Merge branch 'for-3.1' into for-3.2
diff --git a/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt b/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt
new file mode 100644
index 0000000..2c3cd41
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt
@@ -0,0 +1,11 @@
+* Freescale SGTL5000 Stereo Codec
+
+Required properties:
+- compatible : "fsl,sgtl5000".
+
+Example:
+
+codec: sgtl5000@0a {
+ compatible = "fsl,sgtl5000";
+ reg = <0x0a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8731.txt b/Documentation/devicetree/bindings/sound/wm8731.txt
new file mode 100644
index 0000000..15f7004
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8731.txt
@@ -0,0 +1,18 @@
+WM8731 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8731"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8731@1a {
+ compatible = "wlf,wm8731";
+ reg = <0x1a>;
+};
diff --git a/arch/mips/alchemy/devboards/db1200/platform.c b/arch/mips/alchemy/devboards/db1200/platform.c
index fbb5593..dda090b 100644
--- a/arch/mips/alchemy/devboards/db1200/platform.c
+++ b/arch/mips/alchemy/devboards/db1200/platform.c
@@ -422,6 +422,7 @@
},
};
+/* AC97 or I2S device */
static struct platform_device db1200_audio_dev = {
/* name assigned later based on switch setting */
.id = 1, /* PSC ID */
@@ -429,19 +430,32 @@
.resource = au1200_psc1_res,
};
+/* DB1200 ASoC card device */
+static struct platform_device db1200_sound_dev = {
+ /* name assigned later based on switch setting */
+ .id = 1, /* PSC ID */
+};
+
static struct platform_device db1200_stac_dev = {
.name = "ac97-codec",
.id = 1, /* on PSC1 */
};
+static struct platform_device db1200_audiodma_dev = {
+ .name = "au1xpsc-pcm",
+ .id = 1, /* PSC ID */
+};
+
static struct platform_device *db1200_devs[] __initdata = {
NULL, /* PSC0, selected by S6.8 */
&db1200_ide_dev,
&db1200_eth_dev,
&db1200_rtc_dev,
&db1200_nand_dev,
+ &db1200_audiodma_dev,
&db1200_audio_dev,
&db1200_stac_dev,
+ &db1200_sound_dev,
};
static int __init db1200_dev_init(void)
@@ -501,10 +515,12 @@
if (sw == BCSR_SWITCHES_DIP_8) {
bcsr_mod(BCSR_RESETS, 0, BCSR_RESETS_PSC1MUX);
db1200_audio_dev.name = "au1xpsc_i2s";
+ db1200_sound_dev.name = "db1200-i2s";
printk(KERN_INFO " S6.7 ON : PSC1 mode I2S\n");
} else {
bcsr_mod(BCSR_RESETS, BCSR_RESETS_PSC1MUX, 0);
db1200_audio_dev.name = "au1xpsc_ac97";
+ db1200_sound_dev.name = "db1200-ac97";
printk(KERN_INFO " S6.7 OFF: PSC1 mode AC97\n");
}
diff --git a/arch/mips/alchemy/devboards/db1x00/platform.c b/arch/mips/alchemy/devboards/db1x00/platform.c
index 978d5ab..7057d28 100644
--- a/arch/mips/alchemy/devboards/db1x00/platform.c
+++ b/arch/mips/alchemy/devboards/db1x00/platform.c
@@ -19,8 +19,11 @@
*/
#include <linux/init.h>
+#include <linux/interrupt.h>
#include <linux/platform_device.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1000_dma.h>
#include <asm/mach-au1x00/au1xxx.h>
#include <asm/mach-db1x00/bcsr.h>
#include "../platform.h"
@@ -85,6 +88,45 @@
#endif
#endif
+static struct resource alchemy_ac97c_res[] = {
+ [0] = {
+ .start = AU1000_AC97_PHYS_ADDR,
+ .end = AU1000_AC97_PHYS_ADDR + 0xfff,
+ .flags = IORESOURCE_MEM,
+ },
+ [1] = {
+ .start = DMA_ID_AC97C_TX,
+ .end = DMA_ID_AC97C_TX,
+ .flags = IORESOURCE_DMA,
+ },
+ [2] = {
+ .start = DMA_ID_AC97C_RX,
+ .end = DMA_ID_AC97C_RX,
+ .flags = IORESOURCE_DMA,
+ },
+};
+
+static struct platform_device alchemy_ac97c_dev = {
+ .name = "alchemy-ac97c",
+ .id = -1,
+ .resource = alchemy_ac97c_res,
+ .num_resources = ARRAY_SIZE(alchemy_ac97c_res),
+};
+
+static struct platform_device alchemy_ac97c_dma_dev = {
+ .name = "alchemy-pcm-dma",
+ .id = 0,
+};
+
+static struct platform_device db1x00_codec_dev = {
+ .name = "ac97-codec",
+ .id = -1,
+};
+
+static struct platform_device db1x00_audio_dev = {
+ .name = "db1000-audio",
+};
+
static int __init db1xxx_dev_init(void)
{
#ifdef DB1XXX_HAS_PCMCIA
@@ -113,6 +155,12 @@
1);
#endif
db1x_register_norflash(BOARD_FLASH_SIZE, BOARD_FLASH_WIDTH, F_SWAPPED);
+
+ platform_device_register(&db1x00_codec_dev);
+ platform_device_register(&alchemy_ac97c_dma_dev);
+ platform_device_register(&alchemy_ac97c_dev);
+ platform_device_register(&db1x00_audio_dev);
+
return 0;
}
device_initcall(db1xxx_dev_init);
diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h
index f3ee842..6152914 100644
--- a/include/linux/mfd/wm8994/registers.h
+++ b/include/linux/mfd/wm8994/registers.h
@@ -72,6 +72,7 @@
#define WM8994_DC_SERVO_2 0x55
#define WM8994_DC_SERVO_4 0x57
#define WM8994_DC_SERVO_READBACK 0x58
+#define WM8994_DC_SERVO_4E 0x59
#define WM8994_ANALOGUE_HP_1 0x60
#define WM8958_MIC_DETECT_1 0xD0
#define WM8958_MIC_DETECT_2 0xD1
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index e0583b7..350b1b3 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -524,6 +524,8 @@
enum snd_soc_bias_level target_bias_level;
struct list_head list;
+ int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
+
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_dapm;
#endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index aa19f5a..3fe658e 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -19,6 +19,7 @@
#include <linux/workqueue.h>
#include <linux/interrupt.h>
#include <linux/kernel.h>
+#include <linux/regmap.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/control.h>
@@ -260,6 +261,7 @@
enum snd_soc_control_type {
SND_SOC_I2C = 1,
SND_SOC_SPI,
+ SND_SOC_REGMAP,
};
enum snd_soc_compress_type {
@@ -576,6 +578,7 @@
const void *reg_def_copy;
const struct snd_soc_cache_ops *cache_ops;
struct mutex cache_rw_mutex;
+ int val_bytes;
/* dapm */
struct snd_soc_dapm_context dapm;
@@ -634,6 +637,9 @@
void (*seq_notifier)(struct snd_soc_dapm_context *,
enum snd_soc_dapm_type, int);
+ /* codec stream completion event */
+ int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
+
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
@@ -669,6 +675,9 @@
/* platform stream ops */
struct snd_pcm_ops *ops;
+ /* platform stream completion event */
+ int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
+
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig
index a9823fa..77dd0a1 100644
--- a/sound/mips/Kconfig
+++ b/sound/mips/Kconfig
@@ -23,12 +23,15 @@
config SND_AU1X00
- tristate "Au1x00 AC97 Port Driver"
+ tristate "Au1x00 AC97 Port Driver (DEPRECATED)"
depends on SOC_AU1000 || SOC_AU1100 || SOC_AU1500
select SND_PCM
select SND_AC97_CODEC
help
ALSA Sound driver for the Au1x00's AC97 port.
+ Newer drivers for ASoC are available, please do not use
+ this driver as it will be removed in the future.
+
endif # SND_MIPS
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 8224db5..1381db8 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -7,6 +7,8 @@
select SND_PCM
select AC97_BUS if SND_SOC_AC97_BUS
select SND_JACK if INPUT=y || INPUT=SND
+ select REGMAP_I2C if I2C
+ select REGMAP_SPI if SPI_MASTER
---help---
If you want ASoC support, you should say Y here and also to the
@@ -51,6 +53,7 @@
source "sound/soc/omap/Kconfig"
source "sound/soc/kirkwood/Kconfig"
source "sound/soc/mid-x86/Kconfig"
+source "sound/soc/mxs/Kconfig"
source "sound/soc/pxa/Kconfig"
source "sound/soc/samsung/Kconfig"
source "sound/soc/s6000/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 4f91387..9ea8ac8 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -12,6 +12,7 @@
obj-$(CONFIG_SND_SOC) += imx/
obj-$(CONFIG_SND_SOC) += jz4740/
obj-$(CONFIG_SND_SOC) += mid-x86/
+obj-$(CONFIG_SND_SOC) += mxs/
obj-$(CONFIG_SND_SOC) += nuc900/
obj-$(CONFIG_SND_SOC) += omap/
obj-$(CONFIG_SND_SOC) += kirkwood/
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
index 4b67140..6d59254 100644
--- a/sound/soc/au1x/Kconfig
+++ b/sound/soc/au1x/Kconfig
@@ -18,10 +18,38 @@
select SND_AC97_CODEC
select SND_SOC_AC97_BUS
+##
+## Au1000/1500/1100 DMA + AC97C/I2SC
+##
+config SND_SOC_AU1XAUDIO
+ tristate "SoC Audio for Au1000/Au1500/Au1100"
+ depends on MIPS_ALCHEMY
+ help
+ This is a driver set for the AC97 unit and the
+ old DMA controller as found on the Au1000/Au1500/Au1100 chips.
+
+config SND_SOC_AU1XAC97C
+ tristate
+ select AC97_BUS
+ select SND_AC97_CODEC
+ select SND_SOC_AC97_BUS
+
+config SND_SOC_AU1XI2SC
+ tristate
+
##
## Boards
##
+config SND_SOC_DB1000
+ tristate "DB1000 Audio support"
+ depends on SND_SOC_AU1XAUDIO
+ select SND_SOC_AU1XAC97C
+ select SND_SOC_AC97_CODEC
+ help
+ Select this option to enable AC97 audio on the early DB1x00 series
+ of boards (DB1000/DB1500/DB1100).
+
config SND_SOC_DB1200
tristate "DB1200 AC97+I2S audio support"
depends on SND_SOC_AU1XPSC
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
index 1687307..9207105 100644
--- a/sound/soc/au1x/Makefile
+++ b/sound/soc/au1x/Makefile
@@ -3,11 +3,21 @@
snd-soc-au1xpsc-i2s-objs := psc-i2s.o
snd-soc-au1xpsc-ac97-objs := psc-ac97.o
+# Au1000/1500/1100 Audio units
+snd-soc-au1x-dma-objs := dma.o
+snd-soc-au1x-ac97c-objs := ac97c.o
+snd-soc-au1x-i2sc-objs := i2sc.o
+
obj-$(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 0000000..9c05f38
--- /dev/null
+++ b/sound/soc/au1x/ac97c.c
@@ -0,0 +1,365 @@
+/*
+ * Au1000/Au1500/Au1100 AC97C controller driver for ASoC
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * based on the old ALSA driver originally written by
+ * Charles Eidsness <charles@cooper-street.com>
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/mutex.h>
+#include <linux/platform_device.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+
+#include "psc.h"
+
+/* register offsets and bits */
+#define AC97_CONFIG 0x00
+#define AC97_STATUS 0x04
+#define AC97_DATA 0x08
+#define AC97_CMDRESP 0x0c
+#define AC97_ENABLE 0x10
+
+#define CFG_RC(x) (((x) & 0x3ff) << 13) /* valid rx slots mask */
+#define CFG_XS(x) (((x) & 0x3ff) << 3) /* valid tx slots mask */
+#define CFG_SG (1 << 2) /* sync gate */
+#define CFG_SN (1 << 1) /* sync control */
+#define CFG_RS (1 << 0) /* acrst# control */
+#define STAT_XU (1 << 11) /* tx underflow */
+#define STAT_XO (1 << 10) /* tx overflow */
+#define STAT_RU (1 << 9) /* rx underflow */
+#define STAT_RO (1 << 8) /* rx overflow */
+#define STAT_RD (1 << 7) /* codec ready */
+#define STAT_CP (1 << 6) /* command pending */
+#define STAT_TE (1 << 4) /* tx fifo empty */
+#define STAT_TF (1 << 3) /* tx fifo full */
+#define STAT_RE (1 << 1) /* rx fifo empty */
+#define STAT_RF (1 << 0) /* rx fifo full */
+#define CMD_SET_DATA(x) (((x) & 0xffff) << 16)
+#define CMD_GET_DATA(x) ((x) & 0xffff)
+#define CMD_READ (1 << 7)
+#define CMD_WRITE (0 << 7)
+#define CMD_IDX(x) ((x) & 0x7f)
+#define EN_D (1 << 1) /* DISable bit */
+#define EN_CE (1 << 0) /* clock enable bit */
+
+/* how often to retry failed codec register reads/writes */
+#define AC97_RW_RETRIES 5
+
+#define AC97_RATES \
+ SNDRV_PCM_RATE_CONTINUOUS
+
+#define AC97_FMTS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE)
+
+/* instance data. There can be only one, MacLeod!!!!, fortunately there IS only
+ * once AC97C on early Alchemy chips. The newer ones aren't so lucky.
+ */
+static struct au1xpsc_audio_data *ac97c_workdata;
+#define ac97_to_ctx(x) ac97c_workdata
+
+static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg)
+{
+ return __raw_readl(ctx->mmio + reg);
+}
+
+static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v)
+{
+ __raw_writel(v, ctx->mmio + reg);
+ wmb();
+}
+
+static unsigned short au1xac97c_ac97_read(struct snd_ac97 *ac97,
+ unsigned short r)
+{
+ struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+ unsigned int tmo, retry;
+ unsigned long data;
+
+ data = ~0;
+ retry = AC97_RW_RETRIES;
+ do {
+ mutex_lock(&ctx->lock);
+
+ tmo = 5;
+ while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--)
+ udelay(21); /* wait an ac97 frame time */
+ if (!tmo) {
+ pr_debug("ac97rd timeout #1\n");
+ goto next;
+ }
+
+ WR(ctx, AC97_CMDRESP, CMD_IDX(r) | CMD_READ);
+
+ /* stupid errata: data is only valid for 21us, so
+ * poll, Forrest, poll...
+ */
+ tmo = 0x10000;
+ while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--)
+ asm volatile ("nop");
+ data = RD(ctx, AC97_CMDRESP);
+
+ if (!tmo)
+ pr_debug("ac97rd timeout #2\n");
+
+next:
+ mutex_unlock(&ctx->lock);
+ } while (--retry && !tmo);
+
+ pr_debug("AC97RD %04x %04lx %d\n", r, data, retry);
+
+ return retry ? data & 0xffff : 0xffff;
+}
+
+static void au1xac97c_ac97_write(struct snd_ac97 *ac97, unsigned short r,
+ unsigned short v)
+{
+ struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+ unsigned int tmo, retry;
+
+ retry = AC97_RW_RETRIES;
+ do {
+ mutex_lock(&ctx->lock);
+
+ for (tmo = 5; (RD(ctx, AC97_STATUS) & STAT_CP) && tmo; tmo--)
+ udelay(21);
+ if (!tmo) {
+ pr_debug("ac97wr timeout #1\n");
+ goto next;
+ }
+
+ WR(ctx, AC97_CMDRESP, CMD_WRITE | CMD_IDX(r) | CMD_SET_DATA(v));
+
+ for (tmo = 10; (RD(ctx, AC97_STATUS) & STAT_CP) && tmo; tmo--)
+ udelay(21);
+ if (!tmo)
+ pr_debug("ac97wr timeout #2\n");
+next:
+ mutex_unlock(&ctx->lock);
+ } while (--retry && !tmo);
+
+ pr_debug("AC97WR %04x %04x %d\n", r, v, retry);
+}
+
+static void au1xac97c_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+
+ WR(ctx, AC97_CONFIG, ctx->cfg | CFG_SG | CFG_SN);
+ msleep(20);
+ WR(ctx, AC97_CONFIG, ctx->cfg | CFG_SG);
+ WR(ctx, AC97_CONFIG, ctx->cfg);
+}
+
+static void au1xac97c_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+ struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+ int i;
+
+ WR(ctx, AC97_CONFIG, ctx->cfg | CFG_RS);
+ msleep(500);
+ WR(ctx, AC97_CONFIG, ctx->cfg);
+
+ /* wait for codec ready */
+ i = 50;
+ while (((RD(ctx, AC97_STATUS) & STAT_RD) == 0) && --i)
+ msleep(20);
+ if (!i)
+ printk(KERN_ERR "ac97c: codec not ready after cold reset\n");
+}
+
+/* AC97 controller operations */
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = au1xac97c_ac97_read,
+ .write = au1xac97c_ac97_write,
+ .reset = au1xac97c_ac97_cold_reset,
+ .warm_reset = au1xac97c_ac97_warm_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops); /* globals be gone! */
+
+static int alchemy_ac97c_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]);
+ return 0;
+}
+
+static struct snd_soc_dai_ops alchemy_ac97c_ops = {
+ .startup = alchemy_ac97c_startup,
+};
+
+static int au1xac97c_dai_probe(struct snd_soc_dai *dai)
+{
+ return ac97c_workdata ? 0 : -ENODEV;
+}
+
+static struct snd_soc_dai_driver au1xac97c_dai_driver = {
+ .name = "alchemy-ac97c",
+ .ac97_control = 1,
+ .probe = au1xac97c_dai_probe,
+ .playback = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .capture = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .ops = &alchemy_ac97c_ops,
+};
+
+static int __devinit au1xac97c_drvprobe(struct platform_device *pdev)
+{
+ int ret;
+ struct resource *r;
+ struct au1xpsc_audio_data *ctx;
+
+ ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+ if (!ctx)
+ return -ENOMEM;
+
+ mutex_init(&ctx->lock);
+
+ r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!r) {
+ ret = -ENODEV;
+ goto out0;
+ }
+
+ ret = -EBUSY;
+ if (!request_mem_region(r->start, resource_size(r), pdev->name))
+ goto out0;
+
+ ctx->mmio = ioremap_nocache(r->start, resource_size(r));
+ if (!ctx->mmio)
+ goto out1;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!r)
+ goto out1;
+ ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!r)
+ goto out1;
+ ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
+ /* switch it on */
+ WR(ctx, AC97_ENABLE, EN_D | EN_CE);
+ WR(ctx, AC97_ENABLE, EN_CE);
+
+ ctx->cfg = CFG_RC(3) | CFG_XS(3);
+ WR(ctx, AC97_CONFIG, ctx->cfg);
+
+ platform_set_drvdata(pdev, ctx);
+
+ ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver);
+ if (ret)
+ goto out1;
+
+ ac97c_workdata = ctx;
+ return 0;
+
+
+ snd_soc_unregister_dai(&pdev->dev);
+out1:
+ release_mem_region(r->start, resource_size(r));
+out0:
+ kfree(ctx);
+ return ret;
+}
+
+static int __devexit au1xac97c_drvremove(struct platform_device *pdev)
+{
+ struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
+ struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+
+ snd_soc_unregister_dai(&pdev->dev);
+
+ WR(ctx, AC97_ENABLE, EN_D); /* clock off, disable */
+
+ iounmap(ctx->mmio);
+ release_mem_region(r->start, resource_size(r));
+ kfree(ctx);
+
+ ac97c_workdata = NULL; /* MDEV */
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int au1xac97c_drvsuspend(struct device *dev)
+{
+ struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+ WR(ctx, AC97_ENABLE, EN_D); /* clock off, disable */
+
+ return 0;
+}
+
+static int au1xac97c_drvresume(struct device *dev)
+{
+ struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+ WR(ctx, AC97_ENABLE, EN_D | EN_CE);
+ WR(ctx, AC97_ENABLE, EN_CE);
+ WR(ctx, AC97_CONFIG, ctx->cfg);
+
+ return 0;
+}
+
+static const struct dev_pm_ops au1xpscac97_pmops = {
+ .suspend = au1xac97c_drvsuspend,
+ .resume = au1xac97c_drvresume,
+};
+
+#define AU1XPSCAC97_PMOPS (&au1xpscac97_pmops)
+
+#else
+
+#define AU1XPSCAC97_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xac97c_driver = {
+ .driver = {
+ .name = "alchemy-ac97c",
+ .owner = THIS_MODULE,
+ .pm = AU1XPSCAC97_PMOPS,
+ },
+ .probe = au1xac97c_drvprobe,
+ .remove = __devexit_p(au1xac97c_drvremove),
+};
+
+static int __init au1xac97c_load(void)
+{
+ ac97c_workdata = NULL;
+ return platform_driver_register(&au1xac97c_driver);
+}
+
+static void __exit au1xac97c_unload(void)
+{
+ platform_driver_unregister(&au1xac97c_driver);
+}
+
+module_init(au1xac97c_load);
+module_exit(au1xac97c_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c
new file mode 100644
index 0000000..127477a
--- /dev/null
+++ b/sound/soc/au1x/db1000.c
@@ -0,0 +1,75 @@
+/*
+ * DB1000/DB1500/DB1100 ASoC audio fabric support code.
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-db1x00/bcsr.h>
+
+#include "psc.h"
+
+static struct snd_soc_dai_link db1000_ac97_dai = {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .codec_dai_name = "ac97-hifi",
+ .cpu_dai_name = "alchemy-ac97c",
+ .platform_name = "alchemy-pcm-dma.0",
+ .codec_name = "ac97-codec",
+};
+
+static struct snd_soc_card db1000_ac97 = {
+ .name = "DB1000_AC97",
+ .dai_link = &db1000_ac97_dai,
+ .num_links = 1,
+};
+
+static int __devinit db1000_audio_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &db1000_ac97;
+ card->dev = &pdev->dev;
+ return snd_soc_register_card(card);
+}
+
+static int __devexit db1000_audio_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ snd_soc_unregister_card(card);
+ return 0;
+}
+
+static struct platform_driver db1000_audio_driver = {
+ .driver = {
+ .name = "db1000-audio",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = db1000_audio_probe,
+ .remove = __devexit_p(db1000_audio_remove),
+};
+
+static int __init db1000_audio_load(void)
+{
+ return platform_driver_register(&db1000_audio_driver);
+}
+
+static void __exit db1000_audio_unload(void)
+{
+ platform_driver_unregister(&db1000_audio_driver);
+}
+
+module_init(db1000_audio_load);
+module_exit(db1000_audio_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("DB1000/DB1500/DB1100 ASoC audio");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 1d3e258..289312c 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -1,7 +1,7 @@
/*
* DB1200 ASoC audio fabric support code.
*
- * (c) 2008-9 Manuel Lauss <manuel.lauss@gmail.com>
+ * (c) 2008-2011 Manuel Lauss <manuel.lauss@googlemail.com>
*
*/
@@ -21,6 +21,17 @@
#include "../codecs/wm8731.h"
#include "psc.h"
+static struct platform_device_id db1200_pids[] = {
+ {
+ .name = "db1200-ac97",
+ .driver_data = 0,
+ }, {
+ .name = "db1200-i2s",
+ .driver_data = 1,
+ },
+ {},
+};
+
/*------------------------- AC97 PART ---------------------------*/
static struct snd_soc_dai_link db1200_ac97_dai = {
@@ -89,36 +100,47 @@
/*------------------------- COMMON PART ---------------------------*/
-static struct platform_device *db1200_asoc_dev;
+static struct snd_soc_card *db1200_cards[] __devinitdata = {
+ &db1200_ac97_machine,
+ &db1200_i2s_machine,
+};
+
+static int __devinit 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 __devexit 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 = __devexit_p(db1200_audio_remove),
+};
static int __init db1200_audio_load(void)
{
- int ret;
-
- ret = -ENOMEM;
- db1200_asoc_dev = platform_device_alloc("soc-audio", 1); /* PSC1 */
- if (!db1200_asoc_dev)
- goto out;
-
- /* DB1200 board setup set PSC1MUX to preferred audio device */
- if (bcsr_read(BCSR_RESETS) & BCSR_RESETS_PSC1MUX)
- platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_machine);
- else
- platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_machine);
-
- ret = platform_device_add(db1200_asoc_dev);
-
- if (ret) {
- platform_device_put(db1200_asoc_dev);
- db1200_asoc_dev = NULL;
- }
-out:
- return ret;
+ return platform_driver_register(&db1200_audio_driver);
}
static void __exit db1200_audio_unload(void)
{
- platform_device_unregister(db1200_asoc_dev);
+ platform_driver_unregister(&db1200_audio_driver);
}
module_init(db1200_audio_load);
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 20bb53a..d7d04e2 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -169,7 +169,7 @@
au1x_pcm_dbdma_free(pcd);
- if (stype == PCM_RX)
+ if (stype == SNDRV_PCM_STREAM_CAPTURE)
pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id,
DSCR_CMD0_ALWAYS,
au1x_pcm_dmarx_cb, (void *)pcd);
@@ -198,7 +198,7 @@
struct snd_soc_pcm_runtime *rtd = ss->private_data;
struct au1xpsc_audio_dmadata *pcd =
snd_soc_platform_get_drvdata(rtd->platform);
- return &pcd[SUBSTREAM_TYPE(ss)];
+ return &pcd[ss->stream];
}
static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
@@ -212,7 +212,7 @@
if (ret < 0)
goto out;
- stype = SUBSTREAM_TYPE(substream);
+ stype = substream->stream;
pcd = to_dmadata(substream);
DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d "
@@ -255,7 +255,7 @@
au1xxx_dbdma_reset(pcd->ddma_chan);
- if (SUBSTREAM_TYPE(substream) == PCM_RX) {
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
au1x_pcm_queue_rx(pcd);
au1x_pcm_queue_rx(pcd);
} else {
@@ -293,6 +293,16 @@
static int au1xpsc_pcm_open(struct snd_pcm_substream *substream)
{
+ struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream);
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int stype = substream->stream, *dmaids;
+
+ dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ if (!dmaids)
+ return -ENODEV; /* whoa, has ordering changed? */
+
+ pcd->ddma_id = dmaids[stype];
+
snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware);
return 0;
}
@@ -340,36 +350,18 @@
static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev)
{
struct au1xpsc_audio_dmadata *dmadata;
- struct resource *r;
int ret;
dmadata = kzalloc(2 * sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
if (!dmadata)
return -ENOMEM;
- r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
- if (!r) {
- ret = -ENODEV;
- goto out1;
- }
- dmadata[PCM_TX].ddma_id = r->start;
-
- /* RX DMA */
- r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
- if (!r) {
- ret = -ENODEV;
- goto out1;
- }
- dmadata[PCM_RX].ddma_id = r->start;
-
platform_set_drvdata(pdev, dmadata);
ret = snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform);
- if (!ret)
- return ret;
+ if (ret)
+ kfree(dmadata);
-out1:
- kfree(dmadata);
return ret;
}
@@ -405,57 +397,6 @@
module_init(au1xpsc_audio_dbdma_load);
module_exit(au1xpsc_audio_dbdma_unload);
-
-struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev)
-{
- struct resource *res, *r;
- struct platform_device *pd;
- int id[2];
- int ret;
-
- r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
- if (!r)
- return NULL;
- id[0] = r->start;
-
- r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
- if (!r)
- return NULL;
- id[1] = r->start;
-
- res = kzalloc(sizeof(struct resource) * 2, GFP_KERNEL);
- if (!res)
- return NULL;
-
- res[0].start = res[0].end = id[0];
- res[1].start = res[1].end = id[1];
- res[0].flags = res[1].flags = IORESOURCE_DMA;
-
- pd = platform_device_alloc("au1xpsc-pcm", pdev->id);
- if (!pd)
- goto out;
-
- pd->resource = res;
- pd->num_resources = 2;
-
- ret = platform_device_add(pd);
- if (!ret)
- return pd;
-
- platform_device_put(pd);
-out:
- kfree(res);
- return NULL;
-}
-EXPORT_SYMBOL_GPL(au1xpsc_pcm_add);
-
-void au1xpsc_pcm_destroy(struct platform_device *dmapd)
-{
- if (dmapd)
- platform_device_unregister(dmapd);
-}
-EXPORT_SYMBOL_GPL(au1xpsc_pcm_destroy);
-
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver");
MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c
new file mode 100644
index 0000000..7aa5b76
--- /dev/null
+++ b/sound/soc/au1x/dma.c
@@ -0,0 +1,377 @@
+/*
+ * Au1000/Au1500/Au1100 Audio DMA support.
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * copied almost verbatim from the old ALSA driver, written by
+ * Charles Eidsness <charles@cooper-street.com>
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1000_dma.h>
+
+#include "psc.h"
+
+#define ALCHEMY_PCM_FMTS \
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \
+ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | \
+ SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE | \
+ 0)
+
+struct pcm_period {
+ u32 start;
+ u32 relative_end; /* relative to start of buffer */
+ struct pcm_period *next;
+};
+
+struct audio_stream {
+ struct snd_pcm_substream *substream;
+ int dma;
+ struct pcm_period *buffer;
+ unsigned int period_size;
+ unsigned int periods;
+};
+
+struct alchemy_pcm_ctx {
+ struct audio_stream stream[2]; /* playback & capture */
+};
+
+static void au1000_release_dma_link(struct audio_stream *stream)
+{
+ struct pcm_period *pointer;
+ struct pcm_period *pointer_next;
+
+ stream->period_size = 0;
+ stream->periods = 0;
+ pointer = stream->buffer;
+ if (!pointer)
+ return;
+ do {
+ pointer_next = pointer->next;
+ kfree(pointer);
+ pointer = pointer_next;
+ } while (pointer != stream->buffer);
+ stream->buffer = NULL;
+}
+
+static int au1000_setup_dma_link(struct audio_stream *stream,
+ unsigned int period_bytes,
+ unsigned int periods)
+{
+ struct snd_pcm_substream *substream = stream->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_period *pointer;
+ unsigned long dma_start;
+ int i;
+
+ dma_start = virt_to_phys(runtime->dma_area);
+
+ if (stream->period_size == period_bytes &&
+ stream->periods == periods)
+ return 0; /* not changed */
+
+ au1000_release_dma_link(stream);
+
+ stream->period_size = period_bytes;
+ stream->periods = periods;
+
+ stream->buffer = kmalloc(sizeof(struct pcm_period), GFP_KERNEL);
+ if (!stream->buffer)
+ return -ENOMEM;
+ pointer = stream->buffer;
+ for (i = 0; i < periods; i++) {
+ pointer->start = (u32)(dma_start + (i * period_bytes));
+ pointer->relative_end = (u32) (((i+1) * period_bytes) - 0x1);
+ if (i < periods - 1) {
+ pointer->next = kmalloc(sizeof(struct pcm_period),
+ GFP_KERNEL);
+ if (!pointer->next) {
+ au1000_release_dma_link(stream);
+ return -ENOMEM;
+ }
+ pointer = pointer->next;
+ }
+ }
+ pointer->next = stream->buffer;
+ return 0;
+}
+
+static void au1000_dma_stop(struct audio_stream *stream)
+{
+ if (stream->buffer)
+ disable_dma(stream->dma);
+}
+
+static void au1000_dma_start(struct audio_stream *stream)
+{
+ if (!stream->buffer)
+ return;
+
+ init_dma(stream->dma);
+ if (get_dma_active_buffer(stream->dma) == 0) {
+ clear_dma_done0(stream->dma);
+ set_dma_addr0(stream->dma, stream->buffer->start);
+ set_dma_count0(stream->dma, stream->period_size >> 1);
+ set_dma_addr1(stream->dma, stream->buffer->next->start);
+ set_dma_count1(stream->dma, stream->period_size >> 1);
+ } else {
+ clear_dma_done1(stream->dma);
+ set_dma_addr1(stream->dma, stream->buffer->start);
+ set_dma_count1(stream->dma, stream->period_size >> 1);
+ set_dma_addr0(stream->dma, stream->buffer->next->start);
+ set_dma_count0(stream->dma, stream->period_size >> 1);
+ }
+ enable_dma_buffers(stream->dma);
+ start_dma(stream->dma);
+}
+
+static irqreturn_t au1000_dma_interrupt(int irq, void *ptr)
+{
+ struct audio_stream *stream = (struct audio_stream *)ptr;
+ struct snd_pcm_substream *substream = stream->substream;
+
+ switch (get_dma_buffer_done(stream->dma)) {
+ case DMA_D0:
+ stream->buffer = stream->buffer->next;
+ clear_dma_done0(stream->dma);
+ set_dma_addr0(stream->dma, stream->buffer->next->start);
+ set_dma_count0(stream->dma, stream->period_size >> 1);
+ enable_dma_buffer0(stream->dma);
+ break;
+ case DMA_D1:
+ stream->buffer = stream->buffer->next;
+ clear_dma_done1(stream->dma);
+ set_dma_addr1(stream->dma, stream->buffer->next->start);
+ set_dma_count1(stream->dma, stream->period_size >> 1);
+ enable_dma_buffer1(stream->dma);
+ break;
+ case (DMA_D0 | DMA_D1):
+ pr_debug("DMA %d missed interrupt.\n", stream->dma);
+ au1000_dma_stop(stream);
+ au1000_dma_start(stream);
+ break;
+ case (~DMA_D0 & ~DMA_D1):
+ pr_debug("DMA %d empty irq.\n", stream->dma);
+ }
+ snd_pcm_period_elapsed(substream);
+ return IRQ_HANDLED;
+}
+
+static const struct snd_pcm_hardware alchemy_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
+ .formats = ALCHEMY_PCM_FMTS,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .rate_min = SNDRV_PCM_RATE_8000,
+ .rate_max = SNDRV_PCM_RATE_192000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .period_bytes_min = 1024,
+ .period_bytes_max = 16 * 1024 - 1,
+ .periods_min = 4,
+ .periods_max = 255,
+ .buffer_bytes_max = 128 * 1024,
+ .fifo_size = 16,
+};
+
+static inline struct alchemy_pcm_ctx *ss_to_ctx(struct snd_pcm_substream *ss)
+{
+ struct snd_soc_pcm_runtime *rtd = ss->private_data;
+ return snd_soc_platform_get_drvdata(rtd->platform);
+}
+
+static inline struct audio_stream *ss_to_as(struct snd_pcm_substream *ss)
+{
+ struct alchemy_pcm_ctx *ctx = ss_to_ctx(ss);
+ return &(ctx->stream[ss->stream]);
+}
+
+static int alchemy_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream);
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int *dmaids, s = substream->stream;
+ char *name;
+
+ dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ if (!dmaids)
+ return -ENODEV; /* whoa, has ordering changed? */
+
+ /* DMA setup */
+ name = (s == SNDRV_PCM_STREAM_PLAYBACK) ? "audio-tx" : "audio-rx";
+ ctx->stream[s].dma = request_au1000_dma(dmaids[s], name,
+ au1000_dma_interrupt, IRQF_DISABLED,
+ &ctx->stream[s]);
+ set_dma_mode(ctx->stream[s].dma,
+ get_dma_mode(ctx->stream[s].dma) & ~DMA_NC);
+
+ ctx->stream[s].substream = substream;
+ ctx->stream[s].buffer = NULL;
+ snd_soc_set_runtime_hwparams(substream, &alchemy_pcm_hardware);
+
+ return 0;
+}
+
+static int alchemy_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream);
+ int stype = substream->stream;
+
+ ctx->stream[stype].substream = NULL;
+ free_au1000_dma(ctx->stream[stype].dma);
+
+ return 0;
+}
+
+static int alchemy_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct audio_stream *stream = ss_to_as(substream);
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ err = au1000_setup_dma_link(stream,
+ params_period_bytes(hw_params),
+ params_periods(hw_params));
+ if (err)
+ snd_pcm_lib_free_pages(substream);
+
+ return err;
+}
+
+static int alchemy_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct audio_stream *stream = ss_to_as(substream);
+ au1000_release_dma_link(stream);
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int alchemy_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct audio_stream *stream = ss_to_as(substream);
+ int err = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ au1000_dma_start(stream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ au1000_dma_stop(stream);
+ break;
+ default:
+ err = -EINVAL;
+ break;
+ }
+ return err;
+}
+
+static snd_pcm_uframes_t alchemy_pcm_pointer(struct snd_pcm_substream *ss)
+{
+ struct audio_stream *stream = ss_to_as(ss);
+ long location;
+
+ location = get_dma_residue(stream->dma);
+ location = stream->buffer->relative_end - location;
+ if (location == -1)
+ location = 0;
+ return bytes_to_frames(ss->runtime, location);
+}
+
+static struct snd_pcm_ops alchemy_pcm_ops = {
+ .open = alchemy_pcm_open,
+ .close = alchemy_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = alchemy_pcm_hw_params,
+ .hw_free = alchemy_pcm_hw_free,
+ .trigger = alchemy_pcm_trigger,
+ .pointer = alchemy_pcm_pointer,
+};
+
+static void alchemy_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int alchemy_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm *pcm = rtd->pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL), 65536, (4096 * 1024) - 1);
+
+ return 0;
+}
+
+struct snd_soc_platform_driver alchemy_pcm_soc_platform = {
+ .ops = &alchemy_pcm_ops,
+ .pcm_new = alchemy_pcm_new,
+ .pcm_free = alchemy_pcm_free_dma_buffers,
+};
+
+static int __devinit alchemy_pcm_drvprobe(struct platform_device *pdev)
+{
+ struct alchemy_pcm_ctx *ctx;
+ int ret;
+
+ ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+ if (!ctx)
+ return -ENOMEM;
+
+ platform_set_drvdata(pdev, ctx);
+
+ ret = snd_soc_register_platform(&pdev->dev, &alchemy_pcm_soc_platform);
+ if (ret)
+ kfree(ctx);
+
+ return ret;
+}
+
+static int __devexit alchemy_pcm_drvremove(struct platform_device *pdev)
+{
+ struct alchemy_pcm_ctx *ctx = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_platform(&pdev->dev);
+ kfree(ctx);
+
+ return 0;
+}
+
+static struct platform_driver alchemy_pcmdma_driver = {
+ .driver = {
+ .name = "alchemy-pcm-dma",
+ .owner = THIS_MODULE,
+ },
+ .probe = alchemy_pcm_drvprobe,
+ .remove = __devexit_p(alchemy_pcm_drvremove),
+};
+
+static int __init alchemy_pcmdma_load(void)
+{
+ return platform_driver_register(&alchemy_pcmdma_driver);
+}
+
+static void __exit alchemy_pcmdma_unload(void)
+{
+ platform_driver_unregister(&alchemy_pcmdma_driver);
+}
+
+module_init(alchemy_pcmdma_load);
+module_exit(alchemy_pcmdma_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/Au1500/Au1100 Audio DMA driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c
new file mode 100644
index 0000000..b4172fd
--- /dev/null
+++ b/sound/soc/au1x/i2sc.c
@@ -0,0 +1,347 @@
+/*
+ * Au1000/Au1500/Au1100 I2S controller driver for ASoC
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * Note: clock supplied to the I2S controller must be 256x samplerate.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+
+#include "psc.h"
+
+#define I2S_RXTX 0x00
+#define I2S_CFG 0x04
+#define I2S_ENABLE 0x08
+
+#define CFG_XU (1 << 25) /* tx underflow */
+#define CFG_XO (1 << 24)
+#define CFG_RU (1 << 23)
+#define CFG_RO (1 << 22)
+#define CFG_TR (1 << 21)
+#define CFG_TE (1 << 20)
+#define CFG_TF (1 << 19)
+#define CFG_RR (1 << 18)
+#define CFG_RF (1 << 17)
+#define CFG_ICK (1 << 12) /* clock invert */
+#define CFG_PD (1 << 11) /* set to make I2SDIO INPUT */
+#define CFG_LB (1 << 10) /* loopback */
+#define CFG_IC (1 << 9) /* word select invert */
+#define CFG_FM_I2S (0 << 7) /* I2S format */
+#define CFG_FM_LJ (1 << 7) /* left-justified */
+#define CFG_FM_RJ (2 << 7) /* right-justified */
+#define CFG_FM_MASK (3 << 7)
+#define CFG_TN (1 << 6) /* tx fifo en */
+#define CFG_RN (1 << 5) /* rx fifo en */
+#define CFG_SZ_8 (0x08)
+#define CFG_SZ_16 (0x10)
+#define CFG_SZ_18 (0x12)
+#define CFG_SZ_20 (0x14)
+#define CFG_SZ_24 (0x18)
+#define CFG_SZ_MASK (0x1f)
+#define EN_D (1 << 1) /* DISable */
+#define EN_CE (1 << 0) /* clock enable */
+
+/* only limited by clock generator and board design */
+#define AU1XI2SC_RATES \
+ SNDRV_PCM_RATE_CONTINUOUS
+
+#define AU1XI2SC_FMTS \
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | \
+ SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_U18_3BE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \
+ SNDRV_PCM_FMTBIT_S20_3BE | SNDRV_PCM_FMTBIT_U20_3BE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \
+ SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE | \
+ 0)
+
+static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg)
+{
+ return __raw_readl(ctx->mmio + reg);
+}
+
+static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v)
+{
+ __raw_writel(v, ctx->mmio + reg);
+ wmb();
+}
+
+static int au1xi2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(cpu_dai);
+ unsigned long c;
+ int ret;
+
+ ret = -EINVAL;
+ c = ctx->cfg;
+
+ c &= ~CFG_FM_MASK;
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ c |= CFG_FM_I2S;
+ break;
+ case SND_SOC_DAIFMT_MSB:
+ c |= CFG_FM_RJ;
+ break;
+ case SND_SOC_DAIFMT_LSB:
+ c |= CFG_FM_LJ;
+ break;
+ default:
+ goto out;
+ }
+
+ c &= ~(CFG_IC | CFG_ICK); /* IB-IF */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ c |= CFG_IC | CFG_ICK;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ c |= CFG_IC;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ c |= CFG_ICK;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ break;
+ default:
+ goto out;
+ }
+
+ /* I2S controller only supports master */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */
+ break;
+ default:
+ goto out;
+ }
+
+ ret = 0;
+ ctx->cfg = c;
+out:
+ return ret;
+}
+
+static int au1xi2s_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ int stype = SUBSTREAM_TYPE(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ /* power up */
+ WR(ctx, I2S_ENABLE, EN_D | EN_CE);
+ WR(ctx, I2S_ENABLE, EN_CE);
+ ctx->cfg |= (stype == PCM_TX) ? CFG_TN : CFG_RN;
+ WR(ctx, I2S_CFG, ctx->cfg);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ ctx->cfg &= ~((stype == PCM_TX) ? CFG_TN : CFG_RN);
+ WR(ctx, I2S_CFG, ctx->cfg);
+ WR(ctx, I2S_ENABLE, EN_D); /* power off */
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static unsigned long msbits_to_reg(int msbits)
+{
+ switch (msbits) {
+ case 8:
+ return CFG_SZ_8;
+ case 16:
+ return CFG_SZ_16;
+ case 18:
+ return CFG_SZ_18;
+ case 20:
+ return CFG_SZ_20;
+ case 24:
+ return CFG_SZ_24;
+ }
+ return 0;
+}
+
+static int au1xi2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ unsigned long v;
+
+ v = msbits_to_reg(params->msbits);
+ if (!v)
+ return -EINVAL;
+
+ ctx->cfg &= ~CFG_SZ_MASK;
+ ctx->cfg |= v;
+ return 0;
+}
+
+static int au1xi2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]);
+ return 0;
+}
+
+static const struct snd_soc_dai_ops au1xi2s_dai_ops = {
+ .startup = au1xi2s_startup,
+ .trigger = au1xi2s_trigger,
+ .hw_params = au1xi2s_hw_params,
+ .set_fmt = au1xi2s_set_fmt,
+};
+
+static struct snd_soc_dai_driver au1xi2s_dai_driver = {
+ .symmetric_rates = 1,
+ .playback = {
+ .rates = AU1XI2SC_RATES,
+ .formats = AU1XI2SC_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .capture = {
+ .rates = AU1XI2SC_RATES,
+ .formats = AU1XI2SC_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .ops = &au1xi2s_dai_ops,
+};
+
+static int __devinit au1xi2s_drvprobe(struct platform_device *pdev)
+{
+ int ret;
+ struct resource *r;
+ struct au1xpsc_audio_data *ctx;
+
+ ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+ if (!ctx)
+ return -ENOMEM;
+
+ r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!r) {
+ ret = -ENODEV;
+ goto out0;
+ }
+
+ ret = -EBUSY;
+ if (!request_mem_region(r->start, resource_size(r), pdev->name))
+ goto out0;
+
+ ctx->mmio = ioremap_nocache(r->start, resource_size(r));
+ if (!ctx->mmio)
+ goto out1;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!r)
+ goto out1;
+ ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!r)
+ goto out1;
+ ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
+ platform_set_drvdata(pdev, ctx);
+
+ ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver);
+ if (ret)
+ goto out1;
+
+ return 0;
+
+ snd_soc_unregister_dai(&pdev->dev);
+out1:
+ release_mem_region(r->start, resource_size(r));
+out0:
+ kfree(ctx);
+ return ret;
+}
+
+static int __devexit au1xi2s_drvremove(struct platform_device *pdev)
+{
+ struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
+ struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+
+ snd_soc_unregister_dai(&pdev->dev);
+
+ WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */
+
+ iounmap(ctx->mmio);
+ release_mem_region(r->start, resource_size(r));
+ kfree(ctx);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int au1xi2s_drvsuspend(struct device *dev)
+{
+ struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+ WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */
+
+ return 0;
+}
+
+static int au1xi2s_drvresume(struct device *dev)
+{
+ return 0;
+}
+
+static const struct dev_pm_ops au1xi2sc_pmops = {
+ .suspend = au1xi2s_drvsuspend,
+ .resume = au1xi2s_drvresume,
+};
+
+#define AU1XI2SC_PMOPS (&au1xi2sc_pmops)
+
+#else
+
+#define AU1XI2SC_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xi2s_driver = {
+ .driver = {
+ .name = "alchemy-i2sc",
+ .owner = THIS_MODULE,
+ .pm = AU1XI2SC_PMOPS,
+ },
+ .probe = au1xi2s_drvprobe,
+ .remove = __devexit_p(au1xi2s_drvremove),
+};
+
+static int __init au1xi2s_load(void)
+{
+ return platform_driver_register(&au1xi2s_driver);
+}
+
+static void __exit au1xi2s_unload(void)
+{
+ platform_driver_unregister(&au1xi2s_driver);
+}
+
+module_init(au1xi2s_load);
+module_exit(au1xi2s_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/1500/1100 I2S ASoC driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index d0db66f..172eefd 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -41,14 +41,14 @@
(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE)
#define AC97PCR_START(stype) \
- ((stype) == PCM_TX ? PSC_AC97PCR_TS : PSC_AC97PCR_RS)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TS : PSC_AC97PCR_RS)
#define AC97PCR_STOP(stype) \
- ((stype) == PCM_TX ? PSC_AC97PCR_TP : PSC_AC97PCR_RP)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TP : PSC_AC97PCR_RP)
#define AC97PCR_CLRFIFO(stype) \
- ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
#define AC97STAT_BUSY(stype) \
- ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
/* instance data. There can be only one, MacLeod!!!! */
static struct au1xpsc_audio_data *au1xpsc_ac97_workdata;
@@ -215,7 +215,7 @@
{
struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
unsigned long r, ro, stat;
- int chans, t, stype = SUBSTREAM_TYPE(substream);
+ int chans, t, stype = substream->stream;
chans = params_channels(params);
@@ -235,7 +235,7 @@
r |= PSC_AC97CFG_SET_LEN(params->msbits);
/* channels: enable slots for front L/R channel */
- if (stype == PCM_TX) {
+ if (stype == SNDRV_PCM_STREAM_PLAYBACK) {
r &= ~PSC_AC97CFG_TXSLOT_MASK;
r |= PSC_AC97CFG_TXSLOT_ENA(3);
r |= PSC_AC97CFG_TXSLOT_ENA(4);
@@ -294,7 +294,7 @@
int cmd, struct snd_soc_dai *dai)
{
struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
- int ret, stype = SUBSTREAM_TYPE(substream);
+ int ret, stype = substream->stream;
ret = 0;
@@ -324,12 +324,21 @@
return ret;
}
+static int au1xpsc_ac97_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
+ snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]);
+ return 0;
+}
+
static int au1xpsc_ac97_probe(struct snd_soc_dai *dai)
{
return au1xpsc_ac97_workdata ? 0 : -ENODEV;
}
static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
+ .startup = au1xpsc_ac97_startup,
.trigger = au1xpsc_ac97_trigger,
.hw_params = au1xpsc_ac97_hw_params,
};
@@ -379,6 +388,16 @@
if (!wd->mmio)
goto out1;
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!r)
+ goto out2;
+ wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!r)
+ goto out2;
+ wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
/* configuration: max dma trigger threshold, enable ac97 */
wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 |
PSC_AC97CFG_DE_ENABLE;
@@ -401,15 +420,13 @@
ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv);
if (ret)
- goto out1;
+ goto out2;
- wd->dmapd = au1xpsc_pcm_add(pdev);
- if (wd->dmapd) {
- au1xpsc_ac97_workdata = wd;
- return 0;
- }
+ au1xpsc_ac97_workdata = wd;
+ return 0;
- snd_soc_unregister_dai(&pdev->dev);
+out2:
+ iounmap(wd->mmio);
out1:
release_mem_region(r->start, resource_size(r));
out0:
@@ -422,9 +439,6 @@
struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (wd->dmapd)
- au1xpsc_pcm_destroy(wd->dmapd);
-
snd_soc_unregister_dai(&pdev->dev);
/* disable PSC completely */
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index fca0912..7c5ae92 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -42,13 +42,13 @@
(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
#define I2SSTAT_BUSY(stype) \
- ((stype) == PCM_TX ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
#define I2SPCR_START(stype) \
- ((stype) == PCM_TX ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
#define I2SPCR_STOP(stype) \
- ((stype) == PCM_TX ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
#define I2SPCR_CLRFIFO(stype) \
- ((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
@@ -240,7 +240,7 @@
struct snd_soc_dai *dai)
{
struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
- int ret, stype = SUBSTREAM_TYPE(substream);
+ int ret, stype = substream->stream;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
@@ -257,7 +257,16 @@
return ret;
}
+static int au1xpsc_i2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
+ snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]);
+ return 0;
+}
+
static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
+ .startup = au1xpsc_i2s_startup,
.trigger = au1xpsc_i2s_trigger,
.hw_params = au1xpsc_i2s_hw_params,
.set_fmt = au1xpsc_i2s_set_fmt,
@@ -304,6 +313,16 @@
if (!wd->mmio)
goto out1;
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!r)
+ goto out2;
+ wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!r)
+ goto out2;
+ wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
/* preserve PSC clock source set up by platform (dev.platform_data
* is already occupied by soc layer)
*/
@@ -330,15 +349,11 @@
platform_set_drvdata(pdev, wd);
ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv);
- if (ret)
- goto out1;
-
- /* finally add the DMA device for this PSC */
- wd->dmapd = au1xpsc_pcm_add(pdev);
- if (wd->dmapd)
+ if (!ret)
return 0;
- snd_soc_unregister_dai(&pdev->dev);
+out2:
+ iounmap(wd->mmio);
out1:
release_mem_region(r->start, resource_size(r));
out0:
@@ -351,9 +366,6 @@
struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (wd->dmapd)
- au1xpsc_pcm_destroy(wd->dmapd);
-
snd_soc_unregister_dai(&pdev->dev);
au_writel(0, I2S_CFG(wd));
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
index b30eadd..b16b2e0 100644
--- a/sound/soc/au1x/psc.h
+++ b/sound/soc/au1x/psc.h
@@ -1,7 +1,7 @@
/*
- * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ * Alchemy ALSA ASoC audio support.
*
- * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ * (c) 2007-2011 MSC Vertriebsges.m.b.H.,
* Manuel Lauss <manuel.lauss@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
@@ -13,10 +13,6 @@
#ifndef _AU1X_PCM_H
#define _AU1X_PCM_H
-/* DBDMA helpers */
-extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev);
-extern void au1xpsc_pcm_destroy(struct platform_device *dmapd);
-
struct au1xpsc_audio_data {
void __iomem *mmio;
@@ -27,15 +23,9 @@
unsigned long pm[2];
struct mutex lock;
- struct platform_device *dmapd;
+ int dmaids[2];
};
-#define PCM_TX 0
-#define PCM_RX 1
-
-#define SUBSTREAM_TYPE(substream) \
- ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX)
-
/* easy access macros */
#define PSC_CTRL(x) ((unsigned long)((x)->mmio) + PSC_CTRL_OFFSET)
#define PSC_SEL(x) ((unsigned long)((x)->mmio) + PSC_SEL_OFFSET)
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 7e4066e..666fae6 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -20,6 +20,7 @@
#include <linux/regulator/driver.h>
#include <linux/regulator/machine.h>
#include <linux/regulator/consumer.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/tlv.h>
#include <sound/pcm.h>
@@ -1436,10 +1437,17 @@
MODULE_DEVICE_TABLE(i2c, sgtl5000_id);
+static const struct of_device_id sgtl5000_dt_ids[] = {
+ { .compatible = "fsl,sgtl5000", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(i2c, sgtl5000_dt_ids);
+
static struct i2c_driver sgtl5000_i2c_driver = {
.driver = {
.name = "sgtl5000",
.owner = THIS_MODULE,
+ .of_match_table = sgtl5000_dt_ids,
},
.probe = sgtl5000_i2c_probe,
.remove = __devexit_p(sgtl5000_i2c_remove),
diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c
index bcc2089..bbcf9ec 100644
--- a/sound/soc/codecs/wm1250-ev1.c
+++ b/sound/soc/codecs/wm1250-ev1.c
@@ -56,8 +56,26 @@
};
static int __devinit wm1250_ev1_probe(struct i2c_client *i2c,
- const struct i2c_device_id *id)
+ const struct i2c_device_id *i2c_id)
{
+ int ret, id, board, rev;
+
+ board = i2c_smbus_read_byte_data(i2c, 0);
+ if (board < 0) {
+ dev_err(&i2c->dev, "Failed to read ID: %d\n", ret);
+ return ret;
+ }
+
+ id = (board & 0xfe) >> 2;
+ rev = board & 0x3;
+
+ if (id != 1) {
+ dev_err(&i2c->dev, "Unknown board ID %d\n", id);
+ return -ENODEV;
+ }
+
+ dev_info(&i2c->dev, "revision %d\n", rev);
+
return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1,
&wm1250_ev1_dai, 1);
}
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 4fd4d8d..1312009 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -551,7 +551,7 @@
static struct i2c_driver wm8523_i2c_driver = {
.driver = {
- .name = "wm8523-codec",
+ .name = "wm8523",
.owner = THIS_MODULE,
},
.probe = wm8523_i2c_probe,
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 4bbc0a79..95ac665 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -943,7 +943,7 @@
static struct i2c_driver wm8580_i2c_driver = {
.driver = {
- .name = "wm8580-codec",
+ .name = "wm8580",
.owner = THIS_MODULE,
},
.probe = wm8580_i2c_probe,
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 76b4361..f76b6fc 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -22,6 +22,7 @@
#include <linux/platform_device.h>
#include <linux/regulator/consumer.h>
#include <linux/spi/spi.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -607,6 +608,13 @@
.num_dapm_routes = ARRAY_SIZE(wm8731_intercon),
};
+static const struct of_device_id wm8731_of_match[] = {
+ { .compatible = "wlf,wm8731", },
+ { }
+};
+
+MODULE_DEVICE_TABLE(of, wm8731_of_match);
+
#if defined(CONFIG_SPI_MASTER)
static int __devinit wm8731_spi_probe(struct spi_device *spi)
{
@@ -638,6 +646,7 @@
.driver = {
.name = "wm8731",
.owner = THIS_MODULE,
+ .of_match_table = wm8731_of_match,
},
.probe = wm8731_spi_probe,
.remove = __devexit_p(wm8731_spi_remove),
@@ -682,6 +691,7 @@
.driver = {
.name = "wm8731",
.owner = THIS_MODULE,
+ .of_match_table = wm8731_of_match,
},
.probe = wm8731_i2c_probe,
.remove = __devexit_p(wm8731_i2c_remove),
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index ffa2ffe..a702550 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1519,7 +1519,7 @@
static struct spi_driver wm8753_spi_driver = {
.driver = {
- .name = "wm8753-codec",
+ .name = "wm8753",
.owner = THIS_MODULE,
},
.probe = wm8753_spi_probe,
@@ -1563,7 +1563,7 @@
static struct i2c_driver wm8753_i2c_driver = {
.driver = {
- .name = "wm8753-codec",
+ .name = "wm8753",
.owner = THIS_MODULE,
},
.probe = wm8753_i2c_probe,
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 6e85b88..f014e56 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1433,7 +1433,8 @@
int ret, i, val;
wm8993->hubs_data.hp_startup_mode = 1;
- wm8993->hubs_data.dcs_codes = -2;
+ wm8993->hubs_data.dcs_codes_l = -2;
+ wm8993->hubs_data.dcs_codes_r = -2;
wm8993->hubs_data.series_startup = 1;
ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 09e680a..fb5c961 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -107,6 +107,7 @@
case WM8994_LDO_2:
case WM8958_DSP2_EXECCONTROL:
case WM8958_MIC_DETECT_3:
+ case WM8994_DC_SERVO_4E:
return 1;
default:
return 0;
@@ -2972,13 +2973,14 @@
switch (wm8994->revision) {
case 2:
case 3:
- wm8994->hubs.dcs_codes = -5;
+ wm8994->hubs.dcs_codes_l = -5;
+ wm8994->hubs.dcs_codes_r = -5;
wm8994->hubs.hp_startup_mode = 1;
wm8994->hubs.dcs_readback_mode = 1;
wm8994->hubs.series_startup = 1;
break;
default:
- wm8994->hubs.dcs_readback_mode = 1;
+ wm8994->hubs.dcs_readback_mode = 2;
break;
}
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index e763c54..017522e 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -18,6 +18,7 @@
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
+#include <linux/mfd/wm8994/registers.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -116,14 +117,23 @@
{
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
s8 offset;
- u16 reg, reg_l, reg_r, dcs_cfg;
+ u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg;
+
+ switch (hubs->dcs_readback_mode) {
+ case 2:
+ dcs_reg = WM8994_DC_SERVO_4E;
+ break;
+ default:
+ dcs_reg = WM8993_DC_SERVO_3;
+ break;
+ }
/* If we're using a digital only path and have a previously
* callibrated DC servo offset stored then use that. */
if (hubs->class_w && hubs->class_w_dcs) {
dev_dbg(codec->dev, "Using cached DC servo offset %x\n",
hubs->class_w_dcs);
- snd_soc_write(codec, WM8993_DC_SERVO_3, hubs->class_w_dcs);
+ snd_soc_write(codec, dcs_reg, hubs->class_w_dcs);
wait_for_dc_servo(codec,
WM8993_DCS_TRIG_DAC_WR_0 |
WM8993_DCS_TRIG_DAC_WR_1);
@@ -154,8 +164,9 @@
reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
& WM8993_DCS_INTEG_CHAN_1_MASK;
break;
+ case 2:
case 1:
- reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
+ reg = snd_soc_read(codec, dcs_reg);
reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
>> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
@@ -168,24 +179,25 @@
dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
/* Apply correction to DC servo result */
- if (hubs->dcs_codes) {
- dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
- hubs->dcs_codes);
+ if (hubs->dcs_codes_l || hubs->dcs_codes_r) {
+ dev_dbg(codec->dev,
+ "Applying %d/%d code DC servo correction\n",
+ hubs->dcs_codes_l, hubs->dcs_codes_r);
/* HPOUT1R */
offset = reg_r;
- offset += hubs->dcs_codes;
+ offset += hubs->dcs_codes_r;
dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
/* HPOUT1L */
offset = reg_l;
- offset += hubs->dcs_codes;
+ offset += hubs->dcs_codes_l;
dcs_cfg |= (u8)offset;
dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg);
/* Do it */
- snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg);
+ snd_soc_write(codec, dcs_reg, dcs_cfg);
wait_for_dc_servo(codec,
WM8993_DCS_TRIG_DAC_WR_0 |
WM8993_DCS_TRIG_DAC_WR_1);
@@ -217,7 +229,7 @@
/* If we're applying an offset correction then updating the
* callibration would be likely to introduce further offsets. */
- if (hubs->dcs_codes || hubs->no_series_update)
+ if (hubs->dcs_codes_l || hubs->dcs_codes_r || hubs->no_series_update)
return ret;
/* Only need to do this if the outputs are active */
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index 676b125..c674c7a 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -23,7 +23,8 @@
/* This *must* be the first element of the codec->private_data struct */
struct wm_hubs_data {
- int dcs_codes;
+ int dcs_codes_l;
+ int dcs_codes_r;
int dcs_readback_mode;
int hp_startup_mode;
int series_startup;
diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig
new file mode 100644
index 0000000..e4ba8d5
--- /dev/null
+++ b/sound/soc/mxs/Kconfig
@@ -0,0 +1,20 @@
+menuconfig SND_MXS_SOC
+ tristate "SoC Audio for Freescale MXS CPUs"
+ depends on ARCH_MXS
+ select SND_PCM
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the MXS SAIF interface.
+
+
+if SND_MXS_SOC
+
+config SND_SOC_MXS_SGTL5000
+ tristate "SoC Audio support for i.MX boards with sgtl5000"
+ depends on I2C
+ select SND_SOC_SGTL5000
+ help
+ Say Y if you want to add support for SoC audio on an MXS board with
+ a sgtl5000 codec.
+
+endif # SND_MXS_SOC
diff --git a/sound/soc/mxs/Makefile b/sound/soc/mxs/Makefile
new file mode 100644
index 0000000..565b5b5
--- /dev/null
+++ b/sound/soc/mxs/Makefile
@@ -0,0 +1,10 @@
+# MXS Platform Support
+snd-soc-mxs-objs := mxs-saif.o
+snd-soc-mxs-pcm-objs := mxs-pcm.o
+
+obj-$(CONFIG_SND_MXS_SOC) += snd-soc-mxs.o snd-soc-mxs-pcm.o
+
+# i.MX Machine Support
+snd-soc-mxs-sgtl5000-objs := mxs-sgtl5000.o
+
+obj-$(CONFIG_SND_SOC_MXS_SGTL5000) += snd-soc-mxs-sgtl5000.o
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
new file mode 100644
index 0000000..dea5aa4
--- /dev/null
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -0,0 +1,359 @@
+/*
+ * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * Based on sound/soc/imx/imx-pcm-dma-mx2.c
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dmaengine.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <mach/dma.h>
+#include "mxs-pcm.h"
+
+static struct snd_pcm_hardware snd_mxs_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_INTERLEAVED,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_3LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 2,
+ .channels_max = 2,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8192,
+ .periods_min = 1,
+ .periods_max = 52,
+ .buffer_bytes_max = 64 * 1024,
+ .fifo_size = 32,
+
+};
+
+static void audio_dma_irq(void *data)
+{
+ struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+ iprtd->offset += iprtd->period_bytes;
+ iprtd->offset %= iprtd->period_bytes * iprtd->periods;
+ snd_pcm_period_elapsed(substream);
+}
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+ struct mxs_pcm_runtime_data *iprtd = param;
+ struct mxs_pcm_dma_params *dma_params = iprtd->dma_params;
+
+ if (!mxs_dma_is_apbx(chan))
+ return false;
+
+ if (chan->chan_id != dma_params->chan_num)
+ return false;
+
+ chan->private = &iprtd->dma_data;
+
+ return true;
+}
+
+static int mxs_dma_alloc(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+ dma_cap_mask_t mask;
+
+ iprtd->dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+ dma_cap_zero(mask);
+ dma_cap_set(DMA_SLAVE, mask);
+ iprtd->dma_data.chan_irq = iprtd->dma_params->chan_irq;
+ iprtd->dma_chan = dma_request_channel(mask, filter, iprtd);
+ if (!iprtd->dma_chan)
+ return -EINVAL;
+
+ return 0;
+}
+
+static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+ unsigned long dma_addr;
+ struct dma_chan *chan;
+ int ret;
+
+ ret = mxs_dma_alloc(substream, params);
+ if (ret)
+ return ret;
+ chan = iprtd->dma_chan;
+
+ iprtd->size = params_buffer_bytes(params);
+ iprtd->periods = params_periods(params);
+ iprtd->period_bytes = params_period_bytes(params);
+ iprtd->offset = 0;
+ iprtd->period_time = HZ / (params_rate(params) /
+ params_period_size(params));
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ dma_addr = runtime->dma_addr;
+
+ iprtd->buf = substream->dma_buffer.area;
+
+ iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr,
+ iprtd->period_bytes * iprtd->periods,
+ iprtd->period_bytes,
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ DMA_TO_DEVICE : DMA_FROM_DEVICE);
+ if (!iprtd->desc) {
+ dev_err(&chan->dev->device, "cannot prepare slave dma\n");
+ return -EINVAL;
+ }
+
+ iprtd->desc->callback = audio_dma_irq;
+ iprtd->desc->callback_param = substream;
+
+ return 0;
+}
+
+static int snd_mxs_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+ if (iprtd->dma_chan) {
+ dma_release_channel(iprtd->dma_chan);
+ iprtd->dma_chan = NULL;
+ }
+
+ return 0;
+}
+
+static int snd_mxs_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ dmaengine_submit(iprtd->desc);
+
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ dmaengine_terminate_all(iprtd->dma_chan);
+
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_mxs_pcm_pointer(
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+ return bytes_to_frames(substream->runtime, iprtd->offset);
+}
+
+static int snd_mxs_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd;
+ int ret;
+
+ iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL);
+ if (iprtd == NULL)
+ return -ENOMEM;
+ runtime->private_data = iprtd;
+
+ ret = snd_pcm_hw_constraint_integer(substream->runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0) {
+ kfree(iprtd);
+ return ret;
+ }
+
+ snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware);
+
+ return 0;
+}
+
+static int snd_mxs_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+ kfree(iprtd);
+
+ return 0;
+}
+
+static int snd_mxs_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops mxs_pcm_ops = {
+ .open = snd_mxs_open,
+ .close = snd_mxs_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_mxs_pcm_hw_params,
+ .hw_free = snd_mxs_pcm_hw_free,
+ .trigger = snd_mxs_pcm_trigger,
+ .pointer = snd_mxs_pcm_pointer,
+ .mmap = snd_mxs_pcm_mmap,
+};
+
+static int mxs_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = snd_mxs_hardware.buffer_bytes_max;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area)
+ return -ENOMEM;
+ buf->bytes = size;
+
+ return 0;
+}
+
+static u64 mxs_pcm_dmamask = DMA_BIT_MASK(32);
+static int mxs_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &mxs_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+ ret = mxs_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ ret = mxs_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+
+out:
+ return ret;
+}
+
+static void mxs_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_writecombine(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+static struct snd_soc_platform_driver mxs_soc_platform = {
+ .ops = &mxs_pcm_ops,
+ .pcm_new = mxs_pcm_new,
+ .pcm_free = mxs_pcm_free,
+};
+
+static int __devinit mxs_soc_platform_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_platform(&pdev->dev, &mxs_soc_platform);
+}
+
+static int __devexit mxs_soc_platform_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+
+ return 0;
+}
+
+static struct platform_driver mxs_pcm_driver = {
+ .driver = {
+ .name = "mxs-pcm-audio",
+ .owner = THIS_MODULE,
+ },
+ .probe = mxs_soc_platform_probe,
+ .remove = __devexit_p(mxs_soc_platform_remove),
+};
+
+static int __init snd_mxs_pcm_init(void)
+{
+ return platform_driver_register(&mxs_pcm_driver);
+}
+module_init(snd_mxs_pcm_init);
+
+static void __exit snd_mxs_pcm_exit(void)
+{
+ platform_driver_unregister(&mxs_pcm_driver);
+}
+module_exit(snd_mxs_pcm_exit);
diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h
new file mode 100644
index 0000000..f55ac4f
--- /dev/null
+++ b/sound/soc/mxs/mxs-pcm.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef _MXS_PCM_H
+#define _MXS_PCM_H
+
+#include <mach/dma.h>
+
+struct mxs_pcm_dma_params {
+ int chan_irq;
+ int chan_num;
+};
+
+struct mxs_pcm_runtime_data {
+ int period_bytes;
+ int periods;
+ int dma;
+ unsigned long offset;
+ unsigned long size;
+ void *buf;
+ int period_time;
+ struct dma_async_tx_descriptor *desc;
+ struct dma_chan *dma_chan;
+ struct mxs_dma_data dma_data;
+ struct mxs_pcm_dma_params *dma_params;
+};
+
+#endif
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
new file mode 100644
index 0000000..0b3adae
--- /dev/null
+++ b/sound/soc/mxs/mxs-saif.c
@@ -0,0 +1,677 @@
+/*
+ * Copyright 2011 Freescale Semiconductor, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <mach/dma.h>
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/mxs.h>
+
+#include "mxs-saif.h"
+
+static struct mxs_saif *mxs_saif[2];
+
+static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+ switch (clk_id) {
+ case MXS_SAIF_MCLK:
+ saif->mclk = freq;
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/*
+ * Set SAIF clock and MCLK
+ */
+static int mxs_saif_set_clk(struct mxs_saif *saif,
+ unsigned int mclk,
+ unsigned int rate)
+{
+ u32 scr;
+ int ret;
+
+ scr = __raw_readl(saif->base + SAIF_CTRL);
+ scr &= ~BM_SAIF_CTRL_BITCLK_MULT_RATE;
+ scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+
+ /*
+ * Set SAIF clock
+ *
+ * The SAIF clock should be either 384*fs or 512*fs.
+ * If MCLK is used, the SAIF clk ratio need to match mclk ratio.
+ * For 32x mclk, set saif clk as 512*fs.
+ * For 48x mclk, set saif clk as 384*fs.
+ *
+ * If MCLK is not used, we just set saif clk to 512*fs.
+ */
+ if (saif->mclk_in_use) {
+ if (mclk % 32 == 0) {
+ scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+ ret = clk_set_rate(saif->clk, 512 * rate);
+ } else if (mclk % 48 == 0) {
+ scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE;
+ ret = clk_set_rate(saif->clk, 384 * rate);
+ } else {
+ /* SAIF MCLK should be either 32x or 48x */
+ return -EINVAL;
+ }
+ } else {
+ ret = clk_set_rate(saif->clk, 512 * rate);
+ scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+ }
+
+ if (ret)
+ return ret;
+
+ if (!saif->mclk_in_use) {
+ __raw_writel(scr, saif->base + SAIF_CTRL);
+ return 0;
+ }
+
+ /*
+ * Program the over-sample rate for MCLK output
+ *
+ * The available MCLK range is 32x, 48x... 512x. The rate
+ * could be from 8kHz to 192kH.
+ */
+ switch (mclk / rate) {
+ case 32:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(4);
+ break;
+ case 64:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(3);
+ break;
+ case 128:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(2);
+ break;
+ case 256:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(1);
+ break;
+ case 512:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(0);
+ break;
+ case 48:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(3);
+ break;
+ case 96:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(2);
+ break;
+ case 192:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(1);
+ break;
+ case 384:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ __raw_writel(scr, saif->base + SAIF_CTRL);
+
+ return 0;
+}
+
+/*
+ * Put and disable MCLK.
+ */
+int mxs_saif_put_mclk(unsigned int saif_id)
+{
+ struct mxs_saif *saif = mxs_saif[saif_id];
+ u32 stat;
+
+ if (!saif)
+ return -EINVAL;
+
+ stat = __raw_readl(saif->base + SAIF_STAT);
+ if (stat & BM_SAIF_STAT_BUSY) {
+ dev_err(saif->dev, "error: busy\n");
+ return -EBUSY;
+ }
+
+ clk_disable(saif->clk);
+
+ /* disable MCLK output */
+ __raw_writel(BM_SAIF_CTRL_CLKGATE,
+ saif->base + SAIF_CTRL + MXS_SET_ADDR);
+ __raw_writel(BM_SAIF_CTRL_RUN,
+ saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+ saif->mclk_in_use = 0;
+ return 0;
+}
+
+/*
+ * Get MCLK and set clock rate, then enable it
+ *
+ * This interface is used for codecs who are using MCLK provided
+ * by saif.
+ */
+int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk,
+ unsigned int rate)
+{
+ struct mxs_saif *saif = mxs_saif[saif_id];
+ u32 stat;
+ int ret;
+
+ if (!saif)
+ return -EINVAL;
+
+ stat = __raw_readl(saif->base + SAIF_STAT);
+ if (stat & BM_SAIF_STAT_BUSY) {
+ dev_err(saif->dev, "error: busy\n");
+ return -EBUSY;
+ }
+
+ /* Clear Reset */
+ __raw_writel(BM_SAIF_CTRL_SFTRST,
+ saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+ saif->mclk_in_use = 1;
+ ret = mxs_saif_set_clk(saif, mclk, rate);
+ if (ret)
+ return ret;
+
+ ret = clk_enable(saif->clk);
+ if (ret)
+ return ret;
+
+ /* enable MCLK output */
+ __raw_writel(BM_SAIF_CTRL_CLKGATE,
+ saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+ __raw_writel(BM_SAIF_CTRL_RUN,
+ saif->base + SAIF_CTRL + MXS_SET_ADDR);
+
+ return 0;
+}
+
+/*
+ * SAIF DAI format configuration.
+ * Should only be called when port is inactive.
+ */
+static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+ u32 scr, stat;
+ u32 scr0;
+ struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+ stat = __raw_readl(saif->base + SAIF_STAT);
+ if (stat & BM_SAIF_STAT_BUSY) {
+ dev_err(cpu_dai->dev, "error: busy\n");
+ return -EBUSY;
+ }
+
+ scr0 = __raw_readl(saif->base + SAIF_CTRL);
+ scr0 = scr0 & ~BM_SAIF_CTRL_BITCLK_EDGE & ~BM_SAIF_CTRL_LRCLK_POLARITY \
+ & ~BM_SAIF_CTRL_JUSTIFY & ~BM_SAIF_CTRL_DELAY;
+ scr = 0;
+
+ /* DAI mode */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ /* data frame low 1clk before data */
+ scr |= BM_SAIF_CTRL_DELAY;
+ scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ /* data frame high with data */
+ scr &= ~BM_SAIF_CTRL_DELAY;
+ scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+ scr &= ~BM_SAIF_CTRL_JUSTIFY;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* DAI clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_IF:
+ scr |= BM_SAIF_CTRL_BITCLK_EDGE;
+ scr |= BM_SAIF_CTRL_LRCLK_POLARITY;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ scr |= BM_SAIF_CTRL_BITCLK_EDGE;
+ scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ scr &= ~BM_SAIF_CTRL_BITCLK_EDGE;
+ scr |= BM_SAIF_CTRL_LRCLK_POLARITY;
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ scr &= ~BM_SAIF_CTRL_BITCLK_EDGE;
+ scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+ break;
+ }
+
+ /*
+ * Note: We simply just support master mode since SAIF TX can only
+ * work as master.
+ */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ scr &= ~BM_SAIF_CTRL_SLAVE_MODE;
+ __raw_writel(scr | scr0, saif->base + SAIF_CTRL);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int mxs_saif_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+ snd_soc_dai_set_dma_data(cpu_dai, substream, &saif->dma_param);
+
+ /* clear error status to 0 for each re-open */
+ saif->fifo_underrun = 0;
+ saif->fifo_overrun = 0;
+
+ /* Clear Reset for normal operations */
+ __raw_writel(BM_SAIF_CTRL_SFTRST,
+ saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+ return 0;
+}
+
+/*
+ * Should only be called when port is inactive.
+ * although can be called multiple times by upper layers.
+ */
+static int mxs_saif_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+ u32 scr, stat;
+ int ret;
+
+ /* mclk should already be set */
+ if (!saif->mclk && saif->mclk_in_use) {
+ dev_err(cpu_dai->dev, "set mclk first\n");
+ return -EINVAL;
+ }
+
+ stat = __raw_readl(saif->base + SAIF_STAT);
+ if (stat & BM_SAIF_STAT_BUSY) {
+ dev_err(cpu_dai->dev, "error: busy\n");
+ return -EBUSY;
+ }
+
+ /*
+ * Set saif clk based on sample rate.
+ * If mclk is used, we also set mclk, if not, saif->mclk is
+ * default 0, means not used.
+ */
+ ret = mxs_saif_set_clk(saif, saif->mclk, params_rate(params));
+ if (ret) {
+ dev_err(cpu_dai->dev, "unable to get proper clk\n");
+ return ret;
+ }
+
+ scr = __raw_readl(saif->base + SAIF_CTRL);
+
+ scr &= ~BM_SAIF_CTRL_WORD_LENGTH;
+ scr &= ~BM_SAIF_CTRL_BITCLK_48XFS_ENABLE;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ scr |= BF_SAIF_CTRL_WORD_LENGTH(0);
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ scr |= BF_SAIF_CTRL_WORD_LENGTH(4);
+ scr |= BM_SAIF_CTRL_BITCLK_48XFS_ENABLE;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ scr |= BF_SAIF_CTRL_WORD_LENGTH(8);
+ scr |= BM_SAIF_CTRL_BITCLK_48XFS_ENABLE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* Tx/Rx config */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* enable TX mode */
+ scr &= ~BM_SAIF_CTRL_READ_MODE;
+ } else {
+ /* enable RX mode */
+ scr |= BM_SAIF_CTRL_READ_MODE;
+ }
+
+ __raw_writel(scr, saif->base + SAIF_CTRL);
+ return 0;
+}
+
+static int mxs_saif_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+ /* clear clock gate */
+ __raw_writel(BM_SAIF_CTRL_CLKGATE,
+ saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+ /* enable FIFO error irqs */
+ __raw_writel(BM_SAIF_CTRL_FIFO_ERROR_IRQ_EN,
+ saif->base + SAIF_CTRL + MXS_SET_ADDR);
+
+ return 0;
+}
+
+static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ dev_dbg(cpu_dai->dev, "start\n");
+
+ clk_enable(saif->clk);
+ if (!saif->mclk_in_use)
+ __raw_writel(BM_SAIF_CTRL_RUN,
+ saif->base + SAIF_CTRL + MXS_SET_ADDR);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /*
+ * write a data to saif data register to trigger
+ * the transfer
+ */
+ __raw_writel(0, saif->base + SAIF_DATA);
+ } else {
+ /*
+ * read a data from saif data register to trigger
+ * the receive
+ */
+ __raw_readl(saif->base + SAIF_DATA);
+ }
+
+ dev_dbg(cpu_dai->dev, "CTRL 0x%x STAT 0x%x\n",
+ __raw_readl(saif->base + SAIF_CTRL),
+ __raw_readl(saif->base + SAIF_STAT));
+
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ dev_dbg(cpu_dai->dev, "stop\n");
+
+ clk_disable(saif->clk);
+ if (!saif->mclk_in_use)
+ __raw_writel(BM_SAIF_CTRL_RUN,
+ saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+#define MXS_SAIF_RATES SNDRV_PCM_RATE_8000_192000
+#define MXS_SAIF_FORMATS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops mxs_saif_dai_ops = {
+ .startup = mxs_saif_startup,
+ .trigger = mxs_saif_trigger,
+ .prepare = mxs_saif_prepare,
+ .hw_params = mxs_saif_hw_params,
+ .set_sysclk = mxs_saif_set_dai_sysclk,
+ .set_fmt = mxs_saif_set_dai_fmt,
+};
+
+static int mxs_saif_dai_probe(struct snd_soc_dai *dai)
+{
+ struct mxs_saif *saif = dev_get_drvdata(dai->dev);
+
+ snd_soc_dai_set_drvdata(dai, saif);
+
+ return 0;
+}
+
+static struct snd_soc_dai_driver mxs_saif_dai = {
+ .name = "mxs-saif",
+ .probe = mxs_saif_dai_probe,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = MXS_SAIF_RATES,
+ .formats = MXS_SAIF_FORMATS,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = MXS_SAIF_RATES,
+ .formats = MXS_SAIF_FORMATS,
+ },
+ .ops = &mxs_saif_dai_ops,
+};
+
+static irqreturn_t mxs_saif_irq(int irq, void *dev_id)
+{
+ struct mxs_saif *saif = dev_id;
+ unsigned int stat;
+
+ stat = __raw_readl(saif->base + SAIF_STAT);
+ if (!(stat & (BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ |
+ BM_SAIF_STAT_FIFO_OVERFLOW_IRQ)))
+ return IRQ_NONE;
+
+ if (stat & BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ) {
+ dev_dbg(saif->dev, "underrun!!! %d\n", ++saif->fifo_underrun);
+ __raw_writel(BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ,
+ saif->base + SAIF_STAT + MXS_CLR_ADDR);
+ }
+
+ if (stat & BM_SAIF_STAT_FIFO_OVERFLOW_IRQ) {
+ dev_dbg(saif->dev, "overrun!!! %d\n", ++saif->fifo_overrun);
+ __raw_writel(BM_SAIF_STAT_FIFO_OVERFLOW_IRQ,
+ saif->base + SAIF_STAT + MXS_CLR_ADDR);
+ }
+
+ dev_dbg(saif->dev, "SAIF_CTRL %x SAIF_STAT %x\n",
+ __raw_readl(saif->base + SAIF_CTRL),
+ __raw_readl(saif->base + SAIF_STAT));
+
+ return IRQ_HANDLED;
+}
+
+static int mxs_saif_probe(struct platform_device *pdev)
+{
+ struct resource *res;
+ struct mxs_saif *saif;
+ int ret = 0;
+
+ saif = kzalloc(sizeof(*saif), GFP_KERNEL);
+ if (!saif)
+ return -ENOMEM;
+
+ if (pdev->id >= ARRAY_SIZE(mxs_saif))
+ return -EINVAL;
+ mxs_saif[pdev->id] = saif;
+
+ saif->clk = clk_get(&pdev->dev, NULL);
+ if (IS_ERR(saif->clk)) {
+ ret = PTR_ERR(saif->clk);
+ dev_err(&pdev->dev, "Cannot get the clock: %d\n",
+ ret);
+ goto failed_clk;
+ }
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!res) {
+ ret = -ENODEV;
+ dev_err(&pdev->dev, "failed to get io resource: %d\n",
+ ret);
+ goto failed_get_resource;
+ }
+
+ if (!request_mem_region(res->start, resource_size(res), "mxs-saif")) {
+ dev_err(&pdev->dev, "request_mem_region failed\n");
+ ret = -EBUSY;
+ goto failed_get_resource;
+ }
+
+ saif->base = ioremap(res->start, resource_size(res));
+ if (!saif->base) {
+ dev_err(&pdev->dev, "ioremap failed\n");
+ ret = -ENODEV;
+ goto failed_ioremap;
+ }
+
+ res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!res) {
+ ret = -ENODEV;
+ dev_err(&pdev->dev, "failed to get dma resource: %d\n",
+ ret);
+ goto failed_ioremap;
+ }
+ saif->dma_param.chan_num = res->start;
+
+ saif->irq = platform_get_irq(pdev, 0);
+ if (saif->irq < 0) {
+ ret = saif->irq;
+ dev_err(&pdev->dev, "failed to get irq resource: %d\n",
+ ret);
+ goto failed_get_irq1;
+ }
+
+ saif->dev = &pdev->dev;
+ ret = request_irq(saif->irq, mxs_saif_irq, 0, "mxs-saif", saif);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to request irq\n");
+ goto failed_get_irq1;
+ }
+
+ saif->dma_param.chan_irq = platform_get_irq(pdev, 1);
+ if (saif->dma_param.chan_irq < 0) {
+ ret = saif->dma_param.chan_irq;
+ dev_err(&pdev->dev, "failed to get dma irq resource: %d\n",
+ ret);
+ goto failed_get_irq2;
+ }
+
+ platform_set_drvdata(pdev, saif);
+
+ ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai);
+ if (ret) {
+ dev_err(&pdev->dev, "register DAI failed\n");
+ goto failed_register;
+ }
+
+ saif->soc_platform_pdev = platform_device_alloc(
+ "mxs-pcm-audio", pdev->id);
+ if (!saif->soc_platform_pdev) {
+ ret = -ENOMEM;
+ goto failed_pdev_alloc;
+ }
+
+ platform_set_drvdata(saif->soc_platform_pdev, saif);
+ ret = platform_device_add(saif->soc_platform_pdev);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to add soc platform device\n");
+ goto failed_pdev_add;
+ }
+
+ return 0;
+
+failed_pdev_add:
+ platform_device_put(saif->soc_platform_pdev);
+failed_pdev_alloc:
+ snd_soc_unregister_dai(&pdev->dev);
+failed_register:
+failed_get_irq2:
+ free_irq(saif->irq, saif);
+failed_get_irq1:
+ iounmap(saif->base);
+failed_ioremap:
+ release_mem_region(res->start, resource_size(res));
+failed_get_resource:
+ clk_put(saif->clk);
+failed_clk:
+ kfree(saif);
+
+ return ret;
+}
+
+static int __devexit mxs_saif_remove(struct platform_device *pdev)
+{
+ struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ struct mxs_saif *saif = platform_get_drvdata(pdev);
+
+ platform_device_unregister(saif->soc_platform_pdev);
+
+ snd_soc_unregister_dai(&pdev->dev);
+
+ iounmap(saif->base);
+ release_mem_region(res->start, resource_size(res));
+ free_irq(saif->irq, saif);
+
+ clk_put(saif->clk);
+ kfree(saif);
+
+ return 0;
+}
+
+static struct platform_driver mxs_saif_driver = {
+ .probe = mxs_saif_probe,
+ .remove = __devexit_p(mxs_saif_remove),
+
+ .driver = {
+ .name = "mxs-saif",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init mxs_saif_init(void)
+{
+ return platform_driver_register(&mxs_saif_driver);
+}
+
+static void __exit mxs_saif_exit(void)
+{
+ platform_driver_unregister(&mxs_saif_driver);
+}
+
+module_init(mxs_saif_init);
+module_exit(mxs_saif_exit);
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("MXS ASoC SAIF driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h
new file mode 100644
index 0000000..0e2ff8c
--- /dev/null
+++ b/sound/soc/mxs/mxs-saif.h
@@ -0,0 +1,130 @@
+/*
+ * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+
+#ifndef _MXS_SAIF_H
+#define _MXS_SAIF_H
+
+#define SAIF_CTRL 0x0
+#define SAIF_STAT 0x10
+#define SAIF_DATA 0x20
+#define SAIF_VERSION 0X30
+
+/* SAIF_CTRL */
+#define BM_SAIF_CTRL_SFTRST 0x80000000
+#define BM_SAIF_CTRL_CLKGATE 0x40000000
+#define BP_SAIF_CTRL_BITCLK_MULT_RATE 27
+#define BM_SAIF_CTRL_BITCLK_MULT_RATE 0x38000000
+#define BF_SAIF_CTRL_BITCLK_MULT_RATE(v) \
+ (((v) << 27) & BM_SAIF_CTRL_BITCLK_MULT_RATE)
+#define BM_SAIF_CTRL_BITCLK_BASE_RATE 0x04000000
+#define BM_SAIF_CTRL_FIFO_ERROR_IRQ_EN 0x02000000
+#define BM_SAIF_CTRL_FIFO_SERVICE_IRQ_EN 0x01000000
+#define BP_SAIF_CTRL_RSRVD2 21
+#define BM_SAIF_CTRL_RSRVD2 0x00E00000
+
+#define BP_SAIF_CTRL_DMAWAIT_COUNT 16
+#define BM_SAIF_CTRL_DMAWAIT_COUNT 0x001F0000
+#define BF_SAIF_CTRL_DMAWAIT_COUNT(v) \
+ (((v) << 16) & BM_SAIF_CTRL_DMAWAIT_COUNT)
+#define BP_SAIF_CTRL_CHANNEL_NUM_SELECT 14
+#define BM_SAIF_CTRL_CHANNEL_NUM_SELECT 0x0000C000
+#define BF_SAIF_CTRL_CHANNEL_NUM_SELECT(v) \
+ (((v) << 14) & BM_SAIF_CTRL_CHANNEL_NUM_SELECT)
+#define BM_SAIF_CTRL_LRCLK_PULSE 0x00002000
+#define BM_SAIF_CTRL_BIT_ORDER 0x00001000
+#define BM_SAIF_CTRL_DELAY 0x00000800
+#define BM_SAIF_CTRL_JUSTIFY 0x00000400
+#define BM_SAIF_CTRL_LRCLK_POLARITY 0x00000200
+#define BM_SAIF_CTRL_BITCLK_EDGE 0x00000100
+#define BP_SAIF_CTRL_WORD_LENGTH 4
+#define BM_SAIF_CTRL_WORD_LENGTH 0x000000F0
+#define BF_SAIF_CTRL_WORD_LENGTH(v) \
+ (((v) << 4) & BM_SAIF_CTRL_WORD_LENGTH)
+#define BM_SAIF_CTRL_BITCLK_48XFS_ENABLE 0x00000008
+#define BM_SAIF_CTRL_SLAVE_MODE 0x00000004
+#define BM_SAIF_CTRL_READ_MODE 0x00000002
+#define BM_SAIF_CTRL_RUN 0x00000001
+
+/* SAIF_STAT */
+#define BM_SAIF_STAT_PRESENT 0x80000000
+#define BP_SAIF_STAT_RSRVD2 17
+#define BM_SAIF_STAT_RSRVD2 0x7FFE0000
+#define BF_SAIF_STAT_RSRVD2(v) \
+ (((v) << 17) & BM_SAIF_STAT_RSRVD2)
+#define BM_SAIF_STAT_DMA_PREQ 0x00010000
+#define BP_SAIF_STAT_RSRVD1 7
+#define BM_SAIF_STAT_RSRVD1 0x0000FF80
+#define BF_SAIF_STAT_RSRVD1(v) \
+ (((v) << 7) & BM_SAIF_STAT_RSRVD1)
+
+#define BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ 0x00000040
+#define BM_SAIF_STAT_FIFO_OVERFLOW_IRQ 0x00000020
+#define BM_SAIF_STAT_FIFO_SERVICE_IRQ 0x00000010
+#define BP_SAIF_STAT_RSRVD0 1
+#define BM_SAIF_STAT_RSRVD0 0x0000000E
+#define BF_SAIF_STAT_RSRVD0(v) \
+ (((v) << 1) & BM_SAIF_STAT_RSRVD0)
+#define BM_SAIF_STAT_BUSY 0x00000001
+
+/* SAFI_DATA */
+#define BP_SAIF_DATA_PCM_RIGHT 16
+#define BM_SAIF_DATA_PCM_RIGHT 0xFFFF0000
+#define BF_SAIF_DATA_PCM_RIGHT(v) \
+ (((v) << 16) & BM_SAIF_DATA_PCM_RIGHT)
+#define BP_SAIF_DATA_PCM_LEFT 0
+#define BM_SAIF_DATA_PCM_LEFT 0x0000FFFF
+#define BF_SAIF_DATA_PCM_LEFT(v) \
+ (((v) << 0) & BM_SAIF_DATA_PCM_LEFT)
+
+/* SAIF_VERSION */
+#define BP_SAIF_VERSION_MAJOR 24
+#define BM_SAIF_VERSION_MAJOR 0xFF000000
+#define BF_SAIF_VERSION_MAJOR(v) \
+ (((v) << 24) & BM_SAIF_VERSION_MAJOR)
+#define BP_SAIF_VERSION_MINOR 16
+#define BM_SAIF_VERSION_MINOR 0x00FF0000
+#define BF_SAIF_VERSION_MINOR(v) \
+ (((v) << 16) & BM_SAIF_VERSION_MINOR)
+#define BP_SAIF_VERSION_STEP 0
+#define BM_SAIF_VERSION_STEP 0x0000FFFF
+#define BF_SAIF_VERSION_STEP(v) \
+ (((v) << 0) & BM_SAIF_VERSION_STEP)
+
+#define MXS_SAIF_MCLK 0
+
+#include "mxs-pcm.h"
+
+struct mxs_saif {
+ struct device *dev;
+ struct clk *clk;
+ unsigned int mclk;
+ unsigned int mclk_in_use;
+ void __iomem *base;
+ int irq;
+ struct mxs_pcm_dma_params dma_param;
+
+ struct platform_device *soc_platform_pdev;
+ u32 fifo_underrun;
+ u32 fifo_overrun;
+};
+
+extern int mxs_saif_put_mclk(unsigned int saif_id);
+extern int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk,
+ unsigned int rate);
+#endif
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
new file mode 100644
index 0000000..a0d89c9
--- /dev/null
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -0,0 +1,165 @@
+/*
+ * Copyright 2011 Freescale Semiconductor, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-types.h>
+
+#include "../codecs/sgtl5000.h"
+#include "mxs-saif.h"
+
+static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ unsigned int rate = params_rate(params);
+ u32 dai_format, mclk;
+ int ret;
+
+ /* sgtl5000 does not support 512*rate when in 96000 fs */
+ switch (rate) {
+ case 96000:
+ mclk = 256 * rate;
+ break;
+ default:
+ mclk = 512 * rate;
+ break;
+ }
+
+ /* Sgtl5000 sysclk should be >= 8MHz and <= 27M */
+ if (mclk < 8000000 || mclk > 27000000)
+ return -EINVAL;
+
+ /* Set SGTL5000's SYSCLK (provided by SAIF MCLK) */
+ ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk, 0);
+ if (ret)
+ return ret;
+
+ /* The SAIF MCLK should be the same as SGTL5000_SYSCLK */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, MXS_SAIF_MCLK, mclk, 0);
+ if (ret)
+ return ret;
+
+ /* set codec to slave mode */
+ dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS;
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, dai_format);
+ if (ret)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, dai_format);
+ if (ret)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops mxs_sgtl5000_hifi_ops = {
+ .hw_params = mxs_sgtl5000_hw_params,
+};
+
+static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
+ {
+ .name = "HiFi",
+ .stream_name = "HiFi Playback",
+ .codec_dai_name = "sgtl5000",
+ .codec_name = "sgtl5000.0-000a",
+ .cpu_dai_name = "mxs-saif.0",
+ .platform_name = "mxs-pcm-audio.0",
+ .ops = &mxs_sgtl5000_hifi_ops,
+ },
+};
+
+static struct snd_soc_card mxs_sgtl5000 = {
+ .name = "mxs_sgtl5000",
+ .dai_link = mxs_sgtl5000_dai,
+ .num_links = ARRAY_SIZE(mxs_sgtl5000_dai),
+};
+
+static int __devinit mxs_sgtl5000_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &mxs_sgtl5000;
+ int ret;
+
+ /*
+ * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w).
+ * The Sgtl5000 sysclk is derived from saif0 mclk and it's range
+ * should be >= 8MHz and <= 27M.
+ */
+ ret = mxs_saif_get_mclk(0, 44100 * 256, 44100);
+ if (ret)
+ return ret;
+
+ card->dev = &pdev->dev;
+ platform_set_drvdata(pdev, card);
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
+ ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int __devexit mxs_sgtl5000_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ mxs_saif_put_mclk(0);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static struct platform_driver mxs_sgtl5000_audio_driver = {
+ .driver = {
+ .name = "mxs-sgtl5000",
+ .owner = THIS_MODULE,
+ },
+ .probe = mxs_sgtl5000_probe,
+ .remove = __devexit_p(mxs_sgtl5000_remove),
+};
+
+static int __init mxs_sgtl5000_init(void)
+{
+ return platform_driver_register(&mxs_sgtl5000_audio_driver);
+}
+module_init(mxs_sgtl5000_init);
+
+static void __exit mxs_sgtl5000_exit(void)
+{
+ platform_driver_unregister(&mxs_sgtl5000_audio_driver);
+}
+module_exit(mxs_sgtl5000_exit);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("MXS ALSA SoC Machine driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 83ad8ca..ae93aa8 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -105,7 +105,7 @@
if (wordsize + regsize + 2 + 1 != len)
return -EINVAL;
- ret = snd_soc_read(codec , reg);
+ ret = snd_soc_read(codec, reg);
if (ret < 0) {
memset(regbuf, 'X', regsize);
regbuf[regsize] = '\0';
@@ -3141,6 +3141,7 @@
platform->driver = platform_drv;
platform->dapm.dev = dev;
platform->dapm.platform = platform;
+ platform->dapm.stream_event = platform_drv->stream_event;
mutex_lock(&client_mutex);
list_add(&platform->list, &platform_list);
@@ -3253,6 +3254,7 @@
codec->dapm.dev = dev;
codec->dapm.codec = codec;
codec->dapm.seq_notifier = codec_drv->seq_notifier;
+ codec->dapm.stream_event = codec_drv->stream_event;
codec->dev = dev;
codec->driver = codec_drv;
codec->num_dai = num_dai;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7e15914..c265311 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2584,7 +2584,7 @@
{
if (!w->sname || w->dapm != dapm)
continue;
- dev_dbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n",
+ dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n",
w->name, w->sname, stream, event);
if (strstr(w->sname, stream)) {
switch(event) {
@@ -2604,6 +2604,10 @@
}
dapm_power_widgets(dapm, event);
+
+ /* do we need to notify any clients that DAPM stream is complete */
+ if (dapm->stream_event)
+ dapm->stream_event(dapm, event);
}
/**
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index cca490c..e471ed6 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -13,26 +13,13 @@
#include <linux/i2c.h>
#include <linux/spi/spi.h>
+#include <linux/regmap.h>
#include <sound/soc.h>
#include <trace/events/asoc.h>
-#ifdef CONFIG_SPI_MASTER
-static int do_spi_write(void *control, const char *data, int len)
-{
- struct spi_device *spi = control;
- int ret;
-
- ret = spi_write(spi, data, len);
- if (ret < 0)
- return ret;
-
- return len;
-}
-#endif
-
-static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value, const void *data, int len)
+static int hw_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
{
int ret;
@@ -49,13 +36,7 @@
return 0;
}
- ret = codec->hw_write(codec->control_data, data, len);
- if (ret == len)
- return 0;
- if (ret < 0)
- return ret;
- else
- return -EIO;
+ return regmap_write(codec->control_data, reg, value);
}
static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg)
@@ -69,8 +50,11 @@
if (codec->cache_only)
return -1;
- BUG_ON(!codec->hw_read);
- return codec->hw_read(codec, reg);
+ ret = regmap_read(codec->control_data, reg, &val);
+ if (ret == 0)
+ return val;
+ else
+ return ret;
}
ret = snd_soc_cache_read(codec, reg, &val);
@@ -79,183 +63,18 @@
return val;
}
-static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u16 data;
-
- data = cpu_to_be16((reg << 12) | (value & 0xffffff));
-
- return do_hw_write(codec, reg, value, &data, 2);
-}
-
-static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u16 data;
-
- data = cpu_to_be16((reg << 9) | (value & 0x1ff));
-
- return do_hw_write(codec, reg, value, &data, 2);
-}
-
-static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[2];
-
- reg &= 0xff;
- data[0] = reg;
- data[1] = value & 0xff;
-
- return do_hw_write(codec, reg, value, data, 2);
-}
-
-static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[3];
- u16 val = cpu_to_be16(value);
-
- data[0] = reg;
- memcpy(&data[1], &val, sizeof(val));
-
- return do_hw_write(codec, reg, value, data, 3);
-}
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int do_i2c_read(struct snd_soc_codec *codec,
- void *reg, int reglen,
- void *data, int datalen)
-{
- struct i2c_msg xfer[2];
- int ret;
- struct i2c_client *client = codec->control_data;
-
- /* Write register */
- xfer[0].addr = client->addr;
- xfer[0].flags = 0;
- xfer[0].len = reglen;
- xfer[0].buf = reg;
-
- /* Read data */
- xfer[1].addr = client->addr;
- xfer[1].flags = I2C_M_RD;
- xfer[1].len = datalen;
- xfer[1].buf = data;
-
- ret = i2c_transfer(client->adapter, xfer, 2);
- if (ret == 2)
- return 0;
- else if (ret < 0)
- return ret;
- else
- return -EIO;
-}
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u8 reg = r;
- u8 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 1, &data, 1);
- if (ret < 0)
- return 0;
- return data;
-}
-#else
-#define snd_soc_8_8_read_i2c NULL
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u8 reg = r;
- u16 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 1, &data, 2);
- if (ret < 0)
- return 0;
- return (data >> 8) | ((data & 0xff) << 8);
-}
-#else
-#define snd_soc_8_16_read_i2c NULL
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u16 reg = r;
- u8 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 2, &data, 1);
- if (ret < 0)
- return 0;
- return data;
-}
-#else
-#define snd_soc_16_8_read_i2c NULL
-#endif
-
-static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[3];
- u16 rval = cpu_to_be16(reg);
-
- memcpy(data, &rval, sizeof(rval));
- data[2] = value;
-
- return do_hw_write(codec, reg, value, data, 3);
-}
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u16 reg = cpu_to_be16(r);
- u16 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 2, &data, 2);
- if (ret < 0)
- return 0;
- return be16_to_cpu(data);
-}
-#else
-#define snd_soc_16_16_read_i2c NULL
-#endif
-
-static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u16 data[2];
-
- data[0] = cpu_to_be16(reg);
- data[1] = cpu_to_be16(value);
-
- return do_hw_write(codec, reg, value, data, sizeof(data));
-}
-
/* Primitive bulk write support for soc-cache. The data pointed to by
- * `data' needs to already be in the form the hardware expects
- * including any leading register specific data. Any data written
- * through this function will not go through the cache as it only
- * handles writing to volatile or out of bounds registers.
+ * `data' needs to already be in the form the hardware expects. Any
+ * data written through this function will not go through the cache as
+ * it only handles writing to volatile or out of bounds registers.
+ *
+ * This is currently only supported for devices using the regmap API
+ * wrappers.
*/
-static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg,
+static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec,
+ unsigned int reg,
const void *data, size_t len)
{
- int ret;
-
/* To ensure that we don't get out of sync with the cache, check
* whether the base register is volatile or if we've directly asked
* to bypass the cache. Out of bounds registers are considered
@@ -266,66 +85,9 @@
&& reg < codec->driver->reg_cache_size)
return -EINVAL;
- switch (codec->control_type) {
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
- case SND_SOC_I2C:
- ret = i2c_master_send(to_i2c_client(codec->dev), data, len);
- break;
-#endif
-#if defined(CONFIG_SPI_MASTER)
- case SND_SOC_SPI:
- ret = spi_write(to_spi_device(codec->dev), data, len);
- break;
-#endif
- default:
- BUG();
- }
-
- if (ret == len)
- return 0;
- if (ret < 0)
- return ret;
- else
- return -EIO;
+ return regmap_raw_write(codec->control_data, reg, data, len);
}
-static struct {
- int addr_bits;
- int data_bits;
- int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
- unsigned int (*read)(struct snd_soc_codec *, unsigned int);
- unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
-} io_types[] = {
- {
- .addr_bits = 4, .data_bits = 12,
- .write = snd_soc_4_12_write,
- },
- {
- .addr_bits = 7, .data_bits = 9,
- .write = snd_soc_7_9_write,
- },
- {
- .addr_bits = 8, .data_bits = 8,
- .write = snd_soc_8_8_write,
- .i2c_read = snd_soc_8_8_read_i2c,
- },
- {
- .addr_bits = 8, .data_bits = 16,
- .write = snd_soc_8_16_write,
- .i2c_read = snd_soc_8_16_read_i2c,
- },
- {
- .addr_bits = 16, .data_bits = 8,
- .write = snd_soc_16_8_write,
- .i2c_read = snd_soc_16_8_read_i2c,
- },
- {
- .addr_bits = 16, .data_bits = 16,
- .write = snd_soc_16_16_write,
- .i2c_read = snd_soc_16_16_read_i2c,
- },
-};
-
/**
* snd_soc_codec_set_cache_io: Set up standard I/O functions.
*
@@ -349,47 +111,38 @@
int addr_bits, int data_bits,
enum snd_soc_control_type control)
{
- int i;
+ struct regmap_config config;
- for (i = 0; i < ARRAY_SIZE(io_types); i++)
- if (io_types[i].addr_bits == addr_bits &&
- io_types[i].data_bits == data_bits)
- break;
- if (i == ARRAY_SIZE(io_types)) {
- printk(KERN_ERR
- "No I/O functions for %d bit address %d bit data\n",
- addr_bits, data_bits);
- return -EINVAL;
- }
-
- codec->write = io_types[i].write;
+ memset(&config, 0, sizeof(config));
+ codec->write = hw_write;
codec->read = hw_read;
codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
+ config.reg_bits = addr_bits;
+ config.val_bits = data_bits;
+
switch (control) {
case SND_SOC_I2C:
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
- codec->hw_write = (hw_write_t)i2c_master_send;
-#endif
- if (io_types[i].i2c_read)
- codec->hw_read = io_types[i].i2c_read;
-
- codec->control_data = container_of(codec->dev,
- struct i2c_client,
- dev);
+ codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev),
+ &config);
break;
case SND_SOC_SPI:
-#ifdef CONFIG_SPI_MASTER
- codec->hw_write = do_spi_write;
-#endif
-
- codec->control_data = container_of(codec->dev,
- struct spi_device,
- dev);
+ codec->control_data = regmap_init_spi(to_spi_device(codec->dev),
+ &config);
break;
+
+ case SND_SOC_REGMAP:
+ /* Device has made its own regmap arrangements */
+ break;
+
+ default:
+ return -EINVAL;
}
+ if (IS_ERR(codec->control_data))
+ return PTR_ERR(codec->control_data);
+
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);