Merge branch 'topic/digital-mixing' into for-2.6.32
diff --git a/arch/arm/mach-omap2/mcbsp.c b/arch/arm/mach-omap2/mcbsp.c
index a5c0f04..7d22caf 100644
--- a/arch/arm/mach-omap2/mcbsp.c
+++ b/arch/arm/mach-omap2/mcbsp.c
@@ -129,6 +129,7 @@
.rx_irq = INT_24XX_MCBSP1_IRQ_RX,
.tx_irq = INT_24XX_MCBSP1_IRQ_TX,
.ops = &omap2_mcbsp_ops,
+ .buffer_size = 0x6F,
},
{
.phys_base = OMAP34XX_MCBSP2_BASE,
@@ -137,6 +138,7 @@
.rx_irq = INT_24XX_MCBSP2_IRQ_RX,
.tx_irq = INT_24XX_MCBSP2_IRQ_TX,
.ops = &omap2_mcbsp_ops,
+ .buffer_size = 0x3FF,
},
{
.phys_base = OMAP34XX_MCBSP3_BASE,
@@ -145,6 +147,7 @@
.rx_irq = INT_24XX_MCBSP3_IRQ_RX,
.tx_irq = INT_24XX_MCBSP3_IRQ_TX,
.ops = &omap2_mcbsp_ops,
+ .buffer_size = 0x6F,
},
{
.phys_base = OMAP34XX_MCBSP4_BASE,
@@ -153,6 +156,7 @@
.rx_irq = INT_24XX_MCBSP4_IRQ_RX,
.tx_irq = INT_24XX_MCBSP4_IRQ_TX,
.ops = &omap2_mcbsp_ops,
+ .buffer_size = 0x6F,
},
{
.phys_base = OMAP34XX_MCBSP5_BASE,
@@ -161,6 +165,7 @@
.rx_irq = INT_24XX_MCBSP5_IRQ_RX,
.tx_irq = INT_24XX_MCBSP5_IRQ_TX,
.ops = &omap2_mcbsp_ops,
+ .buffer_size = 0x6F,
},
};
#define OMAP34XX_MCBSP_PDATA_SZ ARRAY_SIZE(omap34xx_mcbsp_pdata)
diff --git a/arch/arm/plat-omap/include/mach/mcbsp.h b/arch/arm/plat-omap/include/mach/mcbsp.h
index 57249bb..70e950e 100644
--- a/arch/arm/plat-omap/include/mach/mcbsp.h
+++ b/arch/arm/plat-omap/include/mach/mcbsp.h
@@ -134,6 +134,11 @@
#define OMAP_MCBSP_REG_XCERG 0x74
#define OMAP_MCBSP_REG_XCERH 0x78
#define OMAP_MCBSP_REG_SYSCON 0x8C
+#define OMAP_MCBSP_REG_THRSH2 0x90
+#define OMAP_MCBSP_REG_THRSH1 0x94
+#define OMAP_MCBSP_REG_IRQST 0xA0
+#define OMAP_MCBSP_REG_IRQEN 0xA4
+#define OMAP_MCBSP_REG_WAKEUPEN 0xA8
#define OMAP_MCBSP_REG_XCCR 0xAC
#define OMAP_MCBSP_REG_RCCR 0xB0
@@ -249,8 +254,27 @@
#define RDISABLE 0x0001
/********************** McBSP SYSCONFIG bit definitions ********************/
+#define CLOCKACTIVITY(value) ((value)<<8)
+#define SIDLEMODE(value) ((value)<<3)
+#define ENAWAKEUP 0x0004
#define SOFTRST 0x0002
+/********************** McBSP DMA operating modes **************************/
+#define MCBSP_DMA_MODE_ELEMENT 0
+#define MCBSP_DMA_MODE_THRESHOLD 1
+#define MCBSP_DMA_MODE_FRAME 2
+
+/********************** McBSP WAKEUPEN bit definitions *********************/
+#define XEMPTYEOFEN 0x4000
+#define XRDYEN 0x0400
+#define XEOFEN 0x0200
+#define XFSXEN 0x0100
+#define XSYNCERREN 0x0080
+#define RRDYEN 0x0008
+#define REOFEN 0x0004
+#define RFSREN 0x0002
+#define RSYNCERREN 0x0001
+
/* we don't do multichannel for now */
struct omap_mcbsp_reg_cfg {
u16 spcr2;
@@ -344,6 +368,9 @@
u8 dma_rx_sync, dma_tx_sync;
u16 rx_irq, tx_irq;
struct omap_mcbsp_ops *ops;
+#ifdef CONFIG_ARCH_OMAP34XX
+ u16 buffer_size;
+#endif
};
struct omap_mcbsp {
@@ -377,6 +404,11 @@
struct omap_mcbsp_platform_data *pdata;
struct clk *iclk;
struct clk *fclk;
+#ifdef CONFIG_ARCH_OMAP34XX
+ int dma_op_mode;
+ u16 max_tx_thres;
+ u16 max_rx_thres;
+#endif
};
extern struct omap_mcbsp **mcbsp_ptr;
extern int omap_mcbsp_count;
@@ -385,10 +417,27 @@
void omap_mcbsp_register_board_cfg(struct omap_mcbsp_platform_data *config,
int size);
void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg * config);
+#ifdef CONFIG_ARCH_OMAP34XX
+void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold);
+void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold);
+u16 omap_mcbsp_get_max_tx_threshold(unsigned int id);
+u16 omap_mcbsp_get_max_rx_threshold(unsigned int id);
+int omap_mcbsp_get_dma_op_mode(unsigned int id);
+#else
+static inline void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold)
+{ }
+static inline void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold)
+{ }
+static inline u16 omap_mcbsp_get_max_tx_threshold(unsigned int id) { return 0; }
+static inline u16 omap_mcbsp_get_max_rx_threshold(unsigned int id) { return 0; }
+static inline int omap_mcbsp_get_dma_op_mode(unsigned int id) { return 0; }
+#endif
int omap_mcbsp_request(unsigned int id);
void omap_mcbsp_free(unsigned int id);
void omap_mcbsp_start(unsigned int id, int tx, int rx);
void omap_mcbsp_stop(unsigned int id, int tx, int rx);
+void omap_mcbsp_xmit_enable(unsigned int id, u8 enable);
+void omap_mcbsp_recv_enable(unsigned int id, u8 enable);
void omap_mcbsp_xmit_word(unsigned int id, u32 word);
u32 omap_mcbsp_recv_word(unsigned int id);
diff --git a/arch/arm/plat-omap/mcbsp.c b/arch/arm/plat-omap/mcbsp.c
index a3d2313..b63a720 100644
--- a/arch/arm/plat-omap/mcbsp.c
+++ b/arch/arm/plat-omap/mcbsp.c
@@ -198,6 +198,170 @@
}
EXPORT_SYMBOL(omap_mcbsp_config);
+#ifdef CONFIG_ARCH_OMAP34XX
+/*
+ * omap_mcbsp_set_tx_threshold configures how to deal
+ * with transmit threshold. the threshold value and handler can be
+ * configure in here.
+ */
+void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold)
+{
+ struct omap_mcbsp *mcbsp;
+ void __iomem *io_base;
+
+ if (!cpu_is_omap34xx())
+ return;
+
+ if (!omap_mcbsp_check_valid_id(id)) {
+ printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1);
+ return;
+ }
+ mcbsp = id_to_mcbsp_ptr(id);
+ io_base = mcbsp->io_base;
+
+ OMAP_MCBSP_WRITE(io_base, THRSH2, threshold);
+}
+EXPORT_SYMBOL(omap_mcbsp_set_tx_threshold);
+
+/*
+ * omap_mcbsp_set_rx_threshold configures how to deal
+ * with receive threshold. the threshold value and handler can be
+ * configure in here.
+ */
+void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold)
+{
+ struct omap_mcbsp *mcbsp;
+ void __iomem *io_base;
+
+ if (!cpu_is_omap34xx())
+ return;
+
+ if (!omap_mcbsp_check_valid_id(id)) {
+ printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1);
+ return;
+ }
+ mcbsp = id_to_mcbsp_ptr(id);
+ io_base = mcbsp->io_base;
+
+ OMAP_MCBSP_WRITE(io_base, THRSH1, threshold);
+}
+EXPORT_SYMBOL(omap_mcbsp_set_rx_threshold);
+
+/*
+ * omap_mcbsp_get_max_tx_thres just return the current configured
+ * maximum threshold for transmission
+ */
+u16 omap_mcbsp_get_max_tx_threshold(unsigned int id)
+{
+ struct omap_mcbsp *mcbsp;
+
+ if (!omap_mcbsp_check_valid_id(id)) {
+ printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1);
+ return -ENODEV;
+ }
+ mcbsp = id_to_mcbsp_ptr(id);
+
+ return mcbsp->max_tx_thres;
+}
+EXPORT_SYMBOL(omap_mcbsp_get_max_tx_threshold);
+
+/*
+ * omap_mcbsp_get_max_rx_thres just return the current configured
+ * maximum threshold for reception
+ */
+u16 omap_mcbsp_get_max_rx_threshold(unsigned int id)
+{
+ struct omap_mcbsp *mcbsp;
+
+ if (!omap_mcbsp_check_valid_id(id)) {
+ printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1);
+ return -ENODEV;
+ }
+ mcbsp = id_to_mcbsp_ptr(id);
+
+ return mcbsp->max_rx_thres;
+}
+EXPORT_SYMBOL(omap_mcbsp_get_max_rx_threshold);
+
+/*
+ * omap_mcbsp_get_dma_op_mode just return the current configured
+ * operating mode for the mcbsp channel
+ */
+int omap_mcbsp_get_dma_op_mode(unsigned int id)
+{
+ struct omap_mcbsp *mcbsp;
+ int dma_op_mode;
+
+ if (!omap_mcbsp_check_valid_id(id)) {
+ printk(KERN_ERR "%s: Invalid id (%u)\n", __func__, id + 1);
+ return -ENODEV;
+ }
+ mcbsp = id_to_mcbsp_ptr(id);
+
+ spin_lock_irq(&mcbsp->lock);
+ dma_op_mode = mcbsp->dma_op_mode;
+ spin_unlock_irq(&mcbsp->lock);
+
+ return dma_op_mode;
+}
+EXPORT_SYMBOL(omap_mcbsp_get_dma_op_mode);
+
+static inline void omap34xx_mcbsp_request(struct omap_mcbsp *mcbsp)
+{
+ /*
+ * Enable wakup behavior, smart idle and all wakeups
+ * REVISIT: some wakeups may be unnecessary
+ */
+ if (cpu_is_omap34xx()) {
+ u16 syscon;
+
+ syscon = OMAP_MCBSP_READ(mcbsp->io_base, SYSCON);
+ syscon &= ~(ENAWAKEUP | SIDLEMODE(0x03) | CLOCKACTIVITY(0x03));
+
+ spin_lock_irq(&mcbsp->lock);
+ if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) {
+ syscon |= (ENAWAKEUP | SIDLEMODE(0x02) |
+ CLOCKACTIVITY(0x02));
+ OMAP_MCBSP_WRITE(mcbsp->io_base, WAKEUPEN,
+ XRDYEN | RRDYEN);
+ } else {
+ syscon |= SIDLEMODE(0x01);
+ }
+ spin_unlock_irq(&mcbsp->lock);
+
+ OMAP_MCBSP_WRITE(mcbsp->io_base, SYSCON, syscon);
+ }
+}
+
+static inline void omap34xx_mcbsp_free(struct omap_mcbsp *mcbsp)
+{
+ /*
+ * Disable wakup behavior, smart idle and all wakeups
+ */
+ if (cpu_is_omap34xx()) {
+ u16 syscon;
+
+ syscon = OMAP_MCBSP_READ(mcbsp->io_base, SYSCON);
+ syscon &= ~(ENAWAKEUP | SIDLEMODE(0x03) | CLOCKACTIVITY(0x03));
+ /*
+ * HW bug workaround - If no_idle mode is taken, we need to
+ * go to smart_idle before going to always_idle, or the
+ * device will not hit retention anymore.
+ */
+ syscon |= SIDLEMODE(0x02);
+ OMAP_MCBSP_WRITE(mcbsp->io_base, SYSCON, syscon);
+
+ syscon &= ~(SIDLEMODE(0x03));
+ OMAP_MCBSP_WRITE(mcbsp->io_base, SYSCON, syscon);
+
+ OMAP_MCBSP_WRITE(mcbsp->io_base, WAKEUPEN, 0);
+ }
+}
+#else
+static inline void omap34xx_mcbsp_request(struct omap_mcbsp *mcbsp) {}
+static inline void omap34xx_mcbsp_free(struct omap_mcbsp *mcbsp) {}
+#endif
+
/*
* We can choose between IRQ based or polled IO.
* This needs to be called before omap_mcbsp_request().
@@ -257,6 +421,9 @@
clk_enable(mcbsp->iclk);
clk_enable(mcbsp->fclk);
+ /* Do procedure specific to omap34xx arch, if applicable */
+ omap34xx_mcbsp_request(mcbsp);
+
/*
* Make sure that transmitter, receiver and sample-rate generator are
* not running before activating IRQs.
@@ -305,6 +472,9 @@
if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free)
mcbsp->pdata->ops->free(id);
+ /* Do procedure specific to omap34xx arch, if applicable */
+ omap34xx_mcbsp_free(mcbsp);
+
clk_disable(mcbsp->fclk);
clk_disable(mcbsp->iclk);
@@ -365,7 +535,13 @@
w = OMAP_MCBSP_READ(io_base, SPCR1);
OMAP_MCBSP_WRITE(io_base, SPCR1, w | (rx & 1));
- udelay(100);
+ /*
+ * Worst case: CLKSRG*2 = 8000khz: (1/8000) * 2 * 2 usec
+ * REVISIT: 100us may give enough time for two CLKSRG, however
+ * due to some unknown PM related, clock gating etc. reason it
+ * is now at 500us.
+ */
+ udelay(500);
if (idle) {
/* Start frame sync */
@@ -412,6 +588,58 @@
}
EXPORT_SYMBOL(omap_mcbsp_stop);
+void omap_mcbsp_xmit_enable(unsigned int id, u8 enable)
+{
+ struct omap_mcbsp *mcbsp;
+ void __iomem *io_base;
+ u16 w;
+
+ if (!(cpu_is_omap2430() || cpu_is_omap34xx()))
+ return;
+
+ if (!omap_mcbsp_check_valid_id(id)) {
+ printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1);
+ return;
+ }
+
+ mcbsp = id_to_mcbsp_ptr(id);
+ io_base = mcbsp->io_base;
+
+ w = OMAP_MCBSP_READ(io_base, XCCR);
+
+ if (enable)
+ OMAP_MCBSP_WRITE(io_base, XCCR, w & ~(XDISABLE));
+ else
+ OMAP_MCBSP_WRITE(io_base, XCCR, w | XDISABLE);
+}
+EXPORT_SYMBOL(omap_mcbsp_xmit_enable);
+
+void omap_mcbsp_recv_enable(unsigned int id, u8 enable)
+{
+ struct omap_mcbsp *mcbsp;
+ void __iomem *io_base;
+ u16 w;
+
+ if (!(cpu_is_omap2430() || cpu_is_omap34xx()))
+ return;
+
+ if (!omap_mcbsp_check_valid_id(id)) {
+ printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1);
+ return;
+ }
+
+ mcbsp = id_to_mcbsp_ptr(id);
+ io_base = mcbsp->io_base;
+
+ w = OMAP_MCBSP_READ(io_base, RCCR);
+
+ if (enable)
+ OMAP_MCBSP_WRITE(io_base, RCCR, w & ~(RDISABLE));
+ else
+ OMAP_MCBSP_WRITE(io_base, RCCR, w | RDISABLE);
+}
+EXPORT_SYMBOL(omap_mcbsp_recv_enable);
+
/* polled mcbsp i/o operations */
int omap_mcbsp_pollwrite(unsigned int id, u16 buf)
{
@@ -897,6 +1125,147 @@
}
EXPORT_SYMBOL(omap_mcbsp_set_spi_mode);
+#ifdef CONFIG_ARCH_OMAP34XX
+#define max_thres(m) (mcbsp->pdata->buffer_size)
+#define valid_threshold(m, val) ((val) <= max_thres(m))
+#define THRESHOLD_PROP_BUILDER(prop) \
+static ssize_t prop##_show(struct device *dev, \
+ struct device_attribute *attr, char *buf) \
+{ \
+ struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \
+ \
+ return sprintf(buf, "%u\n", mcbsp->prop); \
+} \
+ \
+static ssize_t prop##_store(struct device *dev, \
+ struct device_attribute *attr, \
+ const char *buf, size_t size) \
+{ \
+ struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \
+ unsigned long val; \
+ int status; \
+ \
+ status = strict_strtoul(buf, 0, &val); \
+ if (status) \
+ return status; \
+ \
+ if (!valid_threshold(mcbsp, val)) \
+ return -EDOM; \
+ \
+ mcbsp->prop = val; \
+ return size; \
+} \
+ \
+static DEVICE_ATTR(prop, 0644, prop##_show, prop##_store);
+
+THRESHOLD_PROP_BUILDER(max_tx_thres);
+THRESHOLD_PROP_BUILDER(max_rx_thres);
+
+static ssize_t dma_op_mode_show(struct device *dev,
+ struct device_attribute *attr, char *buf)
+{
+ struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);
+ int dma_op_mode;
+
+ spin_lock_irq(&mcbsp->lock);
+ dma_op_mode = mcbsp->dma_op_mode;
+ spin_unlock_irq(&mcbsp->lock);
+
+ return sprintf(buf, "current mode: %d\n"
+ "possible mode values are:\n"
+ "%d - %s\n"
+ "%d - %s\n"
+ "%d - %s\n",
+ dma_op_mode,
+ MCBSP_DMA_MODE_ELEMENT, "element mode",
+ MCBSP_DMA_MODE_THRESHOLD, "threshold mode",
+ MCBSP_DMA_MODE_FRAME, "frame mode");
+}
+
+static ssize_t dma_op_mode_store(struct device *dev,
+ struct device_attribute *attr,
+ const char *buf, size_t size)
+{
+ struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);
+ unsigned long val;
+ int status;
+
+ status = strict_strtoul(buf, 0, &val);
+ if (status)
+ return status;
+
+ spin_lock_irq(&mcbsp->lock);
+
+ if (!mcbsp->free) {
+ size = -EBUSY;
+ goto unlock;
+ }
+
+ if (val > MCBSP_DMA_MODE_FRAME || val < MCBSP_DMA_MODE_ELEMENT) {
+ size = -EINVAL;
+ goto unlock;
+ }
+
+ mcbsp->dma_op_mode = val;
+
+unlock:
+ spin_unlock_irq(&mcbsp->lock);
+
+ return size;
+}
+
+static DEVICE_ATTR(dma_op_mode, 0644, dma_op_mode_show, dma_op_mode_store);
+
+static const struct attribute *additional_attrs[] = {
+ &dev_attr_max_tx_thres.attr,
+ &dev_attr_max_rx_thres.attr,
+ &dev_attr_dma_op_mode.attr,
+ NULL,
+};
+
+static const struct attribute_group additional_attr_group = {
+ .attrs = (struct attribute **)additional_attrs,
+};
+
+static inline int __devinit omap_additional_add(struct device *dev)
+{
+ return sysfs_create_group(&dev->kobj, &additional_attr_group);
+}
+
+static inline void __devexit omap_additional_remove(struct device *dev)
+{
+ sysfs_remove_group(&dev->kobj, &additional_attr_group);
+}
+
+static inline void __devinit omap34xx_device_init(struct omap_mcbsp *mcbsp)
+{
+ mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT;
+ if (cpu_is_omap34xx()) {
+ mcbsp->max_tx_thres = max_thres(mcbsp);
+ mcbsp->max_rx_thres = max_thres(mcbsp);
+ /*
+ * REVISIT: Set dmap_op_mode to THRESHOLD as default
+ * for mcbsp2 instances.
+ */
+ if (omap_additional_add(mcbsp->dev))
+ dev_warn(mcbsp->dev,
+ "Unable to create additional controls\n");
+ } else {
+ mcbsp->max_tx_thres = -EINVAL;
+ mcbsp->max_rx_thres = -EINVAL;
+ }
+}
+
+static inline void __devexit omap34xx_device_exit(struct omap_mcbsp *mcbsp)
+{
+ if (cpu_is_omap34xx())
+ omap_additional_remove(mcbsp->dev);
+}
+#else
+static inline void __devinit omap34xx_device_init(struct omap_mcbsp *mcbsp) {}
+static inline void __devexit omap34xx_device_exit(struct omap_mcbsp *mcbsp) {}
+#endif /* CONFIG_ARCH_OMAP34XX */
+
/*
* McBSP1 and McBSP3 are directly mapped on 1610 and 1510.
* 730 has only 2 McBSP, and both of them are MPU peripherals.
@@ -967,6 +1336,10 @@
mcbsp->dev = &pdev->dev;
mcbsp_ptr[id] = mcbsp;
platform_set_drvdata(pdev, mcbsp);
+
+ /* Initialize mcbsp properties for OMAP34XX if needed / applicable */
+ omap34xx_device_init(mcbsp);
+
return 0;
err_fclk:
@@ -990,6 +1363,8 @@
mcbsp->pdata->ops->free)
mcbsp->pdata->ops->free(mcbsp->id);
+ omap34xx_device_exit(mcbsp);
+
clk_disable(mcbsp->fclk);
clk_disable(mcbsp->iclk);
clk_put(mcbsp->fclk);
diff --git a/arch/arm/plat-s3c/include/plat/audio-simtec.h b/arch/arm/plat-s3c/include/plat/audio-simtec.h
new file mode 100644
index 0000000..0f440b9
--- /dev/null
+++ b/arch/arm/plat-s3c/include/plat/audio-simtec.h
@@ -0,0 +1,37 @@
+/* arch/arm/plat-s3c/include/plat/audio-simtec.h
+ *
+ * Copyright 2008 Simtec Electronics
+ * http://armlinux.simtec.co.uk/
+ * Ben Dooks <ben@simtec.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Simtec Audio support.
+*/
+
+/**
+ * struct s3c24xx_audio_simtec_pdata - platform data for simtec audio
+ * @use_mpllin: Select codec clock from MPLLin
+ * @output_cdclk: Need to output CDCLK to the codec
+ * @have_mic: Set if we have a MIC socket
+ * @have_lout: Set if we have a LineOut socket
+ * @amp_gpio: GPIO pin to enable the AMP
+ * @amp_gain: Option GPIO to control AMP gain
+ */
+struct s3c24xx_audio_simtec_pdata {
+ unsigned int use_mpllin:1;
+ unsigned int output_cdclk:1;
+
+ unsigned int have_mic:1;
+ unsigned int have_lout:1;
+
+ int amp_gpio;
+ int amp_gain[2];
+
+ void (*startup)(void);
+};
+
+extern int simtec_audio_add(const char *codec_name,
+ struct s3c24xx_audio_simtec_pdata *pdata);
diff --git a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
index 0fad757..07659da 100644
--- a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
+++ b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
@@ -33,6 +33,11 @@
#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1)
#define S3C2412_IISCON_IIS_ACTIVE (1 << 0)
+#define S3C64XX_IISMOD_BLC_16BIT (0 << 13)
+#define S3C64XX_IISMOD_BLC_8BIT (1 << 13)
+#define S3C64XX_IISMOD_BLC_24BIT (2 << 13)
+#define S3C64XX_IISMOD_BLC_MASK (3 << 13)
+
#define S3C64XX_IISMOD_IMS_PCLK (0 << 10)
#define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10)
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
new file mode 100644
index 0000000..c022736
--- /dev/null
+++ b/include/sound/sh_fsi.h
@@ -0,0 +1,83 @@
+#ifndef __SOUND_FSI_H
+#define __SOUND_FSI_H
+
+/*
+ * Fifo-attached Serial Interface (FSI) support for SH7724
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+/* flags format
+
+ * 0xABCDEEFF
+ *
+ * A: channel size for TDM (input)
+ * B: channel size for TDM (ooutput)
+ * C: inversion
+ * D: mode
+ * E: input format
+ * F: output format
+ */
+
+#include <linux/clk.h>
+#include <sound/soc.h>
+
+/* TDM channel */
+#define SH_FSI_SET_CH_I(x) ((x & 0xF) << 28)
+#define SH_FSI_SET_CH_O(x) ((x & 0xF) << 24)
+
+#define SH_FSI_CH_IMASK 0xF0000000
+#define SH_FSI_CH_OMASK 0x0F000000
+#define SH_FSI_GET_CH_I(x) ((x & SH_FSI_CH_IMASK) >> 28)
+#define SH_FSI_GET_CH_O(x) ((x & SH_FSI_CH_OMASK) >> 24)
+
+/* clock inversion */
+#define SH_FSI_INVERSION_MASK 0x00F00000
+#define SH_FSI_LRM_INV (1 << 20)
+#define SH_FSI_BRM_INV (1 << 21)
+#define SH_FSI_LRS_INV (1 << 22)
+#define SH_FSI_BRS_INV (1 << 23)
+
+/* mode */
+#define SH_FSI_MODE_MASK 0x000F0000
+#define SH_FSI_IN_SLAVE_MODE (1 << 16) /* default master mode */
+#define SH_FSI_OUT_SLAVE_MODE (1 << 17) /* default master mode */
+
+/* DI format */
+#define SH_FSI_FMT_MASK 0x000000FF
+#define SH_FSI_IFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 8)
+#define SH_FSI_OFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 0)
+#define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK)
+#define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK)
+
+#define SH_FSI_FMT_MONO (1 << 0)
+#define SH_FSI_FMT_MONO_DELAY (1 << 1)
+#define SH_FSI_FMT_PCM (1 << 2)
+#define SH_FSI_FMT_I2S (1 << 3)
+#define SH_FSI_FMT_TDM (1 << 4)
+#define SH_FSI_FMT_TDM_DELAY (1 << 5)
+
+#define SH_FSI_IFMT_TDM_CH(x) \
+ (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x))
+#define SH_FSI_IFMT_TDM_DELAY_CH(x) \
+ (SH_FSI_IFMT(TDM_DELAY) | SH_FSI_SET_CH_I(x))
+
+#define SH_FSI_OFMT_TDM_CH(x) \
+ (SH_FSI_OFMT(TDM) | SH_FSI_SET_CH_O(x))
+#define SH_FSI_OFMT_TDM_DELAY_CH(x) \
+ (SH_FSI_OFMT(TDM_DELAY) | SH_FSI_SET_CH_O(x))
+
+struct sh_fsi_platform_info {
+ unsigned long porta_flags;
+ unsigned long portb_flags;
+};
+
+extern struct snd_soc_dai fsi_soc_dai[2];
+extern struct snd_soc_platform fsi_soc_platform;
+
+#endif /* __SOUND_FSI_H */
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 3388405..c1410e3 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -289,6 +289,7 @@
/* dapm sys fs - used by the core */
int snd_soc_dapm_sys_add(struct device *dev);
+void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec);
/* dapm audio pin control and status */
int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin);
diff --git a/include/sound/soc.h b/include/sound/soc.h
index dbb1702..0758a1b 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -416,6 +416,7 @@
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_reg;
struct dentry *debugfs_pop_time;
+ struct dentry *debugfs_dapm;
#endif
};
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 6c00ea4..757b480 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -207,8 +207,8 @@
snprintf(card->longname, sizeof(card->longname),
"%s (%s)", dev->dev.driver->name, card->mixername);
- if (pdata && pdata->codec_data)
- snd_ac97_dev_add_pdata(ac97_bus->codec[0], pdata->codec_pdata);
+ if (pdata && pdata->codec_data[0])
+ snd_ac97_dev_add_pdata(ac97_bus->codec[0], pdata->codec_pdata[0]);
snd_card_set_dev(card, &dev->dev);
ret = snd_card_register(card);
if (ret == 0) {
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 6205f37..743ac6a 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -136,6 +136,9 @@
{
struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+ if (!prtd || !prtd->params)
+ return 0;
+
DCSR(prtd->dma_ch) &= ~DCSR_RUN;
DCSR(prtd->dma_ch) = 0;
DCMD(prtd->dma_ch) = 0;
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 167a5ce..0edca93 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -18,6 +18,7 @@
select SND_SOC_AD73311 if I2C
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
+ select SND_SOC_AK4642 if I2C
select SND_SOC_CS4270 if I2C
select SND_SOC_MAX9877 if I2C
select SND_SOC_PCM3008
@@ -92,6 +93,9 @@
config SND_SOC_AK4535
tristate
+config SND_SOC_AK4642
+ tristate
+
# Cirrus Logic CS4270 Codec
config SND_SOC_CS4270
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index fbab43b..fb4af28 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -5,6 +5,7 @@
snd-soc-ad73311-objs := ad73311.o
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
+snd-soc-ak4642-objs := ak4642.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-cx20442-objs := cx20442.o
snd-soc-l3-objs := l3.o
@@ -54,6 +55,7 @@
obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
+obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
new file mode 100644
index 0000000..e057c7b
--- /dev/null
+++ b/sound/soc/codecs/ak4642.c
@@ -0,0 +1,502 @@
+/*
+ * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * Based on wm8731.c by Richard Purdie
+ * Based on ak4535.c by Richard Purdie
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+/* ** CAUTION **
+ *
+ * This is very simple driver.
+ * It can use headphone output / stereo input only
+ *
+ * AK4642 is not tested.
+ * AK4643 is tested.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "ak4642.h"
+
+#define AK4642_VERSION "0.0.1"
+
+#define PW_MGMT1 0x00
+#define PW_MGMT2 0x01
+#define SG_SL1 0x02
+#define SG_SL2 0x03
+#define MD_CTL1 0x04
+#define MD_CTL2 0x05
+#define TIMER 0x06
+#define ALC_CTL1 0x07
+#define ALC_CTL2 0x08
+#define L_IVC 0x09
+#define L_DVC 0x0a
+#define ALC_CTL3 0x0b
+#define R_IVC 0x0c
+#define R_DVC 0x0d
+#define MD_CTL3 0x0e
+#define MD_CTL4 0x0f
+#define PW_MGMT3 0x10
+#define DF_S 0x11
+#define FIL3_0 0x12
+#define FIL3_1 0x13
+#define FIL3_2 0x14
+#define FIL3_3 0x15
+#define EQ_0 0x16
+#define EQ_1 0x17
+#define EQ_2 0x18
+#define EQ_3 0x19
+#define EQ_4 0x1a
+#define EQ_5 0x1b
+#define FIL1_0 0x1c
+#define FIL1_1 0x1d
+#define FIL1_2 0x1e
+#define FIL1_3 0x1f
+#define PW_MGMT4 0x20
+#define MD_CTL5 0x21
+#define LO_MS 0x22
+#define HP_MS 0x23
+#define SPK_MS 0x24
+
+#define AK4642_CACHEREGNUM 0x25
+
+struct snd_soc_codec_device soc_codec_dev_ak4642;
+
+/* codec private data */
+struct ak4642_priv {
+ struct snd_soc_codec codec;
+ unsigned int sysclk;
+};
+
+static struct snd_soc_codec *ak4642_codec;
+
+/*
+ * ak4642 register cache
+ */
+static const u16 ak4642_reg[AK4642_CACHEREGNUM] = {
+ 0x0000, 0x0000, 0x0001, 0x0000,
+ 0x0002, 0x0000, 0x0000, 0x0000,
+ 0x00e1, 0x00e1, 0x0018, 0x0000,
+ 0x00e1, 0x0018, 0x0011, 0x0008,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000,
+};
+
+/*
+ * read ak4642 register cache
+ */
+static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= AK4642_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write ak4642 register cache
+ */
+static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= AK4642_CACHEREGNUM)
+ return;
+
+ cache[reg] = value;
+}
+
+/*
+ * write to the AK4642 register space
+ */
+static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D8 AK4642 register offset
+ * D7...D0 register data
+ */
+ data[0] = reg & 0xff;
+ data[1] = value & 0xff;
+
+ if (codec->hw_write(codec->control_data, data, 2) == 2) {
+ ak4642_write_reg_cache(codec, reg, value);
+ return 0;
+ } else
+ return -EIO;
+}
+
+static int ak4642_sync(struct snd_soc_codec *codec)
+{
+ u16 *cache = codec->reg_cache;
+ int i, r = 0;
+
+ for (i = 0; i < AK4642_CACHEREGNUM; i++)
+ r |= ak4642_write(codec, i, cache[i]);
+
+ return r;
+};
+
+static int ak4642_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ struct snd_soc_codec *codec = dai->codec;
+
+ if (is_play) {
+ /*
+ * start headphone output
+ *
+ * PLL, Master Mode
+ * Audio I/F Format :MSB justified (ADC & DAC)
+ * Sampling Frequency: 44.1kHz
+ * Digital Volume: −8dB
+ * Bass Boost Level : Middle
+ *
+ * This operation came from example code of
+ * "ASAHI KASEI AK4642" (japanese) manual p97.
+ *
+ * Example code use 0x39, 0x79 value for 0x01 address,
+ * But we need MCKO (0x02) bit now
+ */
+ ak4642_write(codec, 0x05, 0x27);
+ ak4642_write(codec, 0x0f, 0x09);
+ ak4642_write(codec, 0x0e, 0x19);
+ ak4642_write(codec, 0x09, 0x91);
+ ak4642_write(codec, 0x0c, 0x91);
+ ak4642_write(codec, 0x0a, 0x28);
+ ak4642_write(codec, 0x0d, 0x28);
+ ak4642_write(codec, 0x00, 0x64);
+ ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */
+ ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */
+ } else {
+ /*
+ * start stereo input
+ *
+ * PLL Master Mode
+ * Audio I/F Format:MSB justified (ADC & DAC)
+ * Sampling Frequency:44.1kHz
+ * Pre MIC AMP:+20dB
+ * MIC Power On
+ * ALC setting:Refer to Table 35
+ * ALC bit=“1”
+ *
+ * This operation came from example code of
+ * "ASAHI KASEI AK4642" (japanese) manual p94.
+ */
+ ak4642_write(codec, 0x05, 0x27);
+ ak4642_write(codec, 0x02, 0x05);
+ ak4642_write(codec, 0x06, 0x3c);
+ ak4642_write(codec, 0x08, 0xe1);
+ ak4642_write(codec, 0x0b, 0x00);
+ ak4642_write(codec, 0x07, 0x21);
+ ak4642_write(codec, 0x00, 0x41);
+ ak4642_write(codec, 0x10, 0x01);
+ }
+
+ return 0;
+}
+
+static void ak4642_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ struct snd_soc_codec *codec = dai->codec;
+
+ if (is_play) {
+ /* stop headphone output */
+ ak4642_write(codec, 0x01, 0x3b);
+ ak4642_write(codec, 0x01, 0x0b);
+ ak4642_write(codec, 0x00, 0x40);
+ ak4642_write(codec, 0x0e, 0x11);
+ ak4642_write(codec, 0x0f, 0x08);
+ } else {
+ /* stop stereo input */
+ ak4642_write(codec, 0x00, 0x40);
+ ak4642_write(codec, 0x10, 0x00);
+ ak4642_write(codec, 0x07, 0x01);
+ }
+}
+
+static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct ak4642_priv *ak4642 = codec->private_data;
+
+ ak4642->sysclk = freq;
+ return 0;
+}
+
+static struct snd_soc_dai_ops ak4642_dai_ops = {
+ .startup = ak4642_dai_startup,
+ .shutdown = ak4642_dai_shutdown,
+ .set_sysclk = ak4642_dai_set_sysclk,
+};
+
+struct snd_soc_dai ak4642_dai = {
+ .name = "AK4642",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE },
+ .ops = &ak4642_dai_ops,
+};
+EXPORT_SYMBOL_GPL(ak4642_dai);
+
+static int ak4642_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ ak4642_sync(codec);
+ return 0;
+}
+
+/*
+ * initialise the AK4642 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int ak4642_init(struct ak4642_priv *ak4642)
+{
+ struct snd_soc_codec *codec = &ak4642->codec;
+ int ret = 0;
+
+ if (ak4642_codec) {
+ dev_err(codec->dev, "Another ak4642 is registered\n");
+ return -EINVAL;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = ak4642;
+ codec->name = "AK4642";
+ codec->owner = THIS_MODULE;
+ codec->read = ak4642_read_reg_cache;
+ codec->write = ak4642_write;
+ codec->dai = &ak4642_dai;
+ codec->num_dai = 1;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ codec->reg_cache_size = ARRAY_SIZE(ak4642_reg);
+ codec->reg_cache = kmemdup(ak4642_reg,
+ sizeof(ak4642_reg), GFP_KERNEL);
+
+ if (!codec->reg_cache)
+ return -ENOMEM;
+
+ ak4642_dai.dev = codec->dev;
+ ak4642_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ goto reg_cache_err;
+ }
+
+ ret = snd_soc_register_dai(&ak4642_dai);
+ if (ret) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ snd_soc_unregister_codec(codec);
+ goto reg_cache_err;
+ }
+
+ /*
+ * clock setting
+ *
+ * Audio I/F Format: MSB justified (ADC & DAC)
+ * BICK frequency at Master Mode: 64fs
+ * Input Master Clock Select at PLL Mode: 11.2896MHz
+ * MCKO: Enable
+ * Sampling Frequency: 44.1kHz
+ *
+ * This operation came from example code of
+ * "ASAHI KASEI AK4642" (japanese) manual p89.
+ *
+ * please fix-me
+ */
+ ak4642_write(codec, 0x01, 0x08);
+ ak4642_write(codec, 0x04, 0x4a);
+ ak4642_write(codec, 0x05, 0x27);
+ ak4642_write(codec, 0x00, 0x40);
+ ak4642_write(codec, 0x01, 0x0b);
+
+ return ret;
+
+reg_cache_err:
+ kfree(codec->reg_cache);
+ codec->reg_cache = NULL;
+
+ return ret;
+}
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static int ak4642_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct ak4642_priv *ak4642;
+ struct snd_soc_codec *codec;
+ int ret;
+
+ ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL);
+ if (!ak4642)
+ return -ENOMEM;
+
+ codec = &ak4642->codec;
+ codec->dev = &i2c->dev;
+
+ i2c_set_clientdata(i2c, ak4642);
+ codec->control_data = i2c;
+
+ ret = ak4642_init(ak4642);
+ if (ret < 0)
+ printk(KERN_ERR "failed to initialise AK4642\n");
+
+ return ret;
+}
+
+static int ak4642_i2c_remove(struct i2c_client *client)
+{
+ struct ak4642_priv *ak4642 = i2c_get_clientdata(client);
+
+ snd_soc_unregister_dai(&ak4642_dai);
+ snd_soc_unregister_codec(&ak4642->codec);
+ kfree(ak4642->codec.reg_cache);
+ kfree(ak4642);
+ ak4642_codec = NULL;
+
+ return 0;
+}
+
+static const struct i2c_device_id ak4642_i2c_id[] = {
+ { "ak4642", 0 },
+ { "ak4643", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);
+
+static struct i2c_driver ak4642_i2c_driver = {
+ .driver = {
+ .name = "AK4642 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = ak4642_i2c_probe,
+ .remove = ak4642_i2c_remove,
+ .id_table = ak4642_i2c_id,
+};
+
+#endif
+
+static int ak4642_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ int ret;
+
+ if (!ak4642_codec) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = ak4642_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "ak4642: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "ak4642: failed to register card\n");
+ goto card_err;
+ }
+
+ dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION);
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+
+}
+
+/* power down chip */
+static int ak4642_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ak4642 = {
+ .probe = ak4642_probe,
+ .remove = ak4642_remove,
+ .resume = ak4642_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642);
+
+static int __init ak4642_modinit(void)
+{
+ int ret;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ ret = i2c_add_driver(&ak4642_i2c_driver);
+#endif
+ return ret;
+
+}
+module_init(ak4642_modinit);
+
+static void __exit ak4642_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&ak4642_i2c_driver);
+#endif
+
+}
+module_exit(ak4642_exit);
+
+MODULE_DESCRIPTION("Soc AK4642 driver");
+MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ak4642.h b/sound/soc/codecs/ak4642.h
new file mode 100644
index 0000000..e476833
--- /dev/null
+++ b/sound/soc/codecs/ak4642.h
@@ -0,0 +1,20 @@
+/*
+ * ak4642.h -- AK4642 Soc Audio driver
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * Based on ak4535.c
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _AK4642_H
+#define _AK4642_H
+
+extern struct snd_soc_dai ak4642_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ak4642;
+
+#endif
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 126b15b..5d54767 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -53,6 +53,7 @@
/* codec private data */
struct aic3x_priv {
+ struct snd_soc_codec codec;
unsigned int sysclk;
int master;
};
@@ -1156,11 +1157,13 @@
* initialise the AIC3X driver
* register the mixer and dsp interfaces with the kernel
*/
-static int aic3x_init(struct snd_soc_device *socdev)
+static int aic3x_init(struct snd_soc_codec *codec)
{
- struct snd_soc_codec *codec = socdev->card->codec;
- struct aic3x_setup_data *setup = socdev->codec_data;
- int reg, ret = 0;
+ int reg;
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
codec->name = "tlv320aic3x";
codec->owner = THIS_MODULE;
@@ -1177,13 +1180,6 @@
aic3x_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT);
aic3x_write(codec, AIC3X_RESET, SOFT_RESET);
- /* register pcms */
- ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
- if (ret < 0) {
- printk(KERN_ERR "aic3x: failed to create pcms\n");
- goto pcm_err;
- }
-
/* DAC default volume and mute */
aic3x_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON);
aic3x_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON);
@@ -1250,30 +1246,51 @@
/* off, with power on */
aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- /* setup GPIO functions */
- aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4);
- aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4);
-
- snd_soc_add_controls(codec, aic3x_snd_controls,
- ARRAY_SIZE(aic3x_snd_controls));
- aic3x_add_widgets(codec);
- ret = snd_soc_init_card(socdev);
- if (ret < 0) {
- printk(KERN_ERR "aic3x: failed to register card\n");
- goto card_err;
- }
-
- return ret;
-
-card_err:
- snd_soc_free_pcms(socdev);
- snd_soc_dapm_free(socdev);
-pcm_err:
- kfree(codec->reg_cache);
- return ret;
+ return 0;
}
-static struct snd_soc_device *aic3x_socdev;
+static struct snd_soc_codec *aic3x_codec;
+
+static int aic3x_register(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ ret = aic3x_init(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to initialise device\n");
+ return ret;
+ }
+
+ aic3x_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret) {
+ dev_err(codec->dev, "Failed to register codec\n");
+ return ret;
+ }
+
+ ret = snd_soc_register_dai(&aic3x_dai);
+ if (ret) {
+ dev_err(codec->dev, "Failed to register dai\n");
+ snd_soc_unregister_codec(codec);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int aic3x_unregister(struct aic3x_priv *aic3x)
+{
+ aic3x_set_bias_level(&aic3x->codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_unregister_dai(&aic3x_dai);
+ snd_soc_unregister_codec(&aic3x->codec);
+
+ kfree(aic3x);
+ aic3x_codec = NULL;
+
+ return 0;
+}
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
/*
@@ -1288,28 +1305,36 @@
static int aic3x_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
- struct snd_soc_device *socdev = aic3x_socdev;
- struct snd_soc_codec *codec = socdev->card->codec;
- int ret;
+ struct snd_soc_codec *codec;
+ struct aic3x_priv *aic3x;
- i2c_set_clientdata(i2c, codec);
+ aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL);
+ if (aic3x == NULL) {
+ dev_err(&i2c->dev, "failed to create private data\n");
+ return -ENOMEM;
+ }
+
+ codec = &aic3x->codec;
+ codec->dev = &i2c->dev;
+ codec->private_data = aic3x;
codec->control_data = i2c;
+ codec->hw_write = (hw_write_t) i2c_master_send;
- ret = aic3x_init(socdev);
- if (ret < 0)
- printk(KERN_ERR "aic3x: failed to initialise AIC3X\n");
- return ret;
+ i2c_set_clientdata(i2c, aic3x);
+
+ return aic3x_register(codec);
}
static int aic3x_i2c_remove(struct i2c_client *client)
{
- struct snd_soc_codec *codec = i2c_get_clientdata(client);
- kfree(codec->reg_cache);
- return 0;
+ struct aic3x_priv *aic3x = i2c_get_clientdata(client);
+
+ return aic3x_unregister(aic3x);
}
static const struct i2c_device_id aic3x_i2c_id[] = {
{ "tlv320aic3x", 0 },
+ { "tlv320aic33", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
@@ -1320,50 +1345,28 @@
.name = "aic3x I2C Codec",
.owner = THIS_MODULE,
},
- .probe = aic3x_i2c_probe,
+ .probe = aic3x_i2c_probe,
.remove = aic3x_i2c_remove,
.id_table = aic3x_i2c_id,
};
-static int aic3x_add_i2c_device(struct platform_device *pdev,
- const struct aic3x_setup_data *setup)
+static inline void aic3x_i2c_init(void)
{
- struct i2c_board_info info;
- struct i2c_adapter *adapter;
- struct i2c_client *client;
int ret;
ret = i2c_add_driver(&aic3x_i2c_driver);
- if (ret != 0) {
- dev_err(&pdev->dev, "can't add i2c driver\n");
- return ret;
- }
-
- memset(&info, 0, sizeof(struct i2c_board_info));
- info.addr = setup->i2c_address;
- strlcpy(info.type, "tlv320aic3x", I2C_NAME_SIZE);
-
- adapter = i2c_get_adapter(setup->i2c_bus);
- if (!adapter) {
- dev_err(&pdev->dev, "can't get i2c adapter %d\n",
- setup->i2c_bus);
- goto err_driver;
- }
-
- client = i2c_new_device(adapter, &info);
- i2c_put_adapter(adapter);
- if (!client) {
- dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
- (unsigned int)info.addr);
- goto err_driver;
- }
-
- return 0;
-
-err_driver:
- i2c_del_driver(&aic3x_i2c_driver);
- return -ENODEV;
+ if (ret)
+ printk(KERN_ERR "%s: error regsitering i2c driver, %d\n",
+ __func__, ret);
}
+
+static inline void aic3x_i2c_exit(void)
+{
+ i2c_del_driver(&aic3x_i2c_driver);
+}
+#else
+static inline void aic3x_i2c_init(void) { }
+static inline void aic3x_i2c_exit(void) { }
#endif
static int aic3x_probe(struct platform_device *pdev)
@@ -1371,42 +1374,52 @@
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct aic3x_setup_data *setup;
struct snd_soc_codec *codec;
- struct aic3x_priv *aic3x;
int ret = 0;
- printk(KERN_INFO "AIC3X Audio Codec %s\n", AIC3X_VERSION);
-
- setup = socdev->codec_data;
- codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
- if (codec == NULL)
- return -ENOMEM;
-
- aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL);
- if (aic3x == NULL) {
- kfree(codec);
- return -ENOMEM;
+ codec = aic3x_codec;
+ if (!codec) {
+ dev_err(&pdev->dev, "Codec not registered\n");
+ return -ENODEV;
}
- codec->private_data = aic3x;
socdev->card->codec = codec;
- mutex_init(&codec->mutex);
- INIT_LIST_HEAD(&codec->dapm_widgets);
- INIT_LIST_HEAD(&codec->dapm_paths);
+ setup = socdev->codec_data;
- aic3x_socdev = socdev;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- if (setup->i2c_address) {
- codec->hw_write = (hw_write_t) i2c_master_send;
- ret = aic3x_add_i2c_device(pdev, setup);
+ if (!setup) {
+ dev_err(&pdev->dev, "No setup data supplied\n");
+ return -EINVAL;
}
-#else
- /* Add other interfaces here */
-#endif
- if (ret != 0) {
- kfree(codec->private_data);
- kfree(codec);
+ /* setup GPIO functions */
+ aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4);
+ aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "aic3x: failed to create pcms\n");
+ goto pcm_err;
}
+
+ snd_soc_add_controls(codec, aic3x_snd_controls,
+ ARRAY_SIZE(aic3x_snd_controls));
+
+ aic3x_add_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "aic3x: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+pcm_err:
+ kfree(codec->reg_cache);
return ret;
}
@@ -1421,12 +1434,8 @@
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- i2c_unregister_device(codec->control_data);
- i2c_del_driver(&aic3x_i2c_driver);
-#endif
- kfree(codec->private_data);
- kfree(codec);
+
+ kfree(codec->reg_cache);
return 0;
}
@@ -1441,13 +1450,15 @@
static int __init aic3x_modinit(void)
{
- return snd_soc_register_dai(&aic3x_dai);
+ aic3x_i2c_init();
+
+ return 0;
}
module_init(aic3x_modinit);
static void __exit aic3x_exit(void)
{
- snd_soc_unregister_dai(&aic3x_dai);
+ aic3x_i2c_exit();
}
module_exit(aic3x_exit);
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index ac827e5..9af1c88 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -282,8 +282,6 @@
int aic3x_button_pressed(struct snd_soc_codec *codec);
struct aic3x_setup_data {
- int i2c_bus;
- unsigned short i2c_address;
unsigned int gpio_func[2];
};
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index ff9b63b..d998799 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -218,6 +218,8 @@
struct snd_soc_codec codec;
int master;
int sysclk_source;
+ int tdm_slots;
+ int tdm_width;
unsigned int mclk_rate;
unsigned int sysclk_rate;
unsigned int fs;
@@ -519,7 +521,7 @@
dev_dbg(codec->dev, "Using %dHz MCLK\n", wm8993->mclk_rate);
reg = wm8993_read(codec, WM8993_CLOCKING_2);
- reg &= ~WM8993_SYSCLK_SRC;
+ reg &= ~(WM8993_MCLK_DIV | WM8993_SYSCLK_SRC);
if (wm8993->mclk_rate > 13500000) {
reg |= WM8993_MCLK_DIV;
wm8993->sysclk_rate = wm8993->mclk_rate / 2;
@@ -527,8 +529,6 @@
reg &= ~WM8993_MCLK_DIV;
wm8993->sysclk_rate = wm8993->mclk_rate;
}
- reg &= ~WM8993_MCLK_DIV;
- reg &= ~(WM8993_MCLK_DIV | WM8993_SYSCLK_SRC);
wm8993_write(codec, WM8993_CLOCKING_2, reg);
break;
@@ -1189,24 +1189,30 @@
/* What BCLK do we need? */
wm8993->fs = params_rate(params);
wm8993->bclk = 2 * wm8993->fs;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
- wm8993->bclk *= 16;
- break;
- case SNDRV_PCM_FORMAT_S20_3LE:
- wm8993->bclk *= 20;
- aif1 |= 0x8;
- break;
- case SNDRV_PCM_FORMAT_S24_LE:
- wm8993->bclk *= 24;
- aif1 |= 0x10;
- break;
- case SNDRV_PCM_FORMAT_S32_LE:
- wm8993->bclk *= 32;
- aif1 |= 0x18;
- break;
- default:
- return -EINVAL;
+ if (wm8993->tdm_slots) {
+ dev_dbg(codec->dev, "Configuring for %d %d bit TDM slots\n",
+ wm8993->tdm_slots, wm8993->tdm_width);
+ wm8993->bclk *= wm8993->tdm_width * wm8993->tdm_slots;
+ } else {
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ wm8993->bclk *= 16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ wm8993->bclk *= 20;
+ aif1 |= 0x8;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ wm8993->bclk *= 24;
+ aif1 |= 0x10;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ wm8993->bclk *= 32;
+ aif1 |= 0x18;
+ break;
+ default:
+ return -EINVAL;
+ }
}
dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm8993->bclk);
@@ -1325,12 +1331,67 @@
return 0;
}
+static int wm8993_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+ unsigned int rx_mask, int slots, int slot_width)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8993_priv *wm8993 = codec->private_data;
+ int aif1 = 0;
+ int aif2 = 0;
+
+ /* Don't need to validate anything if we're turning off TDM */
+ if (slots == 0) {
+ wm8993->tdm_slots = 0;
+ goto out;
+ }
+
+ /* Note that we allow configurations we can't handle ourselves -
+ * for example, we can generate clocks for slots 2 and up even if
+ * we can't use those slots ourselves.
+ */
+ aif1 |= WM8993_AIFADC_TDM;
+ aif2 |= WM8993_AIFDAC_TDM;
+
+ switch (rx_mask) {
+ case 3:
+ break;
+ case 0xc:
+ aif1 |= WM8993_AIFADC_TDM_CHAN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+
+ switch (tx_mask) {
+ case 3:
+ break;
+ case 0xc:
+ aif2 |= WM8993_AIFDAC_TDM_CHAN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+out:
+ wm8993->tdm_width = slot_width;
+ wm8993->tdm_slots = slots / 2;
+
+ snd_soc_update_bits(codec, WM8993_AUDIO_INTERFACE_1,
+ WM8993_AIFADC_TDM | WM8993_AIFADC_TDM_CHAN, aif1);
+ snd_soc_update_bits(codec, WM8993_AUDIO_INTERFACE_2,
+ WM8993_AIFDAC_TDM | WM8993_AIFDAC_TDM_CHAN, aif2);
+
+ return 0;
+}
+
static struct snd_soc_dai_ops wm8993_ops = {
.set_sysclk = wm8993_set_sysclk,
.set_fmt = wm8993_set_dai_fmt,
.hw_params = wm8993_hw_params,
.digital_mute = wm8993_digital_mute,
.set_pll = wm8993_set_fll,
+ .set_tdm_slot = wm8993_set_tdm_slot,
};
#define WM8993_RATES SNDRV_PCM_RATE_8000_48000
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index fa88b46..e7d2840 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -406,7 +406,7 @@
ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm9705: failed to register card\n");
- goto pcm_err;
+ goto reset_err;
}
return 0;
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index e8fc474..41699bd 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -18,7 +18,6 @@
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
-#include <linux/regulator/consumer.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -474,12 +473,6 @@
SND_SOC_DAPM_MIXER("MIXINR", WM8993_POWER_MANAGEMENT_2, 8, 0,
mixinr, ARRAY_SIZE(mixinr)),
-SND_SOC_DAPM_ADC("ADCL", "Capture", WM8993_POWER_MANAGEMENT_2, 1, 0),
-SND_SOC_DAPM_ADC("ADCR", "Capture", WM8993_POWER_MANAGEMENT_2, 0, 0),
-
-SND_SOC_DAPM_DAC("DACL", "Playback", WM8993_POWER_MANAGEMENT_3, 1, 0),
-SND_SOC_DAPM_DAC("DACR", "Playback", WM8993_POWER_MANAGEMENT_3, 0, 0),
-
SND_SOC_DAPM_MIXER("Left Output Mixer", WM8993_POWER_MANAGEMENT_3, 5, 0,
left_output_mixer, ARRAY_SIZE(left_output_mixer)),
SND_SOC_DAPM_MIXER("Right Output Mixer", WM8993_POWER_MANAGEMENT_3, 4, 0,
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 46c1b0c..0190c1b 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -14,6 +14,7 @@
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
+#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
@@ -207,8 +208,6 @@
/* evm audio private data */
static struct aic3x_setup_data evm_aic3x_setup = {
- .i2c_bus = 1,
- .i2c_address = 0x1b,
};
/* dm6467 evm audio private data */
@@ -251,6 +250,13 @@
static struct platform_device *evm_snd_device;
+/* temporary i2c device creation until this can be moved into the machine
+ * support file.
+*/
+static struct i2c_board_info i2c_device[] = {
+ { I2C_BOARD_INFO("tlv320aic33", 0x1b), }
+};
+
static int __init evm_init(void)
{
struct snd_soc_device *evm_snd_dev_data;
@@ -275,6 +281,8 @@
} else
return -EINVAL;
+ i2c_register_board_info(1, i2c_device, ARRAY_SIZE(i2c_device));
+
evm_snd_device = platform_device_alloc("soc-audio", index);
if (!evm_snd_device)
return -ENOMEM;
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index b60b1df..0a50593 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -22,6 +22,7 @@
*/
#include <linux/clk.h>
+#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -322,8 +323,6 @@
/* Audio private data */
static struct aic3x_setup_data n810_aic33_setup = {
- .i2c_bus = 2,
- .i2c_address = 0x18,
.gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED,
.gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT,
};
@@ -337,6 +336,13 @@
static struct platform_device *n810_snd_device;
+/* temporary i2c device creation until this can be moved into the machine
+ * support file.
+*/
+static struct i2c_board_info i2c_device[] = {
+ { I2C_BOARD_INFO("tlv320aic3x", 0x1b), }
+};
+
static int __init n810_soc_init(void)
{
int err;
@@ -345,6 +351,8 @@
if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax()))
return -ENODEV;
+ i2c_register_board_info(1, i2c_device, ARRAY_SIZE(i2c_device));
+
n810_snd_device = platform_device_alloc("soc-audio", -1);
if (!n810_snd_device)
return -ENOMEM;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 6a837ff..f5387d9 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -139,27 +139,67 @@
static const unsigned long omap34xx_mcbsp_port[][2] = {};
#endif
+static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ int dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp_data->bus_id);
+ int samples;
+
+ /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
+ if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
+ samples = snd_pcm_lib_period_bytes(substream) >> 1;
+ else
+ samples = 1;
+
+ /* Configure McBSP internal buffer usage */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, samples - 1);
+ else
+ omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, samples - 1);
+}
+
static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ int bus_id = mcbsp_data->bus_id;
int err = 0;
- if (cpu_is_omap343x() && mcbsp_data->bus_id == 1) {
+ if (!cpu_dai->active)
+ err = omap_mcbsp_request(bus_id);
+
+ if (cpu_is_omap343x()) {
+ int dma_op_mode = omap_mcbsp_get_dma_op_mode(bus_id);
+ int max_period;
+
/*
* McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer.
* Set constraint for minimum buffer size to the same than FIFO
* size in order to avoid underruns in playback startup because
* HW is keeping the DMA request active until FIFO is filled.
*/
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096, UINT_MAX);
- }
+ if (bus_id == 1)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ 4096, UINT_MAX);
- if (!cpu_dai->active)
- err = omap_mcbsp_request(mcbsp_data->bus_id);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ max_period = omap_mcbsp_get_max_tx_threshold(bus_id);
+ else
+ max_period = omap_mcbsp_get_max_rx_threshold(bus_id);
+
+ max_period++;
+ max_period <<= 1;
+
+ if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ 32, max_period);
+ }
return err;
}
@@ -191,6 +231,11 @@
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
mcbsp_data->active++;
omap_mcbsp_start(mcbsp_data->bus_id, play, !play);
+ /* Make sure data transfer is frame synchronized */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ omap_mcbsp_xmit_enable(mcbsp_data->bus_id, 1);
+ else
+ omap_mcbsp_recv_enable(mcbsp_data->bus_id, 1);
break;
case SNDRV_PCM_TRIGGER_STOP:
@@ -215,7 +260,7 @@
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
- int wlen, channels, wpf;
+ int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT;
unsigned long port;
unsigned int format;
@@ -231,6 +276,12 @@
} else if (cpu_is_omap343x()) {
dma = omap24xx_dma_reqs[bus_id][substream->stream];
port = omap34xx_mcbsp_port[bus_id][substream->stream];
+ omap_mcbsp_dai_dma_params[id][substream->stream].set_threshold =
+ omap_mcbsp_set_threshold;
+ /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
+ if (omap_mcbsp_get_dma_op_mode(bus_id) ==
+ MCBSP_DMA_MODE_THRESHOLD)
+ sync_mode = OMAP_DMA_SYNC_FRAME;
} else {
return -ENODEV;
}
@@ -238,6 +289,7 @@
substream->stream ? "Audio Capture" : "Audio Playback";
omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
+ omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
if (mcbsp_data->configured) {
@@ -321,8 +373,11 @@
/* Generic McBSP register settings */
regs->spcr2 |= XINTM(3) | FREE;
regs->spcr1 |= RINTM(3);
- regs->rcr2 |= RFIG;
- regs->xcr2 |= XFIG;
+ /* RFIG and XFIG are not defined in 34xx */
+ if (!cpu_is_omap34xx()) {
+ regs->rcr2 |= RFIG;
+ regs->xcr2 |= XFIG;
+ }
if (cpu_is_omap2430() || cpu_is_omap34xx()) {
regs->xccr = DXENDLY(1) | XDMAEN;
regs->rccr = RFULL_CYCLE | RDMAEN;
@@ -333,11 +388,15 @@
/* 1-bit data delay */
regs->rcr2 |= RDATDLY(1);
regs->xcr2 |= XDATDLY(1);
+ regs->rccr |= RFULL_CYCLE | RDMAEN | RDISABLE;
+ regs->xccr |= (DXENDLY(1) | XDMAEN | XDISABLE);
break;
case SND_SOC_DAIFMT_DSP_A:
/* 1-bit data delay */
regs->rcr2 |= RDATDLY(1);
regs->xcr2 |= XDATDLY(1);
+ regs->rccr |= RFULL_CYCLE | RDMAEN | RDISABLE;
+ regs->xccr |= (DXENDLY(1) | XDMAEN | XDISABLE);
/* Invert FS polarity configuration */
temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
break;
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 12e14c0..5735945 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -162,7 +162,7 @@
*/
dma_params.data_type = OMAP_DMA_DATA_TYPE_S16;
dma_params.trigger = dma_data->dma_req;
- dma_params.sync_mode = OMAP_DMA_SYNC_ELEMENT;
+ dma_params.sync_mode = dma_data->sync_mode;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
dma_params.src_amode = OMAP_DMA_AMODE_POST_INC;
dma_params.dst_amode = OMAP_DMA_AMODE_CONSTANT;
@@ -195,6 +195,9 @@
else
omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
+ omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
+ omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
+
return 0;
}
@@ -202,6 +205,7 @@
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct omap_runtime_data *prtd = runtime->private_data;
+ struct omap_pcm_dma_data *dma_data = prtd->dma_data;
unsigned long flags;
int ret = 0;
@@ -211,6 +215,10 @@
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
prtd->period_index = 0;
+ /* Configure McBSP internal buffer usage */
+ if (dma_data->set_threshold)
+ dma_data->set_threshold(substream);
+
omap_start_dma(prtd->dma_ch);
break;
@@ -307,7 +315,7 @@
.mmap = omap_pcm_mmap,
};
-static u64 omap_pcm_dmamask = DMA_BIT_MASK(32);
+static u64 omap_pcm_dmamask = DMA_BIT_MASK(64);
static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
int stream)
@@ -357,7 +365,7 @@
if (!card->dev->dma_mask)
card->dev->dma_mask = &omap_pcm_dmamask;
if (!card->dev->coherent_dma_mask)
- card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(64);
if (dai->playback.channels_min) {
ret = omap_pcm_preallocate_dma_buffer(pcm,
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
index 8d9d269..38a821d 100644
--- a/sound/soc/omap/omap-pcm.h
+++ b/sound/soc/omap/omap-pcm.h
@@ -29,6 +29,8 @@
char *name; /* stream identifier */
int dma_req; /* DMA request line */
unsigned long port_addr; /* transmit/receive register */
+ int sync_mode; /* DMA sync mode */
+ void (*set_threshold)(struct snd_pcm_substream *substream);
};
extern struct snd_soc_platform omap_soc_platform;
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 7330e5c..e9ae7b3 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -251,8 +251,8 @@
for (i = 0; i < ARRAY_SIZE(pxa_ac97_dai); i++) {
pxa_ac97_dai[i].dev = &pdev->dev;
- if (pdata && pdata->codec_pdata)
- pxa_ac97_dai[i].ac97_pdata = pdata->codec_pdata;
+ if (pdata && pdata->codec_pdata[0])
+ pxa_ac97_dai[i].ac97_pdata = pdata->codec_pdata[0];
}
/* Punt most of the init to the SoC probe; we may need the machine
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 808de5c..68fef00 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,6 +1,7 @@
config SND_S3C24XX_SOC
tristate "SoC Audio for the Samsung S3CXXXX chips"
- depends on ARCH_S3C2410
+ depends on ARCH_S3C2410 || ARCH_S3C64XX
+ select S3C64XX_DMA if ARCH_S3C64XX
help
Say Y or M if you want to add support for codecs attached to
the S3C24XX AC97 or I2S interfaces. You will also need to
@@ -79,3 +80,22 @@
select SND_S3C24XX_SOC_I2S
select SND_SOC_L3
select SND_SOC_UDA134X
+
+config SND_S3C24XX_SOC_SIMTEC
+ tristate
+ help
+ Internal node for common S3C24XX/Simtec suppor
+
+config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23
+ tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards"
+ depends on SND_S3C24XX_SOC
+ select SND_S3C24XX_SOC_I2S
+ select SND_SOC_TLV320AIC23
+ select SND_S3C24XX_SOC_SIMTEC
+
+config SND_S3C24XX_SOC_SIMTEC_HERMES
+ tristate "SoC I2S Audio support for Simtec Hermes board"
+ depends on SND_S3C24XX_SOC
+ select SND_S3C24XX_SOC_I2S
+ select SND_SOC_TLV320AIC3X
+ select SND_S3C24XX_SOC_SIMTEC
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index eb219b0..99f5a7d 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -20,6 +20,9 @@
snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
+snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o
+snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o
+snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o
obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o
obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
@@ -27,3 +30,7 @@
obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
+obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o
+obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o
+obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o
+
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 1a28317..ebfb2f6 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -357,19 +357,19 @@
#endif
#ifdef CONFIG_PLAT_S3C64XX
- iismod &= ~0x606;
+ iismod &= ~(S3C64XX_IISMOD_BLC_MASK | S3C2412_IISMOD_BCLK_MASK);
/* Sample size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
/* 8 bit sample, 16fs BCLK */
- iismod |= 0x2004;
+ iismod |= (S3C64XX_IISMOD_BLC_8BIT | S3C2412_IISMOD_BCLK_16FS);
break;
case SNDRV_PCM_FORMAT_S16_LE:
/* 16 bit sample, 32fs BCLK */
break;
case SNDRV_PCM_FORMAT_S24_LE:
/* 24 bit sample, 48fs BCLK */
- iismod |= 0x4002;
+ iismod |= (S3C64XX_IISMOD_BLC_24BIT | S3C2412_IISMOD_BCLK_48FS);
break;
}
#endif
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index eecfa5e..8a93196 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -318,6 +318,7 @@
pr_debug("Entered %s\n", __func__);
+ snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware);
prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL);
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c
new file mode 100644
index 0000000..1966e0d
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.c
@@ -0,0 +1,394 @@
+/* sound/soc/s3c24xx/s3c24xx_simtec.c
+ *
+ * Copyright 2009 Simtec Electronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+*/
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <linux/clk.h>
+#include <linux/i2c.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <plat/audio-simtec.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+#include "s3c24xx_simtec.h"
+
+static struct s3c24xx_audio_simtec_pdata *pdata;
+static struct clk *xtal_clk;
+
+static int spk_gain;
+static int spk_unmute;
+
+/**
+ * speaker_gain_get - read the speaker gain setting.
+ * @kcontrol: The control for the speaker gain.
+ * @ucontrol: The value that needs to be updated.
+ *
+ * Read the value for the AMP gain control.
+ */
+static int speaker_gain_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = spk_gain;
+ return 0;
+}
+
+/**
+ * speaker_gain_set - set the value of the speaker amp gain
+ * @value: The value to write.
+ */
+static void speaker_gain_set(int value)
+{
+ gpio_set_value_cansleep(pdata->amp_gain[0], value & 1);
+ gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1);
+}
+
+/**
+ * speaker_gain_put - set the speaker gain setting.
+ * @kcontrol: The control for the speaker gain.
+ * @ucontrol: The value that needs to be set.
+ *
+ * Set the value of the speaker gain from the specified
+ * @ucontrol setting.
+ *
+ * Note, if the speaker amp is muted, then we do not set a gain value
+ * as at-least one of the ICs that is fitted will try and power up even
+ * if the main control is set to off.
+ */
+static int speaker_gain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int value = ucontrol->value.integer.value[0];
+
+ spk_gain = value;
+
+ if (!spk_unmute)
+ speaker_gain_set(value);
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new amp_gain_controls[] = {
+ SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0,
+ speaker_gain_get, speaker_gain_put),
+};
+
+/**
+ * spk_unmute_state - set the unmute state of the speaker
+ * @to: zero to unmute, non-zero to ununmute.
+ */
+static void spk_unmute_state(int to)
+{
+ pr_debug("%s: to=%d\n", __func__, to);
+
+ spk_unmute = to;
+ gpio_set_value(pdata->amp_gpio, to);
+
+ /* if we're umuting, also re-set the gain */
+ if (to && pdata->amp_gain[0] > 0)
+ speaker_gain_set(spk_gain);
+}
+
+/**
+ * speaker_unmute_get - read the speaker unmute setting.
+ * @kcontrol: The control for the speaker gain.
+ * @ucontrol: The value that needs to be updated.
+ *
+ * Read the value for the AMP gain control.
+ */
+static int speaker_unmute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = spk_unmute;
+ return 0;
+}
+
+/**
+ * speaker_unmute_put - set the speaker unmute setting.
+ * @kcontrol: The control for the speaker gain.
+ * @ucontrol: The value that needs to be set.
+ *
+ * Set the value of the speaker gain from the specified
+ * @ucontrol setting.
+ */
+static int speaker_unmute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ spk_unmute_state(ucontrol->value.integer.value[0]);
+ return 0;
+}
+
+/* This is added as a manual control as the speaker amps create clicks
+ * when their power state is changed, which are far more noticeable than
+ * anything produced by the CODEC itself.
+ */
+static const struct snd_kcontrol_new amp_unmute_controls[] = {
+ SOC_SINGLE_EXT("Speaker Switch", 0, 0, 1, 0,
+ speaker_unmute_get, speaker_unmute_put),
+};
+
+void simtec_audio_init(struct snd_soc_codec *codec)
+{
+ if (pdata->amp_gpio > 0) {
+ pr_debug("%s: adding amp routes\n", __func__);
+
+ snd_soc_add_controls(codec, amp_unmute_controls,
+ ARRAY_SIZE(amp_unmute_controls));
+ }
+
+ if (pdata->amp_gain[0] > 0) {
+ pr_debug("%s: adding amp controls\n", __func__);
+ snd_soc_add_controls(codec, amp_gain_controls,
+ ARRAY_SIZE(amp_gain_controls));
+ }
+}
+EXPORT_SYMBOL_GPL(simtec_audio_init);
+
+#define CODEC_CLOCK 12000000
+
+/**
+ * simtec_hw_params - update hardware parameters
+ * @substream: The audio substream instance.
+ * @params: The parameters requested.
+ *
+ * Update the codec data routing and configuration settings
+ * from the supplied data.
+ */
+static int simtec_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->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* Set the CODEC as the bus clock master, I2S */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret) {
+ pr_err("%s: failed set cpu dai format\n", __func__);
+ return ret;
+ }
+
+ /* Set the CODEC as the bus clock master */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret) {
+ pr_err("%s: failed set codec dai format\n", __func__);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0,
+ CODEC_CLOCK, SND_SOC_CLOCK_IN);
+ if (ret) {
+ pr_err( "%s: failed setting codec sysclk\n", __func__);
+ return ret;
+ }
+
+ if (pdata->use_mpllin) {
+ ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL,
+ 0, SND_SOC_CLOCK_OUT);
+
+ if (ret) {
+ pr_err("%s: failed to set MPLLin as clksrc\n",
+ __func__);
+ return ret;
+ }
+ }
+
+ if (pdata->output_cdclk) {
+ int cdclk_scale;
+
+ cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK;
+ cdclk_scale--;
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+ cdclk_scale);
+ }
+
+ return 0;
+}
+
+static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd)
+{
+ /* call any board supplied startup code, this currently only
+ * covers the bast/vr1000 which have a CPLD in the way of the
+ * LRCLK */
+ if (pd->startup)
+ pd->startup();
+
+ return 0;
+}
+
+static struct snd_soc_ops simtec_snd_ops = {
+ .hw_params = simtec_hw_params,
+};
+
+/**
+ * attach_gpio_amp - get and configure the necessary gpios
+ * @dev: The device we're probing.
+ * @pd: The platform data supplied by the board.
+ *
+ * If there is a GPIO based amplifier attached to the board, claim
+ * the necessary GPIO lines for it, and set default values.
+ */
+static int attach_gpio_amp(struct device *dev,
+ struct s3c24xx_audio_simtec_pdata *pd)
+{
+ int ret;
+
+ /* attach gpio amp gain (if any) */
+ if (pdata->amp_gain[0] > 0) {
+ ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0");
+ if (ret) {
+ dev_err(dev, "cannot get amp gpio gain0\n");
+ return ret;
+ }
+
+ ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1");
+ if (ret) {
+ dev_err(dev, "cannot get amp gpio gain1\n");
+ gpio_free(pdata->amp_gain[0]);
+ return ret;
+ }
+
+ gpio_direction_output(pd->amp_gain[0], 0);
+ gpio_direction_output(pd->amp_gain[1], 0);
+ }
+
+ /* note, curently we assume GPA0 isn't valid amp */
+ if (pdata->amp_gpio > 0) {
+ ret = gpio_request(pd->amp_gpio, "gpio-amp");
+ if (ret) {
+ dev_err(dev, "cannot get amp gpio %d (%d)\n",
+ pd->amp_gpio, ret);
+ goto err_amp;
+ }
+
+ /* set the amp off at startup */
+ spk_unmute_state(0);
+ }
+
+ return 0;
+
+err_amp:
+ if (pd->amp_gain[0] > 0) {
+ gpio_free(pd->amp_gain[0]);
+ gpio_free(pd->amp_gain[1]);
+ }
+
+ return ret;
+}
+
+static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd)
+{
+ if (pd->amp_gain[0] > 0) {
+ gpio_free(pd->amp_gain[0]);
+ gpio_free(pd->amp_gain[1]);
+ }
+
+ if (pd->amp_gpio > 0)
+ gpio_free(pd->amp_gpio);
+}
+
+#ifdef CONFIG_PM
+int simtec_audio_resume(struct device *dev)
+{
+ simtec_call_startup(pdata);
+ return 0;
+}
+
+struct dev_pm_ops simtec_audio_pmops = {
+ .resume = simtec_audio_resume,
+};
+EXPORT_SYMBOL_GPL(simtec_audio_pmops);
+#endif
+
+int __devinit simtec_audio_core_probe(struct platform_device *pdev,
+ struct snd_soc_device *socdev)
+{
+ struct platform_device *snd_dev;
+ int ret;
+
+ socdev->card->dai_link->ops = &simtec_snd_ops;
+
+ pdata = pdev->dev.platform_data;
+ if (!pdata) {
+ dev_err(&pdev->dev, "no platform data supplied\n");
+ return -EINVAL;
+ }
+
+ simtec_call_startup(pdata);
+
+ xtal_clk = clk_get(&pdev->dev, "xtal");
+ if (IS_ERR(xtal_clk)) {
+ dev_err(&pdev->dev, "could not get clkout0\n");
+ return -EINVAL;
+ }
+
+ dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk));
+
+ ret = attach_gpio_amp(&pdev->dev, pdata);
+ if (ret)
+ goto err_clk;
+
+ snd_dev = platform_device_alloc("soc-audio", -1);
+ if (!snd_dev) {
+ dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n");
+ ret = -ENOMEM;
+ goto err_gpio;
+ }
+
+ platform_set_drvdata(snd_dev, socdev);
+ socdev->dev = &snd_dev->dev;
+
+ ret = platform_device_add(snd_dev);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to add soc-audio dev\n");
+ goto err_pdev;
+ }
+
+ platform_set_drvdata(pdev, snd_dev);
+ return 0;
+
+err_pdev:
+ platform_device_put(snd_dev);
+
+err_gpio:
+ detach_gpio_amp(pdata);
+
+err_clk:
+ clk_put(xtal_clk);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(simtec_audio_core_probe);
+
+int __devexit simtec_audio_remove(struct platform_device *pdev)
+{
+ struct platform_device *snd_dev = platform_get_drvdata(pdev);
+
+ platform_device_unregister(snd_dev);
+
+ detach_gpio_amp(pdata);
+ clk_put(xtal_clk);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(simtec_audio_remove);
+
+MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.h b/sound/soc/s3c24xx/s3c24xx_simtec.h
new file mode 100644
index 0000000..2714203
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.h
@@ -0,0 +1,22 @@
+/* sound/soc/s3c24xx/s3c24xx_simtec.h
+ *
+ * Copyright 2009 Simtec Electronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+*/
+
+extern void simtec_audio_init(struct snd_soc_codec *codec);
+
+extern int simtec_audio_core_probe(struct platform_device *pdev,
+ struct snd_soc_device *socdev);
+
+extern int simtec_audio_remove(struct platform_device *pdev);
+
+#ifdef CONFIG_PM
+extern struct dev_pm_ops simtec_audio_pmops;
+#define simtec_audio_pm &simtec_audio_pmops
+#else
+#define simtec_audio_pm NULL
+#endif
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
new file mode 100644
index 0000000..8346bd9
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
@@ -0,0 +1,153 @@
+/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
+ *
+ * Copyright 2009 Simtec Electronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+*/
+
+#include <linux/module.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <plat/audio-simtec.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+#include "s3c24xx_simtec.h"
+
+#include "../codecs/tlv320aic3x.h"
+
+static const struct snd_soc_dapm_widget dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("GSM Out", NULL),
+ SND_SOC_DAPM_LINE("GSM In", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+ SND_SOC_DAPM_LINE("ZV", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route base_map[] = {
+ /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */
+
+ { "Headphone Jack", NULL, "HPLOUT" },
+ { "Headphone Jack", NULL, "HPLCOM" },
+ { "Headphone Jack", NULL, "HPROUT" },
+ { "Headphone Jack", NULL, "HPRCOM" },
+
+ /* ZV connected to Line1 */
+
+ { "LINE1L", NULL, "ZV" },
+ { "LINE1R", NULL, "ZV" },
+
+ /* Line In connected to Line2 */
+
+ { "LINE2L", NULL, "Line In" },
+ { "LINE2R", NULL, "Line In" },
+
+ /* Microphone connected to MIC3R and MIC_BIAS */
+
+ { "MIC3L", NULL, "Mic Jack" },
+
+ /* GSM connected to MONO_LOUT and MIC3L (in) */
+
+ { "GSM Out", NULL, "MONO_LOUT" },
+ { "MIC3L", NULL, "GSM In" },
+
+ /* Speaker is connected to LINEOUT{LN,LP,RN,RP}, however we are
+ * not using the DAPM to power it up and down as there it makes
+ * a click when powering up. */
+};
+
+/**
+ * simtec_hermes_init - initialise and add controls
+ * @codec; The codec instance to attach to.
+ *
+ * Attach our controls and configure the necessary codec
+ * mappings for our sound card instance.
+*/
+static int simtec_hermes_init(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, dapm_widgets,
+ ARRAY_SIZE(dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map));
+
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Line In");
+ snd_soc_dapm_enable_pin(codec, "Line Out");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+ simtec_audio_init(codec);
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static struct aic3x_setup_data codec_setup = {
+};
+
+static struct snd_soc_dai_link simtec_dai_aic33 = {
+ .name = "tlv320aic33",
+ .stream_name = "TLV320AIC33",
+ .cpu_dai = &s3c24xx_i2s_dai,
+ .codec_dai = &aic3x_dai,
+ .init = simtec_hermes_init,
+};
+
+/* simtec audio machine driver */
+static struct snd_soc_card snd_soc_machine_simtec_aic33 = {
+ .name = "Simtec-Hermes",
+ .platform = &s3c24xx_soc_platform,
+ .dai_link = &simtec_dai_aic33,
+ .num_links = 1,
+};
+
+/* simtec audio subsystem */
+static struct snd_soc_device simtec_snd_devdata_aic33 = {
+ .card = &snd_soc_machine_simtec_aic33,
+ .codec_dev = &soc_codec_dev_aic3x,
+ .codec_data = &codec_setup,
+};
+
+static int __devinit simtec_audio_hermes_probe(struct platform_device *pd)
+{
+ dev_info(&pd->dev, "probing....\n");
+ return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33);
+}
+
+static struct platform_driver simtec_audio_hermes_platdrv = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "s3c24xx-simtec-hermes-snd",
+ .pm = simtec_audio_pm,
+ },
+ .probe = simtec_audio_hermes_probe,
+ .remove = __devexit_p(simtec_audio_remove),
+};
+
+MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd");
+
+static int __init simtec_hermes_modinit(void)
+{
+ return platform_driver_register(&simtec_audio_hermes_platdrv);
+}
+
+static void __exit simtec_hermes_modexit(void)
+{
+ platform_driver_unregister(&simtec_audio_hermes_platdrv);
+}
+
+module_init(simtec_hermes_modinit);
+module_exit(simtec_hermes_modexit);
+
+MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC Simtec Audio support");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
new file mode 100644
index 0000000..25797e0
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
@@ -0,0 +1,137 @@
+/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
+ *
+ * Copyright 2009 Simtec Electronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+*/
+
+#include <linux/module.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <plat/audio-simtec.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+#include "s3c24xx_simtec.h"
+
+#include "../codecs/tlv320aic23.h"
+
+/* supported machines:
+ *
+ * Machine Connections AMP
+ * ------- ----------- ---
+ * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired)
+ * VR1000 HPOUT, LIN None
+ * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R)
+ * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R)
+ * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R)
+ */
+
+static const struct snd_soc_dapm_widget dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route base_map[] = {
+ { "Headphone Jack", NULL, "LHPOUT"},
+ { "Headphone Jack", NULL, "RHPOUT"},
+
+ { "Line Out", NULL, "LOUT" },
+ { "Line Out", NULL, "ROUT" },
+
+ { "LLINEIN", NULL, "Line In"},
+ { "RLINEIN", NULL, "Line In"},
+
+ { "MICIN", NULL, "Mic Jack"},
+};
+
+/**
+ * simtec_tlv320aic23_init - initialise and add controls
+ * @codec; The codec instance to attach to.
+ *
+ * Attach our controls and configure the necessary codec
+ * mappings for our sound card instance.
+*/
+static int simtec_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, dapm_widgets,
+ ARRAY_SIZE(dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map));
+
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Line In");
+ snd_soc_dapm_enable_pin(codec, "Line Out");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+ simtec_audio_init(codec);
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static struct snd_soc_dai_link simtec_dai_aic23 = {
+ .name = "tlv320aic23",
+ .stream_name = "TLV320AIC23",
+ .cpu_dai = &s3c24xx_i2s_dai,
+ .codec_dai = &tlv320aic23_dai,
+ .init = simtec_tlv320aic23_init,
+};
+
+/* simtec audio machine driver */
+static struct snd_soc_card snd_soc_machine_simtec_aic23 = {
+ .name = "Simtec",
+ .platform = &s3c24xx_soc_platform,
+ .dai_link = &simtec_dai_aic23,
+ .num_links = 1,
+};
+
+/* simtec audio subsystem */
+static struct snd_soc_device simtec_snd_devdata_aic23 = {
+ .card = &snd_soc_machine_simtec_aic23,
+ .codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd)
+{
+ return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23);
+}
+
+static struct platform_driver simtec_audio_tlv320aic23_platdrv = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "s3c24xx-simtec-tlv320aic23",
+ .pm = simtec_audio_pm,
+ },
+ .probe = simtec_audio_tlv320aic23_probe,
+ .remove = __devexit_p(simtec_audio_remove),
+};
+
+MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23");
+
+static int __init simtec_tlv320aic23_modinit(void)
+{
+ return platform_driver_register(&simtec_audio_tlv320aic23_platdrv);
+}
+
+static void __exit simtec_tlv320aic23_modexit(void)
+{
+ platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv);
+}
+
+module_init(simtec_tlv320aic23_modinit);
+module_exit(simtec_tlv320aic23_modexit);
+
+MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC Simtec Audio support");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
index b5f95f9..c1b40ac 100644
--- a/sound/soc/s6000/s6105-ipcam.c
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -14,6 +14,7 @@
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
+#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
@@ -189,8 +190,6 @@
/* s6105 audio private data */
static struct aic3x_setup_data s6105_aic3x_setup = {
- .i2c_bus = 0,
- .i2c_address = 0x18,
};
/* s6105 audio subsystem */
@@ -211,10 +210,19 @@
static struct platform_device *s6105_snd_device;
+/* temporary i2c device creation until this can be moved into the machine
+ * support file.
+*/
+static struct i2c_board_info i2c_device[] = {
+ { I2C_BOARD_INFO("tlv320aic33", 0x18), }
+};
+
static int __init s6105_init(void)
{
int ret;
+ i2c_register_board_info(0, i2c_device, ARRAY_SIZE(i2c_device));
+
s6105_snd_device = platform_device_alloc("soc-audio", -1);
if (!s6105_snd_device)
return -ENOMEM;
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 54bd604..9154b43 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -20,7 +20,12 @@
config SND_SOC_SH4_SSI
tristate
-
+config SND_SOC_SH4_FSI
+ tristate "SH4 FSI support"
+ depends on CPU_SUBTYPE_SH7724
+ select SH_DMA
+ help
+ This option enables FSI sound support
##
## Boards
@@ -35,4 +40,12 @@
This option enables generic sound support for the first
AC97 unit of the SH7760.
+config SND_FSI_AK4642
+ bool "FSI-AK4642 sound support"
+ depends on SND_SOC_SH4_FSI
+ select SND_SOC_AK4642
+ help
+ This option enables generic sound support for the
+ FSI - AK4642 unit
+
endmenu
diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile
index a8e8ab8..a699787 100644
--- a/sound/soc/sh/Makefile
+++ b/sound/soc/sh/Makefile
@@ -5,10 +5,14 @@
## audio units found on some SH-4
snd-soc-hac-objs := hac.o
snd-soc-ssi-objs := ssi.o
+snd-soc-fsi-objs := fsi.o
obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o
obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o
+obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o
## boards
snd-soc-sh7760-ac97-objs := sh7760-ac97.o
+snd-soc-fsi-ak4642-objs := fsi-ak4642.o
obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o
+obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
new file mode 100644
index 0000000..c7af097
--- /dev/null
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -0,0 +1,107 @@
+/*
+ * FSI-AK464x sound support for ms7724se
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * This file is subject to the terms and conditions of the GNU General Public
+ * License. See the file "COPYING" in the main directory of this archive
+ * for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+#include <linux/i2c.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <sound/sh_fsi.h>
+#include <../sound/soc/codecs/ak4642.h>
+
+static struct snd_soc_dai_link fsi_dai_link = {
+ .name = "AK4642",
+ .stream_name = "AK4642",
+ .cpu_dai = &fsi_soc_dai[0], /* fsi */
+ .codec_dai = &ak4642_dai,
+ .ops = NULL,
+};
+
+static struct snd_soc_card fsi_soc_card = {
+ .name = "FSI",
+ .platform = &fsi_soc_platform,
+ .dai_link = &fsi_dai_link,
+ .num_links = 1,
+};
+
+static struct snd_soc_device fsi_snd_devdata = {
+ .card = &fsi_soc_card,
+ .codec_dev = &soc_codec_dev_ak4642,
+};
+
+#define AK4642_BUS 0
+#define AK4642_ADR 0x12
+static int ak4642_add_i2c_device(void)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = AK4642_ADR;
+ strlcpy(info.type, "ak4642", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(AK4642_BUS);
+ if (!adapter) {
+ printk(KERN_DEBUG "can't get i2c adapter\n");
+ return -ENODEV;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ printk(KERN_DEBUG "can't add i2c device\n");
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+static struct platform_device *fsi_snd_device;
+
+static int __init fsi_ak4642_init(void)
+{
+ int ret = -ENOMEM;
+
+ ak4642_add_i2c_device();
+
+ fsi_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!fsi_snd_device)
+ goto out;
+
+ platform_set_drvdata(fsi_snd_device,
+ &fsi_snd_devdata);
+ fsi_snd_devdata.dev = &fsi_snd_device->dev;
+ ret = platform_device_add(fsi_snd_device);
+
+ if (ret)
+ platform_device_put(fsi_snd_device);
+
+out:
+ return ret;
+}
+
+static void __exit fsi_ak4642_exit(void)
+{
+ platform_device_unregister(fsi_snd_device);
+}
+
+module_init(fsi_ak4642_init);
+module_exit(fsi_ak4642_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Generic SH4 FSI-AK4642 sound card");
+MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
new file mode 100644
index 0000000..4412324
--- /dev/null
+++ b/sound/soc/sh/fsi.c
@@ -0,0 +1,1004 @@
+/*
+ * Fifo-attached Serial Interface (FSI) support for SH7724
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * Based on ssi.c
+ * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/list.h>
+#include <linux/clk.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include <sound/sh_fsi.h>
+#include <asm/atomic.h>
+#include <asm/dma.h>
+#include <asm/dma-sh.h>
+
+#define DO_FMT 0x0000
+#define DOFF_CTL 0x0004
+#define DOFF_ST 0x0008
+#define DI_FMT 0x000C
+#define DIFF_CTL 0x0010
+#define DIFF_ST 0x0014
+#define CKG1 0x0018
+#define CKG2 0x001C
+#define DIDT 0x0020
+#define DODT 0x0024
+#define MUTE_ST 0x0028
+#define REG_END MUTE_ST
+
+#define INT_ST 0x0200
+#define IEMSK 0x0204
+#define IMSK 0x0208
+#define MUTE 0x020C
+#define CLK_RST 0x0210
+#define SOFT_RST 0x0214
+#define MREG_START INT_ST
+#define MREG_END SOFT_RST
+
+/* DO_FMT */
+/* DI_FMT */
+#define CR_FMT(param) ((param) << 4)
+# define CR_MONO 0x0
+# define CR_MONO_D 0x1
+# define CR_PCM 0x2
+# define CR_I2S 0x3
+# define CR_TDM 0x4
+# define CR_TDM_D 0x5
+
+/* DOFF_CTL */
+/* DIFF_CTL */
+#define IRQ_HALF 0x00100000
+#define FIFO_CLR 0x00000001
+
+/* DOFF_ST */
+#define ERR_OVER 0x00000010
+#define ERR_UNDER 0x00000001
+
+/* CLK_RST */
+#define B_CLK 0x00000010
+#define A_CLK 0x00000001
+
+/* INT_ST */
+#define INT_B_IN (1 << 12)
+#define INT_B_OUT (1 << 8)
+#define INT_A_IN (1 << 4)
+#define INT_A_OUT (1 << 0)
+
+#define FSI_RATES SNDRV_PCM_RATE_8000_96000
+
+#define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
+
+/************************************************************************
+
+
+ struct
+
+
+************************************************************************/
+struct fsi_priv {
+ void __iomem *base;
+ struct snd_pcm_substream *substream;
+
+ int fifo_max;
+ int chan;
+ int dma_chan;
+
+ int byte_offset;
+ int period_len;
+ int buffer_len;
+ int periods;
+};
+
+struct fsi_master {
+ void __iomem *base;
+ int irq;
+ struct clk *clk;
+ struct fsi_priv fsia;
+ struct fsi_priv fsib;
+ struct sh_fsi_platform_info *info;
+};
+
+static struct fsi_master *master;
+
+/************************************************************************
+
+
+ basic read write function
+
+
+************************************************************************/
+static int __fsi_reg_write(u32 reg, u32 data)
+{
+ /* valid data area is 24bit */
+ data &= 0x00ffffff;
+
+ return ctrl_outl(data, reg);
+}
+
+static u32 __fsi_reg_read(u32 reg)
+{
+ return ctrl_inl(reg);
+}
+
+static int __fsi_reg_mask_set(u32 reg, u32 mask, u32 data)
+{
+ u32 val = __fsi_reg_read(reg);
+
+ val &= ~mask;
+ val |= data & mask;
+
+ return __fsi_reg_write(reg, val);
+}
+
+static int fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data)
+{
+ if (reg > REG_END)
+ return -1;
+
+ return __fsi_reg_write((u32)(fsi->base + reg), data);
+}
+
+static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg)
+{
+ if (reg > REG_END)
+ return 0;
+
+ return __fsi_reg_read((u32)(fsi->base + reg));
+}
+
+static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data)
+{
+ if (reg > REG_END)
+ return -1;
+
+ return __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data);
+}
+
+static int fsi_master_write(u32 reg, u32 data)
+{
+ if ((reg < MREG_START) ||
+ (reg > MREG_END))
+ return -1;
+
+ return __fsi_reg_write((u32)(master->base + reg), data);
+}
+
+static u32 fsi_master_read(u32 reg)
+{
+ if ((reg < MREG_START) ||
+ (reg > MREG_END))
+ return 0;
+
+ return __fsi_reg_read((u32)(master->base + reg));
+}
+
+static int fsi_master_mask_set(u32 reg, u32 mask, u32 data)
+{
+ if ((reg < MREG_START) ||
+ (reg > MREG_END))
+ return -1;
+
+ return __fsi_reg_mask_set((u32)(master->base + reg), mask, data);
+}
+
+/************************************************************************
+
+
+ basic function
+
+
+************************************************************************/
+static struct fsi_priv *fsi_get(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd;
+ struct fsi_priv *fsi = NULL;
+
+ if (!substream || !master)
+ return NULL;
+
+ rtd = substream->private_data;
+ switch (rtd->dai->cpu_dai->id) {
+ case 0:
+ fsi = &master->fsia;
+ break;
+ case 1:
+ fsi = &master->fsib;
+ break;
+ }
+
+ return fsi;
+}
+
+static int fsi_is_port_a(struct fsi_priv *fsi)
+{
+ /* return
+ * 1 : port a
+ * 0 : port b
+ */
+
+ if (fsi == &master->fsia)
+ return 1;
+
+ return 0;
+}
+
+static u32 fsi_get_info_flags(struct fsi_priv *fsi)
+{
+ int is_porta = fsi_is_port_a(fsi);
+
+ return is_porta ? master->info->porta_flags :
+ master->info->portb_flags;
+}
+
+static int fsi_is_master_mode(struct fsi_priv *fsi, int is_play)
+{
+ u32 mode;
+ u32 flags = fsi_get_info_flags(fsi);
+
+ mode = is_play ? SH_FSI_OUT_SLAVE_MODE : SH_FSI_IN_SLAVE_MODE;
+
+ /* return
+ * 1 : master mode
+ * 0 : slave mode
+ */
+
+ return (mode & flags) != mode;
+}
+
+static u32 fsi_port_ab_io_bit(struct fsi_priv *fsi, int is_play)
+{
+ int is_porta = fsi_is_port_a(fsi);
+ u32 data;
+
+ if (is_porta)
+ data = is_play ? (1 << 0) : (1 << 4);
+ else
+ data = is_play ? (1 << 8) : (1 << 12);
+
+ return data;
+}
+
+static void fsi_stream_push(struct fsi_priv *fsi,
+ struct snd_pcm_substream *substream,
+ u32 buffer_len,
+ u32 period_len)
+{
+ fsi->substream = substream;
+ fsi->buffer_len = buffer_len;
+ fsi->period_len = period_len;
+ fsi->byte_offset = 0;
+ fsi->periods = 0;
+}
+
+static void fsi_stream_pop(struct fsi_priv *fsi)
+{
+ fsi->substream = NULL;
+ fsi->buffer_len = 0;
+ fsi->period_len = 0;
+ fsi->byte_offset = 0;
+ fsi->periods = 0;
+}
+
+static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play)
+{
+ u32 status;
+ u32 reg = is_play ? DOFF_ST : DIFF_ST;
+ int residue;
+
+ status = fsi_reg_read(fsi, reg);
+ residue = 0x1ff & (status >> 8);
+ residue *= fsi->chan;
+
+ return residue;
+}
+
+static int fsi_get_residue(struct fsi_priv *fsi, int is_play)
+{
+ int residue;
+ int width;
+ struct snd_pcm_runtime *runtime;
+
+ runtime = fsi->substream->runtime;
+
+ /* get 1 channel data width */
+ width = frames_to_bytes(runtime, 1) / fsi->chan;
+
+ if (2 == width)
+ residue = fsi_get_fifo_residue(fsi, is_play);
+ else
+ residue = get_dma_residue(fsi->dma_chan);
+
+ return residue;
+}
+
+/************************************************************************
+
+
+ basic dma function
+
+
+************************************************************************/
+#define PORTA_DMA 0
+#define PORTB_DMA 1
+
+static int fsi_get_dma_chan(void)
+{
+ if (0 != request_dma(PORTA_DMA, "fsia"))
+ return -EIO;
+
+ if (0 != request_dma(PORTB_DMA, "fsib")) {
+ free_dma(PORTA_DMA);
+ return -EIO;
+ }
+
+ master->fsia.dma_chan = PORTA_DMA;
+ master->fsib.dma_chan = PORTB_DMA;
+
+ return 0;
+}
+
+static void fsi_free_dma_chan(void)
+{
+ dma_wait_for_completion(PORTA_DMA);
+ dma_wait_for_completion(PORTB_DMA);
+ free_dma(PORTA_DMA);
+ free_dma(PORTB_DMA);
+
+ master->fsia.dma_chan = -1;
+ master->fsib.dma_chan = -1;
+}
+
+/************************************************************************
+
+
+ ctrl function
+
+
+************************************************************************/
+static void fsi_irq_enable(struct fsi_priv *fsi, int is_play)
+{
+ u32 data = fsi_port_ab_io_bit(fsi, is_play);
+
+ fsi_master_mask_set(IMSK, data, data);
+ fsi_master_mask_set(IEMSK, data, data);
+}
+
+static void fsi_irq_disable(struct fsi_priv *fsi, int is_play)
+{
+ u32 data = fsi_port_ab_io_bit(fsi, is_play);
+
+ fsi_master_mask_set(IMSK, data, 0);
+ fsi_master_mask_set(IEMSK, data, 0);
+}
+
+static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable)
+{
+ u32 val = fsi_is_port_a(fsi) ? (1 << 0) : (1 << 4);
+
+ if (enable)
+ fsi_master_mask_set(CLK_RST, val, val);
+ else
+ fsi_master_mask_set(CLK_RST, val, 0);
+}
+
+static void fsi_irq_init(struct fsi_priv *fsi, int is_play)
+{
+ u32 data;
+ u32 ctrl;
+
+ data = fsi_port_ab_io_bit(fsi, is_play);
+ ctrl = is_play ? DOFF_CTL : DIFF_CTL;
+
+ /* set IMSK */
+ fsi_irq_disable(fsi, is_play);
+
+ /* set interrupt generation factor */
+ fsi_reg_write(fsi, ctrl, IRQ_HALF);
+
+ /* clear FIFO */
+ fsi_reg_mask_set(fsi, ctrl, FIFO_CLR, FIFO_CLR);
+
+ /* clear interrupt factor */
+ fsi_master_mask_set(INT_ST, data, 0);
+}
+
+static void fsi_soft_all_reset(void)
+{
+ u32 status = fsi_master_read(SOFT_RST);
+
+ /* port AB reset */
+ status &= 0x000000ff;
+ fsi_master_write(SOFT_RST, status);
+ mdelay(10);
+
+ /* soft reset */
+ status &= 0x000000f0;
+ fsi_master_write(SOFT_RST, status);
+ status |= 0x00000001;
+ fsi_master_write(SOFT_RST, status);
+ mdelay(10);
+}
+
+static void fsi_16data_push(struct fsi_priv *fsi,
+ struct snd_pcm_runtime *runtime,
+ int send)
+{
+ u16 *dma_start;
+ u32 snd;
+ int i;
+
+ /* get dma start position for FSI */
+ dma_start = (u16 *)runtime->dma_area;
+ dma_start += fsi->byte_offset / 2;
+
+ /*
+ * soft dma
+ * FSI can not use DMA when 16bpp
+ */
+ for (i = 0; i < send; i++) {
+ snd = (u32)dma_start[i];
+ fsi_reg_write(fsi, DODT, snd << 8);
+ }
+}
+
+static void fsi_32data_push(struct fsi_priv *fsi,
+ struct snd_pcm_runtime *runtime,
+ int send)
+{
+ u32 *dma_start;
+
+ /* get dma start position for FSI */
+ dma_start = (u32 *)runtime->dma_area;
+ dma_start += fsi->byte_offset / 4;
+
+ dma_wait_for_completion(fsi->dma_chan);
+ dma_configure_channel(fsi->dma_chan, (SM_INC|0x400|TS_32|TM_BUR));
+ dma_write(fsi->dma_chan, (u32)dma_start,
+ (u32)(fsi->base + DODT), send * 4);
+}
+
+/* playback interrupt */
+static int fsi_data_push(struct fsi_priv *fsi)
+{
+ struct snd_pcm_runtime *runtime;
+ struct snd_pcm_substream *substream = NULL;
+ int send;
+ int fifo_free;
+ int width;
+
+ if (!fsi ||
+ !fsi->substream ||
+ !fsi->substream->runtime)
+ return -EINVAL;
+
+ runtime = fsi->substream->runtime;
+
+ /* FSI FIFO has limit.
+ * So, this driver can not send periods data at a time
+ */
+ if (fsi->byte_offset >=
+ fsi->period_len * (fsi->periods + 1)) {
+
+ substream = fsi->substream;
+ fsi->periods = (fsi->periods + 1) % runtime->periods;
+
+ if (0 == fsi->periods)
+ fsi->byte_offset = 0;
+ }
+
+ /* get 1 channel data width */
+ width = frames_to_bytes(runtime, 1) / fsi->chan;
+
+ /* get send size for alsa */
+ send = (fsi->buffer_len - fsi->byte_offset) / width;
+
+ /* get FIFO free size */
+ fifo_free = (fsi->fifo_max * fsi->chan) - fsi_get_fifo_residue(fsi, 1);
+
+ /* size check */
+ if (fifo_free < send)
+ send = fifo_free;
+
+ if (2 == width)
+ fsi_16data_push(fsi, runtime, send);
+ else if (4 == width)
+ fsi_32data_push(fsi, runtime, send);
+ else
+ return -EINVAL;
+
+ fsi->byte_offset += send * width;
+
+ fsi_irq_enable(fsi, 1);
+
+ if (substream)
+ snd_pcm_period_elapsed(substream);
+
+ return 0;
+}
+
+static irqreturn_t fsi_interrupt(int irq, void *data)
+{
+ u32 status = fsi_master_read(SOFT_RST) & ~0x00000010;
+ u32 int_st = fsi_master_read(INT_ST);
+
+ /* clear irq status */
+ fsi_master_write(SOFT_RST, status);
+ fsi_master_write(SOFT_RST, status | 0x00000010);
+
+ if (int_st & INT_A_OUT)
+ fsi_data_push(&master->fsia);
+ if (int_st & INT_B_OUT)
+ fsi_data_push(&master->fsib);
+
+ fsi_master_write(INT_ST, 0x0000000);
+
+ return IRQ_HANDLED;
+}
+
+/************************************************************************
+
+
+ dai ops
+
+
+************************************************************************/
+static int fsi_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct fsi_priv *fsi = fsi_get(substream);
+ const char *msg;
+ u32 flags = fsi_get_info_flags(fsi);
+ u32 fmt;
+ u32 reg;
+ u32 data;
+ int is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+ int is_master;
+ int ret = 0;
+
+ clk_enable(master->clk);
+
+ /* CKG1 */
+ data = is_play ? (1 << 0) : (1 << 4);
+ is_master = fsi_is_master_mode(fsi, is_play);
+ if (is_master)
+ fsi_reg_mask_set(fsi, CKG1, data, data);
+ else
+ fsi_reg_mask_set(fsi, CKG1, data, 0);
+
+ /* clock inversion (CKG2) */
+ data = 0;
+ switch (SH_FSI_INVERSION_MASK & flags) {
+ case SH_FSI_LRM_INV:
+ data = 1 << 12;
+ break;
+ case SH_FSI_BRM_INV:
+ data = 1 << 8;
+ break;
+ case SH_FSI_LRS_INV:
+ data = 1 << 4;
+ break;
+ case SH_FSI_BRS_INV:
+ data = 1 << 0;
+ break;
+ }
+ fsi_reg_write(fsi, CKG2, data);
+
+ /* do fmt, di fmt */
+ data = 0;
+ reg = is_play ? DO_FMT : DI_FMT;
+ fmt = is_play ? SH_FSI_GET_OFMT(flags) : SH_FSI_GET_IFMT(flags);
+ switch (fmt) {
+ case SH_FSI_FMT_MONO:
+ msg = "MONO";
+ data = CR_FMT(CR_MONO);
+ fsi->chan = 1;
+ break;
+ case SH_FSI_FMT_MONO_DELAY:
+ msg = "MONO Delay";
+ data = CR_FMT(CR_MONO_D);
+ fsi->chan = 1;
+ break;
+ case SH_FSI_FMT_PCM:
+ msg = "PCM";
+ data = CR_FMT(CR_PCM);
+ fsi->chan = 2;
+ break;
+ case SH_FSI_FMT_I2S:
+ msg = "I2S";
+ data = CR_FMT(CR_I2S);
+ fsi->chan = 2;
+ break;
+ case SH_FSI_FMT_TDM:
+ msg = "TDM";
+ data = CR_FMT(CR_TDM) | (fsi->chan - 1);
+ fsi->chan = is_play ?
+ SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags);
+ break;
+ case SH_FSI_FMT_TDM_DELAY:
+ msg = "TDM Delay";
+ data = CR_FMT(CR_TDM_D) | (fsi->chan - 1);
+ fsi->chan = is_play ?
+ SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags);
+ break;
+ default:
+ dev_err(dai->dev, "unknown format.\n");
+ return -EINVAL;
+ }
+
+ switch (fsi->chan) {
+ case 1:
+ fsi->fifo_max = 256;
+ break;
+ case 2:
+ fsi->fifo_max = 128;
+ break;
+ case 3:
+ case 4:
+ fsi->fifo_max = 64;
+ break;
+ case 5:
+ case 6:
+ case 7:
+ case 8:
+ fsi->fifo_max = 32;
+ break;
+ default:
+ dev_err(dai->dev, "channel size error.\n");
+ return -EINVAL;
+ }
+
+ fsi_reg_write(fsi, reg, data);
+ dev_dbg(dai->dev, "use %s format (%d channel) use %d DMAC\n",
+ msg, fsi->chan, fsi->dma_chan);
+
+ /*
+ * clear clk reset if master mode
+ */
+ if (is_master)
+ fsi_clk_ctrl(fsi, 1);
+
+ /* irq setting */
+ fsi_irq_init(fsi, is_play);
+
+ return ret;
+}
+
+static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct fsi_priv *fsi = fsi_get(substream);
+ int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+
+ fsi_irq_disable(fsi, is_play);
+ fsi_clk_ctrl(fsi, 0);
+
+ clk_disable(master->clk);
+}
+
+static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct fsi_priv *fsi = fsi_get(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ int ret = 0;
+
+ /* capture not supported */
+ if (!is_play)
+ return -ENODEV;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ fsi_stream_push(fsi, substream,
+ frames_to_bytes(runtime, runtime->buffer_size),
+ frames_to_bytes(runtime, runtime->period_size));
+ ret = fsi_data_push(fsi);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ fsi_irq_disable(fsi, is_play);
+ fsi_stream_pop(fsi);
+ break;
+ }
+
+ return ret;
+}
+
+static struct snd_soc_dai_ops fsi_dai_ops = {
+ .startup = fsi_dai_startup,
+ .shutdown = fsi_dai_shutdown,
+ .trigger = fsi_dai_trigger,
+};
+
+/************************************************************************
+
+
+ pcm ops
+
+
+************************************************************************/
+static struct snd_pcm_hardware fsi_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE,
+ .formats = FSI_FMTS,
+ .rates = FSI_RATES,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 64 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8192,
+ .periods_min = 1,
+ .periods_max = 32,
+ .fifo_size = 256,
+};
+
+static int fsi_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ret = 0;
+
+ snd_soc_set_runtime_hwparams(substream, &fsi_pcm_hardware);
+
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+
+ return ret;
+}
+
+static int fsi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static int fsi_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsi_priv *fsi = fsi_get(substream);
+ int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ long location;
+
+ location = (fsi->byte_offset - 1) - fsi_get_residue(fsi, is_play);
+ if (location < 0)
+ location = 0;
+
+ return bytes_to_frames(runtime, location);
+}
+
+static struct snd_pcm_ops fsi_pcm_ops = {
+ .open = fsi_pcm_open,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = fsi_hw_params,
+ .hw_free = fsi_hw_free,
+ .pointer = fsi_pointer,
+};
+
+/************************************************************************
+
+
+ snd_soc_platform
+
+
+************************************************************************/
+#define PREALLOC_BUFFER (32 * 1024)
+#define PREALLOC_BUFFER_MAX (32 * 1024)
+
+static void fsi_pcm_free(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int fsi_pcm_new(struct snd_card *card,
+ struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ /*
+ * dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
+ * in MMAP mode (i.e. aplay -M)
+ */
+ return snd_pcm_lib_preallocate_pages_for_all(
+ pcm,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ PREALLOC_BUFFER, PREALLOC_BUFFER_MAX);
+}
+
+/************************************************************************
+
+
+ alsa struct
+
+
+************************************************************************/
+struct snd_soc_dai fsi_soc_dai[] = {
+ {
+ .name = "FSIA",
+ .id = 0,
+ .playback = {
+ .rates = FSI_RATES,
+ .formats = FSI_FMTS,
+ .channels_min = 1,
+ .channels_max = 8,
+ },
+ /* capture not supported */
+ .ops = &fsi_dai_ops,
+ },
+ {
+ .name = "FSIB",
+ .id = 1,
+ .playback = {
+ .rates = FSI_RATES,
+ .formats = FSI_FMTS,
+ .channels_min = 1,
+ .channels_max = 8,
+ },
+ /* capture not supported */
+ .ops = &fsi_dai_ops,
+ },
+};
+EXPORT_SYMBOL_GPL(fsi_soc_dai);
+
+struct snd_soc_platform fsi_soc_platform = {
+ .name = "fsi-pcm",
+ .pcm_ops = &fsi_pcm_ops,
+ .pcm_new = fsi_pcm_new,
+ .pcm_free = fsi_pcm_free,
+};
+EXPORT_SYMBOL_GPL(fsi_soc_platform);
+
+/************************************************************************
+
+
+ platform function
+
+
+************************************************************************/
+static int fsi_probe(struct platform_device *pdev)
+{
+ struct resource *res;
+ char clk_name[8];
+ unsigned int irq;
+ int ret;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ irq = platform_get_irq(pdev, 0);
+ if (!res || !irq) {
+ dev_err(&pdev->dev, "Not enough FSI platform resources.\n");
+ ret = -ENODEV;
+ goto exit;
+ }
+
+ master = kzalloc(sizeof(*master), GFP_KERNEL);
+ if (!master) {
+ dev_err(&pdev->dev, "Could not allocate master\n");
+ ret = -ENOMEM;
+ goto exit;
+ }
+
+ master->base = ioremap_nocache(res->start, resource_size(res));
+ if (!master->base) {
+ ret = -ENXIO;
+ dev_err(&pdev->dev, "Unable to ioremap FSI registers.\n");
+ goto exit_kfree;
+ }
+
+ master->irq = irq;
+ master->info = pdev->dev.platform_data;
+ master->fsia.base = master->base;
+ master->fsib.base = master->base + 0x40;
+
+ master->fsia.dma_chan = -1;
+ master->fsib.dma_chan = -1;
+
+ ret = fsi_get_dma_chan();
+ if (ret < 0) {
+ dev_err(&pdev->dev, "cannot get dma api\n");
+ goto exit_iounmap;
+ }
+
+ /* FSI is based on SPU mstp */
+ snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id);
+ master->clk = clk_get(NULL, clk_name);
+ if (IS_ERR(master->clk)) {
+ dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name);
+ ret = -EIO;
+ goto exit_free_dma;
+ }
+
+ fsi_soc_dai[0].dev = &pdev->dev;
+ fsi_soc_dai[1].dev = &pdev->dev;
+
+ fsi_soft_all_reset();
+
+ ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master);
+ if (ret) {
+ dev_err(&pdev->dev, "irq request err\n");
+ goto exit_free_dma;
+ }
+
+ ret = snd_soc_register_platform(&fsi_soc_platform);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "cannot snd soc register\n");
+ goto exit_free_irq;
+ }
+
+ return snd_soc_register_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai));
+
+exit_free_irq:
+ free_irq(irq, master);
+exit_free_dma:
+ fsi_free_dma_chan();
+exit_iounmap:
+ iounmap(master->base);
+exit_kfree:
+ kfree(master);
+ master = NULL;
+exit:
+ return ret;
+}
+
+static int fsi_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai));
+ snd_soc_unregister_platform(&fsi_soc_platform);
+
+ clk_put(master->clk);
+
+ fsi_free_dma_chan();
+
+ free_irq(master->irq, master);
+
+ iounmap(master->base);
+ kfree(master);
+ master = NULL;
+ return 0;
+}
+
+static struct platform_driver fsi_driver = {
+ .driver = {
+ .name = "sh_fsi",
+ },
+ .probe = fsi_probe,
+ .remove = fsi_remove,
+};
+
+static int __init fsi_mobile_init(void)
+{
+ return platform_driver_register(&fsi_driver);
+}
+
+static void __exit fsi_mobile_exit(void)
+{
+ platform_driver_unregister(&fsi_driver);
+}
+module_init(fsi_mobile_init);
+module_exit(fsi_mobile_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("SuperH onchip FSI audio driver");
+MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index e984a17c..7ff04ad 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1267,10 +1267,18 @@
if (!codec->debugfs_pop_time)
printk(KERN_WARNING
"Failed to create pop time debugfs file\n");
+
+ codec->debugfs_dapm = debugfs_create_dir("dapm", debugfs_root);
+ if (!codec->debugfs_dapm)
+ printk(KERN_WARNING
+ "Failed to create DAPM debugfs directory\n");
+
+ snd_soc_dapm_debugfs_init(codec);
}
static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
{
+ debugfs_remove_recursive(codec->debugfs_dapm);
debugfs_remove(codec->debugfs_pop_time);
debugfs_remove(codec->debugfs_reg);
}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index a225e5a..0d8b08e 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -37,6 +37,7 @@
#include <linux/bitops.h>
#include <linux/platform_device.h>
#include <linux/jiffies.h>
+#include <linux/debugfs.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -152,8 +153,12 @@
if (card->set_bias_level)
ret = card->set_bias_level(card, level);
- if (ret == 0 && codec->set_bias_level)
- ret = codec->set_bias_level(codec, level);
+ if (ret == 0) {
+ if (codec->set_bias_level)
+ ret = codec->set_bias_level(codec, level);
+ else
+ codec->bias_level = level;
+ }
return ret;
}
@@ -1097,6 +1102,92 @@
}
#endif
+#ifdef CONFIG_DEBUG_FS
+static int dapm_widget_power_open_file(struct inode *inode, struct file *file)
+{
+ file->private_data = inode->i_private;
+ return 0;
+}
+
+static ssize_t dapm_widget_power_read_file(struct file *file,
+ char __user *user_buf,
+ size_t count, loff_t *ppos)
+{
+ struct snd_soc_dapm_widget *w = file->private_data;
+ char *buf;
+ int in, out;
+ ssize_t ret;
+ struct snd_soc_dapm_path *p = NULL;
+
+ buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+
+ in = is_connected_input_ep(w);
+ dapm_clear_walk(w->codec);
+ out = is_connected_output_ep(w);
+ dapm_clear_walk(w->codec);
+
+ ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d\n",
+ w->name, w->power ? "On" : "Off", in, out);
+
+ if (w->active && w->sname)
+ ret += snprintf(buf, PAGE_SIZE - ret, " stream %s active\n",
+ w->sname);
+
+ list_for_each_entry(p, &w->sources, list_sink) {
+ if (p->connect)
+ ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ " in %s %s\n",
+ p->name ? p->name : "static",
+ p->source->name);
+ }
+ list_for_each_entry(p, &w->sinks, list_source) {
+ if (p->connect)
+ ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ " out %s %s\n",
+ p->name ? p->name : "static",
+ p->sink->name);
+ }
+
+ ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
+
+ kfree(buf);
+ return ret;
+}
+
+static const struct file_operations dapm_widget_power_fops = {
+ .open = dapm_widget_power_open_file,
+ .read = dapm_widget_power_read_file,
+};
+
+void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_widget *w;
+ struct dentry *d;
+
+ if (!codec->debugfs_dapm)
+ return;
+
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+ if (!w->name)
+ continue;
+
+ d = debugfs_create_file(w->name, 0444,
+ codec->debugfs_dapm, w,
+ &dapm_widget_power_fops);
+ if (!d)
+ printk(KERN_WARNING
+ "ASoC: Failed to create %s debugfs file\n",
+ w->name);
+ }
+}
+#else
+void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec)
+{
+}
+#endif
+
/* test and update the power status of a mux widget */
static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
struct snd_kcontrol *kcontrol, int mask,