diff options
Diffstat (limited to 'sound/soc/au1x')
| -rw-r--r-- | sound/soc/au1x/Kconfig | 64 | ||||
| -rw-r--r-- | sound/soc/au1x/Makefile | 24 | ||||
| -rw-r--r-- | sound/soc/au1x/ac97c.c | 347 | ||||
| -rw-r--r-- | sound/soc/au1x/db1000.c | 56 | ||||
| -rw-r--r-- | sound/soc/au1x/db1200.c | 197 | ||||
| -rw-r--r-- | sound/soc/au1x/dbdma2.c | 365 | ||||
| -rw-r--r-- | sound/soc/au1x/dma.c | 334 | ||||
| -rw-r--r-- | sound/soc/au1x/i2sc.c | 323 | ||||
| -rw-r--r-- | sound/soc/au1x/psc-ac97.c | 504 | ||||
| -rw-r--r-- | sound/soc/au1x/psc-i2s.c | 421 | ||||
| -rw-r--r-- | sound/soc/au1x/psc.h | 42 |
11 files changed, 2677 insertions, 0 deletions
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig new file mode 100644 index 000000000..a56104040 --- /dev/null +++ b/sound/soc/au1x/Kconfig @@ -0,0 +1,64 @@ +## +## Au1200/Au1550/Au1300 PSC + DBDMA +## +config SND_SOC_AU1XPSC + 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 Au12xx/Au13xx/Au1550 SoC. + +config SND_SOC_AU1XPSC_I2S + tristate + +config SND_SOC_AU1XPSC_AC97 + tristate + select AC97_BUS + 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/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 + 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 new file mode 100644 index 000000000..33183d7fe --- /dev/null +++ b/sound/soc/au1x/Makefile @@ -0,0 +1,24 @@ +# SPDX-License-Identifier: GPL-2.0 +# Au1200/Au1550 PSC audio +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 000000000..66d6c52e7 --- /dev/null +++ b/sound/soc/au1x/ac97c.c @@ -0,0 +1,347 @@ +/* + * 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 = 6; + 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", + .bus_control = true, + .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", + .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 000000000..e97c32798 --- /dev/null +++ b/sound/soc/au1x/db1000.c @@ -0,0 +1,56 @@ +/* + * 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 devm_snd_soc_register_card(&pdev->dev, card); +} + +static struct platform_driver db1000_audio_driver = { + .driver = { + .name = "db1000-audio", + .pm = &snd_soc_pm_ops, + }, + .probe = db1000_audio_probe, +}; + +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 000000000..301e1fc9a --- /dev/null +++ b/sound/soc/au1x/db1200.c @@ -0,0 +1,197 @@ +/* + * 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 const 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; + + /* WM8731 has its own 12MHz crystal */ + snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, + 12000000, SND_SOC_CLOCK_IN); + + return 0; +} + +static const 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", + .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .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", + .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .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", + .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .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 devm_snd_soc_register_card(&pdev->dev, card); +} + +static struct platform_driver db1200_audio_driver = { + .driver = { + .name = "db1200-ac97", + .pm = &snd_soc_pm_ops, + }, + .id_table = db1200_pids, + .probe = db1200_audio_probe, +}; + +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 new file mode 100644 index 000000000..a906560d0 --- /dev/null +++ b/sound/soc/au1x/dbdma2.c @@ -0,0 +1,365 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (c) 2007-2008 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. + * + * DMA glue for Au1x-PSC audio. + * + */ + + +#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/au1xxx_dbdma.h> +#include <asm/mach-au1x00/au1xxx_psc.h> + +#include "psc.h" + +/*#define PCM_DEBUG*/ + +#define DRV_NAME "dbdma2" + +#define MSG(x...) printk(KERN_INFO "au1xpsc_pcm: " x) +#ifdef PCM_DEBUG +#define DBG MSG +#else +#define DBG(x...) do {} while (0) +#endif + +struct au1xpsc_audio_dmadata { + /* DDMA control data */ + unsigned int ddma_id; /* DDMA direction ID for this PSC */ + u32 ddma_chan; /* DDMA context */ + + /* PCM context (for irq handlers) */ + struct snd_pcm_substream *substream; + unsigned long curr_period; /* current segment DDMA is working on */ + unsigned long q_period; /* queue period(s) */ + 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 */ + + /* runtime data */ + int msbits; +}; + +/* + * These settings are somewhat okay, at least on my machine audio plays + * almost skip-free. Especially the 64kB buffer seems to help a LOT. + */ +#define AU1XPSC_PERIOD_MIN_BYTES 1024 +#define AU1XPSC_BUFFER_MIN_BYTES 65536 + +/* 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 | SNDRV_PCM_INFO_BATCH, + .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES, + .period_bytes_max = 4096 * 1024 - 1, + .periods_min = 2, + .periods_max = 4096, /* 2 to as-much-as-you-like */ + .buffer_bytes_max = 4096 * 1024 - 1, + .fifo_size = 16, /* fifo entries of AC97/I2S PSC */ +}; + +static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd) +{ + au1xxx_dbdma_put_source(cd->ddma_chan, cd->dma_area, + cd->period_bytes, DDMA_FLAGS_IE); + + /* update next-to-queue period */ + ++cd->q_period; + cd->dma_area += cd->period_bytes; + if (cd->q_period >= cd->periods) { + cd->q_period = 0; + cd->dma_area = cd->dma_area_s; + } +} + +static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd) +{ + au1xxx_dbdma_put_dest(cd->ddma_chan, cd->dma_area, + cd->period_bytes, DDMA_FLAGS_IE); + + /* update next-to-queue period */ + ++cd->q_period; + cd->dma_area += cd->period_bytes; + if (cd->q_period >= cd->periods) { + cd->q_period = 0; + cd->dma_area = cd->dma_area_s; + } +} + +static void au1x_pcm_dmatx_cb(int irq, void *dev_id) +{ + struct au1xpsc_audio_dmadata *cd = dev_id; + + cd->pos += cd->period_bytes; + if (++cd->curr_period >= cd->periods) { + cd->pos = 0; + cd->curr_period = 0; + } + snd_pcm_period_elapsed(cd->substream); + au1x_pcm_queue_tx(cd); +} + +static void au1x_pcm_dmarx_cb(int irq, void *dev_id) +{ + struct au1xpsc_audio_dmadata *cd = dev_id; + + cd->pos += cd->period_bytes; + if (++cd->curr_period >= cd->periods) { + cd->pos = 0; + cd->curr_period = 0; + } + snd_pcm_period_elapsed(cd->substream); + au1x_pcm_queue_rx(cd); +} + +static void au1x_pcm_dbdma_free(struct au1xpsc_audio_dmadata *pcd) +{ + if (pcd->ddma_chan) { + au1xxx_dbdma_stop(pcd->ddma_chan); + au1xxx_dbdma_reset(pcd->ddma_chan); + au1xxx_dbdma_chan_free(pcd->ddma_chan); + pcd->ddma_chan = 0; + pcd->msbits = 0; + } +} + +/* in case of missing DMA ring or changed TX-source / RX-dest bit widths, + * allocate (or reallocate) a 2-descriptor DMA ring with bit depth according + * to ALSA-supplied sample depth. This is due to limitations in the dbdma api + * (cannot adjust source/dest widths of already allocated descriptor ring). + */ +static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd, + int stype, int msbits) +{ + /* DMA only in 8/16/32 bit widths */ + if (msbits == 24) + msbits = 32; + + /* check current config: correct bits and descriptors allocated? */ + if ((pcd->ddma_chan) && (msbits == pcd->msbits)) + goto out; /* all ok! */ + + au1x_pcm_dbdma_free(pcd); + + 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); + else + pcd->ddma_chan = au1xxx_dbdma_chan_alloc(DSCR_CMD0_ALWAYS, + pcd->ddma_id, + au1x_pcm_dmatx_cb, (void *)pcd); + + if (!pcd->ddma_chan) + return -ENOMEM; + + au1xxx_dbdma_set_devwidth(pcd->ddma_chan, msbits); + au1xxx_dbdma_ring_alloc(pcd->ddma_chan, 2); + + pcd->msbits = msbits; + + au1xxx_dbdma_stop(pcd->ddma_chan); + au1xxx_dbdma_reset(pcd->ddma_chan); + +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 snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct au1xpsc_audio_dmadata *pcd = snd_soc_component_get_drvdata(component); + return &pcd[ss->stream]; +} + +static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct au1xpsc_audio_dmadata *pcd; + int stype, ret; + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) + goto out; + + stype = substream->stream; + pcd = to_dmadata(substream); + + DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %zu " + "runtime->min_align %lu\n", + (unsigned long)runtime->dma_area, + (unsigned long)runtime->dma_addr, runtime->dma_bytes, + runtime->min_align); + + DBG("bits %d frags %d frag_bytes %d is_rx %d\n", params->msbits, + params_periods(params), params_period_bytes(params), stype); + + ret = au1x_pcm_dbdma_realloc(pcd, stype, params->msbits); + if (ret) { + MSG("DDMA channel (re)alloc failed!\n"); + goto out; + } + + pcd->substream = substream; + pcd->period_bytes = params_period_bytes(params); + pcd->periods = params_periods(params); + pcd->dma_area_s = pcd->dma_area = runtime->dma_addr; + pcd->q_period = 0; + pcd->curr_period = 0; + pcd->pos = 0; + + ret = 0; +out: + return ret; +} + +static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_lib_free_pages(substream); + return 0; +} + +static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); + + au1xxx_dbdma_reset(pcd->ddma_chan); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + au1x_pcm_queue_rx(pcd); + au1x_pcm_queue_rx(pcd); + } else { + au1x_pcm_queue_tx(pcd); + au1x_pcm_queue_tx(pcd); + } + + return 0; +} + +static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + u32 c = to_dmadata(substream)->ddma_chan; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + au1xxx_dbdma_start(c); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + au1xxx_dbdma_stop(c); + break; + default: + return -EINVAL; + } + return 0; +} + +static snd_pcm_uframes_t +au1xpsc_pcm_pointer(struct snd_pcm_substream *substream) +{ + 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(to_dmadata(substream)); + return 0; +} + +static const struct snd_pcm_ops au1xpsc_pcm_ops = { + .open = au1xpsc_pcm_open, + .close = au1xpsc_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = au1xpsc_pcm_hw_params, + .hw_free = au1xpsc_pcm_hw_free, + .prepare = au1xpsc_pcm_prepare, + .trigger = au1xpsc_pcm_trigger, + .pointer = au1xpsc_pcm_pointer, +}; + +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; +} + +/* au1xpsc audio platform */ +static struct snd_soc_component_driver au1xpsc_soc_component = { + .name = DRV_NAME, + .ops = &au1xpsc_pcm_ops, + .pcm_new = au1xpsc_pcm_new, +}; + +static int au1xpsc_pcm_drvprobe(struct platform_device *pdev) +{ + struct au1xpsc_audio_dmadata *dmadata; + + dmadata = devm_kcalloc(&pdev->dev, + 2, sizeof(struct au1xpsc_audio_dmadata), + GFP_KERNEL); + if (!dmadata) + return -ENOMEM; + + platform_set_drvdata(pdev, dmadata); + + return devm_snd_soc_register_component(&pdev->dev, + &au1xpsc_soc_component, NULL, 0); +} + +static struct platform_driver au1xpsc_pcm_driver = { + .driver = { + .name = "au1xpsc-pcm", + }, + .probe = au1xpsc_pcm_drvprobe, +}; + +module_platform_driver(au1xpsc_pcm_driver); + +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 000000000..efff1e293 --- /dev/null +++ b/sound/soc/au1x/dma.c @@ -0,0 +1,334 @@ +/* + * 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 DRV_NAME "au1x_dma" + +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; + struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME); + return snd_soc_component_get_drvdata(component); +} + +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 const 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 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_component_driver alchemy_pcm_soc_component = { + .name = DRV_NAME, + .ops = &alchemy_pcm_ops, + .pcm_new = alchemy_pcm_new, +}; + +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 devm_snd_soc_register_component(&pdev->dev, + &alchemy_pcm_soc_component, NULL, 0); +} + +static struct platform_driver alchemy_pcmdma_driver = { + .driver = { + .name = "alchemy-pcm-dma", + }, + .probe = alchemy_pcm_drvprobe, +}; + +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 000000000..450c842c7 --- /dev/null +++ b/sound/soc/au1x/i2sc.c @@ -0,0 +1,323 @@ +/* + * 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", + .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 new file mode 100644 index 000000000..a2050ae5a --- /dev/null +++ b/sound/soc/au1x/psc-ac97.c @@ -0,0 +1,504 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (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 + * published by the Free Software Foundation. + * + * Au1xxx-PSC AC97 glue. + * + */ + +#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> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/mach-au1x00/au1000.h> +#include <asm/mach-au1x00/au1xxx_psc.h> + +#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) + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_48000 + +#define AC97_FMTS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE) + +#define AC97PCR_START(stype) \ + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TS : PSC_AC97PCR_RS) +#define AC97PCR_STOP(stype) \ + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TP : PSC_AC97PCR_RP) +#define AC97PCR_CLRFIFO(stype) \ + ((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) +{ + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); + unsigned short retry, tmo; + unsigned long data; + + __raw_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + wmb(); /* drain writebuffer */ + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + __raw_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), + AC97_CDC(pscdata)); + wmb(); /* drain writebuffer */ + + tmo = 20; + do { + udelay(21); + if (__raw_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD) + break; + } while (--tmo); + + data = __raw_readl(AC97_CDC(pscdata)); + + __raw_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + wmb(); /* drain writebuffer */ + + mutex_unlock(&pscdata->lock); + + if (reg != ((data >> 16) & 0x7f)) + tmo = 1; /* wrong register, try again */ + + } while (--retry && !tmo); + + 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) +{ + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); + unsigned int tmo, retry; + + __raw_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + wmb(); /* drain writebuffer */ + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + __raw_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), + AC97_CDC(pscdata)); + wmb(); /* drain writebuffer */ + + tmo = 20; + do { + udelay(21); + if (__raw_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD) + break; + } while (--tmo); + + __raw_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + wmb(); /* drain writebuffer */ + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); +} + +/* AC97 controller asserts a warm reset */ +static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); + + __raw_writel(PSC_AC97RST_SNC, AC97_RST(pscdata)); + wmb(); /* drain writebuffer */ + msleep(10); + __raw_writel(0, AC97_RST(pscdata)); + wmb(); /* drain writebuffer */ +} + +static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); + int i; + + /* disable PSC during cold reset */ + __raw_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + wmb(); /* drain writebuffer */ + __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(pscdata)); + wmb(); /* drain writebuffer */ + + /* issue cold reset */ + __raw_writel(PSC_AC97RST_RST, AC97_RST(pscdata)); + wmb(); /* drain writebuffer */ + msleep(500); + __raw_writel(0, AC97_RST(pscdata)); + wmb(); /* drain writebuffer */ + + /* enable PSC */ + __raw_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata)); + wmb(); /* drain writebuffer */ + + /* wait for PSC to indicate it's ready */ + i = 1000; + while (!((__raw_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) + msleep(1); + + if (i == 0) { + printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); + return; + } + + /* enable the ac97 function */ + __raw_writel(pscdata->cfg | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + wmb(); /* drain writebuffer */ + + /* wait for AC97 core to become ready */ + i = 1000; + while (!((__raw_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) + msleep(1); + if (i == 0) + printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); +} + +/* AC97 controller operations */ +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, +}; + +static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + 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 = ro = __raw_readl(AC97_CFG(pscdata)); + stat = __raw_readl(AC97_STAT(pscdata)); + + /* already active? */ + if (stat & (PSC_AC97STAT_TB | PSC_AC97STAT_RB)) { + /* reject parameters not currently set up */ + if ((PSC_AC97CFG_GET_LEN(r) != params->msbits) || + (pscdata->rate != params_rate(params))) + return -EINVAL; + } else { + + /* 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 == SNDRV_PCM_STREAM_PLAYBACK) { + r &= ~PSC_AC97CFG_TXSLOT_MASK; + r |= PSC_AC97CFG_TXSLOT_ENA(3); + r |= PSC_AC97CFG_TXSLOT_ENA(4); + } else { + r &= ~PSC_AC97CFG_RXSLOT_MASK; + r |= PSC_AC97CFG_RXSLOT_ENA(3); + r |= PSC_AC97CFG_RXSLOT_ENA(4); + } + + /* 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... */ + __raw_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + wmb(); /* drain writebuffer */ + + /* ...wait for it... */ + t = 100; + while ((__raw_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) && --t) + msleep(1); + + if (!t) + printk(KERN_ERR "PSC-AC97: can't disable!\n"); + + /* ...write config... */ + __raw_writel(r, AC97_CFG(pscdata)); + wmb(); /* drain writebuffer */ + + /* ...enable the AC97 controller again... */ + __raw_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + wmb(); /* drain writebuffer */ + + /* ...and wait for ready bit */ + t = 100; + while ((!(__raw_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) +{ + 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: + __raw_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + wmb(); /* drain writebuffer */ + __raw_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); + wmb(); /* drain writebuffer */ + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + __raw_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); + wmb(); /* drain writebuffer */ + + while (__raw_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype)) + asm volatile ("nop"); + + __raw_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + wmb(); /* drain writebuffer */ + + break; + default: + ret = -EINVAL; + } + 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 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 = { + .bus_control = true, + .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 *iores, *dmares; + unsigned long sel; + struct au1xpsc_audio_data *wd; + + wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data), + GFP_KERNEL); + if (!wd) + return -ENOMEM; + + 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); + + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + return -EBUSY; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; + + 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 */ + wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; + + /* preserve PSC clock source set up by platform */ + sel = __raw_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK; + __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); + wmb(); /* drain writebuffer */ + __raw_writel(0, PSC_SEL(wd)); + wmb(); /* drain writebuffer */ + __raw_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(wd)); + wmb(); /* drain writebuffer */ + + /* 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); + + 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 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 */ + __raw_writel(0, AC97_CFG(wd)); + wmb(); /* drain writebuffer */ + __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); + wmb(); /* drain writebuffer */ + + au1xpsc_ac97_workdata = NULL; /* MDEV */ + + return 0; +} + +#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 */ + wd->pm[0] = __raw_readl(PSC_SEL(wd)); + + __raw_writel(0, AC97_CFG(wd)); + wmb(); /* drain writebuffer */ + __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); + wmb(); /* drain writebuffer */ + + return 0; +} + +static int au1xpsc_ac97_drvresume(struct device *dev) +{ + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + + /* restore PSC clock config */ + __raw_writel(wd->pm[0] | PSC_SEL_PS_AC97MODE, PSC_SEL(wd)); + wmb(); /* drain writebuffer */ + + /* after this point the ac97 core will cold-reset the codec. + * During cold-reset the PSC is reinitialized and the last + * configuration set up in hw_params() is restored. + */ + return 0; +} + +static const struct dev_pm_ops au1xpscac97_pmops = { + .suspend = au1xpsc_ac97_drvsuspend, + .resume = au1xpsc_ac97_drvresume, +}; + +#define AU1XPSCAC97_PMOPS &au1xpscac97_pmops + +#else + +#define AU1XPSCAC97_PMOPS NULL + +#endif + +static struct platform_driver au1xpsc_ac97_driver = { + .driver = { + .name = "au1xpsc_ac97", + .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"); + diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c new file mode 100644 index 000000000..e6eec081e --- /dev/null +++ b/sound/soc/au1x/psc-i2s.c @@ -0,0 +1,421 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (c) 2007-2008 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. + * + * Au1xxx-PSC I2S glue. + * + * 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> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/mach-au1x00/au1000.h> +#include <asm/mach-au1x00/au1xxx_psc.h> + +#include "psc.h" + +/* supported I2S DAI hardware formats */ +#define AU1XPSC_I2S_DAIFMT \ + (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | \ + SND_SOC_DAIFMT_NB_NF) + +/* supported I2S direction */ +#define AU1XPSC_I2S_DIR \ + (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) + +#define AU1XPSC_I2S_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define AU1XPSC_I2S_FMTS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +#define I2SSTAT_BUSY(stype) \ + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB) +#define I2SPCR_START(stype) \ + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TS : PSC_I2SPCR_RS) +#define I2SPCR_STOP(stype) \ + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TP : PSC_I2SPCR_RP) +#define I2SPCR_CLRFIFO(stype) \ + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TC : PSC_I2SPCR_RC) + + +static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(cpu_dai); + unsigned long ct; + int ret; + + ret = -EINVAL; + + ct = pscdata->cfg; + + ct &= ~(PSC_I2SCFG_XM | PSC_I2SCFG_MLJ); /* left-justified */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ct |= PSC_I2SCFG_XM; /* enable I2S mode */ + break; + case SND_SOC_DAIFMT_MSB: + break; + case SND_SOC_DAIFMT_LSB: + ct |= PSC_I2SCFG_MLJ; /* LSB (right-) justified */ + break; + default: + goto out; + } + + ct &= ~(PSC_I2SCFG_BI | PSC_I2SCFG_WI); /* IB-IF */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + ct |= PSC_I2SCFG_BI | PSC_I2SCFG_WI; + break; + case SND_SOC_DAIFMT_NB_IF: + ct |= PSC_I2SCFG_BI; + break; + case SND_SOC_DAIFMT_IB_NF: + ct |= PSC_I2SCFG_WI; + break; + case SND_SOC_DAIFMT_IB_IF: + break; + default: + goto out; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: /* CODEC master */ + ct |= PSC_I2SCFG_MS; /* PSC I2S slave mode */ + break; + case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */ + ct &= ~PSC_I2SCFG_MS; /* PSC I2S Master mode */ + break; + default: + goto out; + } + + pscdata->cfg = ct; + ret = 0; +out: + return ret; +} + +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 = snd_soc_dai_get_drvdata(dai); + + int cfgbits; + unsigned long stat; + + /* check if the PSC is already streaming data */ + stat = __raw_readl(I2S_STAT(pscdata)); + if (stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB)) { + /* reject parameters not currently set up in hardware */ + cfgbits = __raw_readl(I2S_CFG(pscdata)); + if ((PSC_I2SCFG_GET_LEN(cfgbits) != params->msbits) || + (params_rate(params) != pscdata->rate)) + return -EINVAL; + } else { + /* set sample bitdepth */ + pscdata->cfg &= ~(0x1f << 4); + pscdata->cfg |= PSC_I2SCFG_SET_LEN(params->msbits); + /* remember current rate for other stream */ + pscdata->rate = params_rate(params); + } + return 0; +} + +/* Configure PSC late: on my devel systems the codec is I2S master and + * supplies the i2sbitclock __AND__ i2sMclk (!) to the PSC unit. ASoC + * uses aggressive PM and switches the codec off when it is not in use + * which also means the PSC unit doesn't get any clocks and is therefore + * dead. That's why this chunk here gets called from the trigger callback + * because I can be reasonably certain the codec is driving the clocks. + */ +static int au1xpsc_i2s_configure(struct au1xpsc_audio_data *pscdata) +{ + unsigned long tmo; + + /* bring PSC out of sleep, and configure I2S unit */ + __raw_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata)); + wmb(); /* drain writebuffer */ + + tmo = 1000000; + while (!(__raw_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_SR) && tmo) + tmo--; + + if (!tmo) + goto psc_err; + + __raw_writel(0, I2S_CFG(pscdata)); + wmb(); /* drain writebuffer */ + __raw_writel(pscdata->cfg | PSC_I2SCFG_DE_ENABLE, I2S_CFG(pscdata)); + wmb(); /* drain writebuffer */ + + /* wait for I2S controller to become ready */ + tmo = 1000000; + while (!(__raw_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_DR) && tmo) + tmo--; + + if (tmo) + return 0; + +psc_err: + __raw_writel(0, I2S_CFG(pscdata)); + __raw_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata)); + wmb(); /* drain writebuffer */ + return -ETIMEDOUT; +} + +static int au1xpsc_i2s_start(struct au1xpsc_audio_data *pscdata, int stype) +{ + unsigned long tmo, stat; + int ret; + + ret = 0; + + /* if both TX and RX are idle, configure the PSC */ + stat = __raw_readl(I2S_STAT(pscdata)); + if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) { + ret = au1xpsc_i2s_configure(pscdata); + if (ret) + goto out; + } + + __raw_writel(I2SPCR_CLRFIFO(stype), I2S_PCR(pscdata)); + wmb(); /* drain writebuffer */ + __raw_writel(I2SPCR_START(stype), I2S_PCR(pscdata)); + wmb(); /* drain writebuffer */ + + /* wait for start confirmation */ + tmo = 1000000; + while (!(__raw_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo) + tmo--; + + if (!tmo) { + __raw_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata)); + wmb(); /* drain writebuffer */ + ret = -ETIMEDOUT; + } +out: + return ret; +} + +static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) +{ + unsigned long tmo, stat; + + __raw_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata)); + wmb(); /* drain writebuffer */ + + /* wait for stop confirmation */ + tmo = 1000000; + while ((__raw_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo) + tmo--; + + /* if both TX and RX are idle, disable PSC */ + stat = __raw_readl(I2S_STAT(pscdata)); + if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) { + __raw_writel(0, I2S_CFG(pscdata)); + wmb(); /* drain writebuffer */ + __raw_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata)); + wmb(); /* drain writebuffer */ + } + return 0; +} + +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->stream; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + ret = au1xpsc_i2s_start(pscdata, stype); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + ret = au1xpsc_i2s_stop(pscdata, stype); + break; + default: + ret = -EINVAL; + } + 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 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 *iores, *dmares; + unsigned long sel; + struct au1xpsc_audio_data *wd; + + wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data), + GFP_KERNEL); + if (!wd) + return -ENOMEM; + + 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); + + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + return -EBUSY; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; + + 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 = __raw_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK; + __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); + wmb(); /* drain writebuffer */ + __raw_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(wd)); + __raw_writel(0, I2S_CFG(wd)); + wmb(); /* drain writebuffer */ + + /* preconfigure: set max rx/tx fifo depths */ + 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. + */ + + /* 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); + + platform_set_drvdata(pdev, wd); + + return snd_soc_register_component(&pdev->dev, &au1xpsc_i2s_component, + &wd->dai_drv, 1); +} + +static int au1xpsc_i2s_drvremove(struct platform_device *pdev) +{ + struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); + + snd_soc_unregister_component(&pdev->dev); + + __raw_writel(0, I2S_CFG(wd)); + wmb(); /* drain writebuffer */ + __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); + wmb(); /* drain writebuffer */ + + return 0; +} + +#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 */ + wd->pm[0] = __raw_readl(PSC_SEL(wd)); + + __raw_writel(0, I2S_CFG(wd)); + wmb(); /* drain writebuffer */ + __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); + wmb(); /* drain writebuffer */ + + return 0; +} + +static int au1xpsc_i2s_drvresume(struct device *dev) +{ + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + + /* select I2S mode and PSC clock */ + __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); + wmb(); /* drain writebuffer */ + __raw_writel(0, PSC_SEL(wd)); + wmb(); /* drain writebuffer */ + __raw_writel(wd->pm[0], PSC_SEL(wd)); + wmb(); /* drain writebuffer */ + + return 0; +} + +static const struct dev_pm_ops au1xpsci2s_pmops = { + .suspend = au1xpsc_i2s_drvsuspend, + .resume = au1xpsc_i2s_drvresume, +}; + +#define AU1XPSCI2S_PMOPS &au1xpsci2s_pmops + +#else + +#define AU1XPSCI2S_PMOPS NULL + +#endif + +static struct platform_driver au1xpsc_i2s_driver = { + .driver = { + .name = "au1xpsc_i2s", + .pm = AU1XPSCI2S_PMOPS, + }, + .probe = au1xpsc_i2s_drvprobe, + .remove = au1xpsc_i2s_drvremove, +}; + +module_platform_driver(au1xpsc_i2s_driver); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h new file mode 100644 index 000000000..74dffeb64 --- /dev/null +++ b/sound/soc/au1x/psc.h @@ -0,0 +1,42 @@ +/* + * Alchemy ALSA ASoC audio support. + * + * (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. + * + */ + +#ifndef _AU1X_PCM_H +#define _AU1X_PCM_H + +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 mutex lock; + int dmaids[2]; +}; + +/* easy access macros */ +#define PSC_CTRL(x) ((x)->mmio + PSC_CTRL_OFFSET) +#define PSC_SEL(x) ((x)->mmio + PSC_SEL_OFFSET) +#define I2S_STAT(x) ((x)->mmio + PSC_I2SSTAT_OFFSET) +#define I2S_CFG(x) ((x)->mmio + PSC_I2SCFG_OFFSET) +#define I2S_PCR(x) ((x)->mmio + PSC_I2SPCR_OFFSET) +#define AC97_CFG(x) ((x)->mmio + PSC_AC97CFG_OFFSET) +#define AC97_CDC(x) ((x)->mmio + PSC_AC97CDC_OFFSET) +#define AC97_EVNT(x) ((x)->mmio + PSC_AC97EVNT_OFFSET) +#define AC97_PCR(x) ((x)->mmio + PSC_AC97PCR_OFFSET) +#define AC97_RST(x) ((x)->mmio + PSC_AC97RST_OFFSET) +#define AC97_STAT(x) ((x)->mmio + PSC_AC97STAT_OFFSET) + +#endif |