.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,
.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,
.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,
.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,
.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)
void omap_dma_link_lch(int lch_head, int lch_queue)
{
if (omap_dma_in_1510_mode()) {
+ if (lch_head == lch_queue) {
+ dma_write(dma_read(CCR(lch_head)) | (3 << 8),
+ CCR(lch_head));
+ return;
+ }
printk(KERN_ERR "DMA linking is not supported in 1510 mode\n");
BUG();
return;
void omap_dma_unlink_lch(int lch_head, int lch_queue)
{
if (omap_dma_in_1510_mode()) {
+ if (lch_head == lch_queue) {
+ dma_write(dma_read(CCR(lch_head)) & ~(3 << 8),
+ CCR(lch_head));
+ return;
+ }
printk(KERN_ERR "DMA linking is not supported in 1510 mode\n");
BUG();
return;
#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
#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;
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 {
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;
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);
}
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().
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.
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);
}
/* Enable transmitter and receiver */
+ tx &= 1;
w = OMAP_MCBSP_READ(io_base, SPCR2);
- OMAP_MCBSP_WRITE(io_base, SPCR2, w | (tx & 1));
+ OMAP_MCBSP_WRITE(io_base, SPCR2, w | tx);
+ rx &= 1;
w = OMAP_MCBSP_READ(io_base, SPCR1);
- OMAP_MCBSP_WRITE(io_base, SPCR1, w | (rx & 1));
+ OMAP_MCBSP_WRITE(io_base, SPCR1, w | rx);
- 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 */
OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 7));
}
+ if (cpu_is_omap2430() || cpu_is_omap34xx()) {
+ /* Release the transmitter and receiver */
+ w = OMAP_MCBSP_READ(io_base, XCCR);
+ w &= ~(tx ? XDISABLE : 0);
+ OMAP_MCBSP_WRITE(io_base, XCCR, w);
+ w = OMAP_MCBSP_READ(io_base, RCCR);
+ w &= ~(rx ? RDISABLE : 0);
+ OMAP_MCBSP_WRITE(io_base, RCCR, w);
+ }
+
/* Dump McBSP Regs */
omap_mcbsp_dump_reg(id);
}
io_base = mcbsp->io_base;
/* Reset transmitter */
+ tx &= 1;
+ if (cpu_is_omap2430() || cpu_is_omap34xx()) {
+ w = OMAP_MCBSP_READ(io_base, XCCR);
+ w |= (tx ? XDISABLE : 0);
+ OMAP_MCBSP_WRITE(io_base, XCCR, w);
+ }
w = OMAP_MCBSP_READ(io_base, SPCR2);
- OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(tx & 1));
+ OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~tx);
/* Reset receiver */
+ rx &= 1;
+ if (cpu_is_omap2430() || cpu_is_omap34xx()) {
+ w = OMAP_MCBSP_READ(io_base, RCCR);
+ w |= (tx ? RDISABLE : 0);
+ OMAP_MCBSP_WRITE(io_base, RCCR, w);
+ }
w = OMAP_MCBSP_READ(io_base, SPCR1);
- OMAP_MCBSP_WRITE(io_base, SPCR1, w & ~(rx & 1));
+ OMAP_MCBSP_WRITE(io_base, SPCR1, w & ~rx);
idle = !((OMAP_MCBSP_READ(io_base, SPCR2) |
OMAP_MCBSP_READ(io_base, SPCR1)) & 1);
}
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 const char *dma_op_modes[] = {
+ "element", "threshold", "frame",
+};
+
+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, i = 0;
+ ssize_t len = 0;
+ const char * const *s;
+
+ spin_lock_irq(&mcbsp->lock);
+ dma_op_mode = mcbsp->dma_op_mode;
+ spin_unlock_irq(&mcbsp->lock);
+
+ for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) {
+ if (dma_op_mode == i)
+ len += sprintf(buf + len, "[%s] ", *s);
+ else
+ len += sprintf(buf + len, "%s ", *s);
+ }
+ len += sprintf(buf + len, "\n");
+
+ return len;
+}
+
+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);
+ const char * const *s;
+ int i = 0;
+
+ for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++)
+ if (sysfs_streq(buf, *s))
+ break;
+
+ if (i == ARRAY_SIZE(dma_op_modes))
+ return -EINVAL;
+
+ spin_lock_irq(&mcbsp->lock);
+ if (!mcbsp->free) {
+ size = -EBUSY;
+ goto unlock;
+ }
+ mcbsp->dma_op_mode = i;
+
+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.
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:
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);
--- /dev/null
+/* 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);
#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)
#define S3C2412_IISMOD_BCLK_MASK (3 << 1)
#define S3C2412_IISMOD_8BIT (1 << 0)
+#define S3C64XX_IISMOD_CDCLKCON (1 << 12)
+
#define S3C2412_IISPSR_PSREN (1 << 15)
#define S3C2412_IISFIC_TXFLUSH (1 << 15)
--- /dev/null
+#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 */
int div_id, int div);
int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out);
+ int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
/* Digital Audio interface formatting */
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
+int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot);
+
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
/* Digital Audio Interface mute */
*/
int (*set_sysclk)(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir);
- int (*set_pll)(struct snd_soc_dai *dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out);
+ int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
+ unsigned int freq_in, unsigned int freq_out);
int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
/*
int (*set_tdm_slot)(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask,
int slots, int slot_width);
+ int (*set_channel_map)(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot);
int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
/*
.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD}
/* stream domain */
+#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \
+{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \
+ .reg = wreg, .shift = wshift, .invert = winvert }
+#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \
+{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \
+ .reg = wreg, .shift = wshift, .invert = winvert }
#define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert}
/* 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);
snd_soc_dapm_pre, /* machine specific pre widget - exec first */
snd_soc_dapm_post, /* machine specific post widget - exec last */
snd_soc_dapm_supply, /* power/clock supply */
+ snd_soc_dapm_aif_in, /* audio interface input */
+ snd_soc_dapm_aif_out, /* audio interface output */
};
/*
const char *sink;
const char *control;
const char *source;
+
+ /* Note: currently only supported for links where source is a supply */
+ int (*connected)(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink);
};
/* dapm audio path between two widgets */
u32 connect:1; /* source and sink widgets are connected */
u32 walked:1; /* path has been walked */
+ int (*connected)(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink);
+
struct list_head list_source;
struct list_head list_sink;
struct list_head list;
#endif
typedef int (*hw_write_t)(void *,const char* ,int);
-typedef int (*hw_read_t)(void *,char* ,int);
extern struct snd_ac97_bus_ops soc_ac97_ops;
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_reg;
struct dentry *debugfs_pop_time;
+ struct dentry *debugfs_dapm;
#endif
};
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_pdata[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) {
{
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;
#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
- ret = snd_soc_dai_set_pll(codec_dai, 0,
+ ret = snd_soc_dai_set_pll(codec_dai, 0, 0,
clk_get_rate(CODEC_CLK), pll_out);
if (ret < 0) {
pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n",
/*
* Au12x0/Au1550 PSC ALSA ASoC audio support.
*
- * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
- * Manuel Lauss <mano@roarinelk.homelinux.net>
+ * (c) 2007-2009 MSC Vertriebsges.m.b.H.,
+ * Manuel Lauss <manuel.lauss@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
#include <linux/module.h>
#include <linux/device.h>
#include <linux/delay.h>
+#include <linux/mutex.h>
#include <linux/suspend.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include "psc.h"
+/* how often to retry failed codec register reads/writes */
+#define AC97_RW_RETRIES 5
+
#define AC97_DIR \
(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
#define AC97PCR_CLRFIFO(stype) \
((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
+#define AC97STAT_BUSY(stype) \
+ ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
+
/* instance data. There can be only one, MacLeod!!!! */
static struct au1xpsc_audio_data *au1xpsc_ac97_workdata;
{
/* FIXME */
struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
- unsigned short data, tmo;
+ unsigned short data, retry, tmo;
- au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata));
+ au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
au_sync();
- tmo = 1000;
- while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo)
- udelay(2);
+ retry = AC97_RW_RETRIES;
+ do {
+ mutex_lock(&pscdata->lock);
+
+ au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg),
+ AC97_CDC(pscdata));
+ au_sync();
+
+ tmo = 2000;
+ while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD))
+ && --tmo)
+ udelay(2);
- if (!tmo)
- data = 0xffff;
- else
data = au_readl(AC97_CDC(pscdata)) & 0xffff;
- au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
- au_sync();
+ au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+ au_sync();
+
+ mutex_unlock(&pscdata->lock);
+ } while (--retry && !tmo);
- return data;
+ return retry ? data : 0xffff;
}
/* AC97 controller writes to codec register */
{
/* FIXME */
struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
- unsigned int tmo;
+ unsigned int tmo, retry;
- au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata));
+ au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
au_sync();
- tmo = 1000;
- while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo)
+
+ retry = AC97_RW_RETRIES;
+ do {
+ mutex_lock(&pscdata->lock);
+
+ au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff),
+ AC97_CDC(pscdata));
au_sync();
- au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
- au_sync();
+ tmo = 2000;
+ while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD))
+ && --tmo)
+ udelay(2);
+
+ au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+ au_sync();
+
+ mutex_unlock(&pscdata->lock);
+ } while (--retry && !tmo);
}
/* AC97 controller asserts a warm reset */
au_sync();
/* wait for PSC to indicate it's ready */
- i = 100000;
+ i = 1000;
while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i))
- au_sync();
+ msleep(1);
if (i == 0) {
printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n");
au_sync();
/* wait for AC97 core to become ready */
- i = 100000;
+ i = 1000;
while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i))
- au_sync();
+ msleep(1);
if (i == 0)
printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n");
}
{
/* FIXME */
struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
- unsigned long r, stat;
+ unsigned long r, ro, stat;
int chans, stype = SUBSTREAM_TYPE(substream);
chans = params_channels(params);
- r = au_readl(AC97_CFG(pscdata));
+ r = ro = au_readl(AC97_CFG(pscdata));
stat = au_readl(AC97_STAT(pscdata));
/* already active? */
(pscdata->rate != params_rate(params)))
return -EINVAL;
} else {
- /* disable AC97 device controller first */
- au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
- au_sync();
/* set sample bitdepth: REG[24:21]=(BITS-2)/2 */
r &= ~PSC_AC97CFG_LEN_MASK;
r |= PSC_AC97CFG_RXSLOT_ENA(4);
}
- /* finally enable the AC97 controller again */
+ /* do we need to poke the hardware? */
+ if (!(r ^ ro))
+ goto out;
+
+ /* ac97 engine is about to be disabled */
+ mutex_lock(&pscdata->lock);
+
+ /* disable AC97 device controller first... */
+ au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+ au_sync();
+
+ /* ...wait for it... */
+ while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)
+ asm volatile ("nop");
+
+ /* ...write config... */
+ au_writel(r, AC97_CFG(pscdata));
+ au_sync();
+
+ /* ...enable the AC97 controller again... */
au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
au_sync();
+ /* ...and wait for ready bit */
+ while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR))
+ asm volatile ("nop");
+
+ mutex_unlock(&pscdata->lock);
+
pscdata->cfg = r;
pscdata->rate = params_rate(params);
}
+out:
return 0;
}
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
+ au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata));
+ au_sync();
au_writel(AC97PCR_START(stype), AC97_PCR(pscdata));
au_sync();
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata));
au_sync();
+
+ while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype))
+ asm volatile ("nop");
+
+ au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata));
+ au_sync();
+
break;
default:
ret = -EINVAL;
if (!au1xpsc_ac97_workdata)
return -ENOMEM;
+ mutex_init(&au1xpsc_ac97_workdata->lock);
+
r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!r) {
ret = -ENODEV;
goto out1;
/* configuration: max dma trigger threshold, enable ac97 */
- au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 |
- PSC_AC97CFG_TT_FIFO8 |
- PSC_AC97CFG_DE_ENABLE;
+ au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 |
+ PSC_AC97CFG_TT_FIFO8 |
+ PSC_AC97CFG_DE_ENABLE;
/* preserve PSC clock source set up by platform (dev.platform_data
* is already occupied by soc layer)
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver");
-MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
+MODULE_AUTHOR("Manuel Lauss <manuel.lauss@gmail.com>");
unsigned long pm[2];
struct resource *ioarea;
+ struct mutex lock;
};
#define PCM_TX 0
select SND_SOC_AC97_BUS
select SND_BF5XX_SOC_SPORT
+config SND_BF5XX_SOC_AD1836
+ tristate "SoC AD1836 Audio support for BF5xx"
+ depends on SND_BF5XX_TDM
+ select SND_BF5XX_SOC_TDM
+ select SND_SOC_AD1836
+ help
+ Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
config SND_BF5XX_SOC_AD1980
tristate "SoC AD1980/1 Audio support for BF5xx"
depends on SND_BF5XX_AC97
obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o
# Blackfin Machine Support
+snd-ad1836-objs := bf5xx-ad1836.o
snd-ad1980-objs := bf5xx-ad1980.o
snd-ssm2602-objs := bf5xx-ssm2602.o
snd-ad73311-objs := bf5xx-ad73311.o
snd-ad1938-objs := bf5xx-ad1938.o
+obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o
obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
if (!dai->active)
return 0;
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+ ret = sport_set_multichannel(sport, 16, 0x3FF, 1);
+#else
ret = sport_set_multichannel(sport, 16, 0x1F, 1);
+#endif
if (ret) {
pr_err("SPORT is busy!\n");
return -EBUSY;
goto sport_err;
}
/*SPORT works in TDM mode to simulate AC97 transfers*/
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+ ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 1);
+#else
ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1);
+#endif
if (ret) {
pr_err("SPORT is busy!\n");
ret = -EBUSY;
/*
- * linux/sound/arm/bf5xx-ac97.h
+ * sound/soc/blackfin/bf5xx-ac97.h
*
* 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
--- /dev/null
+/*
+ * File: sound/soc/blackfin/bf5xx-ad1836.c
+ * Author: Barry Song <Barry.Song@analog.com>
+ *
+ * Created: Aug 4 2009
+ * Description: Board driver for ad1836 sound chip
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm_params.h>
+
+#include <asm/blackfin.h>
+#include <asm/cacheflush.h>
+#include <asm/irq.h>
+#include <asm/dma.h>
+#include <asm/portmux.h>
+
+#include "../codecs/ad1836.h"
+#include "bf5xx-sport.h"
+
+#include "bf5xx-tdm-pcm.h"
+#include "bf5xx-tdm.h"
+
+static struct snd_soc_card bf5xx_ad1836;
+
+static int bf5xx_ad1836_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ cpu_dai->private_data = sport_handle;
+ return 0;
+}
+
+static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7};
+ int ret = 0;
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI channel mapping */
+ ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map),
+ channel_map, ARRAY_SIZE(channel_map), channel_map);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops bf5xx_ad1836_ops = {
+ .startup = bf5xx_ad1836_startup,
+ .hw_params = bf5xx_ad1836_hw_params,
+};
+
+static struct snd_soc_dai_link bf5xx_ad1836_dai = {
+ .name = "ad1836",
+ .stream_name = "AD1836",
+ .cpu_dai = &bf5xx_tdm_dai,
+ .codec_dai = &ad1836_dai,
+ .ops = &bf5xx_ad1836_ops,
+};
+
+static struct snd_soc_card bf5xx_ad1836 = {
+ .name = "bf5xx_ad1836",
+ .platform = &bf5xx_tdm_soc_platform,
+ .dai_link = &bf5xx_ad1836_dai,
+ .num_links = 1,
+};
+
+static struct snd_soc_device bf5xx_ad1836_snd_devdata = {
+ .card = &bf5xx_ad1836,
+ .codec_dev = &soc_codec_dev_ad1836,
+};
+
+static struct platform_device *bfxx_ad1836_snd_device;
+
+static int __init bf5xx_ad1836_init(void)
+{
+ int ret;
+
+ bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!bfxx_ad1836_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836_snd_devdata);
+ bf5xx_ad1836_snd_devdata.dev = &bfxx_ad1836_snd_device->dev;
+ ret = platform_device_add(bfxx_ad1836_snd_device);
+
+ if (ret)
+ platform_device_put(bfxx_ad1836_snd_device);
+
+ return ret;
+}
+
+static void __exit bf5xx_ad1836_exit(void)
+{
+ platform_device_unregister(bfxx_ad1836_snd_device);
+}
+
+module_init(bf5xx_ad1836_init);
+module_exit(bf5xx_ad1836_exit);
+
+/* Module information */
+MODULE_AUTHOR("Barry Song");
+MODULE_DESCRIPTION("ALSA SoC AD1836 board driver");
+MODULE_LICENSE("GPL");
+
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7};
int ret = 0;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
return ret;
/* set codec DAI slots, 8 channels, all channels are enabled */
- ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8);
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI channel mapping */
+ ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map),
+ channel_map, ARRAY_SIZE(channel_map), channel_map);
if (ret < 0)
return ret;
return 0;
}
-static void bf5xx_i2s_remove(struct snd_soc_dai *dai)
+static void bf5xx_i2s_remove(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
{
pr_debug("%s enter\n", __func__);
peripheral_free_list(&sport_req[sport_num][0]);
#ifdef CONFIG_PM
static int bf5xx_i2s_suspend(struct snd_soc_dai *dai)
{
- struct sport_device *sport =
- (struct sport_device *)dai->private_data;
pr_debug("%s : sport %d\n", __func__, dai->id);
- if (!dai->active)
- return 0;
+
if (dai->capture.active)
- sport_rx_stop(sport);
+ sport_rx_stop(sport_handle);
if (dai->playback.active)
- sport_tx_stop(sport);
+ sport_tx_stop(sport_handle);
return 0;
}
static int bf5xx_i2s_resume(struct snd_soc_dai *dai)
{
int ret;
- struct sport_device *sport =
- (struct sport_device *)dai->private_data;
pr_debug("%s : sport %d\n", __func__, dai->id);
- if (!dai->active)
- return 0;
- ret = sport_config_rx(sport, RFSR | RCKFE, RSFSE|0x1f, 0, 0);
+ ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1,
+ bf5xx_i2s.rcr2, 0, 0);
if (ret) {
pr_err("SPORT is busy!\n");
return -EBUSY;
}
- ret = sport_config_tx(sport, TFSR | TCKFE, TSFSE|0x1f, 0, 0);
+ ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1,
+ bf5xx_i2s.tcr2, 0, 0);
if (ret) {
pr_err("SPORT is busy!\n");
return -EBUSY;
/*
- * linux/sound/arm/bf5xx-i2s.h
+ * sound/soc/blackfin/bf5xx-i2s.h
*
* 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
int sport_tx_start(struct sport_device *sport)
{
- unsigned flags;
+ unsigned long flags;
pr_debug("%s: tx_run:%d, rx_run:%d\n", __func__,
sport->tx_run, sport->rx_run);
if (sport->tx_run)
#include "bf5xx-tdm.h"
#include "bf5xx-sport.h"
-#define PCM_BUFFER_MAX 0x10000
+#define PCM_BUFFER_MAX 0x8000
#define FRAGMENT_SIZE_MIN (4*1024)
#define FRAGMENTS_MIN 2
#define FRAGMENTS_MAX 32
static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count)
{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
+ struct bf5xx_tdm_port *tdm_port = sport->private_data;
unsigned int *src;
unsigned int *dst;
int i;
dst += pos * 8;
while (count--) {
for (i = 0; i < substream->runtime->channels; i++)
- *(dst + i) = *src++;
+ *(dst + tdm_port->tx_map[i]) = *src++;
dst += 8;
}
} else {
src += pos * 8;
while (count--) {
for (i = 0; i < substream->runtime->channels; i++)
- *dst++ = *(src+i);
+ *dst++ = *(src + tdm_port->rx_map[i]);
src += 8;
}
}
#include "bf5xx-sport.h"
#include "bf5xx-tdm.h"
-struct bf5xx_tdm_port {
- u16 tcr1;
- u16 rcr1;
- u16 tcr2;
- u16 rcr2;
- int configured;
-};
-
static struct bf5xx_tdm_port bf5xx_tdm;
static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
bf5xx_tdm.configured = 0;
}
+static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot)
+{
+ int i;
+ unsigned int slot;
+ unsigned int tx_mapped = 0, rx_mapped = 0;
+
+ if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) ||
+ (rx_num > BFIN_TDM_DAI_MAX_SLOTS))
+ return -EINVAL;
+
+ for (i = 0; i < tx_num; i++) {
+ slot = tx_slot[i];
+ if ((slot < BFIN_TDM_DAI_MAX_SLOTS) &&
+ (!(tx_mapped & (1 << slot)))) {
+ bf5xx_tdm.tx_map[i] = slot;
+ tx_mapped |= 1 << slot;
+ } else
+ return -EINVAL;
+ }
+ for (i = 0; i < rx_num; i++) {
+ slot = rx_slot[i];
+ if ((slot < BFIN_TDM_DAI_MAX_SLOTS) &&
+ (!(rx_mapped & (1 << slot)))) {
+ bf5xx_tdm.rx_map[i] = slot;
+ rx_mapped |= 1 << slot;
+ } else
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
#ifdef CONFIG_PM
static int bf5xx_tdm_suspend(struct snd_soc_dai *dai)
{
.hw_params = bf5xx_tdm_hw_params,
.set_fmt = bf5xx_tdm_set_dai_fmt,
.shutdown = bf5xx_tdm_shutdown,
+ .set_channel_map = bf5xx_tdm_set_channel_map,
};
struct snd_soc_dai bf5xx_tdm_dai = {
pr_err("Failed to register DAI: %d\n", ret);
goto sport_config_err;
}
+
+ sport_handle->private_data = &bf5xx_tdm;
return 0;
sport_config_err:
#ifndef _BF5XX_TDM_H
#define _BF5XX_TDM_H
+#define BFIN_TDM_DAI_MAX_SLOTS 8
+struct bf5xx_tdm_port {
+ u16 tcr1;
+ u16 rcr1;
+ u16 tcr2;
+ u16 rcr2;
+ unsigned int tx_map[BFIN_TDM_DAI_MAX_SLOTS];
+ unsigned int rx_map[BFIN_TDM_DAI_MAX_SLOTS];
+ int configured;
+};
+
extern struct snd_soc_dai bf5xx_tdm_dai;
#endif
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_AK4671 if I2C
select SND_SOC_CS4270 if I2C
select SND_SOC_MAX9877 if I2C
select SND_SOC_PCM3008
config SND_SOC_AK4535
tristate
+config SND_SOC_AK4642
+ tristate
+
+config SND_SOC_AK4671
+ tristate
+
# Cirrus Logic CS4270 Codec
config SND_SOC_CS4270
tristate
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-ak4671-objs := ak4671.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-cx20442-objs := cx20442.o
snd-soc-l3-objs := l3.o
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_AK4671) += snd-soc-ak4671.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
#include <linux/init.h>
#include <linux/module.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <sound/core.h>
#include <linux/init.h>
#include <linux/module.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <sound/core.h>
--- /dev/null
+/*
+ * 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");
--- /dev/null
+/*
+ * 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
--- /dev/null
+/*
+ * ak4671.c -- audio driver for AK4671
+ *
+ * Copyright (C) 2009 Samsung Electronics Co.Ltd
+ * Author: Joonyoung Shim <jy0922.shim@samsung.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/delay.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "ak4671.h"
+
+static struct snd_soc_codec *ak4671_codec;
+
+/* codec private data */
+struct ak4671_priv {
+ struct snd_soc_codec codec;
+ u8 reg_cache[AK4671_CACHEREGNUM];
+};
+
+/* ak4671 register cache & default register settings */
+static const u8 ak4671_reg[AK4671_CACHEREGNUM] = {
+ 0x00, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */
+ 0xf6, /* AK4671_PLL_MODE_SELECT0 (0x01) */
+ 0x00, /* AK4671_PLL_MODE_SELECT1 (0x02) */
+ 0x02, /* AK4671_FORMAT_SELECT (0x03) */
+ 0x00, /* AK4671_MIC_SIGNAL_SELECT (0x04) */
+ 0x55, /* AK4671_MIC_AMP_GAIN (0x05) */
+ 0x00, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */
+ 0x00, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */
+ 0xb5, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */
+ 0x00, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */
+ 0x00, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */
+ 0x00, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */
+ 0x00, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */
+ 0x00, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */
+ 0x00, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */
+ 0x00, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */
+ 0x00, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */
+ 0x80, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */
+ 0x91, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */
+ 0x91, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */
+ 0xe1, /* AK4671_ALC_REFERENCE_SELECT (0x14) */
+ 0x00, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */
+ 0x00, /* AK4671_ALC_TIMER_SELECT (0x16) */
+ 0x00, /* AK4671_ALC_MODE_CONTROL (0x17) */
+ 0x02, /* AK4671_MODE_CONTROL1 (0x18) */
+ 0x01, /* AK4671_MODE_CONTROL2 (0x19) */
+ 0x18, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */
+ 0x18, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */
+ 0x00, /* AK4671_SIDETONE_A_CONTROL (0x1c) */
+ 0x02, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */
+ 0x00, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */
+ 0x00, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */
+ 0x00, /* AK4671_FIL3_COEFFICIENT2 (0x20) */
+ 0x00, /* AK4671_FIL3_COEFFICIENT3 (0x21) */
+ 0x00, /* AK4671_EQ_COEFFICIENT0 (0x22) */
+ 0x00, /* AK4671_EQ_COEFFICIENT1 (0x23) */
+ 0x00, /* AK4671_EQ_COEFFICIENT2 (0x24) */
+ 0x00, /* AK4671_EQ_COEFFICIENT3 (0x25) */
+ 0x00, /* AK4671_EQ_COEFFICIENT4 (0x26) */
+ 0x00, /* AK4671_EQ_COEFFICIENT5 (0x27) */
+ 0xa9, /* AK4671_FIL1_COEFFICIENT0 (0x28) */
+ 0x1f, /* AK4671_FIL1_COEFFICIENT1 (0x29) */
+ 0xad, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */
+ 0x20, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */
+ 0x00, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */
+ 0x00, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */
+ 0x00, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */
+ 0x00, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */
+ 0x00, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */
+ 0x00, /* this register not used */
+ 0x00, /* AK4671_E1_COEFFICIENT0 (0x32) */
+ 0x00, /* AK4671_E1_COEFFICIENT1 (0x33) */
+ 0x00, /* AK4671_E1_COEFFICIENT2 (0x34) */
+ 0x00, /* AK4671_E1_COEFFICIENT3 (0x35) */
+ 0x00, /* AK4671_E1_COEFFICIENT4 (0x36) */
+ 0x00, /* AK4671_E1_COEFFICIENT5 (0x37) */
+ 0x00, /* AK4671_E2_COEFFICIENT0 (0x38) */
+ 0x00, /* AK4671_E2_COEFFICIENT1 (0x39) */
+ 0x00, /* AK4671_E2_COEFFICIENT2 (0x3a) */
+ 0x00, /* AK4671_E2_COEFFICIENT3 (0x3b) */
+ 0x00, /* AK4671_E2_COEFFICIENT4 (0x3c) */
+ 0x00, /* AK4671_E2_COEFFICIENT5 (0x3d) */
+ 0x00, /* AK4671_E3_COEFFICIENT0 (0x3e) */
+ 0x00, /* AK4671_E3_COEFFICIENT1 (0x3f) */
+ 0x00, /* AK4671_E3_COEFFICIENT2 (0x40) */
+ 0x00, /* AK4671_E3_COEFFICIENT3 (0x41) */
+ 0x00, /* AK4671_E3_COEFFICIENT4 (0x42) */
+ 0x00, /* AK4671_E3_COEFFICIENT5 (0x43) */
+ 0x00, /* AK4671_E4_COEFFICIENT0 (0x44) */
+ 0x00, /* AK4671_E4_COEFFICIENT1 (0x45) */
+ 0x00, /* AK4671_E4_COEFFICIENT2 (0x46) */
+ 0x00, /* AK4671_E4_COEFFICIENT3 (0x47) */
+ 0x00, /* AK4671_E4_COEFFICIENT4 (0x48) */
+ 0x00, /* AK4671_E4_COEFFICIENT5 (0x49) */
+ 0x00, /* AK4671_E5_COEFFICIENT0 (0x4a) */
+ 0x00, /* AK4671_E5_COEFFICIENT1 (0x4b) */
+ 0x00, /* AK4671_E5_COEFFICIENT2 (0x4c) */
+ 0x00, /* AK4671_E5_COEFFICIENT3 (0x4d) */
+ 0x00, /* AK4671_E5_COEFFICIENT4 (0x4e) */
+ 0x00, /* AK4671_E5_COEFFICIENT5 (0x4f) */
+ 0x88, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */
+ 0x88, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */
+ 0x08, /* AK4671_EQ_CONTRO_10KHZ (0x52) */
+ 0x00, /* AK4671_PCM_IF_CONTROL0 (0x53) */
+ 0x00, /* AK4671_PCM_IF_CONTROL1 (0x54) */
+ 0x00, /* AK4671_PCM_IF_CONTROL2 (0x55) */
+ 0x18, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */
+ 0x18, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */
+ 0x00, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */
+ 0x00, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */
+ 0x00, /* AK4671_SAR_ADC_CONTROL (0x5a) */
+};
+
+/*
+ * LOUT1/ROUT1 output volume control:
+ * from -24 to 6 dB in 6 dB steps (mute instead of -30 dB)
+ */
+static DECLARE_TLV_DB_SCALE(out1_tlv, -3000, 600, 1);
+
+/*
+ * LOUT2/ROUT2 output volume control:
+ * from -33 to 6 dB in 3 dB steps (mute instead of -33 dB)
+ */
+static DECLARE_TLV_DB_SCALE(out2_tlv, -3300, 300, 1);
+
+/*
+ * LOUT3/ROUT3 output volume control:
+ * from -6 to 3 dB in 3 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(out3_tlv, -600, 300, 0);
+
+/*
+ * Mic amp gain control:
+ * from -15 to 30 dB in 3 dB steps
+ * REVISIT: The actual min value(0x01) is -12 dB and the reg value 0x00 is not
+ * available
+ */
+static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -1500, 300, 0);
+
+static const struct snd_kcontrol_new ak4671_snd_controls[] = {
+ /* Common playback gain controls */
+ SOC_SINGLE_TLV("Line Output1 Playback Volume",
+ AK4671_OUTPUT_VOLUME_CONTROL, 0, 0x6, 0, out1_tlv),
+ SOC_SINGLE_TLV("Headphone Output2 Playback Volume",
+ AK4671_OUTPUT_VOLUME_CONTROL, 4, 0xd, 0, out2_tlv),
+ SOC_SINGLE_TLV("Line Output3 Playback Volume",
+ AK4671_LOUT3_POWER_MANAGERMENT, 6, 0x3, 0, out3_tlv),
+
+ /* Common capture gain controls */
+ SOC_DOUBLE_TLV("Mic Amp Capture Volume",
+ AK4671_MIC_AMP_GAIN, 0, 4, 0xf, 0, mic_amp_tlv),
+};
+
+/* event handlers */
+static int ak4671_out2_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u8 reg;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT);
+ reg |= AK4671_MUTEN;
+ snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT);
+ reg &= ~AK4671_MUTEN;
+ snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg);
+ break;
+ }
+
+ return 0;
+}
+
+/* Output Mixers */
+static const struct snd_kcontrol_new ak4671_lout1_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DACL", AK4671_LOUT1_SIGNAL_SELECT, 0, 1, 0),
+ SOC_DAPM_SINGLE("LINL1", AK4671_LOUT1_SIGNAL_SELECT, 1, 1, 0),
+ SOC_DAPM_SINGLE("LINL2", AK4671_LOUT1_SIGNAL_SELECT, 2, 1, 0),
+ SOC_DAPM_SINGLE("LINL3", AK4671_LOUT1_SIGNAL_SELECT, 3, 1, 0),
+ SOC_DAPM_SINGLE("LINL4", AK4671_LOUT1_SIGNAL_SELECT, 4, 1, 0),
+ SOC_DAPM_SINGLE("LOOPL", AK4671_LOUT1_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_rout1_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DACR", AK4671_ROUT1_SIGNAL_SELECT, 0, 1, 0),
+ SOC_DAPM_SINGLE("RINR1", AK4671_ROUT1_SIGNAL_SELECT, 1, 1, 0),
+ SOC_DAPM_SINGLE("RINR2", AK4671_ROUT1_SIGNAL_SELECT, 2, 1, 0),
+ SOC_DAPM_SINGLE("RINR3", AK4671_ROUT1_SIGNAL_SELECT, 3, 1, 0),
+ SOC_DAPM_SINGLE("RINR4", AK4671_ROUT1_SIGNAL_SELECT, 4, 1, 0),
+ SOC_DAPM_SINGLE("LOOPR", AK4671_ROUT1_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_lout2_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DACHL", AK4671_LOUT2_SIGNAL_SELECT, 0, 1, 0),
+ SOC_DAPM_SINGLE("LINH1", AK4671_LOUT2_SIGNAL_SELECT, 1, 1, 0),
+ SOC_DAPM_SINGLE("LINH2", AK4671_LOUT2_SIGNAL_SELECT, 2, 1, 0),
+ SOC_DAPM_SINGLE("LINH3", AK4671_LOUT2_SIGNAL_SELECT, 3, 1, 0),
+ SOC_DAPM_SINGLE("LINH4", AK4671_LOUT2_SIGNAL_SELECT, 4, 1, 0),
+ SOC_DAPM_SINGLE("LOOPHL", AK4671_LOUT2_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_rout2_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DACHR", AK4671_ROUT2_SIGNAL_SELECT, 0, 1, 0),
+ SOC_DAPM_SINGLE("RINH1", AK4671_ROUT2_SIGNAL_SELECT, 1, 1, 0),
+ SOC_DAPM_SINGLE("RINH2", AK4671_ROUT2_SIGNAL_SELECT, 2, 1, 0),
+ SOC_DAPM_SINGLE("RINH3", AK4671_ROUT2_SIGNAL_SELECT, 3, 1, 0),
+ SOC_DAPM_SINGLE("RINH4", AK4671_ROUT2_SIGNAL_SELECT, 4, 1, 0),
+ SOC_DAPM_SINGLE("LOOPHR", AK4671_ROUT2_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_lout3_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DACSL", AK4671_LOUT3_SIGNAL_SELECT, 0, 1, 0),
+ SOC_DAPM_SINGLE("LINS1", AK4671_LOUT3_SIGNAL_SELECT, 1, 1, 0),
+ SOC_DAPM_SINGLE("LINS2", AK4671_LOUT3_SIGNAL_SELECT, 2, 1, 0),
+ SOC_DAPM_SINGLE("LINS3", AK4671_LOUT3_SIGNAL_SELECT, 3, 1, 0),
+ SOC_DAPM_SINGLE("LINS4", AK4671_LOUT3_SIGNAL_SELECT, 4, 1, 0),
+ SOC_DAPM_SINGLE("LOOPSL", AK4671_LOUT3_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_rout3_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DACSR", AK4671_ROUT3_SIGNAL_SELECT, 0, 1, 0),
+ SOC_DAPM_SINGLE("RINS1", AK4671_ROUT3_SIGNAL_SELECT, 1, 1, 0),
+ SOC_DAPM_SINGLE("RINS2", AK4671_ROUT3_SIGNAL_SELECT, 2, 1, 0),
+ SOC_DAPM_SINGLE("RINS3", AK4671_ROUT3_SIGNAL_SELECT, 3, 1, 0),
+ SOC_DAPM_SINGLE("RINS4", AK4671_ROUT3_SIGNAL_SELECT, 4, 1, 0),
+ SOC_DAPM_SINGLE("LOOPSR", AK4671_ROUT3_SIGNAL_SELECT, 5, 1, 0),
+};
+
+/* Input MUXs */
+static const char *ak4671_lin_mux_texts[] =
+ {"LIN1", "LIN2", "LIN3", "LIN4"};
+static const struct soc_enum ak4671_lin_mux_enum =
+ SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 0,
+ ARRAY_SIZE(ak4671_lin_mux_texts),
+ ak4671_lin_mux_texts);
+static const struct snd_kcontrol_new ak4671_lin_mux_control =
+ SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum);
+
+static const char *ak4671_rin_mux_texts[] =
+ {"RIN1", "RIN2", "RIN3", "RIN4"};
+static const struct soc_enum ak4671_rin_mux_enum =
+ SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 2,
+ ARRAY_SIZE(ak4671_rin_mux_texts),
+ ak4671_rin_mux_texts);
+static const struct snd_kcontrol_new ak4671_rin_mux_control =
+ SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum);
+
+static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = {
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("LIN1"),
+ SND_SOC_DAPM_INPUT("RIN1"),
+ SND_SOC_DAPM_INPUT("LIN2"),
+ SND_SOC_DAPM_INPUT("RIN2"),
+ SND_SOC_DAPM_INPUT("LIN3"),
+ SND_SOC_DAPM_INPUT("RIN3"),
+ SND_SOC_DAPM_INPUT("LIN4"),
+ SND_SOC_DAPM_INPUT("RIN4"),
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("LOUT1"),
+ SND_SOC_DAPM_OUTPUT("ROUT1"),
+ SND_SOC_DAPM_OUTPUT("LOUT2"),
+ SND_SOC_DAPM_OUTPUT("ROUT2"),
+ SND_SOC_DAPM_OUTPUT("LOUT3"),
+ SND_SOC_DAPM_OUTPUT("ROUT3"),
+
+ /* DAC */
+ SND_SOC_DAPM_DAC("DAC Left", "Left HiFi Playback",
+ AK4671_AD_DA_POWER_MANAGEMENT, 6, 0),
+ SND_SOC_DAPM_DAC("DAC Right", "Right HiFi Playback",
+ AK4671_AD_DA_POWER_MANAGEMENT, 7, 0),
+
+ /* ADC */
+ SND_SOC_DAPM_ADC("ADC Left", "Left HiFi Capture",
+ AK4671_AD_DA_POWER_MANAGEMENT, 4, 0),
+ SND_SOC_DAPM_ADC("ADC Right", "Right HiFi Capture",
+ AK4671_AD_DA_POWER_MANAGEMENT, 5, 0),
+
+ /* PGA */
+ SND_SOC_DAPM_PGA("LOUT2 Mix Amp",
+ AK4671_LOUT2_POWER_MANAGERMENT, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("ROUT2 Mix Amp",
+ AK4671_LOUT2_POWER_MANAGERMENT, 6, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA("LIN1 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("RIN1 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("LIN2 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 2, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("RIN2 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("LIN3 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("RIN3 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("LIN4 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 6, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("RIN4 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 7, 0, NULL, 0),
+
+ /* Output Mixers */
+ SND_SOC_DAPM_MIXER("LOUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 0, 0,
+ &ak4671_lout1_mixer_controls[0],
+ ARRAY_SIZE(ak4671_lout1_mixer_controls)),
+ SND_SOC_DAPM_MIXER("ROUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 1, 0,
+ &ak4671_rout1_mixer_controls[0],
+ ARRAY_SIZE(ak4671_rout1_mixer_controls)),
+ SND_SOC_DAPM_MIXER_E("LOUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT,
+ 0, 0, &ak4671_lout2_mixer_controls[0],
+ ARRAY_SIZE(ak4671_lout2_mixer_controls),
+ ak4671_out2_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_MIXER_E("ROUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT,
+ 1, 0, &ak4671_rout2_mixer_controls[0],
+ ARRAY_SIZE(ak4671_rout2_mixer_controls),
+ ak4671_out2_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_MIXER("LOUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 0, 0,
+ &ak4671_lout3_mixer_controls[0],
+ ARRAY_SIZE(ak4671_lout3_mixer_controls)),
+ SND_SOC_DAPM_MIXER("ROUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 1, 0,
+ &ak4671_rout3_mixer_controls[0],
+ ARRAY_SIZE(ak4671_rout3_mixer_controls)),
+
+ /* Input MUXs */
+ SND_SOC_DAPM_MUX("LIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 2, 0,
+ &ak4671_lin_mux_control),
+ SND_SOC_DAPM_MUX("RIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 3, 0,
+ &ak4671_rin_mux_control),
+
+ /* Mic Power */
+ SND_SOC_DAPM_MICBIAS("Mic Bias", AK4671_AD_DA_POWER_MANAGEMENT, 1, 0),
+
+ /* Supply */
+ SND_SOC_DAPM_SUPPLY("PMPLL", AK4671_PLL_MODE_SELECT1, 0, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"DAC Left", "NULL", "PMPLL"},
+ {"DAC Right", "NULL", "PMPLL"},
+ {"ADC Left", "NULL", "PMPLL"},
+ {"ADC Right", "NULL", "PMPLL"},
+
+ /* Outputs */
+ {"LOUT1", "NULL", "LOUT1 Mixer"},
+ {"ROUT1", "NULL", "ROUT1 Mixer"},
+ {"LOUT2", "NULL", "LOUT2 Mix Amp"},
+ {"ROUT2", "NULL", "ROUT2 Mix Amp"},
+ {"LOUT3", "NULL", "LOUT3 Mixer"},
+ {"ROUT3", "NULL", "ROUT3 Mixer"},
+
+ {"LOUT1 Mixer", "DACL", "DAC Left"},
+ {"ROUT1 Mixer", "DACR", "DAC Right"},
+ {"LOUT2 Mixer", "DACHL", "DAC Left"},
+ {"ROUT2 Mixer", "DACHR", "DAC Right"},
+ {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"},
+ {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"},
+ {"LOUT3 Mixer", "DACSL", "DAC Left"},
+ {"ROUT3 Mixer", "DACSR", "DAC Right"},
+
+ /* Inputs */
+ {"LIN MUX", "LIN1", "LIN1"},
+ {"LIN MUX", "LIN2", "LIN2"},
+ {"LIN MUX", "LIN3", "LIN3"},
+ {"LIN MUX", "LIN4", "LIN4"},
+
+ {"RIN MUX", "RIN1", "RIN1"},
+ {"RIN MUX", "RIN2", "RIN2"},
+ {"RIN MUX", "RIN3", "RIN3"},
+ {"RIN MUX", "RIN4", "RIN4"},
+
+ {"LIN1", NULL, "Mic Bias"},
+ {"RIN1", NULL, "Mic Bias"},
+ {"LIN2", NULL, "Mic Bias"},
+ {"RIN2", NULL, "Mic Bias"},
+
+ {"ADC Left", "NULL", "LIN MUX"},
+ {"ADC Right", "NULL", "RIN MUX"},
+
+ /* Analog Loops */
+ {"LIN1 Mixing Circuit", "NULL", "LIN1"},
+ {"RIN1 Mixing Circuit", "NULL", "RIN1"},
+ {"LIN2 Mixing Circuit", "NULL", "LIN2"},
+ {"RIN2 Mixing Circuit", "NULL", "RIN2"},
+ {"LIN3 Mixing Circuit", "NULL", "LIN3"},
+ {"RIN3 Mixing Circuit", "NULL", "RIN3"},
+ {"LIN4 Mixing Circuit", "NULL", "LIN4"},
+ {"RIN4 Mixing Circuit", "NULL", "RIN4"},
+
+ {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"},
+ {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"},
+ {"LOUT2 Mixer", "LINH1", "LIN1 Mixing Circuit"},
+ {"ROUT2 Mixer", "RINH1", "RIN1 Mixing Circuit"},
+ {"LOUT3 Mixer", "LINS1", "LIN1 Mixing Circuit"},
+ {"ROUT3 Mixer", "RINS1", "RIN1 Mixing Circuit"},
+
+ {"LOUT1 Mixer", "LINL2", "LIN2 Mixing Circuit"},
+ {"ROUT1 Mixer", "RINR2", "RIN2 Mixing Circuit"},
+ {"LOUT2 Mixer", "LINH2", "LIN2 Mixing Circuit"},
+ {"ROUT2 Mixer", "RINH2", "RIN2 Mixing Circuit"},
+ {"LOUT3 Mixer", "LINS2", "LIN2 Mixing Circuit"},
+ {"ROUT3 Mixer", "RINS2", "RIN2 Mixing Circuit"},
+
+ {"LOUT1 Mixer", "LINL3", "LIN3 Mixing Circuit"},
+ {"ROUT1 Mixer", "RINR3", "RIN3 Mixing Circuit"},
+ {"LOUT2 Mixer", "LINH3", "LIN3 Mixing Circuit"},
+ {"ROUT2 Mixer", "RINH3", "RIN3 Mixing Circuit"},
+ {"LOUT3 Mixer", "LINS3", "LIN3 Mixing Circuit"},
+ {"ROUT3 Mixer", "RINS3", "RIN3 Mixing Circuit"},
+
+ {"LOUT1 Mixer", "LINL4", "LIN4 Mixing Circuit"},
+ {"ROUT1 Mixer", "RINR4", "RIN4 Mixing Circuit"},
+ {"LOUT2 Mixer", "LINH4", "LIN4 Mixing Circuit"},
+ {"ROUT2 Mixer", "RINH4", "RIN4 Mixing Circuit"},
+ {"LOUT3 Mixer", "LINS4", "LIN4 Mixing Circuit"},
+ {"ROUT3 Mixer", "RINS4", "RIN4 Mixing Circuit"},
+};
+
+static int ak4671_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, ak4671_dapm_widgets,
+ ARRAY_SIZE(ak4671_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static int ak4671_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 fs;
+
+ fs = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0);
+ fs &= ~AK4671_FS;
+
+ switch (params_rate(params)) {
+ case 8000:
+ fs |= AK4671_FS_8KHZ;
+ break;
+ case 12000:
+ fs |= AK4671_FS_12KHZ;
+ break;
+ case 16000:
+ fs |= AK4671_FS_16KHZ;
+ break;
+ case 24000:
+ fs |= AK4671_FS_24KHZ;
+ break;
+ case 11025:
+ fs |= AK4671_FS_11_025KHZ;
+ break;
+ case 22050:
+ fs |= AK4671_FS_22_05KHZ;
+ break;
+ case 32000:
+ fs |= AK4671_FS_32KHZ;
+ break;
+ case 44100:
+ fs |= AK4671_FS_44_1KHZ;
+ break;
+ case 48000:
+ fs |= AK4671_FS_48KHZ;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, fs);
+
+ return 0;
+}
+
+static int ak4671_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 pll;
+
+ pll = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0);
+ pll &= ~AK4671_PLL;
+
+ switch (freq) {
+ case 11289600:
+ pll |= AK4671_PLL_11_2896MHZ;
+ break;
+ case 12000000:
+ pll |= AK4671_PLL_12MHZ;
+ break;
+ case 12288000:
+ pll |= AK4671_PLL_12_288MHZ;
+ break;
+ case 13000000:
+ pll |= AK4671_PLL_13MHZ;
+ break;
+ case 13500000:
+ pll |= AK4671_PLL_13_5MHZ;
+ break;
+ case 19200000:
+ pll |= AK4671_PLL_19_2MHZ;
+ break;
+ case 24000000:
+ pll |= AK4671_PLL_24MHZ;
+ break;
+ case 26000000:
+ pll |= AK4671_PLL_26MHZ;
+ break;
+ case 27000000:
+ pll |= AK4671_PLL_27MHZ;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, pll);
+
+ return 0;
+}
+
+static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 mode;
+ u8 format;
+
+ /* set master/slave audio interface */
+ mode = snd_soc_read(codec, AK4671_PLL_MODE_SELECT1);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ mode |= AK4671_M_S;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ mode &= ~(AK4671_M_S);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ format = snd_soc_read(codec, AK4671_FORMAT_SELECT);
+ format &= ~AK4671_DIF;
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ format |= AK4671_DIF_I2S_MODE;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ format |= AK4671_DIF_MSB_MODE;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ format |= AK4671_DIF_DSP_MODE;
+ format |= AK4671_BCKP;
+ format |= AK4671_MSBS;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set mode and format */
+ snd_soc_write(codec, AK4671_PLL_MODE_SELECT1, mode);
+ snd_soc_write(codec, AK4671_FORMAT_SELECT, format);
+
+ return 0;
+}
+
+static int ak4671_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u8 reg;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ case SND_SOC_BIAS_STANDBY:
+ reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT);
+ snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT,
+ reg | AK4671_PMVCM);
+ break;
+ case SND_SOC_BIAS_OFF:
+ snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define AK4671_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000)
+
+#define AK4671_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+
+static struct snd_soc_dai_ops ak4671_dai_ops = {
+ .hw_params = ak4671_hw_params,
+ .set_sysclk = ak4671_set_dai_sysclk,
+ .set_fmt = ak4671_set_dai_fmt,
+};
+
+struct snd_soc_dai ak4671_dai = {
+ .name = "AK4671",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AK4671_RATES,
+ .formats = AK4671_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AK4671_RATES,
+ .formats = AK4671_FORMATS,},
+ .ops = &ak4671_dai_ops,
+};
+EXPORT_SYMBOL_GPL(ak4671_dai);
+
+static int ak4671_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (ak4671_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = ak4671_codec;
+ codec = ak4671_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ snd_soc_add_controls(codec, ak4671_snd_controls,
+ ARRAY_SIZE(ak4671_snd_controls));
+ ak4671_add_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto card_err;
+ }
+
+ ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+static int ak4671_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_ak4671 = {
+ .probe = ak4671_probe,
+ .remove = ak4671_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ak4671);
+
+static int ak4671_register(struct ak4671_priv *ak4671,
+ enum snd_soc_control_type control)
+{
+ int ret;
+ struct snd_soc_codec *codec = &ak4671->codec;
+
+ if (ak4671_codec) {
+ dev_err(codec->dev, "Another AK4671 is registered\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = ak4671;
+ codec->name = "AK4671";
+ codec->owner = THIS_MODULE;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = ak4671_set_bias_level;
+ codec->dai = &ak4671_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = AK4671_CACHEREGNUM;
+ codec->reg_cache = &ak4671->reg_cache;
+
+ memcpy(codec->reg_cache, ak4671_reg, sizeof(ak4671_reg));
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, control);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
+ ak4671_dai.dev = codec->dev;
+ ak4671_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ goto err;
+ }
+
+ ret = snd_soc_register_dai(&ak4671_dai);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ goto err_codec;
+ }
+
+ return 0;
+
+err_codec:
+ snd_soc_unregister_codec(codec);
+err:
+ kfree(ak4671);
+ return ret;
+}
+
+static void ak4671_unregister(struct ak4671_priv *ak4671)
+{
+ ak4671_set_bias_level(&ak4671->codec, SND_SOC_BIAS_OFF);
+ snd_soc_unregister_dai(&ak4671_dai);
+ snd_soc_unregister_codec(&ak4671->codec);
+ kfree(ak4671);
+ ak4671_codec = NULL;
+}
+
+static int __devinit ak4671_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct ak4671_priv *ak4671;
+ struct snd_soc_codec *codec;
+
+ ak4671 = kzalloc(sizeof(struct ak4671_priv), GFP_KERNEL);
+ if (ak4671 == NULL)
+ return -ENOMEM;
+
+ codec = &ak4671->codec;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+
+ i2c_set_clientdata(client, ak4671);
+ codec->control_data = client;
+
+ codec->dev = &client->dev;
+
+ return ak4671_register(ak4671, SND_SOC_I2C);
+}
+
+static __devexit int ak4671_i2c_remove(struct i2c_client *client)
+{
+ struct ak4671_priv *ak4671 = i2c_get_clientdata(client);
+
+ ak4671_unregister(ak4671);
+
+ return 0;
+}
+
+static const struct i2c_device_id ak4671_i2c_id[] = {
+ { "ak4671", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id);
+
+static struct i2c_driver ak4671_i2c_driver = {
+ .driver = {
+ .name = "ak4671",
+ .owner = THIS_MODULE,
+ },
+ .probe = ak4671_i2c_probe,
+ .remove = __devexit_p(ak4671_i2c_remove),
+ .id_table = ak4671_i2c_id,
+};
+
+static int __init ak4671_modinit(void)
+{
+ return i2c_add_driver(&ak4671_i2c_driver);
+}
+module_init(ak4671_modinit);
+
+static void __exit ak4671_exit(void)
+{
+ i2c_del_driver(&ak4671_i2c_driver);
+}
+module_exit(ak4671_exit);
+
+MODULE_DESCRIPTION("ASoC AK4671 codec driver");
+MODULE_AUTHOR("Joonyoung Shim <jy0922.shim@samsung.com>");
+MODULE_LICENSE("GPL");
--- /dev/null
+/*
+ * ak4671.h -- audio driver for AK4671
+ *
+ * Copyright (C) 2009 Samsung Electronics Co.Ltd
+ * Author: Joonyoung Shim <jy0922.shim@samsung.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef _AK4671_H
+#define _AK4671_H
+
+#define AK4671_AD_DA_POWER_MANAGEMENT 0x00
+#define AK4671_PLL_MODE_SELECT0 0x01
+#define AK4671_PLL_MODE_SELECT1 0x02
+#define AK4671_FORMAT_SELECT 0x03
+#define AK4671_MIC_SIGNAL_SELECT 0x04
+#define AK4671_MIC_AMP_GAIN 0x05
+#define AK4671_MIXING_POWER_MANAGEMENT0 0x06
+#define AK4671_MIXING_POWER_MANAGEMENT1 0x07
+#define AK4671_OUTPUT_VOLUME_CONTROL 0x08
+#define AK4671_LOUT1_SIGNAL_SELECT 0x09
+#define AK4671_ROUT1_SIGNAL_SELECT 0x0a
+#define AK4671_LOUT2_SIGNAL_SELECT 0x0b
+#define AK4671_ROUT2_SIGNAL_SELECT 0x0c
+#define AK4671_LOUT3_SIGNAL_SELECT 0x0d
+#define AK4671_ROUT3_SIGNAL_SELECT 0x0e
+#define AK4671_LOUT1_POWER_MANAGERMENT 0x0f
+#define AK4671_LOUT2_POWER_MANAGERMENT 0x10
+#define AK4671_LOUT3_POWER_MANAGERMENT 0x11
+#define AK4671_LCH_INPUT_VOLUME_CONTROL 0x12
+#define AK4671_RCH_INPUT_VOLUME_CONTROL 0x13
+#define AK4671_ALC_REFERENCE_SELECT 0x14
+#define AK4671_DIGITAL_MIXING_CONTROL 0x15
+#define AK4671_ALC_TIMER_SELECT 0x16
+#define AK4671_ALC_MODE_CONTROL 0x17
+#define AK4671_MODE_CONTROL1 0x18
+#define AK4671_MODE_CONTROL2 0x19
+#define AK4671_LCH_OUTPUT_VOLUME_CONTROL 0x1a
+#define AK4671_RCH_OUTPUT_VOLUME_CONTROL 0x1b
+#define AK4671_SIDETONE_A_CONTROL 0x1c
+#define AK4671_DIGITAL_FILTER_SELECT 0x1d
+#define AK4671_FIL3_COEFFICIENT0 0x1e
+#define AK4671_FIL3_COEFFICIENT1 0x1f
+#define AK4671_FIL3_COEFFICIENT2 0x20
+#define AK4671_FIL3_COEFFICIENT3 0x21
+#define AK4671_EQ_COEFFICIENT0 0x22
+#define AK4671_EQ_COEFFICIENT1 0x23
+#define AK4671_EQ_COEFFICIENT2 0x24
+#define AK4671_EQ_COEFFICIENT3 0x25
+#define AK4671_EQ_COEFFICIENT4 0x26
+#define AK4671_EQ_COEFFICIENT5 0x27
+#define AK4671_FIL1_COEFFICIENT0 0x28
+#define AK4671_FIL1_COEFFICIENT1 0x29
+#define AK4671_FIL1_COEFFICIENT2 0x2a
+#define AK4671_FIL1_COEFFICIENT3 0x2b
+#define AK4671_FIL2_COEFFICIENT0 0x2c
+#define AK4671_FIL2_COEFFICIENT1 0x2d
+#define AK4671_FIL2_COEFFICIENT2 0x2e
+#define AK4671_FIL2_COEFFICIENT3 0x2f
+#define AK4671_DIGITAL_FILTER_SELECT2 0x30
+#define AK4671_E1_COEFFICIENT0 0x32
+#define AK4671_E1_COEFFICIENT1 0x33
+#define AK4671_E1_COEFFICIENT2 0x34
+#define AK4671_E1_COEFFICIENT3 0x35
+#define AK4671_E1_COEFFICIENT4 0x36
+#define AK4671_E1_COEFFICIENT5 0x37
+#define AK4671_E2_COEFFICIENT0 0x38
+#define AK4671_E2_COEFFICIENT1 0x39
+#define AK4671_E2_COEFFICIENT2 0x3a
+#define AK4671_E2_COEFFICIENT3 0x3b
+#define AK4671_E2_COEFFICIENT4 0x3c
+#define AK4671_E2_COEFFICIENT5 0x3d
+#define AK4671_E3_COEFFICIENT0 0x3e
+#define AK4671_E3_COEFFICIENT1 0x3f
+#define AK4671_E3_COEFFICIENT2 0x40
+#define AK4671_E3_COEFFICIENT3 0x41
+#define AK4671_E3_COEFFICIENT4 0x42
+#define AK4671_E3_COEFFICIENT5 0x43
+#define AK4671_E4_COEFFICIENT0 0x44
+#define AK4671_E4_COEFFICIENT1 0x45
+#define AK4671_E4_COEFFICIENT2 0x46
+#define AK4671_E4_COEFFICIENT3 0x47
+#define AK4671_E4_COEFFICIENT4 0x48
+#define AK4671_E4_COEFFICIENT5 0x49
+#define AK4671_E5_COEFFICIENT0 0x4a
+#define AK4671_E5_COEFFICIENT1 0x4b
+#define AK4671_E5_COEFFICIENT2 0x4c
+#define AK4671_E5_COEFFICIENT3 0x4d
+#define AK4671_E5_COEFFICIENT4 0x4e
+#define AK4671_E5_COEFFICIENT5 0x4f
+#define AK4671_EQ_CONTROL_250HZ_100HZ 0x50
+#define AK4671_EQ_CONTROL_3500HZ_1KHZ 0x51
+#define AK4671_EQ_CONTRO_10KHZ 0x52
+#define AK4671_PCM_IF_CONTROL0 0x53
+#define AK4671_PCM_IF_CONTROL1 0x54
+#define AK4671_PCM_IF_CONTROL2 0x55
+#define AK4671_DIGITAL_VOLUME_B_CONTROL 0x56
+#define AK4671_DIGITAL_VOLUME_C_CONTROL 0x57
+#define AK4671_SIDETONE_VOLUME_CONTROL 0x58
+#define AK4671_DIGITAL_MIXING_CONTROL2 0x59
+#define AK4671_SAR_ADC_CONTROL 0x5a
+
+#define AK4671_CACHEREGNUM (AK4671_SAR_ADC_CONTROL + 1)
+
+/* Bitfield Definitions */
+
+/* AK4671_AD_DA_POWER_MANAGEMENT (0x00) Fields */
+#define AK4671_PMVCM 0x01
+
+/* AK4671_PLL_MODE_SELECT0 (0x01) Fields */
+#define AK4671_PLL 0x0f
+#define AK4671_PLL_11_2896MHZ (4 << 0)
+#define AK4671_PLL_12_288MHZ (5 << 0)
+#define AK4671_PLL_12MHZ (6 << 0)
+#define AK4671_PLL_24MHZ (7 << 0)
+#define AK4671_PLL_19_2MHZ (8 << 0)
+#define AK4671_PLL_13_5MHZ (12 << 0)
+#define AK4671_PLL_27MHZ (13 << 0)
+#define AK4671_PLL_13MHZ (14 << 0)
+#define AK4671_PLL_26MHZ (15 << 0)
+#define AK4671_FS 0xf0
+#define AK4671_FS_8KHZ (0 << 4)
+#define AK4671_FS_12KHZ (1 << 4)
+#define AK4671_FS_16KHZ (2 << 4)
+#define AK4671_FS_24KHZ (3 << 4)
+#define AK4671_FS_11_025KHZ (5 << 4)
+#define AK4671_FS_22_05KHZ (7 << 4)
+#define AK4671_FS_32KHZ (10 << 4)
+#define AK4671_FS_48KHZ (11 << 4)
+#define AK4671_FS_44_1KHZ (15 << 4)
+
+/* AK4671_PLL_MODE_SELECT1 (0x02) Fields */
+#define AK4671_PMPLL 0x01
+#define AK4671_M_S 0x02
+
+/* AK4671_FORMAT_SELECT (0x03) Fields */
+#define AK4671_DIF 0x03
+#define AK4671_DIF_DSP_MODE (0 << 0)
+#define AK4671_DIF_MSB_MODE (2 << 0)
+#define AK4671_DIF_I2S_MODE (3 << 0)
+#define AK4671_BCKP 0x04
+#define AK4671_MSBS 0x08
+#define AK4671_SDOD 0x10
+
+/* AK4671_LOUT2_POWER_MANAGEMENT (0x10) Fields */
+#define AK4671_MUTEN 0x04
+
+extern struct snd_soc_dai ak4671_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ak4671;
+
+#endif
/* codec private data */
struct aic3x_priv {
+ struct snd_soc_codec codec;
unsigned int sysclk;
int master;
};
* 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;
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);
/* 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);
+ return 0;
+}
- snd_soc_add_controls(codec, aic3x_snd_controls,
- ARRAY_SIZE(aic3x_snd_controls));
- aic3x_add_widgets(codec);
- ret = snd_soc_init_card(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) {
- printk(KERN_ERR "aic3x: failed to register card\n");
- goto card_err;
+ dev_err(codec->dev, "Failed to initialise device\n");
+ return ret;
}
- return ret;
+ aic3x_codec = codec;
-card_err:
- snd_soc_free_pcms(socdev);
- snd_soc_dapm_free(socdev);
-pcm_err:
- kfree(codec->reg_cache);
- return ret;
+ 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 struct snd_soc_device *aic3x_socdev;
+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)
/*
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;
+
+ aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL);
+ if (aic3x == NULL) {
+ dev_err(&i2c->dev, "failed to create private data\n");
+ return -ENOMEM;
+ }
- i2c_set_clientdata(i2c, codec);
+ 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);
.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;
+ if (ret)
+ printk(KERN_ERR "%s: error regsitering i2c driver, %d\n",
+ __func__, ret);
+}
-err_driver:
+static inline void aic3x_i2c_exit(void)
+{
i2c_del_driver(&aic3x_i2c_driver);
- return -ENODEV;
}
+#else
+static inline void aic3x_i2c_init(void) { }
+static inline void aic3x_i2c_exit(void) { }
#endif
static int aic3x_probe(struct platform_device *pdev)
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);
+ codec = aic3x_codec;
+ if (!codec) {
+ dev_err(&pdev->dev, "Codec not registered\n");
+ return -ENODEV;
+ }
+ socdev->card->codec = codec;
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;
+ if (setup) {
+ /* 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);
}
- codec->private_data = aic3x;
- socdev->card->codec = codec;
- mutex_init(&codec->mutex);
- INIT_LIST_HEAD(&codec->dapm_widgets);
- INIT_LIST_HEAD(&codec->dapm_paths);
-
- 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);
+ /* 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;
}
-#else
- /* Add other interfaces here */
-#endif
- if (ret != 0) {
- kfree(codec->private_data);
- kfree(codec);
+ 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;
}
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;
}
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);
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];
};
struct wm8350_jack_data hpl;
struct wm8350_jack_data hpr;
struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
+ int fll_freq_out;
+ int fll_freq_in;
};
static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec,
/* Out4 Capture Mux */
static const struct snd_kcontrol_new wm8350_out4_capture_controls =
-SOC_DAPM_ENUM("Route", wm8350_enum[8]);
+SOC_DAPM_ENUM("Route", wm8350_enum[7]);
static const struct snd_soc_dapm_widget wm8350_dapm_widgets[] = {
}
static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in,
+ int pll_id, int source, unsigned int freq_in,
unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct wm8350 *wm8350 = codec->control_data;
+ struct wm8350_data *priv = codec->private_data;
struct _fll_div fll_div;
int ret = 0;
u16 fll_1, fll_4;
+ if (freq_in == priv->fll_freq_in && freq_out == priv->fll_freq_out)
+ return 0;
+
/* power down FLL - we need to do this for reconfiguration */
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4,
WM8350_FLL_ENA | WM8350_FLL_OSC_ENA);
wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_OSC_ENA);
wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_ENA);
+ priv->fll_freq_out = freq_out;
+ priv->fll_freq_in = freq_in;
+
return 0;
}
}
static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
- unsigned int freq_in, unsigned int freq_out)
+ int source, unsigned int freq_in,
+ unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct wm8400_priv *wm8400 = codec->private_data;
pll_div.k = K;
}
-static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 reg;
return 0;
}
-static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
int offset;
struct snd_soc_codec *codec = codec_dai->codec;
return 0;
snd_soc_write(codec, WM8580_PLLA1 + offset, pll_div.k & 0x1ff);
- snd_soc_write(codec, WM8580_PLLA2 + offset, (pll_div.k >> 9) & 0xff);
+ snd_soc_write(codec, WM8580_PLLA2 + offset, (pll_div.k >> 9) & 0x1ff);
snd_soc_write(codec, WM8580_PLLA3 + offset,
(pll_div.k >> 18 & 0xf) | (pll_div.n << 4));
reg = snd_soc_read(codec, WM8580_PLLA4 + offset);
- reg &= ~0x3f;
+ reg &= ~0x1b;
reg |= pll_div.prescale | pll_div.postscale << 1 |
pll_div.freqmode << 3;
pll_div->k = K;
}
-static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
u16 reg, enable;
int offset;
return 0;
}
-static int wm8900_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8900_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
return wm8900_set_fll(codec_dai->codec, pll_id, freq_in, freq_out);
}
}
/* Untested at the moment */
-static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 reg;
return 0;
}
-static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 reg;
#include <linux/module.h>
#include <linux/moduleparam.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
pll_div.k = K;
}
-static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 reg;
pll_div->k = K;
}
-static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
u16 reg;
struct snd_soc_codec *codec = codec_dai->codec;
return 0;
}
-static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id,
+static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source,
unsigned int Fref, unsigned int Fout)
{
struct snd_soc_codec *codec = dai->codec;
SOC_DAPM_SINGLE("DAC Switch", WM8993_SPEAKER_MIXER, 0, 1, 0),
};
+static const char *aif_text[] = {
+ "Left", "Right"
+};
+
+static const struct soc_enum aifoutl_enum =
+ SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_1, 15, 2, aif_text);
+
+static const struct snd_kcontrol_new aifoutl_mux =
+ SOC_DAPM_ENUM("AIFOUTL Mux", aifoutl_enum);
+
+static const struct soc_enum aifoutr_enum =
+ SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_1, 14, 2, aif_text);
+
+static const struct snd_kcontrol_new aifoutr_mux =
+ SOC_DAPM_ENUM("AIFOUTR Mux", aifoutr_enum);
+
+static const struct soc_enum aifinl_enum =
+ SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_2, 15, 2, aif_text);
+
+static const struct snd_kcontrol_new aifinl_mux =
+ SOC_DAPM_ENUM("AIFINL Mux", aifinl_enum);
+
+static const struct soc_enum aifinr_enum =
+ SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_2, 14, 2, aif_text);
+
+static const struct snd_kcontrol_new aifinr_mux =
+ SOC_DAPM_ENUM("AIFINR Mux", aifinr_enum);
+
+static const char *sidetone_text[] = {
+ "None", "Left", "Right"
+};
+
+static const struct soc_enum sidetonel_enum =
+ SOC_ENUM_SINGLE(WM8993_DIGITAL_SIDE_TONE, 2, 3, sidetone_text);
+
+static const struct snd_kcontrol_new sidetonel_mux =
+ SOC_DAPM_ENUM("Left Sidetone", sidetonel_enum);
+
+static const struct soc_enum sidetoner_enum =
+ SOC_ENUM_SINGLE(WM8993_DIGITAL_SIDE_TONE, 0, 3, sidetone_text);
+
+static const struct snd_kcontrol_new sidetoner_mux =
+ SOC_DAPM_ENUM("Right Sidetone", sidetoner_enum);
+
static const struct snd_soc_dapm_widget wm8993_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8993_BUS_CONTROL_1, 1, 0, clk_sys_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("TOCLK", WM8993_CLOCKING_1, 14, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8993_CLOCKING_3, 0, 0, NULL, 0),
+SND_SOC_DAPM_ADC("ADCL", NULL, WM8993_POWER_MANAGEMENT_2, 1, 0),
+SND_SOC_DAPM_ADC("ADCR", NULL, WM8993_POWER_MANAGEMENT_2, 0, 0),
+
+SND_SOC_DAPM_MUX("AIFOUTL Mux", SND_SOC_NOPM, 0, 0, &aifoutl_mux),
+SND_SOC_DAPM_MUX("AIFOUTR Mux", SND_SOC_NOPM, 0, 0, &aifoutr_mux),
+
+SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0),
-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_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 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_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &aifinl_mux),
+SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &aifinr_mux),
+
+SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &sidetonel_mux),
+SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &sidetoner_mux),
+
+SND_SOC_DAPM_DAC("DACL", NULL, WM8993_POWER_MANAGEMENT_3, 1, 0),
+SND_SOC_DAPM_DAC("DACR", NULL, WM8993_POWER_MANAGEMENT_3, 0, 0),
SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux),
SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux),
{ "ADCR", NULL, "CLK_SYS" },
{ "ADCR", NULL, "CLK_DSP" },
+ { "AIFOUTL Mux", "Left", "ADCL" },
+ { "AIFOUTL Mux", "Right", "ADCR" },
+ { "AIFOUTR Mux", "Left", "ADCL" },
+ { "AIFOUTR Mux", "Right", "ADCR" },
+
+ { "AIFOUTL", NULL, "AIFOUTL Mux" },
+ { "AIFOUTR", NULL, "AIFOUTR Mux" },
+
+ { "DACL Mux", "Left", "AIFINL" },
+ { "DACL Mux", "Right", "AIFINR" },
+ { "DACR Mux", "Left", "AIFINL" },
+ { "DACR Mux", "Right", "AIFINR" },
+
+ { "DACL Sidetone", "Left", "ADCL" },
+ { "DACL Sidetone", "Right", "ADCR" },
+ { "DACR Sidetone", "Left", "ADCL" },
+ { "DACR Sidetone", "Right", "ADCR" },
+
{ "DACL", NULL, "CLK_SYS" },
{ "DACL", NULL, "CLK_DSP" },
+ { "DACL", NULL, "DACL Mux" },
+ { "DACL", NULL, "DACL Sidetone" },
{ "DACR", NULL, "CLK_SYS" },
{ "DACR", NULL, "CLK_DSP" },
+ { "DACR", NULL, "DACR Mux" },
+ { "DACR", NULL, "DACR Sidetone" },
{ "Left Output Mixer", "DAC Switch", "DACL" },
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;
return 0;
}
-static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
return wm9713_set_pll(codec, pll_id, freq_in, freq_out);
#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>
/* Start the DC servo */
snd_soc_update_bits(codec, WM8993_DC_SERVO_0,
- WM8993_DCS_ENA_CHAN_0 |
- WM8993_DCS_ENA_CHAN_1 |
- WM8993_DCS_TRIG_STARTUP_1 |
- WM8993_DCS_TRIG_STARTUP_0,
+ 0xFFFF,
WM8993_DCS_ENA_CHAN_0 |
WM8993_DCS_ENA_CHAN_1 |
WM8993_DCS_TRIG_STARTUP_1 |
WM8993_DCS_TRIG_STARTUP_0);
wait_for_dc_servo(codec);
- snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
- WM8993_DCS_TIMER_PERIOD_01_MASK, 0xa);
reg |= WM8993_HPOUT1R_OUTP | WM8993_HPOUT1R_RMV_SHORT |
WM8993_HPOUT1L_OUTP | WM8993_HPOUT1L_RMV_SHORT;
WM8993_HPOUT1R_DLY |
WM8993_HPOUT1R_OUTP);
- snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
- WM8993_DCS_TIMER_PERIOD_01_MASK, 0);
snd_soc_update_bits(codec, WM8993_DC_SERVO_0,
- WM8993_DCS_ENA_CHAN_0 |
- WM8993_DCS_ENA_CHAN_1, 0);
+ 0xffff, 0);
snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg);
snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
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,
tristate
config SND_DAVINCI_SOC_EVM
- tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM"
+ tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM"
depends on SND_DAVINCI_SOC
- depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM
+ depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM
select SND_DAVINCI_SOC_I2S
select SND_SOC_TLV320AIC3X
help
#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>
unsigned sysclk;
/* ASP1 on DM355 EVM is clocked by an external oscillator */
- if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm())
+ if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() ||
+ machine_is_davinci_dm365_evm())
sysclk = 27000000;
/* ASP0 in DM6446 EVM is clocked by U55, as configured by
.ops = &evm_ops,
};
-/* davinci-evm audio machine driver */
+/* davinci dm6446, dm355 or dm365 evm audio machine driver */
static struct snd_soc_card snd_soc_card_evm = {
.name = "DaVinci EVM",
.platform = &davinci_soc_platform,
.num_links = 1,
};
-/* evm audio private data */
-static struct aic3x_setup_data evm_aic3x_setup = {
- .i2c_bus = 1,
- .i2c_address = 0x1b,
-};
-
-/* dm6467 evm audio private data */
-static struct aic3x_setup_data dm6467_evm_aic3x_setup = {
- .i2c_bus = 1,
- .i2c_address = 0x18,
-};
-
-static struct aic3x_setup_data da8xx_evm_aic3x_setup = {
- .i2c_bus = 1,
- .i2c_address = 0x18,
-};
+static struct aic3x_setup_data aic3x_setup;
/* evm audio subsystem */
static struct snd_soc_device evm_snd_devdata = {
.card = &snd_soc_card_evm,
.codec_dev = &soc_codec_dev_aic3x,
- .codec_data = &evm_aic3x_setup,
+ .codec_data = &aic3x_setup,
};
/* evm audio subsystem */
static struct snd_soc_device dm6467_evm_snd_devdata = {
.card = &dm6467_snd_soc_card_evm,
.codec_dev = &soc_codec_dev_aic3x,
- .codec_data = &dm6467_evm_aic3x_setup,
+ .codec_data = &aic3x_setup,
};
/* evm audio subsystem */
static struct snd_soc_device da830_evm_snd_devdata = {
.card = &da830_snd_soc_card,
.codec_dev = &soc_codec_dev_aic3x,
- .codec_data = &da8xx_evm_aic3x_setup,
+ .codec_data = &aic3x_setup,
};
static struct snd_soc_device da850_evm_snd_devdata = {
.card = &da850_snd_soc_card,
.codec_dev = &soc_codec_dev_aic3x,
- .codec_data = &da8xx_evm_aic3x_setup,
+ .codec_data = &aic3x_setup,
};
static struct platform_device *evm_snd_device;
int index;
int ret;
- if (machine_is_davinci_evm()) {
+ if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) {
evm_snd_dev_data = &evm_snd_devdata;
index = 0;
} else if (machine_is_davinci_dm355_evm()) {
int channel_size)
{
u32 fmt = 0;
+ u32 mask, rotate;
switch (channel_size) {
case DAVINCI_AUDIO_WORD_8:
fmt = 0x03;
+ rotate = 6;
+ mask = 0x000000ff;
break;
case DAVINCI_AUDIO_WORD_12:
fmt = 0x05;
+ rotate = 5;
+ mask = 0x00000fff;
break;
case DAVINCI_AUDIO_WORD_16:
fmt = 0x07;
+ rotate = 4;
+ mask = 0x0000ffff;
break;
case DAVINCI_AUDIO_WORD_20:
fmt = 0x09;
+ rotate = 3;
+ mask = 0x000fffff;
break;
case DAVINCI_AUDIO_WORD_24:
fmt = 0x0B;
+ rotate = 2;
+ mask = 0x00ffffff;
break;
case DAVINCI_AUDIO_WORD_28:
fmt = 0x0D;
+ rotate = 1;
+ mask = 0x0fffffff;
break;
case DAVINCI_AUDIO_WORD_32:
fmt = 0x0F;
+ rotate = 0;
+ mask = 0xffffffff;
break;
default:
RXSSZ(fmt), RXSSZ(0x0F));
mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
TXSSZ(fmt), TXSSZ(0x0F));
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate),
+ TXROT(7));
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate),
+ RXROT(7));
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask);
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask);
+
return 0;
}
printk(KERN_ERR "playback tdm slot %d not supported\n",
dev->tdm_slots);
- mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF);
mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
} else {
/* bit stream is MSB first with no delay */
printk(KERN_ERR "capture tdm slot %d not supported\n",
dev->tdm_slots);
- mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF);
mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
}
}
int size, irq, rc;
const __be32 *prop;
void __iomem *regs;
+ int ret;
/* Fetch the registers and IRQ of the PSC */
irq = irq_of_parse_and_map(op->node, 0);
/* Allocate and initialize the driver private data */
psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL);
if (!psc_dma) {
- iounmap(regs);
- return -ENOMEM;
+ ret = -ENOMEM;
+ goto out_unmap;
}
/* Get the PSC ID */
prop = of_get_property(op->node, "cell-index", &size);
- if (!prop || size < sizeof *prop)
- return -ENODEV;
+ if (!prop || size < sizeof *prop) {
+ ret = -ENODEV;
+ goto out_free;
+ }
spin_lock_init(&psc_dma->lock);
mutex_init(&psc_dma->mutex);
if (!psc_dma->capture.bcom_task ||
!psc_dma->playback.bcom_task) {
dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
- iounmap(regs);
- kfree(psc_dma);
- return -ENODEV;
+ ret = -ENODEV;
+ goto out_free;
}
/* Disable all interrupts and reset the PSC */
&psc_dma_bcom_irq_tx, IRQF_SHARED,
"psc-dma-playback", &psc_dma->playback);
if (rc) {
- free_irq(psc_dma->irq, psc_dma);
- free_irq(psc_dma->capture.irq,
- &psc_dma->capture);
- free_irq(psc_dma->playback.irq,
- &psc_dma->playback);
- return -ENODEV;
+ ret = -ENODEV;
+ goto out_irq;
}
/* Save what we've done so it can be found again later */
/* Tell the ASoC OF helpers about it */
return snd_soc_register_platform(&mpc5200_audio_dma_platform);
+out_irq:
+ free_irq(psc_dma->irq, psc_dma);
+ free_irq(psc_dma->capture.irq, &psc_dma->capture);
+ free_irq(psc_dma->playback.irq, &psc_dma->playback);
+out_free:
+ kfree(psc_dma);
+out_unmap:
+ iounmap(regs);
+ return ret;
}
EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create);
/* codec PLL input is 25 MHz */
- ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG,
+ ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, IGNORED_ARG,
25000000, pll_out);
if (ret < 0) {
printk(KERN_ERR "Error when setting PLL input\n");
*/
#include <linux/clk.h>
+#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
/* 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,
};
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;
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;
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;
}
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;
} 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;
}
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) {
/* 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;
+ regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE;
+ regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE;
}
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
return 0;
}
+static int omap_mcbsp_dai_set_rcvr_src(struct omap_mcbsp_data *mcbsp_data,
+ int clk_id)
+{
+ int sel_bit, set = 0;
+ u16 reg = OMAP2_CONTROL_DEVCONF0;
+
+ if (cpu_class_is_omap1())
+ return -EINVAL; /* TODO: Can this be implemented for OMAP1? */
+ if (mcbsp_data->bus_id != 0)
+ return -EINVAL;
+
+ switch (clk_id) {
+ case OMAP_MCBSP_CLKR_SRC_CLKX:
+ set = 1;
+ case OMAP_MCBSP_CLKR_SRC_CLKR:
+ sel_bit = 3;
+ break;
+ case OMAP_MCBSP_FSR_SRC_FSX:
+ set = 1;
+ case OMAP_MCBSP_FSR_SRC_FSR:
+ sel_bit = 4;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (set)
+ omap_ctrl_writel(omap_ctrl_readl(reg) | (1 << sel_bit), reg);
+ else
+ omap_ctrl_writel(omap_ctrl_readl(reg) & ~(1 << sel_bit), reg);
+
+ return 0;
+}
+
static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq,
int dir)
case OMAP_MCBSP_SYSCLK_CLKR_EXT:
regs->pcr0 |= SCLKME;
break;
+
+ case OMAP_MCBSP_CLKR_SRC_CLKR:
+ case OMAP_MCBSP_CLKR_SRC_CLKX:
+ case OMAP_MCBSP_FSR_SRC_FSR:
+ case OMAP_MCBSP_FSR_SRC_FSX:
+ err = omap_mcbsp_dai_set_rcvr_src(mcbsp_data, clk_id);
+ break;
default:
err = -ENODEV;
}
OMAP_MCBSP_SYSCLK_CLK, /* Internal ICLK */
OMAP_MCBSP_SYSCLK_CLKX_EXT, /* External CLKX pin */
OMAP_MCBSP_SYSCLK_CLKR_EXT, /* External CLKR pin */
+ OMAP_MCBSP_CLKR_SRC_CLKR, /* CLKR from CLKR pin */
+ OMAP_MCBSP_CLKR_SRC_CLKX, /* CLKR from CLKX pin */
+ OMAP_MCBSP_FSR_SRC_FSR, /* FSR from FSR pin */
+ OMAP_MCBSP_FSR_SRC_FSX, /* FSR from FSX pin */
};
/* McBSP dividers */
*/
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;
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;
}
{
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;
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;
.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)
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,
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;
#include "omap-pcm.h"
#include "../codecs/twl4030.h"
-#define TWL4030_INTBR_PMBR1 0x0D
-#define EXTMUTE(value) (value << 2)
+/* TWL4030 PMBR1 Register */
+#define TWL4030_INTBR_PMBR1 0x0D
+/* TWL4030 PMBR1 Register GPIO6 mux bit */
+#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2)
static struct snd_soc_card snd_soc_sdp3430;
static int __init sdp3430_soc_init(void)
{
int ret;
+ u8 pin_mux;
if (!machine_is_omap_3430sdp()) {
pr_debug("Not SDP3430!\n");
*(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
/* Set TWL4030 GPIO6 as EXTMUTE signal */
- twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02),
- TWL4030_MODULE_INTBR);
+ twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux,
+ TWL4030_INTBR_PMBR1);
+ pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03);
+ pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02);
+ twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux,
+ TWL4030_INTBR_PMBR1);
ret = platform_device_add(sdp3430_snd_device);
if (ret)
return ret;
/* set SSP audio pll clock */
- ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps);
+ ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps);
if (ret < 0)
return ret;
/*
* Configure the PLL frequency pxa27x and (afaik - pxa320 only)
*/
-static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
struct ssp_priv *priv = cpu_dai->private_data;
struct ssp_device *ssp = priv->dev.ssp;
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
static int zylonite_wm9713_init(struct snd_soc_codec *codec)
{
if (clk_pout)
- snd_soc_dai_set_pll(&codec->dai[0], 0, clk_get_rate(pout), 0);
+ snd_soc_dai_set_pll(&codec->dai[0], 0, 0,
+ clk_get_rate(pout), 0);
snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
ARRAY_SIZE(zylonite_dapm_widgets));
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out);
+ ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out);
if (ret < 0)
return ret;
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
help
Sat Y if you want to add support for SoC audio on the Jive.
+config SND_S3C64XX_SOC_WM8580
+ tristate "SoC I2S Audio support for WM8580 on SMDK64XX"
+ depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410)
+ depends on BROKEN
+ select SND_SOC_WM8580
+ select SND_S3C64XX_SOC_I2S
+ help
+ Sat Y if you want to add support for SoC audio on the SMDK64XX.
+
config SND_S3C24XX_SOC_SMDK2443_WM9710
tristate "SoC AC97 Audio support for SMDK2443 - WM9710"
depends on SND_S3C24XX_SOC && MACH_SMDK2443
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 && ARCH_S3C2410
+ 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 && ARCH_S3C2410
+ select SND_S3C24XX_SOC_I2S
+ select SND_SOC_TLV320AIC3X
+ select SND_S3C24XX_SOC_SIMTEC
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
+snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o
obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o
obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
obj-$(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-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o
+obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o
+
return ret;
/* codec PLL input is PCLK/4 */
- ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1,
+ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0,
iis_clkrate / 4, pll_out);
if (ret < 0)
return ret;
return ret;
/* codec PLL input is PCLK/4 */
- ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1,
+ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0,
iis_clkrate / 4, pll_out);
if (ret < 0)
return ret;
#include <mach/dma.h>
#include "s3c-i2s-v2.h"
+#include "s3c24xx-pcm.h"
#undef S3C_IIS_V2_SUPPORTED
pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
}
+#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t)
+
/*
* Wait for the LR signal to allow synchronisation to the L/R clock
* from the codec. May only be needed for slave mode.
static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s)
{
u32 iiscon;
- unsigned long timeout = jiffies + msecs_to_jiffies(5);
+ unsigned long loops = msecs_to_loops(5);
pr_debug("Entered %s\n", __func__);
- while (1) {
+ while (--loops) {
iiscon = readl(i2s->regs + S3C2412_IISCON);
if (iiscon & S3C2412_IISCON_LRINDEX)
break;
- if (timeout < jiffies) {
- printk(KERN_ERR "%s: timeout\n", __func__);
- return -ETIMEDOUT;
- }
+ cpu_relax();
+ }
+
+ if (!loops) {
+ printk(KERN_ERR "%s: timeout\n", __func__);
+ return -ETIMEDOUT;
}
return 0;
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_RIGHT_J:
+ iismod |= S3C2412_IISMOD_LR_RLOW;
iismod |= S3C2412_IISMOD_SDF_MSB;
break;
case SND_SOC_DAIFMT_LEFT_J:
+ iismod |= S3C2412_IISMOD_LR_RLOW;
iismod |= S3C2412_IISMOD_SDF_LSB;
break;
case SND_SOC_DAIFMT_I2S:
+ iismod &= ~S3C2412_IISMOD_LR_RLOW;
iismod |= S3C2412_IISMOD_SDF_IIS;
break;
default:
#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
int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
unsigned long irqs;
int ret = 0;
+ int channel = ((struct s3c24xx_pcm_dma_params *)
+ rtd->dai->cpu_dai->dma_data)->channel;
pr_debug("Entered %s\n", __func__);
s3c2412_snd_txctrl(i2s, 1);
local_irq_restore(irqs);
+
+ /*
+ * Load the next buffer to DMA to meet the reqirement
+ * of the auto reload mechanism of S3C24XX.
+ * This call won't bother S3C64XX.
+ */
+ s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+
break;
case SNDRV_PCM_TRIGGER_STOP:
switch (div_id) {
case S3C_I2SV2_DIV_BCLK:
+ if (div > 3) {
+ /* convert value to bit field */
+
+ switch (div) {
+ case 16:
+ div = S3C2412_IISMOD_BCLK_16FS;
+ break;
+
+ case 32:
+ div = S3C2412_IISMOD_BCLK_32FS;
+ break;
+
+ case 24:
+ div = S3C2412_IISMOD_BCLK_24FS;
+ break;
+
+ case 48:
+ div = S3C2412_IISMOD_BCLK_48FS;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ }
+
reg = readl(i2s->regs + S3C2412_IISMOD);
reg &= ~S3C2412_IISMOD_BCLK_MASK;
writel(reg | div, i2s->regs + S3C2412_IISMOD);
}
i2s->iis_pclk = clk_get(dev, "iis");
- if (i2s->iis_pclk == NULL) {
+ if (IS_ERR(i2s->iis_pclk)) {
dev_err(dev, "failed to get iis_clock\n");
iounmap(i2s->regs);
return -ENOENT;
struct snd_soc_dai *dai)
{
u32 ac_glbctrl;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int channel = ((struct s3c24xx_pcm_dma_params *)
+ rtd->dai->cpu_dai->dma_data)->channel;
ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
switch (cmd) {
}
writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+
return 0;
}
int cmd, struct snd_soc_dai *dai)
{
u32 ac_glbctrl;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int channel = ((struct s3c24xx_pcm_dma_params *)
+ rtd->dai->cpu_dai->dma_data)->channel;
ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
switch (cmd) {
}
writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+
return 0;
}
struct snd_soc_dai *dai)
{
int ret = 0;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int channel = ((struct s3c24xx_pcm_dma_params *)
+ rtd->dai->cpu_dai->dma_data)->channel;
pr_debug("Entered %s\n", __func__);
s3c24xx_snd_rxctrl(1);
else
s3c24xx_snd_txctrl(1);
+
+ s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
prtd->state |= ST_RUNNING;
s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START);
- s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED);
break;
case SNDRV_PCM_TRIGGER_STOP:
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);
--- /dev/null
+/* 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");
--- /dev/null
+/* 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
--- /dev/null
+/* 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");
--- /dev/null
+/* 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");
iismod |= S3C64XX_IISMOD_IMS_SYSMUX;
break;
+ case S3C64XX_CLKSRC_CDCLK:
+ switch (dir) {
+ case SND_SOC_CLOCK_IN:
+ iismod |= S3C64XX_IISMOD_CDCLKCON;
+ break;
+ case SND_SOC_CLOCK_OUT:
+ iismod &= ~S3C64XX_IISMOD_CDCLKCON;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+
default:
return -EINVAL;
}
struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai)
{
struct s3c_i2sv2_info *i2s = to_info(dai);
+ u32 iismod = readl(i2s->regs + S3C2412_IISMOD);
- return i2s->iis_cclk;
+ if (iismod & S3C64XX_IISMOD_IMS_SYSMUX)
+ return i2s->iis_cclk;
+ else
+ return i2s->iis_pclk;
}
EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock);
#define S3C64XX_CLKSRC_PCLK (0)
#define S3C64XX_CLKSRC_MUX (1)
+#define S3C64XX_CLKSRC_CDCLK (2)
extern struct snd_soc_dai s3c64xx_i2s_dai[];
--- /dev/null
+/*
+ * smdk64xx_wm8580.c
+ *
+ * Copyright (c) 2009 Samsung Electronics Co. Ltd
+ * Author: Jaswinder Singh <jassi.brar@samsung.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "../codecs/wm8580.h"
+#include "s3c24xx-pcm.h"
+#include "s3c64xx-i2s.h"
+
+#define S3C64XX_I2S_V4 2
+
+/* SMDK64XX has a 12MHZ crystal attached to WM8580 */
+#define SMDK64XX_WM8580_FREQ 12000000
+
+static int smdk64xx_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ unsigned int pll_out;
+ int bfs, rfs, ret;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_U8:
+ case SNDRV_PCM_FORMAT_S8:
+ bfs = 16;
+ break;
+ case SNDRV_PCM_FORMAT_U16_LE:
+ case SNDRV_PCM_FORMAT_S16_LE:
+ bfs = 32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* The Fvco for WM8580 PLLs must fall within [90,100]MHz.
+ * This criterion can't be met if we request PLL output
+ * as {8000x256, 64000x256, 11025x256}Hz.
+ * As a wayout, we rather change rfs to a minimum value that
+ * results in (params_rate(params) * rfs), and itself, acceptable
+ * to both - the CODEC and the CPU.
+ */
+ switch (params_rate(params)) {
+ case 16000:
+ case 22050:
+ case 32000:
+ case 44100:
+ case 48000:
+ case 88200:
+ case 96000:
+ rfs = 256;
+ break;
+ case 64000:
+ rfs = 384;
+ break;
+ case 8000:
+ case 11025:
+ rfs = 512;
+ break;
+ default:
+ return -EINVAL;
+ }
+ pll_out = params_rate(params) * rfs;
+
+ /* Set the Codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* Set the AP DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK,
+ 0, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* We use PCLK for basic ops in SoC-Slave mode */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK,
+ 0, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* Set WM8580 to drive MCLK from it's PLLA */
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK,
+ WM8580_CLKSRC_PLLA);
+ if (ret < 0)
+ return ret;
+
+ /* Explicitly set WM8580-DAC to source from MCLK */
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL,
+ WM8580_CLKSRC_MCLK);
+ if (ret < 0)
+ return ret;
+
+ /* Assuming the CODEC driver evaluates it's rfs too from this call */
+ ret = snd_soc_dai_set_pll(codec_dai, 0, WM8580_PLLA,
+ SMDK64XX_WM8580_FREQ, pll_out);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+/*
+ * SMDK64XX WM8580 DAI operations.
+ */
+static struct snd_soc_ops smdk64xx_ops = {
+ .hw_params = smdk64xx_hw_params,
+};
+
+/* SMDK64xx Playback widgets */
+static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = {
+ SND_SOC_DAPM_HP("Front-L/R", NULL),
+ SND_SOC_DAPM_HP("Center/Sub", NULL),
+ SND_SOC_DAPM_HP("Rear-L/R", NULL),
+};
+
+/* SMDK64xx Capture widgets */
+static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = {
+ SND_SOC_DAPM_MIC("MicIn", NULL),
+ SND_SOC_DAPM_LINE("LineIn", NULL),
+};
+
+/* SMDK-PAIFTX connections */
+static const struct snd_soc_dapm_route audio_map_tx[] = {
+ /* MicIn feeds AINL */
+ {"AINL", NULL, "MicIn"},
+
+ /* LineIn feeds AINL/R */
+ {"AINL", NULL, "LineIn"},
+ {"AINR", NULL, "LineIn"},
+};
+
+/* SMDK-PAIFRX connections */
+static const struct snd_soc_dapm_route audio_map_rx[] = {
+ /* Front Left/Right are fed VOUT1L/R */
+ {"Front-L/R", NULL, "VOUT1L"},
+ {"Front-L/R", NULL, "VOUT1R"},
+
+ /* Center/Sub are fed VOUT2L/R */
+ {"Center/Sub", NULL, "VOUT2L"},
+ {"Center/Sub", NULL, "VOUT2R"},
+
+ /* Rear Left/Right are fed VOUT3L/R */
+ {"Rear-L/R", NULL, "VOUT3L"},
+ {"Rear-L/R", NULL, "VOUT3R"},
+};
+
+static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec)
+{
+ /* Add smdk64xx specific Capture widgets */
+ snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt,
+ ARRAY_SIZE(wm8580_dapm_widgets_cpt));
+
+ /* Set up PAIFTX audio path */
+ snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx));
+
+ /* All enabled by default */
+ snd_soc_dapm_enable_pin(codec, "MicIn");
+ snd_soc_dapm_enable_pin(codec, "LineIn");
+
+ /* signal a DAPM event */
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec)
+{
+ /* Add smdk64xx specific Playback widgets */
+ snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk,
+ ARRAY_SIZE(wm8580_dapm_widgets_pbk));
+
+ /* Set up PAIFRX audio path */
+ snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx));
+
+ /* All enabled by default */
+ snd_soc_dapm_enable_pin(codec, "Front-L/R");
+ snd_soc_dapm_enable_pin(codec, "Center/Sub");
+ snd_soc_dapm_enable_pin(codec, "Rear-L/R");
+
+ /* signal a DAPM event */
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static struct snd_soc_dai_link smdk64xx_dai[] = {
+{ /* Primary Playback i/f */
+ .name = "WM8580 PAIF RX",
+ .stream_name = "Playback",
+ .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4],
+ .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX],
+ .init = smdk64xx_wm8580_init_paifrx,
+ .ops = &smdk64xx_ops,
+},
+{ /* Primary Capture i/f */
+ .name = "WM8580 PAIF TX",
+ .stream_name = "Capture",
+ .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4],
+ .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX],
+ .init = smdk64xx_wm8580_init_paiftx,
+ .ops = &smdk64xx_ops,
+},
+};
+
+static struct snd_soc_card smdk64xx = {
+ .name = "smdk64xx",
+ .platform = &s3c24xx_soc_platform,
+ .dai_link = smdk64xx_dai,
+ .num_links = ARRAY_SIZE(smdk64xx_dai),
+};
+
+static struct snd_soc_device smdk64xx_snd_devdata = {
+ .card = &smdk64xx,
+ .codec_dev = &soc_codec_dev_wm8580,
+};
+
+static struct platform_device *smdk64xx_snd_device;
+
+static int __init smdk64xx_audio_init(void)
+{
+ int ret;
+
+ smdk64xx_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!smdk64xx_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata);
+ smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev;
+ ret = platform_device_add(smdk64xx_snd_device);
+
+ if (ret)
+ platform_device_put(smdk64xx_snd_device);
+
+ return ret;
+}
+module_init(smdk64xx_audio_init);
+
+MODULE_AUTHOR("Jaswinder Singh, jassi.brar@samsung.com");
+MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580");
+MODULE_LICENSE("GPL");
#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>
/* s6105 audio private data */
static struct aic3x_setup_data s6105_aic3x_setup = {
- .i2c_bus = 0,
- .i2c_address = 0x18,
};
/* s6105 audio subsystem */
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;
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
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
## 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
--- /dev/null
+/*
+ * 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>");
--- /dev/null
+/*
+ * 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>");
#define snd_soc_7_9_spi_write NULL
#endif
+static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 *cache = codec->reg_cache;
+ u8 data[2];
+
+ BUG_ON(codec->volatile_register);
+
+ data[0] = reg & 0xff;
+ data[1] = value & 0xff;
+
+ if (reg < codec->reg_cache_size)
+ cache[reg] = value;
+
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 *cache = codec->reg_cache;
+ if (reg >= codec->reg_cache_size)
+ return -1;
+ return cache[reg];
+}
+
static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
unsigned int (*read)(struct snd_soc_codec *, unsigned int);
unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
} io_types[] = {
- { 7, 9, snd_soc_7_9_write, snd_soc_7_9_spi_write, snd_soc_7_9_read },
- { 8, 16, snd_soc_8_16_write, NULL, snd_soc_8_16_read,
- snd_soc_8_16_read_i2c },
+ {
+ .addr_bits = 7, .data_bits = 9,
+ .write = snd_soc_7_9_write, .read = snd_soc_7_9_read,
+ .spi_write = snd_soc_7_9_spi_write
+ },
+ {
+ .addr_bits = 8, .data_bits = 8,
+ .write = snd_soc_8_8_write, .read = snd_soc_8_8_read,
+ },
+ {
+ .addr_bits = 8, .data_bits = 16,
+ .write = snd_soc_8_16_write, .read = snd_soc_8_16_read,
+ .i2c_read = snd_soc_8_16_read_i2c,
+ },
};
/**
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);
}
* snd_soc_dai_set_pll - configure DAI PLL.
* @dai: DAI
* @pll_id: DAI specific PLL ID
+ * @source: DAI specific source for the PLL
* @freq_in: PLL input clock frequency in Hz
* @freq_out: requested PLL output clock frequency in Hz
*
* Configures and enables PLL to generate output clock based on input clock.
*/
-int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
+ unsigned int freq_in, unsigned int freq_out)
{
if (dai->ops && dai->ops->set_pll)
- return dai->ops->set_pll(dai, pll_id, freq_in, freq_out);
+ return dai->ops->set_pll(dai, pll_id, source,
+ freq_in, freq_out);
else
return -EINVAL;
}
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
+/**
+ * snd_soc_dai_set_channel_map - configure DAI audio channel map
+ * @dai: DAI
+ * @tx_num: how many TX channels
+ * @tx_slot: pointer to an array which imply the TX slot number channel
+ * 0~num-1 uses
+ * @rx_num: how many RX channels
+ * @rx_slot: pointer to an array which imply the RX slot number channel
+ * 0~num-1 uses
+ *
+ * configure the relationship between channel number and TDM slot number.
+ */
+int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot)
+{
+ if (dai->ops && dai->ops->set_channel_map)
+ return dai->ops->set_channel_map(dai, tx_num, tx_slot,
+ rx_num, rx_slot);
+ else
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map);
+
/**
* snd_soc_dai_set_tristate - configure DAI system or master clock.
* @dai: DAI
#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>
[snd_soc_dapm_pre] = 0,
[snd_soc_dapm_supply] = 1,
[snd_soc_dapm_micbias] = 2,
- [snd_soc_dapm_mic] = 3,
- [snd_soc_dapm_mux] = 4,
- [snd_soc_dapm_value_mux] = 4,
- [snd_soc_dapm_dac] = 5,
- [snd_soc_dapm_mixer] = 6,
- [snd_soc_dapm_mixer_named_ctl] = 6,
- [snd_soc_dapm_pga] = 7,
- [snd_soc_dapm_adc] = 8,
- [snd_soc_dapm_hp] = 9,
- [snd_soc_dapm_spk] = 9,
- [snd_soc_dapm_post] = 10,
+ [snd_soc_dapm_aif_in] = 3,
+ [snd_soc_dapm_aif_out] = 3,
+ [snd_soc_dapm_mic] = 4,
+ [snd_soc_dapm_mux] = 5,
+ [snd_soc_dapm_value_mux] = 5,
+ [snd_soc_dapm_dac] = 6,
+ [snd_soc_dapm_mixer] = 7,
+ [snd_soc_dapm_mixer_named_ctl] = 7,
+ [snd_soc_dapm_pga] = 8,
+ [snd_soc_dapm_adc] = 9,
+ [snd_soc_dapm_hp] = 10,
+ [snd_soc_dapm_spk] = 10,
+ [snd_soc_dapm_post] = 11,
};
static int dapm_down_seq[] = {
[snd_soc_dapm_micbias] = 8,
[snd_soc_dapm_mux] = 9,
[snd_soc_dapm_value_mux] = 9,
- [snd_soc_dapm_supply] = 10,
- [snd_soc_dapm_post] = 11,
+ [snd_soc_dapm_aif_in] = 10,
+ [snd_soc_dapm_aif_out] = 10,
+ [snd_soc_dapm_supply] = 11,
+ [snd_soc_dapm_post] = 12,
};
static void pop_wait(u32 pop_time)
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;
}
case snd_soc_dapm_micbias:
case snd_soc_dapm_vmid:
case snd_soc_dapm_supply:
+ case snd_soc_dapm_aif_in:
+ case snd_soc_dapm_aif_out:
p->connect = 1;
break;
/* does effect routing - dynamically connected */
if (widget->id == snd_soc_dapm_supply)
return 0;
- if (widget->id == snd_soc_dapm_adc && widget->active)
- return 1;
+ switch (widget->id) {
+ case snd_soc_dapm_adc:
+ case snd_soc_dapm_aif_out:
+ if (widget->active)
+ return 1;
+ default:
+ break;
+ }
if (widget->connected) {
/* connected pin ? */
return 0;
/* active stream ? */
- if (widget->id == snd_soc_dapm_dac && widget->active)
- return 1;
+ switch (widget->id) {
+ case snd_soc_dapm_dac:
+ case snd_soc_dapm_aif_in:
+ if (widget->active)
+ return 1;
+ default:
+ break;
+ }
if (widget->connected) {
/* connected pin ? */
/* Check if one of our outputs is connected */
list_for_each_entry(path, &w->sinks, list_source) {
+ if (path->connected &&
+ !path->connected(path->source, path->sink))
+ continue;
+
if (path->sink && path->sink->power_check &&
path->sink->power_check(path->sink)) {
power = 1;
case snd_soc_dapm_mixer:
case snd_soc_dapm_mixer_named_ctl:
case snd_soc_dapm_supply:
+ case snd_soc_dapm_aif_in:
+ case snd_soc_dapm_aif_out:
if (w->name) {
in = is_connected_input_ep(w);
dapm_clear_walk(w->codec);
}
#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->sname)
+ ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
+ w->sname,
+ w->active ? "active" : "inactive");
+
+ list_for_each_entry(p, &w->sources, list_sink) {
+ if (p->connected && !p->connected(w, p->sink))
+ continue;
+
+ 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->connected && !p->connected(w, p->sink))
+ continue;
+
+ 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,
EXPORT_SYMBOL_GPL(snd_soc_dapm_sync);
static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
- const char *sink, const char *control, const char *source)
+ const struct snd_soc_dapm_route *route)
{
struct snd_soc_dapm_path *path;
struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w;
+ const char *sink = route->sink;
+ const char *control = route->control;
+ const char *source = route->source;
int ret = 0;
/* find src and dest widgets */
path->source = wsource;
path->sink = wsink;
+ path->connected = route->connected;
INIT_LIST_HEAD(&path->list);
INIT_LIST_HEAD(&path->list_source);
INIT_LIST_HEAD(&path->list_sink);
case snd_soc_dapm_pre:
case snd_soc_dapm_post:
case snd_soc_dapm_supply:
+ case snd_soc_dapm_aif_in:
+ case snd_soc_dapm_aif_out:
list_add(&path->list, &codec->dapm_paths);
list_add(&path->list_sink, &wsink->sources);
list_add(&path->list_source, &wsource->sinks);
int i, ret;
for (i = 0; i < num; i++) {
- ret = snd_soc_dapm_add_route(codec, route->sink,
- route->control, route->source);
+ ret = snd_soc_dapm_add_route(codec, route);
if (ret < 0) {
printk(KERN_ERR "Failed to add route %s->%s\n",
route->source,
dapm_new_mux(codec, w);
break;
case snd_soc_dapm_adc:
+ case snd_soc_dapm_aif_out:
w->power_check = dapm_adc_check_power;
break;
case snd_soc_dapm_dac:
+ case snd_soc_dapm_aif_in:
w->power_check = dapm_dac_check_power;
break;
case snd_soc_dapm_pga:
git.openpandora.org / pandora-kernel.git / commitdiff