diff options
Diffstat (limited to 'sound/soc/au1x')
| -rw-r--r-- | sound/soc/au1x/Kconfig | 48 | ||||
| -rw-r--r-- | sound/soc/au1x/Makefile | 14 | ||||
| -rw-r--r-- | sound/soc/au1x/ac97c.c | 348 | ||||
| -rw-r--r-- | sound/soc/au1x/db1000.c | 65 | ||||
| -rw-r--r-- | sound/soc/au1x/db1200.c | 215 | ||||
| -rw-r--r-- | sound/soc/au1x/dbdma2.c | 176 | ||||
| -rw-r--r-- | sound/soc/au1x/dma.c | 344 | ||||
| -rw-r--r-- | sound/soc/au1x/i2sc.c | 324 | ||||
| -rw-r--r-- | sound/soc/au1x/psc-ac97.c | 418 | ||||
| -rw-r--r-- | sound/soc/au1x/psc-i2s.c | 233 | ||||
| -rw-r--r-- | sound/soc/au1x/psc.h | 25 | ||||
| -rw-r--r-- | sound/soc/au1x/sample-ac97.c | 144 |
12 files changed, 1815 insertions, 539 deletions
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig index 410a893aa66..a56104040e8 100644 --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -1,13 +1,13 @@ ## -## Au1200/Au1550 PSC + DBDMA +## Au1200/Au1550/Au1300 PSC + DBDMA ## config SND_SOC_AU1XPSC - tristate "SoC Audio for Au1200/Au1250/Au1550" - depends on SOC_AU1200 || SOC_AU1550 + tristate "SoC Audio for Au12xx/Au13xx/Au1550" + depends on MIPS_ALCHEMY help This option enables support for the Programmable Serial Controllers in AC97 and I2S mode, and the Descriptor-Based DMA - Controller (DBDMA) as found on the Au1200/Au1250/Au1550 SoC. + Controller (DBDMA) as found on the Au12xx/Au13xx/Au1550 SoC. config SND_SOC_AU1XPSC_I2S tristate @@ -18,15 +18,47 @@ 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_SAMPLE_PSC_AC97 - tristate "Sample Au12x0/Au1550 PSC AC97 sound machine" +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/DB1300/DB1550 Audio support" depends on SND_SOC_AU1XPSC select SND_SOC_AU1XPSC_AC97 select SND_SOC_AC97_CODEC + select SND_SOC_WM9712 + select SND_SOC_AU1XPSC_I2S + select SND_SOC_WM8731 help - This is a sample AC97 sound machine for use in Au12x0/Au1550 - based systems which have audio on PSC1 (e.g. Db1200 demoboard). + Select this option to enable audio (AC97 and I2S) on the + Alchemy/AMD/RMI/NetLogic Db1200, Db1550 and Db1300 evaluation boards. + If you need Db1300 touchscreen support, you definitely want to say Y. diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile index 6c6950b8003..920710514ea 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-sample-ac97-objs := sample-ac97.o +snd-soc-db1000-objs := db1000.o +snd-soc-db1200-objs := db1200.o -obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o +obj-$(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 00000000000..c8a2de103c5 --- /dev/null +++ b/sound/soc/au1x/ac97c.c @@ -0,0 +1,348 @@ +/* + * 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 */ +static struct snd_ac97_bus_ops ac97c_bus_ops = { + .read = au1xac97c_ac97_read, + .write = au1xac97c_ac97_write, + .reset = au1xac97c_ac97_cold_reset, + .warm_reset = au1xac97c_ac97_warm_reset, +}; + +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 const 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 const struct snd_soc_component_driver au1xac97c_component = { + .name = "au1xac97c", +}; + +static int au1xac97c_drvprobe(struct platform_device *pdev) +{ + int ret; + struct resource *iores, *dmares; + struct au1xpsc_audio_data *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + mutex_init(&ctx->lock); + + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) + return -ENODEV; + + if (!devm_request_mem_region(&pdev->dev, iores->start, + resource_size(iores), + pdev->name)) + return -EBUSY; + + ctx->mmio = devm_ioremap_nocache(&pdev->dev, iores->start, + resource_size(iores)); + if (!ctx->mmio) + return -EBUSY; + + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + return -EBUSY; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; + + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) + return -EBUSY; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->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_set_ac97_ops(&ac97c_bus_ops); + if (ret) + return ret; + + ret = snd_soc_register_component(&pdev->dev, &au1xac97c_component, + &au1xac97c_dai_driver, 1); + if (ret) + return ret; + + ac97c_workdata = ctx; + return 0; +} + +static int au1xac97c_drvremove(struct platform_device *pdev) +{ + struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev); + + snd_soc_unregister_component(&pdev->dev); + + WR(ctx, AC97_ENABLE, EN_D); /* clock off, disable */ + + 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 = au1xac97c_drvremove, +}; + +module_platform_driver(au1xac97c_driver); + +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 00000000000..376d976bcc2 --- /dev/null +++ b/sound/soc/au1x/db1000.c @@ -0,0 +1,65 @@ +/* + * 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", + .owner = THIS_MODULE, + .dai_link = &db1000_ac97_dai, + .num_links = 1, +}; + +static int 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 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 = db1000_audio_remove, +}; + +module_platform_driver(db1000_audio_driver); + +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 new file mode 100644 index 00000000000..decba87a074 --- /dev/null +++ b/sound/soc/au1x/db1200.c @@ -0,0 +1,215 @@ +/* + * DB1200/DB1300/DB1550 ASoC audio fabric support code. + * + * (c) 2008-2011 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-au1x00/au1xxx_psc.h> +#include <asm/mach-au1x00/au1xxx_dbdma.h> +#include <asm/mach-db1x00/bcsr.h> + +#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, + }, { + .name = "db1300-ac97", + .driver_data = 2, + }, { + .name = "db1300-i2s", + .driver_data = 3, + }, { + .name = "db1550-ac97", + .driver_data = 4, + }, { + .name = "db1550-i2s", + .driver_data = 5, + }, + {}, +}; + +/*------------------------- AC97 PART ---------------------------*/ + +static struct snd_soc_dai_link db1200_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .codec_dai_name = "ac97-hifi", + .cpu_dai_name = "au1xpsc_ac97.1", + .platform_name = "au1xpsc-pcm.1", + .codec_name = "ac97-codec.1", +}; + +static struct snd_soc_card db1200_ac97_machine = { + .name = "DB1200_AC97", + .owner = THIS_MODULE, + .dai_link = &db1200_ac97_dai, + .num_links = 1, +}; + +static struct snd_soc_dai_link db1300_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .codec_dai_name = "wm9712-hifi", + .cpu_dai_name = "au1xpsc_ac97.1", + .platform_name = "au1xpsc-pcm.1", + .codec_name = "wm9712-codec.1", +}; + +static struct snd_soc_card db1300_ac97_machine = { + .name = "DB1300_AC97", + .owner = THIS_MODULE, + .dai_link = &db1300_ac97_dai, + .num_links = 1, +}; + +static struct snd_soc_card db1550_ac97_machine = { + .name = "DB1550_AC97", + .owner = THIS_MODULE, + .dai_link = &db1200_ac97_dai, + .num_links = 1, +}; + +/*------------------------- I2S PART ---------------------------*/ + +static int db1200_i2s_startup(struct snd_pcm_substream *substream) +{ + 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; + int ret; + + /* WM8731 has its own 12MHz crystal */ + snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, + 12000000, SND_SOC_CLOCK_IN); + + /* codec is bitclock and lrclk master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + goto out; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + goto out; + + ret = 0; +out: + return ret; +} + +static struct snd_soc_ops db1200_i2s_wm8731_ops = { + .startup = db1200_i2s_startup, +}; + +static struct snd_soc_dai_link db1200_i2s_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .codec_dai_name = "wm8731-hifi", + .cpu_dai_name = "au1xpsc_i2s.1", + .platform_name = "au1xpsc-pcm.1", + .codec_name = "wm8731.0-001b", + .ops = &db1200_i2s_wm8731_ops, +}; + +static struct snd_soc_card db1200_i2s_machine = { + .name = "DB1200_I2S", + .owner = THIS_MODULE, + .dai_link = &db1200_i2s_dai, + .num_links = 1, +}; + +static struct snd_soc_dai_link db1300_i2s_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .codec_dai_name = "wm8731-hifi", + .cpu_dai_name = "au1xpsc_i2s.2", + .platform_name = "au1xpsc-pcm.2", + .codec_name = "wm8731.0-001b", + .ops = &db1200_i2s_wm8731_ops, +}; + +static struct snd_soc_card db1300_i2s_machine = { + .name = "DB1300_I2S", + .owner = THIS_MODULE, + .dai_link = &db1300_i2s_dai, + .num_links = 1, +}; + +static struct snd_soc_dai_link db1550_i2s_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .codec_dai_name = "wm8731-hifi", + .cpu_dai_name = "au1xpsc_i2s.3", + .platform_name = "au1xpsc-pcm.3", + .codec_name = "wm8731.0-001b", + .ops = &db1200_i2s_wm8731_ops, +}; + +static struct snd_soc_card db1550_i2s_machine = { + .name = "DB1550_I2S", + .owner = THIS_MODULE, + .dai_link = &db1550_i2s_dai, + .num_links = 1, +}; + +/*------------------------- COMMON PART ---------------------------*/ + +static struct snd_soc_card *db1200_cards[] = { + &db1200_ac97_machine, + &db1200_i2s_machine, + &db1300_ac97_machine, + &db1300_i2s_machine, + &db1550_ac97_machine, + &db1550_i2s_machine, +}; + +static int db1200_audio_probe(struct platform_device *pdev) +{ + const struct platform_device_id *pid = platform_get_device_id(pdev); + struct snd_soc_card *card; + + card = db1200_cards[pid->driver_data]; + card->dev = &pdev->dev; + return snd_soc_register_card(card); +} + +static int db1200_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 db1200_audio_driver = { + .driver = { + .name = "db1200-ac97", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .id_table = db1200_pids, + .probe = db1200_audio_probe, + .remove = db1200_audio_remove, +}; + +module_platform_driver(db1200_audio_driver); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("DB1200/DB1300/DB1550 ASoC audio support"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index bc8d654576c..17a24d80473 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -2,7 +2,7 @@ * Au12x0/Au1550 PSC ALSA ASoC audio support. * * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano@roarinelk.homelinux.net> + * Manuel Lauss <manuel.lauss@gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -10,9 +10,6 @@ * * DMA glue for Au1x-PSC audio. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ @@ -51,8 +48,8 @@ struct au1xpsc_audio_dmadata { struct snd_pcm_substream *substream; unsigned long curr_period; /* current segment DDMA is working on */ unsigned long q_period; /* queue period(s) */ - unsigned long dma_area; /* address of queued DMA area */ - unsigned long dma_area_s; /* start address of DMA area */ + dma_addr_t dma_area; /* address of queued DMA area */ + dma_addr_t dma_area_s; /* start address of DMA area */ unsigned long pos; /* current byte position being played */ unsigned long periods; /* number of SG segments in total */ unsigned long period_bytes; /* size in bytes of one SG segment */ @@ -61,9 +58,6 @@ struct au1xpsc_audio_dmadata { int msbits; }; -/* instance data. There can be only one, MacLeod!!!! */ -static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2]; - /* * These settings are somewhat okay, at least on my machine audio plays * almost skip-free. Especially the 64kB buffer seems to help a LOT. @@ -71,19 +65,10 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2]; #define AU1XPSC_PERIOD_MIN_BYTES 1024 #define AU1XPSC_BUFFER_MIN_BYTES 65536 -#define AU1XPSC_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) - /* PCM hardware DMA capabilities - platform specific */ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED, - .formats = AU1XPSC_PCM_FMTS, + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH, .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES, .period_bytes_max = 4096 * 1024 - 1, .periods_min = 2, @@ -94,8 +79,7 @@ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd) { - au1xxx_dbdma_put_source_flags(cd->ddma_chan, - (void *)phys_to_virt(cd->dma_area), + au1xxx_dbdma_put_source(cd->ddma_chan, cd->dma_area, cd->period_bytes, DDMA_FLAGS_IE); /* update next-to-queue period */ @@ -109,9 +93,8 @@ static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd) static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd) { - au1xxx_dbdma_put_dest_flags(cd->ddma_chan, - (void *)phys_to_virt(cd->dma_area), - cd->period_bytes, DDMA_FLAGS_IE); + au1xxx_dbdma_put_dest(cd->ddma_chan, cd->dma_area, + cd->period_bytes, DDMA_FLAGS_IE); /* update next-to-queue period */ ++cd->q_period; @@ -177,7 +160,7 @@ 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); @@ -201,6 +184,14 @@ out: return 0; } +static inline struct au1xpsc_audio_dmadata *to_dmadata(struct snd_pcm_substream *ss) +{ + struct snd_soc_pcm_runtime *rtd = ss->private_data; + struct au1xpsc_audio_dmadata *pcd = + snd_soc_platform_get_drvdata(rtd->platform); + return &pcd[ss->stream]; +} + static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -212,8 +203,8 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, if (ret < 0) goto out; - stype = SUBSTREAM_TYPE(substream); - pcd = au1xpsc_audio_pcmdma[stype]; + stype = substream->stream; + pcd = to_dmadata(substream); DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " "runtime->min_align %d\n", @@ -233,7 +224,7 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, pcd->substream = substream; pcd->period_bytes = params_period_bytes(params); pcd->periods = params_periods(params); - pcd->dma_area_s = pcd->dma_area = (unsigned long)runtime->dma_addr; + pcd->dma_area_s = pcd->dma_area = runtime->dma_addr; pcd->q_period = 0; pcd->curr_period = 0; pcd->pos = 0; @@ -251,12 +242,11 @@ static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream) static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) { - struct au1xpsc_audio_dmadata *pcd = - au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]; + struct au1xpsc_audio_dmadata *pcd = to_dmadata(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 { @@ -269,7 +259,7 @@ static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - u32 c = au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->ddma_chan; + u32 c = to_dmadata(substream)->ddma_chan; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -289,23 +279,32 @@ static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static snd_pcm_uframes_t au1xpsc_pcm_pointer(struct snd_pcm_substream *substream) { - return bytes_to_frames(substream->runtime, - au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->pos); + return bytes_to_frames(substream->runtime, to_dmadata(substream)->pos); } 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; } static int au1xpsc_pcm_close(struct snd_pcm_substream *substream) { - au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]); + au1x_pcm_dbdma_free(to_dmadata(substream)); return 0; } -struct snd_pcm_ops au1xpsc_pcm_ops = { +static struct snd_pcm_ops au1xpsc_pcm_ops = { .open = au1xpsc_pcm_open, .close = au1xpsc_pcm_close, .ioctl = snd_pcm_lib_ioctl, @@ -321,102 +320,57 @@ static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static int au1xpsc_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1); return 0; } -static int au1xpsc_pcm_probe(struct platform_device *pdev) -{ - struct resource *r; - int ret; - - if (au1xpsc_audio_pcmdma[PCM_TX] || au1xpsc_audio_pcmdma[PCM_RX]) - return -EBUSY; - - /* TX DMA */ - au1xpsc_audio_pcmdma[PCM_TX] - = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); - if (!au1xpsc_audio_pcmdma[PCM_TX]) - return -ENOMEM; +/* au1xpsc audio platform */ +static struct snd_soc_platform_driver au1xpsc_soc_platform = { + .ops = &au1xpsc_pcm_ops, + .pcm_new = au1xpsc_pcm_new, + .pcm_free = au1xpsc_pcm_free_dma_buffers, +}; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) { - ret = -ENODEV; - goto out1; - } - (au1xpsc_audio_pcmdma[PCM_TX])->ddma_id = r->start; +static int au1xpsc_pcm_drvprobe(struct platform_device *pdev) +{ + struct au1xpsc_audio_dmadata *dmadata; - /* RX DMA */ - au1xpsc_audio_pcmdma[PCM_RX] - = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); - if (!au1xpsc_audio_pcmdma[PCM_RX]) + dmadata = devm_kzalloc(&pdev->dev, + 2 * sizeof(struct au1xpsc_audio_dmadata), + GFP_KERNEL); + if (!dmadata) return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) { - ret = -ENODEV; - goto out2; - } - (au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start; + platform_set_drvdata(pdev, dmadata); - return 0; - -out2: - kfree(au1xpsc_audio_pcmdma[PCM_RX]); - au1xpsc_audio_pcmdma[PCM_RX] = NULL; -out1: - kfree(au1xpsc_audio_pcmdma[PCM_TX]); - au1xpsc_audio_pcmdma[PCM_TX] = NULL; - return ret; + return snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform); } -static int au1xpsc_pcm_remove(struct platform_device *pdev) +static int au1xpsc_pcm_drvremove(struct platform_device *pdev) { - int i; - - for (i = 0; i < 2; i++) { - if (au1xpsc_audio_pcmdma[i]) { - au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]); - kfree(au1xpsc_audio_pcmdma[i]); - au1xpsc_audio_pcmdma[i] = NULL; - } - } + snd_soc_unregister_platform(&pdev->dev); return 0; } -/* au1xpsc audio platform */ -struct snd_soc_platform au1xpsc_soc_platform = { - .name = "au1xpsc-pcm-dbdma", - .probe = au1xpsc_pcm_probe, - .remove = au1xpsc_pcm_remove, - .pcm_ops = &au1xpsc_pcm_ops, - .pcm_new = au1xpsc_pcm_new, - .pcm_free = au1xpsc_pcm_free_dma_buffers, +static struct platform_driver au1xpsc_pcm_driver = { + .driver = { + .name = "au1xpsc-pcm", + .owner = THIS_MODULE, + }, + .probe = au1xpsc_pcm_drvprobe, + .remove = au1xpsc_pcm_drvremove, }; -EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); - -static int __init au1xpsc_audio_dbdma_init(void) -{ - au1xpsc_audio_pcmdma[PCM_TX] = NULL; - au1xpsc_audio_pcmdma[PCM_RX] = NULL; - return snd_soc_register_platform(&au1xpsc_soc_platform); -} - -static void __exit au1xpsc_audio_dbdma_exit(void) -{ - snd_soc_unregister_platform(&au1xpsc_soc_platform); -} -module_init(au1xpsc_audio_dbdma_init); -module_exit(au1xpsc_audio_dbdma_exit); +module_platform_driver(au1xpsc_pcm_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver"); -MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c new file mode 100644 index 00000000000..e920b60bf6c --- /dev/null +++ b/sound/soc/au1x/dma.c @@ -0,0 +1,344 @@ +/* + * 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" + +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, + .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, 0, + &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; +} + +static 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 alchemy_pcm_drvprobe(struct platform_device *pdev) +{ + struct alchemy_pcm_ctx *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + platform_set_drvdata(pdev, ctx); + + return snd_soc_register_platform(&pdev->dev, &alchemy_pcm_soc_platform); +} + +static int alchemy_pcm_drvremove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver alchemy_pcmdma_driver = { + .driver = { + .name = "alchemy-pcm-dma", + .owner = THIS_MODULE, + }, + .probe = alchemy_pcm_drvprobe, + .remove = alchemy_pcm_drvremove, +}; + +module_platform_driver(alchemy_pcmdma_driver); + +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 00000000000..b3f37f6edbc --- /dev/null +++ b/sound/soc/au1x/i2sc.c @@ -0,0 +1,324 @@ +/* + * 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 const struct snd_soc_component_driver au1xi2s_component = { + .name = "au1xi2s", +}; + +static int au1xi2s_drvprobe(struct platform_device *pdev) +{ + struct resource *iores, *dmares; + struct au1xpsc_audio_data *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) + return -ENODEV; + + if (!devm_request_mem_region(&pdev->dev, iores->start, + resource_size(iores), + pdev->name)) + return -EBUSY; + + ctx->mmio = devm_ioremap_nocache(&pdev->dev, iores->start, + resource_size(iores)); + if (!ctx->mmio) + return -EBUSY; + + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + return -EBUSY; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; + + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) + return -EBUSY; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; + + platform_set_drvdata(pdev, ctx); + + return snd_soc_register_component(&pdev->dev, &au1xi2s_component, + &au1xi2s_dai_driver, 1); +} + +static int au1xi2s_drvremove(struct platform_device *pdev) +{ + struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev); + + snd_soc_unregister_component(&pdev->dev); + + WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */ + + 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 = au1xi2s_drvremove, +}; + +module_platform_driver(au1xi2s_driver); + +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 f0e30aec7f2..986dcec79fa 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano@roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss@gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -10,15 +10,14 @@ * * Au1xxx-PSC AC97 glue. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ #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/suspend.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +28,9 @@ #include "psc.h" +/* how often to retry failed codec register reads/writes */ +#define AC97_RW_RETRIES 5 + #define AC97_DIR \ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) @@ -39,64 +41,113 @@ (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) == 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; +#if 0 + +/* this could theoretically work, but ac97->bus->card->private_data can be NULL + * when snd_ac97_mixer() is called; I don't know if the rest further down the + * chain are always valid either. + */ +static inline struct au1xpsc_audio_data *ac97_to_pscdata(struct snd_ac97 *x) +{ + struct snd_soc_card *c = x->bus->card->private_data; + return snd_soc_dai_get_drvdata(c->rtd->cpu_dai); +} + +#else + +#define ac97_to_pscdata(x) au1xpsc_ac97_workdata + +#endif + /* AC97 controller reads codec register */ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned short data, tmo; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); + unsigned short retry, tmo; + unsigned long data; - au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) - udelay(2); + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), + AC97_CDC(pscdata)); + au_sync(); - if (!tmo) - data = 0xffff; - else - data = au_readl(AC97_CDC(pscdata)) & 0xffff; + tmo = 20; + do { + udelay(21); + if (au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD) + break; + } while (--tmo); - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + data = au_readl(AC97_CDC(pscdata)); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + + if (reg != ((data >> 16) & 0x7f)) + tmo = 1; /* wrong register, try again */ + + } while (--retry && !tmo); - return data; + return retry ? data & 0xffff : 0xffff; } /* AC97 controller writes to codec register */ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned int tmo; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); + unsigned int tmo, retry; - au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), + AC97_CDC(pscdata)); au_sync(); - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + tmo = 20; + do { + udelay(21); + if (au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD) + break; + } while (--tmo); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); } /* AC97 controller asserts a warm reset */ static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); au_writel(PSC_AC97RST_SNC, AC97_RST(pscdata)); au_sync(); @@ -107,8 +158,7 @@ static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97) static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); int i; /* disable PSC during cold reset */ @@ -129,9 +179,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for PSC to indicate it's ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) { printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); @@ -143,34 +193,32 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for AC97 core to become ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops psc_ac97_ops = { .read = au1xpsc_ac97_read, .write = au1xpsc_ac97_write, .reset = au1xpsc_ac97_cold_reset, .warm_reset = au1xpsc_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned long r, stat; - int chans, stype = SUBSTREAM_TYPE(substream); + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); + unsigned long r, ro, stat; + int chans, t, stype = substream->stream; chans = params_channels(params); - r = au_readl(AC97_CFG(pscdata)); + r = ro = au_readl(AC97_CFG(pscdata)); stat = au_readl(AC97_STAT(pscdata)); /* already active? */ @@ -180,16 +228,13 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, (pscdata->rate != params_rate(params))) return -EINVAL; } else { - /* disable AC97 device controller first */ - au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); - au_sync(); /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */ r &= ~PSC_AC97CFG_LEN_MASK; 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); @@ -199,29 +244,64 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_RXSLOT_ENA(4); } - /* finally enable the AC97 controller again */ + /* do we need to poke the hardware? */ + if (!(r ^ ro)) + goto out; + + /* ac97 engine is about to be disabled */ + mutex_lock(&pscdata->lock); + + /* disable AC97 device controller first... */ + au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* ...wait for it... */ + t = 100; + while ((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) && --t) + msleep(1); + + if (!t) + printk(KERN_ERR "PSC-AC97: can't disable!\n"); + + /* ...write config... */ + au_writel(r, AC97_CFG(pscdata)); + au_sync(); + + /* ...enable the AC97 controller again... */ au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); au_sync(); + /* ...and wait for ready bit */ + t = 100; + while ((!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && --t) + msleep(1); + + if (!t) + printk(KERN_ERR "PSC-AC97: can't enable!\n"); + + mutex_unlock(&pscdata->lock); + pscdata->cfg = r; pscdata->rate = params_rate(params); } +out: return 0; } static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - int ret, stype = SUBSTREAM_TYPE(substream); + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); + int ret, stype = substream->stream; ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); au_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); au_sync(); break; @@ -229,6 +309,13 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); au_sync(); + + while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype)) + asm volatile ("nop"); + + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); + break; default: ret = -EINVAL; @@ -236,103 +323,148 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, return ret; } -static int au1xpsc_ac97_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +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 const struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { + .startup = au1xpsc_ac97_startup, + .trigger = au1xpsc_ac97_trigger, + .hw_params = au1xpsc_ac97_hw_params, +}; + +static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = { + .ac97_control = 1, + .probe = au1xpsc_ac97_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 = &au1xpsc_ac97_dai_ops, +}; + +static const struct snd_soc_component_driver au1xpsc_ac97_component = { + .name = "au1xpsc-ac97", +}; + +static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *iores, *dmares; unsigned long sel; + struct au1xpsc_audio_data *wd; - if (au1xpsc_ac97_workdata) - return -EBUSY; - - au1xpsc_ac97_workdata = - kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); - if (!au1xpsc_ac97_workdata) + wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data), + GFP_KERNEL); + if (!wd) return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { - ret = -ENODEV; - goto out0; - } + mutex_init(&wd->lock); + + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + wd->mmio = devm_ioremap_resource(&pdev->dev, iores); + if (IS_ERR(wd->mmio)) + return PTR_ERR(wd->mmio); - ret = -EBUSY; - au1xpsc_ac97_workdata->ioarea = - request_mem_region(r->start, r->end - r->start + 1, - "au1xpsc_ac97"); - if (!au1xpsc_ac97_workdata->ioarea) - goto out0; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + return -EBUSY; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; - au1xpsc_ac97_workdata->mmio = ioremap(r->start, 0xffff); - if (!au1xpsc_ac97_workdata->mmio) - goto out1; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) + return -EBUSY; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; /* configuration: max dma trigger threshold, enable ac97 */ - au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | - PSC_AC97CFG_TT_FIFO8 | - PSC_AC97CFG_DE_ENABLE; + wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; - /* preserve PSC clock source set up by platform (dev.platform_data - * is already occupied by soc layer) - */ - sel = au_readl(PSC_SEL(au1xpsc_ac97_workdata)) & PSC_SEL_CLK_MASK; - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + /* preserve PSC clock source set up by platform */ + sel = au_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK; + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - au_writel(0, PSC_SEL(au1xpsc_ac97_workdata)); + au_writel(0, PSC_SEL(wd)); au_sync(); - au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(au1xpsc_ac97_workdata)); + au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(wd)); au_sync(); - /* next up: cold reset. Dont check for PSC-ready now since - * there may not be any codec clock yet. - */ - return 0; + /* name the DAI like this device instance ("au1xpsc-ac97.PSCINDEX") */ + memcpy(&wd->dai_drv, &au1xpsc_ac97_dai_template, + sizeof(struct snd_soc_dai_driver)); + wd->dai_drv.name = dev_name(&pdev->dev); -out1: - release_resource(au1xpsc_ac97_workdata->ioarea); - kfree(au1xpsc_ac97_workdata->ioarea); -out0: - kfree(au1xpsc_ac97_workdata); - au1xpsc_ac97_workdata = NULL; - return ret; + platform_set_drvdata(pdev, wd); + + ret = snd_soc_set_ac97_ops(&psc_ac97_ops); + if (ret) + return ret; + + ret = snd_soc_register_component(&pdev->dev, &au1xpsc_ac97_component, + &wd->dai_drv, 1); + if (ret) + return ret; + + au1xpsc_ac97_workdata = wd; + return 0; } -static void au1xpsc_ac97_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int au1xpsc_ac97_drvremove(struct platform_device *pdev) { + struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); + + snd_soc_unregister_component(&pdev->dev); + /* disable PSC completely */ - au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_writel(0, AC97_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - iounmap(au1xpsc_ac97_workdata->mmio); - release_resource(au1xpsc_ac97_workdata->ioarea); - kfree(au1xpsc_ac97_workdata->ioarea); - kfree(au1xpsc_ac97_workdata); - au1xpsc_ac97_workdata = NULL; + au1xpsc_ac97_workdata = NULL; /* MDEV */ + + return 0; } -static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai) +#ifdef CONFIG_PM +static int au1xpsc_ac97_drvsuspend(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* save interesting registers and disable PSC */ - au1xpsc_ac97_workdata->pm[0] = - au_readl(PSC_SEL(au1xpsc_ac97_workdata)); + wd->pm[0] = au_readl(PSC_SEL(wd)); - au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_writel(0, AC97_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); return 0; } -static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) +static int au1xpsc_ac97_drvresume(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* restore PSC clock config */ - au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE, - PSC_SEL(au1xpsc_ac97_workdata)); + au_writel(wd->pm[0] | PSC_SEL_PS_AC97MODE, PSC_SEL(wd)); au_sync(); /* after this point the ac97 core will cold-reset the codec. @@ -342,46 +474,32 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) return 0; } -struct snd_soc_dai au1xpsc_ac97_dai = { - .name = "au1xpsc_ac97", - .ac97_control = 1, - .probe = au1xpsc_ac97_probe, - .remove = au1xpsc_ac97_remove, - .suspend = au1xpsc_ac97_suspend, - .resume = au1xpsc_ac97_resume, - .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 = { - .trigger = au1xpsc_ac97_trigger, - .hw_params = au1xpsc_ac97_hw_params, - }, +static struct dev_pm_ops au1xpscac97_pmops = { + .suspend = au1xpsc_ac97_drvsuspend, + .resume = au1xpsc_ac97_drvresume, }; -EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); -static int __init au1xpsc_ac97_init(void) -{ - au1xpsc_ac97_workdata = NULL; - return snd_soc_register_dai(&au1xpsc_ac97_dai); -} +#define AU1XPSCAC97_PMOPS &au1xpscac97_pmops -static void __exit au1xpsc_ac97_exit(void) -{ - snd_soc_unregister_dai(&au1xpsc_ac97_dai); -} +#else + +#define AU1XPSCAC97_PMOPS NULL -module_init(au1xpsc_ac97_init); -module_exit(au1xpsc_ac97_exit); +#endif + +static struct platform_driver au1xpsc_ac97_driver = { + .driver = { + .name = "au1xpsc_ac97", + .owner = THIS_MODULE, + .pm = AU1XPSCAC97_PMOPS, + }, + .probe = au1xpsc_ac97_drvprobe, + .remove = au1xpsc_ac97_drvremove, +}; + +module_platform_driver(au1xpsc_ac97_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); + diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index f916de4400e..fe923a7bdc3 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -2,7 +2,7 @@ * Au12x0/Au1550 PSC ALSA ASoC audio support. * * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano@roarinelk.homelinux.net> + * Manuel Lauss <manuel.lauss@gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -10,14 +10,12 @@ * * Au1xxx-PSC I2S glue. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. * NOTE: so far only PSC slave mode (bit- and frameclock) is supported. */ #include <linux/init.h> #include <linux/module.h> +#include <linux/slab.h> #include <linux/suspend.h> #include <sound/core.h> #include <sound/pcm.h> @@ -44,22 +42,19 @@ (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) -/* instance data. There can be only one, MacLeod!!!! */ -static struct au1xpsc_audio_data *au1xpsc_i2s_workdata; - static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(cpu_dai); unsigned long ct; int ret; @@ -119,7 +114,7 @@ static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); int cfgbits; unsigned long stat; @@ -244,8 +239,8 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; - int ret, stype = SUBSTREAM_TYPE(substream); + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); + int ret, stype = substream->stream; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -262,151 +257,177 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int au1xpsc_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +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 const 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, +}; + +static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = { + .playback = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .capture = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .ops = &au1xpsc_i2s_dai_ops, +}; + +static const struct snd_soc_component_driver au1xpsc_i2s_component = { + .name = "au1xpsc-i2s", +}; + +static int au1xpsc_i2s_drvprobe(struct platform_device *pdev) { - struct resource *r; + struct resource *iores, *dmares; unsigned long sel; int ret; + struct au1xpsc_audio_data *wd; - if (au1xpsc_i2s_workdata) - return -EBUSY; - - au1xpsc_i2s_workdata = - kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); - if (!au1xpsc_i2s_workdata) + wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data), + GFP_KERNEL); + if (!wd) return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { - ret = -ENODEV; - goto out0; - } + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) + return -ENODEV; ret = -EBUSY; - au1xpsc_i2s_workdata->ioarea = - request_mem_region(r->start, r->end - r->start + 1, - "au1xpsc_i2s"); - if (!au1xpsc_i2s_workdata->ioarea) - goto out0; + if (!devm_request_mem_region(&pdev->dev, iores->start, + resource_size(iores), + pdev->name)) + return -EBUSY; + + wd->mmio = devm_ioremap(&pdev->dev, iores->start, + resource_size(iores)); + if (!wd->mmio) + return -EBUSY; + + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + return -EBUSY; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; - au1xpsc_i2s_workdata->mmio = ioremap(r->start, 0xffff); - if (!au1xpsc_i2s_workdata->mmio) - goto out1; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) + return -EBUSY; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) */ - sel = au_readl(PSC_SEL(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK; - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + sel = au_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK; + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(au1xpsc_i2s_workdata)); - au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(wd)); + au_writel(0, I2S_CFG(wd)); au_sync(); /* preconfigure: set max rx/tx fifo depths */ - au1xpsc_i2s_workdata->cfg |= - PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8; + wd->cfg |= PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8; /* don't wait for I2S core to become ready now; clocks may not * be running yet; depending on clock input for PSC a wait might * time out. */ - return 0; + /* name the DAI like this device instance ("au1xpsc-i2s.PSCINDEX") */ + memcpy(&wd->dai_drv, &au1xpsc_i2s_dai_template, + sizeof(struct snd_soc_dai_driver)); + wd->dai_drv.name = dev_name(&pdev->dev); -out1: - release_resource(au1xpsc_i2s_workdata->ioarea); - kfree(au1xpsc_i2s_workdata->ioarea); -out0: - kfree(au1xpsc_i2s_workdata); - au1xpsc_i2s_workdata = NULL; - return ret; + platform_set_drvdata(pdev, wd); + + return snd_soc_register_component(&pdev->dev, &au1xpsc_i2s_component, + &wd->dai_drv, 1); } -static void au1xpsc_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int au1xpsc_i2s_drvremove(struct platform_device *pdev) { - au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); + + snd_soc_unregister_component(&pdev->dev); + + au_writel(0, I2S_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - iounmap(au1xpsc_i2s_workdata->mmio); - release_resource(au1xpsc_i2s_workdata->ioarea); - kfree(au1xpsc_i2s_workdata->ioarea); - kfree(au1xpsc_i2s_workdata); - au1xpsc_i2s_workdata = NULL; + return 0; } -static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai) +#ifdef CONFIG_PM +static int au1xpsc_i2s_drvsuspend(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* save interesting register and disable PSC */ - au1xpsc_i2s_workdata->pm[0] = - au_readl(PSC_SEL(au1xpsc_i2s_workdata)); + wd->pm[0] = au_readl(PSC_SEL(wd)); - au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_writel(0, I2S_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); return 0; } -static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) +static int au1xpsc_i2s_drvresume(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* select I2S mode and PSC clock */ - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - au_writel(0, PSC_SEL(au1xpsc_i2s_workdata)); + au_writel(0, PSC_SEL(wd)); au_sync(); - au_writel(au1xpsc_i2s_workdata->pm[0], - PSC_SEL(au1xpsc_i2s_workdata)); + au_writel(wd->pm[0], PSC_SEL(wd)); au_sync(); return 0; } -struct snd_soc_dai au1xpsc_i2s_dai = { - .name = "au1xpsc_i2s", - .probe = au1xpsc_i2s_probe, - .remove = au1xpsc_i2s_remove, - .suspend = au1xpsc_i2s_suspend, - .resume = au1xpsc_i2s_resume, - .playback = { - .rates = AU1XPSC_I2S_RATES, - .formats = AU1XPSC_I2S_FMTS, - .channels_min = 2, - .channels_max = 8, /* 2 without external help */ - }, - .capture = { - .rates = AU1XPSC_I2S_RATES, - .formats = AU1XPSC_I2S_FMTS, - .channels_min = 2, - .channels_max = 8, /* 2 without external help */ - }, - .ops = { - .trigger = au1xpsc_i2s_trigger, - .hw_params = au1xpsc_i2s_hw_params, - .set_fmt = au1xpsc_i2s_set_fmt, - }, +static struct dev_pm_ops au1xpsci2s_pmops = { + .suspend = au1xpsc_i2s_drvsuspend, + .resume = au1xpsc_i2s_drvresume, }; -EXPORT_SYMBOL(au1xpsc_i2s_dai); -static int __init au1xpsc_i2s_init(void) -{ - au1xpsc_i2s_workdata = NULL; - return snd_soc_register_dai(&au1xpsc_i2s_dai); -} +#define AU1XPSCI2S_PMOPS &au1xpsci2s_pmops -static void __exit au1xpsc_i2s_exit(void) -{ - snd_soc_unregister_dai(&au1xpsc_i2s_dai); -} +#else + +#define AU1XPSCI2S_PMOPS NULL + +#endif + +static struct platform_driver au1xpsc_i2s_driver = { + .driver = { + .name = "au1xpsc_i2s", + .owner = THIS_MODULE, + .pm = AU1XPSCI2S_PMOPS, + }, + .probe = au1xpsc_i2s_drvprobe, + .remove = au1xpsc_i2s_drvremove, +}; -module_init(au1xpsc_i2s_init); -module_exit(au1xpsc_i2s_exit); +module_platform_driver(au1xpsc_i2s_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 8fdb1a04a07..b16b2e02e0c 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -1,42 +1,31 @@ /* - * Au12x0/Au1550 PSC ALSA ASoC audio support. + * Alchemy ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano@roarinelk.homelinux.net> + * (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 * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ #ifndef _AU1X_PCM_H #define _AU1X_PCM_H -extern struct snd_soc_dai au1xpsc_ac97_dai; -extern struct snd_soc_dai au1xpsc_i2s_dai; -extern struct snd_soc_platform au1xpsc_soc_platform; -extern struct snd_ac97_bus_ops soc_ac97_ops; - struct au1xpsc_audio_data { void __iomem *mmio; unsigned long cfg; unsigned long rate; + struct snd_soc_dai_driver dai_drv; + unsigned long pm[2]; - struct resource *ioarea; + struct mutex lock; + 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/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c deleted file mode 100644 index 27683eb7905..00000000000 --- a/sound/soc/au1x/sample-ac97.c +++ /dev/null @@ -1,144 +0,0 @@ -/* - * Sample Au12x0/Au1550 PSC AC97 sound machine. - * - * Copyright (c) 2007-2008 Manuel Lauss <mano@roarinelk.homelinux.net> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms outlined in the file COPYING at the root of this - * source archive. - * - * This is a very generic AC97 sound machine driver for boards which - * have (AC97) audio at PSC1 (e.g. DB1200 demoboards). - */ - -#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 <sound/soc-dapm.h> -#include <asm/mach-au1x00/au1000.h> -#include <asm/mach-au1x00/au1xxx_psc.h> -#include <asm/mach-au1x00/au1xxx_dbdma.h> - -#include "../codecs/ac97.h" -#include "psc.h" - -static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec) -{ - snd_soc_dapm_sync(codec); - return 0; -} - -static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = { - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai = &au1xpsc_ac97_dai, /* see psc-ac97.c */ - .codec_dai = &ac97_dai, /* see codecs/ac97.c */ - .init = au1xpsc_sample_ac97_init, - .ops = NULL, -}; - -static struct snd_soc_card au1xpsc_sample_ac97_machine = { - .name = "Au1xxx PSC AC97 Audio", - .dai_link = &au1xpsc_sample_ac97_dai, - .num_links = 1, -}; - -static struct snd_soc_device au1xpsc_sample_ac97_devdata = { - .card = &au1xpsc_sample_ac97_machine, - .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ - .codec_dev = &soc_codec_dev_ac97, -}; - -static struct resource au1xpsc_psc1_res[] = { - [0] = { - .start = CPHYSADDR(PSC1_BASE_ADDR), - .end = CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff, - .flags = IORESOURCE_MEM, - }, - [1] = { -#ifdef CONFIG_SOC_AU1200 - .start = AU1200_PSC1_INT, - .end = AU1200_PSC1_INT, -#elif defined(CONFIG_SOC_AU1550) - .start = AU1550_PSC1_INT, - .end = AU1550_PSC1_INT, -#endif - .flags = IORESOURCE_IRQ, - }, - [2] = { - .start = DSCR_CMD0_PSC1_TX, - .end = DSCR_CMD0_PSC1_TX, - .flags = IORESOURCE_DMA, - }, - [3] = { - .start = DSCR_CMD0_PSC1_RX, - .end = DSCR_CMD0_PSC1_RX, - .flags = IORESOURCE_DMA, - }, -}; - -static struct platform_device *au1xpsc_sample_ac97_dev; - -static int __init au1xpsc_sample_ac97_load(void) -{ - int ret; - -#ifdef CONFIG_SOC_AU1200 - unsigned long io; - - /* modify sys_pinfunc for AC97 on PSC1 */ - io = au_readl(SYS_PINFUNC); - io |= SYS_PINFUNC_P1C; - io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B); - au_writel(io, SYS_PINFUNC); - au_sync(); -#endif - - ret = -ENOMEM; - - /* setup PSC clock source for AC97 part: external clock provided - * by codec. The psc-ac97.c driver depends on this setting! - */ - au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET); - au_sync(); - - au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1); - if (!au1xpsc_sample_ac97_dev) - goto out; - - au1xpsc_sample_ac97_dev->resource = - kmemdup(au1xpsc_psc1_res, sizeof(struct resource) * - ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL); - au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res); - au1xpsc_sample_ac97_dev->id = 1; - - platform_set_drvdata(au1xpsc_sample_ac97_dev, - &au1xpsc_sample_ac97_devdata); - au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev; - ret = platform_device_add(au1xpsc_sample_ac97_dev); - - if (ret) { - platform_device_put(au1xpsc_sample_ac97_dev); - au1xpsc_sample_ac97_dev = NULL; - } - -out: - return ret; -} - -static void __exit au1xpsc_sample_ac97_exit(void) -{ - platform_device_unregister(au1xpsc_sample_ac97_dev); -} - -module_init(au1xpsc_sample_ac97_load); -module_exit(au1xpsc_sample_ac97_exit); - -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine"); -MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); |