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authormerge <null@invalid>2009-01-22 13:55:32 +0000
committerAndy Green <agreen@octopus.localdomain>2009-01-22 13:55:32 +0000
commitaa6f5ffbdba45aa8e19e5048648fc6c7b25376d3 (patch)
treefbb786d0ac6f8a774fd834e9ce951197e60fbffa /sound/soc/codecs
parentf2d78193eae5dccd3d588d2c8ea0866efc368332 (diff)
MERGE-via-pending-tracking-hist-MERGE-via-stable-tracking-MERGE-via-mokopatches-tracking-fix-stray-endmenu-patch-1232632040-1232632141
pending-tracking-hist top was MERGE-via-stable-tracking-MERGE-via-mokopatches-tracking-fix-stray-endmenu-patch-1232632040-1232632141 / fdf777a63bcb59e0dfd78bfe2c6242e01f6d4eb9 ... parent commitmessage: From: merge <null@invalid> MERGE-via-stable-tracking-hist-MERGE-via-mokopatches-tracking-fix-stray-endmenu-patch-1232632040 stable-tracking-hist top was MERGE-via-mokopatches-tracking-fix-stray-endmenu-patch-1232632040 / 90463bfd2d5a3c8b52f6e6d71024a00e052b0ced ... parent commitmessage: From: merge <null@invalid> MERGE-via-mokopatches-tracking-hist-fix-stray-endmenu-patch mokopatches-tracking-hist top was fix-stray-endmenu-patch / 3630e0be570de8057e7f8d2fe501ed353cdf34e6 ... parent commitmessage: From: Andy Green <andy@openmoko.com> fix-stray-endmenu.patch Signed-off-by: Andy Green <andy@openmoko.com>
Diffstat (limited to 'sound/soc/codecs')
-rw-r--r--sound/soc/codecs/Kconfig79
-rw-r--r--sound/soc/codecs/Makefile12
-rw-r--r--sound/soc/codecs/ac97.c7
-rw-r--r--sound/soc/codecs/ad1980.c24
-rw-r--r--sound/soc/codecs/ad73311.c18
-rw-r--r--sound/soc/codecs/ak4535.c19
-rw-r--r--sound/soc/codecs/cs4270.c38
-rw-r--r--sound/soc/codecs/l3.c91
-rw-r--r--sound/soc/codecs/pcm3008.c212
-rw-r--r--sound/soc/codecs/pcm3008.h25
-rw-r--r--sound/soc/codecs/ssm2602.c57
-rw-r--r--sound/soc/codecs/tlv320aic23.c262
-rw-r--r--sound/soc/codecs/tlv320aic26.c22
-rw-r--r--sound/soc/codecs/tlv320aic3x.c166
-rw-r--r--sound/soc/codecs/tlv320aic3x.h60
-rw-r--r--sound/soc/codecs/twl4030.c1312
-rw-r--r--sound/soc/codecs/twl4030.h226
-rw-r--r--sound/soc/codecs/uda134x.c668
-rw-r--r--sound/soc/codecs/uda134x.h36
-rw-r--r--sound/soc/codecs/uda1380.c29
-rw-r--r--sound/soc/codecs/wm8350.c1583
-rw-r--r--sound/soc/codecs/wm8350.h20
-rw-r--r--sound/soc/codecs/wm8510.c19
-rw-r--r--sound/soc/codecs/wm8580.c134
-rw-r--r--sound/soc/codecs/wm8580.h1
-rw-r--r--sound/soc/codecs/wm8728.c585
-rw-r--r--sound/soc/codecs/wm8728.h30
-rw-r--r--sound/soc/codecs/wm8731.c25
-rw-r--r--sound/soc/codecs/wm8750.c19
-rw-r--r--sound/soc/codecs/wm8753.c39
-rw-r--r--sound/soc/codecs/wm8900.c262
-rw-r--r--sound/soc/codecs/wm8900.h6
-rw-r--r--sound/soc/codecs/wm8903.c268
-rw-r--r--sound/soc/codecs/wm8903.h5
-rw-r--r--sound/soc/codecs/wm8971.c19
-rw-r--r--sound/soc/codecs/wm8990.c43
-rw-r--r--sound/soc/codecs/wm8990.h4
-rw-r--r--sound/soc/codecs/wm9712.c12
-rw-r--r--sound/soc/codecs/wm9713.c46
39 files changed, 5866 insertions, 617 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 38a0e3b620a..d0e0d691ae5 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1,31 +1,50 @@
+# Helper to resolve issues with configs that have SPI enabled but I2C
+# modular, meaning we can't build the codec driver in with I2C support.
+# We use an ordered list of conditional defaults to pick the appropriate
+# setting - SPI can't be modular so that case doesn't need to be covered.
+config SND_SOC_I2C_AND_SPI
+ tristate
+ default m if I2C=m
+ default y if I2C=y
+ default y if SPI_MASTER=y
+
config SND_SOC_ALL_CODECS
tristate "Build all ASoC CODEC drivers"
- depends on I2C
- select SPI
- select SPI_MASTER
- select SND_SOC_AD73311
- select SND_SOC_AK4535
- select SND_SOC_CS4270
- select SND_SOC_SSM2602
- select SND_SOC_TLV320AIC23
- select SND_SOC_TLV320AIC26
- select SND_SOC_TLV320AIC3X
- select SND_SOC_UDA1380
- select SND_SOC_WM8510
- select SND_SOC_WM8580
- select SND_SOC_WM8731
- select SND_SOC_WM8750
- select SND_SOC_WM8753
- select SND_SOC_WM8900
- select SND_SOC_WM8903
- select SND_SOC_WM8971
- select SND_SOC_WM8990
+ select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
+ select SND_SOC_AD1980 if SND_SOC_AC97_BUS
+ select SND_SOC_AD73311 if I2C
+ select SND_SOC_AK4535 if I2C
+ select SND_SOC_CS4270 if I2C
+ select SND_SOC_PCM3008
+ select SND_SOC_SSM2602 if I2C
+ select SND_SOC_TLV320AIC23 if I2C
+ select SND_SOC_TLV320AIC26 if SPI_MASTER
+ select SND_SOC_TLV320AIC3X if I2C
+ select SND_SOC_TWL4030 if TWL4030_CORE
+ select SND_SOC_UDA134X
+ select SND_SOC_UDA1380 if I2C
+ select SND_SOC_WM8350 if MFD_WM8350
+ select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_WM8580 if I2C
+ select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_WM8900 if I2C
+ select SND_SOC_WM8903 if I2C
+ select SND_SOC_WM8971 if I2C
+ select SND_SOC_WM8990 if I2C
+ select SND_SOC_WM9712 if SND_SOC_AC97_BUS
+ select SND_SOC_WM9713 if SND_SOC_AC97_BUS
help
Normally ASoC codec drivers are only built if a machine driver which
uses them is also built since they are only usable with a machine
driver. Selecting this option will allow these drivers to be built
without an explicit machine driver for test and development purposes.
+ Support for the bus types used to access the codecs to be built must
+ be selected separately.
+
If unsure select "N".
@@ -60,6 +79,12 @@ config SND_SOC_CS4270_VD33_ERRATA
bool
depends on SND_SOC_CS4270
+config SND_SOC_L3
+ tristate
+
+config SND_SOC_PCM3008
+ tristate
+
config SND_SOC_SSM2602
tristate
@@ -75,15 +100,29 @@ config SND_SOC_TLV320AIC3X
tristate
depends on I2C
+config SND_SOC_TWL4030
+ tristate
+ depends on TWL4030_CORE
+
+config SND_SOC_UDA134X
+ tristate
+ select SND_SOC_L3
+
config SND_SOC_UDA1380
tristate
+config SND_SOC_WM8350
+ tristate
+
config SND_SOC_WM8510
tristate
config SND_SOC_WM8580
tristate
+config SND_SOC_WM8728
+ tristate
+
config SND_SOC_WM8731
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 90f0a585fc7..c4ddc9aa2bb 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -3,13 +3,19 @@ snd-soc-ad1980-objs := ad1980.o
snd-soc-ad73311-objs := ad73311.o
snd-soc-ak4535-objs := ak4535.o
snd-soc-cs4270-objs := cs4270.o
+snd-soc-l3-objs := l3.o
+snd-soc-pcm3008-objs := pcm3008.o
snd-soc-ssm2602-objs := ssm2602.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
+snd-soc-twl4030-objs := twl4030.o
+snd-soc-uda134x-objs := uda134x.o
snd-soc-uda1380-objs := uda1380.o
+snd-soc-wm8350-objs := wm8350.o
snd-soc-wm8510-objs := wm8510.o
snd-soc-wm8580-objs := wm8580.o
+snd-soc-wm8728-objs := wm8728.o
snd-soc-wm8731-objs := wm8731.o
snd-soc-wm8750-objs := wm8750.o
snd-soc-wm8753-objs := wm8753.o
@@ -25,13 +31,19 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
+obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
+obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
+obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o
+obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o
obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
+obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o
obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o
obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o
+obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o
obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o
obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o
obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index bd1ebdc6c86..fb53e6511af 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -24,7 +24,8 @@
#define AC97_VERSION "0.6"
-static int ac97_prepare(struct snd_pcm_substream *substream)
+static int ac97_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -42,7 +43,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream)
struct snd_soc_dai ac97_dai = {
.name = "AC97 HiFi",
- .type = SND_SOC_DAI_AC97,
+ .ac97_control = 1,
.playback = {
.stream_name = "AC97 Playback",
.channels_min = 1,
@@ -113,7 +114,7 @@ static int ac97_soc_probe(struct platform_device *pdev)
if (ret < 0)
goto bus_err;
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0)
goto bus_err;
return 0;
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 1397b8e06c0..73fdbb4d4a3 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -85,6 +85,9 @@ SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0),
SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1),
SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1),
+SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1),
+SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1),
+
SOC_ENUM("Capture Source", ad1980_cap_src),
SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
@@ -142,10 +145,11 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
struct snd_soc_dai ad1980_dai = {
.name = "AC97",
+ .ac97_control = 1,
.playback = {
.stream_name = "Playback",
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 6,
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE, },
.capture = {
@@ -192,6 +196,7 @@ static int ad1980_soc_probe(struct platform_device *pdev)
struct snd_soc_codec *codec;
int ret = 0;
u16 vendor_id2;
+ u16 ext_status;
printk(KERN_INFO "AD1980 SoC Audio Codec\n");
@@ -234,7 +239,7 @@ static int ad1980_soc_probe(struct platform_device *pdev)
ret = ad1980_reset(codec, 0);
if (ret < 0) {
- printk(KERN_ERR "AC97 link error\n");
+ printk(KERN_ERR "Failed to reset AD1980: AC97 link error\n");
goto reset_err;
}
@@ -253,12 +258,19 @@ static int ad1980_soc_probe(struct platform_device *pdev)
"supported\n");
}
- ac97_write(codec, AC97_MASTER, 0x0000); /* unmute line out volume */
- ac97_write(codec, AC97_PCM, 0x0000); /* unmute PCM out volume */
- ac97_write(codec, AC97_REC_GAIN, 0x0000);/* unmute record volume */
+ /* unmute captures and playbacks volume */
+ ac97_write(codec, AC97_MASTER, 0x0000);
+ ac97_write(codec, AC97_PCM, 0x0000);
+ ac97_write(codec, AC97_REC_GAIN, 0x0000);
+ ac97_write(codec, AC97_CENTER_LFE_MASTER, 0x0000);
+ ac97_write(codec, AC97_SURROUND_MASTER, 0x0000);
+
+ /*power on LFE/CENTER/Surround DACs*/
+ ext_status = ac97_read(codec, AC97_EXTENDED_STATUS);
+ ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
ad1980_add_controls(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "ad1980: failed to register card\n");
goto reset_err;
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
index 37af8607b00..b09289a1e55 100644
--- a/sound/soc/codecs/ad73311.c
+++ b/sound/soc/codecs/ad73311.c
@@ -8,14 +8,10 @@
* 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.
- *
- * Revision history
- * 25th Sep 2008 Initial version.
*/
#include <linux/init.h>
#include <linux/module.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <sound/core.h>
@@ -68,7 +64,7 @@ static int ad73311_soc_probe(struct platform_device *pdev)
goto pcm_err;
}
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "ad73311: failed to register card\n");
goto register_err;
@@ -102,6 +98,18 @@ struct snd_soc_codec_device soc_codec_dev_ad73311 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311);
+static int __init ad73311_init(void)
+{
+ return snd_soc_register_dai(&ad73311_dai);
+}
+module_init(ad73311_init);
+
+static void __exit ad73311_exit(void)
+{
+ snd_soc_unregister_dai(&ad73311_dai);
+}
+module_exit(ad73311_exit);
+
MODULE_DESCRIPTION("ASoC ad73311 driver");
MODULE_AUTHOR("Cliff Cai ");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 2a89b5888e1..81300d8d42c 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -339,7 +339,8 @@ static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai,
}
static int ak4535_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -451,8 +452,6 @@ struct snd_soc_dai ak4535_dai = {
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.ops = {
.hw_params = ak4535_hw_params,
- },
- .dai_ops = {
.set_fmt = ak4535_set_dai_fmt,
.digital_mute = ak4535_mute,
.set_sysclk = ak4535_set_dai_sysclk,
@@ -513,7 +512,7 @@ static int ak4535_init(struct snd_soc_device *socdev)
ak4535_add_controls(codec);
ak4535_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "ak4535: failed to register card\n");
goto card_err;
@@ -689,6 +688,18 @@ struct snd_soc_codec_device soc_codec_dev_ak4535 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535);
+static int __init ak4535_modinit(void)
+{
+ return snd_soc_register_dai(&ak4535_dai);
+}
+module_init(ak4535_modinit);
+
+static void __exit ak4535_exit(void)
+{
+ snd_soc_unregister_dai(&ak4535_dai);
+}
+module_exit(ak4535_exit);
+
MODULE_DESCRIPTION("Soc AK4535 driver");
MODULE_AUTHOR("Richard Purdie");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 0bbd94501d7..f1aa0c34421 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -360,13 +360,14 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg,
/*
* Program the CS4270 with the given hardware parameters.
*
- * The .dai_ops functions are used to provide board-specific data, like
+ * The .ops functions are used to provide board-specific data, like
* input frequencies, to this driver. This function takes that information,
* combines it with the hardware parameters provided, and programs the
* hardware accordingly.
*/
static int cs4270_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -450,6 +451,19 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
return ret;
}
+ /* Disable automatic volume control. It's enabled by default, and
+ * it causes volume change commands to be delayed, sometimes until
+ * after playback has started.
+ */
+
+ reg = cs4270_read_reg_cache(codec, CS4270_TRANS);
+ reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO);
+ ret = cs4270_i2c_write(codec, CS4270_TRANS, reg);
+ if (ret < 0) {
+ printk(KERN_ERR "I2C write failed\n");
+ return ret;
+ }
+
/* Thaw and power-up the codec */
ret = snd_soc_write(codec, CS4270_PWRCTL, 0);
@@ -697,10 +711,10 @@ static int cs4270_probe(struct platform_device *pdev)
if (codec->control_data) {
/* Initialize codec ops */
cs4270_dai.ops.hw_params = cs4270_hw_params;
- cs4270_dai.dai_ops.set_sysclk = cs4270_set_dai_sysclk;
- cs4270_dai.dai_ops.set_fmt = cs4270_set_dai_fmt;
+ cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk;
+ cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt;
#ifdef CONFIG_SND_SOC_CS4270_HWMUTE
- cs4270_dai.dai_ops.digital_mute = cs4270_mute;
+ cs4270_dai.ops.digital_mute = cs4270_mute;
#endif
} else
printk(KERN_INFO "cs4270: no I2C device found, "
@@ -709,7 +723,7 @@ static int cs4270_probe(struct platform_device *pdev)
printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n");
#endif
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "cs4270: failed to register card\n");
goto error_del_driver;
@@ -760,6 +774,18 @@ struct snd_soc_codec_device soc_codec_device_cs4270 = {
};
EXPORT_SYMBOL_GPL(soc_codec_device_cs4270);
+static int __init cs4270_init(void)
+{
+ return snd_soc_register_dai(&cs4270_dai);
+}
+module_init(cs4270_init);
+
+static void __exit cs4270_exit(void)
+{
+ snd_soc_unregister_dai(&cs4270_dai);
+}
+module_exit(cs4270_exit);
+
MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/l3.c b/sound/soc/codecs/l3.c
new file mode 100644
index 00000000000..5353af58862
--- /dev/null
+++ b/sound/soc/codecs/l3.c
@@ -0,0 +1,91 @@
+/*
+ * L3 code
+ *
+ * Copyright (C) 2008, Christian Pellegrin <chripell@evolware.org>
+ *
+ * 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.
+ *
+ *
+ * based on:
+ *
+ * L3 bus algorithm module.
+ *
+ * Copyright (C) 2001 Russell King, All Rights Reserved.
+ *
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/delay.h>
+
+#include <sound/l3.h>
+
+/*
+ * Send one byte of data to the chip. Data is latched into the chip on
+ * the rising edge of the clock.
+ */
+static void sendbyte(struct l3_pins *adap, unsigned int byte)
+{
+ int i;
+
+ for (i = 0; i < 8; i++) {
+ adap->setclk(0);
+ udelay(adap->data_hold);
+ adap->setdat(byte & 1);
+ udelay(adap->data_setup);
+ adap->setclk(1);
+ udelay(adap->clock_high);
+ byte >>= 1;
+ }
+}
+
+/*
+ * Send a set of bytes to the chip. We need to pulse the MODE line
+ * between each byte, but never at the start nor at the end of the
+ * transfer.
+ */
+static void sendbytes(struct l3_pins *adap, const u8 *buf,
+ int len)
+{
+ int i;
+
+ for (i = 0; i < len; i++) {
+ if (i) {
+ udelay(adap->mode_hold);
+ adap->setmode(0);
+ udelay(adap->mode);
+ }
+ adap->setmode(1);
+ udelay(adap->mode_setup);
+ sendbyte(adap, buf[i]);
+ }
+}
+
+int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len)
+{
+ adap->setclk(1);
+ adap->setdat(1);
+ adap->setmode(1);
+ udelay(adap->mode);
+
+ adap->setmode(0);
+ udelay(adap->mode_setup);
+ sendbyte(adap, addr);
+ udelay(adap->mode_hold);
+
+ sendbytes(adap, data, len);
+
+ adap->setclk(1);
+ adap->setdat(1);
+ adap->setmode(0);
+
+ return len;
+}
+EXPORT_SYMBOL_GPL(l3_write);
+
+MODULE_DESCRIPTION("L3 bit-banging driver");
+MODULE_AUTHOR("Christian Pellegrin <chripell@evolware.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
new file mode 100644
index 00000000000..9a3e67e5319
--- /dev/null
+++ b/sound/soc/codecs/pcm3008.c
@@ -0,0 +1,212 @@
+/*
+ * ALSA Soc PCM3008 codec support
+ *
+ * Author: Hugo Villeneuve
+ * Copyright (C) 2008 Lyrtech inc
+ *
+ * Based on AC97 Soc codec, original copyright follow:
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ *
+ * 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.
+ *
+ * Generic PCM3008 support.
+ */
+
+#include <linux/init.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "pcm3008.h"
+
+#define PCM3008_VERSION "0.2"
+
+#define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+struct snd_soc_dai pcm3008_dai = {
+ .name = "PCM3008 HiFi",
+ .playback = {
+ .stream_name = "PCM3008 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = PCM3008_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "PCM3008 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = PCM3008_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+};
+EXPORT_SYMBOL_GPL(pcm3008_dai);
+
+static void pcm3008_gpio_free(struct pcm3008_setup_data *setup)
+{
+ gpio_free(setup->dem0_pin);
+ gpio_free(setup->dem1_pin);
+ gpio_free(setup->pdad_pin);
+ gpio_free(setup->pdda_pin);
+}
+
+static int pcm3008_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct pcm3008_setup_data *setup = socdev->codec_data;
+ int ret = 0;
+
+ printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION);
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (!socdev->codec)
+ return -ENOMEM;
+
+ codec = socdev->codec;
+ mutex_init(&codec->mutex);
+
+ codec->name = "PCM3008";
+ codec->owner = THIS_MODULE;
+ codec->dai = &pcm3008_dai;
+ codec->num_dai = 1;
+ codec->write = NULL;
+ codec->read = NULL;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ /* Register PCMs. */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "pcm3008: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* Register Card. */
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "pcm3008: failed to register card\n");
+ goto card_err;
+ }
+
+ /* DEM1 DEM0 DE-EMPHASIS_MODE
+ * Low Low De-emphasis 44.1 kHz ON
+ * Low High De-emphasis OFF
+ * High Low De-emphasis 48 kHz ON
+ * High High De-emphasis 32 kHz ON
+ */
+
+ /* Configure DEM0 GPIO (turning OFF DAC De-emphasis). */
+ ret = gpio_request(setup->dem0_pin, "codec_dem0");
+ if (ret == 0)
+ ret = gpio_direction_output(setup->dem0_pin, 1);
+ if (ret != 0)
+ goto gpio_err;
+
+ /* Configure DEM1 GPIO (turning OFF DAC De-emphasis). */
+ ret = gpio_request(setup->dem1_pin, "codec_dem1");
+ if (ret == 0)
+ ret = gpio_direction_output(setup->dem1_pin, 0);
+ if (ret != 0)
+ goto gpio_err;
+
+ /* Configure PDAD GPIO. */
+ ret = gpio_request(setup->pdad_pin, "codec_pdad");
+ if (ret == 0)
+ ret = gpio_direction_output(setup->pdad_pin, 1);
+ if (ret != 0)
+ goto gpio_err;
+
+ /* Configure PDDA GPIO. */
+ ret = gpio_request(setup->pdda_pin, "codec_pdda");
+ if (ret == 0)
+ ret = gpio_direction_output(setup->pdda_pin, 1);
+ if (ret != 0)
+ goto gpio_err;
+
+ return ret;
+
+gpio_err:
+ pcm3008_gpio_free(setup);
+card_err:
+ snd_soc_free_pcms(socdev);
+pcm_err:
+ kfree(socdev->codec);
+
+ return ret;
+}
+
+static int pcm3008_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ struct pcm3008_setup_data *setup = socdev->codec_data;
+
+ if (!codec)
+ return 0;
+
+ pcm3008_gpio_free(setup);
+ snd_soc_free_pcms(socdev);
+ kfree(socdev->codec);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int pcm3008_soc_suspend(struct platform_device *pdev, pm_message_t msg)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct pcm3008_setup_data *setup = socdev->codec_data;
+
+ gpio_set_value(setup->pdad_pin, 0);
+ gpio_set_value(setup->pdda_pin, 0);
+
+ return 0;
+}
+
+static int pcm3008_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct pcm3008_setup_data *setup = socdev->codec_data;
+
+ gpio_set_value(setup->pdad_pin, 1);
+ gpio_set_value(setup->pdda_pin, 1);
+
+ return 0;
+}
+#else
+#define pcm3008_soc_suspend NULL
+#define pcm3008_soc_resume NULL
+#endif
+
+struct snd_soc_codec_device soc_codec_dev_pcm3008 = {
+ .probe = pcm3008_soc_probe,
+ .remove = pcm3008_soc_remove,
+ .suspend = pcm3008_soc_suspend,
+ .resume = pcm3008_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_pcm3008);
+
+static int __init pcm3008_init(void)
+{
+ return snd_soc_register_dai(&pcm3008_dai);
+}
+module_init(pcm3008_init);
+
+static void __exit pcm3008_exit(void)
+{
+ snd_soc_unregister_dai(&pcm3008_dai);
+}
+module_exit(pcm3008_exit);
+
+MODULE_DESCRIPTION("Soc PCM3008 driver");
+MODULE_AUTHOR("Hugo Villeneuve");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/pcm3008.h b/sound/soc/codecs/pcm3008.h
new file mode 100644
index 00000000000..d04e87d3c06
--- /dev/null
+++ b/sound/soc/codecs/pcm3008.h
@@ -0,0 +1,25 @@
+/*
+ * PCM3008 ALSA SoC Layer
+ *
+ * Author: Hugo Villeneuve
+ * Copyright (C) 2008 Lyrtech inc
+ *
+ * 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 __LINUX_SND_SOC_PCM3008_H
+#define __LINUX_SND_SOC_PCM3008_H
+
+struct pcm3008_setup_data {
+ unsigned dem0_pin;
+ unsigned dem1_pin;
+ unsigned pdad_pin;
+ unsigned pdda_pin;
+};
+
+extern struct snd_soc_codec_device soc_codec_dev_pcm3008;
+extern struct snd_soc_dai pcm3008_dai;
+
+#endif
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 44ef0dacd56..cac37361676 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -285,16 +285,23 @@ static inline int get_coeff(int mclk, int rate)
}
static int ssm2602_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
u16 srate;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
struct ssm2602_priv *ssm2602 = codec->private_data;
+ struct i2c_client *i2c = codec->control_data;
u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3;
int i = get_coeff(ssm2602->sysclk, params_rate(params));
+ if (substream == ssm2602->slave_substream) {
+ dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n");
+ return 0;
+ }
+
/*no match is found*/
if (i == ARRAY_SIZE(coeff_div))
return -EINVAL;
@@ -324,19 +331,26 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int ssm2602_startup(struct snd_pcm_substream *substream)
+static int ssm2602_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
struct ssm2602_priv *ssm2602 = codec->private_data;
+ struct i2c_client *i2c = codec->control_data;
struct snd_pcm_runtime *master_runtime;
/* The DAI has shared clocks so if we already have a playback or
* capture going then constrain this substream to match it.
+ * TODO: the ssm2602 allows pairs of non-matching PB/REC rates
*/
if (ssm2602->master_substream) {
master_runtime = ssm2602->master_substream->runtime;
+ dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n",
+ master_runtime->sample_bits,
+ master_runtime->rate);
+
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE,
master_runtime->rate,
@@ -354,7 +368,8 @@ static int ssm2602_startup(struct snd_pcm_substream *substream)
return 0;
}
-static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream)
+static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -365,14 +380,21 @@ static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream)
return 0;
}
-static void ssm2602_shutdown(struct snd_pcm_substream *substream)
+static void ssm2602_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
+ struct ssm2602_priv *ssm2602 = codec->private_data;
/* deactivate */
if (!codec->active)
ssm2602_write(codec, SSM2602_ACTIVE, 0);
+
+ if (ssm2602->master_substream == substream)
+ ssm2602->master_substream = ssm2602->slave_substream;
+
+ ssm2602->slave_substream = NULL;
}
static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
@@ -432,10 +454,10 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
iface |= 0x0001;
break;
case SND_SOC_DAIFMT_DSP_A:
- iface |= 0x0003;
+ iface |= 0x0013;
break;
case SND_SOC_DAIFMT_DSP_B:
- iface |= 0x0013;
+ iface |= 0x0003;
break;
default:
return -EINVAL;
@@ -496,6 +518,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
SNDRV_PCM_RATE_96000)
+#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
struct snd_soc_dai ssm2602_dai = {
.name = "SSM2602",
.playback = {
@@ -503,20 +528,18 @@ struct snd_soc_dai ssm2602_dai = {
.channels_min = 2,
.channels_max = 2,
.rates = SSM2602_RATES,
- .formats = SNDRV_PCM_FMTBIT_S32_LE,},
+ .formats = SSM2602_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SSM2602_RATES,
- .formats = SNDRV_PCM_FMTBIT_S32_LE,},
+ .formats = SSM2602_FORMATS,},
.ops = {
.startup = ssm2602_startup,
.prepare = ssm2602_pcm_prepare,
.hw_params = ssm2602_hw_params,
.shutdown = ssm2602_shutdown,
- },
- .dai_ops = {
.digital_mute = ssm2602_mute,
.set_sysclk = ssm2602_set_dai_sysclk,
.set_fmt = ssm2602_set_dai_fmt,
@@ -601,7 +624,7 @@ static int ssm2602_init(struct snd_soc_device *socdev)
ssm2602_add_controls(codec);
ssm2602_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
pr_err("ssm2602: failed to register card\n");
goto card_err;
@@ -770,6 +793,18 @@ struct snd_soc_codec_device soc_codec_dev_ssm2602 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602);
+static int __init ssm2602_modinit(void)
+{
+ return snd_soc_register_dai(&ssm2602_dai);
+}
+module_init(ssm2602_modinit);
+
+static void __exit ssm2602_exit(void)
+{
+ snd_soc_unregister_dai(&ssm2602_dai);
+}
+module_exit(ssm2602_exit);
+
MODULE_DESCRIPTION("ASoC ssm2602 driver");
MODULE_AUTHOR("Cliff Cai");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 44308dac9e1..cfdea007c4c 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -37,12 +37,6 @@
#define AIC23_VERSION "0.1"
-struct tlv320aic23_srate_reg_info {
- u32 sample_rate;
- u8 control; /* SR3, SR2, SR1, SR0 and BOSR */
- u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
-};
-
/*
* AIC23 register cache
*/
@@ -261,20 +255,156 @@ static const struct snd_soc_dapm_route intercon[] = {
};
-/* tlv320aic23 related */
-static const struct tlv320aic23_srate_reg_info srate_reg_info[] = {
- {4000, 0x06, 1}, /* 4000 */
- {8000, 0x06, 0}, /* 8000 */
- {16000, 0x0C, 1}, /* 16000 */
- {22050, 0x11, 1}, /* 22050 */
- {24000, 0x00, 1}, /* 24000 */
- {32000, 0x0C, 0}, /* 32000 */
- {44100, 0x11, 0}, /* 44100 */
- {48000, 0x00, 0}, /* 48000 */
- {88200, 0x1F, 0}, /* 88200 */
- {96000, 0x0E, 0}, /* 96000 */
+/* AIC23 driver data */
+struct aic23 {
+ struct snd_soc_codec codec;
+ int mclk;
+ int requested_adc;
+ int requested_dac;
+};
+
+/*
+ * Common Crystals used
+ * 11.2896 Mhz /128 = *88.2k /192 = 58.8k
+ * 12.0000 Mhz /125 = *96k /136 = 88.235K
+ * 12.2880 Mhz /128 = *96k /192 = 64k
+ * 16.9344 Mhz /128 = 132.3k /192 = *88.2k
+ * 18.4320 Mhz /128 = 144k /192 = *96k
+ */
+
+/*
+ * Normal BOSR 0-256/2 = 128, 1-384/2 = 192
+ * USB BOSR 0-250/2 = 125, 1-272/2 = 136
+ */
+static const int bosr_usb_divisor_table[] = {
+ 128, 125, 192, 136
+};
+#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7))
+#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15))
+static const unsigned short sr_valid_mask[] = {
+ LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/
+ LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/
+ LOWER_GROUP, /* Usb, bosr - 0*/
+ UPPER_GROUP, /* Usb, bosr - 1*/
+};
+/*
+ * Every divisor is a factor of 11*12
+ */
+#define SR_MULT (11*12)
+#define A(x) (x) ? (SR_MULT/x) : 0
+static const unsigned char sr_adc_mult_table[] = {
+ A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1),
+ A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1)
+};
+static const unsigned char sr_dac_mult_table[] = {
+ A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1),
+ A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1)
};
+static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
+ int dac, int dac_l, int dac_h, int need_dac)
+{
+ if ((adc >= adc_l) && (adc <= adc_h) &&
+ (dac >= dac_l) && (dac <= dac_h)) {
+ int diff_adc = need_adc - adc;
+ int diff_dac = need_dac - dac;
+ return abs(diff_adc) + abs(diff_dac);
+ }
+ return UINT_MAX;
+}
+
+static int find_rate(int mclk, u32 need_adc, u32 need_dac)
+{
+ int i, j;
+ int best_i = -1;
+ int best_j = -1;
+ int best_div = 0;
+ unsigned best_score = UINT_MAX;
+ int adc_l, adc_h, dac_l, dac_h;
+
+ need_adc *= SR_MULT;
+ need_dac *= SR_MULT;
+ /*
+ * rates given are +/- 1/32
+ */
+ adc_l = need_adc - (need_adc >> 5);
+ adc_h = need_adc + (need_adc >> 5);
+ dac_l = need_dac - (need_dac >> 5);
+ dac_h = need_dac + (need_dac >> 5);
+ for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) {
+ int base = mclk / bosr_usb_divisor_table[i];
+ int mask = sr_valid_mask[i];
+ for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table);
+ j++, mask >>= 1) {
+ int adc;
+ int dac;
+ int score;
+ if ((mask & 1) == 0)
+ continue;
+ adc = base * sr_adc_mult_table[j];
+ dac = base * sr_dac_mult_table[j];
+ score = get_score(adc, adc_l, adc_h, need_adc,
+ dac, dac_l, dac_h, need_dac);
+ if (best_score > score) {
+ best_score = score;
+ best_i = i;
+ best_j = j;
+ best_div = 0;
+ }
+ score = get_score((adc >> 1), adc_l, adc_h, need_adc,
+ (dac >> 1), dac_l, dac_h, need_dac);
+ /* prefer to have a /2 */
+ if ((score != UINT_MAX) && (best_score >= score)) {
+ best_score = score;
+ best_i = i;
+ best_j = j;
+ best_div = 1;
+ }
+ }
+ }
+ return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT);
+}
+
+#ifdef DEBUG
+static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk,
+ u32 *sample_rate_adc, u32 *sample_rate_dac)
+{
+ int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE);
+ int sr = (src >> 2) & 0x0f;
+ int val = (mclk / bosr_usb_divisor_table[src & 3]);
+ int adc = (val * sr_adc_mult_table[sr]) / SR_MULT;
+ int dac = (val * sr_dac_mult_table[sr]) / SR_MULT;
+ if (src & TLV320AIC23_CLKIN_HALF) {
+ adc >>= 1;
+ dac >>= 1;
+ }
+ *sample_rate_adc = adc;
+ *sample_rate_dac = dac;
+}
+#endif
+
+static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
+ u32 sample_rate_adc, u32 sample_rate_dac)
+{
+ /* Search for the right sample rate */
+ int data = find_rate(mclk, sample_rate_adc, sample_rate_dac);
+ if (data < 0) {
+ printk(KERN_ERR "%s:Invalid rate %u,%u requested\n",
+ __func__, sample_rate_adc, sample_rate_dac);
+ return -EINVAL;
+ }
+ tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
+#ifdef DEBUG
+ {
+ u32 adc, dac;
+ get_current_sample_rates(codec, mclk, &adc, &dac);
+ printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n",
+ adc, dac, data);
+ }
+#endif
+ return 0;
+}
+
static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
{
snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
@@ -288,32 +418,36 @@ static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
}
static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
- u16 iface_reg, data;
- u8 count = 0;
+ u16 iface_reg;
+ int ret;
+ struct aic23 *aic23 = container_of(codec, struct aic23, codec);
+ u32 sample_rate_adc = aic23->requested_adc;
+ u32 sample_rate_dac = aic23->requested_dac;
+ u32 sample_rate = params_rate(params);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ aic23->requested_dac = sample_rate_dac = sample_rate;
+ if (!sample_rate_adc)
+ sample_rate_adc = sample_rate;
+ } else {
+ aic23->requested_adc = sample_rate_adc = sample_rate;
+ if (!sample_rate_dac)
+ sample_rate_dac = sample_rate;
+ }
+ ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc,
+ sample_rate_dac);
+ if (ret < 0)
+ return ret;
iface_reg =
tlv320aic23_read_reg_cache(codec,
TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
-
- /* Search for the right sample rate */
- /* Verify what happens if the rate is not supported
- * now it goes to 96Khz */
- while ((srate_reg_info[count].sample_rate != params_rate(params)) &&
- (count < ARRAY_SIZE(srate_reg_info))) {
- count++;
- }
-
- data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) |
- (srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) |
- TLV320AIC23_USB_CLK_ON;
-
- tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
-
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
@@ -332,7 +466,8 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
+static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -344,17 +479,23 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
return 0;
}
-static void tlv320aic23_shutdown(struct snd_pcm_substream *substream)
+static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
+ struct aic23 *aic23 = container_of(codec, struct aic23, codec);
/* deactivate */
if (!codec->active) {
udelay(50);
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
}
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ aic23->requested_dac = 0;
+ else
+ aic23->requested_adc = 0;
}
static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
@@ -400,7 +541,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
case SND_SOC_DAIFMT_I2S:
iface_reg |= TLV320AIC23_FOR_I2S;
break;
- case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
iface_reg |= TLV320AIC23_FOR_DSP;
break;
case SND_SOC_DAIFMT_RIGHT_J:
@@ -422,12 +563,9 @@ static int tlv320aic23_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;
-
- switch (freq) {
- case 12000000:
- return 0;
- }
- return -EINVAL;
+ struct aic23 *aic23 = container_of(codec, struct aic23, codec);
+ aic23->mclk = freq;
+ return 0;
}
static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
@@ -478,12 +616,10 @@ struct snd_soc_dai tlv320aic23_dai = {
.prepare = tlv320aic23_pcm_prepare,
.hw_params = tlv320aic23_hw_params,
.shutdown = tlv320aic23_shutdown,
- },
- .dai_ops = {
- .digital_mute = tlv320aic23_mute,
- .set_fmt = tlv320aic23_set_dai_fmt,
- .set_sysclk = tlv320aic23_set_dai_sysclk,
- }
+ .digital_mute = tlv320aic23_mute,
+ .set_fmt = tlv320aic23_set_dai_fmt,
+ .set_sysclk = tlv320aic23_set_dai_sysclk,
+ }
};
EXPORT_SYMBOL_GPL(tlv320aic23_dai);
@@ -584,7 +720,7 @@ static int tlv320aic23_init(struct snd_soc_device *socdev)
tlv320aic23_add_controls(codec);
tlv320aic23_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "tlv320aic23: failed to register card\n");
goto card_err;
@@ -659,14 +795,15 @@ static int tlv320aic23_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec;
+ struct aic23 *aic23;
int ret = 0;
printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
- codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
- if (codec == NULL)
+ aic23 = kzalloc(sizeof(struct aic23), GFP_KERNEL);
+ if (aic23 == NULL)
return -ENOMEM;
-
+ codec = &aic23->codec;
socdev->codec = codec;
mutex_init(&codec->mutex);
INIT_LIST_HEAD(&codec->dapm_widgets);
@@ -687,6 +824,7 @@ static int tlv320aic23_remove(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
+ struct aic23 *aic23 = container_of(codec, struct aic23, codec);
if (codec->control_data)
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -697,7 +835,7 @@ static int tlv320aic23_remove(struct platform_device *pdev)
i2c_del_driver(&tlv320aic23_i2c_driver);
#endif
kfree(codec->reg_cache);
- kfree(codec);
+ kfree(aic23);
return 0;
}
@@ -709,6 +847,18 @@ struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23);
+static int __init tlv320aic23_modinit(void)
+{
+ return snd_soc_register_dai(&tlv320aic23_dai);
+}
+module_init(tlv320aic23_modinit);
+
+static void __exit tlv320aic23_exit(void)
+{
+ snd_soc_unregister_dai(&tlv320aic23_dai);
+}
+module_exit(tlv320aic23_exit);
+
MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index bed8a9e63dd..29f2f1a017f 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -125,7 +125,8 @@ static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
* Digital Audio Interface Operations
*/
static int aic26_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -287,8 +288,6 @@ struct snd_soc_dai aic26_dai = {
},
.ops = {
.hw_params = aic26_hw_params,
- },
- .dai_ops = {
.digital_mute = aic26_mute,
.set_sysclk = aic26_set_sysclk,
.set_fmt = aic26_set_fmt,
@@ -360,7 +359,7 @@ static int aic26_probe(struct platform_device *pdev)
/* CODEC is setup, we can register the card now */
dev_dbg(&pdev->dev, "Registering card\n");
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
dev_err(&pdev->dev, "aic26: failed to register card\n");
goto card_err;
@@ -427,7 +426,7 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set);
static int aic26_spi_probe(struct spi_device *spi)
{
struct aic26 *aic26;
- int rc, i, reg;
+ int ret, i, reg;
dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n");
@@ -457,6 +456,14 @@ static int aic26_spi_probe(struct spi_device *spi)
aic26->codec.reg_cache_size = AIC26_NUM_REGS;
aic26->codec.reg_cache = aic26->reg_cache;
+ aic26_dai.dev = &spi->dev;
+ ret = snd_soc_register_dai(&aic26_dai);
+ if (ret != 0) {
+ dev_err(&spi->dev, "Failed to register DAI: %d\n", ret);
+ kfree(aic26);
+ return ret;
+ }
+
/* Reset the codec to power on defaults */
aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00);
@@ -475,8 +482,8 @@ static int aic26_spi_probe(struct spi_device *spi)
/* Register the sysfs files for debugging */
/* Create SysFS files */
- rc = device_create_file(&spi->dev, &dev_attr_keyclick);
- if (rc)
+ ret = device_create_file(&spi->dev, &dev_attr_keyclick);
+ if (ret)
dev_info(&spi->dev, "error creating sysfs files\n");
#if defined(CONFIG_SND_SOC_OF_SIMPLE)
@@ -493,6 +500,7 @@ static int aic26_spi_remove(struct spi_device *spi)
{
struct aic26 *aic26 = dev_get_drvdata(&spi->dev);
+ snd_soc_unregister_dai(&aic26_dai);
kfree(aic26);
return 0;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index cff276ee261..b47a749c5ea 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -253,11 +253,17 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL,
DACR1_2_RLOPM_VOL, 0, 0x7f, 1),
- SOC_DOUBLE_R("Line DAC Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3,
- 0x01, 0),
- SOC_DOUBLE_R("Line PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
- PGAR_2_RLOPM_VOL, 0, 0x7f, 1),
- SOC_DOUBLE_R("Line Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
+ SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0),
+ SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0),
+ SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL,
+ DACR1_2_LLOPM_VOL, 0, 0x7f, 1),
+ SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
+ 0, 0x7f, 1),
+ SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL,
+ 0, 0x7f, 1),
+ SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
+ LINE2R_2_LLOPM_VOL, 0, 0x7f, 1),
+ SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL,
LINE2R_2_RLOPM_VOL, 0, 0x7f, 1),
SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL,
@@ -272,8 +278,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
DACR1_2_HPROUT_VOL, 0, 0x7f, 1),
SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3,
0x01, 0),
- SOC_DOUBLE_R("HP PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL,
+ SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL,
PGAR_2_HPROUT_VOL, 0, 0x7f, 1),
+ SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL,
+ 0, 0x7f, 1),
+ SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL,
+ 0, 0x7f, 1),
SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL,
LINE2R_2_HPROUT_VOL, 0, 0x7f, 1),
@@ -281,8 +291,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
DACR1_2_HPRCOM_VOL, 0, 0x7f, 1),
SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3,
0x01, 0),
- SOC_DOUBLE_R("HPCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL,
- PGAR_2_HPRCOM_VOL, 0, 0x7f, 1),
+ SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL,
+ 0, 0x7f, 1),
+ SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL,
+ 0, 0x7f, 1),
SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL,
LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1),
@@ -333,7 +345,8 @@ SOC_DAPM_ENUM("Route", aic3x_enum[RHPCOM_ENUM]);
/* Left DAC_L1 Mixer */
static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = {
- SOC_DAPM_SINGLE("Line Switch", DACL1_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineL Switch", DACL1_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineR Switch", DACL1_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Mono Switch", DACL1_2_MONOLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("HP Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("HPCOM Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0),
@@ -341,7 +354,8 @@ static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = {
/* Right DAC_R1 Mixer */
static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = {
- SOC_DAPM_SINGLE("Line Switch", DACR1_2_RLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineL Switch", DACR1_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineR Switch", DACR1_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Mono Switch", DACR1_2_MONOLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("HP Switch", DACR1_2_HPROUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("HPCOM Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0),
@@ -350,14 +364,18 @@ static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = {
/* Left PGA Mixer */
static const struct snd_kcontrol_new aic3x_left_pga_mixer_controls[] = {
SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1),
+ SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_LADC_CTRL, 3, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Line2L Switch", LINE2L_2_LADC_CTRL, 3, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1),
+ SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_LADC_CTRL, 0, 1, 1),
};
/* Right PGA Mixer */
static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = {
SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1),
+ SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_RADC_CTRL, 3, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Line2R Switch", LINE2R_2_RADC_CTRL, 3, 1, 1),
+ SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_RADC_CTRL, 4, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1),
};
@@ -379,34 +397,42 @@ SOC_DAPM_ENUM("Route", aic3x_enum[LINE2R_ENUM]);
/* Left PGA Bypass Mixer */
static const struct snd_kcontrol_new aic3x_left_pga_bp_mixer_controls[] = {
- SOC_DAPM_SINGLE("Line Switch", PGAL_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineL Switch", PGAL_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineR Switch", PGAL_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Mono Switch", PGAL_2_MONOLOPM_VOL, 7, 1, 0),
- SOC_DAPM_SINGLE("HP Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0),
- SOC_DAPM_SINGLE("HPCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPL Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPR Switch", PGAL_2_HPROUT_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPLCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPRCOM Switch", PGAL_2_HPRCOM_VOL, 7, 1, 0),
};
/* Right PGA Bypass Mixer */
static const struct snd_kcontrol_new aic3x_right_pga_bp_mixer_controls[] = {
- SOC_DAPM_SINGLE("Line Switch", PGAR_2_RLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineL Switch", PGAR_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineR Switch", PGAR_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Mono Switch", PGAR_2_MONOLOPM_VOL, 7, 1, 0),
- SOC_DAPM_SINGLE("HP Switch", PGAR_2_HPROUT_VOL, 7, 1, 0),
- SOC_DAPM_SINGLE("HPCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPL Switch", PGAR_2_HPLOUT_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPR Switch", PGAR_2_HPROUT_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPLCOM Switch", PGAR_2_HPLCOM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPRCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0),
};
/* Left Line2 Bypass Mixer */
static const struct snd_kcontrol_new aic3x_left_line2_bp_mixer_controls[] = {
- SOC_DAPM_SINGLE("Line Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineL Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineR Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Mono Switch", LINE2L_2_MONOLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("HP Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0),
- SOC_DAPM_SINGLE("HPCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPLCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0),
};
/* Right Line2 Bypass Mixer */
static const struct snd_kcontrol_new aic3x_right_line2_bp_mixer_controls[] = {
- SOC_DAPM_SINGLE("Line Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineL Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineR Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Mono Switch", LINE2R_2_MONOLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("HP Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0),
- SOC_DAPM_SINGLE("HPCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPRCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0),
};
static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
@@ -439,22 +465,26 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
/* Mono Output */
SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0),
- /* Left Inputs to Left ADC */
+ /* Inputs to Left ADC */
SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0),
SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0,
&aic3x_left_pga_mixer_controls[0],
ARRAY_SIZE(aic3x_left_pga_mixer_controls)),
SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0,
&aic3x_left_line1_mux_controls),
+ SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0,
+ &aic3x_left_line1_mux_controls),
SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0,
&aic3x_left_line2_mux_controls),
- /* Right Inputs to Right ADC */
+ /* Inputs to Right ADC */
SND_SOC_DAPM_ADC("Right ADC", "Right Capture",
LINE1R_2_RADC_CTRL, 2, 0),
SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0,
&aic3x_right_pga_mixer_controls[0],
ARRAY_SIZE(aic3x_right_pga_mixer_controls)),
+ SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0,
+ &aic3x_right_line1_mux_controls),
SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0,
&aic3x_right_line1_mux_controls),
SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0,
@@ -531,7 +561,8 @@ static const struct snd_soc_dapm_route intercon[] = {
{"Left DAC Mux", "DAC_L2", "Left DAC"},
{"Left DAC Mux", "DAC_L3", "Left DAC"},
- {"Left DAC_L1 Mixer", "Line Switch", "Left DAC Mux"},
+ {"Left DAC_L1 Mixer", "LineL Switch", "Left DAC Mux"},
+ {"Left DAC_L1 Mixer", "LineR Switch", "Left DAC Mux"},
{"Left DAC_L1 Mixer", "Mono Switch", "Left DAC Mux"},
{"Left DAC_L1 Mixer", "HP Switch", "Left DAC Mux"},
{"Left DAC_L1 Mixer", "HPCOM Switch", "Left DAC Mux"},
@@ -557,7 +588,8 @@ static const struct snd_soc_dapm_route intercon[] = {
{"Right DAC Mux", "DAC_R2", "Right DAC"},
{"Right DAC Mux", "DAC_R3", "Right DAC"},
- {"Right DAC_R1 Mixer", "Line Switch", "Right DAC Mux"},
+ {"Right DAC_R1 Mixer", "LineL Switch", "Right DAC Mux"},
+ {"Right DAC_R1 Mixer", "LineR Switch", "Right DAC Mux"},
{"Right DAC_R1 Mixer", "Mono Switch", "Right DAC Mux"},
{"Right DAC_R1 Mixer", "HP Switch", "Right DAC Mux"},
{"Right DAC_R1 Mixer", "HPCOM Switch", "Right DAC Mux"},
@@ -592,8 +624,10 @@ static const struct snd_soc_dapm_route intercon[] = {
{"Left Line2L Mux", "differential", "LINE2L"},
{"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"},
+ {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"},
{"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"},
{"Left PGA Mixer", "Mic3L Switch", "MIC3L"},
+ {"Left PGA Mixer", "Mic3R Switch", "MIC3R"},
{"Left ADC", NULL, "Left PGA Mixer"},
{"Left ADC", NULL, "GPIO1 dmic modclk"},
@@ -605,18 +639,23 @@ static const struct snd_soc_dapm_route intercon[] = {
{"Right Line2R Mux", "single-ended", "LINE2R"},
{"Right Line2R Mux", "differential", "LINE2R"},
+ {"Right PGA Mixer", "Line1L Switch", "Right Line1L Mux"},
{"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"},
{"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"},
+ {"Right PGA Mixer", "Mic3L Switch", "MIC3L"},
{"Right PGA Mixer", "Mic3R Switch", "MIC3R"},
{"Right ADC", NULL, "Right PGA Mixer"},
{"Right ADC", NULL, "GPIO1 dmic modclk"},
/* Left PGA Bypass */
- {"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "LineL Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "LineR Switch", "Left PGA Mixer"},
{"Left PGA Bypass Mixer", "Mono Switch", "Left PGA Mixer"},
- {"Left PGA Bypass Mixer", "HP Switch", "Left PGA Mixer"},
- {"Left PGA Bypass Mixer", "HPCOM Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "HPL Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "HPR Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "HPLCOM Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "HPRCOM Switch", "Left PGA Mixer"},
{"Left HPCOM Mux", "differential of HPLOUT", "Left PGA Bypass Mixer"},
{"Left HPCOM Mux", "constant VCM", "Left PGA Bypass Mixer"},
@@ -627,10 +666,13 @@ static const struct snd_soc_dapm_route intercon[] = {
{"Left HP Out", NULL, "Left PGA Bypass Mixer"},
/* Right PGA Bypass */
- {"Right PGA Bypass Mixer", "Line Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "LineL Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "LineR Switch", "Right PGA Mixer"},
{"Right PGA Bypass Mixer", "Mono Switch", "Right PGA Mixer"},
- {"Right PGA Bypass Mixer", "HP Switch", "Right PGA Mixer"},
- {"Right PGA Bypass Mixer", "HPCOM Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "HPL Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "HPR Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "HPLCOM Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "HPRCOM Switch", "Right PGA Mixer"},
{"Right HPCOM Mux", "differential of HPROUT", "Right PGA Bypass Mixer"},
{"Right HPCOM Mux", "constant VCM", "Right PGA Bypass Mixer"},
@@ -643,10 +685,11 @@ static const struct snd_soc_dapm_route intercon[] = {
{"Right HP Out", NULL, "Right PGA Bypass Mixer"},
/* Left Line2 Bypass */
- {"Left Line2 Bypass Mixer", "Line Switch", "Left Line2L Mux"},
+ {"Left Line2 Bypass Mixer", "LineL Switch", "Left Line2L Mux"},
+ {"Left Line2 Bypass Mixer", "LineR Switch", "Left Line2L Mux"},
{"Left Line2 Bypass Mixer", "Mono Switch", "Left Line2L Mux"},
{"Left Line2 Bypass Mixer", "HP Switch", "Left Line2L Mux"},
- {"Left Line2 Bypass Mixer", "HPCOM Switch", "Left Line2L Mux"},
+ {"Left Line2 Bypass Mixer", "HPLCOM Switch", "Left Line2L Mux"},
{"Left HPCOM Mux", "differential of HPLOUT", "Left Line2 Bypass Mixer"},
{"Left HPCOM Mux", "constant VCM", "Left Line2 Bypass Mixer"},
@@ -657,10 +700,11 @@ static const struct snd_soc_dapm_route intercon[] = {
{"Left HP Out", NULL, "Left Line2 Bypass Mixer"},
/* Right Line2 Bypass */
- {"Right Line2 Bypass Mixer", "Line Switch", "Right Line2R Mux"},
+ {"Right Line2 Bypass Mixer", "LineL Switch", "Right Line2R Mux"},
+ {"Right Line2 Bypass Mixer", "LineR Switch", "Right Line2R Mux"},
{"Right Line2 Bypass Mixer", "Mono Switch", "Right Line2R Mux"},
{"Right Line2 Bypass Mixer", "HP Switch", "Right Line2R Mux"},
- {"Right Line2 Bypass Mixer", "HPCOM Switch", "Right Line2R Mux"},
+ {"Right Line2 Bypass Mixer", "HPRCOM Switch", "Right Line2R Mux"},
{"Right HPCOM Mux", "differential of HPROUT", "Right Line2 Bypass Mixer"},
{"Right HPCOM Mux", "constant VCM", "Right Line2 Bypass Mixer"},
@@ -694,7 +738,8 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec)
}
static int aic3x_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -846,6 +891,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
struct snd_soc_codec *codec = codec_dai->codec;
struct aic3x_priv *aic3x = codec->private_data;
u8 iface_areg, iface_breg;
+ int delay = 0;
iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f;
iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f;
@@ -871,6 +917,8 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
SND_SOC_DAIFMT_INV_MASK)) {
case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
break;
+ case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF):
+ delay = 1;
case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
iface_breg |= (0x01 << 6);
break;
@@ -887,6 +935,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
/* set iface */
aic3x_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg);
aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg);
+ aic3x_write(codec, AIC3X_ASD_INTF_CTRLC, delay);
return 0;
}
@@ -981,14 +1030,41 @@ int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio)
}
EXPORT_SYMBOL_GPL(aic3x_get_gpio);
+void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect,
+ int headset_debounce, int button_debounce)
+{
+ u8 val;
+
+ val = ((detect & AIC3X_HEADSET_DETECT_MASK)
+ << AIC3X_HEADSET_DETECT_SHIFT) |
+ ((headset_debounce & AIC3X_HEADSET_DEBOUNCE_MASK)
+ << AIC3X_HEADSET_DEBOUNCE_SHIFT) |
+ ((button_debounce & AIC3X_BUTTON_DEBOUNCE_MASK)
+ << AIC3X_BUTTON_DEBOUNCE_SHIFT);
+
+ if (detect & AIC3X_HEADSET_DETECT_MASK)
+ val |= AIC3X_HEADSET_DETECT_ENABLED;
+
+ aic3x_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val);
+}
+EXPORT_SYMBOL_GPL(aic3x_set_headset_detection);
+
int aic3x_headset_detected(struct snd_soc_codec *codec)
{
u8 val;
- aic3x_read(codec, AIC3X_RT_IRQ_FLAGS_REG, &val);
- return (val >> 2) & 1;
+ aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
+ return (val >> 4) & 1;
}
EXPORT_SYMBOL_GPL(aic3x_headset_detected);
+int aic3x_button_pressed(struct snd_soc_codec *codec)
+{
+ u8 val;
+ aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
+ return (val >> 5) & 1;
+}
+EXPORT_SYMBOL_GPL(aic3x_button_pressed);
+
#define AIC3X_RATES SNDRV_PCM_RATE_8000_96000
#define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
@@ -1009,8 +1085,6 @@ struct snd_soc_dai aic3x_dai = {
.formats = AIC3X_FORMATS,},
.ops = {
.hw_params = aic3x_hw_params,
- },
- .dai_ops = {
.digital_mute = aic3x_mute,
.set_sysclk = aic3x_set_dai_sysclk,
.set_fmt = aic3x_set_dai_fmt,
@@ -1152,7 +1226,7 @@ static int aic3x_init(struct snd_soc_device *socdev)
aic3x_add_controls(codec);
aic3x_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "aic3x: failed to register card\n");
goto card_err;
@@ -1341,6 +1415,18 @@ struct snd_soc_codec_device soc_codec_dev_aic3x = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_aic3x);
+static int __init aic3x_modinit(void)
+{
+ return snd_soc_register_dai(&aic3x_dai);
+}
+module_init(aic3x_modinit);
+
+static void __exit aic3x_exit(void)
+{
+ snd_soc_unregister_dai(&aic3x_dai);
+}
+module_exit(aic3x_exit);
+
MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver");
MODULE_AUTHOR("Vladimir Barinov");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index 00a195aa02e..ac827e578c4 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -35,11 +35,15 @@
#define AIC3X_ASD_INTF_CTRLA 8
/* Audio serial data interface control register B */
#define AIC3X_ASD_INTF_CTRLB 9
+/* Audio serial data interface control register C */
+#define AIC3X_ASD_INTF_CTRLC 10
/* Audio overflow status and PLL R value programming register */
#define AIC3X_OVRF_STATUS_AND_PLLR_REG 11
/* Audio codec digital filter control register */
#define AIC3X_CODEC_DFILT_CTRL 12
-
+/* Headset/button press detection register */
+#define AIC3X_HEADSET_DETECT_CTRL_A 13
+#define AIC3X_HEADSET_DETECT_CTRL_B 14
/* ADC PGA Gain control registers */
#define LADC_VOL 15
#define RADC_VOL 16
@@ -48,7 +52,9 @@
#define MIC3LR_2_RADC_CTRL 18
/* Line1 Input control registers */
#define LINE1L_2_LADC_CTRL 19
+#define LINE1R_2_LADC_CTRL 21
#define LINE1R_2_RADC_CTRL 22
+#define LINE1L_2_RADC_CTRL 24
/* Line2 Input control registers */
#define LINE2L_2_LADC_CTRL 20
#define LINE2R_2_RADC_CTRL 23
@@ -79,6 +85,8 @@
#define LINE2L_2_HPLOUT_VOL 45
#define LINE2R_2_HPROUT_VOL 62
#define PGAL_2_HPLOUT_VOL 46
+#define PGAL_2_HPROUT_VOL 60
+#define PGAR_2_HPLOUT_VOL 49
#define PGAR_2_HPROUT_VOL 63
#define DACL1_2_HPLOUT_VOL 47
#define DACR1_2_HPROUT_VOL 64
@@ -88,6 +96,8 @@
#define LINE2L_2_HPLCOM_VOL 52
#define LINE2R_2_HPRCOM_VOL 69
#define PGAL_2_HPLCOM_VOL 53
+#define PGAR_2_HPLCOM_VOL 56
+#define PGAL_2_HPRCOM_VOL 67
#define PGAR_2_HPRCOM_VOL 70
#define DACL1_2_HPLCOM_VOL 54
#define DACR1_2_HPRCOM_VOL 71
@@ -103,11 +113,17 @@
#define MONOLOPM_CTRL 79
/* Line Output Plus/Minus control registers */
#define LINE2L_2_LLOPM_VOL 80
+#define LINE2L_2_RLOPM_VOL 87
+#define LINE2R_2_LLOPM_VOL 83
#define LINE2R_2_RLOPM_VOL 90
#define PGAL_2_LLOPM_VOL 81
+#define PGAL_2_RLOPM_VOL 88
+#define PGAR_2_LLOPM_VOL 84
#define PGAR_2_RLOPM_VOL 91
#define DACL1_2_LLOPM_VOL 82
+#define DACL1_2_RLOPM_VOL 89
#define DACR1_2_RLOPM_VOL 92
+#define DACR1_2_LLOPM_VOL 85
#define LLOPM_CTRL 86
#define RLOPM_CTRL 93
/* GPIO/IRQ registers */
@@ -221,7 +237,49 @@ enum {
void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state);
int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio);
+
+/* headset detection / button API */
+
+/* The AIC3x supports detection of stereo headsets (GND + left + right signal)
+ * and cellular headsets (GND + speaker output + microphone input).
+ * It is recommended to enable MIC bias for this function to work properly.
+ * For more information, please refer to the datasheet. */
+enum {
+ AIC3X_HEADSET_DETECT_OFF = 0,
+ AIC3X_HEADSET_DETECT_STEREO = 1,
+ AIC3X_HEADSET_DETECT_CELLULAR = 2,
+ AIC3X_HEADSET_DETECT_BOTH = 3
+};
+
+enum {
+ AIC3X_HEADSET_DEBOUNCE_16MS = 0,
+ AIC3X_HEADSET_DEBOUNCE_32MS = 1,
+ AIC3X_HEADSET_DEBOUNCE_64MS = 2,
+ AIC3X_HEADSET_DEBOUNCE_128MS = 3,
+ AIC3X_HEADSET_DEBOUNCE_256MS = 4,
+ AIC3X_HEADSET_DEBOUNCE_512MS = 5
+};
+
+enum {
+ AIC3X_BUTTON_DEBOUNCE_0MS = 0,
+ AIC3X_BUTTON_DEBOUNCE_8MS = 1,
+ AIC3X_BUTTON_DEBOUNCE_16MS = 2,
+ AIC3X_BUTTON_DEBOUNCE_32MS = 3
+};
+
+#define AIC3X_HEADSET_DETECT_ENABLED 0x80
+#define AIC3X_HEADSET_DETECT_SHIFT 5
+#define AIC3X_HEADSET_DETECT_MASK 3
+#define AIC3X_HEADSET_DEBOUNCE_SHIFT 2
+#define AIC3X_HEADSET_DEBOUNCE_MASK 7
+#define AIC3X_BUTTON_DEBOUNCE_SHIFT 0
+#define AIC3X_BUTTON_DEBOUNCE_MASK 3
+
+/* see the enums above for valid parameters to this function */
+void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect,
+ int headset_debounce, int button_debounce);
int aic3x_headset_detected(struct snd_soc_codec *codec);
+int aic3x_button_pressed(struct snd_soc_codec *codec);
struct aic3x_setup_data {
int i2c_bus;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
new file mode 100644
index 00000000000..ea370a4f86d
--- /dev/null
+++ b/sound/soc/codecs/twl4030.c
@@ -0,0 +1,1312 @@
+/*
+ * ALSA SoC TWL4030 codec driver
+ *
+ * Author: Steve Sakoman, <steve@sakoman.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#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/i2c/twl4030.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "twl4030.h"
+
+/*
+ * twl4030 register cache & default register settings
+ */
+static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
+ 0x00, /* this register not used */
+ 0x93, /* REG_CODEC_MODE (0x1) */
+ 0xc3, /* REG_OPTION (0x2) */
+ 0x00, /* REG_UNKNOWN (0x3) */
+ 0x00, /* REG_MICBIAS_CTL (0x4) */
+ 0x20, /* REG_ANAMICL (0x5) */
+ 0x00, /* REG_ANAMICR (0x6) */
+ 0x00, /* REG_AVADC_CTL (0x7) */
+ 0x00, /* REG_ADCMICSEL (0x8) */
+ 0x00, /* REG_DIGMIXING (0x9) */
+ 0x0c, /* REG_ATXL1PGA (0xA) */
+ 0x0c, /* REG_ATXR1PGA (0xB) */
+ 0x00, /* REG_AVTXL2PGA (0xC) */
+ 0x00, /* REG_AVTXR2PGA (0xD) */
+ 0x01, /* REG_AUDIO_IF (0xE) */
+ 0x00, /* REG_VOICE_IF (0xF) */
+ 0x00, /* REG_ARXR1PGA (0x10) */
+ 0x00, /* REG_ARXL1PGA (0x11) */
+ 0x6c, /* REG_ARXR2PGA (0x12) */
+ 0x6c, /* REG_ARXL2PGA (0x13) */
+ 0x00, /* REG_VRXPGA (0x14) */
+ 0x00, /* REG_VSTPGA (0x15) */
+ 0x00, /* REG_VRX2ARXPGA (0x16) */
+ 0x0c, /* REG_AVDAC_CTL (0x17) */
+ 0x00, /* REG_ARX2VTXPGA (0x18) */
+ 0x00, /* REG_ARXL1_APGA_CTL (0x19) */
+ 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */
+ 0x4b, /* REG_ARXL2_APGA_CTL (0x1B) */
+ 0x4b, /* REG_ARXR2_APGA_CTL (0x1C) */
+ 0x00, /* REG_ATX2ARXPGA (0x1D) */
+ 0x00, /* REG_BT_IF (0x1E) */
+ 0x00, /* REG_BTPGA (0x1F) */
+ 0x00, /* REG_BTSTPGA (0x20) */
+ 0x00, /* REG_EAR_CTL (0x21) */
+ 0x24, /* REG_HS_SEL (0x22) */
+ 0x0a, /* REG_HS_GAIN_SET (0x23) */
+ 0x00, /* REG_HS_POPN_SET (0x24) */
+ 0x00, /* REG_PREDL_CTL (0x25) */
+ 0x00, /* REG_PREDR_CTL (0x26) */
+ 0x00, /* REG_PRECKL_CTL (0x27) */
+ 0x00, /* REG_PRECKR_CTL (0x28) */
+ 0x00, /* REG_HFL_CTL (0x29) */
+ 0x00, /* REG_HFR_CTL (0x2A) */
+ 0x00, /* REG_ALC_CTL (0x2B) */
+ 0x00, /* REG_ALC_SET1 (0x2C) */
+ 0x00, /* REG_ALC_SET2 (0x2D) */
+ 0x00, /* REG_BOOST_CTL (0x2E) */
+ 0x00, /* REG_SOFTVOL_CTL (0x2F) */
+ 0x00, /* REG_DTMF_FREQSEL (0x30) */
+ 0x00, /* REG_DTMF_TONEXT1H (0x31) */
+ 0x00, /* REG_DTMF_TONEXT1L (0x32) */
+ 0x00, /* REG_DTMF_TONEXT2H (0x33) */
+ 0x00, /* REG_DTMF_TONEXT2L (0x34) */
+ 0x00, /* REG_DTMF_TONOFF (0x35) */
+ 0x00, /* REG_DTMF_WANONOFF (0x36) */
+ 0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */
+ 0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */
+ 0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */
+ 0x16, /* REG_APLL_CTL (0x3A) */
+ 0x00, /* REG_DTMF_CTL (0x3B) */
+ 0x00, /* REG_DTMF_PGA_CTL2 (0x3C) */
+ 0x00, /* REG_DTMF_PGA_CTL1 (0x3D) */
+ 0x00, /* REG_MISC_SET_1 (0x3E) */
+ 0x00, /* REG_PCMBTMUX (0x3F) */
+ 0x00, /* not used (0x40) */
+ 0x00, /* not used (0x41) */
+ 0x00, /* not used (0x42) */
+ 0x00, /* REG_RX_PATH_SEL (0x43) */
+ 0x00, /* REG_VDL_APGA_CTL (0x44) */
+ 0x00, /* REG_VIBRA_CTL (0x45) */
+ 0x00, /* REG_VIBRA_SET (0x46) */
+ 0x00, /* REG_VIBRA_PWM_SET (0x47) */
+ 0x00, /* REG_ANAMIC_GAIN (0x48) */
+ 0x00, /* REG_MISC_SET_2 (0x49) */
+};
+
+/*
+ * read twl4030 register cache
+ */
+static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 *cache = codec->reg_cache;
+
+ return cache[reg];
+}
+
+/*
+ * write twl4030 register cache
+ */
+static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec,
+ u8 reg, u8 value)
+{
+ u8 *cache = codec->reg_cache;
+
+ if (reg >= TWL4030_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the twl4030 register space
+ */
+static int twl4030_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ twl4030_write_reg_cache(codec, reg, value);
+ return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
+}
+
+static void twl4030_clear_codecpdz(struct snd_soc_codec *codec)
+{
+ u8 mode;
+
+ mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
+ twl4030_write(codec, TWL4030_REG_CODEC_MODE,
+ mode & ~TWL4030_CODECPDZ);
+
+ /* REVISIT: this delay is present in TI sample drivers */
+ /* but there seems to be no TRM requirement for it */
+ udelay(10);
+}
+
+static void twl4030_set_codecpdz(struct snd_soc_codec *codec)
+{
+ u8 mode;
+
+ mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
+ twl4030_write(codec, TWL4030_REG_CODEC_MODE,
+ mode | TWL4030_CODECPDZ);
+
+ /* REVISIT: this delay is present in TI sample drivers */
+ /* but there seems to be no TRM requirement for it */
+ udelay(10);
+}
+
+static void twl4030_init_chip(struct snd_soc_codec *codec)
+{
+ int i;
+
+ /* clear CODECPDZ prior to setting register defaults */
+ twl4030_clear_codecpdz(codec);
+
+ /* set all audio section registers to reasonable defaults */
+ for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
+ twl4030_write(codec, i, twl4030_reg[i]);
+
+}
+
+/* Earpiece */
+static const char *twl4030_earpiece_texts[] =
+ {"Off", "DACL1", "DACL2", "DACR1"};
+
+static const unsigned int twl4030_earpiece_values[] =
+ {0x0, 0x1, 0x2, 0x4};
+
+static const struct soc_enum twl4030_earpiece_enum =
+ SOC_VALUE_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, 0x7,
+ ARRAY_SIZE(twl4030_earpiece_texts),
+ twl4030_earpiece_texts,
+ twl4030_earpiece_values);
+
+static const struct snd_kcontrol_new twl4030_dapm_earpiece_control =
+SOC_DAPM_VALUE_ENUM("Route", twl4030_earpiece_enum);
+
+/* PreDrive Left */
+static const char *twl4030_predrivel_texts[] =
+ {"Off", "DACL1", "DACL2", "DACR2"};
+
+static const unsigned int twl4030_predrivel_values[] =
+ {0x0, 0x1, 0x2, 0x4};
+
+static const struct soc_enum twl4030_predrivel_enum =
+ SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, 0x7,
+ ARRAY_SIZE(twl4030_predrivel_texts),
+ twl4030_predrivel_texts,
+ twl4030_predrivel_values);
+
+static const struct snd_kcontrol_new twl4030_dapm_predrivel_control =
+SOC_DAPM_VALUE_ENUM("Route", twl4030_predrivel_enum);
+
+/* PreDrive Right */
+static const char *twl4030_predriver_texts[] =
+ {"Off", "DACR1", "DACR2", "DACL2"};
+
+static const unsigned int twl4030_predriver_values[] =
+ {0x0, 0x1, 0x2, 0x4};
+
+static const struct soc_enum twl4030_predriver_enum =
+ SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7,
+ ARRAY_SIZE(twl4030_predriver_texts),
+ twl4030_predriver_texts,
+ twl4030_predriver_values);
+
+static const struct snd_kcontrol_new twl4030_dapm_predriver_control =
+SOC_DAPM_VALUE_ENUM("Route", twl4030_predriver_enum);
+
+/* Headset Left */
+static const char *twl4030_hsol_texts[] =
+ {"Off", "DACL1", "DACL2"};
+
+static const struct soc_enum twl4030_hsol_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1,
+ ARRAY_SIZE(twl4030_hsol_texts),
+ twl4030_hsol_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_hsol_control =
+SOC_DAPM_ENUM("Route", twl4030_hsol_enum);
+
+/* Headset Right */
+static const char *twl4030_hsor_texts[] =
+ {"Off", "DACR1", "DACR2"};
+
+static const struct soc_enum twl4030_hsor_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4,
+ ARRAY_SIZE(twl4030_hsor_texts),
+ twl4030_hsor_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_hsor_control =
+SOC_DAPM_ENUM("Route", twl4030_hsor_enum);
+
+/* Carkit Left */
+static const char *twl4030_carkitl_texts[] =
+ {"Off", "DACL1", "DACL2"};
+
+static const struct soc_enum twl4030_carkitl_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1,
+ ARRAY_SIZE(twl4030_carkitl_texts),
+ twl4030_carkitl_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_carkitl_control =
+SOC_DAPM_ENUM("Route", twl4030_carkitl_enum);
+
+/* Carkit Right */
+static const char *twl4030_carkitr_texts[] =
+ {"Off", "DACR1", "DACR2"};
+
+static const struct soc_enum twl4030_carkitr_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1,
+ ARRAY_SIZE(twl4030_carkitr_texts),
+ twl4030_carkitr_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_carkitr_control =
+SOC_DAPM_ENUM("Route", twl4030_carkitr_enum);
+
+/* Handsfree Left */
+static const char *twl4030_handsfreel_texts[] =
+ {"Voice", "DACL1", "DACL2", "DACR2"};
+
+static const struct soc_enum twl4030_handsfreel_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0,
+ ARRAY_SIZE(twl4030_handsfreel_texts),
+ twl4030_handsfreel_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control =
+SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum);
+
+/* Handsfree Right */
+static const char *twl4030_handsfreer_texts[] =
+ {"Voice", "DACR1", "DACR2", "DACL2"};
+
+static const struct soc_enum twl4030_handsfreer_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0,
+ ARRAY_SIZE(twl4030_handsfreer_texts),
+ twl4030_handsfreer_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control =
+SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum);
+
+/* Left analog microphone selection */
+static const char *twl4030_analoglmic_texts[] =
+ {"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"};
+
+static const unsigned int twl4030_analoglmic_values[] =
+ {0x0, 0x1, 0x2, 0x4, 0x8};
+
+static const struct soc_enum twl4030_analoglmic_enum =
+ SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0, 0xf,
+ ARRAY_SIZE(twl4030_analoglmic_texts),
+ twl4030_analoglmic_texts,
+ twl4030_analoglmic_values);
+
+static const struct snd_kcontrol_new twl4030_dapm_analoglmic_control =
+SOC_DAPM_VALUE_ENUM("Route", twl4030_analoglmic_enum);
+
+/* Right analog microphone selection */
+static const char *twl4030_analogrmic_texts[] =
+ {"Off", "Sub mic", "AUXR"};
+
+static const unsigned int twl4030_analogrmic_values[] =
+ {0x0, 0x1, 0x4};
+
+static const struct soc_enum twl4030_analogrmic_enum =
+ SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0, 0x5,
+ ARRAY_SIZE(twl4030_analogrmic_texts),
+ twl4030_analogrmic_texts,
+ twl4030_analogrmic_values);
+
+static const struct snd_kcontrol_new twl4030_dapm_analogrmic_control =
+SOC_DAPM_VALUE_ENUM("Route", twl4030_analogrmic_enum);
+
+/* TX1 L/R Analog/Digital microphone selection */
+static const char *twl4030_micpathtx1_texts[] =
+ {"Analog", "Digimic0"};
+
+static const struct soc_enum twl4030_micpathtx1_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 0,
+ ARRAY_SIZE(twl4030_micpathtx1_texts),
+ twl4030_micpathtx1_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_micpathtx1_control =
+SOC_DAPM_ENUM("Route", twl4030_micpathtx1_enum);
+
+/* TX2 L/R Analog/Digital microphone selection */
+static const char *twl4030_micpathtx2_texts[] =
+ {"Analog", "Digimic1"};
+
+static const struct soc_enum twl4030_micpathtx2_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 2,
+ ARRAY_SIZE(twl4030_micpathtx2_texts),
+ twl4030_micpathtx2_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_micpathtx2_control =
+SOC_DAPM_ENUM("Route", twl4030_micpathtx2_enum);
+
+static int micpath_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value;
+ unsigned char adcmicsel, micbias_ctl;
+
+ adcmicsel = twl4030_read_reg_cache(w->codec, TWL4030_REG_ADCMICSEL);
+ micbias_ctl = twl4030_read_reg_cache(w->codec, TWL4030_REG_MICBIAS_CTL);
+ /* Prepare the bits for the given TX path:
+ * shift_l == 0: TX1 microphone path
+ * shift_l == 2: TX2 microphone path */
+ if (e->shift_l) {
+ /* TX2 microphone path */
+ if (adcmicsel & TWL4030_TX2IN_SEL)
+ micbias_ctl |= TWL4030_MICBIAS2_CTL; /* digimic */
+ else
+ micbias_ctl &= ~TWL4030_MICBIAS2_CTL;
+ } else {
+ /* TX1 microphone path */
+ if (adcmicsel & TWL4030_TX1IN_SEL)
+ micbias_ctl |= TWL4030_MICBIAS1_CTL; /* digimic */
+ else
+ micbias_ctl &= ~TWL4030_MICBIAS1_CTL;
+ }
+
+ twl4030_write(w->codec, TWL4030_REG_MICBIAS_CTL, micbias_ctl);
+
+ return 0;
+}
+
+static int handsfree_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value;
+ unsigned char hs_ctl;
+
+ hs_ctl = twl4030_read_reg_cache(w->codec, e->reg);
+
+ if (hs_ctl & TWL4030_HF_CTL_REF_EN) {
+ hs_ctl |= TWL4030_HF_CTL_RAMP_EN;
+ twl4030_write(w->codec, e->reg, hs_ctl);
+ hs_ctl |= TWL4030_HF_CTL_LOOP_EN;
+ twl4030_write(w->codec, e->reg, hs_ctl);
+ hs_ctl |= TWL4030_HF_CTL_HB_EN;
+ twl4030_write(w->codec, e->reg, hs_ctl);
+ } else {
+ hs_ctl &= ~(TWL4030_HF_CTL_RAMP_EN | TWL4030_HF_CTL_LOOP_EN
+ | TWL4030_HF_CTL_HB_EN);
+ twl4030_write(w->codec, e->reg, hs_ctl);
+ }
+
+ return 0;
+}
+
+/*
+ * Some of the gain controls in TWL (mostly those which are associated with
+ * the outputs) are implemented in an interesting way:
+ * 0x0 : Power down (mute)
+ * 0x1 : 6dB
+ * 0x2 : 0 dB
+ * 0x3 : -6 dB
+ * Inverting not going to help with these.
+ * Custom volsw and volsw_2r get/put functions to handle these gain bits.
+ */
+#define SOC_DOUBLE_TLV_TWL4030(xname, xreg, shift_left, shift_right, xmax,\
+ xinvert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_get_volsw_twl4030, \
+ .put = snd_soc_put_volsw_twl4030, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .shift = shift_left, .rshift = shift_right,\
+ .max = xmax, .invert = xinvert} }
+#define SOC_DOUBLE_R_TLV_TWL4030(xname, reg_left, reg_right, xshift, xmax,\
+ xinvert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw_2r, \
+ .get = snd_soc_get_volsw_r2_twl4030,\
+ .put = snd_soc_put_volsw_r2_twl4030, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+ .rshift = xshift, .max = xmax, .invert = xinvert} }
+#define SOC_SINGLE_TLV_TWL4030(xname, xreg, xshift, xmax, xinvert, tlv_array) \
+ SOC_DOUBLE_TLV_TWL4030(xname, xreg, xshift, xshift, xmax, \
+ xinvert, tlv_array)
+
+static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ unsigned int rshift = mc->rshift;
+ int max = mc->max;
+ int mask = (1 << fls(max)) - 1;
+
+ ucontrol->value.integer.value[0] =
+ (snd_soc_read(codec, reg) >> shift) & mask;
+ if (ucontrol->value.integer.value[0])
+ ucontrol->value.integer.value[0] =
+ max + 1 - ucontrol->value.integer.value[0];
+
+ if (shift != rshift) {
+ ucontrol->value.integer.value[1] =
+ (snd_soc_read(codec, reg) >> rshift) & mask;
+ if (ucontrol->value.integer.value[1])
+ ucontrol->value.integer.value[1] =
+ max + 1 - ucontrol->value.integer.value[1];
+ }
+
+ return 0;
+}
+
+static int snd_soc_put_volsw_twl4030(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ unsigned int rshift = mc->rshift;
+ int max = mc->max;
+ int mask = (1 << fls(max)) - 1;
+ unsigned short val, val2, val_mask;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+
+ val_mask = mask << shift;
+ if (val)
+ val = max + 1 - val;
+ val = val << shift;
+ if (shift != rshift) {
+ val2 = (ucontrol->value.integer.value[1] & mask);
+ val_mask |= mask << rshift;
+ if (val2)
+ val2 = max + 1 - val2;
+ val |= val2 << rshift;
+ }
+ return snd_soc_update_bits(codec, reg, val_mask, val);
+}
+
+static int snd_soc_get_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
+ unsigned int shift = mc->shift;
+ int max = mc->max;
+ int mask = (1<<fls(max))-1;
+
+ ucontrol->value.integer.value[0] =
+ (snd_soc_read(codec, reg) >> shift) & mask;
+ ucontrol->value.integer.value[1] =
+ (snd_soc_read(codec, reg2) >> shift) & mask;
+
+ if (ucontrol->value.integer.value[0])
+ ucontrol->value.integer.value[0] =
+ max + 1 - ucontrol->value.integer.value[0];
+ if (ucontrol->value.integer.value[1])
+ ucontrol->value.integer.value[1] =
+ max + 1 - ucontrol->value.integer.value[1];
+
+ return 0;
+}
+
+static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
+ unsigned int shift = mc->shift;
+ int max = mc->max;
+ int mask = (1 << fls(max)) - 1;
+ int err;
+ unsigned short val, val2, val_mask;
+
+ val_mask = mask << shift;
+ val = (ucontrol->value.integer.value[0] & mask);
+ val2 = (ucontrol->value.integer.value[1] & mask);
+
+ if (val)
+ val = max + 1 - val;
+ if (val2)
+ val2 = max + 1 - val2;
+
+ val = val << shift;
+ val2 = val2 << shift;
+
+ err = snd_soc_update_bits(codec, reg, val_mask, val);
+ if (err < 0)
+ return err;
+
+ err = snd_soc_update_bits(codec, reg2, val_mask, val2);
+ return err;
+}
+
+/*
+ * FGAIN volume control:
+ * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB)
+ */
+static DECLARE_TLV_DB_SCALE(digital_fine_tlv, -6300, 100, 1);
+
+/*
+ * CGAIN volume control:
+ * 0 dB to 12 dB in 6 dB steps
+ * value 2 and 3 means 12 dB
+ */
+static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0);
+
+/*
+ * Analog playback gain
+ * -24 dB to 12 dB in 2 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
+
+/*
+ * Gain controls tied to outputs
+ * -6 dB to 6 dB in 6 dB steps (mute instead of -12)
+ */
+static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
+
+/*
+ * Capture gain after the ADCs
+ * from 0 dB to 31 dB in 1 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0);
+
+/*
+ * Gain control for input amplifiers
+ * 0 dB to 30 dB in 6 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new twl4030_snd_controls[] = {
+ /* Common playback gain controls */
+ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume",
+ TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA,
+ 0, 0x3f, 0, digital_fine_tlv),
+ SOC_DOUBLE_R_TLV("DAC2 Digital Fine Playback Volume",
+ TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA,
+ 0, 0x3f, 0, digital_fine_tlv),
+
+ SOC_DOUBLE_R_TLV("DAC1 Digital Coarse Playback Volume",
+ TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA,
+ 6, 0x2, 0, digital_coarse_tlv),
+ SOC_DOUBLE_R_TLV("DAC2 Digital Coarse Playback Volume",
+ TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA,
+ 6, 0x2, 0, digital_coarse_tlv),
+
+ SOC_DOUBLE_R_TLV("DAC1 Analog Playback Volume",
+ TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL,
+ 3, 0x12, 1, analog_tlv),
+ SOC_DOUBLE_R_TLV("DAC2 Analog Playback Volume",
+ TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
+ 3, 0x12, 1, analog_tlv),
+ SOC_DOUBLE_R("DAC1 Analog Playback Switch",
+ TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL,
+ 1, 1, 0),
+ SOC_DOUBLE_R("DAC2 Analog Playback Switch",
+ TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
+ 1, 1, 0),
+
+ /* Separate output gain controls */
+ SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume",
+ TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL,
+ 4, 3, 0, output_tvl),
+
+ SOC_DOUBLE_TLV_TWL4030("Headset Playback Volume",
+ TWL4030_REG_HS_GAIN_SET, 0, 2, 3, 0, output_tvl),
+
+ SOC_DOUBLE_R_TLV_TWL4030("Carkit Playback Volume",
+ TWL4030_REG_PRECKL_CTL, TWL4030_REG_PRECKR_CTL,
+ 4, 3, 0, output_tvl),
+
+ SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
+ TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+
+ /* Common capture gain controls */
+ SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
+ TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA,
+ 0, 0x1f, 0, digital_capture_tlv),
+ SOC_DOUBLE_R_TLV("TX2 Digital Capture Volume",
+ TWL4030_REG_AVTXL2PGA, TWL4030_REG_AVTXR2PGA,
+ 0, 0x1f, 0, digital_capture_tlv),
+
+ SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN,
+ 0, 3, 5, 0, input_gain_tlv),
+};
+
+/* add non dapm controls */
+static int twl4030_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&twl4030_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
+ /* Left channel inputs */
+ SND_SOC_DAPM_INPUT("MAINMIC"),
+ SND_SOC_DAPM_INPUT("HSMIC"),
+ SND_SOC_DAPM_INPUT("AUXL"),
+ SND_SOC_DAPM_INPUT("CARKITMIC"),
+ /* Right channel inputs */
+ SND_SOC_DAPM_INPUT("SUBMIC"),
+ SND_SOC_DAPM_INPUT("AUXR"),
+ /* Digital microphones (Stereo) */
+ SND_SOC_DAPM_INPUT("DIGIMIC0"),
+ SND_SOC_DAPM_INPUT("DIGIMIC1"),
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("OUTL"),
+ SND_SOC_DAPM_OUTPUT("OUTR"),
+ SND_SOC_DAPM_OUTPUT("EARPIECE"),
+ SND_SOC_DAPM_OUTPUT("PREDRIVEL"),
+ SND_SOC_DAPM_OUTPUT("PREDRIVER"),
+ SND_SOC_DAPM_OUTPUT("HSOL"),
+ SND_SOC_DAPM_OUTPUT("HSOR"),
+ SND_SOC_DAPM_OUTPUT("CARKITL"),
+ SND_SOC_DAPM_OUTPUT("CARKITR"),
+ SND_SOC_DAPM_OUTPUT("HFL"),
+ SND_SOC_DAPM_OUTPUT("HFR"),
+
+ /* DACs */
+ SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback",
+ TWL4030_REG_AVDAC_CTL, 0, 0),
+ SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback",
+ TWL4030_REG_AVDAC_CTL, 1, 0),
+ SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback",
+ TWL4030_REG_AVDAC_CTL, 2, 0),
+ SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback",
+ TWL4030_REG_AVDAC_CTL, 3, 0),
+
+ /* Analog PGAs */
+ SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("ARXR2_APGA", TWL4030_REG_ARXR2_APGA_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL,
+ 0, 0, NULL, 0),
+
+ /* Output MUX controls */
+ /* Earpiece */
+ SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_earpiece_control),
+ /* PreDrivL/R */
+ SND_SOC_DAPM_VALUE_MUX("PredriveL Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_predrivel_control),
+ SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_predriver_control),
+ /* HeadsetL/R */
+ SND_SOC_DAPM_MUX("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_hsol_control),
+ SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_hsor_control),
+ /* CarkitL/R */
+ SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_carkitl_control),
+ SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_carkitr_control),
+ /* HandsfreeL/R */
+ SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0,
+ &twl4030_dapm_handsfreel_control, handsfree_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0,
+ &twl4030_dapm_handsfreer_control, handsfree_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+
+ /* Introducing four virtual ADC, since TWL4030 have four channel for
+ capture */
+ SND_SOC_DAPM_ADC("ADC Virtual Left1", "Left Front Capture",
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_ADC("ADC Virtual Right1", "Right Front Capture",
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_ADC("ADC Virtual Left2", "Left Rear Capture",
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_ADC("ADC Virtual Right2", "Right Rear Capture",
+ SND_SOC_NOPM, 0, 0),
+
+ /* Analog/Digital mic path selection.
+ TX1 Left/Right: either analog Left/Right or Digimic0
+ TX2 Left/Right: either analog Left/Right or Digimic1 */
+ SND_SOC_DAPM_MUX_E("TX1 Capture Route", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_micpathtx1_control, micpath_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
+ SND_SOC_DAPM_POST_REG),
+ SND_SOC_DAPM_MUX_E("TX2 Capture Route", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_micpathtx2_control, micpath_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
+ SND_SOC_DAPM_POST_REG),
+
+ /* Analog input muxes with power switch for the physical ADCL/R */
+ SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route",
+ TWL4030_REG_AVADC_CTL, 3, 0, &twl4030_dapm_analoglmic_control),
+ SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route",
+ TWL4030_REG_AVADC_CTL, 1, 0, &twl4030_dapm_analogrmic_control),
+
+ SND_SOC_DAPM_PGA("Analog Left Amplifier",
+ TWL4030_REG_ANAMICL, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Analog Right Amplifier",
+ TWL4030_REG_ANAMICR, 4, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA("Digimic0 Enable",
+ TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Digimic1 Enable",
+ TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0),
+
+ SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0),
+ SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0),
+ SND_SOC_DAPM_MICBIAS("Headset Mic Bias", TWL4030_REG_MICBIAS_CTL, 2, 0),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"ARXL1_APGA", NULL, "DAC Left1"},
+ {"ARXR1_APGA", NULL, "DAC Right1"},
+ {"ARXL2_APGA", NULL, "DAC Left2"},
+ {"ARXR2_APGA", NULL, "DAC Right2"},
+
+ /* Internal playback routings */
+ /* Earpiece */
+ {"Earpiece Mux", "DACL1", "ARXL1_APGA"},
+ {"Earpiece Mux", "DACL2", "ARXL2_APGA"},
+ {"Earpiece Mux", "DACR1", "ARXR1_APGA"},
+ /* PreDrivL */
+ {"PredriveL Mux", "DACL1", "ARXL1_APGA"},
+ {"PredriveL Mux", "DACL2", "ARXL2_APGA"},
+ {"PredriveL Mux", "DACR2", "ARXR2_APGA"},
+ /* PreDrivR */
+ {"PredriveR Mux", "DACR1", "ARXR1_APGA"},
+ {"PredriveR Mux", "DACR2", "ARXR2_APGA"},
+ {"PredriveR Mux", "DACL2", "ARXL2_APGA"},
+ /* HeadsetL */
+ {"HeadsetL Mux", "DACL1", "ARXL1_APGA"},
+ {"HeadsetL Mux", "DACL2", "ARXL2_APGA"},
+ /* HeadsetR */
+ {"HeadsetR Mux", "DACR1", "ARXR1_APGA"},
+ {"HeadsetR Mux", "DACR2", "ARXR2_APGA"},
+ /* CarkitL */
+ {"CarkitL Mux", "DACL1", "ARXL1_APGA"},
+ {"CarkitL Mux", "DACL2", "ARXL2_APGA"},
+ /* CarkitR */
+ {"CarkitR Mux", "DACR1", "ARXR1_APGA"},
+ {"CarkitR Mux", "DACR2", "ARXR2_APGA"},
+ /* HandsfreeL */
+ {"HandsfreeL Mux", "DACL1", "ARXL1_APGA"},
+ {"HandsfreeL Mux", "DACL2", "ARXL2_APGA"},
+ {"HandsfreeL Mux", "DACR2", "ARXR2_APGA"},
+ /* HandsfreeR */
+ {"HandsfreeR Mux", "DACR1", "ARXR1_APGA"},
+ {"HandsfreeR Mux", "DACR2", "ARXR2_APGA"},
+ {"HandsfreeR Mux", "DACL2", "ARXL2_APGA"},
+
+ /* outputs */
+ {"OUTL", NULL, "ARXL2_APGA"},
+ {"OUTR", NULL, "ARXR2_APGA"},
+ {"EARPIECE", NULL, "Earpiece Mux"},
+ {"PREDRIVEL", NULL, "PredriveL Mux"},
+ {"PREDRIVER", NULL, "PredriveR Mux"},
+ {"HSOL", NULL, "HeadsetL Mux"},
+ {"HSOR", NULL, "HeadsetR Mux"},
+ {"CARKITL", NULL, "CarkitL Mux"},
+ {"CARKITR", NULL, "CarkitR Mux"},
+ {"HFL", NULL, "HandsfreeL Mux"},
+ {"HFR", NULL, "HandsfreeR Mux"},
+
+ /* Capture path */
+ {"Analog Left Capture Route", "Main mic", "MAINMIC"},
+ {"Analog Left Capture Route", "Headset mic", "HSMIC"},
+ {"Analog Left Capture Route", "AUXL", "AUXL"},
+ {"Analog Left Capture Route", "Carkit mic", "CARKITMIC"},
+
+ {"Analog Right Capture Route", "Sub mic", "SUBMIC"},
+ {"Analog Right Capture Route", "AUXR", "AUXR"},
+
+ {"Analog Left Amplifier", NULL, "Analog Left Capture Route"},
+ {"Analog Right Amplifier", NULL, "Analog Right Capture Route"},
+
+ {"Digimic0 Enable", NULL, "DIGIMIC0"},
+ {"Digimic1 Enable", NULL, "DIGIMIC1"},
+
+ /* TX1 Left capture path */
+ {"TX1 Capture Route", "Analog", "Analog Left Amplifier"},
+ {"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
+ /* TX1 Right capture path */
+ {"TX1 Capture Route", "Analog", "Analog Right Amplifier"},
+ {"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
+ /* TX2 Left capture path */
+ {"TX2 Capture Route", "Analog", "Analog Left Amplifier"},
+ {"TX2 Capture Route", "Digimic1", "Digimic1 Enable"},
+ /* TX2 Right capture path */
+ {"TX2 Capture Route", "Analog", "Analog Right Amplifier"},
+ {"TX2 Capture Route", "Digimic1", "Digimic1 Enable"},
+
+ {"ADC Virtual Left1", NULL, "TX1 Capture Route"},
+ {"ADC Virtual Right1", NULL, "TX1 Capture Route"},
+ {"ADC Virtual Left2", NULL, "TX2 Capture Route"},
+ {"ADC Virtual Right2", NULL, "TX2 Capture Route"},
+
+};
+
+static int twl4030_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, twl4030_dapm_widgets,
+ ARRAY_SIZE(twl4030_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static void twl4030_power_up(struct snd_soc_codec *codec)
+{
+ u8 anamicl, regmisc1, byte, popn;
+ int i = 0;
+
+ /* set CODECPDZ to turn on codec */
+ twl4030_set_codecpdz(codec);
+
+ /* initiate offset cancellation */
+ anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+ twl4030_write(codec, TWL4030_REG_ANAMICL,
+ anamicl | TWL4030_CNCL_OFFSET_START);
+
+
+ /* wait for offset cancellation to complete */
+ do {
+ /* this takes a little while, so don't slam i2c */
+ udelay(2000);
+ twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
+ TWL4030_REG_ANAMICL);
+ } while ((i++ < 100) &&
+ ((byte & TWL4030_CNCL_OFFSET_START) ==
+ TWL4030_CNCL_OFFSET_START));
+
+ /* anti-pop when changing analog gain */
+ regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
+ twl4030_write(codec, TWL4030_REG_MISC_SET_1,
+ regmisc1 | TWL4030_SMOOTH_ANAVOL_EN);
+
+ /* toggle CODECPDZ as per TRM */
+ twl4030_clear_codecpdz(codec);
+ twl4030_set_codecpdz(codec);
+
+ /* program anti-pop with bias ramp delay */
+ popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+ popn &= TWL4030_RAMP_DELAY;
+ popn |= TWL4030_RAMP_DELAY_645MS;
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+ popn |= TWL4030_VMID_EN;
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+
+ /* enable anti-pop ramp */
+ popn |= TWL4030_RAMP_EN;
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+}
+
+static void twl4030_power_down(struct snd_soc_codec *codec)
+{
+ u8 popn;
+
+ /* disable anti-pop ramp */
+ popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+ popn &= ~TWL4030_RAMP_EN;
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+
+ /* disable bias out */
+ popn &= ~TWL4030_VMID_EN;
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+
+ /* power down */
+ twl4030_clear_codecpdz(codec);
+}
+
+static int twl4030_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ twl4030_power_up(codec);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ /* TODO: develop a twl4030_prepare function */
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* TODO: develop a twl4030_standby function */
+ twl4030_power_down(codec);
+ break;
+ case SND_SOC_BIAS_OFF:
+ twl4030_power_down(codec);
+ break;
+ }
+ codec->bias_level = level;
+
+ return 0;
+}
+
+static int twl4030_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u8 mode, old_mode, format, old_format;
+
+
+ /* bit rate */
+ old_mode = twl4030_read_reg_cache(codec,
+ TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
+ mode = old_mode & ~TWL4030_APLL_RATE;
+
+ switch (params_rate(params)) {
+ case 8000:
+ mode |= TWL4030_APLL_RATE_8000;
+ break;
+ case 11025:
+ mode |= TWL4030_APLL_RATE_11025;
+ break;
+ case 12000:
+ mode |= TWL4030_APLL_RATE_12000;
+ break;
+ case 16000:
+ mode |= TWL4030_APLL_RATE_16000;
+ break;
+ case 22050:
+ mode |= TWL4030_APLL_RATE_22050;
+ break;
+ case 24000:
+ mode |= TWL4030_APLL_RATE_24000;
+ break;
+ case 32000:
+ mode |= TWL4030_APLL_RATE_32000;
+ break;
+ case 44100:
+ mode |= TWL4030_APLL_RATE_44100;
+ break;
+ case 48000:
+ mode |= TWL4030_APLL_RATE_48000;
+ break;
+ default:
+ printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n",
+ params_rate(params));
+ return -EINVAL;
+ }
+
+ if (mode != old_mode) {
+ /* change rate and set CODECPDZ */
+ twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+ twl4030_set_codecpdz(codec);
+ }
+
+ /* sample size */
+ old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+ format = old_format;
+ format &= ~TWL4030_DATA_WIDTH;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ format |= TWL4030_DATA_WIDTH_16S_16W;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ format |= TWL4030_DATA_WIDTH_32S_24W;
+ break;
+ default:
+ printk(KERN_ERR "TWL4030 hw params: unknown format %d\n",
+ params_format(params));
+ return -EINVAL;
+ }
+
+ if (format != old_format) {
+
+ /* clear CODECPDZ before changing format (codec requirement) */
+ twl4030_clear_codecpdz(codec);
+
+ /* change format */
+ twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+
+ /* set CODECPDZ afterwards */
+ twl4030_set_codecpdz(codec);
+ }
+ return 0;
+}
+
+static int twl4030_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;
+ u8 infreq;
+
+ switch (freq) {
+ case 19200000:
+ infreq = TWL4030_APLL_INFREQ_19200KHZ;
+ break;
+ case 26000000:
+ infreq = TWL4030_APLL_INFREQ_26000KHZ;
+ break;
+ case 38400000:
+ infreq = TWL4030_APLL_INFREQ_38400KHZ;
+ break;
+ default:
+ printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n",
+ freq);
+ return -EINVAL;
+ }
+
+ infreq |= TWL4030_APLL_EN;
+ twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq);
+
+ return 0;
+}
+
+static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 old_format, format;
+
+ /* get format */
+ old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+ format = old_format;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ format &= ~(TWL4030_AIF_SLAVE_EN);
+ format &= ~(TWL4030_CLK256FS_EN);
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ format |= TWL4030_AIF_SLAVE_EN;
+ format |= TWL4030_CLK256FS_EN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ format &= ~TWL4030_AIF_FORMAT;
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ format |= TWL4030_AIF_FORMAT_CODEC;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (format != old_format) {
+
+ /* clear CODECPDZ before changing format (codec requirement) */
+ twl4030_clear_codecpdz(codec);
+
+ /* change format */
+ twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+
+ /* set CODECPDZ afterwards */
+ twl4030_set_codecpdz(codec);
+ }
+
+ return 0;
+}
+
+#define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000)
+#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
+
+struct snd_soc_dai twl4030_dai = {
+ .name = "twl4030",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = TWL4030_RATES,
+ .formats = TWL4030_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = TWL4030_RATES,
+ .formats = TWL4030_FORMATS,},
+ .ops = {
+ .hw_params = twl4030_hw_params,
+ .set_sysclk = twl4030_set_dai_sysclk,
+ .set_fmt = twl4030_set_dai_fmt,
+ }
+};
+EXPORT_SYMBOL_GPL(twl4030_dai);
+
+static int twl4030_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int twl4030_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ twl4030_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+/*
+ * initialize the driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+
+static int twl4030_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+
+ printk(KERN_INFO "TWL4030 Audio Codec init \n");
+
+ codec->name = "twl4030";
+ codec->owner = THIS_MODULE;
+ codec->read = twl4030_read_reg_cache;
+ codec->write = twl4030_write;
+ codec->set_bias_level = twl4030_set_bias_level;
+ codec->dai = &twl4030_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = sizeof(twl4030_reg);
+ codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "twl4030: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ twl4030_init_chip(codec);
+
+ /* power on device */
+ twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ twl4030_add_controls(codec);
+ twl4030_add_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "twl4030: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+static struct snd_soc_device *twl4030_socdev;
+
+static int twl4030_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ twl4030_socdev = socdev;
+ twl4030_init(socdev);
+
+ return 0;
+}
+
+static int twl4030_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ printk(KERN_INFO "TWL4030 Audio Codec remove\n");
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_twl4030 = {
+ .probe = twl4030_probe,
+ .remove = twl4030_remove,
+ .suspend = twl4030_suspend,
+ .resume = twl4030_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030);
+
+static int __init twl4030_modinit(void)
+{
+ return snd_soc_register_dai(&twl4030_dai);
+}
+module_init(twl4030_modinit);
+
+static void __exit twl4030_exit(void)
+{
+ snd_soc_unregister_dai(&twl4030_dai);
+}
+module_exit(twl4030_exit);
+
+MODULE_DESCRIPTION("ASoC TWL4030 codec driver");
+MODULE_AUTHOR("Steve Sakoman");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
new file mode 100644
index 00000000000..442e5a82861
--- /dev/null
+++ b/sound/soc/codecs/twl4030.h
@@ -0,0 +1,226 @@
+/*
+ * ALSA SoC TWL4030 codec driver
+ *
+ * Author: Steve Sakoman <steve@sakoman.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __TWL4030_AUDIO_H__
+#define __TWL4030_AUDIO_H__
+
+#define TWL4030_REG_CODEC_MODE 0x1
+#define TWL4030_REG_OPTION 0x2
+#define TWL4030_REG_UNKNOWN 0x3
+#define TWL4030_REG_MICBIAS_CTL 0x4
+#define TWL4030_REG_ANAMICL 0x5
+#define TWL4030_REG_ANAMICR 0x6
+#define TWL4030_REG_AVADC_CTL 0x7
+#define TWL4030_REG_ADCMICSEL 0x8
+#define TWL4030_REG_DIGMIXING 0x9
+#define TWL4030_REG_ATXL1PGA 0xA
+#define TWL4030_REG_ATXR1PGA 0xB
+#define TWL4030_REG_AVTXL2PGA 0xC
+#define TWL4030_REG_AVTXR2PGA 0xD
+#define TWL4030_REG_AUDIO_IF 0xE
+#define TWL4030_REG_VOICE_IF 0xF
+#define TWL4030_REG_ARXR1PGA 0x10
+#define TWL4030_REG_ARXL1PGA 0x11
+#define TWL4030_REG_ARXR2PGA 0x12
+#define TWL4030_REG_ARXL2PGA 0x13
+#define TWL4030_REG_VRXPGA 0x14
+#define TWL4030_REG_VSTPGA 0x15
+#define TWL4030_REG_VRX2ARXPGA 0x16
+#define TWL4030_REG_AVDAC_CTL 0x17
+#define TWL4030_REG_ARX2VTXPGA 0x18
+#define TWL4030_REG_ARXL1_APGA_CTL 0x19
+#define TWL4030_REG_ARXR1_APGA_CTL 0x1A
+#define TWL4030_REG_ARXL2_APGA_CTL 0x1B
+#define TWL4030_REG_ARXR2_APGA_CTL 0x1C
+#define TWL4030_REG_ATX2ARXPGA 0x1D
+#define TWL4030_REG_BT_IF 0x1E
+#define TWL4030_REG_BTPGA 0x1F
+#define TWL4030_REG_BTSTPGA 0x20
+#define TWL4030_REG_EAR_CTL 0x21
+#define TWL4030_REG_HS_SEL 0x22
+#define TWL4030_REG_HS_GAIN_SET 0x23
+#define TWL4030_REG_HS_POPN_SET 0x24
+#define TWL4030_REG_PREDL_CTL 0x25
+#define TWL4030_REG_PREDR_CTL 0x26
+#define TWL4030_REG_PRECKL_CTL 0x27
+#define TWL4030_REG_PRECKR_CTL 0x28
+#define TWL4030_REG_HFL_CTL 0x29
+#define TWL4030_REG_HFR_CTL 0x2A
+#define TWL4030_REG_ALC_CTL 0x2B
+#define TWL4030_REG_ALC_SET1 0x2C
+#define TWL4030_REG_ALC_SET2 0x2D
+#define TWL4030_REG_BOOST_CTL 0x2E
+#define TWL4030_REG_SOFTVOL_CTL 0x2F
+#define TWL4030_REG_DTMF_FREQSEL 0x30
+#define TWL4030_REG_DTMF_TONEXT1H 0x31
+#define TWL4030_REG_DTMF_TONEXT1L 0x32
+#define TWL4030_REG_DTMF_TONEXT2H 0x33
+#define TWL4030_REG_DTMF_TONEXT2L 0x34
+#define TWL4030_REG_DTMF_TONOFF 0x35
+#define TWL4030_REG_DTMF_WANONOFF 0x36
+#define TWL4030_REG_I2S_RX_SCRAMBLE_H 0x37
+#define TWL4030_REG_I2S_RX_SCRAMBLE_M 0x38
+#define TWL4030_REG_I2S_RX_SCRAMBLE_L 0x39
+#define TWL4030_REG_APLL_CTL 0x3A
+#define TWL4030_REG_DTMF_CTL 0x3B
+#define TWL4030_REG_DTMF_PGA_CTL2 0x3C
+#define TWL4030_REG_DTMF_PGA_CTL1 0x3D
+#define TWL4030_REG_MISC_SET_1 0x3E
+#define TWL4030_REG_PCMBTMUX 0x3F
+#define TWL4030_REG_RX_PATH_SEL 0x43
+#define TWL4030_REG_VDL_APGA_CTL 0x44
+#define TWL4030_REG_VIBRA_CTL 0x45
+#define TWL4030_REG_VIBRA_SET 0x46
+#define TWL4030_REG_VIBRA_PWM_SET 0x47
+#define TWL4030_REG_ANAMIC_GAIN 0x48
+#define TWL4030_REG_MISC_SET_2 0x49
+
+#define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1)
+
+/* Bitfield Definitions */
+
+/* TWL4030_CODEC_MODE (0x01) Fields */
+
+#define TWL4030_APLL_RATE 0xF0
+#define TWL4030_APLL_RATE_8000 0x00
+#define TWL4030_APLL_RATE_11025 0x10
+#define TWL4030_APLL_RATE_12000 0x20
+#define TWL4030_APLL_RATE_16000 0x40
+#define TWL4030_APLL_RATE_22050 0x50
+#define TWL4030_APLL_RATE_24000 0x60
+#define TWL4030_APLL_RATE_32000 0x80
+#define TWL4030_APLL_RATE_44100 0x90
+#define TWL4030_APLL_RATE_48000 0xA0
+#define TWL4030_SEL_16K 0x04
+#define TWL4030_CODECPDZ 0x02
+#define TWL4030_OPT_MODE 0x01
+
+/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */
+
+#define TWL4030_MICBIAS2_CTL 0x40
+#define TWL4030_MICBIAS1_CTL 0x20
+#define TWL4030_HSMICBIAS_EN 0x04
+#define TWL4030_MICBIAS2_EN 0x02
+#define TWL4030_MICBIAS1_EN 0x01
+
+/* ANAMICL (0x05) Fields */
+
+#define TWL4030_CNCL_OFFSET_START 0x80
+#define TWL4030_OFFSET_CNCL_SEL 0x60
+#define TWL4030_OFFSET_CNCL_SEL_ARX1 0x00
+#define TWL4030_OFFSET_CNCL_SEL_ARX2 0x20
+#define TWL4030_OFFSET_CNCL_SEL_VRX 0x40
+#define TWL4030_OFFSET_CNCL_SEL_ALL 0x60
+#define TWL4030_MICAMPL_EN 0x10
+#define TWL4030_CKMIC_EN 0x08
+#define TWL4030_AUXL_EN 0x04
+#define TWL4030_HSMIC_EN 0x02
+#define TWL4030_MAINMIC_EN 0x01
+
+/* ANAMICR (0x06) Fields */
+
+#define TWL4030_MICAMPR_EN 0x10
+#define TWL4030_AUXR_EN 0x04
+#define TWL4030_SUBMIC_EN 0x01
+
+/* AVADC_CTL (0x07) Fields */
+
+#define TWL4030_ADCL_EN 0x08
+#define TWL4030_AVADC_CLK_PRIORITY 0x04
+#define TWL4030_ADCR_EN 0x02
+
+/* TWL4030_REG_ADCMICSEL (0x08) Fields */
+
+#define TWL4030_DIGMIC1_EN 0x08
+#define TWL4030_TX2IN_SEL 0x04
+#define TWL4030_DIGMIC0_EN 0x02
+#define TWL4030_TX1IN_SEL 0x01
+
+/* AUDIO_IF (0x0E) Fields */
+
+#define TWL4030_AIF_SLAVE_EN 0x80
+#define TWL4030_DATA_WIDTH 0x60
+#define TWL4030_DATA_WIDTH_16S_16W 0x00
+#define TWL4030_DATA_WIDTH_32S_16W 0x40
+#define TWL4030_DATA_WIDTH_32S_24W 0x60
+#define TWL4030_AIF_FORMAT 0x18
+#define TWL4030_AIF_FORMAT_CODEC 0x00
+#define TWL4030_AIF_FORMAT_LEFT 0x08
+#define TWL4030_AIF_FORMAT_RIGHT 0x10
+#define TWL4030_AIF_FORMAT_TDM 0x18
+#define TWL4030_AIF_TRI_EN 0x04
+#define TWL4030_CLK256FS_EN 0x02
+#define TWL4030_AIF_EN 0x01
+
+/* HS_GAIN_SET (0x23) Fields */
+
+#define TWL4030_HSR_GAIN 0x0C
+#define TWL4030_HSR_GAIN_PWR_DOWN 0x00
+#define TWL4030_HSR_GAIN_PLUS_6DB 0x04
+#define TWL4030_HSR_GAIN_0DB 0x08
+#define TWL4030_HSR_GAIN_MINUS_6DB 0x0C
+#define TWL4030_HSL_GAIN 0x03
+#define TWL4030_HSL_GAIN_PWR_DOWN 0x00
+#define TWL4030_HSL_GAIN_PLUS_6DB 0x01
+#define TWL4030_HSL_GAIN_0DB 0x02
+#define TWL4030_HSL_GAIN_MINUS_6DB 0x03
+
+/* HS_POPN_SET (0x24) Fields */
+
+#define TWL4030_VMID_EN 0x40
+#define TWL4030_EXTMUTE 0x20
+#define TWL4030_RAMP_DELAY 0x1C
+#define TWL4030_RAMP_DELAY_20MS 0x00
+#define TWL4030_RAMP_DELAY_40MS 0x04
+#define TWL4030_RAMP_DELAY_81MS 0x08
+#define TWL4030_RAMP_DELAY_161MS 0x0C
+#define TWL4030_RAMP_DELAY_323MS 0x10
+#define TWL4030_RAMP_DELAY_645MS 0x14
+#define TWL4030_RAMP_DELAY_1291MS 0x18
+#define TWL4030_RAMP_DELAY_2581MS 0x1C
+#define TWL4030_RAMP_EN 0x02
+
+/* HFL_CTL (0x29, 0x2A) Fields */
+#define TWL4030_HF_CTL_HB_EN 0x04
+#define TWL4030_HF_CTL_LOOP_EN 0x08
+#define TWL4030_HF_CTL_RAMP_EN 0x10
+#define TWL4030_HF_CTL_REF_EN 0x20
+
+/* APLL_CTL (0x3A) Fields */
+
+#define TWL4030_APLL_EN 0x10
+#define TWL4030_APLL_INFREQ 0x0F
+#define TWL4030_APLL_INFREQ_19200KHZ 0x05
+#define TWL4030_APLL_INFREQ_26000KHZ 0x06
+#define TWL4030_APLL_INFREQ_38400KHZ 0x0F
+
+/* REG_MISC_SET_1 (0x3E) Fields */
+
+#define TWL4030_CLK64_EN 0x80
+#define TWL4030_SCRAMBLE_EN 0x40
+#define TWL4030_FMLOOP_EN 0x20
+#define TWL4030_SMOOTH_ANAVOL_EN 0x02
+#define TWL4030_DIGMIC_LR_SWAP_EN 0x01
+
+extern struct snd_soc_dai twl4030_dai;
+extern struct snd_soc_codec_device soc_codec_dev_twl4030;
+
+#endif /* End of __TWL4030_AUDIO_H__ */
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
new file mode 100644
index 00000000000..a2c5064a774
--- /dev/null
+++ b/sound/soc/codecs/uda134x.c
@@ -0,0 +1,668 @@
+/*
+ * uda134x.c -- UDA134X ALSA SoC Codec driver
+ *
+ * Modifications by Christian Pellegrin <chripell@evolware.org>
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * Based on the WM87xx drivers by Liam Girdwood and 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/delay.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include <sound/uda134x.h>
+#include <sound/l3.h>
+
+#include "uda134x.h"
+
+
+#define POWER_OFF_ON_STANDBY 1
+/*
+ ALSA SOC usually puts the device in standby mode when it's not used
+ for sometime. If you define POWER_OFF_ON_STANDBY the driver will
+ turn off the ADC/DAC when this callback is invoked and turn it back
+ on when needed. Unfortunately this will result in a very light bump
+ (it can be audible only with good earphones). If this bothers you
+ just comment this line, you will have slightly higher power
+ consumption . Please note that sending the L3 command for ADC is
+ enough to make the bump, so it doesn't make difference if you
+ completely take off power from the codec.
+ */
+
+#define UDA134X_RATES SNDRV_PCM_RATE_8000_48000
+#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE)
+
+struct uda134x_priv {
+ int sysclk;
+ int dai_fmt;
+
+ struct snd_pcm_substream *master_substream;
+ struct snd_pcm_substream *slave_substream;
+};
+
+/* In-data addresses are hard-coded into the reg-cache values */
+static const char uda134x_reg[UDA134X_REGS_NUM] = {
+ /* Extended address registers */
+ 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00,
+ /* Status, data regs */
+ 0x00, 0x83, 0x00, 0x40, 0x80, 0x00,
+};
+
+/*
+ * The codec has no support for reading its registers except for peak level...
+ */
+static inline unsigned int uda134x_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 *cache = codec->reg_cache;
+
+ if (reg >= UDA134X_REGS_NUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * Write the register cache
+ */
+static inline void uda134x_write_reg_cache(struct snd_soc_codec *codec,
+ u8 reg, unsigned int value)
+{
+ u8 *cache = codec->reg_cache;
+
+ if (reg >= UDA134X_REGS_NUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * Write to the uda134x registers
+ *
+ */
+static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ int ret;
+ u8 addr;
+ u8 data = value;
+ struct uda134x_platform_data *pd = codec->control_data;
+
+ pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
+
+ if (reg >= UDA134X_REGS_NUM) {
+ printk(KERN_ERR "%s unkown register: reg: %d",
+ __func__, reg);
+ return -EINVAL;
+ }
+
+ uda134x_write_reg_cache(codec, reg, value);
+
+ switch (reg) {
+ case UDA134X_STATUS0:
+ case UDA134X_STATUS1:
+ addr = UDA134X_STATUS_ADDR;
+ break;
+ case UDA134X_DATA000:
+ case UDA134X_DATA001:
+ case UDA134X_DATA010:
+ addr = UDA134X_DATA0_ADDR;
+ break;
+ case UDA134X_DATA1:
+ addr = UDA134X_DATA1_ADDR;
+ break;
+ default:
+ /* It's an extended address register */
+ addr = (reg | UDA134X_EXTADDR_PREFIX);
+
+ ret = l3_write(&pd->l3,
+ UDA134X_DATA0_ADDR, &addr, 1);
+ if (ret != 1)
+ return -EIO;
+
+ addr = UDA134X_DATA0_ADDR;
+ data = (value | UDA134X_EXTDATA_PREFIX);
+ break;
+ }
+
+ ret = l3_write(&pd->l3,
+ addr, &data, 1);
+ if (ret != 1)
+ return -EIO;
+
+ return 0;
+}
+
+static inline void uda134x_reset(struct snd_soc_codec *codec)
+{
+ u8 reset_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS0);
+ uda134x_write(codec, UDA134X_STATUS0, reset_reg | (1<<6));
+ msleep(1);
+ uda134x_write(codec, UDA134X_STATUS0, reset_reg & ~(1<<6));
+}
+
+static int uda134x_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 mute_reg = uda134x_read_reg_cache(codec, UDA134X_DATA010);
+
+ pr_debug("%s mute: %d\n", __func__, mute);
+
+ if (mute)
+ mute_reg |= (1<<2);
+ else
+ mute_reg &= ~(1<<2);
+
+ uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2));
+
+ return 0;
+}
+
+static int uda134x_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct uda134x_priv *uda134x = codec->private_data;
+ struct snd_pcm_runtime *master_runtime;
+
+ if (uda134x->master_substream) {
+ master_runtime = uda134x->master_substream->runtime;
+
+ pr_debug("%s constraining to %d bits at %d\n", __func__,
+ master_runtime->sample_bits,
+ master_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ master_runtime->rate,
+ master_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ master_runtime->sample_bits,
+ master_runtime->sample_bits);
+
+ uda134x->slave_substream = substream;
+ } else
+ uda134x->master_substream = substream;
+
+ return 0;
+}
+
+static void uda134x_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct uda134x_priv *uda134x = codec->private_data;
+
+ if (uda134x->master_substream == substream)
+ uda134x->master_substream = uda134x->slave_substream;
+
+ uda134x->slave_substream = NULL;
+}
+
+static int uda134x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct uda134x_priv *uda134x = codec->private_data;
+ u8 hw_params;
+
+ if (substream == uda134x->slave_substream) {
+ pr_debug("%s ignoring hw_params for slave substream\n",
+ __func__);
+ return 0;
+ }
+
+ hw_params = uda134x_read_reg_cache(codec, UDA134X_STATUS0);
+ hw_params &= STATUS0_SYSCLK_MASK;
+ hw_params &= STATUS0_DAIFMT_MASK;
+
+ pr_debug("%s sysclk: %d, rate:%d\n", __func__,
+ uda134x->sysclk, params_rate(params));
+
+ /* set SYSCLK / fs ratio */
+ switch (uda134x->sysclk / params_rate(params)) {
+ case 512:
+ break;
+ case 384:
+ hw_params |= (1<<4);
+ break;
+ case 256:
+ hw_params |= (1<<5);
+ break;
+ default:
+ printk(KERN_ERR "%s unsupported fs\n", __func__);
+ return -EINVAL;
+ }
+
+ pr_debug("%s dai_fmt: %d, params_format:%d\n", __func__,
+ uda134x->dai_fmt, params_format(params));
+
+ /* set DAI format and word length */
+ switch (uda134x->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ hw_params |= (1<<1);
+ break;
+ case SNDRV_PCM_FORMAT_S18_3LE:
+ hw_params |= (1<<2);
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ hw_params |= ((1<<2) | (1<<1));
+ break;
+ default:
+ printk(KERN_ERR "%s unsupported format (right)\n",
+ __func__);
+ return -EINVAL;
+ }
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ hw_params |= (1<<3);
+ break;
+ default:
+ printk(KERN_ERR "%s unsupported format\n", __func__);
+ return -EINVAL;
+ }
+
+ uda134x_write(codec, UDA134X_STATUS0, hw_params);
+
+ return 0;
+}
+
+static int uda134x_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 uda134x_priv *uda134x = codec->private_data;
+
+ pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__,
+ clk_id, freq, dir);
+
+ /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
+ because the codec is slave. Of course limitations of the clock
+ master (the IIS controller) apply.
+ We'll error out on set_hw_params if it's not OK */
+ if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) {
+ uda134x->sysclk = freq;
+ return 0;
+ }
+
+ printk(KERN_ERR "%s unsupported sysclk\n", __func__);
+ return -EINVAL;
+}
+
+static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct uda134x_priv *uda134x = codec->private_data;
+
+ pr_debug("%s fmt: %08X\n", __func__, fmt);
+
+ /* codec supports only full slave mode */
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+ printk(KERN_ERR "%s unsupported slave mode\n", __func__);
+ return -EINVAL;
+ }
+
+ /* no support for clock inversion */
+ if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
+ printk(KERN_ERR "%s unsupported clock inversion\n", __func__);
+ return -EINVAL;
+ }
+
+ /* We can't setup DAI format here as it depends on the word bit num */
+ /* so let's just store the value for later */
+ uda134x->dai_fmt = fmt;
+
+ return 0;
+}
+
+static int uda134x_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u8 reg;
+ struct uda134x_platform_data *pd = codec->control_data;
+ int i;
+ u8 *cache = codec->reg_cache;
+
+ pr_debug("%s bias level %d\n", __func__, level);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* ADC, DAC on */
+ reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+ uda134x_write(codec, UDA134X_STATUS1, reg | 0x03);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ /* power on */
+ if (pd->power) {
+ pd->power(1);
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(uda134x_reg); i++)
+ codec->write(codec, i, *cache++);
+ }
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* ADC, DAC power off */
+ reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+ uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03));
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* power off */
+ if (pd->power)
+ pd->power(0);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+static const char *uda134x_dsp_setting[] = {"Flat", "Minimum1",
+ "Minimum2", "Maximum"};
+static const char *uda134x_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const char *uda134x_mixmode[] = {"Differential", "Analog1",
+ "Analog2", "Both"};
+
+static const struct soc_enum uda134x_mixer_enum[] = {
+SOC_ENUM_SINGLE(UDA134X_DATA010, 0, 0x04, uda134x_dsp_setting),
+SOC_ENUM_SINGLE(UDA134X_DATA010, 3, 0x04, uda134x_deemph),
+SOC_ENUM_SINGLE(UDA134X_EA010, 0, 0x04, uda134x_mixmode),
+};
+
+static const struct snd_kcontrol_new uda1341_snd_controls[] = {
+SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
+SOC_SINGLE("Capture Volume", UDA134X_EA010, 2, 0x07, 0),
+SOC_SINGLE("Analog1 Volume", UDA134X_EA000, 0, 0x1F, 1),
+SOC_SINGLE("Analog2 Volume", UDA134X_EA001, 0, 0x1F, 1),
+
+SOC_SINGLE("Mic Sensitivity", UDA134X_EA010, 2, 7, 0),
+SOC_SINGLE("Mic Volume", UDA134X_EA101, 0, 0x1F, 0),
+
+SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0),
+SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0),
+
+SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]),
+SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
+SOC_ENUM("Input Mux", uda134x_mixer_enum[2]),
+
+SOC_SINGLE("AGC Switch", UDA134X_EA100, 4, 1, 0),
+SOC_SINGLE("AGC Target Volume", UDA134X_EA110, 0, 0x03, 1),
+SOC_SINGLE("AGC Timing", UDA134X_EA110, 2, 0x07, 0),
+
+SOC_SINGLE("DAC +6dB Switch", UDA134X_STATUS1, 6, 1, 0),
+SOC_SINGLE("ADC +6dB Switch", UDA134X_STATUS1, 5, 1, 0),
+SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0),
+SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0),
+SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0),
+SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new uda1340_snd_controls[] = {
+SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
+
+SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0),
+SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0),
+
+SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]),
+SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
+
+SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
+};
+
+static int uda134x_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i, n;
+ const struct snd_kcontrol_new *ctrls;
+ struct uda134x_platform_data *pd = codec->control_data;
+
+ switch (pd->model) {
+ case UDA134X_UDA1340:
+ case UDA134X_UDA1344:
+ n = ARRAY_SIZE(uda1340_snd_controls);
+ ctrls = uda1340_snd_controls;
+ break;
+ case UDA134X_UDA1341:
+ n = ARRAY_SIZE(uda1341_snd_controls);
+ ctrls = uda1341_snd_controls;
+ break;
+ default:
+ printk(KERN_ERR "%s unkown codec type: %d",
+ __func__, pd->model);
+ return -EINVAL;
+ }
+
+ for (i = 0; i < n; i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&ctrls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+struct snd_soc_dai uda134x_dai = {
+ .name = "UDA134X",
+ /* playback capabilities */
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA134X_RATES,
+ .formats = UDA134X_FORMATS,
+ },
+ /* capture capabilities */
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA134X_RATES,
+ .formats = UDA134X_FORMATS,
+ },
+ /* pcm operations */
+ .ops = {
+ .startup = uda134x_startup,
+ .shutdown = uda134x_shutdown,
+ .hw_params = uda134x_hw_params,
+ .digital_mute = uda134x_mute,
+ .set_sysclk = uda134x_set_dai_sysclk,
+ .set_fmt = uda134x_set_dai_fmt,
+ }
+};
+EXPORT_SYMBOL(uda134x_dai);
+
+
+static int uda134x_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct uda134x_priv *uda134x;
+ void *codec_setup_data = socdev->codec_data;
+ int ret = -ENOMEM;
+ struct uda134x_platform_data *pd;
+
+ printk(KERN_INFO "UDA134X SoC Audio Codec\n");
+
+ if (!codec_setup_data) {
+ printk(KERN_ERR "UDA134X SoC codec: "
+ "missing L3 bitbang function\n");
+ return -ENODEV;
+ }
+
+ pd = codec_setup_data;
+ switch (pd->model) {
+ case UDA134X_UDA1340:
+ case UDA134X_UDA1341:
+ case UDA134X_UDA1344:
+ break;
+ default:
+ printk(KERN_ERR "UDA134X SoC codec: "
+ "unsupported model %d\n",
+ pd->model);
+ return -EINVAL;
+ }
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->codec == NULL)
+ return ret;
+
+ codec = socdev->codec;
+
+ uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL);
+ if (uda134x == NULL)
+ goto priv_err;
+ codec->private_data = uda134x;
+
+ codec->reg_cache = kmemdup(uda134x_reg, sizeof(uda134x_reg),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ goto reg_err;
+
+ mutex_init(&codec->mutex);
+
+ codec->reg_cache_size = sizeof(uda134x_reg);
+ codec->reg_cache_step = 1;
+
+ codec->name = "UDA134X";
+ codec->owner = THIS_MODULE;
+ codec->dai = &uda134x_dai;
+ codec->num_dai = 1;
+ codec->read = uda134x_read_reg_cache;
+ codec->write = uda134x_write;
+#ifdef POWER_OFF_ON_STANDBY
+ codec->set_bias_level = uda134x_set_bias_level;
+#endif
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->control_data = codec_setup_data;
+
+ if (pd->power)
+ pd->power(1);
+
+ uda134x_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "UDA134X: failed to register pcms\n");
+ goto pcm_err;
+ }
+
+ ret = uda134x_add_controls(codec);
+ if (ret < 0) {
+ printk(KERN_ERR "UDA134X: failed to register controls\n");
+ goto pcm_err;
+ }
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "UDA134X: failed to register card\n");
+ goto card_err;
+ }
+
+ return 0;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+reg_err:
+ kfree(codec->private_data);
+priv_err:
+ kfree(codec);
+ return ret;
+}
+
+/* power down chip */
+static int uda134x_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ kfree(codec->private_data);
+ kfree(codec->reg_cache);
+ kfree(codec);
+
+ return 0;
+}
+
+#if defined(CONFIG_PM)
+static int uda134x_soc_suspend(struct platform_device *pdev,
+ pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int uda134x_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
+ return 0;
+}
+#else
+#define uda134x_soc_suspend NULL
+#define uda134x_soc_resume NULL
+#endif /* CONFIG_PM */
+
+struct snd_soc_codec_device soc_codec_dev_uda134x = {
+ .probe = uda134x_soc_probe,
+ .remove = uda134x_soc_remove,
+ .suspend = uda134x_soc_suspend,
+ .resume = uda134x_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_uda134x);
+
+static int __init uda134x_init(void)
+{
+ return snd_soc_register_dai(&uda134x_dai);
+}
+module_init(uda134x_init);
+
+static void __exit uda134x_exit(void)
+{
+ snd_soc_unregister_dai(&uda134x_dai);
+}
+module_exit(uda134x_exit);
+
+MODULE_DESCRIPTION("UDA134X ALSA soc codec driver");
+MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h
new file mode 100644
index 00000000000..94f440490b3
--- /dev/null
+++ b/sound/soc/codecs/uda134x.h
@@ -0,0 +1,36 @@
+#ifndef _UDA134X_CODEC_H
+#define _UDA134X_CODEC_H
+
+#define UDA134X_L3ADDR 5
+#define UDA134X_DATA0_ADDR ((UDA134X_L3ADDR << 2) | 0)
+#define UDA134X_DATA1_ADDR ((UDA134X_L3ADDR << 2) | 1)
+#define UDA134X_STATUS_ADDR ((UDA134X_L3ADDR << 2) | 2)
+
+#define UDA134X_EXTADDR_PREFIX 0xC0
+#define UDA134X_EXTDATA_PREFIX 0xE0
+
+/* UDA134X registers */
+#define UDA134X_EA000 0
+#define UDA134X_EA001 1
+#define UDA134X_EA010 2
+#define UDA134X_EA011 3
+#define UDA134X_EA100 4
+#define UDA134X_EA101 5
+#define UDA134X_EA110 6
+#define UDA134X_EA111 7
+#define UDA134X_STATUS0 8
+#define UDA134X_STATUS1 9
+#define UDA134X_DATA000 10
+#define UDA134X_DATA001 11
+#define UDA134X_DATA010 12
+#define UDA134X_DATA1 13
+
+#define UDA134X_REGS_NUM 14
+
+#define STATUS0_DAIFMT_MASK (~(7<<1))
+#define STATUS0_SYSCLK_MASK (~(3<<4))
+
+extern struct snd_soc_dai uda134x_dai;
+extern struct snd_soc_codec_device soc_codec_dev_uda134x;
+
+#endif
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index a69ee72a7af..e6bf0844fbf 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -407,7 +407,8 @@ static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
* when the DAI is being clocked by the CPU DAI. It's up to the
* machine and cpu DAI driver to do this before we are called.
*/
-static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
+static int uda1380_pcm_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -439,7 +440,8 @@ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
}
static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -477,7 +479,8 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream)
+static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -560,8 +563,6 @@ struct snd_soc_dai uda1380_dai[] = {
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
.prepare = uda1380_pcm_prepare,
- },
- .dai_ops = {
.digital_mute = uda1380_mute,
.set_fmt = uda1380_set_dai_fmt,
},
@@ -579,8 +580,6 @@ struct snd_soc_dai uda1380_dai[] = {
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
.prepare = uda1380_pcm_prepare,
- },
- .dai_ops = {
.digital_mute = uda1380_mute,
.set_fmt = uda1380_set_dai_fmt,
},
@@ -598,8 +597,6 @@ struct snd_soc_dai uda1380_dai[] = {
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
.prepare = uda1380_pcm_prepare,
- },
- .dai_ops = {
.set_fmt = uda1380_set_dai_fmt,
},
},
@@ -680,7 +677,7 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
/* uda1380 init */
uda1380_add_controls(codec);
uda1380_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
pr_err("uda1380: failed to register card\n");
goto card_err;
@@ -844,6 +841,18 @@ struct snd_soc_codec_device soc_codec_dev_uda1380 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
+static int __init uda1380_modinit(void)
+{
+ return snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
+}
+module_init(uda1380_modinit);
+
+static void __exit uda1380_exit(void)
+{
+ snd_soc_unregister_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
+}
+module_exit(uda1380_exit);
+
MODULE_AUTHOR("Giorgio Padrin");
MODULE_DESCRIPTION("Audio support for codec Philips UDA1380");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
new file mode 100644
index 00000000000..e3989d406f5
--- /dev/null
+++ b/sound/soc/codecs/wm8350.c
@@ -0,0 +1,1583 @@
+/*
+ * wm8350.c -- WM8350 ALSA SoC audio driver
+ *
+ * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC.
+ *
+ * Author: Liam Girdwood <lg@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License 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/platform_device.h>
+#include <linux/mfd/wm8350/audio.h>
+#include <linux/mfd/wm8350/core.h>
+#include <linux/regulator/consumer.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8350.h"
+
+#define WM8350_OUTn_0dB 0x39
+
+#define WM8350_RAMP_NONE 0
+#define WM8350_RAMP_UP 1
+#define WM8350_RAMP_DOWN 2
+
+/* We only include the analogue supplies here; the digital supplies
+ * need to be available well before this driver can be probed.
+ */
+static const char *supply_names[] = {
+ "AVDD",
+ "HPVDD",
+};
+
+struct wm8350_output {
+ u16 active;
+ u16 left_vol;
+ u16 right_vol;
+ u16 ramp;
+ u16 mute;
+};
+
+struct wm8350_data {
+ struct snd_soc_codec codec;
+ struct wm8350_output out1;
+ struct wm8350_output out2;
+ struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
+};
+
+static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ struct wm8350 *wm8350 = codec->control_data;
+ return wm8350->reg_cache[reg];
+}
+
+static unsigned int wm8350_codec_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ struct wm8350 *wm8350 = codec->control_data;
+ return wm8350_reg_read(wm8350, reg);
+}
+
+static int wm8350_codec_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ struct wm8350 *wm8350 = codec->control_data;
+ return wm8350_reg_write(wm8350, reg, value);
+}
+
+/*
+ * Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown.
+ */
+static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec)
+{
+ struct wm8350_data *wm8350_data = codec->private_data;
+ struct wm8350_output *out1 = &wm8350_data->out1;
+ struct wm8350 *wm8350 = codec->control_data;
+ int left_complete = 0, right_complete = 0;
+ u16 reg, val;
+
+ /* left channel */
+ reg = wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME);
+ val = (reg & WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+
+ if (out1->ramp == WM8350_RAMP_UP) {
+ /* ramp step up */
+ if (val < out1->left_vol) {
+ val++;
+ reg &= ~WM8350_OUT1L_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME,
+ reg | (val << WM8350_OUT1L_VOL_SHIFT));
+ } else
+ left_complete = 1;
+ } else if (out1->ramp == WM8350_RAMP_DOWN) {
+ /* ramp step down */
+ if (val > 0) {
+ val--;
+ reg &= ~WM8350_OUT1L_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME,
+ reg | (val << WM8350_OUT1L_VOL_SHIFT));
+ } else
+ left_complete = 1;
+ } else
+ return 1;
+
+ /* right channel */
+ reg = wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME);
+ val = (reg & WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+ if (out1->ramp == WM8350_RAMP_UP) {
+ /* ramp step up */
+ if (val < out1->right_vol) {
+ val++;
+ reg &= ~WM8350_OUT1R_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME,
+ reg | (val << WM8350_OUT1R_VOL_SHIFT));
+ } else
+ right_complete = 1;
+ } else if (out1->ramp == WM8350_RAMP_DOWN) {
+ /* ramp step down */
+ if (val > 0) {
+ val--;
+ reg &= ~WM8350_OUT1R_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME,
+ reg | (val << WM8350_OUT1R_VOL_SHIFT));
+ } else
+ right_complete = 1;
+ }
+
+ /* only hit the update bit if either volume has changed this step */
+ if (!left_complete || !right_complete)
+ wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME, WM8350_OUT1_VU);
+
+ return left_complete & right_complete;
+}
+
+/*
+ * Ramp OUT2 PGA volume to minimise pops at stream startup and shutdown.
+ */
+static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec)
+{
+ struct wm8350_data *wm8350_data = codec->private_data;
+ struct wm8350_output *out2 = &wm8350_data->out2;
+ struct wm8350 *wm8350 = codec->control_data;
+ int left_complete = 0, right_complete = 0;
+ u16 reg, val;
+
+ /* left channel */
+ reg = wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME);
+ val = (reg & WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+ if (out2->ramp == WM8350_RAMP_UP) {
+ /* ramp step up */
+ if (val < out2->left_vol) {
+ val++;
+ reg &= ~WM8350_OUT2L_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME,
+ reg | (val << WM8350_OUT1L_VOL_SHIFT));
+ } else
+ left_complete = 1;
+ } else if (out2->ramp == WM8350_RAMP_DOWN) {
+ /* ramp step down */
+ if (val > 0) {
+ val--;
+ reg &= ~WM8350_OUT2L_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME,
+ reg | (val << WM8350_OUT1L_VOL_SHIFT));
+ } else
+ left_complete = 1;
+ } else
+ return 1;
+
+ /* right channel */
+ reg = wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME);
+ val = (reg & WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+ if (out2->ramp == WM8350_RAMP_UP) {
+ /* ramp step up */
+ if (val < out2->right_vol) {
+ val++;
+ reg &= ~WM8350_OUT2R_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME,
+ reg | (val << WM8350_OUT1R_VOL_SHIFT));
+ } else
+ right_complete = 1;
+ } else if (out2->ramp == WM8350_RAMP_DOWN) {
+ /* ramp step down */
+ if (val > 0) {
+ val--;
+ reg &= ~WM8350_OUT2R_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME,
+ reg | (val << WM8350_OUT1R_VOL_SHIFT));
+ } else
+ right_complete = 1;
+ }
+
+ /* only hit the update bit if either volume has changed this step */
+ if (!left_complete || !right_complete)
+ wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME, WM8350_OUT2_VU);
+
+ return left_complete & right_complete;
+}
+
+/*
+ * This work ramps both output PGAs at stream start/stop time to
+ * minimise pop associated with DAPM power switching.
+ * It's best to enable Zero Cross when ramping occurs to minimise any
+ * zipper noises.
+ */
+static void wm8350_pga_work(struct work_struct *work)
+{
+ struct snd_soc_codec *codec =
+ container_of(work, struct snd_soc_codec, delayed_work.work);
+ struct wm8350_data *wm8350_data = codec->private_data;
+ struct wm8350_output *out1 = &wm8350_data->out1,
+ *out2 = &wm8350_data->out2;
+ int i, out1_complete, out2_complete;
+
+ /* do we need to ramp at all ? */
+ if (out1->ramp == WM8350_RAMP_NONE && out2->ramp == WM8350_RAMP_NONE)
+ return;
+
+ /* PGA volumes have 6 bits of resolution to ramp */
+ for (i = 0; i <= 63; i++) {
+ out1_complete = 1, out2_complete = 1;
+ if (out1->ramp != WM8350_RAMP_NONE)
+ out1_complete = wm8350_out1_ramp_step(codec);
+ if (out2->ramp != WM8350_RAMP_NONE)
+ out2_complete = wm8350_out2_ramp_step(codec);
+
+ /* ramp finished ? */
+ if (out1_complete && out2_complete)
+ break;
+
+ /* we need to delay longer on the up ramp */
+ if (out1->ramp == WM8350_RAMP_UP ||
+ out2->ramp == WM8350_RAMP_UP) {
+ /* delay is longer over 0dB as increases are larger */
+ if (i >= WM8350_OUTn_0dB)
+ schedule_timeout_interruptible(msecs_to_jiffies
+ (2));
+ else
+ schedule_timeout_interruptible(msecs_to_jiffies
+ (1));
+ } else
+ udelay(50); /* doesn't matter if we delay longer */
+ }
+
+ out1->ramp = WM8350_RAMP_NONE;
+ out2->ramp = WM8350_RAMP_NONE;
+}
+
+/*
+ * WM8350 Controls
+ */
+
+static int pga_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct wm8350_data *wm8350_data = codec->private_data;
+ struct wm8350_output *out;
+
+ switch (w->shift) {
+ case 0:
+ case 1:
+ out = &wm8350_data->out1;
+ break;
+ case 2:
+ case 3:
+ out = &wm8350_data->out2;
+ break;
+
+ default:
+ BUG();
+ return -1;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ out->ramp = WM8350_RAMP_UP;
+ out->active = 1;
+
+ if (!delayed_work_pending(&codec->delayed_work))
+ schedule_delayed_work(&codec->delayed_work,
+ msecs_to_jiffies(1));
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ out->ramp = WM8350_RAMP_DOWN;
+ out->active = 0;
+
+ if (!delayed_work_pending(&codec->delayed_work))
+ schedule_delayed_work(&codec->delayed_work,
+ msecs_to_jiffies(1));
+ break;
+ }
+
+ return 0;
+}
+
+static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm8350_data *wm8350_priv = codec->private_data;
+ struct wm8350_output *out = NULL;
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int ret;
+ unsigned int reg = mc->reg;
+ u16 val;
+
+ /* For OUT1 and OUT2 we shadow the values and only actually write
+ * them out when active in order to ensure the amplifier comes on
+ * as quietly as possible. */
+ switch (reg) {
+ case WM8350_LOUT1_VOLUME:
+ out = &wm8350_priv->out1;
+ break;
+ case WM8350_LOUT2_VOLUME:
+ out = &wm8350_priv->out2;
+ break;
+ default:
+ break;
+ }
+
+ if (out) {
+ out->left_vol = ucontrol->value.integer.value[0];
+ out->right_vol = ucontrol->value.integer.value[1];
+ if (!out->active)
+ return 1;
+ }
+
+ ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
+ if (ret < 0)
+ return ret;
+
+ /* now hit the volume update bits (always bit 8) */
+ val = wm8350_codec_read(codec, reg);
+ wm8350_codec_write(codec, reg, val | WM8350_OUT1_VU);
+ return 1;
+}
+
+static int wm8350_get_volsw_2r(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm8350_data *wm8350_priv = codec->private_data;
+ struct wm8350_output *out1 = &wm8350_priv->out1;
+ struct wm8350_output *out2 = &wm8350_priv->out2;
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ unsigned int reg = mc->reg;
+
+ /* If these are cached registers use the cache */
+ switch (reg) {
+ case WM8350_LOUT1_VOLUME:
+ ucontrol->value.integer.value[0] = out1->left_vol;
+ ucontrol->value.integer.value[1] = out1->right_vol;
+ return 0;
+
+ case WM8350_LOUT2_VOLUME:
+ ucontrol->value.integer.value[0] = out2->left_vol;
+ ucontrol->value.integer.value[1] = out2->right_vol;
+ return 0;
+
+ default:
+ break;
+ }
+
+ return snd_soc_get_volsw_2r(kcontrol, ucontrol);
+}
+
+/* double control with volume update */
+#define SOC_WM8350_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
+ xinvert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw_2r, \
+ .get = wm8350_get_volsw_2r, .put = wm8350_put_volsw_2r_vu, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+ .rshift = xshift, .max = xmax, .invert = xinvert}, }
+
+static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
+static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" };
+static const char *wm8350_dacmutem[] = { "Normal", "Soft" };
+static const char *wm8350_dacmutes[] = { "Fast", "Slow" };
+static const char *wm8350_dacfilter[] = { "Normal", "Sloping" };
+static const char *wm8350_adcfilter[] = { "None", "High Pass" };
+static const char *wm8350_adchp[] = { "44.1kHz", "8kHz", "16kHz", "32kHz" };
+static const char *wm8350_lr[] = { "Left", "Right" };
+
+static const struct soc_enum wm8350_enum[] = {
+ SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 4, 4, wm8350_deemp),
+ SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 0, 4, wm8350_pol),
+ SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 14, 2, wm8350_dacmutem),
+ SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 13, 2, wm8350_dacmutes),
+ SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 12, 2, wm8350_dacfilter),
+ SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 15, 2, wm8350_adcfilter),
+ SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 8, 4, wm8350_adchp),
+ SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 0, 4, wm8350_pol),
+ SOC_ENUM_SINGLE(WM8350_INPUT_MIXER_VOLUME, 15, 2, wm8350_lr),
+};
+
+static DECLARE_TLV_DB_LINEAR(pre_amp_tlv, -1200, 3525);
+static DECLARE_TLV_DB_LINEAR(out_pga_tlv, -5700, 600);
+static DECLARE_TLV_DB_SCALE(dac_pcm_tlv, -7163, 36, 1);
+static DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -12700, 50, 1);
+static DECLARE_TLV_DB_SCALE(out_mix_tlv, -1500, 300, 1);
+
+static const unsigned int capture_sd_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 12, TLV_DB_SCALE_ITEM(-3600, 300, 1),
+ 13, 15, TLV_DB_SCALE_ITEM(0, 0, 0),
+};
+
+static const struct snd_kcontrol_new wm8350_snd_controls[] = {
+ SOC_ENUM("Playback Deemphasis", wm8350_enum[0]),
+ SOC_ENUM("Playback DAC Inversion", wm8350_enum[1]),
+ SOC_WM8350_DOUBLE_R_TLV("Playback PCM Volume",
+ WM8350_DAC_DIGITAL_VOLUME_L,
+ WM8350_DAC_DIGITAL_VOLUME_R,
+ 0, 255, 0, dac_pcm_tlv),
+ SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]),
+ SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]),
+ SOC_ENUM("Playback PCM Filter", wm8350_enum[4]),
+ SOC_ENUM("Capture PCM Filter", wm8350_enum[5]),
+ SOC_ENUM("Capture PCM HP Filter", wm8350_enum[6]),
+ SOC_ENUM("Capture ADC Inversion", wm8350_enum[7]),
+ SOC_WM8350_DOUBLE_R_TLV("Capture PCM Volume",
+ WM8350_ADC_DIGITAL_VOLUME_L,
+ WM8350_ADC_DIGITAL_VOLUME_R,
+ 0, 255, 0, adc_pcm_tlv),
+ SOC_DOUBLE_TLV("Capture Sidetone Volume",
+ WM8350_ADC_DIVIDER,
+ 8, 4, 15, 1, capture_sd_tlv),
+ SOC_WM8350_DOUBLE_R_TLV("Capture Volume",
+ WM8350_LEFT_INPUT_VOLUME,
+ WM8350_RIGHT_INPUT_VOLUME,
+ 2, 63, 0, pre_amp_tlv),
+ SOC_DOUBLE_R("Capture ZC Switch",
+ WM8350_LEFT_INPUT_VOLUME,
+ WM8350_RIGHT_INPUT_VOLUME, 13, 1, 0),
+ SOC_SINGLE_TLV("Left Input Left Sidetone Volume",
+ WM8350_OUTPUT_LEFT_MIXER_VOLUME, 1, 7, 0, out_mix_tlv),
+ SOC_SINGLE_TLV("Left Input Right Sidetone Volume",
+ WM8350_OUTPUT_LEFT_MIXER_VOLUME,
+ 5, 7, 0, out_mix_tlv),
+ SOC_SINGLE_TLV("Left Input Bypass Volume",
+ WM8350_OUTPUT_LEFT_MIXER_VOLUME,
+ 9, 7, 0, out_mix_tlv),
+ SOC_SINGLE_TLV("Right Input Left Sidetone Volume",
+ WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+ 1, 7, 0, out_mix_tlv),
+ SOC_SINGLE_TLV("Right Input Right Sidetone Volume",
+ WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+ 5, 7, 0, out_mix_tlv),
+ SOC_SINGLE_TLV("Right Input Bypass Volume",
+ WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+ 13, 7, 0, out_mix_tlv),
+ SOC_SINGLE("Left Input Mixer +20dB Switch",
+ WM8350_INPUT_MIXER_VOLUME_L, 0, 1, 0),
+ SOC_SINGLE("Right Input Mixer +20dB Switch",
+ WM8350_INPUT_MIXER_VOLUME_R, 0, 1, 0),
+ SOC_SINGLE_TLV("Out4 Capture Volume",
+ WM8350_INPUT_MIXER_VOLUME,
+ 1, 7, 0, out_mix_tlv),
+ SOC_WM8350_DOUBLE_R_TLV("Out1 Playback Volume",
+ WM8350_LOUT1_VOLUME,
+ WM8350_ROUT1_VOLUME,
+ 2, 63, 0, out_pga_tlv),
+ SOC_DOUBLE_R("Out1 Playback ZC Switch",
+ WM8350_LOUT1_VOLUME,
+ WM8350_ROUT1_VOLUME, 13, 1, 0),
+ SOC_WM8350_DOUBLE_R_TLV("Out2 Playback Volume",
+ WM8350_LOUT2_VOLUME,
+ WM8350_ROUT2_VOLUME,
+ 2, 63, 0, out_pga_tlv),
+ SOC_DOUBLE_R("Out2 Playback ZC Switch", WM8350_LOUT2_VOLUME,
+ WM8350_ROUT2_VOLUME, 13, 1, 0),
+ SOC_SINGLE("Out2 Right Invert Switch", WM8350_ROUT2_VOLUME, 10, 1, 0),
+ SOC_SINGLE_TLV("Out2 Beep Volume", WM8350_BEEP_VOLUME,
+ 5, 7, 0, out_mix_tlv),
+
+ SOC_DOUBLE_R("Out1 Playback Switch",
+ WM8350_LOUT1_VOLUME,
+ WM8350_ROUT1_VOLUME,
+ 14, 1, 1),
+ SOC_DOUBLE_R("Out2 Playback Switch",
+ WM8350_LOUT2_VOLUME,
+ WM8350_ROUT2_VOLUME,
+ 14, 1, 1),
+};
+
+/*
+ * DAPM Controls
+ */
+
+/* Left Playback Mixer */
+static const struct snd_kcontrol_new wm8350_left_play_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Playback Switch",
+ WM8350_LEFT_MIXER_CONTROL, 11, 1, 0),
+ SOC_DAPM_SINGLE("Left Bypass Switch",
+ WM8350_LEFT_MIXER_CONTROL, 2, 1, 0),
+ SOC_DAPM_SINGLE("Right Playback Switch",
+ WM8350_LEFT_MIXER_CONTROL, 12, 1, 0),
+ SOC_DAPM_SINGLE("Left Sidetone Switch",
+ WM8350_LEFT_MIXER_CONTROL, 0, 1, 0),
+ SOC_DAPM_SINGLE("Right Sidetone Switch",
+ WM8350_LEFT_MIXER_CONTROL, 1, 1, 0),
+};
+
+/* Right Playback Mixer */
+static const struct snd_kcontrol_new wm8350_right_play_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Playback Switch",
+ WM8350_RIGHT_MIXER_CONTROL, 12, 1, 0),
+ SOC_DAPM_SINGLE("Right Bypass Switch",
+ WM8350_RIGHT_MIXER_CONTROL, 3, 1, 0),
+ SOC_DAPM_SINGLE("Left Playback Switch",
+ WM8350_RIGHT_MIXER_CONTROL, 11, 1, 0),
+ SOC_DAPM_SINGLE("Left Sidetone Switch",
+ WM8350_RIGHT_MIXER_CONTROL, 0, 1, 0),
+ SOC_DAPM_SINGLE("Right Sidetone Switch",
+ WM8350_RIGHT_MIXER_CONTROL, 1, 1, 0),
+};
+
+/* Out4 Mixer */
+static const struct snd_kcontrol_new wm8350_out4_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Right Playback Switch",
+ WM8350_OUT4_MIXER_CONTROL, 12, 1, 0),
+ SOC_DAPM_SINGLE("Left Playback Switch",
+ WM8350_OUT4_MIXER_CONTROL, 11, 1, 0),
+ SOC_DAPM_SINGLE("Right Capture Switch",
+ WM8350_OUT4_MIXER_CONTROL, 9, 1, 0),
+ SOC_DAPM_SINGLE("Out3 Playback Switch",
+ WM8350_OUT4_MIXER_CONTROL, 2, 1, 0),
+ SOC_DAPM_SINGLE("Right Mixer Switch",
+ WM8350_OUT4_MIXER_CONTROL, 1, 1, 0),
+ SOC_DAPM_SINGLE("Left Mixer Switch",
+ WM8350_OUT4_MIXER_CONTROL, 0, 1, 0),
+};
+
+/* Out3 Mixer */
+static const struct snd_kcontrol_new wm8350_out3_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Left Playback Switch",
+ WM8350_OUT3_MIXER_CONTROL, 11, 1, 0),
+ SOC_DAPM_SINGLE("Left Capture Switch",
+ WM8350_OUT3_MIXER_CONTROL, 8, 1, 0),
+ SOC_DAPM_SINGLE("Out4 Playback Switch",
+ WM8350_OUT3_MIXER_CONTROL, 3, 1, 0),
+ SOC_DAPM_SINGLE("Left Mixer Switch",
+ WM8350_OUT3_MIXER_CONTROL, 0, 1, 0),
+};
+
+/* Left Input Mixer */
+static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = {
+ SOC_DAPM_SINGLE_TLV("L2 Capture Volume",
+ WM8350_INPUT_MIXER_VOLUME_L, 1, 7, 0, out_mix_tlv),
+ SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
+ WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv),
+ SOC_DAPM_SINGLE("PGA Capture Switch",
+ WM8350_LEFT_INPUT_VOLUME, 14, 1, 0),
+};
+
+/* Right Input Mixer */
+static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = {
+ SOC_DAPM_SINGLE_TLV("L2 Capture Volume",
+ WM8350_INPUT_MIXER_VOLUME_R, 5, 7, 0, out_mix_tlv),
+ SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
+ WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv),
+ SOC_DAPM_SINGLE("PGA Capture Switch",
+ WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0),
+};
+
+/* Left Mic Mixer */
+static const struct snd_kcontrol_new wm8350_left_mic_mixer_controls[] = {
+ SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 1, 1, 0),
+ SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 0, 1, 0),
+ SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 2, 1, 0),
+};
+
+/* Right Mic Mixer */
+static const struct snd_kcontrol_new wm8350_right_mic_mixer_controls[] = {
+ SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 9, 1, 0),
+ SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 8, 1, 0),
+ SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 10, 1, 0),
+};
+
+/* Beep Switch */
+static const struct snd_kcontrol_new wm8350_beep_switch_controls =
+SOC_DAPM_SINGLE("Switch", WM8350_BEEP_VOLUME, 15, 1, 1);
+
+/* Out4 Capture Mux */
+static const struct snd_kcontrol_new wm8350_out4_capture_controls =
+SOC_DAPM_ENUM("Route", wm8350_enum[8]);
+
+static const struct snd_soc_dapm_widget wm8350_dapm_widgets[] = {
+
+ SND_SOC_DAPM_PGA("IN3R PGA", WM8350_POWER_MGMT_2, 11, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IN3L PGA", WM8350_POWER_MGMT_2, 10, 0, NULL, 0),
+ SND_SOC_DAPM_PGA_E("Right Out2 PGA", WM8350_POWER_MGMT_3, 3, 0, NULL,
+ 0, pga_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_PGA_E("Left Out2 PGA", WM8350_POWER_MGMT_3, 2, 0, NULL, 0,
+ pga_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_PGA_E("Right Out1 PGA", WM8350_POWER_MGMT_3, 1, 0, NULL,
+ 0, pga_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_PGA_E("Left Out1 PGA", WM8350_POWER_MGMT_3, 0, 0, NULL, 0,
+ pga_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+ SND_SOC_DAPM_MIXER("Right Capture Mixer", WM8350_POWER_MGMT_2,
+ 7, 0, &wm8350_right_capt_mixer_controls[0],
+ ARRAY_SIZE(wm8350_right_capt_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("Left Capture Mixer", WM8350_POWER_MGMT_2,
+ 6, 0, &wm8350_left_capt_mixer_controls[0],
+ ARRAY_SIZE(wm8350_left_capt_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("Out4 Mixer", WM8350_POWER_MGMT_2, 5, 0,
+ &wm8350_out4_mixer_controls[0],
+ ARRAY_SIZE(wm8350_out4_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("Out3 Mixer", WM8350_POWER_MGMT_2, 4, 0,
+ &wm8350_out3_mixer_controls[0],
+ ARRAY_SIZE(wm8350_out3_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("Right Playback Mixer", WM8350_POWER_MGMT_2, 1, 0,
+ &wm8350_right_play_mixer_controls[0],
+ ARRAY_SIZE(wm8350_right_play_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("Left Playback Mixer", WM8350_POWER_MGMT_2, 0, 0,
+ &wm8350_left_play_mixer_controls[0],
+ ARRAY_SIZE(wm8350_left_play_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("Left Mic Mixer", WM8350_POWER_MGMT_2, 8, 0,
+ &wm8350_left_mic_mixer_controls[0],
+ ARRAY_SIZE(wm8350_left_mic_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("Right Mic Mixer", WM8350_POWER_MGMT_2, 9, 0,
+ &wm8350_right_mic_mixer_controls[0],
+ ARRAY_SIZE(wm8350_right_mic_mixer_controls)),
+
+ /* virtual mixer for Beep and Out2R */
+ SND_SOC_DAPM_MIXER("Out2 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_SWITCH("Beep", WM8350_POWER_MGMT_3, 7, 0,
+ &wm8350_beep_switch_controls),
+
+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture",
+ WM8350_POWER_MGMT_4, 3, 0),
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture",
+ WM8350_POWER_MGMT_4, 2, 0),
+ SND_SOC_DAPM_DAC("Right DAC", "Right Playback",
+ WM8350_POWER_MGMT_4, 5, 0),
+ SND_SOC_DAPM_DAC("Left DAC", "Left Playback",
+ WM8350_POWER_MGMT_4, 4, 0),
+
+ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8350_POWER_MGMT_1, 4, 0),
+
+ SND_SOC_DAPM_MUX("Out4 Capture Channel", SND_SOC_NOPM, 0, 0,
+ &wm8350_out4_capture_controls),
+
+ SND_SOC_DAPM_OUTPUT("OUT1R"),
+ SND_SOC_DAPM_OUTPUT("OUT1L"),
+ SND_SOC_DAPM_OUTPUT("OUT2R"),
+ SND_SOC_DAPM_OUTPUT("OUT2L"),
+ SND_SOC_DAPM_OUTPUT("OUT3"),
+ SND_SOC_DAPM_OUTPUT("OUT4"),
+
+ SND_SOC_DAPM_INPUT("IN1RN"),
+ SND_SOC_DAPM_INPUT("IN1RP"),
+ SND_SOC_DAPM_INPUT("IN2R"),
+ SND_SOC_DAPM_INPUT("IN1LP"),
+ SND_SOC_DAPM_INPUT("IN1LN"),
+ SND_SOC_DAPM_INPUT("IN2L"),
+ SND_SOC_DAPM_INPUT("IN3R"),
+ SND_SOC_DAPM_INPUT("IN3L"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* left playback mixer */
+ {"Left Playback Mixer", "Playback Switch", "Left DAC"},
+ {"Left Playback Mixer", "Left Bypass Switch", "IN3L PGA"},
+ {"Left Playback Mixer", "Right Playback Switch", "Right DAC"},
+ {"Left Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"},
+ {"Left Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"},
+
+ /* right playback mixer */
+ {"Right Playback Mixer", "Playback Switch", "Right DAC"},
+ {"Right Playback Mixer", "Right Bypass Switch", "IN3R PGA"},
+ {"Right Playback Mixer", "Left Playback Switch", "Left DAC"},
+ {"Right Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"},
+ {"Right Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"},
+
+ /* out4 playback mixer */
+ {"Out4 Mixer", "Right Playback Switch", "Right DAC"},
+ {"Out4 Mixer", "Left Playback Switch", "Left DAC"},
+ {"Out4 Mixer", "Right Capture Switch", "Right Capture Mixer"},
+ {"Out4 Mixer", "Out3 Playback Switch", "Out3 Mixer"},
+ {"Out4 Mixer", "Right Mixer Switch", "Right Playback Mixer"},
+ {"Out4 Mixer", "Left Mixer Switch", "Left Playback Mixer"},
+ {"OUT4", NULL, "Out4 Mixer"},
+
+ /* out3 playback mixer */
+ {"Out3 Mixer", "Left Playback Switch", "Left DAC"},
+ {"Out3 Mixer", "Left Capture Switch", "Left Capture Mixer"},
+ {"Out3 Mixer", "Left Mixer Switch", "Left Playback Mixer"},
+ {"Out3 Mixer", "Out4 Playback Switch", "Out4 Mixer"},
+ {"OUT3", NULL, "Out3 Mixer"},
+
+ /* out2 */
+ {"Right Out2 PGA", NULL, "Right Playback Mixer"},
+ {"Left Out2 PGA", NULL, "Left Playback Mixer"},
+ {"OUT2L", NULL, "Left Out2 PGA"},
+ {"OUT2R", NULL, "Right Out2 PGA"},
+
+ /* out1 */
+ {"Right Out1 PGA", NULL, "Right Playback Mixer"},
+ {"Left Out1 PGA", NULL, "Left Playback Mixer"},
+ {"OUT1L", NULL, "Left Out1 PGA"},
+ {"OUT1R", NULL, "Right Out1 PGA"},
+
+ /* ADCs */
+ {"Left ADC", NULL, "Left Capture Mixer"},
+ {"Right ADC", NULL, "Right Capture Mixer"},
+
+ /* Left capture mixer */
+ {"Left Capture Mixer", "L2 Capture Volume", "IN2L"},
+ {"Left Capture Mixer", "L3 Capture Volume", "IN3L PGA"},
+ {"Left Capture Mixer", "PGA Capture Switch", "Left Mic Mixer"},
+ {"Left Capture Mixer", NULL, "Out4 Capture Channel"},
+
+ /* Right capture mixer */
+ {"Right Capture Mixer", "L2 Capture Volume", "IN2R"},
+ {"Right Capture Mixer", "L3 Capture Volume", "IN3R PGA"},
+ {"Right Capture Mixer", "PGA Capture Switch", "Right Mic Mixer"},
+ {"Right Capture Mixer", NULL, "Out4 Capture Channel"},
+
+ /* L3 Inputs */
+ {"IN3L PGA", NULL, "IN3L"},
+ {"IN3R PGA", NULL, "IN3R"},
+
+ /* Left Mic mixer */
+ {"Left Mic Mixer", "INN Capture Switch", "IN1LN"},
+ {"Left Mic Mixer", "INP Capture Switch", "IN1LP"},
+ {"Left Mic Mixer", "IN2 Capture Switch", "IN2L"},
+
+ /* Right Mic mixer */
+ {"Right Mic Mixer", "INN Capture Switch", "IN1RN"},
+ {"Right Mic Mixer", "INP Capture Switch", "IN1RP"},
+ {"Right Mic Mixer", "IN2 Capture Switch", "IN2R"},
+
+ /* out 4 capture */
+ {"Out4 Capture Channel", NULL, "Out4 Mixer"},
+
+ /* Beep */
+ {"Beep", NULL, "IN3R PGA"},
+};
+
+static int wm8350_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8350_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+static int wm8350_add_widgets(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(codec,
+ wm8350_dapm_widgets,
+ ARRAY_SIZE(wm8350_dapm_widgets));
+ if (ret != 0) {
+ dev_err(codec->dev, "dapm control register failed\n");
+ return ret;
+ }
+
+ /* set up audio paths */
+ ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ if (ret != 0) {
+ dev_err(codec->dev, "DAPM route register failed\n");
+ return ret;
+ }
+
+ return snd_soc_dapm_new_widgets(codec);
+}
+
+static int wm8350_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 wm8350 *wm8350 = codec->control_data;
+ u16 fll_4;
+
+ switch (clk_id) {
+ case WM8350_MCLK_SEL_MCLK:
+ wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_1,
+ WM8350_MCLK_SEL);
+ break;
+ case WM8350_MCLK_SEL_PLL_MCLK:
+ case WM8350_MCLK_SEL_PLL_DAC:
+ case WM8350_MCLK_SEL_PLL_ADC:
+ case WM8350_MCLK_SEL_PLL_32K:
+ wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_1,
+ WM8350_MCLK_SEL);
+ fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) &
+ ~WM8350_FLL_CLK_SRC_MASK;
+ wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id);
+ break;
+ }
+
+ /* MCLK direction */
+ if (dir == WM8350_MCLK_DIR_OUT)
+ wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_2,
+ WM8350_MCLK_DIR);
+ else
+ wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_2,
+ WM8350_MCLK_DIR);
+
+ return 0;
+}
+
+static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 val;
+
+ switch (div_id) {
+ case WM8350_ADC_CLKDIV:
+ val = wm8350_codec_read(codec, WM8350_ADC_DIVIDER) &
+ ~WM8350_ADC_CLKDIV_MASK;
+ wm8350_codec_write(codec, WM8350_ADC_DIVIDER, val | div);
+ break;
+ case WM8350_DAC_CLKDIV:
+ val = wm8350_codec_read(codec, WM8350_DAC_CLOCK_CONTROL) &
+ ~WM8350_DAC_CLKDIV_MASK;
+ wm8350_codec_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div);
+ break;
+ case WM8350_BCLK_CLKDIV:
+ val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+ ~WM8350_BCLK_DIV_MASK;
+ wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+ break;
+ case WM8350_OPCLK_CLKDIV:
+ val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+ ~WM8350_OPCLK_DIV_MASK;
+ wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+ break;
+ case WM8350_SYS_CLKDIV:
+ val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+ ~WM8350_MCLK_DIV_MASK;
+ wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+ break;
+ case WM8350_DACLR_CLKDIV:
+ val = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) &
+ ~WM8350_DACLRC_RATE_MASK;
+ wm8350_codec_write(codec, WM8350_DAC_LR_RATE, val | div);
+ break;
+ case WM8350_ADCLR_CLKDIV:
+ val = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) &
+ ~WM8350_ADCLRC_RATE_MASK;
+ wm8350_codec_write(codec, WM8350_ADC_LR_RATE, val | div);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
+ ~(WM8350_AIF_BCLK_INV | WM8350_AIF_LRCLK_INV | WM8350_AIF_FMT_MASK);
+ u16 master = wm8350_codec_read(codec, WM8350_AI_DAC_CONTROL) &
+ ~WM8350_BCLK_MSTR;
+ u16 dac_lrc = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) &
+ ~WM8350_DACLRC_ENA;
+ u16 adc_lrc = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) &
+ ~WM8350_ADCLRC_ENA;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ master |= WM8350_BCLK_MSTR;
+ dac_lrc |= WM8350_DACLRC_ENA;
+ adc_lrc |= WM8350_ADCLRC_ENA;
+ 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 |= 0x2 << 8;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x1 << 8;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x3 << 8;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x3 << 8; /* lg not sure which mode */
+ 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 |= WM8350_AIF_LRCLK_INV | WM8350_AIF_BCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= WM8350_AIF_BCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= WM8350_AIF_LRCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
+ wm8350_codec_write(codec, WM8350_AI_DAC_CONTROL, master);
+ wm8350_codec_write(codec, WM8350_DAC_LR_RATE, dac_lrc);
+ wm8350_codec_write(codec, WM8350_ADC_LR_RATE, adc_lrc);
+ return 0;
+}
+
+static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *codec_dai)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int master = wm8350_codec_cache_read(codec, WM8350_AI_DAC_CONTROL) &
+ WM8350_BCLK_MSTR;
+ int enabled = 0;
+
+ /* Check that the DACs or ADCs are enabled since they are
+ * required for LRC in master mode. The DACs or ADCs need a
+ * valid audio path i.e. pin -> ADC or DAC -> pin before
+ * the LRC will be enabled in master mode. */
+ if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+ return 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) &
+ (WM8350_ADCR_ENA | WM8350_ADCL_ENA);
+ } else {
+ enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) &
+ (WM8350_DACR_ENA | WM8350_DACL_ENA);
+ }
+
+ if (!enabled) {
+ dev_err(codec->dev,
+ "%s: invalid audio path - no clocks available\n",
+ __func__);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *codec_dai)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
+ ~WM8350_AIF_WL_MASK;
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x1 << 10;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x2 << 10;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= 0x3 << 10;
+ break;
+ }
+
+ wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
+ return 0;
+}
+
+static int wm8350_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8350 *wm8350 = codec->control_data;
+
+ if (mute)
+ wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+ else
+ wm8350_clear_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+ return 0;
+}
+
+/* FLL divisors */
+struct _fll_div {
+ int div; /* FLL_OUTDIV */
+ int n;
+ int k;
+ int ratio; /* FLL_FRATIO */
+};
+
+/* The size in bits of the fll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_FLL_SIZE ((1 << 16) * 10)
+
+static inline int fll_factors(struct _fll_div *fll_div, unsigned int input,
+ unsigned int output)
+{
+ u64 Kpart;
+ unsigned int t1, t2, K, Nmod;
+
+ if (output >= 2815250 && output <= 3125000)
+ fll_div->div = 0x4;
+ else if (output >= 5625000 && output <= 6250000)
+ fll_div->div = 0x3;
+ else if (output >= 11250000 && output <= 12500000)
+ fll_div->div = 0x2;
+ else if (output >= 22500000 && output <= 25000000)
+ fll_div->div = 0x1;
+ else {
+ printk(KERN_ERR "wm8350: fll freq %d out of range\n", output);
+ return -EINVAL;
+ }
+
+ if (input > 48000)
+ fll_div->ratio = 1;
+ else
+ fll_div->ratio = 8;
+
+ t1 = output * (1 << (fll_div->div + 1));
+ t2 = input * fll_div->ratio;
+
+ fll_div->n = t1 / t2;
+ Nmod = t1 % t2;
+
+ if (Nmod) {
+ Kpart = FIXED_FLL_SIZE * (long long)Nmod;
+ do_div(Kpart, t2);
+ K = Kpart & 0xFFFFFFFF;
+
+ /* Check if we need to round */
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ K /= 10;
+ fll_div->k = K;
+ } else
+ fll_div->k = 0;
+
+ return 0;
+}
+
+static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in,
+ unsigned int freq_out)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8350 *wm8350 = codec->control_data;
+ struct _fll_div fll_div;
+ int ret = 0;
+ u16 fll_1, fll_4;
+
+ /* power down FLL - we need to do this for reconfiguration */
+ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4,
+ WM8350_FLL_ENA | WM8350_FLL_OSC_ENA);
+
+ if (freq_out == 0 || freq_in == 0)
+ return ret;
+
+ ret = fll_factors(&fll_div, freq_in, freq_out);
+ if (ret < 0)
+ return ret;
+ dev_dbg(wm8350->dev,
+ "FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d",
+ freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div,
+ fll_div.ratio);
+
+ /* set up N.K & dividers */
+ fll_1 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_1) &
+ ~(WM8350_FLL_OUTDIV_MASK | WM8350_FLL_RSP_RATE_MASK | 0xc000);
+ wm8350_codec_write(codec, WM8350_FLL_CONTROL_1,
+ fll_1 | (fll_div.div << 8) | 0x50);
+ wm8350_codec_write(codec, WM8350_FLL_CONTROL_2,
+ (fll_div.ratio << 11) | (fll_div.
+ n & WM8350_FLL_N_MASK));
+ wm8350_codec_write(codec, WM8350_FLL_CONTROL_3, fll_div.k);
+ fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) &
+ ~(WM8350_FLL_FRAC | WM8350_FLL_SLOW_LOCK_REF);
+ wm8350_codec_write(codec, WM8350_FLL_CONTROL_4,
+ fll_4 | (fll_div.k ? WM8350_FLL_FRAC : 0) |
+ (fll_div.ratio == 8 ? WM8350_FLL_SLOW_LOCK_REF : 0));
+
+ /* power FLL on */
+ wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_OSC_ENA);
+ wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_ENA);
+
+ return 0;
+}
+
+static int wm8350_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct wm8350 *wm8350 = codec->control_data;
+ struct wm8350_data *priv = codec->private_data;
+ struct wm8350_audio_platform_data *platform =
+ wm8350->codec.platform_data;
+ u16 pm1;
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+ ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+ pm1 | WM8350_VMID_50K |
+ platform->codec_current_on << 14);
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1);
+ pm1 &= ~WM8350_VMID_MASK;
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+ pm1 | WM8350_VMID_50K);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies),
+ priv->supplies);
+ if (ret != 0)
+ return ret;
+
+ /* Enable the system clock */
+ wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4,
+ WM8350_SYSCLK_ENA);
+
+ /* mute DAC & outputs */
+ wm8350_set_bits(wm8350, WM8350_DAC_MUTE,
+ WM8350_DAC_MUTE_ENA);
+
+ /* discharge cap memory */
+ wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+ platform->dis_out1 |
+ (platform->dis_out2 << 2) |
+ (platform->dis_out3 << 4) |
+ (platform->dis_out4 << 6));
+
+ /* wait for discharge */
+ schedule_timeout_interruptible(msecs_to_jiffies
+ (platform->
+ cap_discharge_msecs));
+
+ /* enable antipop */
+ wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+ (platform->vmid_s_curve << 8));
+
+ /* ramp up vmid */
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+ (platform->
+ codec_current_charge << 14) |
+ WM8350_VMID_5K | WM8350_VMIDEN |
+ WM8350_VBUFEN);
+
+ /* wait for vmid */
+ schedule_timeout_interruptible(msecs_to_jiffies
+ (platform->
+ vmid_charge_msecs));
+
+ /* turn on vmid 300k */
+ pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+ ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+ pm1 |= WM8350_VMID_300K |
+ (platform->codec_current_standby << 14);
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+ pm1);
+
+
+ /* enable analogue bias */
+ pm1 |= WM8350_BIASEN;
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+ /* disable antipop */
+ wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0);
+
+ } else {
+ /* turn on vmid 300k and reduce current */
+ pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+ ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+ pm1 | WM8350_VMID_300K |
+ (platform->
+ codec_current_standby << 14));
+
+ }
+ break;
+
+ case SND_SOC_BIAS_OFF:
+
+ /* mute DAC & enable outputs */
+ wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+
+ wm8350_set_bits(wm8350, WM8350_POWER_MGMT_3,
+ WM8350_OUT1L_ENA | WM8350_OUT1R_ENA |
+ WM8350_OUT2L_ENA | WM8350_OUT2R_ENA);
+
+ /* enable anti pop S curve */
+ wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+ (platform->vmid_s_curve << 8));
+
+ /* turn off vmid */
+ pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+ ~WM8350_VMIDEN;
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+ /* wait */
+ schedule_timeout_interruptible(msecs_to_jiffies
+ (platform->
+ vmid_discharge_msecs));
+
+ wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+ (platform->vmid_s_curve << 8) |
+ platform->dis_out1 |
+ (platform->dis_out2 << 2) |
+ (platform->dis_out3 << 4) |
+ (platform->dis_out4 << 6));
+
+ /* turn off VBuf and drain */
+ pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+ ~(WM8350_VBUFEN | WM8350_VMID_MASK);
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+ pm1 | WM8350_OUTPUT_DRAIN_EN);
+
+ /* wait */
+ schedule_timeout_interruptible(msecs_to_jiffies
+ (platform->drain_msecs));
+
+ pm1 &= ~WM8350_BIASEN;
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+ /* disable anti-pop */
+ wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0);
+
+ wm8350_clear_bits(wm8350, WM8350_LOUT1_VOLUME,
+ WM8350_OUT1L_ENA);
+ wm8350_clear_bits(wm8350, WM8350_ROUT1_VOLUME,
+ WM8350_OUT1R_ENA);
+ wm8350_clear_bits(wm8350, WM8350_LOUT2_VOLUME,
+ WM8350_OUT2L_ENA);
+ wm8350_clear_bits(wm8350, WM8350_ROUT2_VOLUME,
+ WM8350_OUT2R_ENA);
+
+ /* disable clock gen */
+ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4,
+ WM8350_SYSCLK_ENA);
+
+ regulator_bulk_disable(ARRAY_SIZE(priv->supplies),
+ priv->supplies);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+static int wm8350_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8350_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+ wm8350_set_bias_level(codec, SND_SOC_BIAS_ON);
+
+ return 0;
+}
+
+static struct snd_soc_codec *wm8350_codec;
+
+static int wm8350_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct wm8350 *wm8350;
+ struct wm8350_data *priv;
+ int ret;
+ struct wm8350_output *out1;
+ struct wm8350_output *out2;
+
+ BUG_ON(!wm8350_codec);
+
+ socdev->codec = wm8350_codec;
+ codec = socdev->codec;
+ wm8350 = codec->control_data;
+ priv = codec->private_data;
+
+ /* Enable the codec */
+ wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+ /* Enable robust clocking mode in ADC */
+ wm8350_codec_write(codec, WM8350_SECURITY, 0xa7);
+ wm8350_codec_write(codec, 0xde, 0x13);
+ wm8350_codec_write(codec, WM8350_SECURITY, 0);
+
+ /* read OUT1 & OUT2 volumes */
+ out1 = &priv->out1;
+ out2 = &priv->out2;
+ out1->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME) &
+ WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+ out1->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME) &
+ WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+ out2->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME) &
+ WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+ out2->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME) &
+ WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+ wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME, 0);
+ wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME, 0);
+ wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME, 0);
+ wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME, 0);
+
+ /* Latch VU bits & mute */
+ wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME,
+ WM8350_OUT1_VU | WM8350_OUT1L_MUTE);
+ wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME,
+ WM8350_OUT2_VU | WM8350_OUT2L_MUTE);
+ wm8350_set_bits(wm8350, WM8350_ROUT1_VOLUME,
+ WM8350_OUT1_VU | WM8350_OUT1R_MUTE);
+ wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
+ WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
+
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "failed to create pcms\n");
+ return ret;
+ }
+
+ wm8350_add_controls(codec);
+ wm8350_add_widgets(codec);
+
+ wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "failed to register card\n");
+ goto card_err;
+ }
+
+ return 0;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+ return ret;
+}
+
+static int wm8350_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8350 *wm8350 = codec->control_data;
+ int ret;
+
+ /* cancel any work waiting to be queued. */
+ ret = cancel_delayed_work(&codec->delayed_work);
+
+ /* if there was any work waiting then we run it now and
+ * wait for its completion */
+ if (ret) {
+ schedule_delayed_work(&codec->delayed_work, 0);
+ flush_scheduled_work();
+ }
+
+ wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+ return 0;
+}
+
+#define WM8350_RATES (SNDRV_PCM_RATE_8000_96000)
+
+#define WM8350_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_dai wm8350_dai = {
+ .name = "WM8350",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8350_RATES,
+ .formats = WM8350_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8350_RATES,
+ .formats = WM8350_FORMATS,
+ },
+ .ops = {
+ .hw_params = wm8350_pcm_hw_params,
+ .digital_mute = wm8350_mute,
+ .trigger = wm8350_pcm_trigger,
+ .set_fmt = wm8350_set_dai_fmt,
+ .set_sysclk = wm8350_set_dai_sysclk,
+ .set_pll = wm8350_set_fll,
+ .set_clkdiv = wm8350_set_clkdiv,
+ },
+};
+EXPORT_SYMBOL_GPL(wm8350_dai);
+
+struct snd_soc_codec_device soc_codec_dev_wm8350 = {
+ .probe = wm8350_probe,
+ .remove = wm8350_remove,
+ .suspend = wm8350_suspend,
+ .resume = wm8350_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8350);
+
+static int wm8350_codec_probe(struct platform_device *pdev)
+{
+ struct wm8350 *wm8350 = platform_get_drvdata(pdev);
+ struct wm8350_data *priv;
+ struct snd_soc_codec *codec;
+ int ret, i;
+
+ if (wm8350->codec.platform_data == NULL) {
+ dev_err(&pdev->dev, "No audio platform data supplied\n");
+ return -EINVAL;
+ }
+
+ priv = kzalloc(sizeof(struct wm8350_data), GFP_KERNEL);
+ if (priv == NULL)
+ return -ENOMEM;
+
+ for (i = 0; i < ARRAY_SIZE(supply_names); i++)
+ priv->supplies[i].supply = supply_names[i];
+
+ ret = regulator_bulk_get(wm8350->dev, ARRAY_SIZE(priv->supplies),
+ priv->supplies);
+ if (ret != 0)
+ goto err_priv;
+
+ codec = &priv->codec;
+ wm8350->codec.codec = codec;
+
+ wm8350_dai.dev = &pdev->dev;
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+ codec->dev = &pdev->dev;
+ codec->name = "WM8350";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8350_codec_read;
+ codec->write = wm8350_codec_write;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8350_set_bias_level;
+ codec->dai = &wm8350_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = WM8350_MAX_REGISTER;
+ codec->private_data = priv;
+ codec->control_data = wm8350;
+
+ /* Put the codec into reset if it wasn't already */
+ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+ INIT_DELAYED_WORK(&codec->delayed_work, wm8350_pga_work);
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0)
+ goto err_supply;
+
+ wm8350_codec = codec;
+
+ ret = snd_soc_register_dai(&wm8350_dai);
+ if (ret != 0)
+ goto err_codec;
+ return 0;
+
+err_codec:
+ snd_soc_unregister_codec(codec);
+err_supply:
+ regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies);
+err_priv:
+ kfree(priv);
+ wm8350_codec = NULL;
+ return ret;
+}
+
+static int __devexit wm8350_codec_remove(struct platform_device *pdev)
+{
+ struct wm8350 *wm8350 = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = wm8350->codec.codec;
+ struct wm8350_data *priv = codec->private_data;
+
+ snd_soc_unregister_dai(&wm8350_dai);
+ snd_soc_unregister_codec(codec);
+ regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies);
+ kfree(priv);
+ wm8350_codec = NULL;
+ return 0;
+}
+
+static struct platform_driver wm8350_codec_driver = {
+ .driver = {
+ .name = "wm8350-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8350_codec_probe,
+ .remove = __devexit_p(wm8350_codec_remove),
+};
+
+static __init int wm8350_init(void)
+{
+ return platform_driver_register(&wm8350_codec_driver);
+}
+module_init(wm8350_init);
+
+static __exit void wm8350_exit(void)
+{
+ platform_driver_unregister(&wm8350_codec_driver);
+}
+module_exit(wm8350_exit);
+
+MODULE_DESCRIPTION("ASoC WM8350 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm8350-codec");
diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h
new file mode 100644
index 00000000000..cc2887aa6c3
--- /dev/null
+++ b/sound/soc/codecs/wm8350.h
@@ -0,0 +1,20 @@
+/*
+ * wm8350.h - WM8903 audio codec interface
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ *
+ * 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 _WM8350_H
+#define _WM8350_H
+
+#include <sound/soc.h>
+
+extern struct snd_soc_dai wm8350_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8350;
+
+#endif
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index d8ca2da8d63..40f8238df71 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -463,7 +463,8 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai,
}
static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -585,8 +586,6 @@ struct snd_soc_dai wm8510_dai = {
.formats = WM8510_FORMATS,},
.ops = {
.hw_params = wm8510_pcm_hw_params,
- },
- .dai_ops = {
.digital_mute = wm8510_mute,
.set_fmt = wm8510_set_dai_fmt,
.set_clkdiv = wm8510_set_dai_clkdiv,
@@ -659,7 +658,7 @@ static int wm8510_init(struct snd_soc_device *socdev)
wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
wm8510_add_controls(codec);
wm8510_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm8510: failed to register card\n");
goto card_err;
@@ -890,6 +889,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8510 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510);
+static int __init wm8510_modinit(void)
+{
+ return snd_soc_register_dai(&wm8510_dai);
+}
+module_init(wm8510_modinit);
+
+static void __exit wm8510_exit(void)
+{
+ snd_soc_unregister_dai(&wm8510_dai);
+}
+module_exit(wm8510_exit);
+
MODULE_DESCRIPTION("ASoC WM8510 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 627ebfb4209..d004e584529 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -548,13 +548,13 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
* Set PCM DAI bit size and sample rate.
*/
static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai_link *dai = rtd->dai;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
- u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->codec_dai->id);
+ u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id);
paifb &= ~WM8580_AIF_LENGTH_MASK;
/* bit size */
@@ -574,7 +574,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- wm8580_write(codec, WM8580_PAIF3 + dai->codec_dai->id, paifb);
+ wm8580_write(codec, WM8580_PAIF3 + dai->id, paifb);
return 0;
}
@@ -798,8 +798,6 @@ struct snd_soc_dai wm8580_dai[] = {
},
.ops = {
.hw_params = wm8580_paif_hw_params,
- },
- .dai_ops = {
.set_fmt = wm8580_set_paif_dai_fmt,
.set_clkdiv = wm8580_set_dai_clkdiv,
.set_pll = wm8580_set_dai_pll,
@@ -818,8 +816,6 @@ struct snd_soc_dai wm8580_dai[] = {
},
.ops = {
.hw_params = wm8580_paif_hw_params,
- },
- .dai_ops = {
.set_fmt = wm8580_set_paif_dai_fmt,
.set_clkdiv = wm8580_set_dai_clkdiv,
.set_pll = wm8580_set_dai_pll,
@@ -873,7 +869,7 @@ static int wm8580_init(struct snd_soc_device *socdev)
wm8580_add_controls(codec);
wm8580_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm8580: failed to register card\n");
goto card_err;
@@ -900,85 +896,85 @@ static struct snd_soc_device *wm8580_socdev;
* low = 0x1a
* high = 0x1b
*/
-static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-
-/* Magic definition of all other variables and things */
-I2C_CLIENT_INSMOD;
-static struct i2c_driver wm8580_i2c_driver;
-static struct i2c_client client_template;
-
-static int wm8580_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+static int wm8580_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
{
struct snd_soc_device *socdev = wm8580_socdev;
- struct wm8580_setup_data *setup = socdev->codec_data;
struct snd_soc_codec *codec = socdev->codec;
- struct i2c_client *i2c;
int ret;
- if (addr != setup->i2c_address)
- return -ENODEV;
-
- client_template.adapter = adap;
- client_template.addr = addr;
-
- i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
- if (i2c == NULL) {
- kfree(codec);
- return -ENOMEM;
- }
i2c_set_clientdata(i2c, codec);
codec->control_data = i2c;
- ret = i2c_attach_client(i2c);
- if (ret < 0) {
- dev_err(&i2c->dev, "failed to attach codec at addr %x\n", addr);
- goto err;
- }
-
ret = wm8580_init(socdev);
- if (ret < 0) {
+ if (ret < 0)
dev_err(&i2c->dev, "failed to initialise WM8580\n");
- goto err;
- }
-
- return ret;
-
-err:
- kfree(codec);
- kfree(i2c);
return ret;
}
-static int wm8580_i2c_detach(struct i2c_client *client)
+static int wm8580_i2c_remove(struct i2c_client *client)
{
struct snd_soc_codec *codec = i2c_get_clientdata(client);
- i2c_detach_client(client);
kfree(codec->reg_cache);
- kfree(client);
return 0;
}
-static int wm8580_i2c_attach(struct i2c_adapter *adap)
-{
- return i2c_probe(adap, &addr_data, wm8580_codec_probe);
-}
+static const struct i2c_device_id wm8580_i2c_id[] = {
+ { "wm8580", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id);
-/* corgi i2c codec control layer */
static struct i2c_driver wm8580_i2c_driver = {
.driver = {
.name = "WM8580 I2C Codec",
.owner = THIS_MODULE,
},
- .attach_adapter = wm8580_i2c_attach,
- .detach_client = wm8580_i2c_detach,
- .command = NULL,
+ .probe = wm8580_i2c_probe,
+ .remove = wm8580_i2c_remove,
+ .id_table = wm8580_i2c_id,
};
-static struct i2c_client client_template = {
- .name = "WM8580",
- .driver = &wm8580_i2c_driver,
-};
+static int wm8580_add_i2c_device(struct platform_device *pdev,
+ const struct wm8580_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&wm8580_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "wm8580", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+
+ return 0;
+
+err_driver:
+ i2c_del_driver(&wm8580_i2c_driver);
+ return -ENODEV;
+}
#endif
static int wm8580_probe(struct platform_device *pdev)
@@ -1011,11 +1007,8 @@ static int wm8580_probe(struct platform_device *pdev)
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
if (setup->i2c_address) {
- normal_i2c[0] = setup->i2c_address;
codec->hw_write = (hw_write_t)i2c_master_send;
- ret = i2c_add_driver(&wm8580_i2c_driver);
- if (ret != 0)
- printk(KERN_ERR "can't add i2c driver");
+ ret = wm8580_add_i2c_device(pdev, setup);
}
#else
/* Add other interfaces here */
@@ -1034,6 +1027,7 @@ static int wm8580_remove(struct platform_device *pdev)
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_unregister_device(codec->control_data);
i2c_del_driver(&wm8580_i2c_driver);
#endif
kfree(codec->private_data);
@@ -1048,6 +1042,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8580 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580);
+static int __init wm8580_modinit(void)
+{
+ return snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+}
+module_init(wm8580_modinit);
+
+static void __exit wm8580_exit(void)
+{
+ snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+}
+module_exit(wm8580_exit);
+
MODULE_DESCRIPTION("ASoC WM8580 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h
index 589ddaba21d..09e4422f6f2 100644
--- a/sound/soc/codecs/wm8580.h
+++ b/sound/soc/codecs/wm8580.h
@@ -29,6 +29,7 @@
#define WM8580_CLKSRC_NONE 5
struct wm8580_setup_data {
+ int i2c_bus;
unsigned short i2c_address;
};
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
new file mode 100644
index 00000000000..80b11983e13
--- /dev/null
+++ b/sound/soc/codecs/wm8728.c
@@ -0,0 +1,585 @@
+/*
+ * wm8728.c -- WM8728 ALSA SoC Audio driver
+ *
+ * Copyright 2008 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License 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/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8728.h"
+
+struct snd_soc_codec_device soc_codec_dev_wm8728;
+
+/*
+ * We can't read the WM8728 register space so we cache them instead.
+ * Note that the defaults here aren't the physical defaults, we latch
+ * the volume update bits, mute the output and enable infinite zero
+ * detect.
+ */
+static const u16 wm8728_reg_defaults[] = {
+ 0x1ff,
+ 0x1ff,
+ 0x001,
+ 0x100,
+};
+
+static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+ return cache[reg];
+}
+
+static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+ cache[reg] = value;
+}
+
+/*
+ * write to the WM8728 register space
+ */
+static int wm8728_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 WM8728 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8728_write_reg_cache(codec, reg, value);
+
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1);
+
+static const struct snd_kcontrol_new wm8728_snd_controls[] = {
+
+SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL,
+ 0, 255, 0, wm8728_tlv),
+
+SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0),
+};
+
+static int wm8728_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8728_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/*
+ * DAPM controls.
+ */
+static const struct snd_soc_dapm_widget wm8728_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_OUTPUT("VOUTL"),
+SND_SOC_DAPM_OUTPUT("VOUTR"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"VOUTL", NULL, "DAC"},
+ {"VOUTR", NULL, "DAC"},
+};
+
+static int wm8728_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8728_dapm_widgets,
+ ARRAY_SIZE(wm8728_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(codec);
+
+ return 0;
+}
+
+static int wm8728_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+
+ if (mute)
+ wm8728_write(codec, WM8728_DACCTL, mute_reg | 1);
+ else
+ wm8728_write(codec, WM8728_DACCTL, mute_reg & ~1);
+
+ return 0;
+}
+
+static int wm8728_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+
+ dac &= ~0x18;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ dac |= 0x10;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ dac |= 0x08;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8728_write(codec, WM8728_DACCTL, dac);
+
+ return 0;
+}
+
+static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = wm8728_read_reg_cache(codec, WM8728_IFCTL);
+
+ /* Currently only I2S is supported by the driver, though the
+ * hardware is more flexible.
+ */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 1;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* The hardware only support full slave mode */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ iface &= ~0x22;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x20;
+ iface &= ~0x02;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x02;
+ iface &= ~0x20;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x22;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8728_write(codec, WM8728_IFCTL, iface);
+ return 0;
+}
+
+static int wm8728_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg;
+ int i;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* Power everything up... */
+ reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+ wm8728_write(codec, WM8728_DACCTL, reg & ~0x4);
+
+ /* ..then sync in the register cache. */
+ for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++)
+ wm8728_write(codec, i,
+ wm8728_read_reg_cache(codec, i));
+ }
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+ wm8728_write(codec, WM8728_DACCTL, reg | 0x4);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8728_RATES (SNDRV_PCM_RATE_8000_192000)
+
+#define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_dai wm8728_dai = {
+ .name = "WM8728",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = WM8728_RATES,
+ .formats = WM8728_FORMATS,
+ },
+ .ops = {
+ .hw_params = wm8728_hw_params,
+ .digital_mute = wm8728_mute,
+ .set_fmt = wm8728_set_dai_fmt,
+ }
+};
+EXPORT_SYMBOL_GPL(wm8728_dai);
+
+static int wm8728_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int wm8728_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8728_set_bias_level(codec, codec->suspend_bias_level);
+
+ return 0;
+}
+
+/*
+ * initialise the WM8728 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8728_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+
+ codec->name = "WM8728";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8728_read_reg_cache;
+ codec->write = wm8728_write;
+ codec->set_bias_level = wm8728_set_bias_level;
+ codec->dai = &wm8728_dai;
+ codec->num_dai = 1;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->reg_cache_size = ARRAY_SIZE(wm8728_reg_defaults);
+ codec->reg_cache = kmemdup(wm8728_reg_defaults,
+ sizeof(wm8728_reg_defaults),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8728: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* power on device */
+ wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ wm8728_add_controls(codec);
+ wm8728_add_widgets(codec);
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8728: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+static struct snd_soc_device *wm8728_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+/*
+ * WM8728 2 wire address is determined by GPIO5
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+
+static int wm8728_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = wm8728_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = wm8728_init(socdev);
+ if (ret < 0)
+ pr_err("failed to initialise WM8728\n");
+
+ return ret;
+}
+
+static int wm8728_i2c_remove(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ kfree(codec->reg_cache);
+ return 0;
+}
+
+static const struct i2c_device_id wm8728_i2c_id[] = {
+ { "wm8728", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id);
+
+static struct i2c_driver wm8728_i2c_driver = {
+ .driver = {
+ .name = "WM8728 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8728_i2c_probe,
+ .remove = wm8728_i2c_remove,
+ .id_table = wm8728_i2c_id,
+};
+
+static int wm8728_add_i2c_device(struct platform_device *pdev,
+ const struct wm8728_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&wm8728_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "wm8728", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+
+ return 0;
+
+err_driver:
+ i2c_del_driver(&wm8728_i2c_driver);
+ return -ENODEV;
+}
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8728_spi_probe(struct spi_device *spi)
+{
+ struct snd_soc_device *socdev = wm8728_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ codec->control_data = spi;
+
+ ret = wm8728_init(socdev);
+ if (ret < 0)
+ dev_err(&spi->dev, "failed to initialise WM8728\n");
+
+ return ret;
+}
+
+static int __devexit wm8728_spi_remove(struct spi_device *spi)
+{
+ return 0;
+}
+
+static struct spi_driver wm8728_spi_driver = {
+ .driver = {
+ .name = "wm8728",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8728_spi_probe,
+ .remove = __devexit_p(wm8728_spi_remove),
+};
+
+static int wm8728_spi_write(struct spi_device *spi, const char *data, int len)
+{
+ struct spi_transfer t;
+ struct spi_message m;
+ u8 msg[2];
+
+ if (len <= 0)
+ return 0;
+
+ msg[0] = data[0];
+ msg[1] = data[1];
+
+ spi_message_init(&m);
+ memset(&t, 0, (sizeof t));
+
+ t.tx_buf = &msg[0];
+ t.len = len;
+
+ spi_message_add_tail(&t, &m);
+ spi_sync(spi, &m);
+
+ return len;
+}
+#endif /* CONFIG_SPI_MASTER */
+
+static int wm8728_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8728_setup_data *setup;
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ wm8728_socdev = socdev;
+ ret = -ENODEV;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = wm8728_add_i2c_device(pdev, setup);
+ }
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ if (setup->spi) {
+ codec->hw_write = (hw_write_t)wm8728_spi_write;
+ ret = spi_register_driver(&wm8728_spi_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add spi driver");
+ }
+#endif
+
+ if (ret != 0)
+ kfree(codec);
+
+ return ret;
+}
+
+/* power down chip */
+static int wm8728_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_unregister_device(codec->control_data);
+ i2c_del_driver(&wm8728_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8728_spi_driver);
+#endif
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8728 = {
+ .probe = wm8728_probe,
+ .remove = wm8728_remove,
+ .suspend = wm8728_suspend,
+ .resume = wm8728_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8728);
+
+static int __init wm8728_modinit(void)
+{
+ return snd_soc_register_dai(&wm8728_dai);
+}
+module_init(wm8728_modinit);
+
+static void __exit wm8728_exit(void)
+{
+ snd_soc_unregister_dai(&wm8728_dai);
+}
+module_exit(wm8728_exit);
+
+MODULE_DESCRIPTION("ASoC WM8728 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8728.h b/sound/soc/codecs/wm8728.h
new file mode 100644
index 00000000000..d269c132474
--- /dev/null
+++ b/sound/soc/codecs/wm8728.h
@@ -0,0 +1,30 @@
+/*
+ * wm8728.h -- WM8728 ASoC codec driver
+ *
+ * Copyright 2008 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8728_H
+#define _WM8728_H
+
+#define WM8728_DACLVOL 0x00
+#define WM8728_DACRVOL 0x01
+#define WM8728_DACCTL 0x02
+#define WM8728_IFCTL 0x03
+
+struct wm8728_setup_data {
+ int spi;
+ int i2c_bus;
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai wm8728_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8728;
+
+#endif
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 7f8a7e36b33..c444b9f2701 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -264,7 +264,8 @@ static inline int get_coeff(int mclk, int rate)
}
static int wm8731_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -293,7 +294,8 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int wm8731_pcm_prepare(struct snd_pcm_substream *substream)
+static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -305,7 +307,8 @@ static int wm8731_pcm_prepare(struct snd_pcm_substream *substream)
return 0;
}
-static void wm8731_shutdown(struct snd_pcm_substream *substream)
+static void wm8731_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -461,8 +464,6 @@ struct snd_soc_dai wm8731_dai = {
.prepare = wm8731_pcm_prepare,
.hw_params = wm8731_hw_params,
.shutdown = wm8731_shutdown,
- },
- .dai_ops = {
.digital_mute = wm8731_mute,
.set_sysclk = wm8731_set_dai_sysclk,
.set_fmt = wm8731_set_dai_fmt,
@@ -544,7 +545,7 @@ static int wm8731_init(struct snd_soc_device *socdev)
wm8731_add_controls(codec);
wm8731_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm8731: failed to register card\n");
goto card_err;
@@ -792,6 +793,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
+static int __init wm8731_modinit(void)
+{
+ return snd_soc_register_dai(&wm8731_dai);
+}
+module_init(wm8731_modinit);
+
+static void __exit wm8731_exit(void)
+{
+ snd_soc_unregister_dai(&wm8731_dai);
+}
+module_exit(wm8731_exit);
+
MODULE_DESCRIPTION("ASoC WM8731 driver");
MODULE_AUTHOR("Richard Purdie");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 9b7296ee5b0..5997fa68e0d 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -614,7 +614,8 @@ static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai,
}
static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -709,8 +710,6 @@ struct snd_soc_dai wm8750_dai = {
.formats = WM8750_FORMATS,},
.ops = {
.hw_params = wm8750_pcm_hw_params,
- },
- .dai_ops = {
.digital_mute = wm8750_mute,
.set_fmt = wm8750_set_dai_fmt,
.set_sysclk = wm8750_set_dai_sysclk,
@@ -819,7 +818,7 @@ static int wm8750_init(struct snd_soc_device *socdev)
wm8750_add_controls(codec);
wm8750_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm8750: failed to register card\n");
goto card_err;
@@ -1086,6 +1085,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8750 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750);
+static int __init wm8750_modinit(void)
+{
+ return snd_soc_register_dai(&wm8750_dai);
+}
+module_init(wm8750_modinit);
+
+static void __exit wm8750_exit(void)
+{
+ snd_soc_unregister_dai(&wm8750_dai);
+}
+module_exit(wm8750_exit);
+
MODULE_DESCRIPTION("ASoC WM8750 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 8ede5bd66c1..fe1b46b3d71 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -922,7 +922,8 @@ static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai,
* Set PCM DAI bit size and sample rate.
*/
static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -1155,7 +1156,8 @@ static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
* Set PCM DAI bit size and sample rate.
*/
static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -1323,16 +1325,15 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
.channels_min = 1,
.channels_max = 2,
.rates = WM8753_RATES,
- .formats = WM8753_FORMATS,},
+ .formats = WM8753_FORMATS},
.capture = { /* dummy for fast DAI switching */
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8753_RATES,
- .formats = WM8753_FORMATS,},
+ .formats = WM8753_FORMATS},
.ops = {
- .hw_params = wm8753_i2s_hw_params,},
- .dai_ops = {
+ .hw_params = wm8753_i2s_hw_params,
.digital_mute = wm8753_mute,
.set_fmt = wm8753_mode1h_set_dai_fmt,
.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1356,8 +1357,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
.rates = WM8753_RATES,
.formats = WM8753_FORMATS,},
.ops = {
- .hw_params = wm8753_pcm_hw_params,},
- .dai_ops = {
+ .hw_params = wm8753_pcm_hw_params,
.digital_mute = wm8753_mute,
.set_fmt = wm8753_mode1v_set_dai_fmt,
.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1385,8 +1385,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
.rates = WM8753_RATES,
.formats = WM8753_FORMATS,},
.ops = {
- .hw_params = wm8753_pcm_hw_params,},
- .dai_ops = {
+ .hw_params = wm8753_pcm_hw_params,
.digital_mute = wm8753_mute,
.set_fmt = wm8753_mode2_set_dai_fmt,
.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1410,8 +1409,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
.rates = WM8753_RATES,
.formats = WM8753_FORMATS,},
.ops = {
- .hw_params = wm8753_i2s_hw_params,},
- .dai_ops = {
+ .hw_params = wm8753_i2s_hw_params,
.digital_mute = wm8753_mute,
.set_fmt = wm8753_mode3_4_set_dai_fmt,
.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1439,8 +1437,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
.rates = WM8753_RATES,
.formats = WM8753_FORMATS,},
.ops = {
- .hw_params = wm8753_i2s_hw_params,},
- .dai_ops = {
+ .hw_params = wm8753_i2s_hw_params,
.digital_mute = wm8753_mute,
.set_fmt = wm8753_mode3_4_set_dai_fmt,
.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1611,7 +1608,7 @@ static int wm8753_init(struct snd_soc_device *socdev)
wm8753_add_controls(codec);
wm8753_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm8753: failed to register card\n");
goto card_err;
@@ -1886,6 +1883,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8753 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
+static int __init wm8753_modinit(void)
+{
+ return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
+}
+module_init(wm8753_modinit);
+
+static void __exit wm8753_exit(void)
+{
+ snd_soc_unregister_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
+}
+module_exit(wm8753_exit);
+
MODULE_DESCRIPTION("ASoC WM8753 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 3b326c9b558..6767de10ded 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -138,6 +138,10 @@
struct snd_soc_codec_device soc_codec_dev_wm8900;
struct wm8900_priv {
+ struct snd_soc_codec codec;
+
+ u16 reg_cache[WM8900_MAXREG];
+
u32 fll_in; /* FLL input frequency */
u32 fll_out; /* FLL output frequency */
};
@@ -727,7 +731,8 @@ static int wm8900_add_widgets(struct snd_soc_codec *codec)
}
static int wm8900_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -1117,8 +1122,6 @@ struct snd_soc_dai wm8900_dai = {
},
.ops = {
.hw_params = wm8900_hw_params,
- },
- .dai_ops = {
.set_clkdiv = wm8900_set_dai_clkdiv,
.set_pll = wm8900_set_dai_pll,
.set_fmt = wm8900_set_dai_fmt,
@@ -1283,16 +1286,28 @@ static int wm8900_resume(struct platform_device *pdev)
return 0;
}
-/*
- * initialise the WM8900 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8900_init(struct snd_soc_device *socdev)
+static struct snd_soc_codec *wm8900_codec;
+
+static int wm8900_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
{
- struct snd_soc_codec *codec = socdev->codec;
- int ret = 0;
+ struct wm8900_priv *wm8900;
+ struct snd_soc_codec *codec;
unsigned int reg;
- struct i2c_client *i2c_client = socdev->codec->control_data;
+ int ret;
+
+ wm8900 = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL);
+ if (wm8900 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8900->codec;
+ codec->private_data = wm8900;
+ codec->reg_cache = &wm8900->reg_cache[0];
+ codec->reg_cache_size = WM8900_MAXREG;
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
codec->name = "WM8900";
codec->owner = THIS_MODULE;
@@ -1300,33 +1315,28 @@ static int wm8900_init(struct snd_soc_device *socdev)
codec->write = wm8900_write;
codec->dai = &wm8900_dai;
codec->num_dai = 1;
- codec->reg_cache_size = WM8900_MAXREG;
- codec->reg_cache = kmemdup(wm8900_reg_defaults,
- sizeof(wm8900_reg_defaults), GFP_KERNEL);
-
- if (codec->reg_cache == NULL)
- return -ENOMEM;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ codec->control_data = i2c;
+ codec->set_bias_level = wm8900_set_bias_level;
+ codec->dev = &i2c->dev;
reg = wm8900_read(codec, WM8900_REG_ID);
if (reg != 0x8900) {
- dev_err(&i2c_client->dev, "Device is not a WM8900 - ID %x\n",
- reg);
- return -ENODEV;
- }
-
- codec->private_data = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL);
- if (codec->private_data == NULL) {
- ret = -ENOMEM;
- goto priv_err;
+ dev_err(&i2c->dev, "Device is not a WM8900 - ID %x\n", reg);
+ ret = -ENODEV;
+ goto err;
}
/* Read back from the chip */
reg = wm8900_chip_read(codec, WM8900_REG_POWER1);
reg = (reg >> 12) & 0xf;
- dev_info(&i2c_client->dev, "WM8900 revision %d\n", reg);
+ dev_info(&i2c->dev, "WM8900 revision %d\n", reg);
wm8900_reset(codec);
+ /* Turn the chip on */
+ wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
/* Latch the volume update bits */
wm8900_write(codec, WM8900_REG_LINVOL,
wm8900_read(codec, WM8900_REG_LINVOL) | 0x100);
@@ -1352,160 +1362,98 @@ static int wm8900_init(struct snd_soc_device *socdev)
/* Set the DAC and mixer output bias */
wm8900_write(codec, WM8900_REG_OUTBIASCTL, 0x81);
- /* Register pcms */
- ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
- if (ret < 0) {
- dev_err(&i2c_client->dev, "Failed to register new PCMs\n");
- goto pcm_err;
- }
-
- /* Turn the chip on */
- codec->bias_level = SND_SOC_BIAS_OFF;
- wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- wm8900_add_controls(codec);
- wm8900_add_widgets(codec);
-
- ret = snd_soc_register_card(socdev);
- if (ret < 0) {
- dev_err(&i2c_client->dev, "Failed to register card\n");
- goto card_err;
- }
- return ret;
-
-card_err:
- snd_soc_free_pcms(socdev);
- snd_soc_dapm_free(socdev);
-pcm_err:
- kfree(codec->reg_cache);
-priv_err:
- kfree(codec->private_data);
- return ret;
-}
+ wm8900_dai.dev = &i2c->dev;
-static struct snd_soc_device *wm8900_socdev;
+ wm8900_codec = codec;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-
-static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-
-/* Magic definition of all other variables and things */
-I2C_CLIENT_INSMOD;
-
-static struct i2c_driver wm8900_i2c_driver;
-static struct i2c_client client_template;
-
-/* If the i2c layer weren't so broken, we could pass this kind of data
- around */
-static int wm8900_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-{
- struct snd_soc_device *socdev = wm8900_socdev;
- struct wm8900_setup_data *setup = socdev->codec_data;
- struct snd_soc_codec *codec = socdev->codec;
- struct i2c_client *i2c;
- int ret;
-
- if (addr != setup->i2c_address)
- return -ENODEV;
-
- dev_err(&adap->dev, "Probe on %x\n", addr);
-
- client_template.adapter = adap;
- client_template.addr = addr;
-
- i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
- if (i2c == NULL) {
- kfree(codec);
- return -ENOMEM;
- }
- i2c_set_clientdata(i2c, codec);
- codec->control_data = i2c;
-
- ret = i2c_attach_client(i2c);
- if (ret < 0) {
- dev_err(&adap->dev,
- "failed to attach codec at addr %x\n", addr);
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
goto err;
}
- ret = wm8900_init(socdev);
- if (ret < 0) {
- dev_err(&adap->dev, "failed to initialise WM8900\n");
- goto err;
+ ret = snd_soc_register_dai(&wm8900_dai);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to register DAI: %d\n", ret);
+ goto err_codec;
}
+
return ret;
+err_codec:
+ snd_soc_unregister_codec(codec);
err:
- kfree(codec);
- kfree(i2c);
+ kfree(wm8900);
+ wm8900_codec = NULL;
return ret;
}
-static int wm8900_i2c_detach(struct i2c_client *client)
+static int wm8900_i2c_remove(struct i2c_client *client)
{
- struct snd_soc_codec *codec = i2c_get_clientdata(client);
- i2c_detach_client(client);
- kfree(codec->reg_cache);
- kfree(client);
+ snd_soc_unregister_dai(&wm8900_dai);
+ snd_soc_unregister_codec(wm8900_codec);
+
+ wm8900_set_bias_level(wm8900_codec, SND_SOC_BIAS_OFF);
+
+ wm8900_dai.dev = NULL;
+ kfree(wm8900_codec->private_data);
+ wm8900_codec = NULL;
+
return 0;
}
-static int wm8900_i2c_attach(struct i2c_adapter *adap)
-{
- return i2c_probe(adap, &addr_data, wm8900_codec_probe);
-}
+static const struct i2c_device_id wm8900_i2c_id[] = {
+ { "wm8900", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id);
-/* corgi i2c codec control layer */
static struct i2c_driver wm8900_i2c_driver = {
.driver = {
- .name = "WM8900 I2C codec",
+ .name = "WM8900",
.owner = THIS_MODULE,
},
- .attach_adapter = wm8900_i2c_attach,
- .detach_client = wm8900_i2c_detach,
- .command = NULL,
-};
-
-static struct i2c_client client_template = {
- .name = "WM8900",
- .driver = &wm8900_i2c_driver,
+ .probe = wm8900_i2c_probe,
+ .remove = wm8900_i2c_remove,
+ .id_table = wm8900_i2c_id,
};
-#endif
static int wm8900_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct wm8900_setup_data *setup;
struct snd_soc_codec *codec;
int ret = 0;
- dev_info(&pdev->dev, "WM8900 Audio Codec\n");
-
- setup = socdev->codec_data;
- codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
- if (codec == NULL)
- return -ENOMEM;
-
- mutex_init(&codec->mutex);
- INIT_LIST_HEAD(&codec->dapm_widgets);
- INIT_LIST_HEAD(&codec->dapm_paths);
+ if (!wm8900_codec) {
+ dev_err(&pdev->dev, "I2C client not yet instantiated\n");
+ return -ENODEV;
+ }
+ codec = wm8900_codec;
socdev->codec = codec;
- codec->set_bias_level = wm8900_set_bias_level;
+ /* Register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "Failed to register new PCMs\n");
+ goto pcm_err;
+ }
- wm8900_socdev = socdev;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- if (setup->i2c_address) {
- normal_i2c[0] = setup->i2c_address;
- codec->hw_write = (hw_write_t)i2c_master_send;
- ret = i2c_add_driver(&wm8900_i2c_driver);
- if (ret != 0)
- printk(KERN_ERR "can't add i2c driver");
+ wm8900_add_controls(codec);
+ wm8900_add_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "Failed to register card\n");
+ goto card_err;
}
-#else
-#error Non-I2C interfaces not yet supported
-#endif
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
return ret;
}
@@ -1513,17 +1461,9 @@ static int wm8900_probe(struct platform_device *pdev)
static int wm8900_remove(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_codec *codec = socdev->codec;
-
- if (codec->control_data)
- wm8900_set_bias_level(codec, SND_SOC_BIAS_OFF);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- i2c_del_driver(&wm8900_i2c_driver);
-#endif
- kfree(codec);
return 0;
}
@@ -1536,6 +1476,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8900 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8900);
+static int __init wm8900_modinit(void)
+{
+ return i2c_add_driver(&wm8900_i2c_driver);
+}
+module_init(wm8900_modinit);
+
+static void __exit wm8900_exit(void)
+{
+ i2c_del_driver(&wm8900_i2c_driver);
+}
+module_exit(wm8900_exit);
+
MODULE_DESCRIPTION("ASoC WM8900 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfonmicro.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8900.h b/sound/soc/codecs/wm8900.h
index ba450d99e90..fd15007d10c 100644
--- a/sound/soc/codecs/wm8900.h
+++ b/sound/soc/codecs/wm8900.h
@@ -52,12 +52,6 @@
#define WM8900_DAC_CLKDIV_5_5 0x14
#define WM8900_DAC_CLKDIV_6 0x18
-#define WM8900_
-
-struct wm8900_setup_data {
- unsigned short i2c_address;
-};
-
extern struct snd_soc_dai wm8900_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8900;
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ce40d787760..bde74546db4 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -33,19 +33,6 @@
#include "wm8903.h"
-struct wm8903_priv {
- int sysclk;
-
- /* Reference counts */
- int charge_pump_users;
- int class_w_users;
- int playback_active;
- int capture_active;
-
- struct snd_pcm_substream *master_substream;
- struct snd_pcm_substream *slave_substream;
-};
-
/* Register defaults at reset */
static u16 wm8903_reg_defaults[] = {
0x8903, /* R0 - SW Reset and ID */
@@ -223,6 +210,23 @@ static u16 wm8903_reg_defaults[] = {
0x0000, /* R172 - Analogue Output Bias 0 */
};
+struct wm8903_priv {
+ struct snd_soc_codec codec;
+ u16 reg_cache[ARRAY_SIZE(wm8903_reg_defaults)];
+
+ int sysclk;
+
+ /* Reference counts */
+ int charge_pump_users;
+ int class_w_users;
+ int playback_active;
+ int capture_active;
+
+ struct snd_pcm_substream *master_substream;
+ struct snd_pcm_substream *slave_substream;
+};
+
+
static unsigned int wm8903_read_reg_cache(struct snd_soc_codec *codec,
unsigned int reg)
{
@@ -360,6 +364,8 @@ static void wm8903_sync_reg_cache(struct snd_soc_codec *codec, u16 *cache)
static void wm8903_reset(struct snd_soc_codec *codec)
{
wm8903_write(codec, WM8903_SW_RESET_AND_ID, 0);
+ memcpy(codec->reg_cache, wm8903_reg_defaults,
+ sizeof(wm8903_reg_defaults));
}
#define WM8903_OUTPUT_SHORT 0x8
@@ -392,6 +398,7 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
break;
default:
BUG();
+ return -EINVAL; /* Spurious warning from some compilers */
}
switch (w->shift) {
@@ -403,6 +410,7 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
break;
default:
BUG();
+ return -EINVAL; /* Spurious warning from some compilers */
}
if (event & SND_SOC_DAPM_PRE_PMU) {
@@ -773,14 +781,14 @@ static const struct snd_kcontrol_new left_output_mixer[] = {
SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0),
SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0),
SOC_DAPM_SINGLE_W("Left Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 1, 1, 0),
-SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 1, 1, 0),
+SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 0, 1, 0),
};
static const struct snd_kcontrol_new right_output_mixer[] = {
SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 3, 1, 0),
SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 2, 1, 0),
SOC_DAPM_SINGLE_W("Left Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 1, 1, 0),
-SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 1, 1, 0),
+SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 0, 1, 0),
};
static const struct snd_kcontrol_new left_speaker_mixer[] = {
@@ -788,7 +796,7 @@ SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 3, 1, 0),
SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 2, 1, 0),
SOC_DAPM_SINGLE("Left Bypass Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 1, 1, 0),
SOC_DAPM_SINGLE("Right Bypass Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0,
- 1, 1, 0),
+ 0, 1, 0),
};
static const struct snd_kcontrol_new right_speaker_mixer[] = {
@@ -797,7 +805,7 @@ SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0, 2, 1, 0),
SOC_DAPM_SINGLE("Left Bypass Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0,
1, 1, 0),
SOC_DAPM_SINGLE("Right Bypass Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0,
- 1, 1, 0),
+ 0, 1, 0),
};
static const struct snd_soc_dapm_widget wm8903_dapm_widgets[] = {
@@ -989,6 +997,9 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ wm8903_write(codec, WM8903_CLOCK_RATES_2,
+ WM8903_CLK_SYS_ENA);
+
wm8903_run_sequence(codec, 0);
wm8903_sync_reg_cache(codec, codec->reg_cache);
@@ -1019,6 +1030,9 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_OFF:
wm8903_run_sequence(codec, 32);
+ reg = wm8903_read(codec, WM8903_CLOCK_RATES_2);
+ reg &= ~WM8903_CLK_SYS_ENA;
+ wm8903_write(codec, WM8903_CLOCK_RATES_2, reg);
break;
}
@@ -1257,7 +1271,8 @@ static struct {
{ 0, 0 },
};
-static int wm8903_startup(struct snd_pcm_substream *substream)
+static int wm8903_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -1298,7 +1313,8 @@ static int wm8903_startup(struct snd_pcm_substream *substream)
return 0;
}
-static void wm8903_shutdown(struct snd_pcm_substream *substream)
+static void wm8903_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -1317,7 +1333,8 @@ static void wm8903_shutdown(struct snd_pcm_substream *substream)
}
static int wm8903_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -1515,8 +1532,6 @@ struct snd_soc_dai wm8903_dai = {
.startup = wm8903_startup,
.shutdown = wm8903_shutdown,
.hw_params = wm8903_hw_params,
- },
- .dai_ops = {
.digital_mute = wm8903_digital_mute,
.set_fmt = wm8903_set_dai_fmt,
.set_sysclk = wm8903_set_dai_sysclk
@@ -1560,39 +1575,48 @@ static int wm8903_resume(struct platform_device *pdev)
return 0;
}
-/*
- * initialise the WM8903 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8903_init(struct snd_soc_device *socdev)
+static struct snd_soc_codec *wm8903_codec;
+
+static int wm8903_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
{
- struct snd_soc_codec *codec = socdev->codec;
- struct i2c_client *i2c = codec->control_data;
- int ret = 0;
+ struct wm8903_priv *wm8903;
+ struct snd_soc_codec *codec;
+ int ret;
u16 val;
- val = wm8903_hw_read(codec, WM8903_SW_RESET_AND_ID);
- if (val != wm8903_reg_defaults[WM8903_SW_RESET_AND_ID]) {
- dev_err(&i2c->dev,
- "Device with ID register %x is not a WM8903\n", val);
- return -ENODEV;
- }
+ wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL);
+ if (wm8903 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8903->codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->dev = &i2c->dev;
codec->name = "WM8903";
codec->owner = THIS_MODULE;
codec->read = wm8903_read;
codec->write = wm8903_write;
+ codec->hw_write = (hw_write_t)i2c_master_send;
codec->bias_level = SND_SOC_BIAS_OFF;
codec->set_bias_level = wm8903_set_bias_level;
codec->dai = &wm8903_dai;
codec->num_dai = 1;
- codec->reg_cache_size = ARRAY_SIZE(wm8903_reg_defaults);
- codec->reg_cache = kmemdup(wm8903_reg_defaults,
- sizeof(wm8903_reg_defaults),
- GFP_KERNEL);
- if (codec->reg_cache == NULL) {
- dev_err(&i2c->dev, "Failed to allocate register cache\n");
- return -ENOMEM;
+ codec->reg_cache_size = ARRAY_SIZE(wm8903->reg_cache);
+ codec->reg_cache = &wm8903->reg_cache[0];
+ codec->private_data = wm8903;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ val = wm8903_hw_read(codec, WM8903_SW_RESET_AND_ID);
+ if (val != wm8903_reg_defaults[WM8903_SW_RESET_AND_ID]) {
+ dev_err(&i2c->dev,
+ "Device with ID register %x is not a WM8903\n", val);
+ return -ENODEV;
}
val = wm8903_read(codec, WM8903_REVISION_NUMBER);
@@ -1601,16 +1625,6 @@ static int wm8903_init(struct snd_soc_device *socdev)
wm8903_reset(codec);
- /* register pcms */
- ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
- if (ret < 0) {
- dev_err(&i2c->dev, "failed to create pcms\n");
- goto pcm_err;
- }
-
- /* SYSCLK is required for pretty much anything */
- wm8903_write(codec, WM8903_CLOCK_RATES_2, WM8903_CLK_SYS_ENA);
-
/* power on device */
wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -1645,47 +1659,45 @@ static int wm8903_init(struct snd_soc_device *socdev)
val |= WM8903_DAC_MUTEMODE;
wm8903_write(codec, WM8903_DAC_DIGITAL_1, val);
- wm8903_add_controls(codec);
- wm8903_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
- if (ret < 0) {
- dev_err(&i2c->dev, "wm8903: failed to register card\n");
- goto card_err;
+ wm8903_dai.dev = &i2c->dev;
+ wm8903_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+ goto err;
+ }
+
+ ret = snd_soc_register_dai(&wm8903_dai);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to register DAI: %d\n", ret);
+ goto err_codec;
}
return ret;
-card_err:
- snd_soc_free_pcms(socdev);
- snd_soc_dapm_free(socdev);
-pcm_err:
- kfree(codec->reg_cache);
+err_codec:
+ snd_soc_unregister_codec(codec);
+err:
+ wm8903_codec = NULL;
+ kfree(wm8903);
return ret;
}
-static struct snd_soc_device *wm8903_socdev;
-
-static int wm8903_i2c_probe(struct i2c_client *i2c,
- const struct i2c_device_id *id)
+static int wm8903_i2c_remove(struct i2c_client *client)
{
- struct snd_soc_device *socdev = wm8903_socdev;
- struct snd_soc_codec *codec = socdev->codec;
- int ret;
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
- i2c_set_clientdata(i2c, codec);
- codec->control_data = i2c;
+ snd_soc_unregister_dai(&wm8903_dai);
+ snd_soc_unregister_codec(codec);
- ret = wm8903_init(socdev);
- if (ret < 0)
- dev_err(&i2c->dev, "Device initialisation failed\n");
+ wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return ret;
-}
+ kfree(codec->private_data);
+
+ wm8903_codec = NULL;
+ wm8903_dai.dev = NULL;
-static int wm8903_i2c_remove(struct i2c_client *client)
-{
- struct snd_soc_codec *codec = i2c_get_clientdata(client);
- kfree(codec->reg_cache);
return 0;
}
@@ -1709,75 +1721,37 @@ static struct i2c_driver wm8903_i2c_driver = {
static int wm8903_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct wm8903_setup_data *setup;
- struct snd_soc_codec *codec;
- struct wm8903_priv *wm8903;
- struct i2c_board_info board_info;
- struct i2c_adapter *adapter;
- struct i2c_client *i2c_client;
int ret = 0;
- setup = socdev->codec_data;
-
- if (!setup->i2c_address) {
- dev_err(&pdev->dev, "No codec address provided\n");
- return -ENODEV;
+ if (!wm8903_codec) {
+ dev_err(&pdev->dev, "I2C device not yet probed\n");
+ goto err;
}
- codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
- if (codec == NULL)
- return -ENOMEM;
+ socdev->codec = wm8903_codec;
- wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL);
- if (wm8903 == NULL) {
- ret = -ENOMEM;
- goto err_codec;
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "failed to create pcms\n");
+ goto err;
}
- codec->private_data = wm8903;
- socdev->codec = codec;
- mutex_init(&codec->mutex);
- INIT_LIST_HEAD(&codec->dapm_widgets);
- INIT_LIST_HEAD(&codec->dapm_paths);
-
- wm8903_socdev = socdev;
+ wm8903_add_controls(socdev->codec);
+ wm8903_add_widgets(socdev->codec);
- codec->hw_write = (hw_write_t)i2c_master_send;
- ret = i2c_add_driver(&wm8903_i2c_driver);
- if (ret != 0) {
- dev_err(&pdev->dev, "can't add i2c driver\n");
- goto err_priv;
- } else {
- memset(&board_info, 0, sizeof(board_info));
- strlcpy(board_info.type, "wm8903", I2C_NAME_SIZE);
- board_info.addr = setup->i2c_address;
-
- adapter = i2c_get_adapter(setup->i2c_bus);
- if (!adapter) {
- dev_err(&pdev->dev, "Can't get I2C bus %d\n",
- setup->i2c_bus);
- ret = -ENODEV;
- goto err_adapter;
- }
-
- i2c_client = i2c_new_device(adapter, &board_info);
- i2c_put_adapter(adapter);
- if (i2c_client == NULL) {
- dev_err(&pdev->dev,
- "I2C driver registration failed\n");
- ret = -ENODEV;
- goto err_adapter;
- }
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "wm8903: failed to register card\n");
+ goto card_err;
}
return ret;
-err_adapter:
- i2c_del_driver(&wm8903_i2c_driver);
-err_priv:
- kfree(codec->private_data);
-err_codec:
- kfree(codec);
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+err:
return ret;
}
@@ -1792,10 +1766,6 @@ static int wm8903_remove(struct platform_device *pdev)
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
- i2c_unregister_device(socdev->codec->control_data);
- i2c_del_driver(&wm8903_i2c_driver);
- kfree(codec->private_data);
- kfree(codec);
return 0;
}
@@ -1808,6 +1778,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8903 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8903);
+static int __init wm8903_modinit(void)
+{
+ return i2c_add_driver(&wm8903_i2c_driver);
+}
+module_init(wm8903_modinit);
+
+static void __exit wm8903_exit(void)
+{
+ i2c_del_driver(&wm8903_i2c_driver);
+}
+module_exit(wm8903_exit);
+
MODULE_DESCRIPTION("ASoC WM8903 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.cm>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h
index cec622f2f66..0ea27e2b996 100644
--- a/sound/soc/codecs/wm8903.h
+++ b/sound/soc/codecs/wm8903.h
@@ -18,11 +18,6 @@
extern struct snd_soc_dai wm8903_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8903;
-struct wm8903_setup_data {
- int i2c_bus;
- int i2c_address;
-};
-
#define WM8903_MCLK_DIV_2 1
#define WM8903_CLK_SYS 2
#define WM8903_BCLK 3
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index f41a578ddd4..88ead7f8dd9 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -541,7 +541,8 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai,
}
static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -634,8 +635,6 @@ struct snd_soc_dai wm8971_dai = {
.formats = WM8971_FORMATS,},
.ops = {
.hw_params = wm8971_pcm_hw_params,
- },
- .dai_ops = {
.digital_mute = wm8971_mute,
.set_fmt = wm8971_set_dai_fmt,
.set_sysclk = wm8971_set_dai_sysclk,
@@ -748,7 +747,7 @@ static int wm8971_init(struct snd_soc_device *socdev)
wm8971_add_controls(codec);
wm8971_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm8971: failed to register card\n");
goto card_err;
@@ -936,6 +935,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8971 = {
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8971);
+static int __init wm8971_modinit(void)
+{
+ return snd_soc_register_dai(&wm8971_dai);
+}
+module_init(wm8971_modinit);
+
+static void __exit wm8971_exit(void)
+{
+ snd_soc_unregister_dai(&wm8971_dai);
+}
+module_exit(wm8971_exit);
+
MODULE_DESCRIPTION("ASoC WM8971 driver");
MODULE_AUTHOR("Lab126");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 572d22b0880..5b5afc14447 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -106,6 +106,7 @@ static const u16 wm8990_reg[] = {
0x0008, /* R60 - PLL1 */
0x0031, /* R61 - PLL2 */
0x0026, /* R62 - PLL3 */
+ 0x0000, /* R63 - Driver internal */
};
/*
@@ -126,10 +127,9 @@ static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec,
unsigned int reg, unsigned int value)
{
u16 *cache = codec->reg_cache;
- BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1);
- /* Reset register is uncached */
- if (reg == 0)
+ /* Reset register and reserved registers are uncached */
+ if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1)
return;
cache[reg] = value;
@@ -1172,7 +1172,8 @@ static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
* Set PCM DAI bit size and sample rate.
*/
static int wm8990_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -1222,8 +1223,14 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
switch (level) {
case SND_SOC_BIAS_ON:
break;
+
case SND_SOC_BIAS_PREPARE:
+ /* VMID=2*50k */
+ val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) &
+ ~WM8990_VMID_MODE_MASK;
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2);
break;
+
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_OFF) {
/* Enable all output discharge bits */
@@ -1272,10 +1279,17 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN);
- } else {
- /* ON -> standby */
+ /* Enable workaround for ADC clocking issue. */
+ wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0x2);
+ wm8990_write(codec, WM8990_EXT_CTL1, 0xa003);
+ wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0);
}
+
+ /* VMID=2*250k */
+ val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) &
+ ~WM8990_VMID_MODE_MASK;
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4);
break;
case SND_SOC_BIAS_OFF:
@@ -1349,8 +1363,7 @@ struct snd_soc_dai wm8990_dai = {
.rates = WM8990_RATES,
.formats = WM8990_FORMATS,},
.ops = {
- .hw_params = wm8990_hw_params,},
- .dai_ops = {
+ .hw_params = wm8990_hw_params,
.digital_mute = wm8990_mute,
.set_fmt = wm8990_set_dai_fmt,
.set_clkdiv = wm8990_set_dai_clkdiv,
@@ -1449,7 +1462,7 @@ static int wm8990_init(struct snd_soc_device *socdev)
wm8990_add_controls(codec);
wm8990_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm8990: failed to register card\n");
goto card_err;
@@ -1630,6 +1643,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8990 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990);
+static int __init wm8990_modinit(void)
+{
+ return snd_soc_register_dai(&wm8990_dai);
+}
+module_init(wm8990_modinit);
+
+static void __exit wm8990_exit(void)
+{
+ snd_soc_unregister_dai(&wm8990_dai);
+}
+module_exit(wm8990_exit);
+
MODULE_DESCRIPTION("ASoC WM8990 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h
index 0e192f3b078..7114ddc88b4 100644
--- a/sound/soc/codecs/wm8990.h
+++ b/sound/soc/codecs/wm8990.h
@@ -80,8 +80,8 @@
#define WM8990_PLL3 0x3E
#define WM8990_INTDRIVBITS 0x3F
-#define WM8990_REGISTER_COUNT 60
-#define WM8990_MAX_REGISTER 0x3F
+#define WM8990_EXT_ACCESS_ENA 0x75
+#define WM8990_EXT_CTL1 0x7a
/*
* Field Definitions.
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index ffb471e420e..af83d629078 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -487,7 +487,8 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
return 0;
}
-static int ac97_prepare(struct snd_pcm_substream *substream)
+static int ac97_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -507,7 +508,8 @@ static int ac97_prepare(struct snd_pcm_substream *substream)
return ac97_write(codec, reg, runtime->rate);
}
-static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+static int ac97_aux_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -533,7 +535,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
struct snd_soc_dai wm9712_dai[] = {
{
.name = "AC97 HiFi",
- .type = SND_SOC_DAI_AC97_BUS,
+ .ac97_control = 1,
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
@@ -688,7 +690,7 @@ static int wm9712_soc_probe(struct platform_device *pdev)
ret = wm9712_reset(codec, 0);
if (ret < 0) {
- printk(KERN_ERR "AC97 link error\n");
+ printk(KERN_ERR "Failed to reset WM9712: AC97 link error\n");
goto reset_err;
}
@@ -698,7 +700,7 @@ static int wm9712_soc_probe(struct platform_device *pdev)
wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
wm9712_add_controls(codec);
wm9712_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm9712: failed to register card\n");
goto reset_err;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 945b32ed988..f3ca8aaf013 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -928,11 +928,10 @@ static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai,
}
static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->codec;
+ struct snd_soc_codec *codec = dai->codec;
u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3;
switch (params_format(params)) {
@@ -954,11 +953,10 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static void wm9713_voiceshutdown(struct snd_pcm_substream *substream)
+static void wm9713_voiceshutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->codec;
+ struct snd_soc_codec *codec = dai->codec;
u16 status;
/* Gracefully shut down the voice interface. */
@@ -969,12 +967,11 @@ static void wm9713_voiceshutdown(struct snd_pcm_substream *substream)
ac97_write(codec, AC97_EXTENDED_MID, status);
}
-static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
+static int ac97_hifi_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
+ struct snd_soc_codec *codec = dai->codec;
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->codec;
int reg;
u16 vra;
@@ -989,12 +986,11 @@ static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
return ac97_write(codec, reg, runtime->rate);
}
-static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+static int ac97_aux_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
+ struct snd_soc_codec *codec = dai->codec;
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->codec;
u16 vra, xsle;
vra = ac97_read(codec, AC97_EXTENDED_STATUS);
@@ -1028,7 +1024,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
struct snd_soc_dai wm9713_dai[] = {
{
.name = "AC97 HiFi",
- .type = SND_SOC_DAI_AC97_BUS,
+ .ac97_control = 1,
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
@@ -1042,8 +1038,7 @@ struct snd_soc_dai wm9713_dai[] = {
.rates = WM9713_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.ops = {
- .prepare = ac97_hifi_prepare,},
- .dai_ops = {
+ .prepare = ac97_hifi_prepare,
.set_clkdiv = wm9713_set_dai_clkdiv,
.set_pll = wm9713_set_dai_pll,},
},
@@ -1056,8 +1051,7 @@ struct snd_soc_dai wm9713_dai[] = {
.rates = WM9713_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.ops = {
- .prepare = ac97_aux_prepare,},
- .dai_ops = {
+ .prepare = ac97_aux_prepare,
.set_clkdiv = wm9713_set_dai_clkdiv,
.set_pll = wm9713_set_dai_pll,},
},
@@ -1077,8 +1071,7 @@ struct snd_soc_dai wm9713_dai[] = {
.formats = WM9713_PCM_FORMATS,},
.ops = {
.hw_params = wm9713_pcm_hw_params,
- .shutdown = wm9713_voiceshutdown,},
- .dai_ops = {
+ .shutdown = wm9713_voiceshutdown,
.set_clkdiv = wm9713_set_dai_clkdiv,
.set_pll = wm9713_set_dai_pll,
.set_fmt = wm9713_set_dai_fmt,
@@ -1097,6 +1090,8 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm)
}
soc_ac97_ops.reset(codec->ac97);
+ if (soc_ac97_ops.warm_reset)
+ soc_ac97_ops.warm_reset(codec->ac97);
if (ac97_read(codec, 0) != wm9713_reg[0])
return -EIO;
return 0;
@@ -1240,7 +1235,7 @@ static int wm9713_soc_probe(struct platform_device *pdev)
wm9713_reset(codec, 0);
ret = wm9713_reset(codec, 1);
if (ret < 0) {
- printk(KERN_ERR "AC97 link error\n");
+ printk(KERN_ERR "Failed to reset WM9713: AC97 link error\n");
goto reset_err;
}
@@ -1252,7 +1247,7 @@ static int wm9713_soc_probe(struct platform_device *pdev)
wm9713_add_controls(codec);
wm9713_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0)
goto reset_err;
return 0;
@@ -1288,7 +1283,6 @@ static int wm9713_soc_remove(struct platform_device *pdev)
snd_soc_free_ac97_codec(codec);
kfree(codec->private_data);
kfree(codec->reg_cache);
- kfree(codec->dai);
kfree(codec);
return 0;
}