; - /* skip vibra and PLL registers */ - switch (reg) { - case TWL6040_REG_VIBCTLL: - case TWL6040_REG_VIBDATL: - case TWL6040_REG_VIBCTLR: - case TWL6040_REG_VIBDATR: - case TWL6040_REG_HPPLLCTL: - case TWL6040_REG_LPPLLCTL: - case TWL6040_REG_LPPLLDIV: - continue; - default: - break; - } - + for (i = 0; i < ARRAY_SIZE(twl6040_restore_list); i++) { + reg = twl6040_restore_list[i]; twl6040_write(codec, reg, cache[reg]); } } @@ -1325,8 +1257,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, priv->codec_powered = 1; - /* initialize vdd/vss registers with reg_cache */ - twl6040_init_vdd_regs(codec); + twl6040_restore_regs(codec); /* Set external boost GPO */ twl6040_write(codec, TWL6040_REG_GPOCTL, 0x02); @@ -1468,7 +1399,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { .playback = { .stream_name = "Playback", .channels_min = 1, - .channels_max = 2, + .channels_max = 5, .rates = TWL6040_RATES, .formats = TWL6040_FORMATS, }, @@ -1518,8 +1449,8 @@ static struct snd_soc_dai_driver twl6040_dai[] = { .name = "twl6040-vib", .playback = { .stream_name = "Vibra Playback", - .channels_min = 2, - .channels_max = 2, + .channels_min = 1, + .channels_max = 1, .rates = SNDRV_PCM_RATE_CONTINUOUS, .formats = TWL6040_FORMATS, }, @@ -1620,8 +1551,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) goto plugirq_err; } - /* init vio registers */ - twl6040_init_vio_regs(codec); + twl6040_init_chip(codec); /* power on device */ ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index bcc208967917..4523c4cec02b 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 id, board, rev; + + board = i2c_smbus_read_byte_data(i2c, 0); + if (board < 0) { + dev_err(&i2c->dev, "Failed to read ID: %d\n", board); + return board; + } + + 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 + 1); + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, &wm1250_ev1_dai, 1); } diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index db0dced74843..55a4c830e111 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -20,6 +20,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -598,6 +599,11 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8510 = { .reg_cache_default =wm8510_reg, }; +static const struct of_device_id wm8510_of_match[] = { + { .compatible = "wlf,wm8510" }, + { }, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8510_spi_probe(struct spi_device *spi) { @@ -628,6 +634,7 @@ static struct spi_driver wm8510_spi_driver = { .driver = { .name = "wm8510", .owner = THIS_MODULE, + .of_match_table = wm8510_of_match, }, .probe = wm8510_spi_probe, .remove = __devexit_p(wm8510_spi_remove), @@ -671,6 +678,7 @@ static struct i2c_driver wm8510_i2c_driver = { .driver = { .name = "wm8510-codec", .owner = THIS_MODULE, + .of_match_table = wm8510_of_match, }, .probe = wm8510_i2c_probe, .remove = __devexit_p(wm8510_i2c_remove), diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 4fd4d8dca0fc..5355a7a944f7 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -20,6 +20,7 @@ #include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -84,7 +85,7 @@ static const char *wm8523_zd_count_text[] = { static const struct soc_enum wm8523_zc_count = SOC_ENUM_SINGLE(WM8523_ZERO_DETECT, 0, 2, wm8523_zd_count_text); -static const struct snd_kcontrol_new wm8523_snd_controls[] = { +static const struct snd_kcontrol_new wm8523_controls[] = { SOC_DOUBLE_R_TLV("Playback Volume", WM8523_DAC_GAINL, WM8523_DAC_GAINR, 0, 448, 0, dac_tlv), SOC_SINGLE("ZC Switch", WM8523_DAC_CTRL3, 4, 1, 0), @@ -101,22 +102,11 @@ SND_SOC_DAPM_OUTPUT("LINEVOUTL"), SND_SOC_DAPM_OUTPUT("LINEVOUTR"), }; -static const struct snd_soc_dapm_route intercon[] = { +static const struct snd_soc_dapm_route wm8523_dapm_routes[] = { { "LINEVOUTL", NULL, "DAC" }, { "LINEVOUTR", NULL, "DAC" }, }; -static int wm8523_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8523_dapm_widgets, - ARRAY_SIZE(wm8523_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - static struct { int value; int ratio; @@ -479,10 +469,6 @@ static int wm8523_probe(struct snd_soc_codec *codec) /* Bias level configuration will have done an extra enable */ regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); - snd_soc_add_controls(codec, wm8523_snd_controls, - ARRAY_SIZE(wm8523_snd_controls)); - wm8523_add_widgets(codec); - return 0; err_enable: @@ -512,6 +498,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8523 = { .reg_word_size = sizeof(u16), .reg_cache_default = wm8523_reg, .volatile_register = wm8523_volatile_register, + + .controls = wm8523_controls, + .num_controls = ARRAY_SIZE(wm8523_controls), + .dapm_widgets = wm8523_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8523_dapm_widgets), + .dapm_routes = wm8523_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8523_dapm_routes), +}; + +static const struct of_device_id wm8523_of_match[] = { + { .compatible = "wlf,wm8523" }, + { }, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -551,8 +549,9 @@ MODULE_DEVICE_TABLE(i2c, wm8523_i2c_id); static struct i2c_driver wm8523_i2c_driver = { .driver = { - .name = "wm8523-codec", + .name = "wm8523", .owner = THIS_MODULE, + .of_match_table = wm8523_of_match, }, .probe = wm8523_i2c_probe, .remove = __devexit_p(wm8523_i2c_remove), diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 4bbc0a79f01e..4664c3a76c78 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -26,6 +26,7 @@ #include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> @@ -907,6 +908,11 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8580 = { .reg_cache_default = wm8580_reg, }; +static const struct of_device_id wm8580_of_match[] = { + { .compatible = "wlf,wm8580" }, + { }, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -943,8 +949,9 @@ MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id); static struct i2c_driver wm8580_i2c_driver = { .driver = { - .name = "wm8580-codec", + .name = "wm8580", .owner = THIS_MODULE, + .of_match_table = wm8580_of_match, }, .probe = wm8580_i2c_probe, .remove = wm8580_i2c_remove, diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index a537e4af6ae7..8457d3cb5962 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -21,6 +21,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -414,6 +415,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8711 = { .num_dapm_routes = ARRAY_SIZE(wm8711_intercon), }; +static const struct of_device_id wm8711_of_match[] = { + { .compatible = "wlf,wm8711", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8711_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8711_spi_probe(struct spi_device *spi) { @@ -443,8 +450,9 @@ static int __devexit wm8711_spi_remove(struct spi_device *spi) static struct spi_driver wm8711_spi_driver = { .driver = { - .name = "wm8711-codec", + .name = "wm8711", .owner = THIS_MODULE, + .of_match_table = wm8711_of_match, }, .probe = wm8711_spi_probe, .remove = __devexit_p(wm8711_spi_remove), @@ -487,8 +495,9 @@ MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id); static struct i2c_driver wm8711_i2c_driver = { .driver = { - .name = "wm8711-codec", + .name = "wm8711", .owner = THIS_MODULE, + .of_match_table = wm8711_of_match, }, .probe = wm8711_i2c_probe, .remove = __devexit_p(wm8711_i2c_remove), diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 86d4718d3a76..04b027efd5c0 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -19,6 +19,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -269,6 +270,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8728 = { .num_dapm_routes = ARRAY_SIZE(wm8728_intercon), }; +static const struct of_device_id wm8728_of_match[] = { + { .compatible = "wlf,wm8728", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8728_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8728_spi_probe(struct spi_device *spi) { @@ -298,8 +305,9 @@ static int __devexit wm8728_spi_remove(struct spi_device *spi) static struct spi_driver wm8728_spi_driver = { .driver = { - .name = "wm8728-codec", + .name = "wm8728", .owner = THIS_MODULE, + .of_match_table = wm8728_of_match, }, .probe = wm8728_spi_probe, .remove = __devexit_p(wm8728_spi_remove), @@ -342,8 +350,9 @@ MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id); static struct i2c_driver wm8728_i2c_driver = { .driver = { - .name = "wm8728-codec", + .name = "wm8728", .owner = THIS_MODULE, + .of_match_table = wm8728_of_match, }, .probe = wm8728_i2c_probe, .remove = __devexit_p(wm8728_i2c_remove), 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/wm8737.c b/sound/soc/codecs/wm8737.c index 30c67d06a904..f6aef58845c2 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -20,6 +20,7 @@ #include <linux/regulator/consumer.h> #include <linux/spi/spi.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -634,6 +635,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8737 = { .reg_cache_default = wm8737_reg, }; +static const struct of_device_id wm8737_of_match[] = { + { .compatible = "wlf,wm8737", }, + { } +}; + +MODULE_DEVICE_TABLE(of, wm8737_of_match); + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8737_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -673,6 +681,7 @@ static struct i2c_driver wm8737_i2c_driver = { .driver = { .name = "wm8737", .owner = THIS_MODULE, + .of_match_table = wm8737_of_match, }, .probe = wm8737_i2c_probe, .remove = __devexit_p(wm8737_i2c_remove), @@ -711,6 +720,7 @@ static struct spi_driver wm8737_spi_driver = { .driver = { .name = "wm8737", .owner = THIS_MODULE, + .of_match_table = wm8737_of_match, }, .probe = wm8737_spi_probe, .remove = __devexit_p(wm8737_spi_remove), diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 25af901fe813..78c9e5ab3fa5 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -17,9 +17,11 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/spi/spi.h> #include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -422,17 +424,35 @@ static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); int ret = 0; + int i; + + for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) + wm8741->supplies[i].supply = wm8741_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8741->control_type); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; + goto err_enable; } ret = wm8741_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); - return ret; + goto err_enable; } /* Change some default settings - latch VU */ @@ -451,58 +471,61 @@ static int wm8741_probe(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Successful registration\n"); return ret; + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); +err: + return ret; +} + +static int wm8741_remove(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + + return 0; } static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .probe = wm8741_probe, + .remove = wm8741_remove, .resume = wm8741_resume, .reg_cache_size = ARRAY_SIZE(wm8741_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8741_reg_defaults, }; +static const struct of_device_id wm8741_of_match[] = { + { .compatible = "wlf,wm8741", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8741_of_match); + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8741_priv *wm8741; - int ret, i; + int ret; wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL); if (wm8741 == NULL) return -ENOMEM; - for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) - wm8741->supplies[i].supply = wm8741_supply_names[i]; - - ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8741->supplies), - wm8741->supplies); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); - goto err; - } - - ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies), - wm8741->supplies); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; - } - i2c_set_clientdata(i2c, wm8741); wm8741->control_type = SND_SOC_I2C; - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8741, &wm8741_dai, 1); - if (ret < 0) - goto err_enable; - return ret; + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_wm8741, &wm8741_dai, 1); + if (ret != 0) + goto err; -err_enable: - regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + return ret; -err_get: - regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); err: kfree(wm8741); return ret; @@ -510,10 +533,7 @@ err: static int wm8741_i2c_remove(struct i2c_client *client) { - struct wm8741_priv *wm8741 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); kfree(i2c_get_clientdata(client)); return 0; } @@ -526,8 +546,9 @@ MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id); static struct i2c_driver wm8741_i2c_driver = { .driver = { - .name = "wm8741-codec", + .name = "wm8741", .owner = THIS_MODULE, + .of_match_table = wm8741_of_match, }, .probe = wm8741_i2c_probe, .remove = wm8741_i2c_remove, @@ -535,6 +556,44 @@ static struct i2c_driver wm8741_i2c_driver = { }; #endif +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8741_spi_probe(struct spi_device *spi) +{ + struct wm8741_priv *wm8741; + int ret; + + wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL); + if (wm8741 == NULL) + return -ENOMEM; + + wm8741->control_type = SND_SOC_SPI; + spi_set_drvdata(spi, wm8741); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_wm8741, &wm8741_dai, 1); + if (ret < 0) + kfree(wm8741); + return ret; +} + +static int __devexit wm8741_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + kfree(spi_get_drvdata(spi)); + return 0; +} + +static struct spi_driver wm8741_spi_driver = { + .driver = { + .name = "wm8741", + .owner = THIS_MODULE, + .of_match_table = wm8741_of_match, + }, + .probe = wm8741_spi_probe, + .remove = __devexit_p(wm8741_spi_remove), +}; +#endif /* CONFIG_SPI_MASTER */ + static int __init wm8741_modinit(void) { int ret = 0; @@ -544,6 +603,13 @@ static int __init wm8741_modinit(void) if (ret != 0) pr_err("Failed to register WM8741 I2C driver: %d\n", ret); #endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8741_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register wm8741 SPI driver: %d\n", + ret); + } +#endif return ret; } @@ -551,6 +617,9 @@ module_init(wm8741_modinit); static void __exit wm8741_exit(void) { +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8741_spi_driver); +#endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8741_i2c_driver); #endif diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index d0003cc3bcd6..15f03721ec6f 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -21,6 +21,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -751,6 +752,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8750 = { .reg_cache_default = wm8750_reg, }; +static const struct of_device_id wm8750_of_match[] = { + { .compatible = "wlf,wm8750", }, + { .compatible = "wlf,wm8987", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8750_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8750_spi_probe(struct spi_device *spi) { @@ -787,8 +795,9 @@ MODULE_DEVICE_TABLE(spi, wm8750_spi_ids); static struct spi_driver wm8750_spi_driver = { .driver = { - .name = "wm8750-codec", + .name = "wm8750", .owner = THIS_MODULE, + .of_match_table = wm8750_of_match, }, .id_table = wm8750_spi_ids, .probe = wm8750_spi_probe, @@ -833,8 +842,9 @@ MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); static struct i2c_driver wm8750_i2c_driver = { .driver = { - .name = "wm8750-codec", + .name = "wm8750", .owner = THIS_MODULE, + .of_match_table = wm8750_of_match, }, .probe = wm8750_i2c_probe, .remove = __devexit_p(wm8750_i2c_remove), diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index ffa2ffe5ec11..fe04a101d657 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -38,6 +38,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> @@ -1490,6 +1491,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8753 = { .reg_cache_default = wm8753_reg, }; +static const struct of_device_id wm8753_of_match[] = { + { .compatible = "wlf,wm8753", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8753_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8753_spi_probe(struct spi_device *spi) { @@ -1519,8 +1526,9 @@ static int __devexit wm8753_spi_remove(struct spi_device *spi) static struct spi_driver wm8753_spi_driver = { .driver = { - .name = "wm8753-codec", + .name = "wm8753", .owner = THIS_MODULE, + .of_match_table = wm8753_of_match, }, .probe = wm8753_spi_probe, .remove = __devexit_p(wm8753_spi_remove), @@ -1563,8 +1571,9 @@ MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id); static struct i2c_driver wm8753_i2c_driver = { .driver = { - .name = "wm8753-codec", + .name = "wm8753", .owner = THIS_MODULE, + .of_match_table = wm8753_of_match, }, .probe = wm8753_i2c_probe, .remove = __devexit_p(wm8753_i2c_remove), diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 19b92baa9e8c..aa05e6507f84 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -14,6 +14,7 @@ #include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/of_device.h> #include <linux/pm.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> @@ -684,6 +685,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8770 = { .reg_cache_default = wm8770_reg_defs }; +static const struct of_device_id wm8770_of_match[] = { + { .compatible = "wlf,wm8770", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8770_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8770_spi_probe(struct spi_device *spi) { @@ -715,6 +722,7 @@ static struct spi_driver wm8770_spi_driver = { .driver = { .name = "wm8770", .owner = THIS_MODULE, + .of_match_table = wm8770_of_match, }, .probe = wm8770_spi_probe, .remove = __devexit_p(wm8770_spi_remove) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 8e7953b1b790..00d8846fae8a 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -18,6 +18,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> @@ -215,8 +216,6 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream, int ratio_shift, master; int i; - iface = 0; - switch (dai->driver->id) { case WM8776_DAI_DAC: iface_reg = WM8776_DACIFCTRL; @@ -232,20 +231,23 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Set word length */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(params_format(params))) { + case 16: + iface = 0; + case 20: + iface = 0x10; break; - case SNDRV_PCM_FORMAT_S20_3LE: - iface |= 0x10; + case 24: + iface = 0x20; break; - case SNDRV_PCM_FORMAT_S24_LE: - iface |= 0x20; - break; - case SNDRV_PCM_FORMAT_S32_LE: - iface |= 0x30; + case 32: + iface = 0x30; break; + default: + dev_err(codec->dev, "Unsupported sample size: %i\n", + snd_pcm_format_width(params_format(params))); + return -EINVAL; } /* Only need to set MCLK/LRCLK ratio if we're master */ @@ -320,11 +322,6 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define WM8776_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) - - #define WM8776_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -349,7 +346,9 @@ static struct snd_soc_dai_driver wm8776_dai[] = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = WM8776_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 32000, + .rate_max = 192000, .formats = WM8776_FORMATS, }, .ops = &wm8776_dac_ops, @@ -361,7 +360,9 @@ static struct snd_soc_dai_driver wm8776_dai[] = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = WM8776_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 32000, + .rate_max = 96000, .formats = WM8776_FORMATS, }, .ops = &wm8776_adc_ops, @@ -452,6 +453,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8776 = { .reg_cache_default = wm8776_reg, }; +static const struct of_device_id wm8776_of_match[] = { + { .compatible = "wlf,wm8776", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8776_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8776_spi_probe(struct spi_device *spi) { @@ -481,8 +488,9 @@ static int __devexit wm8776_spi_remove(struct spi_device *spi) static struct spi_driver wm8776_spi_driver = { .driver = { - .name = "wm8776-codec", + .name = "wm8776", .owner = THIS_MODULE, + .of_match_table = wm8776_of_match, }, .probe = wm8776_spi_probe, .remove = __devexit_p(wm8776_spi_remove), @@ -525,8 +533,9 @@ MODULE_DEVICE_TABLE(i2c, wm8776_i2c_id); static struct i2c_driver wm8776_i2c_driver = { .driver = { - .name = "wm8776-codec", + .name = "wm8776", .owner = THIS_MODULE, + .of_match_table = wm8776_of_match, }, .probe = wm8776_i2c_probe, .remove = __devexit_p(wm8776_i2c_remove), diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 9a5e67c5a6bd..9ee072b85975 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -16,6 +16,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/of_device.h> #include <linux/spi/spi.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> @@ -717,6 +718,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .volatile_register = wm8804_volatile }; +static const struct of_device_id wm8804_of_match[] = { + { .compatible = "wlf,wm8804", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8804_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8804_spi_probe(struct spi_device *spi) { @@ -748,6 +755,7 @@ static struct spi_driver wm8804_spi_driver = { .driver = { .name = "wm8804", .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, }, .probe = wm8804_spi_probe, .remove = __devexit_p(wm8804_spi_remove) @@ -792,6 +800,7 @@ static struct i2c_driver wm8804_i2c_driver = { .driver = { .name = "wm8804", .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, }, .probe = wm8804_i2c_probe, .remove = __devexit_p(wm8804_i2c_remove), diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 1725550c293e..3676b38838d8 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -63,6 +63,8 @@ struct wm8962_priv { int fll_fref; int fll_fout; + u16 dsp2_ena; + struct delayed_work mic_work; struct snd_soc_jack *jack; @@ -837,7 +839,7 @@ static const struct wm8962_reg_access { [40] = { 0x00FF, 0x01FF, 0x0000 }, /* R40 - SPKOUTL volume */ [41] = { 0x00FF, 0x01FF, 0x0000 }, /* R41 - SPKOUTR volume */ - [47] = { 0x000F, 0x0000, 0x0000 }, /* R47 - Thermal Shutdown Status */ + [47] = { 0x000F, 0x0000, 0xFFFF }, /* R47 - Thermal Shutdown Status */ [48] = { 0x7EC7, 0x7E07, 0xFFFF }, /* R48 - Additional Control (4) */ [49] = { 0x00D3, 0x00D7, 0xFFFF }, /* R49 - Class D Control 1 */ [51] = { 0x0047, 0x0047, 0x0000 }, /* R51 - Class D Control 2 */ @@ -965,7 +967,7 @@ static const struct wm8962_reg_access { [584] = { 0x002D, 0x002D, 0x0000 }, /* R584 - IRQ Debounce */ [586] = { 0xC000, 0xC000, 0x0000 }, /* R586 - MICINT Source Pol */ [768] = { 0x0001, 0x0001, 0x0000 }, /* R768 - DSP2 Power Management */ - [1037] = { 0x0000, 0x003F, 0x0000 }, /* R1037 - DSP2_ExecControl */ + [1037] = { 0x0000, 0x003F, 0xFFFF }, /* R1037 - DSP2_ExecControl */ [4096] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4096 - Write Sequencer 0 */ [4097] = { 0x00FF, 0x00FF, 0x0000 }, /* R4097 - Write Sequencer 1 */ [4098] = { 0x070F, 0x070F, 0x0000 }, /* R4098 - Write Sequencer 2 */ @@ -1986,6 +1988,122 @@ static const unsigned int classd_tlv[] = { }; static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static int wm8962_dsp2_write_config(struct snd_soc_codec *codec) +{ + return 0; +} + +static int wm8962_dsp2_set_enable(struct snd_soc_codec *codec, u16 val) +{ + u16 adcl = snd_soc_read(codec, WM8962_LEFT_ADC_VOLUME); + u16 adcr = snd_soc_read(codec, WM8962_RIGHT_ADC_VOLUME); + u16 dac = snd_soc_read(codec, WM8962_ADC_DAC_CONTROL_1); + + /* Mute the ADCs and DACs */ + snd_soc_write(codec, WM8962_LEFT_ADC_VOLUME, 0); + snd_soc_write(codec, WM8962_RIGHT_ADC_VOLUME, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1, + WM8962_DAC_MUTE, WM8962_DAC_MUTE); + + snd_soc_write(codec, WM8962_SOUNDSTAGE_ENABLES_0, val); + + /* Restore the ADCs and DACs */ + snd_soc_write(codec, WM8962_LEFT_ADC_VOLUME, adcl); + snd_soc_write(codec, WM8962_RIGHT_ADC_VOLUME, adcr); + snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1, + WM8962_DAC_MUTE, dac); + + return 0; +} + +static int wm8962_dsp2_start(struct snd_soc_codec *codec) +{ + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + wm8962_dsp2_write_config(codec); + + snd_soc_write(codec, WM8962_DSP2_EXECCONTROL, WM8962_DSP2_RUNR); + + wm8962_dsp2_set_enable(codec, wm8962->dsp2_ena); + + return 0; +} + +static int wm8962_dsp2_stop(struct snd_soc_codec *codec) +{ + wm8962_dsp2_set_enable(codec, 0); + + snd_soc_write(codec, WM8962_DSP2_EXECCONTROL, WM8962_DSP2_STOP); + + return 0; +} + +#define WM8962_DSP2_ENABLE(xname, xshift) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = wm8962_dsp2_ena_info, \ + .get = wm8962_dsp2_ena_get, .put = wm8962_dsp2_ena_put, \ + .private_value = xshift } + +static int wm8962_dsp2_ena_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} + +static int wm8962_dsp2_ena_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int shift = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !!(wm8962->dsp2_ena & 1 << shift); + + return 0; +} + +static int wm8962_dsp2_ena_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int shift = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + int old = wm8962->dsp2_ena; + int ret = 0; + int dsp2_running = snd_soc_read(codec, WM8962_DSP2_POWER_MANAGEMENT) & + WM8962_DSP2_ENA; + + mutex_lock(&codec->mutex); + + if (ucontrol->value.integer.value[0]) + wm8962->dsp2_ena |= 1 << shift; + else + wm8962->dsp2_ena &= ~(1 << shift); + + if (wm8962->dsp2_ena == old) + goto out; + + ret = 1; + + if (dsp2_running) { + if (wm8962->dsp2_ena) + wm8962_dsp2_set_enable(codec, wm8962->dsp2_ena); + else + wm8962_dsp2_stop(codec); + } + +out: + mutex_unlock(&codec->mutex); + + return ret; +} + /* The VU bits for the headphones are in a different register to the mute * bits and only take effect on the PGA if it is actually powered. */ @@ -2049,6 +2167,14 @@ static const char *cap_hpf_mode_text[] = { static const struct soc_enum cap_hpf_mode = SOC_ENUM_SINGLE(WM8962_ADC_DAC_CONTROL_2, 10, 2, cap_hpf_mode_text); + +static const char *cap_lhpf_mode_text[] = { + "LPF", "HPF" +}; + +static const struct soc_enum cap_lhpf_mode = + SOC_ENUM_SINGLE(WM8962_LHPF1, 1, 2, cap_lhpf_mode_text); + static const struct snd_kcontrol_new wm8962_snd_controls[] = { SOC_DOUBLE("Input Mixer Switch", WM8962_INPUT_MIXER_CONTROL_1, 3, 2, 1, 1), @@ -2077,6 +2203,8 @@ SOC_DOUBLE_R("Capture ZC Switch", WM8962_LEFT_INPUT_VOLUME, SOC_SINGLE("Capture HPF Switch", WM8962_ADC_DAC_CONTROL_1, 0, 1, 1), SOC_ENUM("Capture HPF Mode", cap_hpf_mode), SOC_SINGLE("Capture HPF Cutoff", WM8962_ADC_DAC_CONTROL_2, 7, 7, 0), +SOC_SINGLE("Capture LHPF Switch", WM8962_LHPF1, 0, 1, 0), +SOC_ENUM("Capture LHPF Mode", cap_lhpf_mode), SOC_DOUBLE_R_TLV("Sidetone Volume", WM8962_DAC_DSP_MIXING_1, WM8962_DAC_DSP_MIXING_2, 4, 12, 0, st_tlv), @@ -2134,6 +2262,11 @@ SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B4_GAIN_SHIFT, 31, 0, eq_tlv), SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv), + +WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT), +WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT), +WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT), +WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT), }; static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = { @@ -2395,6 +2528,31 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, } } +static int dsp2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (wm8962->dsp2_ena) + wm8962_dsp2_start(codec); + break; + + case SND_SOC_DAPM_PRE_PMD: + if (wm8962->dsp2_ena) + wm8962_dsp2_stop(codec); + break; + + default: + BUG(); + return -EINVAL; + } + + return 0; +} + static const char *st_text[] = { "None", "Right", "Left" }; static const struct soc_enum str_enum = @@ -2517,6 +2675,9 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, sysclk_event, SND_SOC_DAPM_SUPPLY("Charge Pump", WM8962_CHARGE_PUMP_1, 0, 0, cp_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY_S("DSP2", 1, WM8962_DSP2_POWER_MANAGEMENT, + WM8962_DSP2_ENA_SHIFT, 0, dsp2_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_MIXER("INPGAL", WM8962_LEFT_INPUT_PGA_CONTROL, 4, 0, inpgal, ARRAY_SIZE(inpgal)), @@ -2612,11 +2773,13 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "ADCL", NULL, "TOCLK" }, { "ADCL", NULL, "MIXINL" }, { "ADCL", NULL, "DMIC" }, + { "ADCL", NULL, "DSP2" }, { "ADCR", NULL, "SYSCLK" }, { "ADCR", NULL, "TOCLK" }, { "ADCR", NULL, "MIXINR" }, { "ADCR", NULL, "DMIC" }, + { "ADCR", NULL, "DSP2" }, { "STL", "Left", "ADCL" }, { "STL", "Right", "ADCR" }, @@ -2628,11 +2791,13 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "DACL", NULL, "TOCLK" }, { "DACL", NULL, "Beep" }, { "DACL", NULL, "STL" }, + { "DACL", NULL, "DSP2" }, { "DACR", NULL, "SYSCLK" }, { "DACR", NULL, "TOCLK" }, { "DACR", NULL, "Beep" }, { "DACR", NULL, "STR" }, + { "DACR", NULL, "DSP2" }, { "HPMIXL", "IN4L Switch", "IN4L" }, { "HPMIXL", "IN4R Switch", "IN4R" }, @@ -3403,12 +3568,16 @@ static irqreturn_t wm8962_irq(int irq, void *data) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int mask; int active; + int reg; mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK); active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); active &= ~mask; + if (!active) + return IRQ_NONE; + /* Acknowledge the interrupts */ snd_soc_write(codec, WM8962_INTERRUPT_STATUS_2, active); @@ -3420,9 +3589,21 @@ static irqreturn_t wm8962_irq(int irq, void *data) if (active & WM8962_FIFOS_ERR_EINT) dev_err(codec->dev, "FIFO error\n"); - if (active & WM8962_TEMP_SHUT_EINT) + if (active & WM8962_TEMP_SHUT_EINT) { dev_crit(codec->dev, "Thermal shutdown\n"); + reg = snd_soc_read(codec, WM8962_THERMAL_SHUTDOWN_STATUS); + + if (reg & WM8962_TEMP_ERR_HP) + dev_crit(codec->dev, "Headphone thermal error\n"); + if (reg & WM8962_TEMP_WARN_HP) + dev_crit(codec->dev, "Headphone thermal warning\n"); + if (reg & WM8962_TEMP_ERR_SPK) + dev_crit(codec->dev, "Speaker thermal error\n"); + if (reg & WM8962_TEMP_WARN_SPK) + dev_crit(codec->dev, "Speaker thermal warning\n"); + } + if (active & (WM8962_MICSCD_EINT | WM8962_MICD_EINT)) { dev_dbg(codec->dev, "Microphone event detected\n"); @@ -3479,31 +3660,6 @@ int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(wm8962_mic_detect); -#ifdef CONFIG_PM -static int wm8962_resume(struct snd_soc_codec *codec) -{ - u16 *reg_cache = codec->reg_cache; - int i; - - /* Restore the registers */ - for (i = 1; i < codec->driver->reg_cache_size; i++) { - switch (i) { - case WM8962_SOFTWARE_RESET: - continue; - default: - break; - } - - if (reg_cache[i] != wm8962_reg[i]) - snd_soc_write(codec, i, reg_cache[i]); - } - - return 0; -} -#else -#define wm8962_resume NULL -#endif - #if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) static int beep_rates[] = { 500, 1000, 2000, 4000, @@ -4015,7 +4171,6 @@ static int wm8962_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { .probe = wm8962_probe, .remove = wm8962_remove, - .resume = wm8962_resume, .set_bias_level = wm8962_set_bias_level, .reg_cache_size = WM8962_MAX_REGISTER + 1, .reg_word_size = sizeof(u16), diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 6e85b8869af7..eec8e1435116 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -847,6 +847,7 @@ SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8993_BUS_CONTROL_1, 1, 0, clk_sys_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("TOCLK", WM8993_CLOCKING_1, 14, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8993_CLOCKING_3, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_ADC("ADCL", NULL, WM8993_POWER_MANAGEMENT_2, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, WM8993_POWER_MANAGEMENT_2, 0, 0), @@ -880,6 +881,9 @@ SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), }; static const struct snd_soc_dapm_route routes[] = { + { "MICBIAS1", NULL, "VMID" }, + { "MICBIAS2", NULL, "VMID" }, + { "ADCL", NULL, "CLK_SYS" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCR", NULL, "CLK_SYS" }, @@ -1433,7 +1437,8 @@ 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-tables.c b/sound/soc/codecs/wm8994-tables.c index a87adbd05ee1..df5a8b9a250f 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -1073,8 +1073,8 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x0000, 0x0000 }, /* R1069 */ { 0x0000, 0x0000 }, /* R1070 */ { 0x0000, 0x0000 }, /* R1071 */ - { 0x0000, 0x0000 }, /* R1072 */ - { 0x0000, 0x0000 }, /* R1073 */ + { 0x006F, 0x006F }, /* R1072 - AIF1 DAC1 Noise Gate */ + { 0x006F, 0x006F }, /* R1073 - AIF1 DAC2 Noise Gate */ { 0x0000, 0x0000 }, /* R1074 */ { 0x0000, 0x0000 }, /* R1075 */ { 0x0000, 0x0000 }, /* R1076 */ @@ -1329,7 +1329,7 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x0000, 0x0000 }, /* R1325 */ { 0x0000, 0x0000 }, /* R1326 */ { 0x0000, 0x0000 }, /* R1327 */ - { 0x0000, 0x0000 }, /* R1328 */ + { 0x006F, 0x006F }, /* R1328 - AIF2 DAC Noise Gate */ { 0x0000, 0x0000 }, /* R1329 */ { 0x0000, 0x0000 }, /* R1330 */ { 0x0000, 0x0000 }, /* R1331 */ @@ -1635,8 +1635,8 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R58 - MICBIAS */ 0x000D, /* R59 - LDO 1 */ 0x0003, /* R60 - LDO 2 */ - 0x0000, /* R61 */ - 0x0000, /* R62 */ + 0x0039, /* R61 - MICBIAS1 */ + 0x0039, /* R62 - MICBIAS2 */ 0x0000, /* R63 */ 0x0000, /* R64 */ 0x0000, /* R65 */ @@ -2646,8 +2646,8 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R1069 */ 0x0000, /* R1070 */ 0x0000, /* R1071 */ - 0x0000, /* R1072 */ - 0x0000, /* R1073 */ + 0x0068, /* R1072 - AIF1 DAC1 Noise Gate */ + 0x0068, /* R1073 - AIF1 DAC2 Noise Gate */ 0x0000, /* R1074 */ 0x0000, /* R1075 */ 0x0000, /* R1076 */ @@ -2902,7 +2902,7 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R1325 */ 0x0000, /* R1326 */ 0x0000, /* R1327 */ - 0x0000, /* R1328 */ + 0x0068, /* R1328 - AIF2 DAC Noise Gate */ 0x0000, /* R1329 */ 0x0000, /* R1330 */ 0x0000, /* R1331 */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b393f9fac97a..e5372675123d 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; @@ -281,6 +282,7 @@ static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); static const DECLARE_TLV_DB_SCALE(st_tlv, -3600, 300, 0); static const DECLARE_TLV_DB_SCALE(wm8994_3d_tlv, -1600, 183, 0); static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); #define WM8994_DRC_SWITCH(xname, reg, shift) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -660,8 +662,45 @@ SOC_SINGLE_TLV("AIF2 EQ5 Volume", WM8994_AIF2_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static const char *wm8958_ng_text[] = { + "30ms", "125ms", "250ms", "500ms", +}; + +static const struct soc_enum wm8958_aif1dac1_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF1_DAC1_NOISE_GATE, + WM8958_AIF1DAC1_NG_THR_SHIFT, 4, wm8958_ng_text); + +static const struct soc_enum wm8958_aif1dac2_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF1_DAC2_NOISE_GATE, + WM8958_AIF1DAC2_NG_THR_SHIFT, 4, wm8958_ng_text); + +static const struct soc_enum wm8958_aif2dac_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF2_DAC_NOISE_GATE, + WM8958_AIF2DAC_NG_THR_SHIFT, 4, wm8958_ng_text); + static const struct snd_kcontrol_new wm8958_snd_controls[] = { SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv), + +SOC_SINGLE("AIF1DAC1 Noise Gate Switch", WM8958_AIF1_DAC1_NOISE_GATE, + WM8958_AIF1DAC1_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF1DAC1 Noise Gate Hold Time", wm8958_aif1dac1_ng_hold), +SOC_SINGLE_TLV("AIF1DAC1 Noise Gate Threshold Volume", + WM8958_AIF1_DAC1_NOISE_GATE, WM8958_AIF1DAC1_NG_THR_SHIFT, + 7, 1, ng_tlv), + +SOC_SINGLE("AIF1DAC2 Noise Gate Switch", WM8958_AIF1_DAC2_NOISE_GATE, + WM8958_AIF1DAC2_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF1DAC2 Noise Gate Hold Time", wm8958_aif1dac2_ng_hold), +SOC_SINGLE_TLV("AIF1DAC2 Noise Gate Threshold Volume", + WM8958_AIF1_DAC2_NOISE_GATE, WM8958_AIF1DAC2_NG_THR_SHIFT, + 7, 1, ng_tlv), + +SOC_SINGLE("AIF2DAC Noise Gate Switch", WM8958_AIF2_DAC_NOISE_GATE, + WM8958_AIF2DAC_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF2DAC Noise Gate Hold Time", wm8958_aif2dac_ng_hold), +SOC_SINGLE_TLV("AIF2DAC Noise Gate Threshold Volume", + WM8958_AIF2_DAC_NOISE_GATE, WM8958_AIF2DAC_NG_THR_SHIFT, + 7, 1, ng_tlv), }; static int clk_sys_event(struct snd_soc_dapm_widget *w, @@ -681,6 +720,97 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, return 0; } +static void vmid_reference(struct snd_soc_codec *codec) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + wm8994->vmid_refcount++; + + dev_dbg(codec->dev, "Referencing VMID, refcount is now %d\n", + wm8994->vmid_refcount); + + if (wm8994->vmid_refcount == 1) { + /* Startup bias, VMID ramp & buffer */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (0x11 << WM8994_VMID_RAMP_SHIFT)); + + /* Main bias enable, VMID=2x40k */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | + WM8994_VMID_SEL_MASK, + WM8994_BIAS_ENA | 0x2); + + msleep(20); + } +} + +static void vmid_dereference(struct snd_soc_codec *codec) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + wm8994->vmid_refcount--; + + dev_dbg(codec->dev, "Dereferencing VMID, refcount is now %d\n", + wm8994->vmid_refcount); + + if (wm8994->vmid_refcount == 0) { + /* Switch over to startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (1 << WM8994_VMID_RAMP_SHIFT)); + + /* Disable main biases */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | + WM8994_VMID_SEL_MASK, 0); + + /* Discharge line */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH); + + msleep(5); + + /* Switch off startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, 0); + } +} + +static int vmid_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + vmid_reference(codec); + break; + + case SND_SOC_DAPM_POST_PMD: + vmid_dereference(codec); + break; + } + + return 0; +} + static void wm8994_update_class_w(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -1208,6 +1338,8 @@ SND_SOC_DAPM_INPUT("Clock"), SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, vmid_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1525,6 +1657,8 @@ static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { static const struct snd_soc_dapm_route wm8994_intercon[] = { { "AIF2DACL", NULL, "AIF2DAC Mux" }, { "AIF2DACR", NULL, "AIF2DAC Mux" }, + { "MICBIAS1", NULL, "VMID" }, + { "MICBIAS2", NULL, "VMID" }, }; static const struct snd_soc_dapm_route wm8958_intercon[] = { @@ -1629,10 +1763,12 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, unsigned int freq_in, unsigned int freq_out) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; int reg_offset, ret; struct fll_div fll; u16 reg, aif1, aif2; unsigned long timeout; + bool was_enabled; aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) & WM8994_AIF1CLK_ENA; @@ -1653,6 +1789,9 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, return -EINVAL; } + reg = snd_soc_read(codec, WM8994_FLL1_CONTROL_1 + reg_offset); + was_enabled = reg & WM8994_FLL1_ENA; + switch (src) { case 0: /* Allow no source specification when stopping */ @@ -1719,6 +1858,21 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, /* Enable (with fractional mode if required) */ if (freq_out) { + /* Enable VMID if we need it */ + if (!was_enabled) { + switch (control->type) { + case WM8994: + vmid_reference(codec); + break; + case WM8958: + if (wm8994->revision < 1) + vmid_reference(codec); + break; + default: + break; + } + } + if (fll.k) reg = WM8994_FLL1_ENA | WM8994_FLL1_FRAC; else @@ -1736,6 +1890,20 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, } else { msleep(5); } + } else { + if (was_enabled) { + switch (control->type) { + case WM8994: + vmid_dereference(codec); + break; + case WM8958: + if (wm8994->revision < 1) + vmid_dereference(codec); + break; + default: + break; + } + } } wm8994->fll[id].in = freq_in; @@ -1852,9 +2020,6 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_PREPARE: - /* VMID=2x40k */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_VMID_SEL_MASK, 0x2); break; case SND_SOC_BIAS_STANDBY: @@ -1896,65 +2061,13 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, WM8994_LINEOUT2_DISCH, WM8994_LINEOUT1_DISCH | WM8994_LINEOUT2_DISCH); - - /* Startup bias, VMID ramp & buffer */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - (0x11 << WM8994_VMID_RAMP_SHIFT)); - - /* Main bias enable, VMID=2x40k */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_BIAS_ENA | - WM8994_VMID_SEL_MASK, - WM8994_BIAS_ENA | 0x2); - - msleep(20); } - /* VMID=2x500k */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_VMID_SEL_MASK, 0x4); break; case SND_SOC_BIAS_OFF: if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { - /* Switch over to startup biases */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_BIAS_SRC | - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, - WM8994_BIAS_SRC | - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - (1 << WM8994_VMID_RAMP_SHIFT)); - - /* Disable main biases */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_BIAS_ENA | - WM8994_VMID_SEL_MASK, 0); - - /* Discharge line */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_1, - WM8994_LINEOUT1_DISCH | - WM8994_LINEOUT2_DISCH, - WM8994_LINEOUT1_DISCH | - WM8994_LINEOUT2_DISCH); - - msleep(5); - - /* Switch off startup biases */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_BIAS_SRC | - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, 0); - wm8994->cur_fw = NULL; pm_runtime_put(codec->dev); @@ -2384,6 +2497,21 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) return snd_soc_update_bits(codec, reg, mask, val); } +static int wm8994_aif2_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* Disable the pulls on the AIF if we're using it to save power. */ + snd_soc_update_bits(codec, WM8994_GPIO_3, + WM8994_GPN_PU | WM8994_GPN_PD, 0); + snd_soc_update_bits(codec, WM8994_GPIO_4, + WM8994_GPN_PU | WM8994_GPN_PD, 0); + snd_soc_update_bits(codec, WM8994_GPIO_5, + WM8994_GPN_PU | WM8994_GPN_PD, 0); + + return 0; +} + #define WM8994_RATES SNDRV_PCM_RATE_8000_96000 #define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ @@ -2451,6 +2579,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, + .probe = wm8994_aif2_probe, .ops = &wm8994_aif2_dai_ops, }, { @@ -2916,6 +3045,24 @@ static irqreturn_t wm8994_fifo_error(int irq, void *data) return IRQ_HANDLED; } +static irqreturn_t wm8994_temp_warn(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_err(codec->dev, "Thermal warning\n"); + + return IRQ_HANDLED; +} + +static irqreturn_t wm8994_temp_shut(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_crit(codec->dev, "Thermal shutdown\n"); + + return IRQ_HANDLED; +} + static int wm8994_codec_probe(struct snd_soc_codec *codec) { struct wm8994 *control; @@ -2972,13 +3119,14 @@ 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; } break; @@ -2993,6 +3141,10 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, wm8994_fifo_error, "FIFO error", codec); + wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_WARN, + wm8994_temp_warn, "Thermal warning", codec); + wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_SHUT, + wm8994_temp_shut, "Thermal shutdown", codec); ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, wm_hubs_dcs_done, "DC servo done", @@ -3257,6 +3409,8 @@ err_irq: wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec); err: kfree(wm8994); return ret; @@ -3279,6 +3433,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec); switch (control->type) { case WM8994: diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 1ab2266039f7..f4f1355efc82 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -83,6 +83,8 @@ struct wm8994_priv { struct completion fll_locked[2]; bool fll_locked_irq; + int vmid_refcount; + int dac_rates[2]; int lrclk_shared[2]; diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 5ad873fda814..74ae5995a786 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1573,9 +1573,7 @@ static int wm8995_resume(struct snd_soc_codec *codec) static int wm8995_remove(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; - struct i2c_client *i2c; - i2c = container_of(codec->dev, struct i2c_client, dev); wm8995 = snd_soc_codec_get_drvdata(codec); wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -1642,6 +1640,7 @@ static int wm8995_probe(struct snd_soc_codec *codec) if (ret != 0x8995) { dev_err(codec->dev, "Invalid device ID: %#x\n", ret); + ret = -EINVAL; goto err_reg_enable; } diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 0cdb9d105671..833df74c5584 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -41,12 +41,11 @@ #define HPOUT2L 4 #define HPOUT2R 8 -#define WM8996_NUM_SUPPLIES 4 +#define WM8996_NUM_SUPPLIES 3 static const char *wm8996_supply_names[WM8996_NUM_SUPPLIES] = { "DBVDD", "AVDD1", "AVDD2", - "CPVDD", }; struct wm8996_priv { @@ -71,6 +70,8 @@ struct wm8996_priv { struct regulator_bulk_data supplies[WM8996_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8996_NUM_SUPPLIES]; + struct regulator *cpvdd; + int bg_ena; struct wm8996_pdata pdata; @@ -112,7 +113,6 @@ static int wm8996_regulator_event_##n(struct notifier_block *nb, \ WM8996_REGULATOR_EVENT(0) WM8996_REGULATOR_EVENT(1) WM8996_REGULATOR_EVENT(2) -WM8996_REGULATOR_EVENT(3) static const u16 wm8996_reg[WM8996_MAX_REGISTER] = { [WM8996_SOFTWARE_RESET] = 0x8996, @@ -414,6 +414,7 @@ static const DECLARE_TLV_DB_SCALE(out_digital_tlv, -1200, 150, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -900, 75, 0); static const DECLARE_TLV_DB_SCALE(spk_tlv, -900, 150, 0); static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(threedstereo_tlv, -1600, 183, 1); static const char *sidetone_hpf_text[] = { "2.9kHz", "1.5kHz", "735Hz", "403Hz", "196Hz", "98Hz", "49Hz" @@ -608,6 +609,14 @@ SOC_SINGLE("DAC High Performance Switch", WM8996_OVERSAMPLING, 0, 1, 0), SOC_SINGLE("DAC Soft Mute Switch", WM8996_DAC_SOFTMUTE, 1, 1, 0), SOC_SINGLE("DAC Slow Soft Mute Switch", WM8996_DAC_SOFTMUTE, 0, 1, 0), +SOC_SINGLE("DSP1 3D Stereo Switch", WM8996_DSP1_RX_FILTERS_2, 8, 1, 0), +SOC_SINGLE("DSP2 3D Stereo Switch", WM8996_DSP2_RX_FILTERS_2, 8, 1, 0), + +SOC_SINGLE_TLV("DSP1 3D Stereo Volume", WM8996_DSP1_RX_FILTERS_2, 10, 15, + 0, threedstereo_tlv), +SOC_SINGLE_TLV("DSP2 3D Stereo Volume", WM8996_DSP2_RX_FILTERS_2, 10, 15, + 0, threedstereo_tlv), + SOC_DOUBLE_TLV("Digital Output 1 Volume", WM8996_DAC1_HPOUT1_VOLUME, 0, 4, 8, 0, out_digital_tlv), SOC_DOUBLE_TLV("Digital Output 2 Volume", WM8996_DAC2_HPOUT2_VOLUME, 0, 4, @@ -658,19 +667,75 @@ SOC_SINGLE_TLV("DSP2 EQ B5 Volume", WM8996_DSP2_RX_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static void wm8996_bg_enable(struct snd_soc_codec *codec) +{ + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + + wm8996->bg_ena++; + if (wm8996->bg_ena == 1) { + snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, + WM8996_BG_ENA, WM8996_BG_ENA); + msleep(2); + } +} + +static void wm8996_bg_disable(struct snd_soc_codec *codec) +{ + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + + wm8996->bg_ena--; + if (!wm8996->bg_ena) + snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, + WM8996_BG_ENA, 0); +} + +static int bg_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wm8996_bg_enable(codec); + break; + case SND_SOC_DAPM_POST_PMD: + wm8996_bg_disable(codec); + break; + default: + BUG(); + ret = -EINVAL; + } + + return ret; +} + static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = w->codec; + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(wm8996->cpvdd); + if (ret != 0) + dev_err(codec->dev, "Failed to enable CPVDD: %d\n", + ret); + break; case SND_SOC_DAPM_POST_PMU: msleep(5); break; + case SND_SOC_DAPM_POST_PMD: + regulator_disable_deferred(wm8996->cpvdd, 20); + break; default: BUG(); - return -EINVAL; + ret = -EINVAL; } - return 0; + return ret; } static int rmv_short_event(struct snd_soc_dapm_widget *w, @@ -698,7 +763,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) { struct i2c_client *i2c = to_i2c_client(codec->dev); struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); - int i, ret; + int ret; unsigned long timeout = 200; snd_soc_write(codec, WM8996_DC_SERVO_2, mask); @@ -713,15 +778,12 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) } else { msleep(1); - if (--i) { - timeout = 0; - break; - } + timeout--; } ret = snd_soc_read(codec, WM8996_DC_SERVO_2); dev_dbg(codec->dev, "DC servo state: %x\n", ret); - } while (ret & mask); + } while (timeout && ret & mask); if (timeout == 0) dev_err(codec->dev, "DC servo timed out for %x\n", mask); @@ -979,9 +1041,12 @@ SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event, - SND_SOC_DAPM_POST_PMU), - + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("Bandgap", SND_SOC_NOPM, 0, 0, bg_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICB1 Audio", WM8996_MICBIAS_1, 4, 1, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICB2 Audio", WM8996_MICBIAS_2, 4, 1, NULL, 0), SND_SOC_DAPM_MICBIAS("MICB2", WM8996_POWER_MANAGEMENT_1, 9, 0), SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0), @@ -1035,14 +1100,14 @@ SND_SOC_DAPM_DAC("DAC2R", NULL, WM8996_POWER_MANAGEMENT_5, 2, 0), SND_SOC_DAPM_DAC("DAC1L", NULL, WM8996_POWER_MANAGEMENT_5, 1, 0), SND_SOC_DAPM_DAC("DAC1R", NULL, WM8996_POWER_MANAGEMENT_5, 0, 0), -SND_SOC_DAPM_AIF_IN("AIF2RX1", "AIF2 Playback", 1, +SND_SOC_DAPM_AIF_IN("AIF2RX1", "AIF2 Playback", 0, WM8996_POWER_MANAGEMENT_4, 9, 0), -SND_SOC_DAPM_AIF_IN("AIF2RX0", "AIF2 Playback", 2, +SND_SOC_DAPM_AIF_IN("AIF2RX0", "AIF2 Playback", 1, WM8996_POWER_MANAGEMENT_4, 8, 0), -SND_SOC_DAPM_AIF_IN("AIF2TX1", "AIF2 Capture", 1, +SND_SOC_DAPM_AIF_IN("AIF2TX1", "AIF2 Capture", 0, WM8996_POWER_MANAGEMENT_6, 9, 0), -SND_SOC_DAPM_AIF_IN("AIF2TX0", "AIF2 Capture", 2, +SND_SOC_DAPM_AIF_IN("AIF2TX0", "AIF2 Capture", 1, WM8996_POWER_MANAGEMENT_6, 8, 0), SND_SOC_DAPM_AIF_IN("AIF1RX5", "AIF1 Playback", 5, @@ -1137,17 +1202,23 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "Charge Pump", NULL, "SYSCLK" }, { "MICB1", NULL, "LDO2" }, + { "MICB1", NULL, "MICB1 Audio" }, + { "MICB1", NULL, "Bandgap" }, { "MICB2", NULL, "LDO2" }, + { "MICB2", NULL, "MICB2 Audio" }, + { "MICB2", NULL, "Bandgap" }, { "IN1L PGA", NULL, "IN2LN" }, { "IN1L PGA", NULL, "IN2LP" }, { "IN1L PGA", NULL, "IN1LN" }, { "IN1L PGA", NULL, "IN1LP" }, + { "IN1L PGA", NULL, "Bandgap" }, { "IN1R PGA", NULL, "IN2RN" }, { "IN1R PGA", NULL, "IN2RP" }, { "IN1R PGA", NULL, "IN1RN" }, { "IN1R PGA", NULL, "IN1RP" }, + { "IN1R PGA", NULL, "Bandgap" }, { "ADCL", NULL, "IN1L PGA" }, @@ -1281,6 +1352,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "DAC2R", NULL, "DAC2R Mixer" }, { "HPOUT2L PGA", NULL, "Charge Pump" }, + { "HPOUT2L PGA", NULL, "Bandgap" }, { "HPOUT2L PGA", NULL, "DAC2L" }, { "HPOUT2L_DLY", NULL, "HPOUT2L PGA" }, { "HPOUT2L_DCS", NULL, "HPOUT2L_DLY" }, @@ -1288,6 +1360,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "HPOUT2L_RMV_SHORT", NULL, "HPOUT2L_OUTP" }, { "HPOUT2R PGA", NULL, "Charge Pump" }, + { "HPOUT2R PGA", NULL, "Bandgap" }, { "HPOUT2R PGA", NULL, "DAC2R" }, { "HPOUT2R_DLY", NULL, "HPOUT2R PGA" }, { "HPOUT2R_DCS", NULL, "HPOUT2R_DLY" }, @@ -1295,6 +1368,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "HPOUT2R_RMV_SHORT", NULL, "HPOUT2R_OUTP" }, { "HPOUT1L PGA", NULL, "Charge Pump" }, + { "HPOUT1L PGA", NULL, "Bandgap" }, { "HPOUT1L PGA", NULL, "DAC1L" }, { "HPOUT1L_DLY", NULL, "HPOUT1L PGA" }, { "HPOUT1L_DCS", NULL, "HPOUT1L_DLY" }, @@ -1302,6 +1376,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "HPOUT1L_RMV_SHORT", NULL, "HPOUT1L_OUTP" }, { "HPOUT1R PGA", NULL, "Charge Pump" }, + { "HPOUT1R PGA", NULL, "Bandgap" }, { "HPOUT1R PGA", NULL, "DAC1R" }, { "HPOUT1R_DLY", NULL, "HPOUT1R PGA" }, { "HPOUT1R_DCS", NULL, "HPOUT1R_DLY" }, @@ -1620,14 +1695,7 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - break; - case SND_SOC_BIAS_PREPARE: - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { - snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, - WM8996_BG_ENA, WM8996_BG_ENA); - msleep(2); - } break; case SND_SOC_BIAS_STANDBY: @@ -1650,9 +1718,6 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, codec->cache_only = false; snd_soc_cache_sync(codec); } - - snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, - WM8996_BG_ENA, 0); break; case SND_SOC_BIAS_OFF: @@ -2041,7 +2106,7 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, struct i2c_client *i2c = to_i2c_client(codec->dev); struct _fll_div fll_div; unsigned long timeout; - int ret, reg; + int ret, reg, retry; /* Any change? */ if (source == wm8996->fll_src && Fref == wm8996->fll_fref && @@ -2057,6 +2122,8 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1, WM8996_FLL_ENA, 0); + wm8996_bg_disable(codec); + return 0; } @@ -2111,6 +2178,11 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda); + /* Enable the bandgap if it's not already enabled */ + ret = snd_soc_read(codec, WM8996_FLL_CONTROL_1); + if (!(ret & WM8996_FLL_ENA)) + wm8996_bg_enable(codec); + /* Clear any pending completions (eg, from failed startups) */ try_wait_for_completion(&wm8996->fll_lock); @@ -2128,17 +2200,29 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, else timeout = msecs_to_jiffies(2); - /* Allow substantially longer if we've actually got the IRQ */ + /* Allow substantially longer if we've actually got the IRQ, poll + * at a slightly higher rate if we don't. + */ if (i2c->irq) - timeout *= 1000; + timeout *= 10; + else + timeout /= 2; - ret = wait_for_completion_timeout(&wm8996->fll_lock, timeout); + for (retry = 0; retry < 10; retry++) { + ret = wait_for_completion_timeout(&wm8996->fll_lock, + timeout); + if (ret != 0) { + WARN_ON(!i2c->irq); + break; + } - if (ret == 0 && i2c->irq) { + ret = snd_soc_read(codec, WM8996_INTERRUPT_RAW_STATUS_2); + if (ret & WM8996_FLL_LOCK_STS) + break; + } + if (retry == 10) { dev_err(codec->dev, "Timed out waiting for FLL\n"); ret = -ETIMEDOUT; - } else { - ret = 0; } dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); @@ -2297,12 +2381,94 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, /* Enable interrupts and we're off */ snd_soc_update_bits(codec, WM8996_INTERRUPT_STATUS_2_MASK, - WM8996_IM_MICD_EINT, 0); + WM8996_IM_MICD_EINT | WM8996_HP_DONE_EINT, 0); return 0; } EXPORT_SYMBOL_GPL(wm8996_detect); +static void wm8996_hpdet_irq(struct snd_soc_codec *codec) +{ + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + int val, reg, report; + + /* Assume headphone in error conditions; we need to report + * something or we stall our state machine. + */ + report = SND_JACK_HEADPHONE; + + reg = snd_soc_read(codec, WM8996_HEADPHONE_DETECT_2); + if (reg < 0) { + dev_err(codec->dev, "Failed to read HPDET status\n"); + goto out; + } + + if (!(reg & WM8996_HP_DONE)) { + dev_err(codec->dev, "Got HPDET IRQ but HPDET is busy\n"); + goto out; + } + + val = reg & WM8996_HP_LVL_MASK; + + dev_dbg(codec->dev, "HPDET measured %d ohms\n", val); + + /* If we've got high enough impedence then report as line, + * otherwise assume headphone. + */ + if (val >= 126) + report = SND_JACK_LINEOUT; + else + report = SND_JACK_HEADPHONE; + +out: + if (wm8996->jack_mic) + report |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(wm8996->jack, report, + SND_JACK_LINEOUT | SND_JACK_HEADSET); + + wm8996->detecting = false; + + /* If the output isn't running re-clamp it */ + if (!(snd_soc_read(codec, WM8996_POWER_MANAGEMENT_1) & + (WM8996_HPOUT1L_ENA | WM8996_HPOUT1R_RMV_SHORT))) + snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT, 0); + + /* Go back to looking at the microphone */ + snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_1, + WM8996_JD_MODE_MASK, 0); + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA, + WM8996_MICD_ENA); + + snd_soc_dapm_disable_pin(&codec->dapm, "Bandgap"); + snd_soc_dapm_sync(&codec->dapm); +} + +static void wm8996_hpdet_start(struct snd_soc_codec *codec) +{ + /* Unclamp the output, we can't measure while we're shorting it */ + snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT); + + /* We need bandgap for HPDET */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "Bandgap"); + snd_soc_dapm_sync(&codec->dapm); + + /* Go into headphone detect left mode */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA, 0); + snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_1, + WM8996_JD_MODE_MASK, 1); + + /* Trigger a measurement */ + snd_soc_update_bits(codec, WM8996_HEADPHONE_DETECT_1, + WM8996_HP_POLL, WM8996_HP_POLL); +} + static void wm8996_micd(struct snd_soc_codec *codec) { struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); @@ -2323,28 +2489,36 @@ static void wm8996_micd(struct snd_soc_codec *codec) wm8996->jack_mic = false; wm8996->detecting = true; snd_soc_jack_report(wm8996->jack, 0, - SND_JACK_HEADSET | SND_JACK_BTN_0); + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0); + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_RATE_MASK, WM8996_MICD_RATE_MASK); return; } - /* If the measurement is very high we've got a microphone but - * do a little debounce to account for mechanical issues. + /* If the measurement is very high we've got a microphone, + * either we just detected one or if we already reported then + * we've got a button release event. */ if (val & 0x400) { - dev_dbg(codec->dev, "Microphone detected\n"); - snd_soc_jack_report(wm8996->jack, SND_JACK_HEADSET, - SND_JACK_HEADSET | SND_JACK_BTN_0); - wm8996->jack_mic = true; - wm8996->detecting = false; - - /* Increase poll rate to give better responsiveness - * for buttons */ - snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - 5 << WM8996_MICD_RATE_SHIFT); + if (wm8996->detecting) { + dev_dbg(codec->dev, "Microphone detected\n"); + wm8996->jack_mic = true; + wm8996_hpdet_start(codec); + + /* Increase poll rate to give better responsiveness + * for buttons */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, + WM8996_MICD_RATE_MASK, + 5 << WM8996_MICD_RATE_SHIFT); + } else { + dev_dbg(codec->dev, "Mic button up\n"); + snd_soc_jack_report(wm8996->jack, 0, SND_JACK_BTN_0); + } + + return; } /* If we detected a lower impedence during initial startup @@ -2376,15 +2550,11 @@ static void wm8996_micd(struct snd_soc_codec *codec) if (val & 0x3fc) { if (wm8996->jack_mic) { dev_dbg(codec->dev, "Mic button detected\n"); - snd_soc_jack_report(wm8996->jack, - SND_JACK_HEADSET | SND_JACK_BTN_0, - SND_JACK_HEADSET | SND_JACK_BTN_0); - } else { - dev_dbg(codec->dev, "Headphone detected\n"); - snd_soc_jack_report(wm8996->jack, - SND_JACK_HEADPHONE, - SND_JACK_HEADSET | + snd_soc_jack_report(wm8996->jack, SND_JACK_BTN_0, SND_JACK_BTN_0); + } else if (wm8996->detecting) { + dev_dbg(codec->dev, "Headphone detected\n"); + wm8996_hpdet_start(codec); /* Increase the detection rate a bit for * responsiveness. @@ -2392,8 +2562,6 @@ static void wm8996_micd(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_RATE_MASK, 7 << WM8996_MICD_RATE_SHIFT); - - wm8996->detecting = false; } } } @@ -2412,6 +2580,9 @@ static irqreturn_t wm8996_irq(int irq, void *data) } irq_val &= ~snd_soc_read(codec, WM8996_INTERRUPT_STATUS_2_MASK); + if (!irq_val) + return IRQ_NONE; + snd_soc_write(codec, WM8996_INTERRUPT_STATUS_2, irq_val); if (irq_val & (WM8996_DCS_DONE_01_EINT | WM8996_DCS_DONE_23_EINT)) { @@ -2430,10 +2601,10 @@ static irqreturn_t wm8996_irq(int irq, void *data) if (irq_val & WM8996_MICD_EINT) wm8996_micd(codec); - if (irq_val) - return IRQ_HANDLED; - else - return IRQ_NONE; + if (irq_val & WM8996_HP_DONE_EINT) + wm8996_hpdet_irq(codec); + + return IRQ_HANDLED; } static irqreturn_t wm8996_edge_irq(int irq, void *data) @@ -2548,7 +2719,13 @@ static int wm8996_probe(struct snd_soc_codec *codec) wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0; wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1; wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2; - wm8996->disable_nb[3].notifier_call = wm8996_regulator_event_3; + + wm8996->cpvdd = regulator_get(&i2c->dev, "CPVDD"); + if (IS_ERR(wm8996->cpvdd)) { + ret = PTR_ERR(wm8996->cpvdd); + dev_err(&i2c->dev, "Failed to get CPVDD: %d\n", ret); + goto err_get; + } /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) { @@ -2565,7 +2742,7 @@ static int wm8996_probe(struct snd_soc_codec *codec) wm8996->supplies); if (ret != 0) { dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; + goto err_cpvdd; } if (wm8996->pdata.ldo_ena >= 0) { @@ -2808,6 +2985,8 @@ err_enable: gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); +err_cpvdd: + regulator_put(wm8996->cpvdd); err_get: regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); err: @@ -2831,6 +3010,7 @@ static int wm8996_remove(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) regulator_unregister_notifier(wm8996->supplies[i].consumer, &wm8996->disable_nb[i]); + regulator_put(wm8996->cpvdd); regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); return 0; diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index a4691321f9b3..f32ab1ee9647 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1120,8 +1120,8 @@ static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } -static int wm9081_set_sysclk(struct snd_soc_codec *codec, - int clk_id, unsigned int freq, int dir) +static int wm9081_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) { struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 4de12203e611..f2f3077928da 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -139,7 +139,6 @@ static const u16 wm9090_reg_defaults[] = { /* This struct is used to save the context */ struct wm9090_priv { - struct mutex mutex; struct wm9090_platform_data pdata; void *control_data; }; @@ -663,7 +662,6 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm9090); wm9090->control_data = i2c; - mutex_init(&wm9090->mutex); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9090, NULL, 0); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index e763c54c55dc..ca8ce03510f4 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 */ @@ -699,6 +711,11 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "IN1L PGA", "IN1LP Switch", "IN1LP" }, { "IN1L PGA", "IN1LN Switch", "IN1LN" }, + { "IN1L PGA", NULL, "VMID" }, + { "IN1R PGA", NULL, "VMID" }, + { "IN2L PGA", NULL, "VMID" }, + { "IN2R PGA", NULL, "VMID" }, + { "IN1R PGA", "IN1RP Switch", "IN1RP" }, { "IN1R PGA", "IN1RN Switch", "IN1RN" }, @@ -716,12 +733,14 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "MIXINL", NULL, "Direct Voice" }, { "MIXINL", NULL, "IN1LP" }, { "MIXINL", NULL, "Left Output Mixer" }, + { "MIXINL", NULL, "VMID" }, { "MIXINR", "IN1R Switch", "IN1R PGA" }, { "MIXINR", "IN2R Switch", "IN2R PGA" }, { "MIXINR", NULL, "Direct Voice" }, { "MIXINR", NULL, "IN1RP" }, { "MIXINR", NULL, "Right Output Mixer" }, + { "MIXINR", NULL, "VMID" }, { "ADCL", NULL, "MIXINL" }, { "ADCR", NULL, "MIXINR" }, @@ -752,6 +771,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Earpiece Mixer", "Left Output Switch", "Left Output PGA" }, { "Earpiece Mixer", "Right Output Switch", "Right Output PGA" }, + { "Earpiece Driver", NULL, "VMID" }, { "Earpiece Driver", NULL, "Earpiece Mixer" }, { "HPOUT2N", NULL, "Earpiece Driver" }, { "HPOUT2P", NULL, "Earpiece Driver" }, @@ -774,9 +794,11 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "SPKR Boost", "SPKR Switch", "SPKR" }, { "SPKR Boost", "SPKL Switch", "SPKL" }, + { "SPKL Driver", NULL, "VMID" }, { "SPKL Driver", NULL, "SPKL Boost" }, { "SPKL Driver", NULL, "CLK_SYS" }, + { "SPKR Driver", NULL, "VMID" }, { "SPKR Driver", NULL, "SPKR Boost" }, { "SPKR Driver", NULL, "CLK_SYS" }, @@ -790,12 +812,18 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Headphone PGA", NULL, "Left Headphone Mux" }, { "Headphone PGA", NULL, "Right Headphone Mux" }, + { "Headphone PGA", NULL, "VMID" }, { "Headphone PGA", NULL, "CLK_SYS" }, { "Headphone PGA", NULL, "Headphone Supply" }, { "HPOUT1L", NULL, "Headphone PGA" }, { "HPOUT1R", NULL, "Headphone PGA" }, + { "LINEOUT1N Driver", NULL, "VMID" }, + { "LINEOUT1P Driver", NULL, "VMID" }, + { "LINEOUT2N Driver", NULL, "VMID" }, + { "LINEOUT2P Driver", NULL, "VMID" }, + { "LINEOUT1N", NULL, "LINEOUT1N Driver" }, { "LINEOUT1P", NULL, "LINEOUT1P Driver" }, { "LINEOUT2N", NULL, "LINEOUT2N Driver" }, 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/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 8566238db2a5..7173df254a91 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -732,16 +732,19 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, davinci_hw_param(dev, substream->stream); switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_S8: dma_params->data_type = 1; word_length = DAVINCI_AUDIO_WORD_8; break; + case SNDRV_PCM_FORMAT_U16_LE: case SNDRV_PCM_FORMAT_S16_LE: dma_params->data_type = 2; word_length = DAVINCI_AUDIO_WORD_16; break; + case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; word_length = DAVINCI_AUDIO_WORD_32; @@ -818,6 +821,13 @@ static struct snd_soc_dai_ops davinci_mcasp_dai_ops = { }; +#define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + static struct snd_soc_dai_driver davinci_mcasp_dai[] = { { .name = "davinci-mcasp.0", @@ -825,17 +835,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .channels_min = 2, .channels_max = 2, .rates = DAVINCI_MCASP_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, + .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, .channels_max = 2, .rates = DAVINCI_MCASP_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, + .formats = DAVINCI_MCASP_PCM_FMTS, }, .ops = &davinci_mcasp_dai_ops, @@ -846,7 +852,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .channels_min = 1, .channels_max = 384, .rates = DAVINCI_MCASP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = DAVINCI_MCASP_PCM_FMTS, }, .ops = &davinci_mcasp_dai_ops, }, diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index a49e667373bc..d5fe08cc5db7 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -180,7 +180,6 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - int link = prtd->asp_link[0]; unsigned int period_size; unsigned int dma_offset; dma_addr_t dma_pos; @@ -198,7 +197,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) fifo_level = prtd->params->fifo_level; pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " - "dma_ptr = %x period_size=%x\n", link, dma_pos, period_size); + "dma_ptr = %x period_size=%x\n", prtd->asp_link[0], dma_pos, + period_size); data_type = prtd->params->data_type; count = period_size / data_type; @@ -222,17 +222,19 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) } acnt = prtd->params->acnt; - edma_set_src(link, src, INCR, W8BIT); - edma_set_dest(link, dst, INCR, W8BIT); + edma_set_src(prtd->asp_link[0], src, INCR, W8BIT); + edma_set_dest(prtd->asp_link[0], dst, INCR, W8BIT); - edma_set_src_index(link, src_bidx, src_cidx); - edma_set_dest_index(link, dst_bidx, dst_cidx); + edma_set_src_index(prtd->asp_link[0], src_bidx, src_cidx); + edma_set_dest_index(prtd->asp_link[0], dst_bidx, dst_cidx); if (!fifo_level) - edma_set_transfer_params(link, acnt, count, 1, 0, ASYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, + ASYNC); else - edma_set_transfer_params(link, acnt, fifo_level, count, - fifo_level, ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, + count, fifo_level, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) @@ -305,7 +307,6 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) unsigned int acnt = params->acnt; /* divide by 2 for ping/pong */ unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1; - int link = prtd->asp_link[1]; unsigned int fifo_level = prtd->params->fifo_level; unsigned int count; if ((data_type == 0) || (data_type > 4)) { @@ -316,28 +317,26 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) dma_addr_t asp_src_pong = iram_dma->addr + ping_size; ram_src_cidx = ping_size; ram_dst_cidx = -ping_size; - edma_set_src(link, asp_src_pong, INCR, W8BIT); + edma_set_src(prtd->asp_link[1], asp_src_pong, INCR, W8BIT); - link = prtd->asp_link[0]; - edma_set_src_index(link, data_type, data_type * fifo_level); - link = prtd->asp_link[1]; - edma_set_src_index(link, data_type, data_type * fifo_level); + edma_set_src_index(prtd->asp_link[0], data_type, + data_type * fifo_level); + edma_set_src_index(prtd->asp_link[1], data_type, + data_type * fifo_level); - link = prtd->ram_link; - edma_set_src(link, runtime->dma_addr, INCR, W32BIT); + edma_set_src(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); } else { dma_addr_t asp_dst_pong = iram_dma->addr + ping_size; ram_src_cidx = -ping_size; ram_dst_cidx = ping_size; - edma_set_dest(link, asp_dst_pong, INCR, W8BIT); + edma_set_dest(prtd->asp_link[1], asp_dst_pong, INCR, W8BIT); - link = prtd->asp_link[0]; - edma_set_dest_index(link, data_type, data_type * fifo_level); - link = prtd->asp_link[1]; - edma_set_dest_index(link, data_type, data_type * fifo_level); + edma_set_dest_index(prtd->asp_link[0], data_type, + data_type * fifo_level); + edma_set_dest_index(prtd->asp_link[1], data_type, + data_type * fifo_level); - link = prtd->ram_link; - edma_set_dest(link, runtime->dma_addr, INCR, W32BIT); + edma_set_dest(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); } if (!fifo_level) { @@ -354,10 +353,9 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) count, fifo_level, ABSYNC); } - link = prtd->ram_link; - edma_set_src_index(link, ping_size, ram_src_cidx); - edma_set_dest_index(link, ping_size, ram_dst_cidx); - edma_set_transfer_params(link, ping_size, 2, + edma_set_src_index(prtd->ram_link, ping_size, ram_src_cidx); + edma_set_dest_index(prtd->ram_link, ping_size, ram_dst_cidx); + edma_set_transfer_params(prtd->ram_link, ping_size, 2, runtime->periods, 2, ASYNC); /* init master params */ @@ -406,32 +404,32 @@ static int request_ping_pong(struct snd_pcm_substream *substream, { dma_addr_t asp_src_ping; dma_addr_t asp_dst_ping; - int link; + int ret; struct davinci_pcm_dma_params *params = prtd->params; /* Request ram master channel */ - link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, + ret = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, davinci_pcm_dma_irq, substream, prtd->params->ram_chan_q); - if (link < 0) + if (ret < 0) goto exit1; /* Request ram link channel */ - link = prtd->ram_link = edma_alloc_slot( + ret = prtd->ram_link = edma_alloc_slot( EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit2; - link = prtd->asp_link[1] = edma_alloc_slot( + ret = prtd->asp_link[1] = edma_alloc_slot( EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit3; prtd->ram_link2 = -1; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - link = prtd->ram_link2 = edma_alloc_slot( + ret = prtd->ram_link2 = edma_alloc_slot( EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit4; } /* circle ping-pong buffers */ @@ -448,36 +446,33 @@ static int request_ping_pong(struct snd_pcm_substream *substream, asp_dst_ping = iram_dma->addr; } /* ping */ - link = prtd->asp_link[0]; - edma_set_src(link, asp_src_ping, INCR, W16BIT); - edma_set_dest(link, asp_dst_ping, INCR, W16BIT); - edma_set_src_index(link, 0, 0); - edma_set_dest_index(link, 0, 0); + edma_set_src(prtd->asp_link[0], asp_src_ping, INCR, W16BIT); + edma_set_dest(prtd->asp_link[0], asp_dst_ping, INCR, W16BIT); + edma_set_src_index(prtd->asp_link[0], 0, 0); + edma_set_dest_index(prtd->asp_link[0], 0, 0); - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); prtd->asp_params.opt |= TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(link, &prtd->asp_params); + edma_write_slot(prtd->asp_link[0], &prtd->asp_params); /* pong */ - link = prtd->asp_link[1]; - edma_set_src(link, asp_src_ping, INCR, W16BIT); - edma_set_dest(link, asp_dst_ping, INCR, W16BIT); - edma_set_src_index(link, 0, 0); - edma_set_dest_index(link, 0, 0); + edma_set_src(prtd->asp_link[1], asp_src_ping, INCR, W16BIT); + edma_set_dest(prtd->asp_link[1], asp_dst_ping, INCR, W16BIT); + edma_set_src_index(prtd->asp_link[1], 0, 0); + edma_set_dest_index(prtd->asp_link[1], 0, 0); - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[1], &prtd->asp_params); prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); /* interrupt after every pong completion */ prtd->asp_params.opt |= TCINTEN | TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(link, &prtd->asp_params); + edma_write_slot(prtd->asp_link[1], &prtd->asp_params); /* ram */ - link = prtd->ram_link; - edma_set_src(link, iram_dma->addr, INCR, W32BIT); - edma_set_dest(link, iram_dma->addr, INCR, W32BIT); + edma_set_src(prtd->ram_link, iram_dma->addr, INCR, W32BIT); + edma_set_dest(prtd->ram_link, iram_dma->addr, INCR, W32BIT); pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u," "for asp:%u %u %u\n", __func__, prtd->ram_channel, prtd->ram_link, prtd->ram_link2, @@ -494,7 +489,7 @@ exit2: edma_free_channel(prtd->ram_channel); prtd->ram_channel = -1; exit1: - return link; + return ret; } static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) @@ -502,22 +497,22 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) struct snd_dma_buffer *iram_dma; struct davinci_runtime_data *prtd = substream->runtime->private_data; struct davinci_pcm_dma_params *params = prtd->params; - int link; + int ret; if (!params) return -ENODEV; /* Request asp master DMA channel */ - link = prtd->asp_channel = edma_alloc_channel(params->channel, + ret = prtd->asp_channel = edma_alloc_channel(params->channel, davinci_pcm_dma_irq, substream, prtd->params->asp_chan_q); - if (link < 0) + if (ret < 0) goto exit1; /* Request asp link channels */ - link = prtd->asp_link[0] = edma_alloc_slot( + ret = prtd->asp_link[0] = edma_alloc_slot( EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit2; iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; @@ -537,17 +532,17 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) * the buffer and its length (ccnt) ... use it as a template * so davinci_pcm_enqueue_dma() takes less time in IRQ. */ - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); prtd->asp_params.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); - prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(link) << 5; - edma_write_slot(link, &prtd->asp_params); + prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5; + edma_write_slot(prtd->asp_link[0], &prtd->asp_params); return 0; exit2: edma_free_channel(prtd->asp_channel); prtd->asp_channel = -1; exit1: - return link; + return ret; } static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index d3aa15119d26..0134d4e9131c 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -28,12 +28,6 @@ #include <mach/hardware.h> #include "ep93xx-pcm.h" -#define edb93xx_has_audio() (machine_is_edb9301() || \ - machine_is_edb9302() || \ - machine_is_edb9302a() || \ - machine_is_edb9307a() || \ - machine_is_edb9315a()) - static int edb93xx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -94,49 +88,61 @@ static struct snd_soc_card snd_soc_edb93xx = { .num_links = 1, }; -static struct platform_device *edb93xx_snd_device; - -static int __init edb93xx_init(void) +static int __devinit edb93xx_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_edb93xx; int ret; - if (!edb93xx_has_audio()) - return -ENODEV; - ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, EP93XX_SYSCON_I2SCLKDIV_ORIDE | EP93XX_SYSCON_I2SCLKDIV_SPOL); if (ret) return ret; - edb93xx_snd_device = platform_device_alloc("soc-audio", -1); - if (!edb93xx_snd_device) { - ret = -ENOMEM; - goto free_i2s; + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + ep93xx_i2s_release(); } - platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx); - ret = platform_device_add(edb93xx_snd_device); - if (ret) - goto device_put; + return ret; +} - return 0; +static int __devexit edb93xx_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); -device_put: - platform_device_put(edb93xx_snd_device); -free_i2s: + snd_soc_unregister_card(card); ep93xx_i2s_release(); - return ret; + + return 0; +} + +static struct platform_driver edb93xx_driver = { + .driver = { + .name = "edb93xx-audio", + .owner = THIS_MODULE, + }, + .probe = edb93xx_probe, + .remove = __devexit_p(edb93xx_remove), +}; + +static int __init edb93xx_init(void) +{ + return platform_driver_register(&edb93xx_driver); } module_init(edb93xx_init); static void __exit edb93xx_exit(void) { - platform_device_unregister(edb93xx_snd_device); - ep93xx_i2s_release(); + platform_driver_unregister(&edb93xx_driver); } module_exit(edb93xx_exit); MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>"); MODULE_DESCRIPTION("ALSA SoC EDB93xx"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:edb93xx-audio"); diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 8dfd3ad84b19..d00230a591b1 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -355,3 +355,4 @@ module_exit(ep93xx_soc_platform_exit); MODULE_AUTHOR("Ryan Mallon"); MODULE_DESCRIPTION("EP93xx ALSA PCM interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:ep93xx-pcm-audio"); diff --git a/sound/soc/ep93xx/simone.c b/sound/soc/ep93xx/simone.c index 286817946c56..968cb316d511 100644 --- a/sound/soc/ep93xx/simone.c +++ b/sound/soc/ep93xx/simone.c @@ -39,53 +39,61 @@ static struct snd_soc_card snd_soc_simone = { }; static struct platform_device *simone_snd_ac97_device; -static struct platform_device *simone_snd_device; -static int __init simone_init(void) +static int __devinit simone_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_simone; int ret; - if (!machine_is_sim_one()) - return -ENODEV; - - simone_snd_ac97_device = platform_device_alloc("ac97-codec", -1); - if (!simone_snd_ac97_device) - return -ENOMEM; + simone_snd_ac97_device = platform_device_register_simple("ac97-codec", + -1, NULL, 0); + if (IS_ERR(simone_snd_ac97_device)) + return PTR_ERR(simone_snd_ac97_device); - ret = platform_device_add(simone_snd_ac97_device); - if (ret) - goto fail1; + card->dev = &pdev->dev; - simone_snd_device = platform_device_alloc("soc-audio", -1); - if (!simone_snd_device) { - ret = -ENOMEM; - goto fail2; + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + platform_device_unregister(simone_snd_ac97_device); } - platform_set_drvdata(simone_snd_device, &snd_soc_simone); - ret = platform_device_add(simone_snd_device); - if (ret) - goto fail3; + return ret; +} + +static int __devexit simone_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + platform_device_unregister(simone_snd_ac97_device); return 0; +} -fail3: - platform_device_put(simone_snd_device); -fail2: - platform_device_del(simone_snd_ac97_device); -fail1: - platform_device_put(simone_snd_ac97_device); - return ret; +static struct platform_driver simone_driver = { + .driver = { + .name = "simone-audio", + .owner = THIS_MODULE, + }, + .probe = simone_probe, + .remove = __devexit_p(simone_remove), +}; + +static int __init simone_init(void) +{ + return platform_driver_register(&simone_driver); } module_init(simone_init); static void __exit simone_exit(void) { - platform_device_unregister(simone_snd_device); - platform_device_unregister(simone_snd_ac97_device); + platform_driver_unregister(&simone_driver); } module_exit(simone_exit); MODULE_DESCRIPTION("ALSA SoC Simplemachines Sim.One"); MODULE_AUTHOR("Mika Westerberg <mika.westerberg@iki.fi>"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:simone-audio"); diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c index c8aa8a5003ca..f74ac54c285a 100644 --- a/sound/soc/ep93xx/snappercl15.c +++ b/sound/soc/ep93xx/snappercl15.c @@ -104,37 +104,56 @@ static struct snd_soc_card snd_soc_snappercl15 = { .num_links = 1, }; -static struct platform_device *snappercl15_snd_device; - -static int __init snappercl15_init(void) +static int __devinit snappercl15_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_snappercl15; int ret; - if (!machine_is_snapper_cl15()) - return -ENODEV; - ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, EP93XX_SYSCON_I2SCLKDIV_ORIDE | EP93XX_SYSCON_I2SCLKDIV_SPOL); if (ret) return ret; - snappercl15_snd_device = platform_device_alloc("soc-audio", -1); - if (!snappercl15_snd_device) - return -ENOMEM; - - platform_set_drvdata(snappercl15_snd_device, &snd_soc_snappercl15); - ret = platform_device_add(snappercl15_snd_device); - if (ret) - platform_device_put(snappercl15_snd_device); + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + ep93xx_i2s_release(); + } return ret; } -static void __exit snappercl15_exit(void) +static int __devexit snappercl15_remove(struct platform_device *pdev) { - platform_device_unregister(snappercl15_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); ep93xx_i2s_release(); + + return 0; +} + +static struct platform_driver snappercl15_driver = { + .driver = { + .name = "snappercl15-audio", + .owner = THIS_MODULE, + }, + .probe = snappercl15_probe, + .remove = __devexit_p(snappercl15_remove), +}; + +static int __init snappercl15_init(void) +{ + return platform_driver_register(&snappercl15_driver); +} + +static void __exit snappercl15_exit(void) +{ + platform_driver_unregister(&snappercl15_driver); } module_init(snappercl15_init); @@ -143,4 +162,4 @@ module_exit(snappercl15_exit); MODULE_AUTHOR("Ryan Mallon"); MODULE_DESCRIPTION("ALSA SoC Snapper CL15"); MODULE_LICENSE("GPL"); - +MODULE_ALIAS("platform:snappercl15-audio"); diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index cb50598338e9..ef15402a3bc4 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -297,7 +297,6 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; static u64 fsl_dma_dmamask = DMA_BIT_MASK(36); int ret; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index d48afea5d93d..0268cf989736 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -78,7 +78,6 @@ * @second_stream: pointer to second stream * @playback: the number of playback streams opened * @capture: the number of capture streams opened - * @asynchronous: 0=synchronous mode, 1=asynchronous mode * @cpu_dai: the CPU DAI for this device * @dev_attr: the sysfs device attribute structure * @stats: SSI statistics @@ -90,9 +89,6 @@ struct fsl_ssi_private { unsigned int irq; struct snd_pcm_substream *first_stream; struct snd_pcm_substream *second_stream; - unsigned int playback; - unsigned int capture; - int asynchronous; unsigned int fifo_depth; struct snd_soc_dai_driver cpu_dai_drv; struct device_attribute dev_attr; @@ -281,24 +277,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_ssi_private *ssi_private = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; /* * If this is the first stream opened, then request the IRQ * and initialize the SSI registers. */ - if (!ssi_private->playback && !ssi_private->capture) { + if (!ssi_private->first_stream) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - int ret; - - /* The 'name' should not have any slashes in it. */ - ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, - ssi_private->name, ssi_private); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not claim irq %u\n", ssi_private->irq); - return ret; - } + + ssi_private->first_stream = substream; /* * Section 16.5 of the MPC8610 reference manual says that the @@ -316,7 +306,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE - | (ssi_private->asynchronous ? 0 : CCSR_SSI_SCR_SYN)); + | (synchronous ? CCSR_SSI_SCR_SYN : 0)); out_be32(&ssi->stcr, CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -333,7 +323,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * master. */ - /* 4. Enable the interrupts and DMA requests */ + /* Enable the interrupts and DMA requests */ out_be32(&ssi->sier, SIER_FLAGS); /* @@ -362,58 +352,47 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * this is bad is because at this point, the PCM driver has not * finished initializing the DMA controller. */ - } + } else { + if (synchronous) { + struct snd_pcm_runtime *first_runtime = + ssi_private->first_stream->runtime; + /* + * This is the second stream open, and we're in + * synchronous mode, so we need to impose sample + * sample size constraints. This is because STCCR is + * used for playback and capture in synchronous mode, + * so there's no way to specify different word + * lengths. + * + * Note that this can cause a race condition if the + * second stream is opened before the first stream is + * fully initialized. We provide some protection by + * checking to make sure the first stream is + * initialized, but it's not perfect. ALSA sometimes + * re-initializes the driver with a different sample + * rate or size. If the second stream is opened + * before the first stream has received its final + * parameters, then the second stream may be + * constrained to the wrong sample rate or size. + */ + if (!first_runtime->sample_bits) { + dev_err(substream->pcm->card->dev, + "set sample size in %s stream first\n", + substream->stream == + SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + return -EAGAIN; + } - if (!ssi_private->first_stream) - ssi_private->first_stream = substream; - else { - /* This is the second stream open, so we need to impose sample - * rate and maybe sample size constraints. Note that this can - * cause a race condition if the second stream is opened before - * the first stream is fully initialized. - * - * We provide some protection by checking to make sure the first - * stream is initialized, but it's not perfect. ALSA sometimes - * re-initializes the driver with a different sample rate or - * size. If the second stream is opened before the first stream - * has received its final parameters, then the second stream may - * be constrained to the wrong sample rate or size. - * - * FIXME: This code does not handle opening and closing streams - * repeatedly. If you open two streams and then close the first - * one, you may not be able to open another stream until you - * close the second one as well. - */ - struct snd_pcm_runtime *first_runtime = - ssi_private->first_stream->runtime; - - if (!first_runtime->sample_bits) { - dev_err(substream->pcm->card->dev, - "set sample size in %s stream first\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? "capture" : "playback"); - return -EAGAIN; - } - - /* If we're in synchronous mode, then we need to constrain - * the sample size as well. We don't support independent sample - * rates in asynchronous mode. - */ - if (!ssi_private->asynchronous) snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, first_runtime->sample_bits, first_runtime->sample_bits); + } ssi_private->second_stream = substream; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ssi_private->playback++; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ssi_private->capture++; - return 0; } @@ -434,24 +413,35 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int sample_size = + snd_pcm_format_width(params_format(hw_params)); + u32 wl = CCSR_SSI_SxCCR_WL(sample_size); + int enabled = in_be32(&ssi->scr) & CCSR_SSI_SCR_SSIEN; - if (substream == ssi_private->first_stream) { - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - unsigned int sample_size = - snd_pcm_format_width(params_format(hw_params)); - u32 wl = CCSR_SSI_SxCCR_WL(sample_size); + /* + * If we're in synchronous mode, and the SSI is already enabled, + * then STCCR is already set properly. + */ + if (enabled && ssi_private->cpu_dai_drv.symmetric_rates) + return 0; - /* The SSI should always be disabled at this points (SSIEN=0) */ + /* + * FIXME: The documentation says that SxCCR[WL] should not be + * modified while the SSI is enabled. The only time this can + * happen is if we're trying to do simultaneous playback and + * capture in asynchronous mode. Unfortunately, I have been enable + * to get that to work at all on the P1022DS. Therefore, we don't + * bother to disable/enable the SSI when setting SxCCR[WL], because + * the SSI will stop anyway. Maybe one day, this will get fixed. + */ - /* In synchronous mode, the SSI uses STCCR for capture */ - if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || - !ssi_private->asynchronous) - clrsetbits_be32(&ssi->stccr, - CCSR_SSI_SxCCR_WL_MASK, wl); - else - clrsetbits_be32(&ssi->srccr, - CCSR_SSI_SxCCR_WL_MASK, wl); - } + /* In synchronous mode, the SSI uses STCCR for capture */ + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || + ssi_private->cpu_dai_drv.symmetric_rates) + clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); + else + clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); return 0; } @@ -474,7 +464,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) setbits32(&ssi->scr, @@ -510,27 +499,18 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ssi_private->playback--; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ssi_private->capture--; - if (ssi_private->first_stream == substream) ssi_private->first_stream = ssi_private->second_stream; ssi_private->second_stream = NULL; /* - * If this is the last active substream, disable the SSI and release - * the IRQ. + * If this is the last active substream, disable the SSI. */ - if (!ssi_private->playback && !ssi_private->capture) { + if (!ssi_private->first_stream) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); - - free_irq(ssi_private->irq, ssi_private); } } @@ -675,22 +655,33 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) ret = of_address_to_resource(np, 0, &res); if (ret) { dev_err(&pdev->dev, "could not determine device resources\n"); - kfree(ssi_private); - return ret; + goto error_kmalloc; } ssi_private->ssi = of_iomap(np, 0); if (!ssi_private->ssi) { dev_err(&pdev->dev, "could not map device resources\n"); - kfree(ssi_private); - return -ENOMEM; + ret = -ENOMEM; + goto error_kmalloc; } ssi_private->ssi_phys = res.start; + ssi_private->irq = irq_of_parse_and_map(np, 0); + if (ssi_private->irq == NO_IRQ) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + ret = -ENXIO; + goto error_iomap; + } + + /* The 'name' should not have any slashes in it. */ + ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name, + ssi_private); + if (ret < 0) { + dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); + goto error_irqmap; + } /* Are the RX and the TX clocks locked? */ - if (of_find_property(np, "fsl,ssi-asynchronous", NULL)) - ssi_private->asynchronous = 1; - else + if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) ssi_private->cpu_dai_drv.symmetric_rates = 1; /* Determine the FIFO depth. */ @@ -711,7 +702,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "could not create sysfs %s file\n", ssi_private->dev_attr.attr.name); - goto error; + goto error_irq; } /* Register with ASoC */ @@ -720,7 +711,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &ssi_private->cpu_dai_drv); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); - goto error; + goto error_dev; } /* Trigger the machine driver's probe function. The platform driver @@ -741,18 +732,28 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (IS_ERR(ssi_private->pdev)) { ret = PTR_ERR(ssi_private->pdev); dev_err(&pdev->dev, "failed to register platform: %d\n", ret); - goto error; + goto error_dai; } return 0; -error: +error_dai: snd_soc_unregister_dai(&pdev->dev); + +error_dev: dev_set_drvdata(&pdev->dev, NULL); - if (dev_attr) - device_remove_file(&pdev->dev, dev_attr); + device_remove_file(&pdev->dev, dev_attr); + +error_irq: + free_irq(ssi_private->irq, ssi_private); + +error_irqmap: irq_dispose_mapping(ssi_private->irq); + +error_iomap: iounmap(ssi_private->ssi); + +error_kmalloc: kfree(ssi_private); return ret; @@ -766,6 +767,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) snd_soc_unregister_dai(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); + free_irq(ssi_private->irq, ssi_private); + irq_dispose_mapping(ssi_private->irq); + kfree(ssi_private); dev_set_drvdata(&pdev->dev, NULL); diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 358f0baaf71b..31af405bda84 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -505,7 +505,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) return 0; error_sound: - platform_device_unregister(sound_device); + platform_device_put(sound_device); error: kfree(machine_data); error_alloc: diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index fcb862eb0c73..2c064a9824ad 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -267,7 +267,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) if (bus < 0) return bus; - snprintf(buf, len, "%s-codec.%u-%04x", temp, bus, addr); + snprintf(buf, len, "%s.%u-%04x", temp, bus, addr); return 0; } @@ -506,7 +506,7 @@ static int p1022_ds_probe(struct platform_device *pdev) error: if (sound_device) - platform_device_unregister(sound_device); + platform_device_put(sound_device); kfree(mdata); error_put: diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 7945625e0e08..c8527ead3736 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -242,23 +242,22 @@ static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; + struct snd_pcm_substream *substream; int ret; ret = imx_pcm_new(rtd); if (ret) return ret; - if (dai->driver->playback.channels_min) { - struct snd_pcm_substream *substream = - pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (substream) { struct snd_dma_buffer *buf = &substream->dma_buffer; imx_ssi_fiq_tx_buffer = (unsigned long)buf->area; } - if (dai->driver->capture.channels_min) { - struct snd_pcm_substream *substream = - pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + if (substream) { struct snd_dma_buffer *buf = &substream->dma_buffer; imx_ssi_fiq_rx_buffer = (unsigned long)buf->area; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 10a8e2783751..4297cb6af42e 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -357,8 +357,8 @@ int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; int ret; - ret = dma_mmap_coherent(NULL, vma, runtime->dma_area, - runtime->dma_addr, runtime->dma_bytes); + ret = dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, runtime->dma_area, @@ -399,14 +399,14 @@ int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) card->dev->dma_mask = &imx_pcm_dmamask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = imx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = imx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 3e7826058efe..9925d20ab0a3 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -226,13 +226,18 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) static int sst_platform_open(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime; + struct snd_pcm_runtime *runtime = substream->runtime; struct sst_runtime_stream *stream; int ret_val = 0; pr_debug("sst_platform_open called\n"); - runtime = substream->runtime; - runtime->hw = sst_platform_pcm_hw; + + snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); + ret_val = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret_val < 0) + return ret_val; + stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; @@ -259,8 +264,8 @@ static int sst_platform_open(struct snd_pcm_substream *substream) return ret_val; } runtime->private_data = stream; - return snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); + + return 0; } static int sst_platform_close(struct snd_pcm_substream *substream) @@ -469,7 +474,7 @@ static struct platform_driver sst_platform_driver = { static int __init sst_soc_platform_init(void) { pr_debug("sst_soc_platform_init called\n"); - return platform_driver_register(&sst_platform_driver); + return platform_driver_register(&sst_platform_driver); } module_init(sst_soc_platform_init); 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..401944cf4560 --- /dev/null +++ b/sound/soc/mxs/mxs-saif.c @@ -0,0 +1,797 @@ +/* + * 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 <linux/time.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/saif.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]; + +/* + * SAIF is a little different with other normal SOC DAIs on clock using. + * + * For MXS, two SAIF modules are instantiated on-chip. + * Each SAIF has a set of clock pins and can be operating in master + * mode simultaneously if they are connected to different off-chip codecs. + * Also, one of the two SAIFs can master or drive the clock pins while the + * other SAIF, in slave mode, receives clocking from the master SAIF. + * This also means that both SAIFs must operate at the same sample rate. + * + * We abstract this as each saif has a master, the master could be + * himself or other saifs. In the generic saif driver, saif does not need + * to know the different clkmux. Saif only needs to know who is his master + * and operating his master to generate the proper clock rate for him. + * The master id is provided in mach-specific layer according to different + * clkmux setting. + */ + +static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + 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; +} + +/* + * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK + * is provided by other SAIF, we provide a interface here to get its master + * from its master_id. + * Note that the master could be himself. + */ +static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif) +{ + return mxs_saif[saif->master_id]; +} + +/* + * Set SAIF clock and MCLK + */ +static int mxs_saif_set_clk(struct mxs_saif *saif, + unsigned int mclk, + unsigned int rate) +{ + u32 scr; + int ret; + struct mxs_saif *master_saif; + + dev_dbg(saif->dev, "mclk %d rate %d\n", mclk, rate); + + /* Set master saif to generate proper clock */ + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + + dev_dbg(saif->dev, "master saif%d\n", master_saif->id); + + /* Checking if can playback and capture simutaneously */ + if (master_saif->ongoing && rate != master_saif->cur_rate) { + dev_err(saif->dev, + "can not change clock, master saif%d(rate %d) is ongoing\n", + master_saif->id, master_saif->cur_rate); + return -EINVAL; + } + + scr = __raw_readl(master_saif->base + SAIF_CTRL); + scr &= ~BM_SAIF_CTRL_BITCLK_MULT_RATE; + scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; + + /* + * 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 (master_saif->mclk_in_use) { + if (mclk % 32 == 0) { + scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; + ret = clk_set_rate(master_saif->clk, 512 * rate); + } else if (mclk % 48 == 0) { + scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE; + ret = clk_set_rate(master_saif->clk, 384 * rate); + } else { + /* SAIF MCLK should be either 32x or 48x */ + return -EINVAL; + } + } else { + ret = clk_set_rate(master_saif->clk, 512 * rate); + scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; + } + + if (ret) + return ret; + + master_saif->cur_rate = rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(scr, master_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, master_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; + struct mxs_saif *master_saif; + + if (!saif) + return -EINVAL; + + /* Clear Reset */ + __raw_writel(BM_SAIF_CTRL_SFTRST, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + + /* FIXME: need clear clk gate for register r/w */ + __raw_writel(BM_SAIF_CTRL_CLKGATE, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + + master_saif = mxs_saif_get_master(saif); + if (saif != master_saif) { + dev_err(saif->dev, "can not get mclk from a non-master saif\n"); + return -EINVAL; + } + + stat = __raw_readl(saif->base + SAIF_STAT); + if (stat & BM_SAIF_STAT_BUSY) { + dev_err(saif->dev, "error: busy\n"); + return -EBUSY; + } + + 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_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. + * Here the master is relative to codec side. + * Saif internally could be slave when working on EXTMASTER mode. + * We just hide this to machine driver. + */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + if (saif->id == saif->master_id) + scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + else + scr |= BM_SAIF_CTRL_SLAVE_MODE; + + __raw_writel(scr | scr0, saif->base + SAIF_CTRL); + break; + default: + 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); + + /* clear clock gate */ + __raw_writel(BM_SAIF_CTRL_CLKGATE, + 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); + + /* 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); + struct mxs_saif *master_saif; + u32 delay; + + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dev_dbg(cpu_dai->dev, "start\n"); + + clk_enable(master_saif->clk); + if (!master_saif->mclk_in_use) + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_SET_ADDR); + + /* + * If the saif's master is not himself, we also need to enable + * itself clk for its internal basic logic to work. + */ + if (saif != master_saif) { + clk_enable(saif->clk); + __raw_writel(BM_SAIF_CTRL_RUN, + saif->base + SAIF_CTRL + MXS_SET_ADDR); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* + * 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); + } + + master_saif->ongoing = 1; + + dev_dbg(saif->dev, "CTRL 0x%x STAT 0x%x\n", + __raw_readl(saif->base + SAIF_CTRL), + __raw_readl(saif->base + SAIF_STAT)); + + dev_dbg(master_saif->dev, "CTRL 0x%x STAT 0x%x\n", + __raw_readl(master_saif->base + SAIF_CTRL), + __raw_readl(master_saif->base + SAIF_STAT)); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dev_dbg(cpu_dai->dev, "stop\n"); + + /* wait a while for the current sample to complete */ + delay = USEC_PER_SEC / master_saif->cur_rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(delay); + } + clk_disable(master_saif->clk); + + if (saif != master_saif) { + __raw_writel(BM_SAIF_CTRL_RUN, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(delay); + clk_disable(saif->clk); + } + + master_saif->ongoing = 0; + + break; + default: + 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; + struct mxs_saif_platform_data *pdata; + int ret = 0; + + if (pdev->id >= ARRAY_SIZE(mxs_saif)) + return -EINVAL; + + pdata = pdev->dev.platform_data; + if (pdata && pdata->init) { + ret = pdata->init(); + if (ret) + return ret; + } + + saif = kzalloc(sizeof(*saif), GFP_KERNEL); + if (!saif) + return -ENOMEM; + + mxs_saif[pdev->id] = saif; + saif->id = pdev->id; + + saif->master_id = saif->id; + if (pdata && pdata->get_master_id) { + saif->master_id = pdata->get_master_id(saif->id); + if (saif->master_id < 0 || + saif->master_id >= ARRAY_SIZE(mxs_saif)) + return -EINVAL; + } + + saif->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(saif->clk)) { + 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..12c91e4eb941 --- /dev/null +++ b/sound/soc/mxs/mxs-saif.h @@ -0,0 +1,134 @@ +/* + * 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; + unsigned int id; + unsigned int master_id; + unsigned int cur_rate; + unsigned int ongoing; + + 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..7fbeaec06eb4 --- /dev/null +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -0,0 +1,173 @@ +/* + * 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 Tx", + .stream_name = "HiFi Playback", + .codec_dai_name = "sgtl5000", + .codec_name = "sgtl5000.0-000a", + .cpu_dai_name = "mxs-saif.0", + .platform_name = "mxs-pcm-audio.0", + .ops = &mxs_sgtl5000_hifi_ops, + }, { + .name = "HiFi Rx", + .stream_name = "HiFi Capture", + .codec_dai_name = "sgtl5000", + .codec_name = "sgtl5000.0-000a", + .cpu_dai_name = "mxs-saif.1", + .platform_name = "mxs-pcm-audio.1", + .ops = &mxs_sgtl5000_hifi_ops, + }, +}; + +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/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index d589ef14e917..e46d5516e000 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -318,7 +318,6 @@ static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32); static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; if (!card->dev->dma_mask) diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 1a591f1ebfbd..b899a3bc8f42 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -306,8 +306,10 @@ static int __init raumfeld_audio_init(void) &snd_soc_raumfeld_connector); ret = platform_device_add(raumfeld_audio_device); - if (ret < 0) + if (ret < 0) { + platform_device_put(raumfeld_audio_device); return ret; + } raumfeld_enable_audio(true); return 0; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index b253d864868a..ce920e3cfea1 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -312,7 +312,7 @@ static struct snd_soc_dai_link spitz_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8750-codec.0-001b", + .codec_name = "wm8750.0-001b", .init = spitz_wm8750_init, .ops = &spitz_ops, }; diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index d69d9fc32233..4b81ffd87566 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -198,7 +198,7 @@ static struct snd_soc_dai_link z2_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8750-codec.0-001b", + .codec_name = "wm8750.0-001b", .init = z2_wm8750_init, .ops = &z2_ops, }; diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 80c85fd64e1a..55efc2bdf0bd 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -446,7 +446,6 @@ static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32); static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime) { struct snd_card *card = runtime->card->snd_card; - struct snd_soc_dai *dai = runtime->cpu_dai; struct snd_pcm *pcm = runtime->pcm; struct s6000_pcm_dma_params *params; int res; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 65f980ef2870..dd3b3eac0805 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -63,7 +63,7 @@ config SND_SOC_SAMSUNG_SMDK_WM8580 config SND_SOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210 || MACH_SMDK4212) select SND_SOC_WM8994 select SND_SAMSUNG_I2S help @@ -158,7 +158,7 @@ config SND_SOC_GONI_AQUILA_WM8994 config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310) + depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310 || MACH_SMDK4212) select SND_SAMSUNG_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. @@ -173,7 +173,7 @@ config SND_SOC_SMDK_WM8580_PCM config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310) + depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310 || MACH_SMDK4212) select SND_SOC_WM8994 select SND_SAMSUNG_PCM help diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 14eb6ea69e7c..ed8f13a29c85 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -131,7 +131,7 @@ static struct snd_soc_dai_link jive_dai = { .cpu_dai_name = "s3c2412-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-001a", + .codec_name = "wm8750.0-001a", .init = jive_wm8750_init, .ops = &jive_ops, }; diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 841ab14c1100..7ab8e2c29216 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -69,10 +69,10 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; s3c2412_i2s.iis_cclk = clk_get(dai->dev, "i2sclk"); - if (s3c2412_i2s.iis_cclk == NULL) { + if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); iounmap(s3c2412_i2s.regs); - return -ENODEV; + return PTR_ERR(s3c2412_i2s.iis_cclk); } /* Set MPLL as the source for IIS CLK */ diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 63d8849d80bd..21c92e2e3007 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -383,10 +383,10 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) return -ENXIO; s3c24xx_i2s.iis_clk = clk_get(dai->dev, "iis"); - if (s3c24xx_i2s.iis_clk == NULL) { + if (IS_ERR(s3c24xx_i2s.iis_clk)) { pr_err("failed to get iis_clock\n"); iounmap(s3c24xx_i2s.regs); - return -ENODEV; + return PTR_ERR(s3c24xx_i2s.iis_clk); } clk_enable(s3c24xx_i2s.iis_clk); diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index dc9d551f6788..65c1cfd47d8a 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -66,17 +66,17 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) pr_debug("%s %d\n", __func__, clk_users); if (clk_users == 0) { xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); - if (!xtal) { + if (IS_ERR(xtal)) { printk(KERN_ERR "%s cannot get xtal\n", __func__); - ret = -EBUSY; + ret = PTR_ERR(xtal); } else { pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, "pclk"); - if (!pclk) { + if (IS_ERR(pclk)) { printk(KERN_ERR "%s cannot get pclk\n", __func__); clk_put(xtal); - ret = -EBUSY; + ret = PTR_ERR(pclk); } } if (!ret) { diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 0a2c4f223038..bbd14768ecd3 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -207,7 +207,7 @@ static struct snd_soc_dai_link smartq_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-0x1a", + .codec_name = "wm8750.0-0x1a", .init = smartq_wm8987_init, .ops = &smartq_hifi_ops, }, diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 3d26f6607aa4..20deecf3b243 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -210,7 +210,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .init = smdk_wm8580_init_paifrx, .ops = &smdk_ops, }, @@ -220,7 +220,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8580-hifi-capture", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .init = smdk_wm8580_init_paiftx, .ops = &smdk_ops, }, @@ -230,7 +230,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-i2s.x", .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .init = smdk_wm8580_init_paifrx, .ops = &smdk_ops, }, diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index 0d12092df164..4b9c73477ce0 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -127,7 +127,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-pcm.0", .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .ops = &smdk_wm8580_pcm_ops, }, { .name = "WM8580 PAIF PCM TX", @@ -135,7 +135,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-pcm.0", .codec_dai_name = "wm8580-hifi-capture", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .ops = &smdk_wm8580_pcm_ops, }, }; diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 590e9274b062..b9e213f6cc06 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -125,10 +125,6 @@ static struct snd_soc_jack_pin speyside_headset_pins[] = { .pin = "Headset Mic", .mask = SND_JACK_MICROPHONE, }, - { - .pin = "Headphone", - .mask = SND_JACK_HEADPHONE, - }, }; /* Default the headphone selection to active high */ @@ -171,7 +167,8 @@ static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd) gpio_direction_output(WM8996_HPSEL_GPIO, speyside_jack_polarity); ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0, &speyside_headset); if (ret) return ret; @@ -227,7 +224,7 @@ static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) snd_soc_dapm_nc_pin(dapm, "LINEOUT"); /* At any time the WM9081 is active it will have this clock */ - return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, + return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, 48000 * 256, 0); } @@ -252,6 +249,7 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Main AMIC"), SOC_DAPM_PIN_SWITCH("WM1250 Input"), SOC_DAPM_PIN_SWITCH("WM1250 Output"), + SOC_DAPM_PIN_SWITCH("Headphone"), }; static struct snd_soc_dapm_widget widgets[] = { diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 72535f2daaf2..3820a6b057dc 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -31,13 +31,13 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, 32768, - 44100 * 256); + 44100 * 512); if (ret < 0) pr_err("Failed to start FLL: %d\n", ret); ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, - 44100 * 256, + 44100 * 512, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("Failed to set SYSCLK: %d\n", ret); diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 20b7f3b003a3..143c705ac27b 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -548,9 +548,6 @@ static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec, static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec) { - const struct snd_soc_codec_driver *codec_drv; - - codec_drv = codec->driver; return DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); } @@ -868,10 +865,6 @@ static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec) static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) { - const struct snd_soc_codec_driver *codec_drv; - - codec_drv = codec->driver; - if (codec->reg_def_copy) codec->reg_cache = kmemdup(codec->reg_def_copy, codec->reg_size, GFP_KERNEL); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2ef014af215..10e5cdeeb18e 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'; @@ -143,7 +143,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf, step = codec->driver->reg_cache_step; for (i = 0; i < codec->driver->reg_cache_size; i += step) { - if (codec->readable_register && !codec->readable_register(codec, i)) + if (!snd_soc_codec_readable_register(codec, i)) continue; if (codec->driver->display_register) { count += codec->driver->display_register(codec, buf + count, @@ -244,7 +244,6 @@ static ssize_t codec_reg_write_file(struct file *file, size_t buf_size; char *start = buf; unsigned long reg, value; - int step = 1; struct snd_soc_codec *codec = file->private_data; buf_size = min(count, (sizeof(buf)-1)); @@ -252,9 +251,6 @@ static ssize_t codec_reg_write_file(struct file *file, return -EFAULT; buf[buf_size] = 0; - if (codec->driver->reg_cache_step) - step = codec->driver->reg_cache_step; - while (*start == ' ') start++; reg = simple_strtoul(start, &start, 16); @@ -956,6 +952,8 @@ static int soc_probe_codec(struct snd_soc_card *card, snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, driver->num_dapm_widgets); + codec->dapm.idle_bias_off = driver->idle_bias_off; + if (driver->probe) { ret = driver->probe(codec); if (ret < 0) { @@ -2668,7 +2666,7 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, if (dai->driver && dai->driver->ops->set_sysclk) return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); else if (dai->codec && dai->codec->driver->set_sysclk) - return dai->codec->driver->set_sysclk(dai->codec, clk_id, + return dai->codec->driver->set_sysclk(dai->codec, clk_id, 0, freq, dir); else return -EINVAL; @@ -2679,16 +2677,18 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); * snd_soc_codec_set_sysclk - configure CODEC system or master clock. * @codec: CODEC * @clk_id: DAI specific clock ID + * @source: Source for the clock * @freq: new clock frequency in Hz * @dir: new clock direction - input/output. * * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. */ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, - unsigned int freq, int dir) + int source, unsigned int freq, int dir) { if (codec->driver->set_sysclk) - return codec->driver->set_sysclk(codec, clk_id, freq, dir); + return codec->driver->set_sysclk(codec, clk_id, source, + freq, dir); else return -EINVAL; } @@ -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 d67c637557a7..4a440b52dd7a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -443,6 +443,11 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) if (path->name != (char *)w->kcontrol_news[i].name) continue; + if (w->kcontrols[i]) { + path->kcontrol = w->kcontrols[i]; + continue; + } + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + sizeof(struct snd_soc_dapm_widget *), wlist = kzalloc(wlistsize, GFP_KERNEL); @@ -579,8 +584,8 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) name + prefix_len, prefix); ret = snd_ctl_add(card, kcontrol); if (ret < 0) { - dev_err(dapm->dev, - "asoc: failed to add kcontrol %s\n", w->name); + dev_err(dapm->dev, "failed to add kcontrol %s: %d\n", + w->name, ret); kfree(wlist); return ret; } @@ -1556,7 +1561,6 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, /* found, now check type */ found = 1; path->connect = connect; - break; } if (found) @@ -2584,7 +2588,7 @@ 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 +2608,10 @@ 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); } /** diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index a62f7dd4ba96..66fcccd79efe 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -13,26 +13,14 @@ #include <linux/i2c.h> #include <linux/spi/spi.h> +#include <linux/regmap.h> #include <sound/soc.h> #include <trace/events/asoc.h> -#ifdef CONFIG_SPI_MASTER -static int do_spi_write(void *control, const char *data, int len) -{ - struct spi_device *spi = control; - int ret; - - ret = spi_write(spi, data, len); - if (ret < 0) - return ret; - - return len; -} -#endif - -static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value, const void *data, int len) +#ifdef CONFIG_REGMAP +static int hw_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) { int ret; @@ -49,13 +37,7 @@ static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } - ret = codec->hw_write(codec->control_data, data, len); - if (ret == len) - return 0; - if (ret < 0) - return ret; - else - return -EIO; + return regmap_write(codec->control_data, reg, value); } static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) @@ -69,8 +51,11 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) if (codec->cache_only) return -1; - BUG_ON(!codec->hw_read); - return codec->hw_read(codec, reg); + ret = regmap_read(codec->control_data, reg, &val); + if (ret == 0) + return val; + else + return ret; } ret = snd_soc_cache_read(codec, reg, &val); @@ -79,202 +64,18 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) return val; } -static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data; - - data = cpu_to_be16((reg << 12) | (value & 0xffffff)); - - return do_hw_write(codec, reg, value, &data, 2); -} - -static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data; - - data = cpu_to_be16((reg << 9) | (value & 0x1ff)); - - return do_hw_write(codec, reg, value, &data, 2); -} - -static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - reg &= 0xff; - data[0] = reg; - data[1] = value & 0xff; - - return do_hw_write(codec, reg, value, data, 2); -} - -static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - u16 val = cpu_to_be16(value); - - data[0] = reg; - memcpy(&data[1], &val, sizeof(val)); - - return do_hw_write(codec, reg, value, data, 3); -} - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int do_i2c_read(struct snd_soc_codec *codec, - void *reg, int reglen, - void *data, int datalen) -{ - struct i2c_msg xfer[2]; - int ret; - struct i2c_client *client = codec->control_data; - - /* Write register */ - xfer[0].addr = client->addr; - xfer[0].flags = 0; - xfer[0].len = reglen; - xfer[0].buf = reg; - - /* Read data */ - xfer[1].addr = client->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = datalen; - xfer[1].buf = data; - - ret = i2c_transfer(client->adapter, xfer, 2); - if (ret == 2) - return 0; - else if (ret < 0) - return ret; - else - return -EIO; -} -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u8 reg = r; - u8 data; - int ret; - - ret = do_i2c_read(codec, ®, 1, &data, 1); - if (ret < 0) - return 0; - return data; -} -#else -#define snd_soc_8_8_read_i2c NULL -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u8 reg = r; - u16 data; - int ret; - - ret = do_i2c_read(codec, ®, 1, &data, 2); - if (ret < 0) - return 0; - return (data >> 8) | ((data & 0xff) << 8); -} -#else -#define snd_soc_8_16_read_i2c NULL -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u16 reg = r; - u8 data; - int ret; - - ret = do_i2c_read(codec, ®, 2, &data, 1); - if (ret < 0) - return 0; - return data; -} -#else -#define snd_soc_16_8_read_i2c NULL -#endif - -#if defined(CONFIG_SPI_MASTER) -static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec, - unsigned int r) -{ - struct spi_device *spi = codec->control_data; - - const u16 reg = cpu_to_be16(r | 0x100); - u8 data; - int ret; - - ret = spi_write_then_read(spi, ®, 2, &data, 1); - if (ret < 0) - return 0; - return data; -} -#else -#define snd_soc_16_8_read_spi NULL -#endif - -static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - u16 rval = cpu_to_be16(reg); - - memcpy(data, &rval, sizeof(rval)); - data[2] = value; - - return do_hw_write(codec, reg, value, data, 3); -} - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u16 reg = cpu_to_be16(r); - u16 data; - int ret; - - ret = do_i2c_read(codec, ®, 2, &data, 2); - if (ret < 0) - return 0; - return be16_to_cpu(data); -} -#else -#define snd_soc_16_16_read_i2c NULL -#endif - -static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data[2]; - - data[0] = cpu_to_be16(reg); - data[1] = cpu_to_be16(value); - - return do_hw_write(codec, reg, value, data, sizeof(data)); -} - /* Primitive bulk write support for soc-cache. The data pointed to by - * `data' needs to already be in the form the hardware expects - * including any leading register specific data. Any data written - * through this function will not go through the cache as it only - * handles writing to volatile or out of bounds registers. + * `data' needs to already be in the form the hardware expects. Any + * data written through this function will not go through the cache as + * it only handles writing to volatile or out of bounds registers. + * + * This is currently only supported for devices using the regmap API + * wrappers. */ -static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg, +static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, + unsigned int reg, const void *data, size_t len) { - int ret; - /* To ensure that we don't get out of sync with the cache, check * whether the base register is volatile or if we've directly asked * to bypass the cache. Out of bounds registers are considered @@ -285,68 +86,9 @@ static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int r && reg < codec->driver->reg_cache_size) return -EINVAL; - switch (codec->control_type) { -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) - case SND_SOC_I2C: - ret = i2c_master_send(to_i2c_client(codec->dev), data, len); - break; -#endif -#if defined(CONFIG_SPI_MASTER) - case SND_SOC_SPI: - ret = spi_write(to_spi_device(codec->dev), data, len); - break; -#endif - default: - BUG(); - } - - if (ret == len) - return 0; - if (ret < 0) - return ret; - else - return -EIO; + return regmap_raw_write(codec->control_data, reg, data, len); } -static struct { - int addr_bits; - int data_bits; - int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); - unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int); -} io_types[] = { - { - .addr_bits = 4, .data_bits = 12, - .write = snd_soc_4_12_write, - }, - { - .addr_bits = 7, .data_bits = 9, - .write = snd_soc_7_9_write, - }, - { - .addr_bits = 8, .data_bits = 8, - .write = snd_soc_8_8_write, - .i2c_read = snd_soc_8_8_read_i2c, - }, - { - .addr_bits = 8, .data_bits = 16, - .write = snd_soc_8_16_write, - .i2c_read = snd_soc_8_16_read_i2c, - }, - { - .addr_bits = 16, .data_bits = 8, - .write = snd_soc_16_8_write, - .i2c_read = snd_soc_16_8_read_i2c, - .spi_read = snd_soc_16_8_read_spi, - }, - { - .addr_bits = 16, .data_bits = 16, - .write = snd_soc_16_16_write, - .i2c_read = snd_soc_16_16_read_i2c, - }, -}; - /** * snd_soc_codec_set_cache_io: Set up standard I/O functions. * @@ -370,50 +112,51 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int addr_bits, int data_bits, enum snd_soc_control_type control) { - int i; - - for (i = 0; i < ARRAY_SIZE(io_types); i++) - if (io_types[i].addr_bits == addr_bits && - io_types[i].data_bits == data_bits) - break; - if (i == ARRAY_SIZE(io_types)) { - printk(KERN_ERR - "No I/O functions for %d bit address %d bit data\n", - addr_bits, data_bits); - return -EINVAL; - } + struct regmap_config config; - codec->write = io_types[i].write; + memset(&config, 0, sizeof(config)); + codec->write = hw_write; codec->read = hw_read; codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; + config.reg_bits = addr_bits; + config.val_bits = data_bits; + switch (control) { +#if defined(CONFIG_REGMAP_I2C) || defined(CONFIG_REGMAP_I2C_MODULE) case SND_SOC_I2C: -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) - codec->hw_write = (hw_write_t)i2c_master_send; -#endif - if (io_types[i].i2c_read) - codec->hw_read = io_types[i].i2c_read; - - codec->control_data = container_of(codec->dev, - struct i2c_client, - dev); + codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev), + &config); break; +#endif +#if defined(CONFIG_REGMAP_SPI) || defined(CONFIG_REGMAP_SPI_MODULE) case SND_SOC_SPI: -#ifdef CONFIG_SPI_MASTER - codec->hw_write = do_spi_write; + codec->control_data = regmap_init_spi(to_spi_device(codec->dev), + &config); + break; #endif - if (io_types[i].spi_read) - codec->hw_read = io_types[i].spi_read; - codec->control_data = container_of(codec->dev, - struct spi_device, - dev); + case SND_SOC_REGMAP: + /* Device has made its own regmap arrangements */ break; + + default: + return -EINVAL; } + if (IS_ERR(codec->control_data)) + return PTR_ERR(codec->control_data); + return 0; } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); - +#else +int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, + int addr_bits, int data_bits, + enum snd_soc_control_type control) +{ + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); +#endif diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2879c883eebc..1aee9fcdf650 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -27,8 +27,6 @@ #include <sound/soc.h> #include <sound/initval.h> -static DEFINE_MUTEX(pcm_mutex); - static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; |