ASoC: Merge dai_ops factor out
Merge Eric Maio's patch to merge snd_soc_dai_ops out of line. Fixed
merge issues and updated drivers, plus an issue with the ops for the two
s3c2443 AC97 DAIs having been merged.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
diff --git a/arch/arm/mach-pxa/include/mach/regs-ssp.h b/arch/arm/mach-pxa/include/mach/regs-ssp.h
index 3c04cde..f43905a 100644
--- a/arch/arm/mach-pxa/include/mach/regs-ssp.h
+++ b/arch/arm/mach-pxa/include/mach/regs-ssp.h
@@ -47,7 +47,7 @@
#define SSCR0_TUM (1 << 23) /* Transmit FIFO underrun interrupt mask */
#define SSCR0_FRDC (0x07000000) /* Frame rate divider control (mask) */
#define SSCR0_SlotsPerFrm(x) (((x) - 1) << 24) /* Time slots per frame [1..8] */
-#define SSCR0_ADC (1 << 30) /* Audio clock select */
+#define SSCR0_ACS (1 << 30) /* Audio clock select */
#define SSCR0_MOD (1 << 31) /* Mode (normal or network) */
#endif
@@ -106,6 +106,11 @@
#define SSSR_TINT (1 << 19) /* Receiver Time-out Interrupt */
#define SSSR_PINT (1 << 18) /* Peripheral Trailing Byte Interrupt */
+#if defined(CONFIG_PXA3xx)
+#define SSPSP_EDMYSTOP(x) ((x) << 28) /* Extended Dummy Stop */
+#define SSPSP_EDMYSTRT(x) ((x) << 26) /* Extended Dummy Start */
+#endif
+
#define SSPSP_FSRT (1 << 25) /* Frame Sync Relative Timing */
#define SSPSP_DMYSTOP(x) ((x) << 23) /* Dummy Stop */
#define SSPSP_SFRMWDTH(x) ((x) << 16) /* Serial Frame Width */
diff --git a/arch/arm/mach-s3c2410/dma.c b/arch/arm/mach-s3c2410/dma.c
index 552b4c7..440c014 100644
--- a/arch/arm/mach-s3c2410/dma.c
+++ b/arch/arm/mach-s3c2410/dma.c
@@ -28,7 +28,7 @@
#include <mach/regs-mem.h>
#include <mach/regs-lcd.h>
#include <mach/regs-sdi.h>
-#include <asm/plat-s3c24xx/regs-iis.h>
+#include <plat/regs-iis.h>
#include <plat/regs-spi.h>
static struct s3c24xx_dma_map __initdata s3c2410_dma_mappings[] = {
diff --git a/arch/arm/mach-s3c2410/include/mach/io.h b/arch/arm/mach-s3c2410/include/mach/io.h
index 9813dbf..c477771 100644
--- a/arch/arm/mach-s3c2410/include/mach/io.h
+++ b/arch/arm/mach-s3c2410/include/mach/io.h
@@ -9,7 +9,7 @@
#ifndef __ASM_ARM_ARCH_IO_H
#define __ASM_ARM_ARCH_IO_H
-#include <mach/hardware.h>
+#include <mach/map.h>
#define IO_SPACE_LIMIT 0xffffffff
diff --git a/arch/arm/mach-s3c2412/dma.c b/arch/arm/mach-s3c2412/dma.c
index 919856c..9e34785 100644
--- a/arch/arm/mach-s3c2412/dma.c
+++ b/arch/arm/mach-s3c2412/dma.c
@@ -29,8 +29,8 @@
#include <mach/regs-mem.h>
#include <mach/regs-lcd.h>
#include <mach/regs-sdi.h>
-#include <asm/plat-s3c24xx/regs-s3c2412-iis.h>
-#include <asm/plat-s3c24xx/regs-iis.h>
+#include <plat/regs-s3c2412-iis.h>
+#include <plat/regs-iis.h>
#include <plat/regs-spi.h>
#define MAP(x) { (x)| DMA_CH_VALID, (x)| DMA_CH_VALID, (x)| DMA_CH_VALID, (x)| DMA_CH_VALID }
diff --git a/arch/arm/mach-s3c2440/dma.c b/arch/arm/mach-s3c2440/dma.c
index 5b5ee0b..69b6cf3 100644
--- a/arch/arm/mach-s3c2440/dma.c
+++ b/arch/arm/mach-s3c2440/dma.c
@@ -28,7 +28,7 @@
#include <mach/regs-mem.h>
#include <mach/regs-lcd.h>
#include <mach/regs-sdi.h>
-#include <asm/plat-s3c24xx/regs-iis.h>
+#include <plat/regs-iis.h>
#include <plat/regs-spi.h>
static struct s3c24xx_dma_map __initdata s3c2440_dma_mappings[] = {
diff --git a/arch/arm/mach-s3c2443/dma.c b/arch/arm/mach-s3c2443/dma.c
index 2a58a4d..8430e58 100644
--- a/arch/arm/mach-s3c2443/dma.c
+++ b/arch/arm/mach-s3c2443/dma.c
@@ -29,7 +29,7 @@
#include <mach/regs-mem.h>
#include <mach/regs-lcd.h>
#include <mach/regs-sdi.h>
-#include <asm/plat-s3c24xx/regs-iis.h>
+#include <plat/regs-iis.h>
#include <plat/regs-spi.h>
#define MAP(x) { \
diff --git a/arch/arm/mach-s3c2410/include/mach/audio.h b/arch/arm/plat-s3c/include/plat/audio.h
similarity index 100%
rename from arch/arm/mach-s3c2410/include/mach/audio.h
rename to arch/arm/plat-s3c/include/plat/audio.h
diff --git a/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
similarity index 93%
rename from include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h
rename to arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
index 25d4058..0fad757 100644
--- a/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h
+++ b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
@@ -33,6 +33,9 @@
#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1)
#define S3C2412_IISCON_IIS_ACTIVE (1 << 0)
+#define S3C64XX_IISMOD_IMS_PCLK (0 << 10)
+#define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10)
+
#define S3C2412_IISMOD_MASTER_INTERNAL (0 << 10)
#define S3C2412_IISMOD_MASTER_EXTERNAL (1 << 10)
#define S3C2412_IISMOD_SLAVE (2 << 10)
@@ -44,8 +47,8 @@
#define S3C2412_IISMOD_LR_LLOW (0 << 7)
#define S3C2412_IISMOD_LR_RLOW (1 << 7)
#define S3C2412_IISMOD_SDF_IIS (0 << 5)
-#define S3C2412_IISMOD_SDF_MSB (0 << 5)
-#define S3C2412_IISMOD_SDF_LSB (0 << 5)
+#define S3C2412_IISMOD_SDF_MSB (1 << 5)
+#define S3C2412_IISMOD_SDF_LSB (2 << 5)
#define S3C2412_IISMOD_SDF_MASK (3 << 5)
#define S3C2412_IISMOD_RCLK_256FS (0 << 3)
#define S3C2412_IISMOD_RCLK_512FS (1 << 3)
diff --git a/arch/arm/plat-s3c24xx/clock-dclk.c b/arch/arm/plat-s3c24xx/clock-dclk.c
index 5b75a79..35219dc 100644
--- a/arch/arm/plat-s3c24xx/clock-dclk.c
+++ b/arch/arm/plat-s3c24xx/clock-dclk.c
@@ -18,6 +18,7 @@
#include <mach/regs-clock.h>
#include <mach/regs-gpio.h>
+#include <mach/hardware.h>
#include <plat/clock.h>
#include <plat/cpu.h>
diff --git a/include/asm-arm/plat-s3c24xx/regs-iis.h b/arch/arm/plat-s3c24xx/include/plat/regs-iis.h
similarity index 100%
rename from include/asm-arm/plat-s3c24xx/regs-iis.h
rename to arch/arm/plat-s3c24xx/include/plat/regs-iis.h
diff --git a/include/linux/mfd/wm8400-audio.h b/include/linux/mfd/wm8400-audio.h
index b6640e0..e06ed3e 100644
--- a/include/linux/mfd/wm8400-audio.h
+++ b/include/linux/mfd/wm8400-audio.h
@@ -1181,6 +1181,7 @@
#define WM8400_FLL_OUTDIV_SHIFT 0 /* FLL_OUTDIV - [2:0] */
#define WM8400_FLL_OUTDIV_WIDTH 3 /* FLL_OUTDIV - [2:0] */
+struct wm8400;
void wm8400_reset_codec_reg_cache(struct wm8400 *wm8400);
#endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 0e77352..a40bc6f 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -16,6 +16,8 @@
#include <linux/platform_device.h>
#include <linux/types.h>
#include <linux/workqueue.h>
+#include <linux/interrupt.h>
+#include <linux/kernel.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/control.h>
@@ -168,6 +170,9 @@
struct snd_soc_ac97_ops;
struct snd_soc_jack;
struct snd_soc_jack_pin;
+#ifdef CONFIG_GPIOLIB
+struct snd_soc_jack_gpio;
+#endif
typedef int (*hw_write_t)(void *,const char* ,int);
typedef int (*hw_read_t)(void *,char* ,int);
@@ -194,6 +199,12 @@
void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask);
int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
struct snd_soc_jack_pin *pins);
+#ifdef CONFIG_GPIOLIB
+int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
+ struct snd_soc_jack_gpio *gpios);
+void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
+ struct snd_soc_jack_gpio *gpios);
+#endif
/* codec IO */
#define snd_soc_read(codec, reg) codec->read(codec, reg)
@@ -264,6 +275,27 @@
bool invert;
};
+/**
+ * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection
+ *
+ * @gpio: gpio number
+ * @name: gpio name
+ * @report: value to report when jack detected
+ * @invert: report presence in low state
+ * @debouce_time: debouce time in ms
+ */
+#ifdef CONFIG_GPIOLIB
+struct snd_soc_jack_gpio {
+ unsigned int gpio;
+ const char *name;
+ int report;
+ int invert;
+ int debounce_time;
+ struct snd_soc_jack *jack;
+ struct work_struct work;
+};
+#endif
+
struct snd_soc_jack {
struct snd_jack *jack;
struct snd_soc_card *card;
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 43dd8ce..7065753 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -164,41 +164,41 @@
*/
switch (params_rate(params)) {
case 48000:
- pll_out = 12288000;
- mclk_div = WM8510_MCLKDIV_1;
- bclk = WM8510_BCLKDIV_8;
- break;
-
- case 44100:
- pll_out = 11289600;
- mclk_div = WM8510_MCLKDIV_1;
- bclk = WM8510_BCLKDIV_8;
- break;
-
- case 22050:
- pll_out = 11289600;
+ pll_out = 24576000;
mclk_div = WM8510_MCLKDIV_2;
bclk = WM8510_BCLKDIV_8;
break;
- case 16000:
- pll_out = 12288000;
- mclk_div = WM8510_MCLKDIV_3;
+ case 44100:
+ pll_out = 22579200;
+ mclk_div = WM8510_MCLKDIV_2;
bclk = WM8510_BCLKDIV_8;
break;
- case 11025:
- pll_out = 11289600;
+ case 22050:
+ pll_out = 22579200;
mclk_div = WM8510_MCLKDIV_4;
bclk = WM8510_BCLKDIV_8;
break;
- case 8000:
- pll_out = 12288000;
+ case 16000:
+ pll_out = 24576000;
mclk_div = WM8510_MCLKDIV_6;
bclk = WM8510_BCLKDIV_8;
break;
+ case 11025:
+ pll_out = 22579200;
+ mclk_div = WM8510_MCLKDIV_8;
+ bclk = WM8510_BCLKDIV_8;
+ break;
+
+ case 8000:
+ pll_out = 24576000;
+ mclk_div = WM8510_MCLKDIV_12;
+ bclk = WM8510_BCLKDIV_8;
+ break;
+
default:
pr_warning("playpaq_wm8510: Unsupported sample rate %d\n",
params_rate(params));
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 5885702..8a935f2 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -357,8 +357,8 @@
sport_err:
#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
gpio_err:
+#endif
peripheral_free_list(sport_req[sport_num]);
peripheral_err:
free_page((unsigned long)cmd_count);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 628a591..b6c7f7a 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -14,6 +14,7 @@
select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
select SND_SOC_AD1980 if SND_SOC_AC97_BUS
select SND_SOC_AD73311 if I2C
+ select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
select SND_SOC_CS4270 if I2C
select SND_SOC_PCM3008
@@ -25,6 +26,7 @@
select SND_SOC_UDA134X
select SND_SOC_UDA1380 if I2C
select SND_SOC_WM8350 if MFD_WM8350
+ select SND_SOC_WM8400 if MFD_WM8400
select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8580 if I2C
select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI
@@ -60,6 +62,9 @@
config SND_SOC_AD73311
tristate
+config SND_SOC_AK4104
+ tristate
+
config SND_SOC_AK4535
tristate
@@ -106,6 +111,9 @@
config SND_SOC_WM8350
tristate
+config SND_SOC_WM8400
+ tristate
+
config SND_SOC_WM8510
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 3664cdc..030d245 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,6 +1,7 @@
snd-soc-ac97-objs := ac97.o
snd-soc-ad1980-objs := ad1980.o
snd-soc-ad73311-objs := ad73311.o
+snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-l3-objs := l3.o
@@ -13,6 +14,7 @@
snd-soc-uda134x-objs := uda134x.o
snd-soc-uda1380-objs := uda1380.o
snd-soc-wm8350-objs := wm8350.o
+snd-soc-wm8400-objs := wm8400.o
snd-soc-wm8510-objs := wm8510.o
snd-soc-wm8580-objs := wm8580.o
snd-soc-wm8728-objs := wm8728.o
@@ -30,6 +32,7 @@
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.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_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
@@ -42,6 +45,7 @@
obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o
obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o
+obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o
obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o
obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o
obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o
diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h
index 507ce0c..569573d 100644
--- a/sound/soc/codecs/ad73311.h
+++ b/sound/soc/codecs/ad73311.h
@@ -70,7 +70,7 @@
#define REGD_IGS(x) (x & 0x7)
#define REGD_RMOD (1 << 3)
#define REGD_OGS(x) ((x & 0x7) << 4)
-#define REGD_MUTE (x << 7)
+#define REGD_MUTE (1 << 7)
/* Control register E */
#define CTRL_REG_E (4 << 8)
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
new file mode 100644
index 0000000..4d47bc4
--- /dev/null
+++ b/sound/soc/codecs/ak4104.c
@@ -0,0 +1,365 @@
+/*
+ * AK4104 ALSA SoC (ASoC) driver
+ *
+ * Copyright (c) 2009 Daniel Mack <daniel@caiaq.de>
+ *
+ * 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 <sound/core.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <linux/spi/spi.h>
+#include <sound/asoundef.h>
+
+#include "ak4104.h"
+
+/* AK4104 registers addresses */
+#define AK4104_REG_CONTROL1 0x00
+#define AK4104_REG_RESERVED 0x01
+#define AK4104_REG_CONTROL2 0x02
+#define AK4104_REG_TX 0x03
+#define AK4104_REG_CHN_STATUS(x) ((x) + 0x04)
+#define AK4104_NUM_REGS 10
+
+#define AK4104_REG_MASK 0x1f
+#define AK4104_READ 0xc0
+#define AK4104_WRITE 0xe0
+#define AK4104_RESERVED_VAL 0x5b
+
+/* Bit masks for AK4104 registers */
+#define AK4104_CONTROL1_RSTN (1 << 0)
+#define AK4104_CONTROL1_PW (1 << 1)
+#define AK4104_CONTROL1_DIF0 (1 << 2)
+#define AK4104_CONTROL1_DIF1 (1 << 3)
+
+#define AK4104_CONTROL2_SEL0 (1 << 0)
+#define AK4104_CONTROL2_SEL1 (1 << 1)
+#define AK4104_CONTROL2_MODE (1 << 2)
+
+#define AK4104_TX_TXE (1 << 0)
+#define AK4104_TX_V (1 << 1)
+
+#define DRV_NAME "ak4104"
+
+struct ak4104_private {
+ struct snd_soc_codec codec;
+ u8 reg_cache[AK4104_NUM_REGS];
+};
+
+static int ak4104_fill_cache(struct snd_soc_codec *codec)
+{
+ int i;
+ u8 *reg_cache = codec->reg_cache;
+ struct spi_device *spi = codec->control_data;
+
+ for (i = 0; i < codec->reg_cache_size; i++) {
+ int ret = spi_w8r8(spi, i | AK4104_READ);
+ if (ret < 0) {
+ dev_err(&spi->dev, "SPI write failure\n");
+ return ret;
+ }
+
+ reg_cache[i] = ret;
+ }
+
+ return 0;
+}
+
+static unsigned int ak4104_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 *reg_cache = codec->reg_cache;
+
+ if (reg >= codec->reg_cache_size)
+ return -EINVAL;
+
+ return reg_cache[reg];
+}
+
+static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 *cache = codec->reg_cache;
+ struct spi_device *spi = codec->control_data;
+
+ if (reg >= codec->reg_cache_size)
+ return -EINVAL;
+
+ reg &= AK4104_REG_MASK;
+ reg |= AK4104_WRITE;
+
+ /* only write to the hardware if value has changed */
+ if (cache[reg] != value) {
+ u8 tmp[2] = { reg, value };
+ if (spi_write(spi, tmp, sizeof(tmp))) {
+ dev_err(&spi->dev, "SPI write failed\n");
+ return -EIO;
+ }
+
+ cache[reg] = value;
+ }
+
+ return 0;
+}
+
+static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int format)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int val = 0;
+
+ val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1);
+ if (val < 0)
+ return val;
+
+ val &= ~(AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1);
+
+ /* set DAI format */
+ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ val |= AK4104_CONTROL1_DIF0;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ val |= AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1;
+ break;
+ default:
+ dev_err(codec->dev, "invalid dai format\n");
+ return -EINVAL;
+ }
+
+ /* This device can only be slave */
+ if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+ return -EINVAL;
+
+ return ak4104_spi_write(codec, AK4104_REG_CONTROL1, val);
+}
+
+static int ak4104_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ int val = 0;
+
+ /* set the IEC958 bits: consumer mode, no copyright bit */
+ val |= IEC958_AES0_CON_NOT_COPYRIGHT;
+ ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(0), val);
+
+ val = 0;
+
+ switch (params_rate(params)) {
+ case 44100:
+ val |= IEC958_AES3_CON_FS_44100;
+ break;
+ case 48000:
+ val |= IEC958_AES3_CON_FS_48000;
+ break;
+ case 32000:
+ val |= IEC958_AES3_CON_FS_32000;
+ break;
+ default:
+ dev_err(codec->dev, "unsupported sampling rate\n");
+ return -EINVAL;
+ }
+
+ return ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(3), val);
+}
+
+static struct snd_soc_dai_ops ak4101_dai_ops = {
+ .hw_params = ak4104_hw_params,
+ .set_fmt = ak4104_set_dai_fmt,
+};
+
+struct snd_soc_dai ak4104_dai = {
+ .name = DRV_NAME,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_32000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S24_LE
+ },
+ .ops = &ak4101_dai_ops,
+};
+
+static struct snd_soc_codec *ak4104_codec;
+
+static int ak4104_spi_probe(struct spi_device *spi)
+{
+ struct snd_soc_codec *codec;
+ struct ak4104_private *ak4104;
+ int ret, val;
+
+ spi->bits_per_word = 8;
+ spi->mode = SPI_MODE_0;
+ ret = spi_setup(spi);
+ if (ret < 0)
+ return ret;
+
+ ak4104 = kzalloc(sizeof(struct ak4104_private), GFP_KERNEL);
+ if (!ak4104) {
+ dev_err(&spi->dev, "could not allocate codec\n");
+ return -ENOMEM;
+ }
+
+ codec = &ak4104->codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->dev = &spi->dev;
+ codec->name = DRV_NAME;
+ codec->owner = THIS_MODULE;
+ codec->dai = &ak4104_dai;
+ codec->num_dai = 1;
+ codec->private_data = ak4104;
+ codec->control_data = spi;
+ codec->reg_cache = ak4104->reg_cache;
+ codec->reg_cache_size = AK4104_NUM_REGS;
+
+ /* read all regs and fill the cache */
+ ret = ak4104_fill_cache(codec);
+ if (ret < 0) {
+ dev_err(&spi->dev, "failed to fill register cache\n");
+ return ret;
+ }
+
+ /* read the 'reserved' register - according to the datasheet, it
+ * should contain 0x5b. Not a good way to verify the presence of
+ * the device, but there is no hardware ID register. */
+ if (ak4104_read_reg_cache(codec, AK4104_REG_RESERVED) !=
+ AK4104_RESERVED_VAL) {
+ ret = -ENODEV;
+ goto error_free_codec;
+ }
+
+ /* set power-up and non-reset bits */
+ val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1);
+ val |= AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN;
+ ret = ak4104_spi_write(codec, AK4104_REG_CONTROL1, val);
+ if (ret < 0)
+ goto error_free_codec;
+
+ /* enable transmitter */
+ val = ak4104_read_reg_cache(codec, AK4104_REG_TX);
+ val |= AK4104_TX_TXE;
+ ret = ak4104_spi_write(codec, AK4104_REG_TX, val);
+ if (ret < 0)
+ goto error_free_codec;
+
+ ak4104_codec = codec;
+ ret = snd_soc_register_dai(&ak4104_dai);
+ if (ret < 0) {
+ dev_err(&spi->dev, "failed to register DAI\n");
+ goto error_free_codec;
+ }
+
+ spi_set_drvdata(spi, ak4104);
+ dev_info(&spi->dev, "SPI device initialized\n");
+ return 0;
+
+error_free_codec:
+ kfree(ak4104);
+ ak4104_dai.dev = NULL;
+ return ret;
+}
+
+static int __devexit ak4104_spi_remove(struct spi_device *spi)
+{
+ int ret, val;
+ struct ak4104_private *ak4104 = spi_get_drvdata(spi);
+
+ val = ak4104_read_reg_cache(&ak4104->codec, AK4104_REG_CONTROL1);
+ if (val < 0)
+ return val;
+
+ /* clear power-up and non-reset bits */
+ val &= ~(AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN);
+ ret = ak4104_spi_write(&ak4104->codec, AK4104_REG_CONTROL1, val);
+ if (ret < 0)
+ return ret;
+
+ ak4104_codec = NULL;
+ kfree(ak4104);
+ return 0;
+}
+
+static int ak4104_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = ak4104_codec;
+ int ret;
+
+ /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */
+ socdev->card->codec = 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\n");
+ return ret;
+ }
+
+ /* Register the socdev */
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card\n");
+ snd_soc_free_pcms(socdev);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int ak4104_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ snd_soc_free_pcms(socdev);
+ return 0;
+};
+
+struct snd_soc_codec_device soc_codec_device_ak4104 = {
+ .probe = ak4104_probe,
+ .remove = ak4104_remove
+};
+EXPORT_SYMBOL_GPL(soc_codec_device_ak4104);
+
+static struct spi_driver ak4104_spi_driver = {
+ .driver = {
+ .name = DRV_NAME,
+ .owner = THIS_MODULE,
+ },
+ .probe = ak4104_spi_probe,
+ .remove = __devexit_p(ak4104_spi_remove),
+};
+
+static int __init ak4104_init(void)
+{
+ pr_info("Asahi Kasei AK4104 ALSA SoC Codec Driver\n");
+ return spi_register_driver(&ak4104_spi_driver);
+}
+module_init(ak4104_init);
+
+static void __exit ak4104_exit(void)
+{
+ spi_unregister_driver(&ak4104_spi_driver);
+}
+module_exit(ak4104_exit);
+
+MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
+MODULE_DESCRIPTION("Asahi Kasei AK4104 ALSA SoC driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/codecs/ak4104.h b/sound/soc/codecs/ak4104.h
new file mode 100644
index 0000000..eb88fe7
--- /dev/null
+++ b/sound/soc/codecs/ak4104.h
@@ -0,0 +1,7 @@
+#ifndef _AK4104_H
+#define _AK4104_H
+
+extern struct snd_soc_dai ak4104_dai;
+extern struct snd_soc_codec_device soc_codec_device_ak4104;
+
+#endif
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 7ae3d65..2137670 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -12,14 +12,13 @@
*
* Current features/limitations:
*
- * 1) Software mode is supported. Stand-alone mode is not supported.
- * 2) Only I2C is supported, not SPI
- * 3) Only Master mode is supported, not Slave.
- * 4) The machine driver's 'startup' function must call
- * cs4270_set_dai_sysclk() with the value of MCLK.
- * 5) Only I2S and left-justified modes are supported
- * 6) Power management is not supported
- * 7) The only supported control is volume and hardware mute (if enabled)
+ * - Software mode is supported. Stand-alone mode is not supported.
+ * - Only I2C is supported, not SPI
+ * - Support for master and slave mode
+ * - The machine driver's 'startup' function must call
+ * cs4270_set_dai_sysclk() with the value of MCLK.
+ * - Only I2S and left-justified modes are supported
+ * - Power management is not supported
*/
#include <linux/module.h>
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 535d8ce..86bb15c 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -584,12 +584,11 @@
/* Save the current volume */
hs_gain = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_GAIN_SET);
+ hs_pop = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_POPN_SET);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
/* Do the anti-pop/bias ramp enable according to the TRM */
- hs_pop = TWL4030_RAMP_DELAY_645MS;
- twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
hs_pop |= TWL4030_VMID_EN;
twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
/* Is this needed? Can we just use whatever gain here? */
@@ -603,8 +602,6 @@
break;
case SND_SOC_DAPM_POST_PMD:
/* Do the anti-pop/bias ramp disable according to the TRM */
- hs_pop = twl4030_read_reg_cache(w->codec,
- TWL4030_REG_HS_POPN_SET);
hs_pop &= ~TWL4030_RAMP_EN;
twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
/* Bypass the reg_cache to mute the headset */
@@ -847,6 +844,17 @@
*/
static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0);
+static const char *twl4030_rampdelay_texts[] = {
+ "27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms",
+ "437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms",
+ "3495/2581/1748 ms"
+};
+
+static const struct soc_enum twl4030_rampdelay_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_HS_POPN_SET, 2,
+ ARRAY_SIZE(twl4030_rampdelay_texts),
+ twl4030_rampdelay_texts);
+
static const struct snd_kcontrol_new twl4030_snd_controls[] = {
/* Common playback gain controls */
SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume",
@@ -901,6 +909,8 @@
SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN,
0, 3, 5, 0, input_gain_tlv),
+
+ SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum),
};
static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index cafa768..5b21594 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -25,6 +25,7 @@
#include <linux/ioctl.h>
#include <linux/delay.h>
#include <linux/i2c.h>
+#include <linux/workqueue.h>
#include <sound/core.h>
#include <sound/control.h>
#include <sound/initval.h>
@@ -35,7 +36,8 @@
#include "uda1380.h"
-#define UDA1380_VERSION "0.6"
+static struct work_struct uda1380_work;
+static struct snd_soc_codec *uda1380_codec;
/*
* uda1380 register cache
@@ -52,6 +54,8 @@
0x0000, 0x8000, 0x0002, 0x0000,
};
+static unsigned long uda1380_cache_dirty;
+
/*
* read uda1380 register cache
*/
@@ -73,8 +77,11 @@
u16 reg, unsigned int value)
{
u16 *cache = codec->reg_cache;
+
if (reg >= UDA1380_CACHEREGNUM)
return;
+ if ((reg >= 0x10) && (cache[reg] != value))
+ set_bit(reg - 0x10, &uda1380_cache_dirty);
cache[reg] = value;
}
@@ -113,6 +120,8 @@
(data[0]<<8) | data[1]);
return -EIO;
}
+ if (reg >= 0x10)
+ clear_bit(reg - 0x10, &uda1380_cache_dirty);
return 0;
} else
return -EIO;
@@ -120,6 +129,20 @@
#define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0)
+static void uda1380_flush_work(struct work_struct *work)
+{
+ int bit, reg;
+
+ for_each_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) {
+ reg = 0x10 + bit;
+ pr_debug("uda1380: flush reg %x val %x:\n", reg,
+ uda1380_read_reg_cache(uda1380_codec, reg));
+ uda1380_write(uda1380_codec, reg,
+ uda1380_read_reg_cache(uda1380_codec, reg));
+ clear_bit(bit, &uda1380_cache_dirty);
+ }
+}
+
/* declarations of ALSA reg_elem_REAL controls */
static const char *uda1380_deemp[] = {
"None",
@@ -254,7 +277,6 @@
SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0), /* DA_POL_INV */
SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum), /* SEL_NS */
SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum), /* MIX_POS, MIX */
- SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0), /* SILENCE, force DAC output to silence */
SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0), /* SDET_ON */
SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum), /* SD_VALUE */
SOC_ENUM("Oversampling Input", uda1380_os_enum), /* OS */
@@ -377,8 +399,9 @@
iface |= R01_SFORI_MSB | R01_SFORO_MSB;
}
- if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
- iface |= R01_SIM;
+ /* DATAI is slave only, so in single-link mode, this has to be slave */
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+ return -EINVAL;
uda1380_write(codec, UDA1380_IFACE, iface);
@@ -406,6 +429,10 @@
iface |= R01_SFORI_MSB;
}
+ /* DATAI is slave only, so this has to be slave */
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+ return -EINVAL;
+
uda1380_write(codec, UDA1380_IFACE, iface);
return 0;
@@ -440,41 +467,28 @@
return 0;
}
-/*
- * Flush reg cache
- * We can only write the interpolator and decimator registers
- * when the DAI is being clocked by the CPU DAI. It's up to the
- * machine and cpu DAI driver to do this before we are called.
- */
-static int uda1380_pcm_prepare(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
+static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
- int reg, reg_start, reg_end, clk;
+ int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- reg_start = UDA1380_MVOL;
- reg_end = UDA1380_MIXER;
- } else {
- reg_start = UDA1380_DEC;
- reg_end = UDA1380_AGC;
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ uda1380_write_reg_cache(codec, UDA1380_MIXER,
+ mixer & ~R14_SILENCE);
+ schedule_work(&uda1380_work);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ uda1380_write_reg_cache(codec, UDA1380_MIXER,
+ mixer | R14_SILENCE);
+ schedule_work(&uda1380_work);
+ break;
}
-
- /* FIXME disable DAC_CLK */
- clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
- uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK);
-
- for (reg = reg_start; reg <= reg_end; reg++) {
- pr_debug("uda1380: flush reg %x val %x:", reg,
- uda1380_read_reg_cache(codec, reg));
- uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg));
- }
-
- /* FIXME restore DAC_CLK */
- uda1380_write(codec, UDA1380_CLK, clk);
-
return 0;
}
@@ -540,24 +554,6 @@
uda1380_write(codec, UDA1380_CLK, clk);
}
-static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute)
-{
- struct snd_soc_codec *codec = codec_dai->codec;
- u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM;
-
- /* FIXME: mute(codec,0) is called when the magician clock is already
- * set to WSPLL, but for some unknown reason writing to interpolator
- * registers works only when clocked by SYSCLK */
- u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
- uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk);
- if (mute)
- uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM);
- else
- uda1380_write(codec, UDA1380_DEEMP, mute_reg);
- uda1380_write(codec, UDA1380_CLK, clk);
- return 0;
-}
-
static int uda1380_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
@@ -586,23 +582,21 @@
static struct snd_soc_dai_ops uda1380_dai_ops = {
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
- .prepare = uda1380_pcm_prepare,
- .digital_mute = uda1380_mute,
+ .trigger = uda1380_trigger,
.set_fmt = uda1380_set_dai_fmt_both,
};
static struct snd_soc_dai_ops uda1380_dai_ops_playback = {
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
- .prepare = uda1380_pcm_prepare,
- .digital_mute = uda1380_mute,
+ .trigger = uda1380_trigger,
.set_fmt = uda1380_set_dai_fmt_playback,
};
static struct snd_soc_dai_ops uda1380_dai_ops_capture = {
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
- .prepare = uda1380_pcm_prepare,
+ .trigger = uda1380_trigger,
.set_fmt = uda1380_set_dai_fmt_capture,
};
@@ -700,6 +694,9 @@
codec->reg_cache_step = 1;
uda1380_reset(codec);
+ uda1380_codec = codec;
+ INIT_WORK(&uda1380_work, uda1380_flush_work);
+
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
@@ -832,8 +829,6 @@
struct snd_soc_codec *codec;
int ret;
- pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION);
-
setup = socdev->codec_data;
codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (codec == NULL)
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
new file mode 100644
index 0000000..4e1ceff
--- /dev/null
+++ b/sound/soc/codecs/wm8400.c
@@ -0,0 +1,1481 @@
+/*
+ * wm8400.c -- WM8400 ALSA Soc Audio driver
+ *
+ * Copyright 2008, 2009 Wolfson Microelectronics PLC.
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.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/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
+#include <linux/mfd/wm8400-audio.h>
+#include <linux/mfd/wm8400-private.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 <sound/tlv.h>
+
+#include "wm8400.h"
+
+/* Fake register for internal state */
+#define WM8400_INTDRIVBITS (WM8400_REGISTER_COUNT + 1)
+#define WM8400_INMIXL_PWR 0
+#define WM8400_AINLMUX_PWR 1
+#define WM8400_INMIXR_PWR 2
+#define WM8400_AINRMUX_PWR 3
+
+static struct regulator_bulk_data power[] = {
+ {
+ .supply = "I2S1VDD",
+ },
+ {
+ .supply = "I2S2VDD",
+ },
+ {
+ .supply = "DCVDD",
+ },
+ {
+ .supply = "FLLVDD",
+ },
+ {
+ .supply = "HPVDD",
+ },
+ {
+ .supply = "SPKVDD",
+ },
+};
+
+/* codec private data */
+struct wm8400_priv {
+ struct snd_soc_codec codec;
+ struct wm8400 *wm8400;
+ u16 fake_register;
+ unsigned int sysclk;
+ unsigned int pcmclk;
+ struct work_struct work;
+};
+
+static inline unsigned int wm8400_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ struct wm8400_priv *wm8400 = codec->private_data;
+
+ if (reg == WM8400_INTDRIVBITS)
+ return wm8400->fake_register;
+ else
+ return wm8400_reg_read(wm8400->wm8400, reg);
+}
+
+/*
+ * write to the wm8400 register space
+ */
+static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ struct wm8400_priv *wm8400 = codec->private_data;
+
+ if (reg == WM8400_INTDRIVBITS) {
+ wm8400->fake_register = value;
+ return 0;
+ } else
+ return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value);
+}
+
+static void wm8400_codec_reset(struct snd_soc_codec *codec)
+{
+ struct wm8400_priv *wm8400 = codec->private_data;
+
+ wm8400_reset_codec_reg_cache(wm8400->wm8400);
+}
+
+static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
+
+static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000);
+
+static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, -2100, 0);
+
+static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600);
+
+static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0);
+
+static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0);
+
+static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763);
+
+static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0);
+
+static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int reg = mc->reg;
+ int ret;
+ u16 val;
+
+ ret = snd_soc_put_volsw(kcontrol, ucontrol);
+ if (ret < 0)
+ return ret;
+
+ /* now hit the volume update bits (always bit 8) */
+ val = wm8400_read(codec, reg);
+ return wm8400_write(codec, reg, val | 0x0100);
+}
+
+#define WM8400_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_get_volsw, .put = wm8400_outpga_put_volsw_vu, \
+ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+
+static const char *wm8400_digital_sidetone[] =
+ {"None", "Left ADC", "Right ADC", "Reserved"};
+
+static const struct soc_enum wm8400_left_digital_sidetone_enum =
+SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE,
+ WM8400_ADC_TO_DACL_SHIFT, 2, wm8400_digital_sidetone);
+
+static const struct soc_enum wm8400_right_digital_sidetone_enum =
+SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE,
+ WM8400_ADC_TO_DACR_SHIFT, 2, wm8400_digital_sidetone);
+
+static const char *wm8400_adcmode[] =
+ {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"};
+
+static const struct soc_enum wm8400_right_adcmode_enum =
+SOC_ENUM_SINGLE(WM8400_ADC_CTRL, WM8400_ADC_HPF_CUT_SHIFT, 3, wm8400_adcmode);
+
+static const struct snd_kcontrol_new wm8400_snd_controls[] = {
+/* INMIXL */
+SOC_SINGLE("LIN12 PGA Boost", WM8400_INPUT_MIXER3, WM8400_L12MNBST_SHIFT,
+ 1, 0),
+SOC_SINGLE("LIN34 PGA Boost", WM8400_INPUT_MIXER3, WM8400_L34MNBST_SHIFT,
+ 1, 0),
+/* INMIXR */
+SOC_SINGLE("RIN12 PGA Boost", WM8400_INPUT_MIXER3, WM8400_R12MNBST_SHIFT,
+ 1, 0),
+SOC_SINGLE("RIN34 PGA Boost", WM8400_INPUT_MIXER3, WM8400_R34MNBST_SHIFT,
+ 1, 0),
+
+/* LOMIX */
+SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8400_OUTPUT_MIXER3,
+ WM8400_LLI3LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER3,
+ WM8400_LR12LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER3,
+ WM8400_LL12LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8400_OUTPUT_MIXER5,
+ WM8400_LRI3LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8400_OUTPUT_MIXER5,
+ WM8400_LRBLOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8400_OUTPUT_MIXER5,
+ WM8400_LRBLOVOL_SHIFT, 7, 0, out_mix_tlv),
+
+/* ROMIX */
+SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8400_OUTPUT_MIXER4,
+ WM8400_RRI3ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER4,
+ WM8400_RL12ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER4,
+ WM8400_RR12ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8400_OUTPUT_MIXER6,
+ WM8400_RLI3ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8400_OUTPUT_MIXER6,
+ WM8400_RLBROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8400_OUTPUT_MIXER6,
+ WM8400_RRBROVOL_SHIFT, 7, 0, out_mix_tlv),
+
+/* LOUT */
+WM8400_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8400_LEFT_OUTPUT_VOLUME,
+ WM8400_LOUTVOL_SHIFT, WM8400_LOUTVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("LOUT ZC", WM8400_LEFT_OUTPUT_VOLUME, WM8400_LOZC_SHIFT, 1, 0),
+
+/* ROUT */
+WM8400_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8400_RIGHT_OUTPUT_VOLUME,
+ WM8400_ROUTVOL_SHIFT, WM8400_ROUTVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("ROUT ZC", WM8400_RIGHT_OUTPUT_VOLUME, WM8400_ROZC_SHIFT, 1, 0),
+
+/* LOPGA */
+WM8400_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8400_LEFT_OPGA_VOLUME,
+ WM8400_LOPGAVOL_SHIFT, WM8400_LOPGAVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("LOPGA ZC Switch", WM8400_LEFT_OPGA_VOLUME,
+ WM8400_LOPGAZC_SHIFT, 1, 0),
+
+/* ROPGA */
+WM8400_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8400_RIGHT_OPGA_VOLUME,
+ WM8400_ROPGAVOL_SHIFT, WM8400_ROPGAVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("ROPGA ZC Switch", WM8400_RIGHT_OPGA_VOLUME,
+ WM8400_ROPGAZC_SHIFT, 1, 0),
+
+SOC_SINGLE("LON Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+ WM8400_LONMUTE_SHIFT, 1, 0),
+SOC_SINGLE("LOP Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+ WM8400_LOPMUTE_SHIFT, 1, 0),
+SOC_SINGLE("LOP Attenuation Switch", WM8400_LINE_OUTPUTS_VOLUME,
+ WM8400_LOATTN_SHIFT, 1, 0),
+SOC_SINGLE("RON Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+ WM8400_RONMUTE_SHIFT, 1, 0),
+SOC_SINGLE("ROP Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+ WM8400_ROPMUTE_SHIFT, 1, 0),
+SOC_SINGLE("ROP Attenuation Switch", WM8400_LINE_OUTPUTS_VOLUME,
+ WM8400_ROATTN_SHIFT, 1, 0),
+
+SOC_SINGLE("OUT3 Mute Switch", WM8400_OUT3_4_VOLUME,
+ WM8400_OUT3MUTE_SHIFT, 1, 0),
+SOC_SINGLE("OUT3 Attenuation Switch", WM8400_OUT3_4_VOLUME,
+ WM8400_OUT3ATTN_SHIFT, 1, 0),
+
+SOC_SINGLE("OUT4 Mute Switch", WM8400_OUT3_4_VOLUME,
+ WM8400_OUT4MUTE_SHIFT, 1, 0),
+SOC_SINGLE("OUT4 Attenuation Switch", WM8400_OUT3_4_VOLUME,
+ WM8400_OUT4ATTN_SHIFT, 1, 0),
+
+SOC_SINGLE("Speaker Mode Switch", WM8400_CLASSD1,
+ WM8400_CDMODE_SHIFT, 1, 0),
+
+SOC_SINGLE("Speaker Output Attenuation Volume", WM8400_SPEAKER_VOLUME,
+ WM8400_SPKATTN_SHIFT, WM8400_SPKATTN_MASK, 0),
+SOC_SINGLE("Speaker DC Boost Volume", WM8400_CLASSD3,
+ WM8400_DCGAIN_SHIFT, 6, 0),
+SOC_SINGLE("Speaker AC Boost Volume", WM8400_CLASSD3,
+ WM8400_ACGAIN_SHIFT, 6, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume",
+ WM8400_LEFT_DAC_DIGITAL_VOLUME, WM8400_DACL_VOL_SHIFT,
+ 127, 0, out_dac_tlv),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume",
+ WM8400_RIGHT_DAC_DIGITAL_VOLUME, WM8400_DACR_VOL_SHIFT,
+ 127, 0, out_dac_tlv),
+
+SOC_ENUM("Left Digital Sidetone", wm8400_left_digital_sidetone_enum),
+SOC_ENUM("Right Digital Sidetone", wm8400_right_digital_sidetone_enum),
+
+SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8400_DIGITAL_SIDE_TONE,
+ WM8400_ADCL_DAC_SVOL_SHIFT, 15, 0, out_sidetone_tlv),
+SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8400_DIGITAL_SIDE_TONE,
+ WM8400_ADCR_DAC_SVOL_SHIFT, 15, 0, out_sidetone_tlv),
+
+SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8400_ADC_CTRL,
+ WM8400_ADC_HPF_ENA_SHIFT, 1, 0),
+
+SOC_ENUM("ADC HPF Mode", wm8400_right_adcmode_enum),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume",
+ WM8400_LEFT_ADC_DIGITAL_VOLUME,
+ WM8400_ADCL_VOL_SHIFT,
+ WM8400_ADCL_VOL_MASK,
+ 0,
+ in_adc_tlv),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume",
+ WM8400_RIGHT_ADC_DIGITAL_VOLUME,
+ WM8400_ADCR_VOL_SHIFT,
+ WM8400_ADCR_VOL_MASK,
+ 0,
+ in_adc_tlv),
+
+WM8400_OUTPGA_SINGLE_R_TLV("LIN12 Volume",
+ WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+ WM8400_LIN12VOL_SHIFT,
+ WM8400_LIN12VOL_MASK,
+ 0,
+ in_pga_tlv),
+
+SOC_SINGLE("LIN12 ZC Switch", WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+ WM8400_LI12ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("LIN12 Mute Switch", WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+ WM8400_LI12MUTE_SHIFT, 1, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("LIN34 Volume",
+ WM8400_LEFT_LINE_INPUT_3_4_VOLUME,
+ WM8400_LIN34VOL_SHIFT,
+ WM8400_LIN34VOL_MASK,
+ 0,
+ in_pga_tlv),
+
+SOC_SINGLE("LIN34 ZC Switch", WM8400_LEFT_LINE_INPUT_3_4_VOLUME,
+ WM8400_LI34ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("LIN34 Mute Switch", WM8400_LEFT_LINE_INPUT_3_4_VOLUME,
+ WM8400_LI34MUTE_SHIFT, 1, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("RIN12 Volume",
+ WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+ WM8400_RIN12VOL_SHIFT,
+ WM8400_RIN12VOL_MASK,
+ 0,
+ in_pga_tlv),
+
+SOC_SINGLE("RIN12 ZC Switch", WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+ WM8400_RI12ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("RIN12 Mute Switch", WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+ WM8400_RI12MUTE_SHIFT, 1, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("RIN34 Volume",
+ WM8400_RIGHT_LINE_INPUT_3_4_VOLUME,
+ WM8400_RIN34VOL_SHIFT,
+ WM8400_RIN34VOL_MASK,
+ 0,
+ in_pga_tlv),
+
+SOC_SINGLE("RIN34 ZC Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME,
+ WM8400_RI34ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME,
+ WM8400_RI34MUTE_SHIFT, 1, 0),
+
+};
+
+/* add non dapm controls */
+static int wm8400_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8400_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8400_snd_controls[i],codec,
+ NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/*
+ * _DAPM_ Controls
+ */
+
+static int inmixer_event (struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ u16 reg, fakepower;
+
+ reg = wm8400_read(w->codec, WM8400_POWER_MANAGEMENT_2);
+ fakepower = wm8400_read(w->codec, WM8400_INTDRIVBITS);
+
+ if (fakepower & ((1 << WM8400_INMIXL_PWR) |
+ (1 << WM8400_AINLMUX_PWR))) {
+ reg |= WM8400_AINL_ENA;
+ } else {
+ reg &= ~WM8400_AINL_ENA;
+ }
+
+ if (fakepower & ((1 << WM8400_INMIXR_PWR) |
+ (1 << WM8400_AINRMUX_PWR))) {
+ reg |= WM8400_AINR_ENA;
+ } else {
+ reg &= ~WM8400_AINL_ENA;
+ }
+ wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg);
+
+ return 0;
+}
+
+static int outmixer_event (struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol * kcontrol, int event)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ u32 reg_shift = mc->shift;
+ int ret = 0;
+ u16 reg;
+
+ switch (reg_shift) {
+ case WM8400_SPEAKER_MIXER | (WM8400_LDSPK << 8) :
+ reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER1);
+ if (reg & WM8400_LDLO) {
+ printk(KERN_WARNING
+ "Cannot set as Output Mixer 1 LDLO Set\n");
+ ret = -1;
+ }
+ break;
+ case WM8400_SPEAKER_MIXER | (WM8400_RDSPK << 8):
+ reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER2);
+ if (reg & WM8400_RDRO) {
+ printk(KERN_WARNING
+ "Cannot set as Output Mixer 2 RDRO Set\n");
+ ret = -1;
+ }
+ break;
+ case WM8400_OUTPUT_MIXER1 | (WM8400_LDLO << 8):
+ reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER);
+ if (reg & WM8400_LDSPK) {
+ printk(KERN_WARNING
+ "Cannot set as Speaker Mixer LDSPK Set\n");
+ ret = -1;
+ }
+ break;
+ case WM8400_OUTPUT_MIXER2 | (WM8400_RDRO << 8):
+ reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER);
+ if (reg & WM8400_RDSPK) {
+ printk(KERN_WARNING
+ "Cannot set as Speaker Mixer RDSPK Set\n");
+ ret = -1;
+ }
+ break;
+ }
+
+ return ret;
+}
+
+/* INMIX dB values */
+static const unsigned int in_mix_tlv[] = {
+ TLV_DB_RANGE_HEAD(1),
+ 0,7, TLV_DB_LINEAR_ITEM(-1200, 600),
+};
+
+/* Left In PGA Connections */
+static const struct snd_kcontrol_new wm8400_dapm_lin12_pga_controls[] = {
+SOC_DAPM_SINGLE("LIN1 Switch", WM8400_INPUT_MIXER2, WM8400_LMN1_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LIN2 Switch", WM8400_INPUT_MIXER2, WM8400_LMP2_SHIFT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8400_dapm_lin34_pga_controls[] = {
+SOC_DAPM_SINGLE("LIN3 Switch", WM8400_INPUT_MIXER2, WM8400_LMN3_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LIN4 Switch", WM8400_INPUT_MIXER2, WM8400_LMP4_SHIFT, 1, 0),
+};
+
+/* Right In PGA Connections */
+static const struct snd_kcontrol_new wm8400_dapm_rin12_pga_controls[] = {
+SOC_DAPM_SINGLE("RIN1 Switch", WM8400_INPUT_MIXER2, WM8400_RMN1_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RIN2 Switch", WM8400_INPUT_MIXER2, WM8400_RMP2_SHIFT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8400_dapm_rin34_pga_controls[] = {
+SOC_DAPM_SINGLE("RIN3 Switch", WM8400_INPUT_MIXER2, WM8400_RMN3_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RIN4 Switch", WM8400_INPUT_MIXER2, WM8400_RMP4_SHIFT, 1, 0),
+};
+
+/* INMIXL */
+static const struct snd_kcontrol_new wm8400_dapm_inmixl_controls[] = {
+SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8400_INPUT_MIXER3,
+ WM8400_LDBVOL_SHIFT, WM8400_LDBVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8400_INPUT_MIXER5, WM8400_LI2BVOL_SHIFT,
+ 7, 0, in_mix_tlv),
+SOC_DAPM_SINGLE("LINPGA12 Switch", WM8400_INPUT_MIXER3, WM8400_L12MNB_SHIFT,
+ 1, 0),
+SOC_DAPM_SINGLE("LINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT,
+ 1, 0),
+};
+
+/* INMIXR */
+static const struct snd_kcontrol_new wm8400_dapm_inmixr_controls[] = {
+SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8400_INPUT_MIXER4,
+ WM8400_RDBVOL_SHIFT, WM8400_RDBVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8400_INPUT_MIXER6, WM8400_RI2BVOL_SHIFT,
+ 7, 0, in_mix_tlv),
+SOC_DAPM_SINGLE("RINPGA12 Switch", WM8400_INPUT_MIXER3, WM8400_L12MNB_SHIFT,
+ 1, 0),
+SOC_DAPM_SINGLE("RINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT,
+ 1, 0),
+};
+
+/* AINLMUX */
+static const char *wm8400_ainlmux[] =
+ {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"};
+
+static const struct soc_enum wm8400_ainlmux_enum =
+SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINLMODE_SHIFT,
+ ARRAY_SIZE(wm8400_ainlmux), wm8400_ainlmux);
+
+static const struct snd_kcontrol_new wm8400_dapm_ainlmux_controls =
+SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum);
+
+/* DIFFINL */
+
+/* AINRMUX */
+static const char *wm8400_ainrmux[] =
+ {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"};
+
+static const struct soc_enum wm8400_ainrmux_enum =
+SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINRMODE_SHIFT,
+ ARRAY_SIZE(wm8400_ainrmux), wm8400_ainrmux);
+
+static const struct snd_kcontrol_new wm8400_dapm_ainrmux_controls =
+SOC_DAPM_ENUM("Route", wm8400_ainrmux_enum);
+
+/* RXVOICE */
+static const struct snd_kcontrol_new wm8400_dapm_rxvoice_controls[] = {
+SOC_DAPM_SINGLE_TLV("LIN4/RXN", WM8400_INPUT_MIXER5, WM8400_LR4BVOL_SHIFT,
+ WM8400_LR4BVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("RIN4/RXP", WM8400_INPUT_MIXER6, WM8400_RL4BVOL_SHIFT,
+ WM8400_RL4BVOL_MASK, 0, in_mix_tlv),
+};
+
+/* LOMIX */
+static const struct snd_kcontrol_new wm8400_dapm_lomix_controls[] = {
+SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8400_OUTPUT_MIXER1,
+ WM8400_LRBLO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8400_OUTPUT_MIXER1,
+ WM8400_LLBLO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8400_OUTPUT_MIXER1,
+ WM8400_LRI3LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8400_OUTPUT_MIXER1,
+ WM8400_LLI3LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER1,
+ WM8400_LR12LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER1,
+ WM8400_LL12LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8400_OUTPUT_MIXER1,
+ WM8400_LDLO_SHIFT, 1, 0),
+};
+
+/* ROMIX */
+static const struct snd_kcontrol_new wm8400_dapm_romix_controls[] = {
+SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8400_OUTPUT_MIXER2,
+ WM8400_RLBRO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8400_OUTPUT_MIXER2,
+ WM8400_RRBRO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8400_OUTPUT_MIXER2,
+ WM8400_RLI3RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8400_OUTPUT_MIXER2,
+ WM8400_RRI3RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER2,
+ WM8400_RL12RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER2,
+ WM8400_RR12RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8400_OUTPUT_MIXER2,
+ WM8400_RDRO_SHIFT, 1, 0),
+};
+
+/* LONMIX */
+static const struct snd_kcontrol_new wm8400_dapm_lonmix_controls[] = {
+SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8400_LINE_MIXER1,
+ WM8400_LLOPGALON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8400_LINE_MIXER1,
+ WM8400_LROPGALON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8400_LINE_MIXER1,
+ WM8400_LOPLON_SHIFT, 1, 0),
+};
+
+/* LOPMIX */
+static const struct snd_kcontrol_new wm8400_dapm_lopmix_controls[] = {
+SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8400_LINE_MIXER1,
+ WM8400_LR12LOP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8400_LINE_MIXER1,
+ WM8400_LL12LOP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8400_LINE_MIXER1,
+ WM8400_LLOPGALOP_SHIFT, 1, 0),
+};
+
+/* RONMIX */
+static const struct snd_kcontrol_new wm8400_dapm_ronmix_controls[] = {
+SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8400_LINE_MIXER2,
+ WM8400_RROPGARON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8400_LINE_MIXER2,
+ WM8400_RLOPGARON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8400_LINE_MIXER2,
+ WM8400_ROPRON_SHIFT, 1, 0),
+};
+
+/* ROPMIX */
+static const struct snd_kcontrol_new wm8400_dapm_ropmix_controls[] = {
+SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8400_LINE_MIXER2,
+ WM8400_RL12ROP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8400_LINE_MIXER2,
+ WM8400_RR12ROP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8400_LINE_MIXER2,
+ WM8400_RROPGAROP_SHIFT, 1, 0),
+};
+
+/* OUT3MIX */
+static const struct snd_kcontrol_new wm8400_dapm_out3mix_controls[] = {
+SOC_DAPM_SINGLE("OUT3MIX LIN4/RXP Bypass Switch", WM8400_OUT3_4_MIXER,
+ WM8400_LI4O3_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8400_OUT3_4_MIXER,
+ WM8400_LPGAO3_SHIFT, 1, 0),
+};
+
+/* OUT4MIX */
+static const struct snd_kcontrol_new wm8400_dapm_out4mix_controls[] = {
+SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8400_OUT3_4_MIXER,
+ WM8400_RPGAO4_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("OUT4MIX RIN4/RXP Bypass Switch", WM8400_OUT3_4_MIXER,
+ WM8400_RI4O4_SHIFT, 1, 0),
+};
+
+/* SPKMIX */
+static const struct snd_kcontrol_new wm8400_dapm_spkmix_controls[] = {
+SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8400_SPEAKER_MIXER,
+ WM8400_LI2SPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8400_SPEAKER_MIXER,
+ WM8400_LB2SPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8400_SPEAKER_MIXER,
+ WM8400_LOPGASPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8400_SPEAKER_MIXER,
+ WM8400_LDSPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8400_SPEAKER_MIXER,
+ WM8400_RDSPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8400_SPEAKER_MIXER,
+ WM8400_ROPGASPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8400_SPEAKER_MIXER,
+ WM8400_RL12ROP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8400_SPEAKER_MIXER,
+ WM8400_RI2SPK_SHIFT, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8400_dapm_widgets[] = {
+/* Input Side */
+/* Input Lines */
+SND_SOC_DAPM_INPUT("LIN1"),
+SND_SOC_DAPM_INPUT("LIN2"),
+SND_SOC_DAPM_INPUT("LIN3"),
+SND_SOC_DAPM_INPUT("LIN4/RXN"),
+SND_SOC_DAPM_INPUT("RIN3"),
+SND_SOC_DAPM_INPUT("RIN4/RXP"),
+SND_SOC_DAPM_INPUT("RIN1"),
+SND_SOC_DAPM_INPUT("RIN2"),
+SND_SOC_DAPM_INPUT("Internal ADC Source"),
+
+/* DACs */
+SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8400_POWER_MANAGEMENT_2,
+ WM8400_ADCL_ENA_SHIFT, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8400_POWER_MANAGEMENT_2,
+ WM8400_ADCR_ENA_SHIFT, 0),
+
+/* Input PGAs */
+SND_SOC_DAPM_MIXER("LIN12 PGA", WM8400_POWER_MANAGEMENT_2,
+ WM8400_LIN12_ENA_SHIFT,
+ 0, &wm8400_dapm_lin12_pga_controls[0],
+ ARRAY_SIZE(wm8400_dapm_lin12_pga_controls)),
+SND_SOC_DAPM_MIXER("LIN34 PGA", WM8400_POWER_MANAGEMENT_2,
+ WM8400_LIN34_ENA_SHIFT,
+ 0, &wm8400_dapm_lin34_pga_controls[0],
+ ARRAY_SIZE(wm8400_dapm_lin34_pga_controls)),
+SND_SOC_DAPM_MIXER("RIN12 PGA", WM8400_POWER_MANAGEMENT_2,
+ WM8400_RIN12_ENA_SHIFT,
+ 0, &wm8400_dapm_rin12_pga_controls[0],
+ ARRAY_SIZE(wm8400_dapm_rin12_pga_controls)),
+SND_SOC_DAPM_MIXER("RIN34 PGA", WM8400_POWER_MANAGEMENT_2,
+ WM8400_RIN34_ENA_SHIFT,
+ 0, &wm8400_dapm_rin34_pga_controls[0],
+ ARRAY_SIZE(wm8400_dapm_rin34_pga_controls)),
+
+/* INMIXL */
+SND_SOC_DAPM_MIXER_E("INMIXL", WM8400_INTDRIVBITS, WM8400_INMIXL_PWR, 0,
+ &wm8400_dapm_inmixl_controls[0],
+ ARRAY_SIZE(wm8400_dapm_inmixl_controls),
+ inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* AINLMUX */
+SND_SOC_DAPM_MUX_E("AILNMUX", WM8400_INTDRIVBITS, WM8400_AINLMUX_PWR, 0,
+ &wm8400_dapm_ainlmux_controls, inmixer_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* INMIXR */
+SND_SOC_DAPM_MIXER_E("INMIXR", WM8400_INTDRIVBITS, WM8400_INMIXR_PWR, 0,
+ &wm8400_dapm_inmixr_controls[0],
+ ARRAY_SIZE(wm8400_dapm_inmixr_controls),
+ inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* AINRMUX */
+SND_SOC_DAPM_MUX_E("AIRNMUX", WM8400_INTDRIVBITS, WM8400_AINRMUX_PWR, 0,
+ &wm8400_dapm_ainrmux_controls, inmixer_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* Output Side */
+/* DACs */
+SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8400_POWER_MANAGEMENT_3,
+ WM8400_DACL_ENA_SHIFT, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8400_POWER_MANAGEMENT_3,
+ WM8400_DACR_ENA_SHIFT, 0),
+
+/* LOMIX */
+SND_SOC_DAPM_MIXER_E("LOMIX", WM8400_POWER_MANAGEMENT_3,
+ WM8400_LOMIX_ENA_SHIFT,
+ 0, &wm8400_dapm_lomix_controls[0],
+ ARRAY_SIZE(wm8400_dapm_lomix_controls),
+ outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+/* LONMIX */
+SND_SOC_DAPM_MIXER("LONMIX", WM8400_POWER_MANAGEMENT_3, WM8400_LON_ENA_SHIFT,
+ 0, &wm8400_dapm_lonmix_controls[0],
+ ARRAY_SIZE(wm8400_dapm_lonmix_controls)),
+
+/* LOPMIX */
+SND_SOC_DAPM_MIXER("LOPMIX", WM8400_POWER_MANAGEMENT_3, WM8400_LOP_ENA_SHIFT,
+ 0, &wm8400_dapm_lopmix_controls[0],
+ ARRAY_SIZE(wm8400_dapm_lopmix_controls)),
+
+/* OUT3MIX */
+SND_SOC_DAPM_MIXER("OUT3MIX", WM8400_POWER_MANAGEMENT_1, WM8400_OUT3_ENA_SHIFT,
+ 0, &wm8400_dapm_out3mix_controls[0],
+ ARRAY_SIZE(wm8400_dapm_out3mix_controls)),
+
+/* SPKMIX */
+SND_SOC_DAPM_MIXER_E("SPKMIX", WM8400_POWER_MANAGEMENT_1, WM8400_SPK_ENA_SHIFT,
+ 0, &wm8400_dapm_spkmix_controls[0],
+ ARRAY_SIZE(wm8400_dapm_spkmix_controls), outmixer_event,
+ SND_SOC_DAPM_PRE_REG),
+
+/* OUT4MIX */
+SND_SOC_DAPM_MIXER("OUT4MIX", WM8400_POWER_MANAGEMENT_1, WM8400_OUT4_ENA_SHIFT,
+ 0, &wm8400_dapm_out4mix_controls[0],
+ ARRAY_SIZE(wm8400_dapm_out4mix_controls)),
+
+/* ROPMIX */
+SND_SOC_DAPM_MIXER("ROPMIX", WM8400_POWER_MANAGEMENT_3, WM8400_ROP_ENA_SHIFT,
+ 0, &wm8400_dapm_ropmix_controls[0],
+ ARRAY_SIZE(wm8400_dapm_ropmix_controls)),
+
+/* RONMIX */
+SND_SOC_DAPM_MIXER("RONMIX", WM8400_POWER_MANAGEMENT_3, WM8400_RON_ENA_SHIFT,
+ 0, &wm8400_dapm_ronmix_controls[0],
+ ARRAY_SIZE(wm8400_dapm_ronmix_controls)),
+
+/* ROMIX */
+SND_SOC_DAPM_MIXER_E("ROMIX", WM8400_POWER_MANAGEMENT_3,
+ WM8400_ROMIX_ENA_SHIFT,
+ 0, &wm8400_dapm_romix_controls[0],
+ ARRAY_SIZE(wm8400_dapm_romix_controls),
+ outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+/* LOUT PGA */
+SND_SOC_DAPM_PGA("LOUT PGA", WM8400_POWER_MANAGEMENT_1, WM8400_LOUT_ENA_SHIFT,
+ 0, NULL, 0),
+
+/* ROUT PGA */
+SND_SOC_DAPM_PGA("ROUT PGA", WM8400_POWER_MANAGEMENT_1, WM8400_ROUT_ENA_SHIFT,
+ 0, NULL, 0),
+
+/* LOPGA */
+SND_SOC_DAPM_PGA("LOPGA", WM8400_POWER_MANAGEMENT_3, WM8400_LOPGA_ENA_SHIFT, 0,
+ NULL, 0),
+
+/* ROPGA */
+SND_SOC_DAPM_PGA("ROPGA", WM8400_POWER_MANAGEMENT_3, WM8400_ROPGA_ENA_SHIFT, 0,
+ NULL, 0),
+
+/* MICBIAS */
+SND_SOC_DAPM_MICBIAS("MICBIAS", WM8400_POWER_MANAGEMENT_1,
+ WM8400_MIC1BIAS_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_OUTPUT("LON"),
+SND_SOC_DAPM_OUTPUT("LOP"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("SPKN"),
+SND_SOC_DAPM_OUTPUT("SPKP"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_OUTPUT("OUT4"),
+SND_SOC_DAPM_OUTPUT("ROP"),
+SND_SOC_DAPM_OUTPUT("RON"),
+
+SND_SOC_DAPM_OUTPUT("Internal DAC Sink"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Make DACs turn on when playing even if not mixed into any outputs */
+ {"Internal DAC Sink", NULL, "Left DAC"},
+ {"Internal DAC Sink", NULL, "Right DAC"},
+
+ /* Make ADCs turn on when recording
+ * even if not mixed from any inputs */
+ {"Left ADC", NULL, "Internal ADC Source"},
+ {"Right ADC", NULL, "Internal ADC Source"},
+
+ /* Input Side */
+ /* LIN12 PGA */
+ {"LIN12 PGA", "LIN1 Switch", "LIN1"},
+ {"LIN12 PGA", "LIN2 Switch", "LIN2"},
+ /* LIN34 PGA */
+ {"LIN34 PGA", "LIN3 Switch", "LIN3"},
+ {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"},
+ /* INMIXL */
+ {"INMIXL", "Record Left Volume", "LOMIX"},
+ {"INMIXL", "LIN2 Volume", "LIN2"},
+ {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
+ {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
+ /* AILNMUX */
+ {"AILNMUX", "INMIXL Mix", "INMIXL"},
+ {"AILNMUX", "DIFFINL Mix", "LIN12 PGA"},
+ {"AILNMUX", "DIFFINL Mix", "LIN34 PGA"},
+ {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
+ {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+ /* ADC */
+ {"Left ADC", NULL, "AILNMUX"},
+
+ /* RIN12 PGA */
+ {"RIN12 PGA", "RIN1 Switch", "RIN1"},
+ {"RIN12 PGA", "RIN2 Switch", "RIN2"},
+ /* RIN34 PGA */
+ {"RIN34 PGA", "RIN3 Switch", "RIN3"},
+ {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"},
+ /* INMIXL */
+ {"INMIXR", "Record Right Volume", "ROMIX"},
+ {"INMIXR", "RIN2 Volume", "RIN2"},
+ {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
+ {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
+ /* AIRNMUX */
+ {"AIRNMUX", "INMIXR Mix", "INMIXR"},
+ {"AIRNMUX", "DIFFINR Mix", "RIN12 PGA"},
+ {"AIRNMUX", "DIFFINR Mix", "RIN34 PGA"},
+ {"AIRNMUX", "RXVOICE Mix", "LIN4/RXN"},
+ {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+ /* ADC */
+ {"Right ADC", NULL, "AIRNMUX"},
+
+ /* LOMIX */
+ {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
+ {"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"},
+ {"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+ {"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+ {"LOMIX", "LOMIX Right ADC Bypass Switch", "AIRNMUX"},
+ {"LOMIX", "LOMIX Left ADC Bypass Switch", "AILNMUX"},
+ {"LOMIX", "LOMIX Left DAC Switch", "Left DAC"},
+
+ /* ROMIX */
+ {"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"},
+ {"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"},
+ {"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+ {"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+ {"ROMIX", "ROMIX Right ADC Bypass Switch", "AIRNMUX"},
+ {"ROMIX", "ROMIX Left ADC Bypass Switch", "AILNMUX"},
+ {"ROMIX", "ROMIX Right DAC Switch", "Right DAC"},
+
+ /* SPKMIX */
+ {"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"},
+ {"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"},
+ {"SPKMIX", "SPKMIX LADC Bypass Switch", "AILNMUX"},
+ {"SPKMIX", "SPKMIX RADC Bypass Switch", "AIRNMUX"},
+ {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"},
+ {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"},
+ {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"},
+ {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"},
+
+ /* LONMIX */
+ {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"},
+ {"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"},
+ {"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"},
+
+ /* LOPMIX */
+ {"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+ {"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+ {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
+
+ /* OUT3MIX */
+ {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"},
+ {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
+
+ /* OUT4MIX */
+ {"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"},
+ {"OUT4MIX", "OUT4MIX RIN4/RXP Bypass Switch", "RIN4/RXP"},
+
+ /* RONMIX */
+ {"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"},
+ {"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"},
+ {"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"},
+
+ /* ROPMIX */
+ {"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+ {"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+ {"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"},
+
+ /* Out Mixer PGAs */
+ {"LOPGA", NULL, "LOMIX"},
+ {"ROPGA", NULL, "ROMIX"},
+
+ {"LOUT PGA", NULL, "LOMIX"},
+ {"ROUT PGA", NULL, "ROMIX"},
+
+ /* Output Pins */
+ {"LON", NULL, "LONMIX"},
+ {"LOP", NULL, "LOPMIX"},
+ {"OUT3", NULL, "OUT3MIX"},
+ {"LOUT", NULL, "LOUT PGA"},
+ {"SPKN", NULL, "SPKMIX"},
+ {"ROUT", NULL, "ROUT PGA"},
+ {"OUT4", NULL, "OUT4MIX"},
+ {"ROP", NULL, "ROPMIX"},
+ {"RON", NULL, "RONMIX"},
+};
+
+static int wm8400_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8400_dapm_widgets,
+ ARRAY_SIZE(wm8400_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+/*
+ * Clock after FLL and dividers
+ */
+static int wm8400_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8400_priv *wm8400 = codec->private_data;
+
+ wm8400->sysclk = freq;
+ return 0;
+}
+
+/*
+ * Sets ADC and Voice DAC format.
+ */
+static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 audio1, audio3;
+
+ audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1);
+ audio3 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_3);
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ audio3 &= ~WM8400_AIF_MSTR1;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ audio3 |= WM8400_AIF_MSTR1;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ audio1 &= ~WM8400_AIF_FMT_MASK;
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ audio1 |= WM8400_AIF_FMT_I2S;
+ audio1 &= ~WM8400_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ audio1 |= WM8400_AIF_FMT_RIGHTJ;
+ audio1 &= ~WM8400_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ audio1 |= WM8400_AIF_FMT_LEFTJ;
+ audio1 &= ~WM8400_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ audio1 |= WM8400_AIF_FMT_DSP;
+ audio1 &= ~WM8400_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ audio1 |= WM8400_AIF_FMT_DSP | WM8400_AIF_LRCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
+ wm8400_write(codec, WM8400_AUDIO_INTERFACE_3, audio3);
+ return 0;
+}
+
+static int wm8400_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ switch (div_id) {
+ case WM8400_MCLK_DIV:
+ reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+ ~WM8400_MCLK_DIV_MASK;
+ wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+ break;
+ case WM8400_DACCLK_DIV:
+ reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+ ~WM8400_DAC_CLKDIV_MASK;
+ wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+ break;
+ case WM8400_ADCCLK_DIV:
+ reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+ ~WM8400_ADC_CLKDIV_MASK;
+ wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+ break;
+ case WM8400_BCLK_DIV:
+ reg = wm8400_read(codec, WM8400_CLOCKING_1) &
+ ~WM8400_BCLK_DIV_MASK;
+ wm8400_write(codec, WM8400_CLOCKING_1, reg | div);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+/*
+ * Set PCM DAI bit size and sample rate.
+ */
+static int wm8400_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ u16 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1);
+
+ audio1 &= ~WM8400_AIF_WL_MASK;
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ audio1 |= WM8400_AIF_WL_20BITS;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ audio1 |= WM8400_AIF_WL_24BITS;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ audio1 |= WM8400_AIF_WL_32BITS;
+ break;
+ }
+
+ wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
+ return 0;
+}
+
+static int wm8400_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 val = wm8400_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE;
+
+ if (mute)
+ wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
+ else
+ wm8400_write(codec, WM8400_DAC_CTRL, val);
+
+ return 0;
+}
+
+/* TODO: set bias for best performance at standby */
+static int wm8400_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct wm8400_priv *wm8400 = codec->private_data;
+ u16 val;
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ /* VMID=2*50k */
+ val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) &
+ ~WM8400_VMID_MODE_MASK;
+ wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ ret = regulator_bulk_enable(ARRAY_SIZE(power),
+ &power[0]);
+ if (ret != 0) {
+ dev_err(wm8400->wm8400->dev,
+ "Failed to enable regulators: %d\n",
+ ret);
+ return ret;
+ }
+
+ wm8400_write(codec, WM8400_POWER_MANAGEMENT_1,
+ WM8400_CODEC_ENA | WM8400_SYSCLK_ENA);
+
+ /* Enable all output discharge bits */
+ wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE |
+ WM8400_DIS_RLINE | WM8400_DIS_OUT3 |
+ WM8400_DIS_OUT4 | WM8400_DIS_LOUT |
+ WM8400_DIS_ROUT);
+
+ /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */
+ wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+ WM8400_BUFDCOPEN | WM8400_POBCTRL);
+
+ msleep(500);
+
+ /* Enable outputs */
+ val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+ val |= WM8400_SPK_ENA | WM8400_OUT3_ENA |
+ WM8400_OUT4_ENA | WM8400_LOUT_ENA |
+ WM8400_ROUT_ENA;
+ wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+ /* disable all output discharge bits */
+ wm8400_write(codec, WM8400_ANTIPOP1, 0);
+
+ /* Enable VREF & VMID at 2x50k */
+ val |= 0x2 | WM8400_VREF_ENA;
+ wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+ msleep(600);
+
+ /* Enable BUFIOEN */
+ wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+ WM8400_BUFDCOPEN | WM8400_POBCTRL |
+ WM8400_BUFIOEN);
+
+ /* Disable outputs */
+ val &= ~(WM8400_SPK_ENA | WM8400_OUT3_ENA |
+ WM8400_OUT4_ENA | WM8400_LOUT_ENA |
+ WM8400_ROUT_ENA);
+ wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+ /* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+ wm8400_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN);
+ }
+
+ /* VMID=2*300k */
+ val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) &
+ ~WM8400_VMID_MODE_MASK;
+ wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ /* Enable POBCTRL and SOFT_ST */
+ wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+ WM8400_POBCTRL | WM8400_BUFIOEN);
+
+ /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */
+ wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+ WM8400_BUFDCOPEN | WM8400_POBCTRL |
+ WM8400_BUFIOEN);
+
+ /* mute DAC */
+ val = wm8400_read(codec, WM8400_DAC_CTRL);
+ wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
+
+ /* Enable any disabled outputs */
+ val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+ val |= WM8400_SPK_ENA | WM8400_OUT3_ENA |
+ WM8400_OUT4_ENA | WM8400_LOUT_ENA |
+ WM8400_ROUT_ENA;
+ wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+ /* Disable VMID */
+ val &= ~WM8400_VMID_MODE_MASK;
+ wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+ msleep(300);
+
+ /* Enable all output discharge bits */
+ wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE |
+ WM8400_DIS_RLINE | WM8400_DIS_OUT3 |
+ WM8400_DIS_OUT4 | WM8400_DIS_LOUT |
+ WM8400_DIS_ROUT);
+
+ /* Disable VREF */
+ val &= ~WM8400_VREF_ENA;
+ wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+ /* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+ wm8400_write(codec, WM8400_ANTIPOP2, 0x0);
+
+ ret = regulator_bulk_disable(ARRAY_SIZE(power),
+ &power[0]);
+ if (ret != 0)
+ return ret;
+
+ break;
+ }
+
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8400_RATES SNDRV_PCM_RATE_8000_96000
+
+#define WM8400_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8400_dai_ops = {
+ .hw_params = wm8400_hw_params,
+ .digital_mute = wm8400_mute,
+ .set_fmt = wm8400_set_dai_fmt,
+ .set_clkdiv = wm8400_set_dai_clkdiv,
+ .set_sysclk = wm8400_set_dai_sysclk,
+};
+
+/*
+ * The WM8400 supports 2 different and mutually exclusive DAI
+ * configurations.
+ *
+ * 1. ADC/DAC on Primary Interface
+ * 2. ADC on Primary Interface/DAC on secondary
+ */
+struct snd_soc_dai wm8400_dai = {
+/* ADC/DAC on primary */
+ .name = "WM8400 ADC/DAC Primary",
+ .id = 1,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8400_RATES,
+ .formats = WM8400_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8400_RATES,
+ .formats = WM8400_FORMATS,
+ },
+ .ops = &wm8400_dai_ops,
+};
+EXPORT_SYMBOL_GPL(wm8400_dai);
+
+static int wm8400_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int wm8400_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+
+static struct snd_soc_codec *wm8400_codec;
+
+static int wm8400_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret;
+
+ if (!wm8400_codec) {
+ dev_err(&pdev->dev, "wm8400 not yet discovered\n");
+ return -ENODEV;
+ }
+ codec = wm8400_codec;
+
+ socdev->card->codec = codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ wm8400_add_controls(codec);
+ wm8400_add_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+/* power down chip */
+static int wm8400_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_wm8400 = {
+ .probe = wm8400_probe,
+ .remove = wm8400_remove,
+ .suspend = wm8400_suspend,
+ .resume = wm8400_resume,
+};
+
+static void wm8400_probe_deferred(struct work_struct *work)
+{
+ struct wm8400_priv *priv = container_of(work, struct wm8400_priv,
+ work);
+ struct snd_soc_codec *codec = &priv->codec;
+ int ret;
+
+ /* charge output caps */
+ wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* We're done, tell the subsystem. */
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(priv->wm8400->dev,
+ "Failed to register codec: %d\n", ret);
+ goto err;
+ }
+
+ ret = snd_soc_register_dai(&wm8400_dai);
+ if (ret != 0) {
+ dev_err(priv->wm8400->dev,
+ "Failed to register DAI: %d\n", ret);
+ goto err_codec;
+ }
+
+ return;
+
+err_codec:
+ snd_soc_unregister_codec(codec);
+err:
+ wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int wm8400_codec_probe(struct platform_device *dev)
+{
+ struct wm8400_priv *priv;
+ int ret;
+ u16 reg;
+ struct snd_soc_codec *codec;
+
+ priv = kzalloc(sizeof(struct wm8400_priv), GFP_KERNEL);
+ if (priv == NULL)
+ return -ENOMEM;
+
+ codec = &priv->codec;
+ codec->private_data = priv;
+ codec->control_data = dev->dev.driver_data;
+ priv->wm8400 = dev->dev.driver_data;
+
+ ret = regulator_bulk_get(priv->wm8400->dev,
+ ARRAY_SIZE(power), &power[0]);
+ if (ret != 0) {
+ dev_err(&dev->dev, "Failed to get regulators: %d\n", ret);
+ goto err;
+ }
+
+ codec->dev = &dev->dev;
+ wm8400_dai.dev = &dev->dev;
+
+ codec->name = "WM8400";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8400_read;
+ codec->write = wm8400_write;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8400_set_bias_level;
+ codec->dai = &wm8400_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = WM8400_REGISTER_COUNT;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+ INIT_WORK(&priv->work, wm8400_probe_deferred);
+
+ wm8400_codec_reset(codec);
+
+ reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+ wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA);
+
+ /* Latch volume update bits */
+ reg = wm8400_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME);
+ wm8400_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+ reg & WM8400_IPVU);
+ reg = wm8400_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME);
+ wm8400_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+ reg & WM8400_IPVU);
+
+ wm8400_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
+ wm8400_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
+
+ wm8400_codec = codec;
+
+ if (!schedule_work(&priv->work)) {
+ ret = -EINVAL;
+ goto err_regulator;
+ }
+
+ return 0;
+
+err_regulator:
+ wm8400_codec = NULL;
+ regulator_bulk_free(ARRAY_SIZE(power), power);
+err:
+ kfree(priv);
+ return ret;
+}
+
+static int __exit wm8400_codec_remove(struct platform_device *dev)
+{
+ struct wm8400_priv *priv = wm8400_codec->private_data;
+ u16 reg;
+
+ snd_soc_unregister_dai(&wm8400_dai);
+ snd_soc_unregister_codec(wm8400_codec);
+
+ reg = wm8400_read(wm8400_codec, WM8400_POWER_MANAGEMENT_1);
+ wm8400_write(wm8400_codec, WM8400_POWER_MANAGEMENT_1,
+ reg & (~WM8400_CODEC_ENA));
+
+ regulator_bulk_free(ARRAY_SIZE(power), power);
+ kfree(priv);
+
+ wm8400_codec = NULL;
+
+ return 0;
+}
+
+static struct platform_driver wm8400_codec_driver = {
+ .driver = {
+ .name = "wm8400-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8400_codec_probe,
+ .remove = __exit_p(wm8400_codec_remove),
+};
+
+static int __init wm8400_codec_init(void)
+{
+ return platform_driver_register(&wm8400_codec_driver);
+}
+module_init(wm8400_codec_init);
+
+static void __exit wm8400_codec_exit(void)
+{
+ platform_driver_unregister(&wm8400_codec_driver);
+}
+module_exit(wm8400_codec_exit);
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8400);
+
+MODULE_DESCRIPTION("ASoC WM8400 driver");
+MODULE_AUTHOR("Mark Brown");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm8400-codec");
diff --git a/sound/soc/codecs/wm8400.h b/sound/soc/codecs/wm8400.h
new file mode 100644
index 0000000..79c5934
--- /dev/null
+++ b/sound/soc/codecs/wm8400.h
@@ -0,0 +1,62 @@
+/*
+ * wm8400.h -- audio driver for WM8400
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.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 _WM8400_CODEC_H
+#define _WM8400_CODEC_H
+
+#define WM8400_MCLK_DIV 0
+#define WM8400_DACCLK_DIV 1
+#define WM8400_ADCCLK_DIV 2
+#define WM8400_BCLK_DIV 3
+
+#define WM8400_MCLK_DIV_1 0x400
+#define WM8400_MCLK_DIV_2 0x800
+
+#define WM8400_DAC_CLKDIV_1 0x00
+#define WM8400_DAC_CLKDIV_1_5 0x04
+#define WM8400_DAC_CLKDIV_2 0x08
+#define WM8400_DAC_CLKDIV_3 0x0c
+#define WM8400_DAC_CLKDIV_4 0x10
+#define WM8400_DAC_CLKDIV_5_5 0x14
+#define WM8400_DAC_CLKDIV_6 0x18
+
+#define WM8400_ADC_CLKDIV_1 0x00
+#define WM8400_ADC_CLKDIV_1_5 0x20
+#define WM8400_ADC_CLKDIV_2 0x40
+#define WM8400_ADC_CLKDIV_3 0x60
+#define WM8400_ADC_CLKDIV_4 0x80
+#define WM8400_ADC_CLKDIV_5_5 0xa0
+#define WM8400_ADC_CLKDIV_6 0xc0
+
+
+#define WM8400_BCLK_DIV_1 (0x0 << 1)
+#define WM8400_BCLK_DIV_1_5 (0x1 << 1)
+#define WM8400_BCLK_DIV_2 (0x2 << 1)
+#define WM8400_BCLK_DIV_3 (0x3 << 1)
+#define WM8400_BCLK_DIV_4 (0x4 << 1)
+#define WM8400_BCLK_DIV_5_5 (0x5 << 1)
+#define WM8400_BCLK_DIV_6 (0x6 << 1)
+#define WM8400_BCLK_DIV_8 (0x7 << 1)
+#define WM8400_BCLK_DIV_11 (0x8 << 1)
+#define WM8400_BCLK_DIV_12 (0x9 << 1)
+#define WM8400_BCLK_DIV_16 (0xA << 1)
+#define WM8400_BCLK_DIV_22 (0xB << 1)
+#define WM8400_BCLK_DIV_24 (0xC << 1)
+#define WM8400_BCLK_DIV_32 (0xD << 1)
+#define WM8400_BCLK_DIV_44 (0xE << 1)
+#define WM8400_BCLK_DIV_48 (0xF << 1)
+
+extern struct snd_soc_dai wm8400_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8400;
+
+#endif
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index cc975a6..6a4cea0 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -336,7 +336,7 @@
return 0;
}
- pll_factors(freq_out*8, freq_in);
+ pll_factors(freq_out*4, freq_in);
wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n);
wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18);
@@ -367,7 +367,7 @@
wm8510_write(codec, WM8510_GPIO, reg | div);
break;
case WM8510_MCLKDIV:
- reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f;
+ reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x11f;
wm8510_write(codec, WM8510_CLOCK, reg | div);
break;
case WM8510_ADCCLK:
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index ee0af23..27f9e23 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -35,8 +35,6 @@
#include "wm8580.h"
-#define WM8580_VERSION "0.1"
-
struct pll_state {
unsigned int in;
unsigned int out;
@@ -976,8 +974,6 @@
struct wm8580_priv *wm8580;
int ret = 0;
- pr_info("WM8580 Audio Codec %s\n", WM8580_VERSION);
-
setup = socdev->codec_data;
codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (codec == NULL)
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index cc6e57f..a6e8f3f 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -51,11 +51,6 @@
#include "wm8753.h"
-#ifdef CONFIG_SPI_MASTER
-static struct spi_driver wm8753_spi_driver;
-static int wm8753_spi_write(struct spi_device *spi, const char *data, int len);
-#endif
-
static int caps_charge = 2000;
module_param(caps_charge, int, 0);
MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index b502741..bd7392c 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -20,7 +20,7 @@
config SND_DAVINCI_SOC_SFFSDR
tristate "SoC Audio support for SFFSDR"
- depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR
+ depends on SND_DAVINCI_SOC && MACH_SFFSDR
select SND_DAVINCI_SOC_I2S
select SND_SOC_PCM3008
select SFFSDR_FPGA
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
index 0bf81ab..40eccfe 100644
--- a/sound/soc/davinci/davinci-sffsdr.c
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -36,9 +36,16 @@
#include "davinci-pcm.h"
#include "davinci-i2s.h"
+/*
+ * CLKX and CLKR are the inputs for the Sample Rate Generator.
+ * FSX and FSR are outputs, driven by the sample Rate Generator.
+ */
+#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \
+ SND_SOC_DAIFMT_CBM_CFS | \
+ SND_SOC_DAIFMT_IB_NF)
+
static int sffsdr_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *dai)
+ 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;
@@ -56,13 +63,8 @@
}
#endif
- /* Set cpu DAI configuration:
- * CLKX and CLKR are the inputs for the Sample Rate Generator.
- * FSX and FSR are outputs, driven by the sample Rate Generator. */
- ret = snd_soc_dai_set_fmt(cpu_dai,
- SND_SOC_DAIFMT_RIGHT_J |
- SND_SOC_DAIFMT_CBM_CFS |
- SND_SOC_DAIFMT_IB_NF);
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT);
if (ret < 0)
return ret;
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 58a3fa4..b3eb857 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -142,7 +142,8 @@
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_JOINT_DUPLEX,
+ SNDRV_PCM_INFO_JOINT_DUPLEX |
+ SNDRV_PCM_INFO_PAUSE,
.formats = FSLDMA_PCM_FORMATS,
.rates = FSLDMA_PCM_RATES,
.rate_min = 5512,
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 0fddd43..169bca2 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -72,6 +72,7 @@
* @dev: struct device pointer
* @playback: the number of playback streams opened
* @capture: the number of capture streams opened
+ * @asynchronous: 0=synchronous mode, 1=asynchronous mode
* @cpu_dai: the CPU DAI for this device
* @dev_attr: the sysfs device attribute structure
* @stats: SSI statistics
@@ -86,6 +87,7 @@
struct device *dev;
unsigned int playback;
unsigned int capture;
+ int asynchronous;
struct snd_soc_dai cpu_dai;
struct device_attribute dev_attr;
@@ -301,9 +303,10 @@
*
* FIXME: Little-endian samples require a different shift dir
*/
- clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK,
- CCSR_SSI_SCR_TFR_CLK_DIS |
- CCSR_SSI_SCR_I2S_MODE_SLAVE | CCSR_SSI_SCR_SYN);
+ clrsetbits_be32(&ssi->scr,
+ CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN,
+ CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE
+ | (ssi_private->asynchronous ? 0 : CCSR_SSI_SCR_SYN));
out_be32(&ssi->stcr,
CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
@@ -382,10 +385,15 @@
SNDRV_PCM_HW_PARAM_RATE,
first_runtime->rate, first_runtime->rate);
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
- first_runtime->sample_bits,
- first_runtime->sample_bits);
+ /* If we're in synchronous mode, then we need to constrain
+ * the sample size as well. We don't support independent sample
+ * rates in asynchronous mode.
+ */
+ if (!ssi_private->asynchronous)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ first_runtime->sample_bits,
+ first_runtime->sample_bits);
ssi_private->second_stream = substream;
}
@@ -421,13 +429,18 @@
struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
unsigned int sample_size =
snd_pcm_format_width(params_format(hw_params));
- u32 wl;
+ u32 wl = CCSR_SSI_SxCCR_WL(sample_size);
/* The SSI should always be disabled at this points (SSIEN=0) */
- wl = CCSR_SSI_SxCCR_WL(sample_size);
/* In synchronous mode, the SSI uses STCCR for capture */
- clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
+ if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ||
+ !ssi_private->asynchronous)
+ clrsetbits_be32(&ssi->stccr,
+ CCSR_SSI_SxCCR_WL_MASK, wl);
+ else
+ clrsetbits_be32(&ssi->srccr,
+ CCSR_SSI_SxCCR_WL_MASK, wl);
}
return 0;
@@ -451,28 +464,33 @@
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_RESUME:
+ clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
setbits32(&ssi->scr,
CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
} else {
- clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+ long timeout = jiffies + 10;
+
setbits32(&ssi->scr,
CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
- /*
- * I think we need this delay to allow time for the SSI
- * to put data into its FIFO. Without it, ALSA starts
- * to complain about overruns.
+ /* Wait until the SSI has filled its FIFO. Without this
+ * delay, ALSA complains about overruns. When the FIFO
+ * is full, the DMA controller initiates its first
+ * transfer. Until then, however, the DMA's DAR
+ * register is zero, which translates to an
+ * out-of-bounds pointer. This makes ALSA think an
+ * overrun has occurred.
*/
- mdelay(1);
+ while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) &&
+ (jiffies < timeout));
+ if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0))
+ return -EIO;
}
break;
case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
clrbits32(&ssi->scr, CCSR_SSI_SCR_TE);
@@ -655,6 +673,7 @@
ssi_private->ssi_phys = ssi_info->ssi_phys;
ssi_private->irq = ssi_info->irq;
ssi_private->dev = ssi_info->dev;
+ ssi_private->asynchronous = ssi_info->asynchronous;
ssi_private->dev->driver_data = fsl_ssi_dai;
@@ -705,6 +724,14 @@
}
EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai);
+static int __init fsl_ssi_init(void)
+{
+ printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n");
+
+ return 0;
+}
+module_init(fsl_ssi_init);
+
MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h
index 83b44d7..eade01f 100644
--- a/sound/soc/fsl/fsl_ssi.h
+++ b/sound/soc/fsl/fsl_ssi.h
@@ -208,6 +208,7 @@
* ssi_phys: physical address of the SSI registers
* irq: IRQ of this SSI
* dev: struct device, used to create the sysfs statistics file
+ * asynchronous: 0=synchronous mode, 1=asynchronous mode
*/
struct fsl_ssi_info {
unsigned int id;
@@ -215,6 +216,7 @@
dma_addr_t ssi_phys;
unsigned int irq;
struct device *dev;
+ int asynchronous;
};
struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info);
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index acf39a6..ef67d1c 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -353,6 +353,11 @@
}
ssi_info.irq = machine_data->ssi_irq;
+ /* Do we want to use asynchronous mode? */
+ ssi_info.asynchronous =
+ of_find_property(np, "fsl,ssi-asynchronous", NULL) ? 1 : 0;
+ if (ssi_info.asynchronous)
+ dev_info(&ofdev->dev, "using asynchronous mode\n");
/* Map the global utilities registers. */
guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts");
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index cd41a94..a952a4e 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -186,13 +186,6 @@
return -ENODEV;
}
- if (clk_get_usecount(tlv320aic23_mclk) > 0) {
- /* MCLK is already in use */
- printk(KERN_WARNING
- "MCLK in use at %d Hz. We change it to %d Hz\n",
- (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
- }
-
/*
* Configure 12 MHz output on MCLK.
*/
@@ -205,9 +198,8 @@
}
}
- printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n",
- (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK,
- clk_get_usecount(tlv320aic23_mclk));
+ printk(KERN_INFO "MCLK = %d [%d]\n",
+ (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
return 0;
err1:
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index e226fa7..715c648 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -28,6 +28,7 @@
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
+#include <sound/jack.h>
#include <asm/mach-types.h>
#include <mach/hardware.h>
@@ -81,12 +82,97 @@
.hw_params = sdp3430_hw_params,
};
+/* SDP3430 machine DAPM */
+static const struct snd_soc_dapm_widget sdp3430_twl4030_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Ext Mic", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_HP("Headset Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* External Mics: MAINMIC, SUBMIC with bias*/
+ {"MAINMIC", NULL, "Mic Bias 1"},
+ {"SUBMIC", NULL, "Mic Bias 2"},
+ {"Mic Bias 1", NULL, "Ext Mic"},
+ {"Mic Bias 2", NULL, "Ext Mic"},
+
+ /* External Speakers: HFL, HFR */
+ {"Ext Spk", NULL, "HFL"},
+ {"Ext Spk", NULL, "HFR"},
+
+ /* Headset: HSMIC (with bias), HSOL, HSOR */
+ {"Headset Jack", NULL, "HSOL"},
+ {"Headset Jack", NULL, "HSOR"},
+ {"HSMIC", NULL, "Headset Mic Bias"},
+ {"Headset Mic Bias", NULL, "Headset Jack"},
+};
+
+static int sdp3430_twl4030_init(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ /* Add SDP3430 specific widgets */
+ ret = snd_soc_dapm_new_controls(codec, sdp3430_twl4030_dapm_widgets,
+ ARRAY_SIZE(sdp3430_twl4030_dapm_widgets));
+ if (ret)
+ return ret;
+
+ /* Set up SDP3430 specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ /* SDP3430 connected pins */
+ snd_soc_dapm_enable_pin(codec, "Ext Mic");
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+
+ /* TWL4030 not connected pins */
+ snd_soc_dapm_nc_pin(codec, "AUXL");
+ snd_soc_dapm_nc_pin(codec, "AUXR");
+ snd_soc_dapm_nc_pin(codec, "CARKITMIC");
+ snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
+ snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
+
+ snd_soc_dapm_nc_pin(codec, "OUTL");
+ snd_soc_dapm_nc_pin(codec, "OUTR");
+ snd_soc_dapm_nc_pin(codec, "EARPIECE");
+ snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
+ snd_soc_dapm_nc_pin(codec, "PREDRIVER");
+ snd_soc_dapm_nc_pin(codec, "CARKITL");
+ snd_soc_dapm_nc_pin(codec, "CARKITR");
+
+ ret = snd_soc_dapm_sync(codec);
+
+ return ret;
+}
+
+/* Headset jack */
+static struct snd_soc_jack hs_jack;
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+ {
+ .pin = "Headset Jack",
+ .mask = SND_JACK_HEADSET,
+ },
+};
+
+/* Headset jack detection gpios */
+static struct snd_soc_jack_gpio hs_jack_gpios[] = {
+ {
+ .gpio = (OMAP_MAX_GPIO_LINES + 2),
+ .name = "hsdet-gpio",
+ .report = SND_JACK_HEADSET,
+ .debounce_time = 200,
+ },
+};
+
/* Digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link sdp3430_dai = {
.name = "TWL4030",
.stream_name = "TWL4030",
.cpu_dai = &omap_mcbsp_dai[0],
.codec_dai = &twl4030_dai,
+ .init = sdp3430_twl4030_init,
.ops = &sdp3430_ops,
};
@@ -130,7 +216,21 @@
if (ret)
goto err1;
- return 0;
+ /* Headset jack detection */
+ ret = snd_soc_jack_new(&snd_soc_sdp3430, "SDP3430 headset jack",
+ SND_JACK_HEADSET, &hs_jack);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+ hs_jack_pins);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
+ hs_jack_gpios);
+
+ return ret;
err1:
printk(KERN_ERR "Unable to add platform device\n");
@@ -142,6 +242,9 @@
static void __exit sdp3430_soc_exit(void)
{
+ snd_soc_jack_free_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
+ hs_jack_gpios);
+
platform_device_unregister(sdp3430_snd_device);
}
module_exit(sdp3430_soc_exit);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 3e18064..d3fa635 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -300,7 +300,7 @@
int val;
u32 sscr0 = ssp_read_reg(ssp, SSCR0) &
- ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+ ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
dev_dbg(&ssp->pdev->dev,
"pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n",
@@ -328,7 +328,7 @@
case PXA_SSP_CLK_AUDIO:
priv->sysclk = 0;
ssp_set_scr(&priv->dev, 1);
- sscr0 |= SSCR0_ADC;
+ sscr0 |= SSCR0_ACS;
break;
default:
return -ENODEV;
@@ -522,9 +522,20 @@
u32 sscr1;
u32 sspsp;
+ /* check if we need to change anything at all */
+ if (priv->dai_fmt == fmt)
+ return 0;
+
+ /* we can only change the settings if the port is not in use */
+ if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) {
+ dev_err(&ssp->pdev->dev,
+ "can't change hardware dai format: stream is in use");
+ return -EINVAL;
+ }
+
/* reset port settings */
sscr0 = ssp_read_reg(ssp, SSCR0) &
- (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+ (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7);
sspsp = 0;
@@ -644,8 +655,7 @@
sscr0 |= SSCR0_FPCKE;
#endif
sscr0 |= SSCR0_DataSize(16);
- if (params_channels(params) > 1)
- sscr0 |= SSCR0_EDSS;
+ /* use network mode (2 slots) for 16 bit stereo */
break;
case SNDRV_PCM_FORMAT_S24_LE:
sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8));
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 11cd0f2..cf80904 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -106,13 +106,13 @@
static int pxa2xx_ac97_probe(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
- return pxa2xx_ac97_hw_probe(pdev);
+ return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev));
}
static void pxa2xx_ac97_remove(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
- pxa2xx_ac97_hw_remove(pdev);
+ pxa2xx_ac97_hw_remove(to_platform_device(dai->dev));
}
static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
@@ -230,15 +230,45 @@
EXPORT_SYMBOL_GPL(pxa_ac97_dai);
EXPORT_SYMBOL_GPL(soc_ac97_ops);
+static int __devinit pxa2xx_ac97_dev_probe(struct platform_device *pdev)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(pxa_ac97_dai); i++)
+ pxa_ac97_dai[i].dev = &pdev->dev;
+
+ /* Punt most of the init to the SoC probe; we may need the machine
+ * driver to do interesting things with the clocking to get us up
+ * and running.
+ */
+ return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+}
+
+static int __devexit pxa2xx_ac97_dev_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+
+ return 0;
+}
+
+static struct platform_driver pxa2xx_ac97_driver = {
+ .probe = pxa2xx_ac97_dev_probe,
+ .remove = __devexit_p(pxa2xx_ac97_dev_remove),
+ .driver = {
+ .name = "pxa2xx-ac97",
+ .owner = THIS_MODULE,
+ },
+};
+
static int __init pxa_ac97_init(void)
{
- return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+ return platform_driver_register(&pxa2xx_ac97_driver);
}
module_init(pxa_ac97_init);
static void __exit pxa_ac97_exit(void)
{
- snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+ platform_driver_unregister(&pxa2xx_ac97_driver);
}
module_exit(pxa_ac97_exit);
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index 0140a25..9f6116e 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -127,8 +127,11 @@
if (ret < 0)
return ret;
- /* We're not really in network mode but the emulation wants this. */
- ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1);
+ /* Use network mode for stereo, one slot per channel. */
+ if (params_channels(params) > 1)
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 2);
+ else
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1);
if (ret < 0)
return ret;
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index e05a710..2f3a21e 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,19 +1,31 @@
config SND_S3C24XX_SOC
- tristate "SoC Audio for the Samsung S3C24XX chips"
- depends on ARCH_S3C2410
+ tristate "SoC Audio for the Samsung S3CXXXX chips"
+ depends on ARCH_S3C2410 || ARCH_S3C64XX
help
Say Y or M if you want to add support for codecs attached to
- the S3C24XX AC97, I2S or SSP interface. You will also need
- to select the audio interfaces to support below.
+ the S3C24XX and S3C64XX AC97, I2S or SSP interface. You will
+ also need to select the audio interfaces to support below.
config SND_S3C24XX_SOC_I2S
tristate
+ select S3C2410_DMA
+
+config SND_S3C_I2SV2_SOC
+ tristate
config SND_S3C2412_SOC_I2S
tristate
+ select SND_S3C_I2SV2_SOC
+ select S3C2410_DMA
+
+config SND_S3C64XX_SOC_I2S
+ tristate
+ select SND_S3C_I2SV2_SOC
+ select S3C64XX_DMA
config SND_S3C2443_SOC_AC97
tristate
+ select S3C2410_DMA
select AC97_BUS
select SND_SOC_AC97_BUS
@@ -26,6 +38,14 @@
Say Y if you want to add support for SoC audio on smdk2440
with the WM8753.
+config SND_S3C24XX_SOC_JIVE_WM8750
+ tristate "SoC I2S Audio support for Jive"
+ depends on SND_S3C24XX_SOC && MACH_JIVE
+ select SND_SOC_WM8750
+ select SND_S3C2412_SOC_I2S
+ help
+ Sat Y if you want to add support for SoC audio on the Jive.
+
config SND_S3C24XX_SOC_SMDK2443_WM9710
tristate "SoC AC97 Audio support for SMDK2443 - WM9710"
depends on SND_S3C24XX_SOC && MACH_SMDK2443
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 96b3f3f..07a93a2 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -2,19 +2,25 @@
snd-soc-s3c24xx-objs := s3c24xx-pcm.o
snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o
+snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o
snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o
+snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o
obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o
obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o
obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o
obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
+obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o
+obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o
# S3C24XX Machine Support
+snd-soc-jive-wm8750-objs := jive_wm8750.o
snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.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
diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c
new file mode 100644
index 0000000..3206379
--- /dev/null
+++ b/sound/soc/s3c24xx/jive_wm8750.c
@@ -0,0 +1,201 @@
+/* sound/soc/s3c24xx/jive_wm8750.c
+ *
+ * Copyright 2007,2008 Simtec Electronics
+ *
+ * Based on sound/soc/pxa/spitz.c
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * 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/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c2412-i2s.h"
+
+#include "../codecs/wm8750.h"
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ { "Headphone Jack", NULL, "LOUT1" },
+ { "Headphone Jack", NULL, "ROUT1" },
+ { "Internal Speaker", NULL, "LOUT2" },
+ { "Internal Speaker", NULL, "ROUT2" },
+ { "LINPUT1", NULL, "Line Input" },
+ { "RINPUT1", NULL, "Line Input" },
+};
+
+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Internal Speaker", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static int jive_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;
+ struct s3c_i2sv2_rate_calc div;
+ unsigned int clk = 0;
+ int ret = 0;
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ clk = 12288000;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 11289600;
+ break;
+ }
+
+ s3c_i2sv2_calc_rate(&div, NULL, params_rate(params),
+ s3c2412_get_iisclk());
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER,
+ div.clk_div - 1);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops jive_ops = {
+ .hw_params = jive_hw_params,
+};
+
+static int jive_wm8750_init(struct snd_soc_codec *codec)
+{
+ int err;
+
+ /* These endpoints are not being used. */
+ snd_soc_dapm_nc_pin(codec, "LINPUT2");
+ snd_soc_dapm_nc_pin(codec, "RINPUT2");
+ snd_soc_dapm_nc_pin(codec, "LINPUT3");
+ snd_soc_dapm_nc_pin(codec, "RINPUT3");
+ snd_soc_dapm_nc_pin(codec, "OUT3");
+ snd_soc_dapm_nc_pin(codec, "MONO");
+
+ /* Add jive specific widgets */
+ err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+ ARRAY_SIZE(wm8750_dapm_widgets));
+ if (err) {
+ printk(KERN_ERR "%s: failed to add widgets (%d)\n",
+ __func__, err);
+ return err;
+ }
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static struct snd_soc_dai_link jive_dai = {
+ .name = "wm8750",
+ .stream_name = "WM8750",
+ .cpu_dai = &s3c2412_i2s_dai,
+ .codec_dai = &wm8750_dai,
+ .init = jive_wm8750_init,
+ .ops = &jive_ops,
+};
+
+/* jive audio machine driver */
+static struct snd_soc_machine snd_soc_machine_jive = {
+ .name = "Jive",
+ .dai_link = &jive_dai,
+ .num_links = 1,
+};
+
+/* jive audio private data */
+static struct wm8750_setup_data jive_wm8750_setup = {
+};
+
+/* jive audio subsystem */
+static struct snd_soc_device jive_snd_devdata = {
+ .machine = &snd_soc_machine_jive,
+ .platform = &s3c24xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm8750_spi,
+ .codec_data = &jive_wm8750_setup,
+};
+
+static struct platform_device *jive_snd_device;
+
+static int __init jive_init(void)
+{
+ int ret;
+
+ if (!machine_is_jive())
+ return 0;
+
+ printk("JIVE WM8750 Audio support\n");
+
+ jive_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!jive_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(jive_snd_device, &jive_snd_devdata);
+ jive_snd_devdata.dev = &jive_snd_device->dev;
+ ret = platform_device_add(jive_snd_device);
+
+ if (ret)
+ platform_device_put(jive_snd_device);
+
+ return ret;
+}
+
+static void __exit jive_exit(void)
+{
+ platform_device_unregister(jive_snd_device);
+}
+
+module_init(jive_init);
+module_exit(jive_exit);
+
+MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC Jive Audio support");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 286e11a..5f6aeec 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -29,25 +29,17 @@
#include <mach/regs-clock.h>
#include <mach/regs-gpio.h>
#include <mach/hardware.h>
-#include <mach/audio.h>
+#include <plat/audio.h>
#include <linux/io.h>
#include <mach/spi-gpio.h>
-#include <asm/plat-s3c24xx/regs-iis.h>
+#include <plat/regs-iis.h>
#include "../codecs/wm8753.h"
#include "lm4857.h"
#include "s3c24xx-pcm.h"
#include "s3c24xx-i2s.h"
-/* Debugging stuff */
-#define S3C24XX_SOC_NEO1973_WM8753_DEBUG 0
-#if S3C24XX_SOC_NEO1973_WM8753_DEBUG
-#define DBG(x...) printk(KERN_DEBUG "s3c24xx-soc-neo1973-wm8753: " x)
-#else
-#define DBG(x...)
-#endif
-
/* define the scenarios */
#define NEO_AUDIO_OFF 0
#define NEO_GSM_CALL_AUDIO_HANDSET 1
@@ -72,7 +64,7 @@
int ret = 0;
unsigned long iis_clkrate;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
iis_clkrate = s3c24xx_i2s_get_clockrate();
@@ -158,7 +150,7 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
/* disable the PLL */
return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
@@ -181,7 +173,7 @@
int ret = 0;
unsigned long iis_clkrate;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
iis_clkrate = s3c24xx_i2s_get_clockrate();
@@ -224,7 +216,7 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
/* disable the PLL */
return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
@@ -246,7 +238,7 @@
static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
{
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
switch (neo1973_scenario) {
case NEO_AUDIO_OFF:
@@ -330,7 +322,7 @@
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
if (neo1973_scenario == ucontrol->value.integer.value[0])
return 0;
@@ -344,7 +336,7 @@
static void lm4857_write_regs(void)
{
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
if (i2c_master_send(i2c, lm4857_regs, 4) != 4)
printk(KERN_ERR "lm4857: i2c write failed\n");
@@ -357,7 +349,7 @@
int shift = (kcontrol->private_value >> 8) & 0x0F;
int mask = (kcontrol->private_value >> 16) & 0xFF;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask;
return 0;
@@ -385,7 +377,7 @@
{
u8 value = lm4857_regs[LM4857_CTRL] & 0x0F;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
if (value)
value -= 5;
@@ -399,7 +391,7 @@
{
u8 value = ucontrol->value.integer.value[0];
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
if (value)
value += 5;
@@ -508,7 +500,7 @@
{
int i, err;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
/* set up NC codec pins */
snd_soc_dapm_nc_pin(codec, "LOUT2");
@@ -593,7 +585,7 @@
static int lm4857_i2c_probe(struct i2c_client *client,
const struct i2c_device_id *id)
{
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
i2c = client;
@@ -603,7 +595,7 @@
static int lm4857_i2c_remove(struct i2c_client *client)
{
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
i2c = NULL;
@@ -614,7 +606,7 @@
static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
{
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
dev_dbg(&dev->dev, "lm4857_suspend\n");
lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf;
@@ -627,7 +619,7 @@
static int lm4857_resume(struct i2c_client *dev)
{
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
if (lm4857_state) {
lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f);
@@ -638,7 +630,7 @@
static void lm4857_shutdown(struct i2c_client *dev)
{
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
dev_dbg(&dev->dev, "lm4857_shutdown\n");
lm4857_regs[LM4857_CTRL] &= 0xf0;
@@ -669,7 +661,7 @@
{
int ret;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
if (!machine_is_neo1973_gta01()) {
printk(KERN_INFO
@@ -700,7 +692,7 @@
static void __exit neo1973_exit(void)
{
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
i2c_del_driver(&lm4857_i2c_driver);
platform_device_unregister(neo1973_snd_device);
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
new file mode 100644
index 0000000..295a4c9
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -0,0 +1,638 @@
+/* sound/soc/s3c24xx/s3c-i2c-v2.c
+ *
+ * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs.
+ *
+ * Copyright (c) 2006 Wolfson Microelectronics PLC.
+ * Graeme Gregory graeme.gregory@wolfsonmicro.com
+ * linux@wolfsonmicro.com
+ *
+ * Copyright (c) 2008, 2007, 2004-2005 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 as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/kernel.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <plat/regs-s3c2412-iis.h>
+
+#include <plat/audio.h>
+#include <mach/dma.h>
+
+#include "s3c-i2s-v2.h"
+
+#define S3C2412_I2S_DEBUG_CON 0
+
+static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
+{
+ return cpu_dai->private_data;
+}
+
+#define bit_set(v, b) (((v) & (b)) ? 1 : 0)
+
+#if S3C2412_I2S_DEBUG_CON
+static void dbg_showcon(const char *fn, u32 con)
+{
+ printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn,
+ bit_set(con, S3C2412_IISCON_LRINDEX),
+ bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY),
+ bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY),
+ bit_set(con, S3C2412_IISCON_TXFIFO_FULL),
+ bit_set(con, S3C2412_IISCON_RXFIFO_FULL));
+
+ printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n",
+ fn,
+ bit_set(con, S3C2412_IISCON_TXDMA_PAUSE),
+ bit_set(con, S3C2412_IISCON_RXDMA_PAUSE),
+ bit_set(con, S3C2412_IISCON_TXCH_PAUSE),
+ bit_set(con, S3C2412_IISCON_RXCH_PAUSE));
+ printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn,
+ bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE),
+ bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE),
+ bit_set(con, S3C2412_IISCON_IIS_ACTIVE));
+}
+#else
+static inline void dbg_showcon(const char *fn, u32 con)
+{
+}
+#endif
+
+
+/* Turn on or off the transmission path. */
+void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
+{
+ void __iomem *regs = i2s->regs;
+ u32 fic, con, mod;
+
+ pr_debug("%s(%d)\n", __func__, on);
+
+ fic = readl(regs + S3C2412_IISFIC);
+ con = readl(regs + S3C2412_IISCON);
+ mod = readl(regs + S3C2412_IISMOD);
+
+ pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+
+ if (on) {
+ con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE;
+ con &= ~S3C2412_IISCON_TXDMA_PAUSE;
+ con &= ~S3C2412_IISCON_TXCH_PAUSE;
+
+ switch (mod & S3C2412_IISMOD_MODE_MASK) {
+ case S3C2412_IISMOD_MODE_TXONLY:
+ case S3C2412_IISMOD_MODE_TXRX:
+ /* do nothing, we are in the right mode */
+ break;
+
+ case S3C2412_IISMOD_MODE_RXONLY:
+ mod &= ~S3C2412_IISMOD_MODE_MASK;
+ mod |= S3C2412_IISMOD_MODE_TXRX;
+ break;
+
+ default:
+ dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n");
+ }
+
+ writel(con, regs + S3C2412_IISCON);
+ writel(mod, regs + S3C2412_IISMOD);
+ } else {
+ /* Note, we do not have any indication that the FIFO problems
+ * tha the S3C2410/2440 had apply here, so we should be able
+ * to disable the DMA and TX without resetting the FIFOS.
+ */
+
+ con |= S3C2412_IISCON_TXDMA_PAUSE;
+ con |= S3C2412_IISCON_TXCH_PAUSE;
+ con &= ~S3C2412_IISCON_TXDMA_ACTIVE;
+
+ switch (mod & S3C2412_IISMOD_MODE_MASK) {
+ case S3C2412_IISMOD_MODE_TXRX:
+ mod &= ~S3C2412_IISMOD_MODE_MASK;
+ mod |= S3C2412_IISMOD_MODE_RXONLY;
+ break;
+
+ case S3C2412_IISMOD_MODE_TXONLY:
+ mod &= ~S3C2412_IISMOD_MODE_MASK;
+ con &= ~S3C2412_IISCON_IIS_ACTIVE;
+ break;
+
+ default:
+ dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n");
+ }
+
+ writel(mod, regs + S3C2412_IISMOD);
+ writel(con, regs + S3C2412_IISCON);
+ }
+
+ fic = readl(regs + S3C2412_IISFIC);
+ dbg_showcon(__func__, con);
+ pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+}
+EXPORT_SYMBOL_GPL(s3c2412_snd_txctrl);
+
+void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
+{
+ void __iomem *regs = i2s->regs;
+ u32 fic, con, mod;
+
+ pr_debug("%s(%d)\n", __func__, on);
+
+ fic = readl(regs + S3C2412_IISFIC);
+ con = readl(regs + S3C2412_IISCON);
+ mod = readl(regs + S3C2412_IISMOD);
+
+ pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+
+ if (on) {
+ con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE;
+ con &= ~S3C2412_IISCON_RXDMA_PAUSE;
+ con &= ~S3C2412_IISCON_RXCH_PAUSE;
+
+ switch (mod & S3C2412_IISMOD_MODE_MASK) {
+ case S3C2412_IISMOD_MODE_TXRX:
+ case S3C2412_IISMOD_MODE_RXONLY:
+ /* do nothing, we are in the right mode */
+ break;
+
+ case S3C2412_IISMOD_MODE_TXONLY:
+ mod &= ~S3C2412_IISMOD_MODE_MASK;
+ mod |= S3C2412_IISMOD_MODE_TXRX;
+ break;
+
+ default:
+ dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+ }
+
+ writel(mod, regs + S3C2412_IISMOD);
+ writel(con, regs + S3C2412_IISCON);
+ } else {
+ /* See txctrl notes on FIFOs. */
+
+ con &= ~S3C2412_IISCON_RXDMA_ACTIVE;
+ con |= S3C2412_IISCON_RXDMA_PAUSE;
+ con |= S3C2412_IISCON_RXCH_PAUSE;
+
+ switch (mod & S3C2412_IISMOD_MODE_MASK) {
+ case S3C2412_IISMOD_MODE_RXONLY:
+ con &= ~S3C2412_IISCON_IIS_ACTIVE;
+ mod &= ~S3C2412_IISMOD_MODE_MASK;
+ break;
+
+ case S3C2412_IISMOD_MODE_TXRX:
+ mod &= ~S3C2412_IISMOD_MODE_MASK;
+ mod |= S3C2412_IISMOD_MODE_TXONLY;
+ break;
+
+ default:
+ dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+ }
+
+ writel(con, regs + S3C2412_IISCON);
+ writel(mod, regs + S3C2412_IISMOD);
+ }
+
+ fic = readl(regs + S3C2412_IISFIC);
+ pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+}
+EXPORT_SYMBOL_GPL(s3c2412_snd_rxctrl);
+
+/*
+ * 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);
+
+ pr_debug("Entered %s\n", __func__);
+
+ while (1) {
+ iiscon = readl(i2s->regs + S3C2412_IISCON);
+ if (iiscon & S3C2412_IISCON_LRINDEX)
+ break;
+
+ if (timeout < jiffies) {
+ printk(KERN_ERR "%s: timeout\n", __func__);
+ return -ETIMEDOUT;
+ }
+ }
+
+ return 0;
+}
+
+/*
+ * Set S3C2412 I2S DAI format
+ */
+static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
+ u32 iismod;
+
+ pr_debug("Entered %s\n", __func__);
+
+ iismod = readl(i2s->regs + S3C2412_IISMOD);
+ pr_debug("hw_params r: IISMOD: %x \n", iismod);
+
+#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
+#define IISMOD_MASTER_MASK S3C2412_IISMOD_MASTER_MASK
+#define IISMOD_SLAVE S3C2412_IISMOD_SLAVE
+#define IISMOD_MASTER S3C2412_IISMOD_MASTER_INTERNAL
+#endif
+
+#if defined(CONFIG_PLAT_S3C64XX)
+/* From Rev1.1 datasheet, we have two master and two slave modes:
+ * IMS[11:10]:
+ * 00 = master mode, fed from PCLK
+ * 01 = master mode, fed from CLKAUDIO
+ * 10 = slave mode, using PCLK
+ * 11 = slave mode, using I2SCLK
+ */
+#define IISMOD_MASTER_MASK (1 << 11)
+#define IISMOD_SLAVE (1 << 11)
+#define IISMOD_MASTER (0x0)
+#endif
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ i2s->master = 0;
+ iismod &= ~IISMOD_MASTER_MASK;
+ iismod |= IISMOD_SLAVE;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ i2s->master = 1;
+ iismod &= ~IISMOD_MASTER_MASK;
+ iismod |= IISMOD_MASTER;
+ break;
+ default:
+ pr_debug("unknwon master/slave format\n");
+ return -EINVAL;
+ }
+
+ iismod &= ~S3C2412_IISMOD_SDF_MASK;
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_RIGHT_J:
+ iismod |= S3C2412_IISMOD_SDF_MSB;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iismod |= S3C2412_IISMOD_SDF_LSB;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ iismod |= S3C2412_IISMOD_SDF_IIS;
+ break;
+ default:
+ pr_debug("Unknown data format\n");
+ return -EINVAL;
+ }
+
+ writel(iismod, i2s->regs + S3C2412_IISMOD);
+ pr_debug("hw_params w: IISMOD: %x \n", iismod);
+ return 0;
+}
+
+static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *socdai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai_link *dai = rtd->dai;
+ struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai);
+ u32 iismod;
+
+ pr_debug("Entered %s\n", __func__);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dai->cpu_dai->dma_data = i2s->dma_playback;
+ else
+ dai->cpu_dai->dma_data = i2s->dma_capture;
+
+ /* Working copies of register */
+ iismod = readl(i2s->regs + S3C2412_IISMOD);
+ pr_debug("%s: r: IISMOD: %x\n", __func__, iismod);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ iismod |= S3C2412_IISMOD_8BIT;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ iismod &= ~S3C2412_IISMOD_8BIT;
+ break;
+ }
+
+ writel(iismod, i2s->regs + S3C2412_IISMOD);
+ pr_debug("%s: w: IISMOD: %x\n", __func__, iismod);
+ return 0;
+}
+
+static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct s3c_i2sv2_info *i2s = to_info(rtd->dai->cpu_dai);
+ int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
+ unsigned long irqs;
+ int ret = 0;
+
+ pr_debug("Entered %s\n", __func__);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ /* On start, ensure that the FIFOs are cleared and reset. */
+
+ writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH,
+ i2s->regs + S3C2412_IISFIC);
+
+ /* clear again, just in case */
+ writel(0x0, i2s->regs + S3C2412_IISFIC);
+
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (!i2s->master) {
+ ret = s3c2412_snd_lrsync(i2s);
+ if (ret)
+ goto exit_err;
+ }
+
+ local_irq_save(irqs);
+
+ if (capture)
+ s3c2412_snd_rxctrl(i2s, 1);
+ else
+ s3c2412_snd_txctrl(i2s, 1);
+
+ local_irq_restore(irqs);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ local_irq_save(irqs);
+
+ if (capture)
+ s3c2412_snd_rxctrl(i2s, 0);
+ else
+ s3c2412_snd_txctrl(i2s, 0);
+
+ local_irq_restore(irqs);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+exit_err:
+ return ret;
+}
+
+/*
+ * Set S3C2412 Clock dividers
+ */
+static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
+ int div_id, int div)
+{
+ struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
+ u32 reg;
+
+ pr_debug("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div);
+
+ switch (div_id) {
+ case S3C_I2SV2_DIV_BCLK:
+ reg = readl(i2s->regs + S3C2412_IISMOD);
+ reg &= ~S3C2412_IISMOD_BCLK_MASK;
+ writel(reg | div, i2s->regs + S3C2412_IISMOD);
+
+ pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD));
+ break;
+
+ case S3C_I2SV2_DIV_RCLK:
+ if (div > 3) {
+ /* convert value to bit field */
+
+ switch (div) {
+ case 256:
+ div = S3C2412_IISMOD_RCLK_256FS;
+ break;
+
+ case 384:
+ div = S3C2412_IISMOD_RCLK_384FS;
+ break;
+
+ case 512:
+ div = S3C2412_IISMOD_RCLK_512FS;
+ break;
+
+ case 768:
+ div = S3C2412_IISMOD_RCLK_768FS;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ }
+
+ reg = readl(i2s->regs + S3C2412_IISMOD);
+ reg &= ~S3C2412_IISMOD_RCLK_MASK;
+ writel(reg | div, i2s->regs + S3C2412_IISMOD);
+ pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD));
+ break;
+
+ case S3C_I2SV2_DIV_PRESCALER:
+ if (div >= 0) {
+ writel((div << 8) | S3C2412_IISPSR_PSREN,
+ i2s->regs + S3C2412_IISPSR);
+ } else {
+ writel(0x0, i2s->regs + S3C2412_IISPSR);
+ }
+ pr_debug("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR));
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+/* default table of all avaialable root fs divisors */
+static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 };
+
+int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
+ unsigned int *fstab,
+ unsigned int rate, struct clk *clk)
+{
+ unsigned long clkrate = clk_get_rate(clk);
+ unsigned int div;
+ unsigned int fsclk;
+ unsigned int actual;
+ unsigned int fs;
+ unsigned int fsdiv;
+ signed int deviation = 0;
+ unsigned int best_fs = 0;
+ unsigned int best_div = 0;
+ unsigned int best_rate = 0;
+ unsigned int best_deviation = INT_MAX;
+
+ if (fstab == NULL)
+ fstab = iis_fs_tab;
+
+ for (fs = 0; fs < ARRAY_SIZE(iis_fs_tab); fs++) {
+ fsdiv = iis_fs_tab[fs];
+
+ fsclk = clkrate / fsdiv;
+ div = fsclk / rate;
+
+ if ((fsclk % rate) > (rate / 2))
+ div++;
+
+ if (div <= 1)
+ continue;
+
+ actual = clkrate / (fsdiv * div);
+ deviation = actual - rate;
+
+ printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n",
+ fsdiv, div, actual, deviation);
+
+ deviation = abs(deviation);
+
+ if (deviation < best_deviation) {
+ best_fs = fsdiv;
+ best_div = div;
+ best_rate = actual;
+ best_deviation = deviation;
+ }
+
+ if (deviation == 0)
+ break;
+ }
+
+ printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n",
+ best_fs, best_div, best_rate);
+
+ info->fs_div = best_fs;
+ info->clk_div = best_div;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate);
+
+int s3c_i2sv2_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai,
+ struct s3c_i2sv2_info *i2s,
+ unsigned long base)
+{
+ struct device *dev = &pdev->dev;
+
+ i2s->dev = dev;
+
+ /* record our i2s structure for later use in the callbacks */
+ dai->private_data = i2s;
+
+ i2s->regs = ioremap(base, 0x100);
+ if (i2s->regs == NULL) {
+ dev_err(dev, "cannot ioremap registers\n");
+ return -ENXIO;
+ }
+
+ i2s->iis_pclk = clk_get(dev, "iis");
+ if (i2s->iis_pclk == NULL) {
+ dev_err(dev, "failed to get iis_clock\n");
+ iounmap(i2s->regs);
+ return -ENOENT;
+ }
+
+ clk_enable(i2s->iis_pclk);
+
+ s3c2412_snd_txctrl(i2s, 0);
+ s3c2412_snd_rxctrl(i2s, 0);
+
+ return 0;
+}
+
+EXPORT_SYMBOL_GPL(s3c_i2sv2_probe);
+
+#ifdef CONFIG_PM
+static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
+{
+ struct s3c_i2sv2_info *i2s = to_info(dai);
+ u32 iismod;
+
+ if (dai->active) {
+ i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD);
+ i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON);
+ i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR);
+
+ /* some basic suspend checks */
+
+ iismod = readl(i2s->regs + S3C2412_IISMOD);
+
+ if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
+ pr_warning("%s: RXDMA active?\n", __func__);
+
+ if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
+ pr_warning("%s: TXDMA active?\n", __func__);
+
+ if (iismod & S3C2412_IISCON_IIS_ACTIVE)
+ pr_warning("%s: IIS active\n", __func__);
+ }
+
+ return 0;
+}
+
+static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
+{
+ struct s3c_i2sv2_info *i2s = to_info(dai);
+
+ pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n",
+ dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
+
+ if (dai->active) {
+ writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
+ writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD);
+ writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR);
+
+ writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH,
+ i2s->regs + S3C2412_IISFIC);
+
+ ndelay(250);
+ writel(0x0, i2s->regs + S3C2412_IISFIC);
+ }
+
+ return 0;
+}
+#else
+#define s3c2412_i2s_suspend NULL
+#define s3c2412_i2s_resume NULL
+#endif
+
+int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
+{
+ dai->ops.trigger = s3c2412_i2s_trigger;
+ dai->ops.hw_params = s3c2412_i2s_hw_params;
+ dai->ops.set_fmt = s3c2412_i2s_set_fmt;
+ dai->ops.set_clkdiv = s3c2412_i2s_set_clkdiv;
+
+ dai->suspend = s3c2412_i2s_suspend;
+ dai->resume = s3c2412_i2s_resume;
+
+ return snd_soc_register_dai(dai);
+}
+
+EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai);
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h
new file mode 100644
index 0000000..f66854a
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.h
@@ -0,0 +1,90 @@
+/* sound/soc/s3c24xx/s3c-i2s-v2.h
+ *
+ * ALSA Soc Audio Layer - S3C_I2SV2 I2S driver
+ *
+ * Copyright (c) 2007 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 as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+*/
+
+/* This code is the core support for the I2S block found in a number of
+ * Samsung SoC devices which is unofficially named I2S-V2. Currently the
+ * S3C2412 and the S3C64XX series use this block to provide 1 or 2 I2S
+ * channels via configurable GPIO.
+ */
+
+#ifndef __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H
+#define __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H __FILE__
+
+#define S3C_I2SV2_DIV_BCLK (1)
+#define S3C_I2SV2_DIV_RCLK (2)
+#define S3C_I2SV2_DIV_PRESCALER (3)
+
+/**
+ * struct s3c_i2sv2_info - S3C I2S-V2 information
+ * @dev: The parent device passed to use from the probe.
+ * @regs: The pointer to the device registe block.
+ * @master: True if the I2S core is the I2S bit clock master.
+ * @dma_playback: DMA information for playback channel.
+ * @dma_capture: DMA information for capture channel.
+ * @suspend_iismod: PM save for the IISMOD register.
+ * @suspend_iiscon: PM save for the IISCON register.
+ * @suspend_iispsr: PM save for the IISPSR register.
+ *
+ * This is the private codec state for the hardware associated with an
+ * I2S channel such as the register mappings and clock sources.
+ */
+struct s3c_i2sv2_info {
+ struct device *dev;
+ void __iomem *regs;
+
+ struct clk *iis_pclk;
+ struct clk *iis_cclk;
+ struct clk *iis_clk;
+
+ unsigned char master;
+
+ struct s3c24xx_pcm_dma_params *dma_playback;
+ struct s3c24xx_pcm_dma_params *dma_capture;
+
+ u32 suspend_iismod;
+ u32 suspend_iiscon;
+ u32 suspend_iispsr;
+};
+
+struct s3c_i2sv2_rate_calc {
+ unsigned int clk_div; /* for prescaler */
+ unsigned int fs_div; /* for root frame clock */
+};
+
+extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
+ unsigned int *fstab,
+ unsigned int rate, struct clk *clk);
+
+/**
+ * s3c_i2sv2_probe - probe for i2s device helper
+ * @pdev: The platform device supplied to the original probe.
+ * @dai: The ASoC DAI structure supplied to the original probe.
+ * @i2s: Our local i2s structure to fill in.
+ * @base: The base address for the registers.
+ */
+extern int s3c_i2sv2_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai,
+ struct s3c_i2sv2_info *i2s,
+ unsigned long base);
+
+/**
+ * s3c_i2sv2_register_dai - register dai with soc core
+ * @dai: The snd_soc_dai structure to register
+ *
+ * Fill in any missing fields and then register the given dai with the
+ * soc core.
+ */
+extern int s3c_i2sv2_register_dai(struct snd_soc_dai *dai);
+
+#endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 382d7ee..1ca3cda 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -22,6 +22,7 @@
#include <linux/delay.h>
#include <linux/clk.h>
#include <linux/kernel.h>
+#include <linux/io.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -30,26 +31,16 @@
#include <sound/soc.h>
#include <mach/hardware.h>
-#include <linux/io.h>
-#include <asm/dma.h>
+#include <plat/regs-s3c2412-iis.h>
-#include <asm/plat-s3c24xx/regs-s3c2412-iis.h>
-
-#include <mach/regs-gpio.h>
-#include <mach/audio.h>
+#include <plat/regs-gpio.h>
+#include <plat/audio.h>
#include <mach/dma.h>
#include "s3c24xx-pcm.h"
#include "s3c2412-i2s.h"
#define S3C2412_I2S_DEBUG 0
-#define S3C2412_I2S_DEBUG_CON 0
-
-#if S3C2412_I2S_DEBUG
-#define DBG(x...) printk(KERN_INFO x)
-#else
-#define DBG(x...) do { } while (0)
-#endif
static struct s3c2410_dma_client s3c2412_dma_client_out = {
.name = "I2S PCM Stereo out"
@@ -73,431 +64,7 @@
.dma_size = 4,
};
-struct s3c2412_i2s_info {
- struct device *dev;
- void __iomem *regs;
- struct clk *iis_clk;
- struct clk *iis_pclk;
- struct clk *iis_cclk;
-
- u32 suspend_iismod;
- u32 suspend_iiscon;
- u32 suspend_iispsr;
-};
-
-static struct s3c2412_i2s_info s3c2412_i2s;
-
-#define bit_set(v, b) (((v) & (b)) ? 1 : 0)
-
-#if S3C2412_I2S_DEBUG_CON
-static void dbg_showcon(const char *fn, u32 con)
-{
- printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn,
- bit_set(con, S3C2412_IISCON_LRINDEX),
- bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY),
- bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY),
- bit_set(con, S3C2412_IISCON_TXFIFO_FULL),
- bit_set(con, S3C2412_IISCON_RXFIFO_FULL));
-
- printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n",
- fn,
- bit_set(con, S3C2412_IISCON_TXDMA_PAUSE),
- bit_set(con, S3C2412_IISCON_RXDMA_PAUSE),
- bit_set(con, S3C2412_IISCON_TXCH_PAUSE),
- bit_set(con, S3C2412_IISCON_RXCH_PAUSE));
- printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn,
- bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE),
- bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE),
- bit_set(con, S3C2412_IISCON_IIS_ACTIVE));
-}
-#else
-static inline void dbg_showcon(const char *fn, u32 con)
-{
-}
-#endif
-
-/* Turn on or off the transmission path. */
-static void s3c2412_snd_txctrl(int on)
-{
- struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
- void __iomem *regs = i2s->regs;
- u32 fic, con, mod;
-
- DBG("%s(%d)\n", __func__, on);
-
- fic = readl(regs + S3C2412_IISFIC);
- con = readl(regs + S3C2412_IISCON);
- mod = readl(regs + S3C2412_IISMOD);
-
- DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
-
- if (on) {
- con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE;
- con &= ~S3C2412_IISCON_TXDMA_PAUSE;
- con &= ~S3C2412_IISCON_TXCH_PAUSE;
-
- switch (mod & S3C2412_IISMOD_MODE_MASK) {
- case S3C2412_IISMOD_MODE_TXONLY:
- case S3C2412_IISMOD_MODE_TXRX:
- /* do nothing, we are in the right mode */
- break;
-
- case S3C2412_IISMOD_MODE_RXONLY:
- mod &= ~S3C2412_IISMOD_MODE_MASK;
- mod |= S3C2412_IISMOD_MODE_TXRX;
- break;
-
- default:
- dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n");
- }
-
- writel(con, regs + S3C2412_IISCON);
- writel(mod, regs + S3C2412_IISMOD);
- } else {
- /* Note, we do not have any indication that the FIFO problems
- * tha the S3C2410/2440 had apply here, so we should be able
- * to disable the DMA and TX without resetting the FIFOS.
- */
-
- con |= S3C2412_IISCON_TXDMA_PAUSE;
- con |= S3C2412_IISCON_TXCH_PAUSE;
- con &= ~S3C2412_IISCON_TXDMA_ACTIVE;
-
- switch (mod & S3C2412_IISMOD_MODE_MASK) {
- case S3C2412_IISMOD_MODE_TXRX:
- mod &= ~S3C2412_IISMOD_MODE_MASK;
- mod |= S3C2412_IISMOD_MODE_RXONLY;
- break;
-
- case S3C2412_IISMOD_MODE_TXONLY:
- mod &= ~S3C2412_IISMOD_MODE_MASK;
- con &= ~S3C2412_IISCON_IIS_ACTIVE;
- break;
-
- default:
- dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n");
- }
-
- writel(mod, regs + S3C2412_IISMOD);
- writel(con, regs + S3C2412_IISCON);
- }
-
- fic = readl(regs + S3C2412_IISFIC);
- dbg_showcon(__func__, con);
- DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
-}
-
-static void s3c2412_snd_rxctrl(int on)
-{
- struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
- void __iomem *regs = i2s->regs;
- u32 fic, con, mod;
-
- DBG("%s(%d)\n", __func__, on);
-
- fic = readl(regs + S3C2412_IISFIC);
- con = readl(regs + S3C2412_IISCON);
- mod = readl(regs + S3C2412_IISMOD);
-
- DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
-
- if (on) {
- con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE;
- con &= ~S3C2412_IISCON_RXDMA_PAUSE;
- con &= ~S3C2412_IISCON_RXCH_PAUSE;
-
- switch (mod & S3C2412_IISMOD_MODE_MASK) {
- case S3C2412_IISMOD_MODE_TXRX:
- case S3C2412_IISMOD_MODE_RXONLY:
- /* do nothing, we are in the right mode */
- break;
-
- case S3C2412_IISMOD_MODE_TXONLY:
- mod &= ~S3C2412_IISMOD_MODE_MASK;
- mod |= S3C2412_IISMOD_MODE_TXRX;
- break;
-
- default:
- dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
- }
-
- writel(mod, regs + S3C2412_IISMOD);
- writel(con, regs + S3C2412_IISCON);
- } else {
- /* See txctrl notes on FIFOs. */
-
- con &= ~S3C2412_IISCON_RXDMA_ACTIVE;
- con |= S3C2412_IISCON_RXDMA_PAUSE;
- con |= S3C2412_IISCON_RXCH_PAUSE;
-
- switch (mod & S3C2412_IISMOD_MODE_MASK) {
- case S3C2412_IISMOD_MODE_RXONLY:
- con &= ~S3C2412_IISCON_IIS_ACTIVE;
- mod &= ~S3C2412_IISMOD_MODE_MASK;
- break;
-
- case S3C2412_IISMOD_MODE_TXRX:
- mod &= ~S3C2412_IISMOD_MODE_MASK;
- mod |= S3C2412_IISMOD_MODE_TXONLY;
- break;
-
- default:
- dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
- }
-
- writel(con, regs + S3C2412_IISCON);
- writel(mod, regs + S3C2412_IISMOD);
- }
-
- fic = readl(regs + S3C2412_IISFIC);
- DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
-}
-
-
-/*
- * 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(void)
-{
- u32 iiscon;
- unsigned long timeout = jiffies + msecs_to_jiffies(5);
-
- DBG("Entered %s\n", __func__);
-
- while (1) {
- iiscon = readl(s3c2412_i2s.regs + S3C2412_IISCON);
- if (iiscon & S3C2412_IISCON_LRINDEX)
- break;
-
- if (timeout < jiffies) {
- printk(KERN_ERR "%s: timeout\n", __func__);
- return -ETIMEDOUT;
- }
- }
-
- return 0;
-}
-
-/*
- * Check whether CPU is the master or slave
- */
-static inline int s3c2412_snd_is_clkmaster(void)
-{
- u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
-
- DBG("Entered %s\n", __func__);
-
- iismod &= S3C2412_IISMOD_MASTER_MASK;
- return !(iismod == S3C2412_IISMOD_SLAVE);
-}
-
-/*
- * Set S3C2412 I2S DAI format
- */
-static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
- unsigned int fmt)
-{
- u32 iismod;
-
-
- DBG("Entered %s\n", __func__);
-
- iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
- DBG("hw_params r: IISMOD: %x \n", iismod);
-
- switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
- case SND_SOC_DAIFMT_CBM_CFM:
- iismod &= ~S3C2412_IISMOD_MASTER_MASK;
- iismod |= S3C2412_IISMOD_SLAVE;
- break;
- case SND_SOC_DAIFMT_CBS_CFS:
- iismod &= ~S3C2412_IISMOD_MASTER_MASK;
- iismod |= S3C2412_IISMOD_MASTER_INTERNAL;
- break;
- default:
- DBG("unknwon master/slave format\n");
- return -EINVAL;
- }
-
- iismod &= ~S3C2412_IISMOD_SDF_MASK;
-
- switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
- case SND_SOC_DAIFMT_RIGHT_J:
- iismod |= S3C2412_IISMOD_SDF_MSB;
- break;
- case SND_SOC_DAIFMT_LEFT_J:
- iismod |= S3C2412_IISMOD_SDF_LSB;
- break;
- case SND_SOC_DAIFMT_I2S:
- iismod |= S3C2412_IISMOD_SDF_IIS;
- break;
- default:
- DBG("Unknown data format\n");
- return -EINVAL;
- }
-
- writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD);
- DBG("hw_params w: IISMOD: %x \n", iismod);
- return 0;
-}
-
-static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- u32 iismod;
-
- DBG("Entered %s\n", __func__);
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_out;
- else
- rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_in;
-
- /* Working copies of register */
- iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
- DBG("%s: r: IISMOD: %x\n", __func__, iismod);
-
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S8:
- iismod |= S3C2412_IISMOD_8BIT;
- break;
- case SNDRV_PCM_FORMAT_S16_LE:
- iismod &= ~S3C2412_IISMOD_8BIT;
- break;
- }
-
- writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD);
- DBG("%s: w: IISMOD: %x\n", __func__, iismod);
- return 0;
-}
-
-static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
- struct snd_soc_dai *dai)
-{
- int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
- unsigned long irqs;
- int ret = 0;
-
- DBG("Entered %s\n", __func__);
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- /* On start, ensure that the FIFOs are cleared and reset. */
-
- writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH,
- s3c2412_i2s.regs + S3C2412_IISFIC);
-
- /* clear again, just in case */
- writel(0x0, s3c2412_i2s.regs + S3C2412_IISFIC);
-
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- if (!s3c2412_snd_is_clkmaster()) {
- ret = s3c2412_snd_lrsync();
- if (ret)
- goto exit_err;
- }
-
- local_irq_save(irqs);
-
- if (capture)
- s3c2412_snd_rxctrl(1);
- else
- s3c2412_snd_txctrl(1);
-
- local_irq_restore(irqs);
- break;
-
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- local_irq_save(irqs);
-
- if (capture)
- s3c2412_snd_rxctrl(0);
- else
- s3c2412_snd_txctrl(0);
-
- local_irq_restore(irqs);
- break;
- default:
- ret = -EINVAL;
- break;
- }
-
-exit_err:
- return ret;
-}
-
-/* default table of all avaialable root fs divisors */
-static unsigned int s3c2412_iis_fs[] = { 256, 512, 384, 768, 0 };
-
-int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info,
- unsigned int *fstab,
- unsigned int rate, struct clk *clk)
-{
- unsigned long clkrate = clk_get_rate(clk);
- unsigned int div;
- unsigned int fsclk;
- unsigned int actual;
- unsigned int fs;
- unsigned int fsdiv;
- signed int deviation = 0;
- unsigned int best_fs = 0;
- unsigned int best_div = 0;
- unsigned int best_rate = 0;
- unsigned int best_deviation = INT_MAX;
-
-
- if (fstab == NULL)
- fstab = s3c2412_iis_fs;
-
- for (fs = 0;; fs++) {
- fsdiv = s3c2412_iis_fs[fs];
-
- if (fsdiv == 0)
- break;
-
- fsclk = clkrate / fsdiv;
- div = fsclk / rate;
-
- if ((fsclk % rate) > (rate / 2))
- div++;
-
- if (div <= 1)
- continue;
-
- actual = clkrate / (fsdiv * div);
- deviation = actual - rate;
-
- printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n",
- fsdiv, div, actual, deviation);
-
- deviation = abs(deviation);
-
- if (deviation < best_deviation) {
- best_fs = fsdiv;
- best_div = div;
- best_rate = actual;
- best_deviation = deviation;
- }
-
- if (deviation == 0)
- break;
- }
-
- printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n",
- best_fs, best_div, best_rate);
-
- info->fs_div = best_fs;
- info->clk_div = best_div;
-
- return 0;
-}
-EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate);
+static struct s3c_i2sv2_info s3c2412_i2s;
/*
* Set S3C2412 Clock source
@@ -507,15 +74,17 @@
{
u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
- DBG("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id,
+ pr_debug("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id,
freq, dir);
switch (clk_id) {
case S3C2412_CLKSRC_PCLK:
+ s3c2412_i2s.master = 1;
iismod &= ~S3C2412_IISMOD_MASTER_MASK;
iismod |= S3C2412_IISMOD_MASTER_INTERNAL;
break;
case S3C2412_CLKSRC_I2SCLK:
+ s3c2412_i2s.master = 0;
iismod &= ~S3C2412_IISMOD_MASTER_MASK;
iismod |= S3C2412_IISMOD_MASTER_EXTERNAL;
break;
@@ -527,74 +96,6 @@
return 0;
}
-/*
- * Set S3C2412 Clock dividers
- */
-static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
- int div_id, int div)
-{
- struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
- u32 reg;
-
- DBG("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div);
-
- switch (div_id) {
- case S3C2412_DIV_BCLK:
- reg = readl(i2s->regs + S3C2412_IISMOD);
- reg &= ~S3C2412_IISMOD_BCLK_MASK;
- writel(reg | div, i2s->regs + S3C2412_IISMOD);
-
- DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD));
- break;
-
- case S3C2412_DIV_RCLK:
- if (div > 3) {
- /* convert value to bit field */
-
- switch (div) {
- case 256:
- div = S3C2412_IISMOD_RCLK_256FS;
- break;
-
- case 384:
- div = S3C2412_IISMOD_RCLK_384FS;
- break;
-
- case 512:
- div = S3C2412_IISMOD_RCLK_512FS;
- break;
-
- case 768:
- div = S3C2412_IISMOD_RCLK_768FS;
- break;
-
- default:
- return -EINVAL;
- }
- }
-
- reg = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
- reg &= ~S3C2412_IISMOD_RCLK_MASK;
- writel(reg | div, i2s->regs + S3C2412_IISMOD);
- DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD));
- break;
-
- case S3C2412_DIV_PRESCALER:
- if (div >= 0) {
- writel((div << 8) | S3C2412_IISPSR_PSREN,
- i2s->regs + S3C2412_IISPSR);
- } else {
- writel(0x0, i2s->regs + S3C2412_IISPSR);
- }
- DBG("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR));
- break;
-
- default:
- return -EINVAL;
- }
-
- return 0;
-}
struct clk *s3c2412_get_iisclk(void)
{
@@ -606,34 +107,30 @@
static int s3c2412_i2s_probe(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
- DBG("Entered %s\n", __func__);
+ int ret;
- s3c2412_i2s.dev = &pdev->dev;
+ pr_debug("Entered %s\n", __func__);
- s3c2412_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100);
- if (s3c2412_i2s.regs == NULL)
- return -ENXIO;
+ ret = s3c_i2sv2_probe(pdev, dai, &s3c2412_i2s, S3C2410_PA_IIS);
+ if (ret)
+ return ret;
- s3c2412_i2s.iis_pclk = clk_get(&pdev->dev, "iis");
- if (s3c2412_i2s.iis_pclk == NULL) {
- DBG("failed to get iis_clock\n");
- iounmap(s3c2412_i2s.regs);
- return -ENODEV;
- }
+ s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in;
+ s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out;
s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk");
if (s3c2412_i2s.iis_cclk == NULL) {
- DBG("failed to get i2sclk clock\n");
+ pr_debug("failed to get i2sclk clock\n");
iounmap(s3c2412_i2s.regs);
return -ENODEV;
}
- clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll"));
+ /* Set MPLL as the source for IIS CLK */
- clk_enable(s3c2412_i2s.iis_pclk);
+ clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll"));
clk_enable(s3c2412_i2s.iis_cclk);
- s3c2412_i2s.iis_clk = s3c2412_i2s.iis_pclk;
+ s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk;
/* Configure the I2S pins in correct mode */
s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK);
@@ -642,86 +139,22 @@
s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI);
s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO);
- s3c2412_snd_txctrl(0);
- s3c2412_snd_rxctrl(0);
-
return 0;
}
-#ifdef CONFIG_PM
-static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
-{
- struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
- u32 iismod;
-
- if (dai->active) {
- i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD);
- i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON);
- i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR);
-
- /* some basic suspend checks */
-
- iismod = readl(i2s->regs + S3C2412_IISMOD);
-
- if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
- pr_warning("%s: RXDMA active?\n", __func__);
-
- if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
- pr_warning("%s: TXDMA active?\n", __func__);
-
- if (iismod & S3C2412_IISCON_IIS_ACTIVE)
- pr_warning("%s: IIS active\n", __func__);
- }
-
- return 0;
-}
-
-static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
-{
- struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
-
- pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n",
- dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
-
- if (dai->active) {
- writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
- writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD);
- writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR);
-
- writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH,
- i2s->regs + S3C2412_IISFIC);
-
- ndelay(250);
- writel(0x0, i2s->regs + S3C2412_IISFIC);
-
- }
-
- return 0;
-}
-#else
-#define s3c2412_i2s_suspend NULL
-#define s3c2412_i2s_resume NULL
-#endif /* CONFIG_PM */
-
#define S3C2412_I2S_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 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = {
- .trigger = s3c2412_i2s_trigger,
- .hw_params = s3c2412_i2s_hw_params,
- .set_fmt = s3c2412_i2s_set_fmt,
- .set_clkdiv = s3c2412_i2s_set_clkdiv,
.set_sysclk = s3c2412_i2s_set_sysclk,
};
struct snd_soc_dai s3c2412_i2s_dai = {
- .name = "s3c2412-i2s",
- .id = 0,
- .probe = s3c2412_i2s_probe,
- .suspend = s3c2412_i2s_suspend,
- .resume = s3c2412_i2s_resume,
+ .name = "s3c2412-i2s",
+ .id = 0,
+ .probe = s3c2412_i2s_probe,
.playback = {
.channels_min = 2,
.channels_max = 2,
@@ -740,7 +173,7 @@
static int __init s3c2412_i2s_init(void)
{
- return snd_soc_register_dai(&s3c2412_i2s_dai);
+ return s3c_i2sv2_register_dai(&s3c2412_i2s_dai);
}
module_init(s3c2412_i2s_init);
@@ -750,7 +183,6 @@
}
module_exit(s3c2412_i2s_exit);
-
/* Module information */
MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
MODULE_DESCRIPTION("S3C2412 I2S SoC Interface");
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h
index aac08a2..92848e5 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.h
+++ b/sound/soc/s3c24xx/s3c2412-i2s.h
@@ -15,9 +15,11 @@
#ifndef __SND_SOC_S3C24XX_S3C2412_I2S_H
#define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__
-#define S3C2412_DIV_BCLK (1)
-#define S3C2412_DIV_RCLK (2)
-#define S3C2412_DIV_PRESCALER (3)
+#include "s3c-i2s-v2.h"
+
+#define S3C2412_DIV_BCLK S3C_I2SV2_DIV_BCLK
+#define S3C2412_DIV_RCLK S3C_I2SV2_DIV_RCLK
+#define S3C2412_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER
#define S3C2412_CLKSRC_PCLK (0)
#define S3C2412_CLKSRC_I2SCLK (1)
@@ -26,13 +28,4 @@
extern struct snd_soc_dai s3c2412_i2s_dai;
-struct s3c2412_rate_calc {
- unsigned int clk_div; /* for prescaler */
- unsigned int fs_div; /* for root frame clock */
-};
-
-extern int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info,
- unsigned int *fstab,
- unsigned int rate, struct clk *clk);
-
#endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 83ea623..3698f70 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -31,7 +31,7 @@
#include <plat/regs-ac97.h>
#include <mach/regs-gpio.h>
#include <mach/regs-clock.h>
-#include <mach/audio.h>
+#include <plat/audio.h>
#include <asm/dma.h>
#include <mach/dma.h>
@@ -360,6 +360,11 @@
.trigger = s3c2443_ac97_trigger,
};
+static struct snd_soc_dai_ops s3c2443_ac97_mic_dai_ops = {
+ .hw_params = s3c2443_ac97_hw_mic_params,
+ .trigger = s3c2443_ac97_mic_trigger,
+};
+
struct snd_soc_dai s3c2443_ac97_dai[] = {
{
.name = "s3c2443-ac97",
@@ -391,7 +396,7 @@
.channels_max = 1,
.rates = s3c2443_AC97_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
- .ops = &s3c2443_ac97_dai_ops,
+ .ops = &s3c2443_ac97_mic_dai_ops,
},
};
EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 4473fb5..cc06696 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -30,22 +30,15 @@
#include <mach/hardware.h>
#include <mach/regs-gpio.h>
#include <mach/regs-clock.h>
-#include <mach/audio.h>
+#include <plat/audio.h>
#include <asm/dma.h>
#include <mach/dma.h>
-#include <asm/plat-s3c24xx/regs-iis.h>
+#include <plat/regs-iis.h>
#include "s3c24xx-pcm.h"
#include "s3c24xx-i2s.h"
-#define S3C24XX_I2S_DEBUG 0
-#if S3C24XX_I2S_DEBUG
-#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x)
-#else
-#define DBG(x...)
-#endif
-
static struct s3c2410_dma_client s3c24xx_dma_client_out = {
.name = "I2S PCM Stereo out"
};
@@ -84,13 +77,13 @@
u32 iiscon;
u32 iismod;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
- DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+ pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon);
if (on) {
iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE;
@@ -120,7 +113,7 @@
writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
}
- DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+ pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon);
}
static void s3c24xx_snd_rxctrl(int on)
@@ -129,13 +122,13 @@
u32 iiscon;
u32 iismod;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
- DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+ pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon);
if (on) {
iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE;
@@ -165,7 +158,7 @@
writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
}
- DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+ pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon);
}
/*
@@ -177,7 +170,7 @@
u32 iiscon;
int timeout = 50; /* 5ms */
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
while (1) {
iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
@@ -197,7 +190,7 @@
*/
static inline int s3c24xx_snd_is_clkmaster(void)
{
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1;
}
@@ -210,10 +203,10 @@
{
u32 iismod;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
- DBG("hw_params r: IISMOD: %lx \n", iismod);
+ pr_debug("hw_params r: IISMOD: %x \n", iismod);
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
@@ -238,7 +231,7 @@
}
writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
- DBG("hw_params w: IISMOD: %lx \n", iismod);
+ pr_debug("hw_params w: IISMOD: %x \n", iismod);
return 0;
}
@@ -249,7 +242,7 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
u32 iismod;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out;
@@ -258,7 +251,7 @@
/* Working copies of register */
iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
- DBG("hw_params r: IISMOD: %lx\n", iismod);
+ pr_debug("hw_params r: IISMOD: %x\n", iismod);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
@@ -276,7 +269,7 @@
}
writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
- DBG("hw_params w: IISMOD: %lx\n", iismod);
+ pr_debug("hw_params w: IISMOD: %x\n", iismod);
return 0;
}
@@ -285,7 +278,7 @@
{
int ret = 0;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
@@ -327,7 +320,7 @@
{
u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
iismod &= ~S3C2440_IISMOD_MPLL;
@@ -353,7 +346,7 @@
{
u32 reg;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
switch (div_id) {
case S3C24XX_DIV_BCLK:
@@ -389,7 +382,7 @@
static int s3c24xx_i2s_probe(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100);
if (s3c24xx_i2s.regs == NULL)
@@ -397,7 +390,7 @@
s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis");
if (s3c24xx_i2s.iis_clk == NULL) {
- DBG("failed to get iis_clock\n");
+ pr_err("failed to get iis_clock\n");
iounmap(s3c24xx_i2s.regs);
return -ENODEV;
}
@@ -421,7 +414,7 @@
#ifdef CONFIG_PM
static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai)
{
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
@@ -435,7 +428,7 @@
static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai)
{
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
clk_enable(s3c24xx_i2s.iis_clk);
writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON);
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index ba1ae09..a9d68fa 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -29,17 +29,10 @@
#include <asm/dma.h>
#include <mach/hardware.h>
#include <mach/dma.h>
-#include <mach/audio.h>
+#include <plat/audio.h>
#include "s3c24xx-pcm.h"
-#define S3C24XX_PCM_DEBUG 0
-#if S3C24XX_PCM_DEBUG
-#define DBG(x...) printk(KERN_DEBUG "s3c24xx-pcm: " x)
-#else
-#define DBG(x...)
-#endif
-
static const struct snd_pcm_hardware s3c24xx_pcm_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
@@ -84,16 +77,16 @@
dma_addr_t pos = prtd->dma_pos;
int ret;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
while (prtd->dma_loaded < prtd->dma_limit) {
unsigned long len = prtd->dma_period;
- DBG("dma_loaded: %d\n", prtd->dma_loaded);
+ pr_debug("dma_loaded: %d\n", prtd->dma_loaded);
if ((pos + len) > prtd->dma_end) {
len = prtd->dma_end - pos;
- DBG(KERN_DEBUG "%s: corrected dma len %ld\n",
+ pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n",
__func__, len);
}
@@ -119,7 +112,7 @@
struct snd_pcm_substream *substream = dev_id;
struct s3c24xx_runtime_data *prtd;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR)
return;
@@ -148,7 +141,7 @@
unsigned long totbytes = params_buffer_bytes(params);
int ret = 0;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
/* return if this is a bufferless transfer e.g.
* codec <--> BT codec or GSM modem -- lg FIXME */
@@ -161,14 +154,14 @@
/* prepare DMA */
prtd->params = dma;
- DBG("params %p, client %p, channel %d\n", prtd->params,
+ pr_debug("params %p, client %p, channel %d\n", prtd->params,
prtd->params->client, prtd->params->channel);
ret = s3c2410_dma_request(prtd->params->channel,
prtd->params->client, NULL);
if (ret < 0) {
- DBG(KERN_ERR "failed to get dma channel\n");
+ printk(KERN_ERR "failed to get dma channel\n");
return ret;
}
}
@@ -196,7 +189,7 @@
{
struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
/* TODO - do we need to ensure DMA flushed */
snd_pcm_set_runtime_buffer(substream, NULL);
@@ -214,7 +207,7 @@
struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
int ret = 0;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
/* return if this is a bufferless transfer e.g.
* codec <--> BT codec or GSM modem -- lg FIXME */
@@ -259,7 +252,7 @@
struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
int ret = 0;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
spin_lock(&prtd->lock);
@@ -297,7 +290,7 @@
unsigned long res;
dma_addr_t src, dst;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
spin_lock(&prtd->lock);
s3c2410_dma_getposition(prtd->params->channel, &src, &dst);
@@ -309,7 +302,7 @@
spin_unlock(&prtd->lock);
- DBG("Pointer %x %x\n", src, dst);
+ pr_debug("Pointer %x %x\n", src, dst);
/* we seem to be getting the odd error from the pcm library due
* to out-of-bounds pointers. this is maybe due to the dma engine
@@ -330,7 +323,7 @@
struct snd_pcm_runtime *runtime = substream->runtime;
struct s3c24xx_runtime_data *prtd;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware);
@@ -349,10 +342,10 @@
struct snd_pcm_runtime *runtime = substream->runtime;
struct s3c24xx_runtime_data *prtd = runtime->private_data;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
if (!prtd)
- DBG("s3c24xx_pcm_close called with prtd == NULL\n");
+ pr_debug("s3c24xx_pcm_close called with prtd == NULL\n");
kfree(prtd);
@@ -364,7 +357,7 @@
{
struct snd_pcm_runtime *runtime = substream->runtime;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
return dma_mmap_writecombine(substream->pcm->card->dev, vma,
runtime->dma_area,
@@ -390,7 +383,7 @@
struct snd_dma_buffer *buf = &substream->dma_buffer;
size_t size = s3c24xx_pcm_hardware.buffer_bytes_max;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
buf->dev.type = SNDRV_DMA_TYPE_DEV;
buf->dev.dev = pcm->card->dev;
@@ -409,7 +402,7 @@
struct snd_dma_buffer *buf;
int stream;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
for (stream = 0; stream < 2; stream++) {
substream = pcm->streams[stream].substream;
@@ -433,7 +426,7 @@
{
int ret = 0;
- DBG("Entered %s\n", __func__);
+ pr_debug("Entered %s\n", __func__);
if (!card->dev->dma_mask)
card->dev->dma_mask = &s3c24xx_pcm_dmamask;
diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
index a0a4d183..8e79a41 100644
--- a/sound/soc/s3c24xx/s3c24xx_uda134x.c
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -22,7 +22,7 @@
#include <sound/s3c24xx_uda134x.h>
#include <sound/uda134x.h>
-#include <asm/plat-s3c24xx/regs-iis.h>
+#include <plat/regs-iis.h>
#include "s3c24xx-pcm.h"
#include "s3c24xx-i2s.h"
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
new file mode 100644
index 0000000..6e1e85d
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -0,0 +1,220 @@
+/* sound/soc/s3c24xx/s3c64xx-i2s.c
+ *
+ * ALSA SoC Audio Layer - S3C64XX I2S driver
+ *
+ * Copyright 2008 Openmoko, Inc.
+ * Copyright 2008 Simtec Electronics
+ * Ben Dooks <ben@simtec.co.uk>
+ * http://armlinux.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.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/kernel.h>
+#include <linux/gpio.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <plat/regs-s3c2412-iis.h>
+#include <plat/gpio-bank-d.h>
+#include <plat/gpio-bank-e.h>
+#include <plat/gpio-cfg.h>
+#include <plat/audio.h>
+
+#include <mach/map.h>
+#include <mach/dma.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c64xx-i2s.h"
+
+static struct s3c2410_dma_client s3c64xx_dma_client_out = {
+ .name = "I2S PCM Stereo out"
+};
+
+static struct s3c2410_dma_client s3c64xx_dma_client_in = {
+ .name = "I2S PCM Stereo in"
+};
+
+static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = {
+ [0] = {
+ .channel = DMACH_I2S0_OUT,
+ .client = &s3c64xx_dma_client_out,
+ .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD,
+ .dma_size = 4,
+ },
+ [1] = {
+ .channel = DMACH_I2S1_OUT,
+ .client = &s3c64xx_dma_client_out,
+ .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD,
+ .dma_size = 4,
+ },
+};
+
+static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_in[2] = {
+ [0] = {
+ .channel = DMACH_I2S0_IN,
+ .client = &s3c64xx_dma_client_in,
+ .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD,
+ .dma_size = 4,
+ },
+ [1] = {
+ .channel = DMACH_I2S1_IN,
+ .client = &s3c64xx_dma_client_in,
+ .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD,
+ .dma_size = 4,
+ },
+};
+
+static struct s3c_i2sv2_info s3c64xx_i2s[2];
+
+static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
+{
+ return cpu_dai->private_data;
+}
+
+static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
+ u32 iismod = readl(i2s->regs + S3C2412_IISMOD);
+
+ switch (clk_id) {
+ case S3C64XX_CLKSRC_PCLK:
+ iismod &= ~S3C64XX_IISMOD_IMS_SYSMUX;
+ break;
+
+ case S3C64XX_CLKSRC_MUX:
+ iismod |= S3C64XX_IISMOD_IMS_SYSMUX;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ writel(iismod, i2s->regs + S3C2412_IISMOD);
+
+ return 0;
+}
+
+
+unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *dai)
+{
+ struct s3c_i2sv2_info *i2s = to_info(dai);
+
+ return clk_get_rate(i2s->iis_cclk);
+}
+EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clockrate);
+
+static int s3c64xx_i2s_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct device *dev = &pdev->dev;
+ struct s3c_i2sv2_info *i2s;
+ int ret;
+
+ dev_dbg(dev, "%s: probing dai %d\n", __func__, pdev->id);
+
+ if (pdev->id < 0 || pdev->id > ARRAY_SIZE(s3c64xx_i2s)) {
+ dev_err(dev, "id %d out of range\n", pdev->id);
+ return -EINVAL;
+ }
+
+ i2s = &s3c64xx_i2s[pdev->id];
+
+ ret = s3c_i2sv2_probe(pdev, dai, i2s,
+ pdev->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0);
+ if (ret)
+ return ret;
+
+ i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id];
+ i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id];
+
+ i2s->iis_cclk = clk_get(dev, "audio-bus");
+ if (IS_ERR(i2s->iis_cclk)) {
+ dev_err(dev, "failed to get audio-bus");
+ iounmap(i2s->regs);
+ return -ENODEV;
+ }
+
+ /* configure GPIO for i2s port */
+ switch (pdev->id) {
+ case 0:
+ s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK);
+ s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK);
+ s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK);
+ s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI);
+ s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0);
+ break;
+ case 1:
+ s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK);
+ s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK);
+ s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK);
+ s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI);
+ s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0);
+ }
+
+ return 0;
+}
+
+
+#define S3C64XX_I2S_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 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define S3C64XX_I2S_FMTS \
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE)
+
+struct snd_soc_dai s3c64xx_i2s_dai = {
+ .name = "s3c64xx-i2s",
+ .id = 0,
+ .probe = s3c64xx_i2s_probe,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = S3C64XX_I2S_RATES,
+ .formats = S3C64XX_I2S_FMTS,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = S3C64XX_I2S_RATES,
+ .formats = S3C64XX_I2S_FMTS,
+ },
+ .ops = {
+ .set_sysclk = s3c64xx_i2s_set_sysclk,
+ },
+};
+EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai);
+
+static int __init s3c64xx_i2s_init(void)
+{
+ return s3c_i2sv2_register_dai(&s3c64xx_i2s_dai);
+}
+module_init(s3c64xx_i2s_init);
+
+static void __exit s3c64xx_i2s_exit(void)
+{
+ snd_soc_unregister_dai(&s3c64xx_i2s_dai);
+}
+module_exit(s3c64xx_i2s_exit);
+
+/* Module information */
+MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("S3C64XX I2S SoC Interface");
+MODULE_LICENSE("GPL");
+
+
+
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h
new file mode 100644
index 0000000..b7ffe3c
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.h
@@ -0,0 +1,31 @@
+/* sound/soc/s3c24xx/s3c64xx-i2s.h
+ *
+ * ALSA SoC Audio Layer - S3C64XX I2S driver
+ *
+ * Copyright 2008 Openmoko, Inc.
+ * Copyright 2008 Simtec Electronics
+ * Ben Dooks <ben@simtec.co.uk>
+ * http://armlinux.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.
+ */
+
+#ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H
+#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__
+
+#include "s3c-i2s-v2.h"
+
+#define S3C64XX_DIV_BCLK S3C_I2SV2_DIV_BCLK
+#define S3C64XX_DIV_RCLK S3C_I2SV2_DIV_RCLK
+#define S3C64XX_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER
+
+#define S3C64XX_CLKSRC_PCLK (0)
+#define S3C64XX_CLKSRC_MUX (1)
+
+extern struct snd_soc_dai s3c64xx_i2s_dai;
+
+extern unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *cpu_dai);
+
+#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 4b8dbbf..735903a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -332,7 +332,7 @@
* kcontrol name.
*/
name_len = strlen(w->kcontrols[i].name) + 1;
- if (w->id == snd_soc_dapm_mixer)
+ if (w->id != snd_soc_dapm_mixer_named_ctl)
name_len += 1 + strlen(w->name);
path->long_name = kmalloc(name_len, GFP_KERNEL);
@@ -341,15 +341,14 @@
return -ENOMEM;
switch (w->id) {
- case snd_soc_dapm_mixer:
default:
snprintf(path->long_name, name_len, "%s %s",
w->name, w->kcontrols[i].name);
- break;
+ break;
case snd_soc_dapm_mixer_named_ctl:
snprintf(path->long_name, name_len, "%s",
w->kcontrols[i].name);
- break;
+ break;
}
path->long_name[name_len - 1] = '\0';
@@ -523,6 +522,137 @@
EXPORT_SYMBOL_GPL(dapm_reg_event);
/*
+ * Scan a single DAPM widget for a complete audio path and update the
+ * power status appropriately.
+ */
+static int dapm_power_widget(struct snd_soc_codec *codec, int event,
+ struct snd_soc_dapm_widget *w)
+{
+ int in, out, power_change, power, ret;
+
+ /* vmid - no action */
+ if (w->id == snd_soc_dapm_vmid)
+ return 0;
+
+ /* active ADC */
+ if (w->id == snd_soc_dapm_adc && w->active) {
+ in = is_connected_input_ep(w);
+ dapm_clear_walk(w->codec);
+ w->power = (in != 0) ? 1 : 0;
+ dapm_update_bits(w);
+ return 0;
+ }
+
+ /* active DAC */
+ if (w->id == snd_soc_dapm_dac && w->active) {
+ out = is_connected_output_ep(w);
+ dapm_clear_walk(w->codec);
+ w->power = (out != 0) ? 1 : 0;
+ dapm_update_bits(w);
+ return 0;
+ }
+
+ /* pre and post event widgets */
+ if (w->id == snd_soc_dapm_pre) {
+ if (!w->event)
+ return 0;
+
+ if (event == SND_SOC_DAPM_STREAM_START) {
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_PRE_PMU);
+ if (ret < 0)
+ return ret;
+ } else if (event == SND_SOC_DAPM_STREAM_STOP) {
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_PRE_PMD);
+ if (ret < 0)
+ return ret;
+ }
+ return 0;
+ }
+ if (w->id == snd_soc_dapm_post) {
+ if (!w->event)
+ return 0;
+
+ if (event == SND_SOC_DAPM_STREAM_START) {
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_POST_PMU);
+ if (ret < 0)
+ return ret;
+ } else if (event == SND_SOC_DAPM_STREAM_STOP) {
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_POST_PMD);
+ if (ret < 0)
+ return ret;
+ }
+ return 0;
+ }
+
+ /* all other widgets */
+ in = is_connected_input_ep(w);
+ dapm_clear_walk(w->codec);
+ out = is_connected_output_ep(w);
+ dapm_clear_walk(w->codec);
+ power = (out != 0 && in != 0) ? 1 : 0;
+ power_change = (w->power == power) ? 0 : 1;
+ w->power = power;
+
+ if (!power_change)
+ return 0;
+
+ /* call any power change event handlers */
+ if (w->event)
+ pr_debug("power %s event for %s flags %x\n",
+ w->power ? "on" : "off",
+ w->name, w->event_flags);
+
+ /* power up pre event */
+ if (power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
+ ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* power down pre event */
+ if (!power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
+ ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* Lower PGA volume to reduce pops */
+ if (w->id == snd_soc_dapm_pga && !power)
+ dapm_set_pga(w, power);
+
+ dapm_update_bits(w);
+
+ /* Raise PGA volume to reduce pops */
+ if (w->id == snd_soc_dapm_pga && power)
+ dapm_set_pga(w, power);
+
+ /* power up post event */
+ if (power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_POST_PMU);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* power down post event */
+ if (!power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
+ ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
+ if (ret < 0)
+ return ret;
+ }
+
+ return 0;
+}
+
+/*
* Scan each dapm widget for complete audio path.
* A complete path is a route that has valid endpoints i.e.:-
*
@@ -534,7 +664,7 @@
static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
{
struct snd_soc_dapm_widget *w;
- int in, out, i, c = 1, *seq = NULL, ret = 0, power_change, power;
+ int i, c = 1, *seq = NULL, ret = 0;
/* do we have a sequenced stream event */
if (event == SND_SOC_DAPM_STREAM_START) {
@@ -545,135 +675,20 @@
seq = dapm_down_seq;
}
- for(i = 0; i < c; i++) {
+ for (i = 0; i < c; i++) {
list_for_each_entry(w, &codec->dapm_widgets, list) {
/* is widget in stream order */
if (seq && seq[i] && w->id != seq[i])
continue;
- /* vmid - no action */
- if (w->id == snd_soc_dapm_vmid)
- continue;
-
- /* active ADC */
- if (w->id == snd_soc_dapm_adc && w->active) {
- in = is_connected_input_ep(w);
- dapm_clear_walk(w->codec);
- w->power = (in != 0) ? 1 : 0;
- dapm_update_bits(w);
- continue;
- }
-
- /* active DAC */
- if (w->id == snd_soc_dapm_dac && w->active) {
- out = is_connected_output_ep(w);
- dapm_clear_walk(w->codec);
- w->power = (out != 0) ? 1 : 0;
- dapm_update_bits(w);
- continue;
- }
-
- /* pre and post event widgets */
- if (w->id == snd_soc_dapm_pre) {
- if (!w->event)
- continue;
-
- if (event == SND_SOC_DAPM_STREAM_START) {
- ret = w->event(w,
- NULL, SND_SOC_DAPM_PRE_PMU);
- if (ret < 0)
- return ret;
- } else if (event == SND_SOC_DAPM_STREAM_STOP) {
- ret = w->event(w,
- NULL, SND_SOC_DAPM_PRE_PMD);
- if (ret < 0)
- return ret;
- }
- continue;
- }
- if (w->id == snd_soc_dapm_post) {
- if (!w->event)
- continue;
-
- if (event == SND_SOC_DAPM_STREAM_START) {
- ret = w->event(w,
- NULL, SND_SOC_DAPM_POST_PMU);
- if (ret < 0)
- return ret;
- } else if (event == SND_SOC_DAPM_STREAM_STOP) {
- ret = w->event(w,
- NULL, SND_SOC_DAPM_POST_PMD);
- if (ret < 0)
- return ret;
- }
- continue;
- }
-
- /* all other widgets */
- in = is_connected_input_ep(w);
- dapm_clear_walk(w->codec);
- out = is_connected_output_ep(w);
- dapm_clear_walk(w->codec);
- power = (out != 0 && in != 0) ? 1 : 0;
- power_change = (w->power == power) ? 0: 1;
- w->power = power;
-
- if (!power_change)
- continue;
-
- /* call any power change event handlers */
- if (w->event)
- pr_debug("power %s event for %s flags %x\n",
- w->power ? "on" : "off",
- w->name, w->event_flags);
-
- /* power up pre event */
- if (power && w->event &&
- (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
- ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
- if (ret < 0)
- return ret;
- }
-
- /* power down pre event */
- if (!power && w->event &&
- (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
- ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
- if (ret < 0)
- return ret;
- }
-
- /* Lower PGA volume to reduce pops */
- if (w->id == snd_soc_dapm_pga && !power)
- dapm_set_pga(w, power);
-
- dapm_update_bits(w);
-
- /* Raise PGA volume to reduce pops */
- if (w->id == snd_soc_dapm_pga && power)
- dapm_set_pga(w, power);
-
- /* power up post event */
- if (power && w->event &&
- (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
- ret = w->event(w,
- NULL, SND_SOC_DAPM_POST_PMU);
- if (ret < 0)
- return ret;
- }
-
- /* power down post event */
- if (!power && w->event &&
- (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
- ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
- if (ret < 0)
- return ret;
- }
+ ret = dapm_power_widget(codec, event, w);
+ if (ret != 0)
+ return ret;
}
}
- return ret;
+ return 0;
}
#ifdef DEBUG
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index ab64a30..28346fb 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -14,6 +14,10 @@
#include <sound/jack.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
+#include <linux/gpio.h>
+#include <linux/interrupt.h>
+#include <linux/workqueue.h>
+#include <linux/delay.h>
/**
* snd_soc_jack_new - Create a new jack
@@ -136,3 +140,128 @@
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins);
+
+#ifdef CONFIG_GPIOLIB
+/* gpio detect */
+static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio)
+{
+ struct snd_soc_jack *jack = gpio->jack;
+ int enable;
+ int report;
+
+ if (gpio->debounce_time > 0)
+ mdelay(gpio->debounce_time);
+
+ enable = gpio_get_value(gpio->gpio);
+ if (gpio->invert)
+ enable = !enable;
+
+ if (enable)
+ report = gpio->report;
+ else
+ report = 0;
+
+ snd_soc_jack_report(jack, report, gpio->report);
+}
+
+/* irq handler for gpio pin */
+static irqreturn_t gpio_handler(int irq, void *data)
+{
+ struct snd_soc_jack_gpio *gpio = data;
+
+ schedule_work(&gpio->work);
+
+ return IRQ_HANDLED;
+}
+
+/* gpio work */
+static void gpio_work(struct work_struct *work)
+{
+ struct snd_soc_jack_gpio *gpio;
+
+ gpio = container_of(work, struct snd_soc_jack_gpio, work);
+ snd_soc_jack_gpio_detect(gpio);
+}
+
+/**
+ * snd_soc_jack_add_gpios - Associate GPIO pins with an ASoC jack
+ *
+ * @jack: ASoC jack
+ * @count: number of pins
+ * @gpios: array of gpio pins
+ *
+ * This function will request gpio, set data direction and request irq
+ * for each gpio in the array.
+ */
+int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
+ struct snd_soc_jack_gpio *gpios)
+{
+ int i, ret;
+
+ for (i = 0; i < count; i++) {
+ if (!gpio_is_valid(gpios[i].gpio)) {
+ printk(KERN_ERR "Invalid gpio %d\n",
+ gpios[i].gpio);
+ ret = -EINVAL;
+ goto undo;
+ }
+ if (!gpios[i].name) {
+ printk(KERN_ERR "No name for gpio %d\n",
+ gpios[i].gpio);
+ ret = -EINVAL;
+ goto undo;
+ }
+
+ ret = gpio_request(gpios[i].gpio, gpios[i].name);
+ if (ret)
+ goto undo;
+
+ ret = gpio_direction_input(gpios[i].gpio);
+ if (ret)
+ goto err;
+
+ ret = request_irq(gpio_to_irq(gpios[i].gpio),
+ gpio_handler,
+ IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING,
+ jack->card->dev->driver->name,
+ &gpios[i]);
+ if (ret)
+ goto err;
+
+ INIT_WORK(&gpios[i].work, gpio_work);
+ gpios[i].jack = jack;
+ }
+
+ return 0;
+
+err:
+ gpio_free(gpios[i].gpio);
+undo:
+ snd_soc_jack_free_gpios(jack, i, gpios);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_add_gpios);
+
+/**
+ * snd_soc_jack_free_gpios - Release GPIO pins' resources of an ASoC jack
+ *
+ * @jack: ASoC jack
+ * @count: number of pins
+ * @gpios: array of gpio pins
+ *
+ * Release gpio and irq resources for gpio pins associated with an ASoC jack.
+ */
+void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
+ struct snd_soc_jack_gpio *gpios)
+{
+ int i;
+
+ for (i = 0; i < count; i++) {
+ free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]);
+ gpio_free(gpios[i].gpio);
+ gpios[i].jack = NULL;
+ }
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_free_gpios);
+#endif /* CONFIG_GPIOLIB */
