Merge branch 'for-2.6.32' into for-2.6.33
diff --git a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
index 07659da..abf2fbc 100644
--- a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
+++ b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
@@ -67,6 +67,8 @@
#define S3C2412_IISMOD_BCLK_MASK (3 << 1)
#define S3C2412_IISMOD_8BIT (1 << 0)
+#define S3C64XX_IISMOD_CDCLKCON (1 << 12)
+
#define S3C2412_IISPSR_PSREN (1 << 15)
#define S3C2412_IISFIC_TXFLUSH (1 << 15)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 97ca9af..ca24e7f 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -30,6 +30,7 @@
#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */
#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */
#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
+#define SND_SOC_DAIFMT_PDM 6 /* Pulse density modulation */
/* left and right justified also known as MSB and LSB respectively */
#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
@@ -106,7 +107,7 @@
int div_id, int div);
int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out);
+ int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
/* Digital Audio interface formatting */
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
@@ -114,6 +115,10 @@
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
+int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot);
+
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
/* Digital Audio Interface mute */
@@ -136,8 +141,8 @@
*/
int (*set_sysclk)(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir);
- int (*set_pll)(struct snd_soc_dai *dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out);
+ int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
+ unsigned int freq_in, unsigned int freq_out);
int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
/*
@@ -148,6 +153,9 @@
int (*set_tdm_slot)(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask,
int slots, int slot_width);
+ int (*set_channel_map)(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot);
int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
/*
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index c1410e3..c5c95e1d 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -206,6 +206,12 @@
.get = snd_soc_dapm_get_enum_double, \
.put = snd_soc_dapm_put_enum_double, \
.private_value = (unsigned long)&xenum }
+#define SOC_DAPM_ENUM_VIRT(xname, xenum) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_enum_double, \
+ .get = snd_soc_dapm_get_enum_virt, \
+ .put = snd_soc_dapm_put_enum_virt, \
+ .private_value = (unsigned long)&xenum }
#define SOC_DAPM_VALUE_ENUM(xname, xenum) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_enum_double, \
@@ -260,6 +266,10 @@
struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
+int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
@@ -333,6 +343,10 @@
const char *sink;
const char *control;
const char *source;
+
+ /* Note: currently only supported for links where source is a supply */
+ int (*connected)(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink);
};
/* dapm audio path between two widgets */
@@ -349,6 +363,9 @@
u32 connect:1; /* source and sink widgets are connected */
u32 walked:1; /* path has been walked */
+ int (*connected)(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink);
+
struct list_head list_source;
struct list_head list_sink;
struct list_head list;
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 475cb7e..0b1f917a 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -413,6 +413,7 @@
unsigned int num_dai;
#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_codec_root;
struct dentry *debugfs_reg;
struct dentry *debugfs_pop_time;
struct dentry *debugfs_dapm;
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 9eb610c..9df4c68 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -268,7 +268,7 @@
#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
- ret = snd_soc_dai_set_pll(codec_dai, 0,
+ ret = snd_soc_dai_set_pll(codec_dai, 0, 0,
clk_get_rate(CODEC_CLK), pll_out);
if (ret < 0) {
pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n",
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index cd361e3..0f45a3f 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -52,6 +52,7 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7};
int ret = 0;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
@@ -65,6 +66,12 @@
if (ret < 0)
return ret;
+ /* set cpu DAI channel mapping */
+ ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map),
+ channel_map, ARRAY_SIZE(channel_map), channel_map);
+ if (ret < 0)
+ return ret;
+
return 0;
}
diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c
index 08269e9..2ef1e50 100644
--- a/sound/soc/blackfin/bf5xx-ad1938.c
+++ b/sound/soc/blackfin/bf5xx-ad1938.c
@@ -61,6 +61,7 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7};
int ret = 0;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
@@ -75,7 +76,13 @@
return ret;
/* set codec DAI slots, 8 channels, all channels are enabled */
- ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8);
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI channel mapping */
+ ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map),
+ channel_map, ARRAY_SIZE(channel_map), channel_map);
if (ret < 0)
return ret;
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 084b688..3e6ada0 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -49,7 +49,6 @@
u16 rcr1;
u16 tcr2;
u16 rcr2;
- int counter;
int configured;
};
@@ -133,16 +132,6 @@
return ret;
}
-static int bf5xx_i2s_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- pr_debug("%s enter\n", __func__);
-
- /*this counter is used for counting how many pcm streams are opened*/
- bf5xx_i2s.counter++;
- return 0;
-}
-
static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -201,9 +190,8 @@
struct snd_soc_dai *dai)
{
pr_debug("%s enter\n", __func__);
- bf5xx_i2s.counter--;
/* No active stream, SPORT is allowed to be configured again. */
- if (!bf5xx_i2s.counter)
+ if (!dai->active)
bf5xx_i2s.configured = 0;
}
@@ -284,7 +272,6 @@
SNDRV_PCM_FMTBIT_S32_LE)
static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = {
- .startup = bf5xx_i2s_startup,
.shutdown = bf5xx_i2s_shutdown,
.hw_params = bf5xx_i2s_hw_params,
.set_fmt = bf5xx_i2s_set_dai_fmt,
diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c
index ccb5e82..a8c73cb 100644
--- a/sound/soc/blackfin/bf5xx-tdm-pcm.c
+++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c
@@ -43,7 +43,7 @@
#include "bf5xx-tdm.h"
#include "bf5xx-sport.h"
-#define PCM_BUFFER_MAX 0x10000
+#define PCM_BUFFER_MAX 0x8000
#define FRAGMENT_SIZE_MIN (4*1024)
#define FRAGMENTS_MIN 2
#define FRAGMENTS_MAX 32
@@ -177,6 +177,9 @@
static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count)
{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
+ struct bf5xx_tdm_port *tdm_port = sport->private_data;
unsigned int *src;
unsigned int *dst;
int i;
@@ -188,7 +191,7 @@
dst += pos * 8;
while (count--) {
for (i = 0; i < substream->runtime->channels; i++)
- *(dst + i) = *src++;
+ *(dst + tdm_port->tx_map[i]) = *src++;
dst += 8;
}
} else {
@@ -198,7 +201,7 @@
src += pos * 8;
while (count--) {
for (i = 0; i < substream->runtime->channels; i++)
- *dst++ = *(src+i);
+ *dst++ = *(src + tdm_port->rx_map[i]);
src += 8;
}
}
diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
index ff546e9..4b36012 100644
--- a/sound/soc/blackfin/bf5xx-tdm.c
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -46,14 +46,6 @@
#include "bf5xx-sport.h"
#include "bf5xx-tdm.h"
-struct bf5xx_tdm_port {
- u16 tcr1;
- u16 rcr1;
- u16 tcr2;
- u16 rcr2;
- int configured;
-};
-
static struct bf5xx_tdm_port bf5xx_tdm;
static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
@@ -181,6 +173,40 @@
bf5xx_tdm.configured = 0;
}
+static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot)
+{
+ int i;
+ unsigned int slot;
+ unsigned int tx_mapped = 0, rx_mapped = 0;
+
+ if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) ||
+ (rx_num > BFIN_TDM_DAI_MAX_SLOTS))
+ return -EINVAL;
+
+ for (i = 0; i < tx_num; i++) {
+ slot = tx_slot[i];
+ if ((slot < BFIN_TDM_DAI_MAX_SLOTS) &&
+ (!(tx_mapped & (1 << slot)))) {
+ bf5xx_tdm.tx_map[i] = slot;
+ tx_mapped |= 1 << slot;
+ } else
+ return -EINVAL;
+ }
+ for (i = 0; i < rx_num; i++) {
+ slot = rx_slot[i];
+ if ((slot < BFIN_TDM_DAI_MAX_SLOTS) &&
+ (!(rx_mapped & (1 << slot)))) {
+ bf5xx_tdm.rx_map[i] = slot;
+ rx_mapped |= 1 << slot;
+ } else
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
#ifdef CONFIG_PM
static int bf5xx_tdm_suspend(struct snd_soc_dai *dai)
{
@@ -235,6 +261,7 @@
.hw_params = bf5xx_tdm_hw_params,
.set_fmt = bf5xx_tdm_set_dai_fmt,
.shutdown = bf5xx_tdm_shutdown,
+ .set_channel_map = bf5xx_tdm_set_channel_map,
};
struct snd_soc_dai bf5xx_tdm_dai = {
@@ -300,6 +327,8 @@
pr_err("Failed to register DAI: %d\n", ret);
goto sport_config_err;
}
+
+ sport_handle->private_data = &bf5xx_tdm;
return 0;
sport_config_err:
diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h
index 618ec3d..04189a1 100644
--- a/sound/soc/blackfin/bf5xx-tdm.h
+++ b/sound/soc/blackfin/bf5xx-tdm.h
@@ -9,6 +9,17 @@
#ifndef _BF5XX_TDM_H
#define _BF5XX_TDM_H
+#define BFIN_TDM_DAI_MAX_SLOTS 8
+struct bf5xx_tdm_port {
+ u16 tcr1;
+ u16 rcr1;
+ u16 tcr2;
+ u16 rcr2;
+ unsigned int tx_map[BFIN_TDM_DAI_MAX_SLOTS];
+ unsigned int rx_map[BFIN_TDM_DAI_MAX_SLOTS];
+ int configured;
+};
+
extern struct snd_soc_dai bf5xx_tdm_dai;
#endif
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 0edca93..3c46f34 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -19,6 +19,7 @@
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
select SND_SOC_AK4642 if I2C
+ select SND_SOC_AK4671 if I2C
select SND_SOC_CS4270 if I2C
select SND_SOC_MAX9877 if I2C
select SND_SOC_PCM3008
@@ -36,6 +37,7 @@
select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8523 if I2C
select SND_SOC_WM8580 if I2C
+ select SND_SOC_WM8711 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI
@@ -96,6 +98,9 @@
config SND_SOC_AK4642
tristate
+config SND_SOC_AK4671
+ tristate
+
# Cirrus Logic CS4270 Codec
config SND_SOC_CS4270
tristate
@@ -160,6 +165,9 @@
config SND_SOC_WM8580
tristate
+config SND_SOC_WM8711
+ tristate
+
config SND_SOC_WM8728
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index fb4af28..fc1c458 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -6,6 +6,7 @@
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
snd-soc-ak4642-objs := ak4642.o
+snd-soc-ak4671-objs := ak4671.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-cx20442-objs := cx20442.o
snd-soc-l3-objs := l3.o
@@ -24,6 +25,7 @@
snd-soc-wm8510-objs := wm8510.o
snd-soc-wm8523-objs := wm8523.o
snd-soc-wm8580-objs := wm8580.o
+snd-soc-wm8711-objs := wm8711.o
snd-soc-wm8728-objs := wm8728.o
snd-soc-wm8731-objs := wm8731.o
snd-soc-wm8750-objs := wm8750.o
@@ -56,6 +58,7 @@
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
+obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
@@ -74,6 +77,7 @@
obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o
obj-$(CONFIG_SND_SOC_WM8523) += snd-soc-wm8523.o
obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o
+obj-$(CONFIG_SND_SOC_WM8711) += snd-soc-wm8711.o
obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o
obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o
obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
new file mode 100644
index 0000000..b61214d
--- /dev/null
+++ b/sound/soc/codecs/ak4671.c
@@ -0,0 +1,825 @@
+/*
+ * ak4671.c -- audio driver for AK4671
+ *
+ * Copyright (C) 2009 Samsung Electronics Co.Ltd
+ * Author: Joonyoung Shim <jy0922.shim@samsung.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/delay.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "ak4671.h"
+
+static struct snd_soc_codec *ak4671_codec;
+
+/* codec private data */
+struct ak4671_priv {
+ struct snd_soc_codec codec;
+ u8 reg_cache[AK4671_CACHEREGNUM];
+};
+
+/* ak4671 register cache & default register settings */
+static const u8 ak4671_reg[AK4671_CACHEREGNUM] = {
+ 0x00, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */
+ 0xf6, /* AK4671_PLL_MODE_SELECT0 (0x01) */
+ 0x00, /* AK4671_PLL_MODE_SELECT1 (0x02) */
+ 0x02, /* AK4671_FORMAT_SELECT (0x03) */
+ 0x00, /* AK4671_MIC_SIGNAL_SELECT (0x04) */
+ 0x55, /* AK4671_MIC_AMP_GAIN (0x05) */
+ 0x00, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */
+ 0x00, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */
+ 0xb5, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */
+ 0x00, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */
+ 0x00, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */
+ 0x00, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */
+ 0x00, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */
+ 0x00, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */
+ 0x00, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */
+ 0x00, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */
+ 0x00, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */
+ 0x80, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */
+ 0x91, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */
+ 0x91, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */
+ 0xe1, /* AK4671_ALC_REFERENCE_SELECT (0x14) */
+ 0x00, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */
+ 0x00, /* AK4671_ALC_TIMER_SELECT (0x16) */
+ 0x00, /* AK4671_ALC_MODE_CONTROL (0x17) */
+ 0x02, /* AK4671_MODE_CONTROL1 (0x18) */
+ 0x01, /* AK4671_MODE_CONTROL2 (0x19) */
+ 0x18, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */
+ 0x18, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */
+ 0x00, /* AK4671_SIDETONE_A_CONTROL (0x1c) */
+ 0x02, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */
+ 0x00, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */
+ 0x00, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */
+ 0x00, /* AK4671_FIL3_COEFFICIENT2 (0x20) */
+ 0x00, /* AK4671_FIL3_COEFFICIENT3 (0x21) */
+ 0x00, /* AK4671_EQ_COEFFICIENT0 (0x22) */
+ 0x00, /* AK4671_EQ_COEFFICIENT1 (0x23) */
+ 0x00, /* AK4671_EQ_COEFFICIENT2 (0x24) */
+ 0x00, /* AK4671_EQ_COEFFICIENT3 (0x25) */
+ 0x00, /* AK4671_EQ_COEFFICIENT4 (0x26) */
+ 0x00, /* AK4671_EQ_COEFFICIENT5 (0x27) */
+ 0xa9, /* AK4671_FIL1_COEFFICIENT0 (0x28) */
+ 0x1f, /* AK4671_FIL1_COEFFICIENT1 (0x29) */
+ 0xad, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */
+ 0x20, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */
+ 0x00, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */
+ 0x00, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */
+ 0x00, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */
+ 0x00, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */
+ 0x00, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */
+ 0x00, /* this register not used */
+ 0x00, /* AK4671_E1_COEFFICIENT0 (0x32) */
+ 0x00, /* AK4671_E1_COEFFICIENT1 (0x33) */
+ 0x00, /* AK4671_E1_COEFFICIENT2 (0x34) */
+ 0x00, /* AK4671_E1_COEFFICIENT3 (0x35) */
+ 0x00, /* AK4671_E1_COEFFICIENT4 (0x36) */
+ 0x00, /* AK4671_E1_COEFFICIENT5 (0x37) */
+ 0x00, /* AK4671_E2_COEFFICIENT0 (0x38) */
+ 0x00, /* AK4671_E2_COEFFICIENT1 (0x39) */
+ 0x00, /* AK4671_E2_COEFFICIENT2 (0x3a) */
+ 0x00, /* AK4671_E2_COEFFICIENT3 (0x3b) */
+ 0x00, /* AK4671_E2_COEFFICIENT4 (0x3c) */
+ 0x00, /* AK4671_E2_COEFFICIENT5 (0x3d) */
+ 0x00, /* AK4671_E3_COEFFICIENT0 (0x3e) */
+ 0x00, /* AK4671_E3_COEFFICIENT1 (0x3f) */
+ 0x00, /* AK4671_E3_COEFFICIENT2 (0x40) */
+ 0x00, /* AK4671_E3_COEFFICIENT3 (0x41) */
+ 0x00, /* AK4671_E3_COEFFICIENT4 (0x42) */
+ 0x00, /* AK4671_E3_COEFFICIENT5 (0x43) */
+ 0x00, /* AK4671_E4_COEFFICIENT0 (0x44) */
+ 0x00, /* AK4671_E4_COEFFICIENT1 (0x45) */
+ 0x00, /* AK4671_E4_COEFFICIENT2 (0x46) */
+ 0x00, /* AK4671_E4_COEFFICIENT3 (0x47) */
+ 0x00, /* AK4671_E4_COEFFICIENT4 (0x48) */
+ 0x00, /* AK4671_E4_COEFFICIENT5 (0x49) */
+ 0x00, /* AK4671_E5_COEFFICIENT0 (0x4a) */
+ 0x00, /* AK4671_E5_COEFFICIENT1 (0x4b) */
+ 0x00, /* AK4671_E5_COEFFICIENT2 (0x4c) */
+ 0x00, /* AK4671_E5_COEFFICIENT3 (0x4d) */
+ 0x00, /* AK4671_E5_COEFFICIENT4 (0x4e) */
+ 0x00, /* AK4671_E5_COEFFICIENT5 (0x4f) */
+ 0x88, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */
+ 0x88, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */
+ 0x08, /* AK4671_EQ_CONTRO_10KHZ (0x52) */
+ 0x00, /* AK4671_PCM_IF_CONTROL0 (0x53) */
+ 0x00, /* AK4671_PCM_IF_CONTROL1 (0x54) */
+ 0x00, /* AK4671_PCM_IF_CONTROL2 (0x55) */
+ 0x18, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */
+ 0x18, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */
+ 0x00, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */
+ 0x00, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */
+ 0x00, /* AK4671_SAR_ADC_CONTROL (0x5a) */
+};
+
+/*
+ * LOUT1/ROUT1 output volume control:
+ * from -24 to 6 dB in 6 dB steps (mute instead of -30 dB)
+ */
+static DECLARE_TLV_DB_SCALE(out1_tlv, -3000, 600, 1);
+
+/*
+ * LOUT2/ROUT2 output volume control:
+ * from -33 to 6 dB in 3 dB steps (mute instead of -33 dB)
+ */
+static DECLARE_TLV_DB_SCALE(out2_tlv, -3300, 300, 1);
+
+/*
+ * LOUT3/ROUT3 output volume control:
+ * from -6 to 3 dB in 3 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(out3_tlv, -600, 300, 0);
+
+/*
+ * Mic amp gain control:
+ * from -15 to 30 dB in 3 dB steps
+ * REVISIT: The actual min value(0x01) is -12 dB and the reg value 0x00 is not
+ * available
+ */
+static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -1500, 300, 0);
+
+static const struct snd_kcontrol_new ak4671_snd_controls[] = {
+ /* Common playback gain controls */
+ SOC_SINGLE_TLV("Line Output1 Playback Volume",
+ AK4671_OUTPUT_VOLUME_CONTROL, 0, 0x6, 0, out1_tlv),
+ SOC_SINGLE_TLV("Headphone Output2 Playback Volume",
+ AK4671_OUTPUT_VOLUME_CONTROL, 4, 0xd, 0, out2_tlv),
+ SOC_SINGLE_TLV("Line Output3 Playback Volume",
+ AK4671_LOUT3_POWER_MANAGERMENT, 6, 0x3, 0, out3_tlv),
+
+ /* Common capture gain controls */
+ SOC_DOUBLE_TLV("Mic Amp Capture Volume",
+ AK4671_MIC_AMP_GAIN, 0, 4, 0xf, 0, mic_amp_tlv),
+};
+
+/* event handlers */
+static int ak4671_out2_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u8 reg;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT);
+ reg |= AK4671_MUTEN;
+ snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT);
+ reg &= ~AK4671_MUTEN;
+ snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg);
+ break;
+ }
+
+ return 0;
+}
+
+/* Output Mixers */
+static const struct snd_kcontrol_new ak4671_lout1_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DACL", AK4671_LOUT1_SIGNAL_SELECT, 0, 1, 0),
+ SOC_DAPM_SINGLE("LINL1", AK4671_LOUT1_SIGNAL_SELECT, 1, 1, 0),
+ SOC_DAPM_SINGLE("LINL2", AK4671_LOUT1_SIGNAL_SELECT, 2, 1, 0),
+ SOC_DAPM_SINGLE("LINL3", AK4671_LOUT1_SIGNAL_SELECT, 3, 1, 0),
+ SOC_DAPM_SINGLE("LINL4", AK4671_LOUT1_SIGNAL_SELECT, 4, 1, 0),
+ SOC_DAPM_SINGLE("LOOPL", AK4671_LOUT1_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_rout1_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DACR", AK4671_ROUT1_SIGNAL_SELECT, 0, 1, 0),
+ SOC_DAPM_SINGLE("RINR1", AK4671_ROUT1_SIGNAL_SELECT, 1, 1, 0),
+ SOC_DAPM_SINGLE("RINR2", AK4671_ROUT1_SIGNAL_SELECT, 2, 1, 0),
+ SOC_DAPM_SINGLE("RINR3", AK4671_ROUT1_SIGNAL_SELECT, 3, 1, 0),
+ SOC_DAPM_SINGLE("RINR4", AK4671_ROUT1_SIGNAL_SELECT, 4, 1, 0),
+ SOC_DAPM_SINGLE("LOOPR", AK4671_ROUT1_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_lout2_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DACHL", AK4671_LOUT2_SIGNAL_SELECT, 0, 1, 0),
+ SOC_DAPM_SINGLE("LINH1", AK4671_LOUT2_SIGNAL_SELECT, 1, 1, 0),
+ SOC_DAPM_SINGLE("LINH2", AK4671_LOUT2_SIGNAL_SELECT, 2, 1, 0),
+ SOC_DAPM_SINGLE("LINH3", AK4671_LOUT2_SIGNAL_SELECT, 3, 1, 0),
+ SOC_DAPM_SINGLE("LINH4", AK4671_LOUT2_SIGNAL_SELECT, 4, 1, 0),
+ SOC_DAPM_SINGLE("LOOPHL", AK4671_LOUT2_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_rout2_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DACHR", AK4671_ROUT2_SIGNAL_SELECT, 0, 1, 0),
+ SOC_DAPM_SINGLE("RINH1", AK4671_ROUT2_SIGNAL_SELECT, 1, 1, 0),
+ SOC_DAPM_SINGLE("RINH2", AK4671_ROUT2_SIGNAL_SELECT, 2, 1, 0),
+ SOC_DAPM_SINGLE("RINH3", AK4671_ROUT2_SIGNAL_SELECT, 3, 1, 0),
+ SOC_DAPM_SINGLE("RINH4", AK4671_ROUT2_SIGNAL_SELECT, 4, 1, 0),
+ SOC_DAPM_SINGLE("LOOPHR", AK4671_ROUT2_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_lout3_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DACSL", AK4671_LOUT3_SIGNAL_SELECT, 0, 1, 0),
+ SOC_DAPM_SINGLE("LINS1", AK4671_LOUT3_SIGNAL_SELECT, 1, 1, 0),
+ SOC_DAPM_SINGLE("LINS2", AK4671_LOUT3_SIGNAL_SELECT, 2, 1, 0),
+ SOC_DAPM_SINGLE("LINS3", AK4671_LOUT3_SIGNAL_SELECT, 3, 1, 0),
+ SOC_DAPM_SINGLE("LINS4", AK4671_LOUT3_SIGNAL_SELECT, 4, 1, 0),
+ SOC_DAPM_SINGLE("LOOPSL", AK4671_LOUT3_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_rout3_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DACSR", AK4671_ROUT3_SIGNAL_SELECT, 0, 1, 0),
+ SOC_DAPM_SINGLE("RINS1", AK4671_ROUT3_SIGNAL_SELECT, 1, 1, 0),
+ SOC_DAPM_SINGLE("RINS2", AK4671_ROUT3_SIGNAL_SELECT, 2, 1, 0),
+ SOC_DAPM_SINGLE("RINS3", AK4671_ROUT3_SIGNAL_SELECT, 3, 1, 0),
+ SOC_DAPM_SINGLE("RINS4", AK4671_ROUT3_SIGNAL_SELECT, 4, 1, 0),
+ SOC_DAPM_SINGLE("LOOPSR", AK4671_ROUT3_SIGNAL_SELECT, 5, 1, 0),
+};
+
+/* Input MUXs */
+static const char *ak4671_lin_mux_texts[] =
+ {"LIN1", "LIN2", "LIN3", "LIN4"};
+static const struct soc_enum ak4671_lin_mux_enum =
+ SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 0,
+ ARRAY_SIZE(ak4671_lin_mux_texts),
+ ak4671_lin_mux_texts);
+static const struct snd_kcontrol_new ak4671_lin_mux_control =
+ SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum);
+
+static const char *ak4671_rin_mux_texts[] =
+ {"RIN1", "RIN2", "RIN3", "RIN4"};
+static const struct soc_enum ak4671_rin_mux_enum =
+ SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 2,
+ ARRAY_SIZE(ak4671_rin_mux_texts),
+ ak4671_rin_mux_texts);
+static const struct snd_kcontrol_new ak4671_rin_mux_control =
+ SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum);
+
+static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = {
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("LIN1"),
+ SND_SOC_DAPM_INPUT("RIN1"),
+ SND_SOC_DAPM_INPUT("LIN2"),
+ SND_SOC_DAPM_INPUT("RIN2"),
+ SND_SOC_DAPM_INPUT("LIN3"),
+ SND_SOC_DAPM_INPUT("RIN3"),
+ SND_SOC_DAPM_INPUT("LIN4"),
+ SND_SOC_DAPM_INPUT("RIN4"),
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("LOUT1"),
+ SND_SOC_DAPM_OUTPUT("ROUT1"),
+ SND_SOC_DAPM_OUTPUT("LOUT2"),
+ SND_SOC_DAPM_OUTPUT("ROUT2"),
+ SND_SOC_DAPM_OUTPUT("LOUT3"),
+ SND_SOC_DAPM_OUTPUT("ROUT3"),
+
+ /* DAC */
+ SND_SOC_DAPM_DAC("DAC Left", "Left HiFi Playback",
+ AK4671_AD_DA_POWER_MANAGEMENT, 6, 0),
+ SND_SOC_DAPM_DAC("DAC Right", "Right HiFi Playback",
+ AK4671_AD_DA_POWER_MANAGEMENT, 7, 0),
+
+ /* ADC */
+ SND_SOC_DAPM_ADC("ADC Left", "Left HiFi Capture",
+ AK4671_AD_DA_POWER_MANAGEMENT, 4, 0),
+ SND_SOC_DAPM_ADC("ADC Right", "Right HiFi Capture",
+ AK4671_AD_DA_POWER_MANAGEMENT, 5, 0),
+
+ /* PGA */
+ SND_SOC_DAPM_PGA("LOUT2 Mix Amp",
+ AK4671_LOUT2_POWER_MANAGERMENT, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("ROUT2 Mix Amp",
+ AK4671_LOUT2_POWER_MANAGERMENT, 6, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA("LIN1 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("RIN1 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("LIN2 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 2, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("RIN2 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("LIN3 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("RIN3 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("LIN4 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 6, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("RIN4 Mixing Circuit",
+ AK4671_MIXING_POWER_MANAGEMENT1, 7, 0, NULL, 0),
+
+ /* Output Mixers */
+ SND_SOC_DAPM_MIXER("LOUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 0, 0,
+ &ak4671_lout1_mixer_controls[0],
+ ARRAY_SIZE(ak4671_lout1_mixer_controls)),
+ SND_SOC_DAPM_MIXER("ROUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 1, 0,
+ &ak4671_rout1_mixer_controls[0],
+ ARRAY_SIZE(ak4671_rout1_mixer_controls)),
+ SND_SOC_DAPM_MIXER_E("LOUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT,
+ 0, 0, &ak4671_lout2_mixer_controls[0],
+ ARRAY_SIZE(ak4671_lout2_mixer_controls),
+ ak4671_out2_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_MIXER_E("ROUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT,
+ 1, 0, &ak4671_rout2_mixer_controls[0],
+ ARRAY_SIZE(ak4671_rout2_mixer_controls),
+ ak4671_out2_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_MIXER("LOUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 0, 0,
+ &ak4671_lout3_mixer_controls[0],
+ ARRAY_SIZE(ak4671_lout3_mixer_controls)),
+ SND_SOC_DAPM_MIXER("ROUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 1, 0,
+ &ak4671_rout3_mixer_controls[0],
+ ARRAY_SIZE(ak4671_rout3_mixer_controls)),
+
+ /* Input MUXs */
+ SND_SOC_DAPM_MUX("LIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 2, 0,
+ &ak4671_lin_mux_control),
+ SND_SOC_DAPM_MUX("RIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 3, 0,
+ &ak4671_rin_mux_control),
+
+ /* Mic Power */
+ SND_SOC_DAPM_MICBIAS("Mic Bias", AK4671_AD_DA_POWER_MANAGEMENT, 1, 0),
+
+ /* Supply */
+ SND_SOC_DAPM_SUPPLY("PMPLL", AK4671_PLL_MODE_SELECT1, 0, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"DAC Left", "NULL", "PMPLL"},
+ {"DAC Right", "NULL", "PMPLL"},
+ {"ADC Left", "NULL", "PMPLL"},
+ {"ADC Right", "NULL", "PMPLL"},
+
+ /* Outputs */
+ {"LOUT1", "NULL", "LOUT1 Mixer"},
+ {"ROUT1", "NULL", "ROUT1 Mixer"},
+ {"LOUT2", "NULL", "LOUT2 Mix Amp"},
+ {"ROUT2", "NULL", "ROUT2 Mix Amp"},
+ {"LOUT3", "NULL", "LOUT3 Mixer"},
+ {"ROUT3", "NULL", "ROUT3 Mixer"},
+
+ {"LOUT1 Mixer", "DACL", "DAC Left"},
+ {"ROUT1 Mixer", "DACR", "DAC Right"},
+ {"LOUT2 Mixer", "DACHL", "DAC Left"},
+ {"ROUT2 Mixer", "DACHR", "DAC Right"},
+ {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"},
+ {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"},
+ {"LOUT3 Mixer", "DACSL", "DAC Left"},
+ {"ROUT3 Mixer", "DACSR", "DAC Right"},
+
+ /* Inputs */
+ {"LIN MUX", "LIN1", "LIN1"},
+ {"LIN MUX", "LIN2", "LIN2"},
+ {"LIN MUX", "LIN3", "LIN3"},
+ {"LIN MUX", "LIN4", "LIN4"},
+
+ {"RIN MUX", "RIN1", "RIN1"},
+ {"RIN MUX", "RIN2", "RIN2"},
+ {"RIN MUX", "RIN3", "RIN3"},
+ {"RIN MUX", "RIN4", "RIN4"},
+
+ {"LIN1", NULL, "Mic Bias"},
+ {"RIN1", NULL, "Mic Bias"},
+ {"LIN2", NULL, "Mic Bias"},
+ {"RIN2", NULL, "Mic Bias"},
+
+ {"ADC Left", "NULL", "LIN MUX"},
+ {"ADC Right", "NULL", "RIN MUX"},
+
+ /* Analog Loops */
+ {"LIN1 Mixing Circuit", "NULL", "LIN1"},
+ {"RIN1 Mixing Circuit", "NULL", "RIN1"},
+ {"LIN2 Mixing Circuit", "NULL", "LIN2"},
+ {"RIN2 Mixing Circuit", "NULL", "RIN2"},
+ {"LIN3 Mixing Circuit", "NULL", "LIN3"},
+ {"RIN3 Mixing Circuit", "NULL", "RIN3"},
+ {"LIN4 Mixing Circuit", "NULL", "LIN4"},
+ {"RIN4 Mixing Circuit", "NULL", "RIN4"},
+
+ {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"},
+ {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"},
+ {"LOUT2 Mixer", "LINH1", "LIN1 Mixing Circuit"},
+ {"ROUT2 Mixer", "RINH1", "RIN1 Mixing Circuit"},
+ {"LOUT3 Mixer", "LINS1", "LIN1 Mixing Circuit"},
+ {"ROUT3 Mixer", "RINS1", "RIN1 Mixing Circuit"},
+
+ {"LOUT1 Mixer", "LINL2", "LIN2 Mixing Circuit"},
+ {"ROUT1 Mixer", "RINR2", "RIN2 Mixing Circuit"},
+ {"LOUT2 Mixer", "LINH2", "LIN2 Mixing Circuit"},
+ {"ROUT2 Mixer", "RINH2", "RIN2 Mixing Circuit"},
+ {"LOUT3 Mixer", "LINS2", "LIN2 Mixing Circuit"},
+ {"ROUT3 Mixer", "RINS2", "RIN2 Mixing Circuit"},
+
+ {"LOUT1 Mixer", "LINL3", "LIN3 Mixing Circuit"},
+ {"ROUT1 Mixer", "RINR3", "RIN3 Mixing Circuit"},
+ {"LOUT2 Mixer", "LINH3", "LIN3 Mixing Circuit"},
+ {"ROUT2 Mixer", "RINH3", "RIN3 Mixing Circuit"},
+ {"LOUT3 Mixer", "LINS3", "LIN3 Mixing Circuit"},
+ {"ROUT3 Mixer", "RINS3", "RIN3 Mixing Circuit"},
+
+ {"LOUT1 Mixer", "LINL4", "LIN4 Mixing Circuit"},
+ {"ROUT1 Mixer", "RINR4", "RIN4 Mixing Circuit"},
+ {"LOUT2 Mixer", "LINH4", "LIN4 Mixing Circuit"},
+ {"ROUT2 Mixer", "RINH4", "RIN4 Mixing Circuit"},
+ {"LOUT3 Mixer", "LINS4", "LIN4 Mixing Circuit"},
+ {"ROUT3 Mixer", "RINS4", "RIN4 Mixing Circuit"},
+};
+
+static int ak4671_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, ak4671_dapm_widgets,
+ ARRAY_SIZE(ak4671_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static int ak4671_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 fs;
+
+ fs = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0);
+ fs &= ~AK4671_FS;
+
+ switch (params_rate(params)) {
+ case 8000:
+ fs |= AK4671_FS_8KHZ;
+ break;
+ case 12000:
+ fs |= AK4671_FS_12KHZ;
+ break;
+ case 16000:
+ fs |= AK4671_FS_16KHZ;
+ break;
+ case 24000:
+ fs |= AK4671_FS_24KHZ;
+ break;
+ case 11025:
+ fs |= AK4671_FS_11_025KHZ;
+ break;
+ case 22050:
+ fs |= AK4671_FS_22_05KHZ;
+ break;
+ case 32000:
+ fs |= AK4671_FS_32KHZ;
+ break;
+ case 44100:
+ fs |= AK4671_FS_44_1KHZ;
+ break;
+ case 48000:
+ fs |= AK4671_FS_48KHZ;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, fs);
+
+ return 0;
+}
+
+static int ak4671_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 pll;
+
+ pll = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0);
+ pll &= ~AK4671_PLL;
+
+ switch (freq) {
+ case 11289600:
+ pll |= AK4671_PLL_11_2896MHZ;
+ break;
+ case 12000000:
+ pll |= AK4671_PLL_12MHZ;
+ break;
+ case 12288000:
+ pll |= AK4671_PLL_12_288MHZ;
+ break;
+ case 13000000:
+ pll |= AK4671_PLL_13MHZ;
+ break;
+ case 13500000:
+ pll |= AK4671_PLL_13_5MHZ;
+ break;
+ case 19200000:
+ pll |= AK4671_PLL_19_2MHZ;
+ break;
+ case 24000000:
+ pll |= AK4671_PLL_24MHZ;
+ break;
+ case 26000000:
+ pll |= AK4671_PLL_26MHZ;
+ break;
+ case 27000000:
+ pll |= AK4671_PLL_27MHZ;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, pll);
+
+ return 0;
+}
+
+static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 mode;
+ u8 format;
+
+ /* set master/slave audio interface */
+ mode = snd_soc_read(codec, AK4671_PLL_MODE_SELECT1);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ mode |= AK4671_M_S;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ mode &= ~(AK4671_M_S);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ format = snd_soc_read(codec, AK4671_FORMAT_SELECT);
+ format &= ~AK4671_DIF;
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ format |= AK4671_DIF_I2S_MODE;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ format |= AK4671_DIF_MSB_MODE;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ format |= AK4671_DIF_DSP_MODE;
+ format |= AK4671_BCKP;
+ format |= AK4671_MSBS;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set mode and format */
+ snd_soc_write(codec, AK4671_PLL_MODE_SELECT1, mode);
+ snd_soc_write(codec, AK4671_FORMAT_SELECT, format);
+
+ return 0;
+}
+
+static int ak4671_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u8 reg;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ case SND_SOC_BIAS_STANDBY:
+ reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT);
+ snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT,
+ reg | AK4671_PMVCM);
+ break;
+ case SND_SOC_BIAS_OFF:
+ snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define AK4671_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000)
+
+#define AK4671_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+
+static struct snd_soc_dai_ops ak4671_dai_ops = {
+ .hw_params = ak4671_hw_params,
+ .set_sysclk = ak4671_set_dai_sysclk,
+ .set_fmt = ak4671_set_dai_fmt,
+};
+
+struct snd_soc_dai ak4671_dai = {
+ .name = "AK4671",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AK4671_RATES,
+ .formats = AK4671_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AK4671_RATES,
+ .formats = AK4671_FORMATS,},
+ .ops = &ak4671_dai_ops,
+};
+EXPORT_SYMBOL_GPL(ak4671_dai);
+
+static int ak4671_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (ak4671_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = ak4671_codec;
+ codec = ak4671_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ snd_soc_add_controls(codec, ak4671_snd_controls,
+ ARRAY_SIZE(ak4671_snd_controls));
+ ak4671_add_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto card_err;
+ }
+
+ ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+static int ak4671_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ak4671 = {
+ .probe = ak4671_probe,
+ .remove = ak4671_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ak4671);
+
+static int ak4671_register(struct ak4671_priv *ak4671,
+ enum snd_soc_control_type control)
+{
+ int ret;
+ struct snd_soc_codec *codec = &ak4671->codec;
+
+ if (ak4671_codec) {
+ dev_err(codec->dev, "Another AK4671 is registered\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = ak4671;
+ codec->name = "AK4671";
+ codec->owner = THIS_MODULE;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = ak4671_set_bias_level;
+ codec->dai = &ak4671_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = AK4671_CACHEREGNUM;
+ codec->reg_cache = &ak4671->reg_cache;
+
+ memcpy(codec->reg_cache, ak4671_reg, sizeof(ak4671_reg));
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, control);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
+ ak4671_dai.dev = codec->dev;
+ ak4671_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ goto err;
+ }
+
+ ret = snd_soc_register_dai(&ak4671_dai);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ goto err_codec;
+ }
+
+ return 0;
+
+err_codec:
+ snd_soc_unregister_codec(codec);
+err:
+ kfree(ak4671);
+ return ret;
+}
+
+static void ak4671_unregister(struct ak4671_priv *ak4671)
+{
+ ak4671_set_bias_level(&ak4671->codec, SND_SOC_BIAS_OFF);
+ snd_soc_unregister_dai(&ak4671_dai);
+ snd_soc_unregister_codec(&ak4671->codec);
+ kfree(ak4671);
+ ak4671_codec = NULL;
+}
+
+static int __devinit ak4671_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct ak4671_priv *ak4671;
+ struct snd_soc_codec *codec;
+
+ ak4671 = kzalloc(sizeof(struct ak4671_priv), GFP_KERNEL);
+ if (ak4671 == NULL)
+ return -ENOMEM;
+
+ codec = &ak4671->codec;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+
+ i2c_set_clientdata(client, ak4671);
+ codec->control_data = client;
+
+ codec->dev = &client->dev;
+
+ return ak4671_register(ak4671, SND_SOC_I2C);
+}
+
+static __devexit int ak4671_i2c_remove(struct i2c_client *client)
+{
+ struct ak4671_priv *ak4671 = i2c_get_clientdata(client);
+
+ ak4671_unregister(ak4671);
+
+ return 0;
+}
+
+static const struct i2c_device_id ak4671_i2c_id[] = {
+ { "ak4671", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id);
+
+static struct i2c_driver ak4671_i2c_driver = {
+ .driver = {
+ .name = "ak4671",
+ .owner = THIS_MODULE,
+ },
+ .probe = ak4671_i2c_probe,
+ .remove = __devexit_p(ak4671_i2c_remove),
+ .id_table = ak4671_i2c_id,
+};
+
+static int __init ak4671_modinit(void)
+{
+ return i2c_add_driver(&ak4671_i2c_driver);
+}
+module_init(ak4671_modinit);
+
+static void __exit ak4671_exit(void)
+{
+ i2c_del_driver(&ak4671_i2c_driver);
+}
+module_exit(ak4671_exit);
+
+MODULE_DESCRIPTION("ASoC AK4671 codec driver");
+MODULE_AUTHOR("Joonyoung Shim <jy0922.shim@samsung.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ak4671.h b/sound/soc/codecs/ak4671.h
new file mode 100644
index 0000000..e2fad96
--- /dev/null
+++ b/sound/soc/codecs/ak4671.h
@@ -0,0 +1,156 @@
+/*
+ * ak4671.h -- audio driver for AK4671
+ *
+ * Copyright (C) 2009 Samsung Electronics Co.Ltd
+ * Author: Joonyoung Shim <jy0922.shim@samsung.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef _AK4671_H
+#define _AK4671_H
+
+#define AK4671_AD_DA_POWER_MANAGEMENT 0x00
+#define AK4671_PLL_MODE_SELECT0 0x01
+#define AK4671_PLL_MODE_SELECT1 0x02
+#define AK4671_FORMAT_SELECT 0x03
+#define AK4671_MIC_SIGNAL_SELECT 0x04
+#define AK4671_MIC_AMP_GAIN 0x05
+#define AK4671_MIXING_POWER_MANAGEMENT0 0x06
+#define AK4671_MIXING_POWER_MANAGEMENT1 0x07
+#define AK4671_OUTPUT_VOLUME_CONTROL 0x08
+#define AK4671_LOUT1_SIGNAL_SELECT 0x09
+#define AK4671_ROUT1_SIGNAL_SELECT 0x0a
+#define AK4671_LOUT2_SIGNAL_SELECT 0x0b
+#define AK4671_ROUT2_SIGNAL_SELECT 0x0c
+#define AK4671_LOUT3_SIGNAL_SELECT 0x0d
+#define AK4671_ROUT3_SIGNAL_SELECT 0x0e
+#define AK4671_LOUT1_POWER_MANAGERMENT 0x0f
+#define AK4671_LOUT2_POWER_MANAGERMENT 0x10
+#define AK4671_LOUT3_POWER_MANAGERMENT 0x11
+#define AK4671_LCH_INPUT_VOLUME_CONTROL 0x12
+#define AK4671_RCH_INPUT_VOLUME_CONTROL 0x13
+#define AK4671_ALC_REFERENCE_SELECT 0x14
+#define AK4671_DIGITAL_MIXING_CONTROL 0x15
+#define AK4671_ALC_TIMER_SELECT 0x16
+#define AK4671_ALC_MODE_CONTROL 0x17
+#define AK4671_MODE_CONTROL1 0x18
+#define AK4671_MODE_CONTROL2 0x19
+#define AK4671_LCH_OUTPUT_VOLUME_CONTROL 0x1a
+#define AK4671_RCH_OUTPUT_VOLUME_CONTROL 0x1b
+#define AK4671_SIDETONE_A_CONTROL 0x1c
+#define AK4671_DIGITAL_FILTER_SELECT 0x1d
+#define AK4671_FIL3_COEFFICIENT0 0x1e
+#define AK4671_FIL3_COEFFICIENT1 0x1f
+#define AK4671_FIL3_COEFFICIENT2 0x20
+#define AK4671_FIL3_COEFFICIENT3 0x21
+#define AK4671_EQ_COEFFICIENT0 0x22
+#define AK4671_EQ_COEFFICIENT1 0x23
+#define AK4671_EQ_COEFFICIENT2 0x24
+#define AK4671_EQ_COEFFICIENT3 0x25
+#define AK4671_EQ_COEFFICIENT4 0x26
+#define AK4671_EQ_COEFFICIENT5 0x27
+#define AK4671_FIL1_COEFFICIENT0 0x28
+#define AK4671_FIL1_COEFFICIENT1 0x29
+#define AK4671_FIL1_COEFFICIENT2 0x2a
+#define AK4671_FIL1_COEFFICIENT3 0x2b
+#define AK4671_FIL2_COEFFICIENT0 0x2c
+#define AK4671_FIL2_COEFFICIENT1 0x2d
+#define AK4671_FIL2_COEFFICIENT2 0x2e
+#define AK4671_FIL2_COEFFICIENT3 0x2f
+#define AK4671_DIGITAL_FILTER_SELECT2 0x30
+#define AK4671_E1_COEFFICIENT0 0x32
+#define AK4671_E1_COEFFICIENT1 0x33
+#define AK4671_E1_COEFFICIENT2 0x34
+#define AK4671_E1_COEFFICIENT3 0x35
+#define AK4671_E1_COEFFICIENT4 0x36
+#define AK4671_E1_COEFFICIENT5 0x37
+#define AK4671_E2_COEFFICIENT0 0x38
+#define AK4671_E2_COEFFICIENT1 0x39
+#define AK4671_E2_COEFFICIENT2 0x3a
+#define AK4671_E2_COEFFICIENT3 0x3b
+#define AK4671_E2_COEFFICIENT4 0x3c
+#define AK4671_E2_COEFFICIENT5 0x3d
+#define AK4671_E3_COEFFICIENT0 0x3e
+#define AK4671_E3_COEFFICIENT1 0x3f
+#define AK4671_E3_COEFFICIENT2 0x40
+#define AK4671_E3_COEFFICIENT3 0x41
+#define AK4671_E3_COEFFICIENT4 0x42
+#define AK4671_E3_COEFFICIENT5 0x43
+#define AK4671_E4_COEFFICIENT0 0x44
+#define AK4671_E4_COEFFICIENT1 0x45
+#define AK4671_E4_COEFFICIENT2 0x46
+#define AK4671_E4_COEFFICIENT3 0x47
+#define AK4671_E4_COEFFICIENT4 0x48
+#define AK4671_E4_COEFFICIENT5 0x49
+#define AK4671_E5_COEFFICIENT0 0x4a
+#define AK4671_E5_COEFFICIENT1 0x4b
+#define AK4671_E5_COEFFICIENT2 0x4c
+#define AK4671_E5_COEFFICIENT3 0x4d
+#define AK4671_E5_COEFFICIENT4 0x4e
+#define AK4671_E5_COEFFICIENT5 0x4f
+#define AK4671_EQ_CONTROL_250HZ_100HZ 0x50
+#define AK4671_EQ_CONTROL_3500HZ_1KHZ 0x51
+#define AK4671_EQ_CONTRO_10KHZ 0x52
+#define AK4671_PCM_IF_CONTROL0 0x53
+#define AK4671_PCM_IF_CONTROL1 0x54
+#define AK4671_PCM_IF_CONTROL2 0x55
+#define AK4671_DIGITAL_VOLUME_B_CONTROL 0x56
+#define AK4671_DIGITAL_VOLUME_C_CONTROL 0x57
+#define AK4671_SIDETONE_VOLUME_CONTROL 0x58
+#define AK4671_DIGITAL_MIXING_CONTROL2 0x59
+#define AK4671_SAR_ADC_CONTROL 0x5a
+
+#define AK4671_CACHEREGNUM (AK4671_SAR_ADC_CONTROL + 1)
+
+/* Bitfield Definitions */
+
+/* AK4671_AD_DA_POWER_MANAGEMENT (0x00) Fields */
+#define AK4671_PMVCM 0x01
+
+/* AK4671_PLL_MODE_SELECT0 (0x01) Fields */
+#define AK4671_PLL 0x0f
+#define AK4671_PLL_11_2896MHZ (4 << 0)
+#define AK4671_PLL_12_288MHZ (5 << 0)
+#define AK4671_PLL_12MHZ (6 << 0)
+#define AK4671_PLL_24MHZ (7 << 0)
+#define AK4671_PLL_19_2MHZ (8 << 0)
+#define AK4671_PLL_13_5MHZ (12 << 0)
+#define AK4671_PLL_27MHZ (13 << 0)
+#define AK4671_PLL_13MHZ (14 << 0)
+#define AK4671_PLL_26MHZ (15 << 0)
+#define AK4671_FS 0xf0
+#define AK4671_FS_8KHZ (0 << 4)
+#define AK4671_FS_12KHZ (1 << 4)
+#define AK4671_FS_16KHZ (2 << 4)
+#define AK4671_FS_24KHZ (3 << 4)
+#define AK4671_FS_11_025KHZ (5 << 4)
+#define AK4671_FS_22_05KHZ (7 << 4)
+#define AK4671_FS_32KHZ (10 << 4)
+#define AK4671_FS_48KHZ (11 << 4)
+#define AK4671_FS_44_1KHZ (15 << 4)
+
+/* AK4671_PLL_MODE_SELECT1 (0x02) Fields */
+#define AK4671_PMPLL 0x01
+#define AK4671_M_S 0x02
+
+/* AK4671_FORMAT_SELECT (0x03) Fields */
+#define AK4671_DIF 0x03
+#define AK4671_DIF_DSP_MODE (0 << 0)
+#define AK4671_DIF_MSB_MODE (2 << 0)
+#define AK4671_DIF_I2S_MODE (3 << 0)
+#define AK4671_BCKP 0x04
+#define AK4671_MSBS 0x08
+#define AK4671_SDOD 0x10
+
+/* AK4671_LOUT2_POWER_MANAGEMENT (0x10) Fields */
+#define AK4671_MUTEN 0x04
+
+extern struct snd_soc_dai ak4671_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ak4671;
+
+#endif
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 593d5b9..72abc5a 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1101,7 +1101,7 @@
}
static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in,
+ int pll_id, int source, unsigned int freq_in,
unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index b9ef4d9..9cb8e50 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1011,7 +1011,8 @@
}
static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
- unsigned int freq_in, unsigned int freq_out)
+ int source, unsigned int freq_in,
+ unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct wm8400_priv *wm8400 = codec->private_data;
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 060d5d0..5702435 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -271,8 +271,8 @@
pll_div.k = K;
}
-static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 reg;
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 6bded8c..3be5c0b 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -407,8 +407,8 @@
return 0;
}
-static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
int offset;
struct snd_soc_codec *codec = codec_dai->codec;
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
new file mode 100644
index 0000000..90ec8c5
--- /dev/null
+++ b/sound/soc/codecs/wm8711.c
@@ -0,0 +1,659 @@
+/*
+ * wm8711.c -- WM8711 ALSA SoC Audio driver
+ *
+ * Copyright 2006 Wolfson Microelectronics
+ *
+ * Author: Mike Arthur <linux@wolfsonmicro.com>
+ *
+ * Based on wm8731.c by Richard Purdie
+ *
+ * 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/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.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/tlv.h>
+#include <sound/initval.h>
+
+#include "wm8711.h"
+
+static struct snd_soc_codec *wm8711_codec;
+
+/* codec private data */
+struct wm8711_priv {
+ struct snd_soc_codec codec;
+ u16 reg_cache[WM8711_CACHEREGNUM];
+ unsigned int sysclk;
+};
+
+/*
+ * wm8711 register cache
+ * We can't read the WM8711 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ * There is no point in caching the reset register
+ */
+static const u16 wm8711_reg[WM8711_CACHEREGNUM] = {
+ 0x0079, 0x0079, 0x000a, 0x0008,
+ 0x009f, 0x000a, 0x0000, 0x0000
+};
+
+#define wm8711_reset(c) snd_soc_write(c, WM8711_RESET, 0)
+
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+
+static const struct snd_kcontrol_new wm8711_snd_controls[] = {
+
+SOC_DOUBLE_R_TLV("Master Playback Volume", WM8711_LOUT1V, WM8711_ROUT1V,
+ 0, 127, 0, out_tlv),
+SOC_DOUBLE_R("Master Playback ZC Switch", WM8711_LOUT1V, WM8711_ROUT1V,
+ 7, 1, 0),
+
+};
+
+/* Output Mixer */
+static const struct snd_kcontrol_new wm8711_output_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8711_APANA, 3, 1, 0),
+SOC_DAPM_SINGLE("HiFi Playback Switch", WM8711_APANA, 4, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8711_dapm_widgets[] = {
+SND_SOC_DAPM_MIXER("Output Mixer", WM8711_PWR, 4, 1,
+ &wm8711_output_mixer_controls[0],
+ ARRAY_SIZE(wm8711_output_mixer_controls)),
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8711_PWR, 3, 1),
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("LHPOUT"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_OUTPUT("RHPOUT"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ /* output mixer */
+ {"Output Mixer", "Line Bypass Switch", "Line Input"},
+ {"Output Mixer", "HiFi Playback Switch", "DAC"},
+
+ /* outputs */
+ {"RHPOUT", NULL, "Output Mixer"},
+ {"ROUT", NULL, "Output Mixer"},
+ {"LHPOUT", NULL, "Output Mixer"},
+ {"LOUT", NULL, "Output Mixer"},
+};
+
+static int wm8711_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8711_dapm_widgets,
+ ARRAY_SIZE(wm8711_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+struct _coeff_div {
+ u32 mclk;
+ u32 rate;
+ u16 fs;
+ u8 sr:4;
+ u8 bosr:1;
+ u8 usb:1;
+};
+
+/* codec mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+ /* 48k */
+ {12288000, 48000, 256, 0x0, 0x0, 0x0},
+ {18432000, 48000, 384, 0x0, 0x1, 0x0},
+ {12000000, 48000, 250, 0x0, 0x0, 0x1},
+
+ /* 32k */
+ {12288000, 32000, 384, 0x6, 0x0, 0x0},
+ {18432000, 32000, 576, 0x6, 0x1, 0x0},
+ {12000000, 32000, 375, 0x6, 0x0, 0x1},
+
+ /* 8k */
+ {12288000, 8000, 1536, 0x3, 0x0, 0x0},
+ {18432000, 8000, 2304, 0x3, 0x1, 0x0},
+ {11289600, 8000, 1408, 0xb, 0x0, 0x0},
+ {16934400, 8000, 2112, 0xb, 0x1, 0x0},
+ {12000000, 8000, 1500, 0x3, 0x0, 0x1},
+
+ /* 96k */
+ {12288000, 96000, 128, 0x7, 0x0, 0x0},
+ {18432000, 96000, 192, 0x7, 0x1, 0x0},
+ {12000000, 96000, 125, 0x7, 0x0, 0x1},
+
+ /* 44.1k */
+ {11289600, 44100, 256, 0x8, 0x0, 0x0},
+ {16934400, 44100, 384, 0x8, 0x1, 0x0},
+ {12000000, 44100, 272, 0x8, 0x1, 0x1},
+
+ /* 88.2k */
+ {11289600, 88200, 128, 0xf, 0x0, 0x0},
+ {16934400, 88200, 192, 0xf, 0x1, 0x0},
+ {12000000, 88200, 136, 0xf, 0x1, 0x1},
+};
+
+static inline int get_coeff(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+ return i;
+ }
+ return 0;
+}
+
+static int wm8711_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8711_priv *wm8711 = codec->private_data;
+ u16 iface = snd_soc_read(codec, WM8711_IFACE) & 0xfffc;
+ int i = get_coeff(wm8711->sysclk, params_rate(params));
+ u16 srate = (coeff_div[i].sr << 2) |
+ (coeff_div[i].bosr << 1) | coeff_div[i].usb;
+
+ snd_soc_write(codec, WM8711_SRATE, srate);
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0008;
+ break;
+ }
+
+ snd_soc_write(codec, WM8711_IFACE, iface);
+ return 0;
+}
+
+static int wm8711_pcm_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ /* set active */
+ snd_soc_write(codec, WM8711_ACTIVE, 0x0001);
+
+ return 0;
+}
+
+static void wm8711_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ /* deactivate */
+ if (!codec->active) {
+ udelay(50);
+ snd_soc_write(codec, WM8711_ACTIVE, 0x0);
+ }
+}
+
+static int wm8711_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = snd_soc_read(codec, WM8711_APDIGI) & 0xfff7;
+
+ if (mute)
+ snd_soc_write(codec, WM8711_APDIGI, mute_reg | 0x8);
+ else
+ snd_soc_write(codec, WM8711_APDIGI, mute_reg);
+
+ return 0;
+}
+
+static int wm8711_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 wm8711_priv *wm8711 = codec->private_data;
+
+ switch (freq) {
+ case 11289600:
+ case 12000000:
+ case 12288000:
+ case 16934400:
+ case 18432000:
+ wm8711->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
+static int wm8711_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface |= 0x0040;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x0013;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set iface */
+ snd_soc_write(codec, WM8711_IFACE, iface);
+ return 0;
+}
+
+
+static int wm8711_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg = snd_soc_read(codec, WM8711_PWR) & 0xff7f;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ snd_soc_write(codec, WM8711_PWR, reg);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ snd_soc_write(codec, WM8711_PWR, reg | 0x0040);
+ break;
+ case SND_SOC_BIAS_OFF:
+ snd_soc_write(codec, WM8711_ACTIVE, 0x0);
+ snd_soc_write(codec, WM8711_PWR, 0xffff);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8711_RATES SNDRV_PCM_RATE_8000_96000
+
+#define WM8711_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8711_ops = {
+ .prepare = wm8711_pcm_prepare,
+ .hw_params = wm8711_hw_params,
+ .shutdown = wm8711_shutdown,
+ .digital_mute = wm8711_mute,
+ .set_sysclk = wm8711_set_dai_sysclk,
+ .set_fmt = wm8711_set_dai_fmt,
+};
+
+struct snd_soc_dai wm8711_dai = {
+ .name = "WM8711",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8711_RATES,
+ .formats = WM8711_FORMATS,
+ },
+ .ops = &wm8711_ops,
+};
+EXPORT_SYMBOL_GPL(wm8711_dai);
+
+static int wm8711_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;
+
+ snd_soc_write(codec, WM8711_ACTIVE, 0x0);
+ wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8711_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8711_reg); i++) {
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+ wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ wm8711_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+static int wm8711_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (wm8711_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = wm8711_codec;
+ codec = wm8711_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ snd_soc_add_controls(codec, wm8711_snd_controls,
+ ARRAY_SIZE(wm8711_snd_controls));
+ wm8711_add_widgets(codec);
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+/* power down chip */
+static int wm8711_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_wm8711 = {
+ .probe = wm8711_probe,
+ .remove = wm8711_remove,
+ .suspend = wm8711_suspend,
+ .resume = wm8711_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8711);
+
+static int wm8711_register(struct wm8711_priv *wm8711,
+ enum snd_soc_control_type control)
+{
+ int ret;
+ struct snd_soc_codec *codec = &wm8711->codec;
+ u16 reg;
+
+ if (wm8711_codec) {
+ dev_err(codec->dev, "Another WM8711 is registered\n");
+ return -EINVAL;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = wm8711;
+ codec->name = "WM8711";
+ codec->owner = THIS_MODULE;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8711_set_bias_level;
+ codec->dai = &wm8711_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = WM8711_CACHEREGNUM;
+ codec->reg_cache = &wm8711->reg_cache;
+
+ memcpy(codec->reg_cache, wm8711_reg, sizeof(wm8711_reg));
+
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
+ ret = wm8711_reset(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset\n");
+ goto err;
+ }
+
+ wm8711_dai.dev = codec->dev;
+
+ wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* Latch the update bits */
+ reg = snd_soc_read(codec, WM8711_LOUT1V);
+ snd_soc_write(codec, WM8711_LOUT1V, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8711_ROUT1V);
+ snd_soc_write(codec, WM8711_ROUT1V, reg | 0x0100);
+
+ wm8711_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ goto err;
+ }
+
+ ret = snd_soc_register_dai(&wm8711_dai);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ goto err_codec;
+ }
+
+ return 0;
+
+err_codec:
+ snd_soc_unregister_codec(codec);
+err:
+ kfree(wm8711);
+ return ret;
+}
+
+static void wm8711_unregister(struct wm8711_priv *wm8711)
+{
+ wm8711_set_bias_level(&wm8711->codec, SND_SOC_BIAS_OFF);
+ snd_soc_unregister_dai(&wm8711_dai);
+ snd_soc_unregister_codec(&wm8711->codec);
+ kfree(wm8711);
+ wm8711_codec = NULL;
+}
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8711_spi_probe(struct spi_device *spi)
+{
+ struct snd_soc_codec *codec;
+ struct wm8711_priv *wm8711;
+
+ wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL);
+ if (wm8711 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8711->codec;
+ codec->control_data = spi;
+ codec->dev = &spi->dev;
+
+ dev_set_drvdata(&spi->dev, wm8711);
+
+ return wm8711_register(wm8711, SND_SOC_SPI);
+}
+
+static int __devexit wm8711_spi_remove(struct spi_device *spi)
+{
+ struct wm8711_priv *wm8711 = dev_get_drvdata(&spi->dev);
+
+ wm8711_unregister(wm8711);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int wm8711_spi_suspend(struct spi_device *spi, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&spi->dev);
+}
+
+static int wm8711_spi_resume(struct spi_device *spi)
+{
+ return snd_soc_resume_device(&spi->dev);
+}
+#else
+#define wm8711_spi_suspend NULL
+#define wm8711_spi_resume NULL
+#endif
+
+static struct spi_driver wm8711_spi_driver = {
+ .driver = {
+ .name = "wm8711",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8711_spi_probe,
+ .suspend = wm8711_spi_suspend,
+ .resume = wm8711_spi_resume,
+ .remove = __devexit_p(wm8711_spi_remove),
+};
+#endif /* CONFIG_SPI_MASTER */
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static __devinit int wm8711_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct wm8711_priv *wm8711;
+ struct snd_soc_codec *codec;
+
+ wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL);
+ if (wm8711 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8711->codec;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+
+ i2c_set_clientdata(i2c, wm8711);
+ codec->control_data = i2c;
+
+ codec->dev = &i2c->dev;
+
+ return wm8711_register(wm8711, SND_SOC_I2C);
+}
+
+static __devexit int wm8711_i2c_remove(struct i2c_client *client)
+{
+ struct wm8711_priv *wm8711 = i2c_get_clientdata(client);
+ wm8711_unregister(wm8711);
+ return 0;
+}
+
+static const struct i2c_device_id wm8711_i2c_id[] = {
+ { "wm8711", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id);
+
+static struct i2c_driver wm8711_i2c_driver = {
+ .driver = {
+ .name = "WM8711 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8711_i2c_probe,
+ .remove = __devexit_p(wm8711_i2c_remove),
+ .id_table = wm8711_i2c_id,
+};
+#endif
+
+static int __init wm8711_modinit(void)
+{
+ int ret;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ ret = i2c_add_driver(&wm8711_i2c_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "Failed to register WM8711 I2C driver: %d\n",
+ ret);
+ }
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ ret = spi_register_driver(&wm8711_spi_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "Failed to register WM8711 SPI driver: %d\n",
+ ret);
+ }
+#endif
+ return 0;
+}
+module_init(wm8711_modinit);
+
+static void __exit wm8711_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&wm8711_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8711_spi_driver);
+#endif
+}
+module_exit(wm8711_exit);
+
+MODULE_DESCRIPTION("ASoC WM8711 driver");
+MODULE_AUTHOR("Mike Arthur");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8711.h b/sound/soc/codecs/wm8711.h
new file mode 100644
index 0000000..381e84a
--- /dev/null
+++ b/sound/soc/codecs/wm8711.h
@@ -0,0 +1,42 @@
+/*
+ * wm8711.h -- WM8711 Soc Audio driver
+ *
+ * Copyright 2006 Wolfson Microelectronics
+ *
+ * Author: Mike Arthur <linux@wolfsonmicro.com>
+ *
+ * Based on wm8731.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8711_H
+#define _WM8711_H
+
+/* WM8711 register space */
+
+#define WM8711_LOUT1V 0x02
+#define WM8711_ROUT1V 0x03
+#define WM8711_APANA 0x04
+#define WM8711_APDIGI 0x05
+#define WM8711_PWR 0x06
+#define WM8711_IFACE 0x07
+#define WM8711_SRATE 0x08
+#define WM8711_ACTIVE 0x09
+#define WM8711_RESET 0x0f
+
+#define WM8711_CACHEREGNUM 8
+
+#define WM8711_SYSCLK 0
+#define WM8711_DAI 0
+
+struct wm8711_setup_data {
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai wm8711_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8711;
+
+#endif
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 5ad677c..9b27efb 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -724,8 +724,8 @@
pll_div->k = K;
}
-static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
u16 reg, enable;
int offset;
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 5e9c855..882604e 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -814,8 +814,8 @@
return 0;
}
-static int wm8900_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8900_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
return wm8900_set_fll(codec_dai->codec, pll_id, freq_in, freq_out);
}
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 1ef2454..1685cfb 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -536,8 +536,8 @@
}
/* Untested at the moment */
-static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 reg;
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index f59703b..416fb3c 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -540,8 +540,8 @@
return 0;
}
-static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 reg;
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 98d663a..eff2933 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -281,36 +281,38 @@
}
struct pll_ {
- unsigned int pre_div:4; /* prescale - 1 */
+ unsigned int pre_div:1;
unsigned int n:4;
unsigned int k;
};
-static struct pll_ pll_div;
-
/* The size in bits of the pll divide multiplied by 10
* to allow rounding later */
#define FIXED_PLL_SIZE ((1 << 24) * 10)
-static void pll_factors(unsigned int target, unsigned int source)
+static void pll_factors(struct pll_ *pll_div,
+ unsigned int target, unsigned int source)
{
unsigned long long Kpart;
unsigned int K, Ndiv, Nmod;
+ /* There is a fixed divide by 4 in the output path */
+ target *= 4;
+
Ndiv = target / source;
if (Ndiv < 6) {
- source >>= 1;
- pll_div.pre_div = 1;
+ source /= 2;
+ pll_div->pre_div = 1;
Ndiv = target / source;
} else
- pll_div.pre_div = 0;
+ pll_div->pre_div = 0;
if ((Ndiv < 6) || (Ndiv > 12))
printk(KERN_WARNING
"WM8974 N value %u outwith recommended range!\n",
Ndiv);
- pll_div.n = Ndiv;
+ pll_div->n = Ndiv;
Nmod = target % source;
Kpart = FIXED_PLL_SIZE * (long long)Nmod;
@@ -325,13 +327,14 @@
/* Move down to proper range now rounding is done */
K /= 10;
- pll_div.k = K;
+ pll_div->k = K;
}
-static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
+ struct pll_ pll_div;
u16 reg;
if (freq_in == 0 || freq_out == 0) {
@@ -345,7 +348,7 @@
return 0;
}
- pll_factors(freq_out*4, freq_in);
+ pll_factors(&pll_div, freq_out, freq_in);
snd_soc_write(codec, WM8974_PLLN, (pll_div.pre_div << 4) | pll_div.n);
snd_soc_write(codec, WM8974_PLLK1, pll_div.k >> 18);
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 2d702db..f657e9a 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -972,8 +972,8 @@
pll_div->k = K;
}
-static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
u16 reg;
struct snd_soc_codec *codec = codec_dai->codec;
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index d998799..dac3977 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -422,7 +422,7 @@
return 0;
}
-static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id,
+static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source,
unsigned int Fref, unsigned int Fout)
{
struct snd_soc_codec *codec = dai->codec;
@@ -1572,33 +1572,15 @@
/* Use automatic clock configuration */
snd_soc_update_bits(codec, WM8993_CLOCKING_4, WM8993_SR_MODE, 0);
- if (!wm8993->pdata.lineout1_diff)
- snd_soc_update_bits(codec, WM8993_LINE_MIXER1,
- WM8993_LINEOUT1_MODE,
- WM8993_LINEOUT1_MODE);
- if (!wm8993->pdata.lineout2_diff)
- snd_soc_update_bits(codec, WM8993_LINE_MIXER2,
- WM8993_LINEOUT2_MODE,
- WM8993_LINEOUT2_MODE);
-
- if (wm8993->pdata.lineout1fb)
- snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
- WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB);
-
- if (wm8993->pdata.lineout2fb)
- snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
- WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB);
-
- /* Apply the microphone bias/detection configuration - the
- * platform data is directly applicable to the register. */
- snd_soc_update_bits(codec, WM8993_MICBIAS,
- WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK |
- WM8993_MICB1_LVL | WM8993_MICB2_LVL,
- wm8993->pdata.jd_scthr << WM8993_JD_SCTHR_SHIFT |
- wm8993->pdata.jd_thr << WM8993_JD_THR_SHIFT |
- wm8993->pdata.micbias1_lvl |
- wm8993->pdata.micbias1_lvl << 1);
-
+ wm_hubs_handle_analogue_pdata(codec, wm8993->pdata.lineout1_diff,
+ wm8993->pdata.lineout2_diff,
+ wm8993->pdata.lineout1fb,
+ wm8993->pdata.lineout2fb,
+ wm8993->pdata.jd_scthr,
+ wm8993->pdata.jd_thr,
+ wm8993->pdata.micbias1_lvl,
+ wm8993->pdata.micbias2_lvl);
+
ret = wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
if (ret != 0)
goto err;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index abed37a..ca3d449 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -800,8 +800,8 @@
return 0;
}
-static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
return wm9713_set_pll(codec, pll_id, freq_in, freq_out);
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index e542027..810a563 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -738,6 +738,41 @@
}
EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_routes);
+int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec,
+ int lineout1_diff, int lineout2_diff,
+ int lineout1fb, int lineout2fb,
+ int jd_scthr, int jd_thr, int micbias1_lvl,
+ int micbias2_lvl)
+{
+ if (!lineout1_diff)
+ snd_soc_update_bits(codec, WM8993_LINE_MIXER1,
+ WM8993_LINEOUT1_MODE,
+ WM8993_LINEOUT1_MODE);
+ if (!lineout2_diff)
+ snd_soc_update_bits(codec, WM8993_LINE_MIXER2,
+ WM8993_LINEOUT2_MODE,
+ WM8993_LINEOUT2_MODE);
+
+ if (lineout1fb)
+ snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
+ WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB);
+
+ if (lineout2fb)
+ snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
+ WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB);
+
+ snd_soc_update_bits(codec, WM8993_MICBIAS,
+ WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK |
+ WM8993_MICB1_LVL | WM8993_MICB2_LVL,
+ jd_scthr << WM8993_JD_SCTHR_SHIFT |
+ jd_thr << WM8993_JD_THR_SHIFT |
+ micbias1_lvl |
+ micbias2_lvl << WM8993_MICB2_LVL_SHIFT);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(wm_hubs_handle_analogue_pdata);
+
MODULE_DESCRIPTION("Shared support for Wolfson hubs products");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index ec09cb6..36d3fba 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -20,5 +20,10 @@
extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *);
extern int wm_hubs_add_analogue_routes(struct snd_soc_codec *, int, int);
+extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *,
+ int lineout1_diff, int lineout2_diff,
+ int lineout1fb, int lineout2fb,
+ int jd_scthr, int jd_thr,
+ int micbias1_lvl, int micbias2_lvl);
#endif
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 4dfd4ad..047ee39 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -13,9 +13,9 @@
tristate
config SND_DAVINCI_SOC_EVM
- tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM"
+ tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM"
depends on SND_DAVINCI_SOC
- depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM
+ depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM
select SND_DAVINCI_SOC_I2S
select SND_SOC_TLV320AIC3X
help
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 67414f6..7ccbe66 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -45,7 +45,8 @@
unsigned sysclk;
/* ASP1 on DM355 EVM is clocked by an external oscillator */
- if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm())
+ if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() ||
+ machine_is_davinci_dm365_evm())
sysclk = 27000000;
/* ASP0 in DM6446 EVM is clocked by U55, as configured by
@@ -176,7 +177,7 @@
.ops = &evm_ops,
};
-/* davinci-evm audio machine driver */
+/* davinci dm6446, dm355 or dm365 evm audio machine driver */
static struct snd_soc_card snd_soc_card_evm = {
.name = "DaVinci EVM",
.platform = &davinci_soc_platform,
@@ -243,7 +244,7 @@
int index;
int ret;
- if (machine_is_davinci_evm()) {
+ if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) {
evm_snd_dev_data = &evm_snd_devdata;
index = 0;
} else if (machine_is_davinci_dm355_evm()) {
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 4ae7070..2ab8093 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -397,6 +397,8 @@
}
dma_params->acnt = dma_params->data_type;
+ dma_params->fifo_level = 0;
+
rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1);
xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 5d1f98a..50ad051 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -714,16 +714,13 @@
struct davinci_pcm_dma_params *dma_params =
&dev->dma_params[substream->stream];
int word_length;
- u8 numevt;
+ u8 fifo_level;
davinci_hw_common_param(dev, substream->stream);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- numevt = dev->txnumevt;
+ fifo_level = dev->txnumevt;
else
- numevt = dev->rxnumevt;
-
- if (!numevt)
- numevt = 1;
+ fifo_level = dev->rxnumevt;
if (dev->op_mode == DAVINCI_MCASP_DIT_MODE)
davinci_hw_dit_param(dev);
@@ -751,12 +748,12 @@
return -EINVAL;
}
- if (dev->version == MCASP_VERSION_2) {
- dma_params->data_type *= numevt;
- dma_params->acnt = 4 * numevt;
- } else
+ if (dev->version == MCASP_VERSION_2 && !fifo_level)
+ dma_params->acnt = 4;
+ else
dma_params->acnt = dma_params->data_type;
+ dma_params->fifo_level = fifo_level;
davinci_config_channel_size(dev, word_length);
return 0;
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index c73a915..fb10f1d 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -66,38 +66,53 @@
dma_addr_t dma_pos;
dma_addr_t src, dst;
unsigned short src_bidx, dst_bidx;
+ unsigned short src_cidx, dst_cidx;
unsigned int data_type;
unsigned short acnt;
unsigned int count;
+ unsigned int fifo_level;
period_size = snd_pcm_lib_period_bytes(substream);
dma_offset = prtd->period * period_size;
dma_pos = runtime->dma_addr + dma_offset;
+ fifo_level = prtd->params->fifo_level;
pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d "
"dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size);
data_type = prtd->params->data_type;
count = period_size / data_type;
+ if (fifo_level)
+ count /= fifo_level;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
src = dma_pos;
dst = prtd->params->dma_addr;
src_bidx = data_type;
dst_bidx = 0;
+ src_cidx = data_type * fifo_level;
+ dst_cidx = 0;
} else {
src = prtd->params->dma_addr;
dst = dma_pos;
src_bidx = 0;
dst_bidx = data_type;
+ src_cidx = 0;
+ dst_cidx = data_type * fifo_level;
}
acnt = prtd->params->acnt;
edma_set_src(lch, src, INCR, W8BIT);
edma_set_dest(lch, dst, INCR, W8BIT);
- edma_set_src_index(lch, src_bidx, 0);
- edma_set_dest_index(lch, dst_bidx, 0);
- edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC);
+
+ edma_set_src_index(lch, src_bidx, src_cidx);
+ edma_set_dest_index(lch, dst_bidx, dst_cidx);
+
+ if (!fifo_level)
+ edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC);
+ else
+ edma_set_transfer_params(lch, acnt, fifo_level, count,
+ fifo_level, ABSYNC);
prtd->period++;
if (unlikely(prtd->period >= runtime->periods))
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
index 8746606..c8b0d2b 100644
--- a/sound/soc/davinci/davinci-pcm.h
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -23,6 +23,7 @@
enum dma_event_q eventq_no; /* event queue number */
unsigned char data_type; /* xfer data type */
unsigned char convert_mono_stereo;
+ unsigned int fifo_level;
};
diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c
index e4dcb53..0267d2d 100644
--- a/sound/soc/imx/mx27vis_wm8974.c
+++ b/sound/soc/imx/mx27vis_wm8974.c
@@ -157,7 +157,7 @@
/* codec PLL input is 25 MHz */
- ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG,
+ ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, IGNORED_ARG,
25000000, pll_out);
if (ret < 0) {
printk(KERN_ERR "Error when setting PLL input\n");
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 9f7c61e..4c8d99a 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -213,7 +213,7 @@
return ret;
/* set SSP audio pll clock */
- ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps);
+ ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps);
if (ret < 0)
return ret;
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index d11a6d7..3bd7712 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -305,8 +305,8 @@
/*
* Configure the PLL frequency pxa27x and (afaik - pxa320 only)
*/
-static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
{
struct ssp_priv *priv = cpu_dai->private_data;
struct ssp_device *ssp = priv->dev.ssp;
@@ -760,13 +760,13 @@
.resume = pxa_ssp_resume,
.playback = {
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 8,
.rates = PXA_SSP_RATES,
.formats = PXA_SSP_FORMATS,
},
.capture = {
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 8,
.rates = PXA_SSP_RATES,
.formats = PXA_SSP_FORMATS,
},
@@ -780,13 +780,13 @@
.resume = pxa_ssp_resume,
.playback = {
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 8,
.rates = PXA_SSP_RATES,
.formats = PXA_SSP_FORMATS,
},
.capture = {
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 8,
.rates = PXA_SSP_RATES,
.formats = PXA_SSP_FORMATS,
},
@@ -801,13 +801,13 @@
.resume = pxa_ssp_resume,
.playback = {
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 8,
.rates = PXA_SSP_RATES,
.formats = PXA_SSP_FORMATS,
},
.capture = {
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 8,
.rates = PXA_SSP_RATES,
.formats = PXA_SSP_FORMATS,
},
@@ -822,13 +822,13 @@
.resume = pxa_ssp_resume,
.playback = {
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 8,
.rates = PXA_SSP_RATES,
.formats = PXA_SSP_FORMATS,
},
.capture = {
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 8,
.rates = PXA_SSP_RATES,
.formats = PXA_SSP_FORMATS,
},
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index 9a386b4..dd678ae 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -74,7 +74,8 @@
static int zylonite_wm9713_init(struct snd_soc_codec *codec)
{
if (clk_pout)
- snd_soc_dai_set_pll(&codec->dai[0], 0, clk_get_rate(pout), 0);
+ snd_soc_dai_set_pll(&codec->dai[0], 0, 0,
+ clk_get_rate(pout), 0);
snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
ARRAY_SIZE(zylonite_dapm_widgets));
@@ -128,7 +129,7 @@
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out);
+ ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out);
if (ret < 0)
return ret;
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 923428f..d7912f1 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -56,6 +56,15 @@
help
Sat Y if you want to add support for SoC audio on the Jive.
+config SND_S3C64XX_SOC_WM8580
+ tristate "SoC I2S Audio support for WM8580 on SMDK64XX"
+ depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410)
+ depends on BROKEN
+ select SND_SOC_WM8580
+ select SND_S3C64XX_SOC_I2S
+ help
+ Sat Y if you want to add support for SoC audio on the SMDK64XX.
+
config SND_S3C24XX_SOC_SMDK2443_WM9710
tristate "SoC AC97 Audio support for SMDK2443 - WM9710"
depends on SND_S3C24XX_SOC && MACH_SMDK2443
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 99f5a7d..7790406 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -23,6 +23,7 @@
snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o
snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o
snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o
+snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o
obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o
obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
@@ -33,4 +34,5 @@
obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o
obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o
obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o
+obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o
diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
index 0c52e36..26409a9 100644
--- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
@@ -119,7 +119,7 @@
return ret;
/* codec PLL input is PCLK/4 */
- ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1,
+ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0,
iis_clkrate / 4, pll_out);
if (ret < 0)
return ret;
@@ -133,7 +133,7 @@
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
/* disable the PLL */
- return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
+ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0);
}
/*
@@ -183,7 +183,7 @@
return ret;
/* configue and enable PLL for 12.288MHz output */
- ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2,
+ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0,
iis_clkrate / 4, 12288000);
if (ret < 0)
return ret;
@@ -197,7 +197,7 @@
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
/* disable the PLL */
- return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
+ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0);
}
static struct snd_soc_ops neo1973_gta02_voice_ops = {
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 906709e..c9b7948 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -137,7 +137,7 @@
return ret;
/* codec PLL input is PCLK/4 */
- ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1,
+ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0,
iis_clkrate / 4, pll_out);
if (ret < 0)
return ret;
@@ -153,7 +153,7 @@
pr_debug("Entered %s\n", __func__);
/* disable the PLL */
- return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
+ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0);
}
/*
@@ -203,7 +203,7 @@
return ret;
/* configue and enable PLL for 12.288MHz output */
- ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2,
+ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0,
iis_clkrate / 4, 12288000);
if (ret < 0)
return ret;
@@ -219,7 +219,7 @@
pr_debug("Entered %s\n", __func__);
/* disable the PLL */
- return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
+ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0);
}
static struct snd_soc_ops neo1973_voice_ops = {
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 9bc4aa3..11c45a3 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -312,12 +312,15 @@
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_RIGHT_J:
+ iismod |= S3C2412_IISMOD_LR_RLOW;
iismod |= S3C2412_IISMOD_SDF_MSB;
break;
case SND_SOC_DAIFMT_LEFT_J:
+ iismod |= S3C2412_IISMOD_LR_RLOW;
iismod |= S3C2412_IISMOD_SDF_LSB;
break;
case SND_SOC_DAIFMT_I2S:
+ iismod &= ~S3C2412_IISMOD_LR_RLOW;
iismod |= S3C2412_IISMOD_SDF_IIS;
break;
default:
@@ -467,6 +470,31 @@
switch (div_id) {
case S3C_I2SV2_DIV_BCLK:
+ if (div > 3) {
+ /* convert value to bit field */
+
+ switch (div) {
+ case 16:
+ div = S3C2412_IISMOD_BCLK_16FS;
+ break;
+
+ case 32:
+ div = S3C2412_IISMOD_BCLK_32FS;
+ break;
+
+ case 24:
+ div = S3C2412_IISMOD_BCLK_24FS;
+ break;
+
+ case 48:
+ div = S3C2412_IISMOD_BCLK_48FS;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ }
+
reg = readl(i2s->regs + S3C2412_IISMOD);
reg &= ~S3C2412_IISMOD_BCLK_MASK;
writel(reg | div, i2s->regs + S3C2412_IISMOD);
@@ -626,7 +654,7 @@
}
i2s->iis_pclk = clk_get(dev, "iis");
- if (i2s->iis_pclk == NULL) {
+ if (IS_ERR(i2s->iis_pclk)) {
dev_err(dev, "failed to get iis_clock\n");
iounmap(i2s->regs);
return -ENOENT;
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 3c06c40..43fb253 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -99,6 +99,19 @@
iismod |= S3C64XX_IISMOD_IMS_SYSMUX;
break;
+ case S3C64XX_CLKSRC_CDCLK:
+ switch (dir) {
+ case SND_SOC_CLOCK_IN:
+ iismod |= S3C64XX_IISMOD_CDCLKCON;
+ break;
+ case SND_SOC_CLOCK_OUT:
+ iismod &= ~S3C64XX_IISMOD_CDCLKCON;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+
default:
return -EINVAL;
}
@@ -111,8 +124,12 @@
struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai)
{
struct s3c_i2sv2_info *i2s = to_info(dai);
+ u32 iismod = readl(i2s->regs + S3C2412_IISMOD);
- return i2s->iis_cclk;
+ if (iismod & S3C64XX_IISMOD_IMS_SYSMUX)
+ return i2s->iis_cclk;
+ else
+ return i2s->iis_pclk;
}
EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock);
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h
index 02148ce..abe7253 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.h
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.h
@@ -25,6 +25,7 @@
#define S3C64XX_CLKSRC_PCLK (0)
#define S3C64XX_CLKSRC_MUX (1)
+#define S3C64XX_CLKSRC_CDCLK (2)
extern struct snd_soc_dai s3c64xx_i2s_dai[];
diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c
new file mode 100644
index 0000000..482aaf1
--- /dev/null
+++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c
@@ -0,0 +1,273 @@
+/*
+ * smdk64xx_wm8580.c
+ *
+ * Copyright (c) 2009 Samsung Electronics Co. Ltd
+ * Author: Jaswinder Singh <jassi.brar@samsung.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "../codecs/wm8580.h"
+#include "s3c24xx-pcm.h"
+#include "s3c64xx-i2s.h"
+
+#define S3C64XX_I2S_V4 2
+
+/* SMDK64XX has a 12MHZ crystal attached to WM8580 */
+#define SMDK64XX_WM8580_FREQ 12000000
+
+static int smdk64xx_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ unsigned int pll_out;
+ int bfs, rfs, ret;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_U8:
+ case SNDRV_PCM_FORMAT_S8:
+ bfs = 16;
+ break;
+ case SNDRV_PCM_FORMAT_U16_LE:
+ case SNDRV_PCM_FORMAT_S16_LE:
+ bfs = 32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* The Fvco for WM8580 PLLs must fall within [90,100]MHz.
+ * This criterion can't be met if we request PLL output
+ * as {8000x256, 64000x256, 11025x256}Hz.
+ * As a wayout, we rather change rfs to a minimum value that
+ * results in (params_rate(params) * rfs), and itself, acceptable
+ * to both - the CODEC and the CPU.
+ */
+ switch (params_rate(params)) {
+ case 16000:
+ case 22050:
+ case 32000:
+ case 44100:
+ case 48000:
+ case 88200:
+ case 96000:
+ rfs = 256;
+ break;
+ case 64000:
+ rfs = 384;
+ break;
+ case 8000:
+ case 11025:
+ rfs = 512;
+ break;
+ default:
+ return -EINVAL;
+ }
+ pll_out = params_rate(params) * rfs;
+
+ /* Set the Codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* Set the AP DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK,
+ 0, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* We use PCLK for basic ops in SoC-Slave mode */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK,
+ 0, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* Set WM8580 to drive MCLK from it's PLLA */
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK,
+ WM8580_CLKSRC_PLLA);
+ if (ret < 0)
+ return ret;
+
+ /* Explicitly set WM8580-DAC to source from MCLK */
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL,
+ WM8580_CLKSRC_MCLK);
+ if (ret < 0)
+ return ret;
+
+ /* Assuming the CODEC driver evaluates it's rfs too from this call */
+ ret = snd_soc_dai_set_pll(codec_dai, 0, WM8580_PLLA,
+ SMDK64XX_WM8580_FREQ, pll_out);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+/*
+ * SMDK64XX WM8580 DAI operations.
+ */
+static struct snd_soc_ops smdk64xx_ops = {
+ .hw_params = smdk64xx_hw_params,
+};
+
+/* SMDK64xx Playback widgets */
+static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = {
+ SND_SOC_DAPM_HP("Front-L/R", NULL),
+ SND_SOC_DAPM_HP("Center/Sub", NULL),
+ SND_SOC_DAPM_HP("Rear-L/R", NULL),
+};
+
+/* SMDK64xx Capture widgets */
+static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = {
+ SND_SOC_DAPM_MIC("MicIn", NULL),
+ SND_SOC_DAPM_LINE("LineIn", NULL),
+};
+
+/* SMDK-PAIFTX connections */
+static const struct snd_soc_dapm_route audio_map_tx[] = {
+ /* MicIn feeds AINL */
+ {"AINL", NULL, "MicIn"},
+
+ /* LineIn feeds AINL/R */
+ {"AINL", NULL, "LineIn"},
+ {"AINR", NULL, "LineIn"},
+};
+
+/* SMDK-PAIFRX connections */
+static const struct snd_soc_dapm_route audio_map_rx[] = {
+ /* Front Left/Right are fed VOUT1L/R */
+ {"Front-L/R", NULL, "VOUT1L"},
+ {"Front-L/R", NULL, "VOUT1R"},
+
+ /* Center/Sub are fed VOUT2L/R */
+ {"Center/Sub", NULL, "VOUT2L"},
+ {"Center/Sub", NULL, "VOUT2R"},
+
+ /* Rear Left/Right are fed VOUT3L/R */
+ {"Rear-L/R", NULL, "VOUT3L"},
+ {"Rear-L/R", NULL, "VOUT3R"},
+};
+
+static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec)
+{
+ /* Add smdk64xx specific Capture widgets */
+ snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt,
+ ARRAY_SIZE(wm8580_dapm_widgets_cpt));
+
+ /* Set up PAIFTX audio path */
+ snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx));
+
+ /* All enabled by default */
+ snd_soc_dapm_enable_pin(codec, "MicIn");
+ snd_soc_dapm_enable_pin(codec, "LineIn");
+
+ /* signal a DAPM event */
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec)
+{
+ /* Add smdk64xx specific Playback widgets */
+ snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk,
+ ARRAY_SIZE(wm8580_dapm_widgets_pbk));
+
+ /* Set up PAIFRX audio path */
+ snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx));
+
+ /* All enabled by default */
+ snd_soc_dapm_enable_pin(codec, "Front-L/R");
+ snd_soc_dapm_enable_pin(codec, "Center/Sub");
+ snd_soc_dapm_enable_pin(codec, "Rear-L/R");
+
+ /* signal a DAPM event */
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static struct snd_soc_dai_link smdk64xx_dai[] = {
+{ /* Primary Playback i/f */
+ .name = "WM8580 PAIF RX",
+ .stream_name = "Playback",
+ .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4],
+ .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX],
+ .init = smdk64xx_wm8580_init_paifrx,
+ .ops = &smdk64xx_ops,
+},
+{ /* Primary Capture i/f */
+ .name = "WM8580 PAIF TX",
+ .stream_name = "Capture",
+ .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4],
+ .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX],
+ .init = smdk64xx_wm8580_init_paiftx,
+ .ops = &smdk64xx_ops,
+},
+};
+
+static struct snd_soc_card smdk64xx = {
+ .name = "smdk64xx",
+ .platform = &s3c24xx_soc_platform,
+ .dai_link = smdk64xx_dai,
+ .num_links = ARRAY_SIZE(smdk64xx_dai),
+};
+
+static struct snd_soc_device smdk64xx_snd_devdata = {
+ .card = &smdk64xx,
+ .codec_dev = &soc_codec_dev_wm8580,
+};
+
+static struct platform_device *smdk64xx_snd_device;
+
+static int __init smdk64xx_audio_init(void)
+{
+ int ret;
+
+ smdk64xx_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!smdk64xx_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata);
+ smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev;
+ ret = platform_device_add(smdk64xx_snd_device);
+
+ if (ret)
+ platform_device_put(smdk64xx_snd_device);
+
+ return ret;
+}
+module_init(smdk64xx_audio_init);
+
+MODULE_AUTHOR("Jaswinder Singh, jassi.brar@samsung.com");
+MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index c8ceddc..d2505e8 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -77,6 +77,35 @@
#define snd_soc_7_9_spi_write NULL
#endif
+static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 *cache = codec->reg_cache;
+ u8 data[2];
+
+ BUG_ON(codec->volatile_register);
+
+ data[0] = reg & 0xff;
+ data[1] = value & 0xff;
+
+ if (reg < codec->reg_cache_size)
+ cache[reg] = value;
+
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 *cache = codec->reg_cache;
+ if (reg >= codec->reg_cache_size)
+ return -1;
+ return cache[reg];
+}
+
static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
@@ -150,9 +179,20 @@
unsigned int (*read)(struct snd_soc_codec *, unsigned int);
unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
} io_types[] = {
- { 7, 9, snd_soc_7_9_write, snd_soc_7_9_spi_write, snd_soc_7_9_read },
- { 8, 16, snd_soc_8_16_write, NULL, snd_soc_8_16_read,
- snd_soc_8_16_read_i2c },
+ {
+ .addr_bits = 7, .data_bits = 9,
+ .write = snd_soc_7_9_write, .read = snd_soc_7_9_read,
+ .spi_write = snd_soc_7_9_spi_write
+ },
+ {
+ .addr_bits = 8, .data_bits = 8,
+ .write = snd_soc_8_8_write, .read = snd_soc_8_8_read,
+ },
+ {
+ .addr_bits = 8, .data_bits = 16,
+ .write = snd_soc_8_16_write, .read = snd_soc_8_16_read,
+ .i2c_read = snd_soc_8_16_read_i2c,
+ },
};
/**
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 7ff04ad..1dec9d2 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1254,21 +1254,39 @@
static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
{
+ char codec_root[128];
+
+ if (codec->dev)
+ snprintf(codec_root, sizeof(codec_root),
+ "%s.%s", codec->name, dev_name(codec->dev));
+ else
+ snprintf(codec_root, sizeof(codec_root),
+ "%s", codec->name);
+
+ codec->debugfs_codec_root = debugfs_create_dir(codec_root,
+ debugfs_root);
+ if (!codec->debugfs_codec_root) {
+ printk(KERN_WARNING
+ "ASoC: Failed to create codec debugfs directory\n");
+ return;
+ }
+
codec->debugfs_reg = debugfs_create_file("codec_reg", 0644,
- debugfs_root, codec,
- &codec_reg_fops);
+ codec->debugfs_codec_root,
+ codec, &codec_reg_fops);
if (!codec->debugfs_reg)
printk(KERN_WARNING
"ASoC: Failed to create codec register debugfs file\n");
codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744,
- debugfs_root,
+ codec->debugfs_codec_root,
&codec->pop_time);
if (!codec->debugfs_pop_time)
printk(KERN_WARNING
"Failed to create pop time debugfs file\n");
- codec->debugfs_dapm = debugfs_create_dir("dapm", debugfs_root);
+ codec->debugfs_dapm = debugfs_create_dir("dapm",
+ codec->debugfs_codec_root);
if (!codec->debugfs_dapm)
printk(KERN_WARNING
"Failed to create DAPM debugfs directory\n");
@@ -1278,9 +1296,7 @@
static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
{
- debugfs_remove_recursive(codec->debugfs_dapm);
- debugfs_remove(codec->debugfs_pop_time);
- debugfs_remove(codec->debugfs_reg);
+ debugfs_remove_recursive(codec->debugfs_codec_root);
}
#else
@@ -2197,16 +2213,18 @@
* snd_soc_dai_set_pll - configure DAI PLL.
* @dai: DAI
* @pll_id: DAI specific PLL ID
+ * @source: DAI specific source for the PLL
* @freq_in: PLL input clock frequency in Hz
* @freq_out: requested PLL output clock frequency in Hz
*
* Configures and enables PLL to generate output clock based on input clock.
*/
-int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out)
+int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
+ unsigned int freq_in, unsigned int freq_out)
{
if (dai->ops && dai->ops->set_pll)
- return dai->ops->set_pll(dai, pll_id, freq_in, freq_out);
+ return dai->ops->set_pll(dai, pll_id, source,
+ freq_in, freq_out);
else
return -EINVAL;
}
@@ -2251,6 +2269,30 @@
EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
/**
+ * snd_soc_dai_set_channel_map - configure DAI audio channel map
+ * @dai: DAI
+ * @tx_num: how many TX channels
+ * @tx_slot: pointer to an array which imply the TX slot number channel
+ * 0~num-1 uses
+ * @rx_num: how many RX channels
+ * @rx_slot: pointer to an array which imply the RX slot number channel
+ * 0~num-1 uses
+ *
+ * configure the relationship between channel number and TDM slot number.
+ */
+int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot)
+{
+ if (dai->ops && dai->ops->set_channel_map)
+ return dai->ops->set_channel_map(dai, tx_num, tx_slot,
+ rx_num, rx_slot);
+ else
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map);
+
+/**
* snd_soc_dai_set_tristate - configure DAI system or master clock.
* @dai: DAI
* @tristate: tristate enable
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8de6f9d..d2af872 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -719,6 +719,10 @@
/* Check if one of our outputs is connected */
list_for_each_entry(path, &w->sinks, list_source) {
+ if (path->connected &&
+ !path->connected(path->source, path->sink))
+ continue;
+
if (path->sink && path->sink->power_check &&
path->sink->power_check(path->sink)) {
power = 1;
@@ -1138,6 +1142,9 @@
w->active ? "active" : "inactive");
list_for_each_entry(p, &w->sources, list_sink) {
+ if (p->connected && !p->connected(w, p->sink))
+ continue;
+
if (p->connect)
ret += snprintf(buf + ret, PAGE_SIZE - ret,
" in %s %s\n",
@@ -1145,6 +1152,9 @@
p->source->name);
}
list_for_each_entry(p, &w->sinks, list_source) {
+ if (p->connected && !p->connected(w, p->sink))
+ continue;
+
if (p->connect)
ret += snprintf(buf + ret, PAGE_SIZE - ret,
" out %s %s\n",
@@ -1192,8 +1202,8 @@
/* test and update the power status of a mux widget */
static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
- struct snd_kcontrol *kcontrol, int mask,
- int mux, int val, struct soc_enum *e)
+ struct snd_kcontrol *kcontrol, int change,
+ int mux, struct soc_enum *e)
{
struct snd_soc_dapm_path *path;
int found = 0;
@@ -1202,7 +1212,7 @@
widget->id != snd_soc_dapm_value_mux)
return -ENODEV;
- if (!snd_soc_test_bits(widget->codec, e->reg, mask, val))
+ if (!change)
return 0;
/* find dapm widget path assoc with kcontrol */
@@ -1387,10 +1397,13 @@
EXPORT_SYMBOL_GPL(snd_soc_dapm_sync);
static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
- const char *sink, const char *control, const char *source)
+ const struct snd_soc_dapm_route *route)
{
struct snd_soc_dapm_path *path;
struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w;
+ const char *sink = route->sink;
+ const char *control = route->control;
+ const char *source = route->source;
int ret = 0;
/* find src and dest widgets */
@@ -1414,6 +1427,7 @@
path->source = wsource;
path->sink = wsink;
+ path->connected = route->connected;
INIT_LIST_HEAD(&path->list);
INIT_LIST_HEAD(&path->list_source);
INIT_LIST_HEAD(&path->list_sink);
@@ -1514,8 +1528,7 @@
int i, ret;
for (i = 0; i < num; i++) {
- ret = snd_soc_dapm_add_route(codec, route->sink,
- route->control, route->source);
+ ret = snd_soc_dapm_add_route(codec, route);
if (ret < 0) {
printk(KERN_ERR "Failed to add route %s->%s\n",
route->source,
@@ -1752,7 +1765,7 @@
{
struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int val, mux;
+ unsigned int val, mux, change;
unsigned int mask, bitmask;
int ret = 0;
@@ -1772,20 +1785,21 @@
mutex_lock(&widget->codec->mutex);
widget->value = val;
- dapm_mux_update_power(widget, kcontrol, mask, mux, val, e);
- if (widget->event) {
- if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
- ret = widget->event(widget,
- kcontrol, SND_SOC_DAPM_PRE_REG);
- if (ret < 0)
- goto out;
- }
- ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
- if (widget->event_flags & SND_SOC_DAPM_POST_REG)
- ret = widget->event(widget,
- kcontrol, SND_SOC_DAPM_POST_REG);
- } else
- ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
+ change = snd_soc_test_bits(widget->codec, e->reg, mask, val);
+ dapm_mux_update_power(widget, kcontrol, change, mux, e);
+
+ if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
+ ret = widget->event(widget,
+ kcontrol, SND_SOC_DAPM_PRE_REG);
+ if (ret < 0)
+ goto out;
+ }
+
+ ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
+
+ if (widget->event_flags & SND_SOC_DAPM_POST_REG)
+ ret = widget->event(widget,
+ kcontrol, SND_SOC_DAPM_POST_REG);
out:
mutex_unlock(&widget->codec->mutex);
@@ -1794,6 +1808,54 @@
EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double);
/**
+ * snd_soc_dapm_get_enum_virt - Get virtual DAPM mux
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = widget->value;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt);
+
+/**
+ * snd_soc_dapm_put_enum_virt - Set virtual DAPM mux
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ struct soc_enum *e =
+ (struct soc_enum *)kcontrol->private_value;
+ int change;
+ int ret = 0;
+
+ if (ucontrol->value.enumerated.item[0] >= e->max)
+ return -EINVAL;
+
+ mutex_lock(&widget->codec->mutex);
+
+ change = widget->value != ucontrol->value.enumerated.item[0];
+ widget->value = ucontrol->value.enumerated.item[0];
+ dapm_mux_update_power(widget, kcontrol, change, widget->value, e);
+
+ mutex_unlock(&widget->codec->mutex);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt);
+
+/**
* snd_soc_dapm_get_value_enum_double - dapm semi enumerated double mixer get
* callback
* @kcontrol: mixer control
@@ -1851,7 +1913,7 @@
{
struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int val, mux;
+ unsigned int val, mux, change;
unsigned int mask;
int ret = 0;
@@ -1869,20 +1931,21 @@
mutex_lock(&widget->codec->mutex);
widget->value = val;
- dapm_mux_update_power(widget, kcontrol, mask, mux, val, e);
- if (widget->event) {
- if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
- ret = widget->event(widget,
- kcontrol, SND_SOC_DAPM_PRE_REG);
- if (ret < 0)
- goto out;
- }
- ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
- if (widget->event_flags & SND_SOC_DAPM_POST_REG)
- ret = widget->event(widget,
- kcontrol, SND_SOC_DAPM_POST_REG);
- } else
- ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
+ change = snd_soc_test_bits(widget->codec, e->reg, mask, val);
+ dapm_mux_update_power(widget, kcontrol, change, mux, e);
+
+ if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
+ ret = widget->event(widget,
+ kcontrol, SND_SOC_DAPM_PRE_REG);
+ if (ret < 0)
+ goto out;
+ }
+
+ ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
+
+ if (widget->event_flags & SND_SOC_DAPM_POST_REG)
+ ret = widget->event(widget,
+ kcontrol, SND_SOC_DAPM_POST_REG);
out:
mutex_unlock(&widget->codec->mutex);