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-rw-r--r--sound/soc/codecs/Kconfig16
-rw-r--r--sound/soc/codecs/Makefile4
-rw-r--r--sound/soc/codecs/ad193x.c41
-rw-r--r--sound/soc/codecs/ad193x.h5
-rw-r--r--sound/soc/codecs/ak4642.c32
-rw-r--r--sound/soc/codecs/cs42l51.c763
-rw-r--r--sound/soc/codecs/cs42l51.h163
-rw-r--r--sound/soc/codecs/da7210.c9
-rw-r--r--sound/soc/codecs/jz4740.c511
-rw-r--r--sound/soc/codecs/jz4740.h20
-rw-r--r--sound/soc/codecs/spdif_transciever.c94
-rw-r--r--sound/soc/codecs/spdif_transciever.h1
-rw-r--r--sound/soc/codecs/tlv320aic23.c7
-rw-r--r--sound/soc/codecs/tlv320dac33.c57
-rw-r--r--sound/soc/codecs/twl4030.c291
-rw-r--r--sound/soc/codecs/twl4030.h3
-rw-r--r--sound/soc/codecs/twl6040.c2
-rw-r--r--sound/soc/codecs/uda134x.c64
-rw-r--r--sound/soc/codecs/uda134x.h5
-rw-r--r--sound/soc/codecs/wm2000.c2
-rw-r--r--sound/soc/codecs/wm8750.c11
-rw-r--r--sound/soc/codecs/wm8960.c99
-rw-r--r--sound/soc/codecs/wm8990.c4
-rw-r--r--sound/soc/codecs/wm8994.c43
-rw-r--r--sound/soc/codecs/wm8994.h3
25 files changed, 2050 insertions, 200 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 31ac553..ea1f5ed 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -22,9 +22,11 @@ config SND_SOC_ALL_CODECS
select SND_SOC_AK4642 if I2C
select SND_SOC_AK4671 if I2C
select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
+ select SND_SOC_CS42L51 if I2C
select SND_SOC_CS4270 if I2C
- select SND_SOC_MAX9877 if I2C
select SND_SOC_DA7210 if I2C
+ select SND_SOC_JZ4740 if SOC_JZ4740
+ select SND_SOC_MAX9877 if I2C
select SND_SOC_PCM3008
select SND_SOC_SPDIF
select SND_SOC_SSM2602 if I2C
@@ -120,13 +122,13 @@ config SND_SOC_AK4671
config SND_SOC_CQ0093VC
tristate
+config SND_SOC_CS42L51
+ tristate
+
# Cirrus Logic CS4270 Codec
config SND_SOC_CS4270
tristate
-config SND_SOC_DA7210
- tristate
-
# Cirrus Logic CS4270 Codec VD = 3.3V Errata
# Select if you are affected by the errata where the part will not function
# if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will
@@ -138,9 +140,15 @@ config SND_SOC_CS4270_VD33_ERRATA
config SND_SOC_CX20442
tristate
+config SND_SOC_JZ4740_CODEC
+ tristate
+
config SND_SOC_L3
tristate
+config SND_SOC_DA7210
+ tristate
+
config SND_SOC_PCM3008
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 91429ea..d8d9eeb 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -9,6 +9,7 @@ snd-soc-ak4535-objs := ak4535.o
snd-soc-ak4642-objs := ak4642.o
snd-soc-ak4671-objs := ak4671.o
snd-soc-cq93vc-objs := cq93vc.o
+snd-soc-cs42l51-objs := cs42l51.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-cx20442-objs := cx20442.o
snd-soc-da7210-objs := da7210.o
@@ -56,6 +57,7 @@ snd-soc-wm9705-objs := wm9705.o
snd-soc-wm9712-objs := wm9712.o
snd-soc-wm9713-objs := wm9713.o
snd-soc-wm-hubs-objs := wm_hubs.o
+snd-soc-jz4740-codec-objs := jz4740.o
# Amp
snd-soc-max9877-objs := max9877.o
@@ -74,10 +76,12 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o
obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
+obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
+obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index c8ca114..1def75e 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -24,6 +24,7 @@
/* codec private data */
struct ad193x_priv {
+ unsigned int sysclk;
struct snd_soc_codec codec;
u8 reg_cache[AD193X_NUM_REGS];
};
@@ -251,15 +252,32 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
return 0;
}
+static int ad193x_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 ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
+ switch (freq) {
+ case 12288000:
+ case 18432000:
+ case 24576000:
+ case 36864000:
+ ad193x->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
static int ad193x_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- int word_len = 0, reg = 0;
+ int word_len = 0, reg = 0, master_rate = 0;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
+ struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
/* bit size */
switch (params_format(params)) {
@@ -275,6 +293,25 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
break;
}
+ switch (ad193x->sysclk) {
+ case 12288000:
+ master_rate = AD193X_PLL_INPUT_256;
+ break;
+ case 18432000:
+ master_rate = AD193X_PLL_INPUT_384;
+ break;
+ case 24576000:
+ master_rate = AD193X_PLL_INPUT_512;
+ break;
+ case 36864000:
+ master_rate = AD193X_PLL_INPUT_768;
+ break;
+ }
+
+ reg = snd_soc_read(codec, AD193X_PLL_CLK_CTRL0);
+ reg = (reg & AD193X_PLL_INPUT_MASK) | master_rate;
+ snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg);
+
reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len;
snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
@@ -348,6 +385,7 @@ static int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type)
/* pll input: mclki/xi */
snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */
snd_soc_write(codec, AD193X_PLL_CLK_CTRL1, 0x04);
+ ad193x->sysclk = 12288000;
ret = snd_soc_register_codec(codec);
if (ret != 0) {
@@ -383,6 +421,7 @@ static struct snd_soc_dai_ops ad193x_dai_ops = {
.hw_params = ad193x_hw_params,
.digital_mute = ad193x_mute,
.set_tdm_slot = ad193x_set_tdm_slot,
+ .set_sysclk = ad193x_set_dai_sysclk,
.set_fmt = ad193x_set_dai_fmt,
};
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index a03c880..654ba64 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -11,6 +11,11 @@
#define AD193X_PLL_CLK_CTRL0 0x800
#define AD193X_PLL_POWERDOWN 0x01
+#define AD193X_PLL_INPUT_MASK (~0x6)
+#define AD193X_PLL_INPUT_256 (0 << 1)
+#define AD193X_PLL_INPUT_384 (1 << 1)
+#define AD193X_PLL_INPUT_512 (2 << 1)
+#define AD193X_PLL_INPUT_768 (3 << 1)
#define AD193X_PLL_CLK_CTRL1 0x801
#define AD193X_DAC_CTRL0 0x802
#define AD193X_DAC_POWERDOWN 0x01
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 7528a54..60b83b4 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -22,20 +22,13 @@
* AK4643 is tested.
*/
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/init.h>
#include <linux/delay.h>
-#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/slab.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 "ak4642.h"
@@ -111,6 +104,23 @@
struct snd_soc_codec_device soc_codec_dev_ak4642;
+/*
+ * Playback Volume (table 39)
+ *
+ * max : 0x00 : +12.0 dB
+ * ( 0.5 dB step )
+ * min : 0xFE : -115.0 dB
+ * mute: 0xFF
+ */
+static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1);
+
+static const struct snd_kcontrol_new ak4642_snd_controls[] = {
+
+ SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
+ 0, 0xFF, 1, out_tlv),
+};
+
+
/* codec private data */
struct ak4642_priv {
struct snd_soc_codec codec;
@@ -204,7 +214,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
*
* PLL, Master Mode
* Audio I/F Format :MSB justified (ADC & DAC)
- * Digital Volume: -8dB
* Bass Boost Level : Middle
*
* This operation came from example code of
@@ -214,8 +223,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
ak4642_write(codec, 0x0e, 0x19);
ak4642_write(codec, 0x09, 0x91);
ak4642_write(codec, 0x0c, 0x91);
- ak4642_write(codec, 0x0a, 0x28);
- ak4642_write(codec, 0x0d, 0x28);
ak4642_write(codec, 0x00, 0x64);
snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP);
snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN);
@@ -548,6 +555,9 @@ static int ak4642_probe(struct platform_device *pdev)
goto pcm_err;
}
+ snd_soc_add_controls(ak4642_codec, ak4642_snd_controls,
+ ARRAY_SIZE(ak4642_snd_controls));
+
dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION);
return ret;
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
new file mode 100644
index 0000000..dd9b855
--- /dev/null
+++ b/sound/soc/codecs/cs42l51.c
@@ -0,0 +1,763 @@
+/*
+ * cs42l51.c
+ *
+ * ASoC Driver for Cirrus Logic CS42L51 codecs
+ *
+ * Copyright (c) 2010 Arnaud Patard <apatard@mandriva.com>
+ *
+ * Based on cs4270.c - Copyright (c) Freescale Semiconductor
+ *
+ * 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.
+ *
+ * For now:
+ * - Only I2C is support. Not SPI
+ * - master mode *NOT* supported
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/pcm.h>
+#include <linux/i2c.h>
+
+#include "cs42l51.h"
+
+enum master_slave_mode {
+ MODE_SLAVE,
+ MODE_SLAVE_AUTO,
+ MODE_MASTER,
+};
+
+struct cs42l51_private {
+ unsigned int mclk;
+ unsigned int audio_mode; /* The mode (I2S or left-justified) */
+ enum master_slave_mode func;
+ struct snd_soc_codec codec;
+ u8 reg_cache[CS42L51_NUMREGS];
+};
+
+static struct snd_soc_codec *cs42l51_codec;
+
+#define CS42L51_FORMATS ( \
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE)
+
+static int cs42l51_fill_cache(struct snd_soc_codec *codec)
+{
+ u8 *cache = codec->reg_cache + 1;
+ struct i2c_client *i2c_client = codec->control_data;
+ s32 length;
+
+ length = i2c_smbus_read_i2c_block_data(i2c_client,
+ CS42L51_FIRSTREG | 0x80, CS42L51_NUMREGS, cache);
+ if (length != CS42L51_NUMREGS) {
+ dev_err(&i2c_client->dev,
+ "I2C read failure, addr=0x%x (ret=%d vs %d)\n",
+ i2c_client->addr, length, CS42L51_NUMREGS);
+ return -EIO;
+ }
+
+ return 0;
+}
+
+static int cs42l51_i2c_probe(struct i2c_client *i2c_client,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_codec *codec;
+ struct cs42l51_private *cs42l51;
+ int ret = 0;
+ int reg;
+
+ if (cs42l51_codec)
+ return -EBUSY;
+
+ /* Verify that we have a CS42L51 */
+ ret = i2c_smbus_read_byte_data(i2c_client, CS42L51_CHIP_REV_ID);
+ if (ret < 0) {
+ dev_err(&i2c_client->dev, "failed to read I2C\n");
+ goto error;
+ }
+
+ if ((ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) &&
+ (ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) {
+ dev_err(&i2c_client->dev, "Invalid chip id\n");
+ ret = -ENODEV;
+ goto error;
+ }
+
+ dev_info(&i2c_client->dev, "found device cs42l51 rev %d\n",
+ ret & 7);
+
+ cs42l51 = kzalloc(sizeof(struct cs42l51_private), GFP_KERNEL);
+ if (!cs42l51) {
+ dev_err(&i2c_client->dev, "could not allocate codec\n");
+ return -ENOMEM;
+ }
+ codec = &cs42l51->codec;
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->dev = &i2c_client->dev;
+ codec->name = "CS42L51";
+ codec->owner = THIS_MODULE;
+ codec->dai = &cs42l51_dai;
+ codec->num_dai = 1;
+ snd_soc_codec_set_drvdata(codec, cs42l51);
+
+ codec->control_data = i2c_client;
+ codec->reg_cache = cs42l51->reg_cache;
+ codec->reg_cache_size = CS42L51_NUMREGS;
+ i2c_set_clientdata(i2c_client, codec);
+
+ ret = cs42l51_fill_cache(codec);
+ if (ret < 0) {
+ dev_err(&i2c_client->dev, "failed to fill register cache\n");
+ goto error_alloc;
+ }
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+ if (ret < 0) {
+ dev_err(&i2c_client->dev, "Failed to set cache I/O: %d\n", ret);
+ goto error_alloc;
+ }
+
+ /*
+ * DAC configuration
+ * - Use signal processor
+ * - auto mute
+ * - vol changes immediate
+ * - no de-emphasize
+ */
+ reg = CS42L51_DAC_CTL_DATA_SEL(1)
+ | CS42L51_DAC_CTL_AMUTE | CS42L51_DAC_CTL_DACSZ(0);
+ ret = snd_soc_write(codec, CS42L51_DAC_CTL, reg);
+ if (ret < 0)
+ goto error_alloc;
+
+ cs42l51_dai.dev = codec->dev;
+ cs42l51_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ goto error_alloc;
+ }
+
+ ret = snd_soc_register_dai(&cs42l51_dai);
+ if (ret < 0) {
+ dev_err(&i2c_client->dev, "failed to register DAIe\n");
+ goto error_reg;
+ }
+
+ return 0;
+
+error_reg:
+ snd_soc_unregister_codec(codec);
+error_alloc:
+ kfree(cs42l51);
+error:
+ return ret;
+}
+
+static int cs42l51_i2c_remove(struct i2c_client *client)
+{
+ struct cs42l51_private *cs42l51 = i2c_get_clientdata(client);
+ snd_soc_unregister_dai(&cs42l51_dai);
+ snd_soc_unregister_codec(cs42l51_codec);
+ cs42l51_codec = NULL;
+ kfree(cs42l51);
+ return 0;
+}
+
+
+static const struct i2c_device_id cs42l51_id[] = {
+ {"cs42l51", 0},
+ {}
+};
+MODULE_DEVICE_TABLE(i2c, cs42l51_id);
+
+static struct i2c_driver cs42l51_i2c_driver = {
+ .driver = {
+ .name = "CS42L51 I2C",
+ .owner = THIS_MODULE,
+ },
+ .id_table = cs42l51_id,
+ .probe = cs42l51_i2c_probe,
+ .remove = cs42l51_i2c_remove,
+};
+
+static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned long value = snd_soc_read(codec, CS42L51_PCM_MIXER)&3;
+
+ switch (value) {
+ default:
+ case 0:
+ ucontrol->value.integer.value[0] = 0;
+ break;
+ /* same value : (L+R)/2 and (R+L)/2 */
+ case 1:
+ case 2:
+ ucontrol->value.integer.value[0] = 1;
+ break;
+ case 3:
+ ucontrol->value.integer.value[0] = 2;
+ break;
+ }
+
+ return 0;
+}
+
+#define CHAN_MIX_NORMAL 0x00
+#define CHAN_MIX_BOTH 0x55
+#define CHAN_MIX_SWAP 0xFF
+
+static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned char val;
+
+ switch (ucontrol->value.integer.value[0]) {
+ default:
+ case 0:
+ val = CHAN_MIX_NORMAL;
+ break;
+ case 1:
+ val = CHAN_MIX_BOTH;
+ break;
+ case 2:
+ val = CHAN_MIX_SWAP;
+ break;
+ }
+
+ snd_soc_write(codec, CS42L51_PCM_MIXER, val);
+
+ return 1;
+}
+
+static const DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -5150, 50, 0);
+static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0);
+/* This is a lie. after -102 db, it stays at -102 */
+/* maybe a range would be better */
+static const DECLARE_TLV_DB_SCALE(aout_tlv, -11550, 50, 0);
+
+static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0);
+static const char *chan_mix[] = {
+ "L R",
+ "L+R",
+ "R L",
+};
+
+static const struct soc_enum cs42l51_chan_mix =
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(chan_mix), chan_mix);
+
+static const struct snd_kcontrol_new cs42l51_snd_controls[] = {
+ SOC_DOUBLE_R_SX_TLV("PCM Playback Volume",
+ CS42L51_PCMA_VOL, CS42L51_PCMB_VOL,
+ 7, 0xffffff99, 0x18, adc_pcm_tlv),
+ SOC_DOUBLE_R("PCM Playback Switch",
+ CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1),
+ SOC_DOUBLE_R_SX_TLV("Analog Playback Volume",
+ CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL,
+ 8, 0xffffff19, 0x18, aout_tlv),
+ SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
+ CS42L51_ADCA_VOL, CS42L51_ADCB_VOL,
+ 7, 0xffffff99, 0x18, adc_pcm_tlv),
+ SOC_DOUBLE_R("ADC Mixer Switch",
+ CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1),
+ SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0),
+ SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0),
+ SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0),
+ SOC_SINGLE("Zero Cross Switch", CS42L51_DAC_CTL, 0, 0, 0),
+ SOC_DOUBLE_TLV("Mic Boost Volume",
+ CS42L51_MIC_CTL, 0, 1, 1, 0, boost_tlv),
+ SOC_SINGLE_TLV("Bass Volume", CS42L51_TONE_CTL, 0, 0xf, 1, tone_tlv),
+ SOC_SINGLE_TLV("Treble Volume", CS42L51_TONE_CTL, 4, 0xf, 1, tone_tlv),
+ SOC_ENUM_EXT("PCM channel mixer",
+ cs42l51_chan_mix,
+ cs42l51_get_chan_mix, cs42l51_set_chan_mix),
+};
+
+/*
+ * to power down, one must:
+ * 1.) Enable the PDN bit
+ * 2.) enable power-down for the select channels
+ * 3.) disable the PDN bit.
+ */
+static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ unsigned long value;
+
+ value = snd_soc_read(w->codec, CS42L51_POWER_CTL1);
+ value &= ~CS42L51_POWER_CTL1_PDN;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMD:
+ value |= CS42L51_POWER_CTL1_PDN;
+ break;
+ default:
+ case SND_SOC_DAPM_POST_PMD:
+ break;
+ }
+ snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1,
+ CS42L51_POWER_CTL1_PDN, value);
+
+ return 0;
+}
+
+static const char *cs42l51_dac_names[] = {"Direct PCM",
+ "DSP PCM", "ADC"};
+static const struct soc_enum cs42l51_dac_mux_enum =
+ SOC_ENUM_SINGLE(CS42L51_DAC_CTL, 6, 3, cs42l51_dac_names);
+static const struct snd_kcontrol_new cs42l51_dac_mux_controls =
+ SOC_DAPM_ENUM("Route", cs42l51_dac_mux_enum);
+
+static const char *cs42l51_adcl_names[] = {"AIN1 Left", "AIN2 Left",
+ "MIC Left", "MIC+preamp Left"};
+static const struct soc_enum cs42l51_adcl_mux_enum =
+ SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 4, 4, cs42l51_adcl_names);
+static const struct snd_kcontrol_new cs42l51_adcl_mux_controls =
+ SOC_DAPM_ENUM("Route", cs42l51_adcl_mux_enum);
+
+static const char *cs42l51_adcr_names[] = {"AIN1 Right", "AIN2 Right",
+ "MIC Right", "MIC+preamp Right"};
+static const struct soc_enum cs42l51_adcr_mux_enum =
+ SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 6, 4, cs42l51_adcr_names);
+static const struct snd_kcontrol_new cs42l51_adcr_mux_controls =
+ SOC_DAPM_ENUM("Route", cs42l51_adcr_mux_enum);
+
+static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = {
+ SND_SOC_DAPM_MICBIAS("Mic Bias", CS42L51_MIC_POWER_CTL, 1, 1),
+ SND_SOC_DAPM_PGA_E("Left PGA", CS42L51_POWER_CTL1, 3, 1, NULL, 0,
+ cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+ SND_SOC_DAPM_PGA_E("Right PGA", CS42L51_POWER_CTL1, 4, 1, NULL, 0,
+ cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+ SND_SOC_DAPM_ADC_E("Left ADC", "Left HiFi Capture",
+ CS42L51_POWER_CTL1, 1, 1,
+ cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+ SND_SOC_DAPM_ADC_E("Right ADC", "Right HiFi Capture",
+ CS42L51_POWER_CTL1, 2, 1,
+ cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+ SND_SOC_DAPM_DAC_E("Left DAC", "Left HiFi Playback",
+ CS42L51_POWER_CTL1, 5, 1,
+ cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+ SND_SOC_DAPM_DAC_E("Right DAC", "Right HiFi Playback",
+ CS42L51_POWER_CTL1, 6, 1,
+ cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+
+ /* analog/mic */
+ SND_SOC_DAPM_INPUT("AIN1L"),
+ SND_SOC_DAPM_INPUT("AIN1R"),
+ SND_SOC_DAPM_INPUT("AIN2L"),
+ SND_SOC_DAPM_INPUT("AIN2R"),
+ SND_SOC_DAPM_INPUT("MICL"),
+ SND_SOC_DAPM_INPUT("MICR"),
+
+ SND_SOC_DAPM_MIXER("Mic Preamp Left",
+ CS42L51_MIC_POWER_CTL, 2, 1, NULL, 0),
+ SND_SOC_DAPM_MIXER("Mic Preamp Right",
+ CS42L51_MIC_POWER_CTL, 3, 1, NULL, 0),
+
+ /* HP */
+ SND_SOC_DAPM_OUTPUT("HPL"),
+ SND_SOC_DAPM_OUTPUT("HPR"),
+
+ /* mux */
+ SND_SOC_DAPM_MUX("DAC Mux", SND_SOC_NOPM, 0, 0,
+ &cs42l51_dac_mux_controls),
+ SND_SOC_DAPM_MUX("PGA-ADC Mux Left", SND_SOC_NOPM, 0, 0,
+ &cs42l51_adcl_mux_controls),
+ SND_SOC_DAPM_MUX("PGA-ADC Mux Right", SND_SOC_NOPM, 0, 0,
+ &cs42l51_adcr_mux_controls),
+};
+
+static const struct snd_soc_dapm_route cs42l51_routes[] = {
+ {"HPL", NULL, "Left DAC"},
+ {"HPR", NULL, "Right DAC"},
+
+ {"Left ADC", NULL, "Left PGA"},
+ {"Right ADC", NULL, "Right PGA"},
+
+ {"Mic Preamp Left", NULL, "MICL"},
+ {"Mic Preamp Right", NULL, "MICR"},
+
+ {"PGA-ADC Mux Left", "AIN1 Left", "AIN1L" },
+ {"PGA-ADC Mux Left", "AIN2 Left", "AIN2L" },
+ {"PGA-ADC Mux Left", "MIC Left", "MICL" },
+ {"PGA-ADC Mux Left", "MIC+preamp Left", "Mic Preamp Left" },
+ {"PGA-ADC Mux Right", "AIN1 Right", "AIN1R" },
+ {"PGA-ADC Mux Right", "AIN2 Right", "AIN2R" },
+ {"PGA-ADC Mux Right", "MIC Right", "MICR" },
+ {"PGA-ADC Mux Right", "MIC+preamp Right", "Mic Preamp Right" },
+
+ {"Left PGA", NULL, "PGA-ADC Mux Left"},
+ {"Right PGA", NULL, "PGA-ADC Mux Right"},
+};
+
+static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int format)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_LEFT_J:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ cs42l51->audio_mode = format & SND_SOC_DAIFMT_FORMAT_MASK;
+ break;
+ default:
+ dev_err(codec->dev, "invalid DAI format\n");
+ ret = -EINVAL;
+ }
+
+ switch (format & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ cs42l51->func = MODE_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ cs42l51->func = MODE_SLAVE_AUTO;
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+struct cs42l51_ratios {
+ unsigned int ratio;
+ unsigned char speed_mode;
+ unsigned char mclk;
+};
+
+static struct cs42l51_ratios slave_ratios[] = {
+ { 512, CS42L51_QSM_MODE, 0 }, { 768, CS42L51_QSM_MODE, 0 },
+ { 1024, CS42L51_QSM_MODE, 0 }, { 1536, CS42L51_QSM_MODE, 0 },
+ { 2048, CS42L51_QSM_MODE, 0 }, { 3072, CS42L51_QSM_MODE, 0 },
+ { 256, CS42L51_HSM_MODE, 0 }, { 384, CS42L51_HSM_MODE, 0 },
+ { 512, CS42L51_HSM_MODE, 0 }, { 768, CS42L51_HSM_MODE, 0 },
+ { 1024, CS42L51_HSM_MODE, 0 }, { 1536, CS42L51_HSM_MODE, 0 },
+ { 128, CS42L51_SSM_MODE, 0 }, { 192, CS42L51_SSM_MODE, 0 },
+ { 256, CS42L51_SSM_MODE, 0 }, { 384, CS42L51_SSM_MODE, 0 },
+ { 512, CS42L51_SSM_MODE, 0 }, { 768, CS42L51_SSM_MODE, 0 },
+ { 128, CS42L51_DSM_MODE, 0 }, { 192, CS42L51_DSM_MODE, 0 },
+ { 256, CS42L51_DSM_MODE, 0 }, { 384, CS42L51_DSM_MODE, 0 },
+};
+
+static struct cs42l51_ratios slave_auto_ratios[] = {
+ { 1024, CS42L51_QSM_MODE, 0 }, { 1536, CS42L51_QSM_MODE, 0 },
+ { 2048, CS42L51_QSM_MODE, 1 }, { 3072, CS42L51_QSM_MODE, 1 },
+ { 512, CS42L51_HSM_MODE, 0 }, { 768, CS42L51_HSM_MODE, 0 },
+ { 1024, CS42L51_HSM_MODE, 1 }, { 1536, CS42L51_HSM_MODE, 1 },
+ { 256, CS42L51_SSM_MODE, 0 }, { 384, CS42L51_SSM_MODE, 0 },
+ { 512, CS42L51_SSM_MODE, 1 }, { 768, CS42L51_SSM_MODE, 1 },
+ { 128, CS42L51_DSM_MODE, 0 }, { 192, CS42L51_DSM_MODE, 0 },
+ { 256, CS42L51_DSM_MODE, 1 }, { 384, CS42L51_DSM_MODE, 1 },
+};
+
+static int cs42l51_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 cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
+ struct cs42l51_ratios *ratios = NULL;
+ int nr_ratios = 0;
+ unsigned int rates = 0;
+ unsigned int rate_min = -1;
+ unsigned int rate_max = 0;
+ int i;
+
+ cs42l51->mclk = freq;
+
+ switch (cs42l51->func) {
+ case MODE_MASTER:
+ return -EINVAL;
+ case MODE_SLAVE:
+ ratios = slave_ratios;
+ nr_ratios = ARRAY_SIZE(slave_ratios);
+ break;
+ case MODE_SLAVE_AUTO:
+ ratios = slave_auto_ratios;
+ nr_ratios = ARRAY_SIZE(slave_auto_ratios);
+ break;
+ }
+
+ for (i = 0; i < nr_ratios; i++) {
+ unsigned int rate = freq / ratios[i].ratio;
+ rates |= snd_pcm_rate_to_rate_bit(rate);
+ if (rate < rate_min)
+ rate_min = rate;
+ if (rate > rate_max)
+ rate_max = rate;
+ }
+ rates &= ~SNDRV_PCM_RATE_KNOT;
+
+ if (!rates) {
+ dev_err(codec->dev, "could not find a valid sample rate\n");
+ return -EINVAL;
+ }
+
+ codec_dai->playback.rates = rates;
+ codec_dai->playback.rate_min = rate_min;
+ codec_dai->playback.rate_max = rate_max;
+
+ codec_dai->capture.rates = rates;
+ codec_dai->capture.rate_min = rate_min;
+ codec_dai->capture.rate_max = rate_max;
+
+ return 0;
+}
+
+static int cs42l51_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+ unsigned int i;
+ unsigned int rate;
+ unsigned int ratio;
+ struct cs42l51_ratios *ratios = NULL;
+ int nr_ratios = 0;
+ int intf_ctl, power_ctl, fmt;
+
+ switch (cs42l51->func) {
+ case MODE_MASTER:
+ return -EINVAL;
+ case MODE_SLAVE:
+ ratios = slave_ratios;
+ nr_ratios = ARRAY_SIZE(slave_ratios);
+ break;
+ case MODE_SLAVE_AUTO:
+ ratios = slave_auto_ratios;
+ nr_ratios = ARRAY_SIZE(slave_auto_ratios);
+ break;
+ }
+
+ /* Figure out which MCLK/LRCK ratio to use */
+ rate = params_rate(params); /* Sampling rate, in Hz */
+ ratio = cs42l51->mclk / rate; /* MCLK/LRCK ratio */
+ for (i = 0; i < nr_ratios; i++) {
+ if (ratios[i].ratio == ratio)
+ break;
+ }
+
+ if (i == nr_ratios) {
+ /* We did not find a matching ratio */
+ dev_err(codec->dev, "could not find matching ratio\n");
+ return -EINVAL;
+ }
+
+ intf_ctl = snd_soc_read(codec, CS42L51_INTF_CTL);
+ power_ctl = snd_soc_read(codec, CS42L51_MIC_POWER_CTL);
+
+ intf_ctl &= ~(CS42L51_INTF_CTL_MASTER | CS42L51_INTF_CTL_ADC_I2S
+ | CS42L51_INTF_CTL_DAC_FORMAT(7));
+ power_ctl &= ~(CS42L51_MIC_POWER_CTL_SPEED(3)
+ | CS42L51_MIC_POWER_CTL_MCLK_DIV2);
+
+ switch (cs42l51->func) {
+ case MODE_MASTER:
+ intf_ctl |= CS42L51_INTF_CTL_MASTER;
+ power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(ratios[i].speed_mode);
+ break;
+ case MODE_SLAVE:
+ power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(ratios[i].speed_mode);
+ break;
+ case MODE_SLAVE_AUTO:
+ power_ctl |= CS42L51_MIC_POWER_CTL_AUTO;
+ break;
+ }
+
+ switch (cs42l51->audio_mode) {
+ case SND_SOC_DAIFMT_I2S:
+ intf_ctl |= CS42L51_INTF_CTL_ADC_I2S;
+ intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_I2S);
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_LJ24);
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ case SNDRV_PCM_FORMAT_S16_BE:
+ fmt = CS42L51_DAC_DIF_RJ16;
+ break;
+ case SNDRV_PCM_FORMAT_S18_3LE:
+ case SNDRV_PCM_FORMAT_S18_3BE:
+ fmt = CS42L51_DAC_DIF_RJ18;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ case SNDRV_PCM_FORMAT_S20_3BE:
+ fmt = CS42L51_DAC_DIF_RJ20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ case SNDRV_PCM_FORMAT_S24_BE:
+ fmt = CS42L51_DAC_DIF_RJ24;
+ break;
+ default:
+ dev_err(codec->dev, "unknown format\n");
+ return -EINVAL;
+ }
+ intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(fmt);
+ break;
+ default:
+ dev_err(codec->dev, "unknown format\n");
+ return -EINVAL;
+ }
+
+ if (ratios[i].mclk)
+ power_ctl |= CS42L51_MIC_POWER_CTL_MCLK_DIV2;
+
+ ret = snd_soc_write(codec, CS42L51_INTF_CTL, intf_ctl);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_write(codec, CS42L51_MIC_POWER_CTL, power_ctl);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int cs42l51_dai_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ int reg;
+ int mask = CS42L51_DAC_OUT_CTL_DACA_MUTE|CS42L51_DAC_OUT_CTL_DACB_MUTE;
+
+ reg = snd_soc_read(codec, CS42L51_DAC_OUT_CTL);
+
+ if (mute)
+ reg |= mask;
+ else
+ reg &= ~mask;
+
+ return snd_soc_write(codec, CS42L51_DAC_OUT_CTL, reg);
+}
+
+static struct snd_soc_dai_ops cs42l51_dai_ops = {
+ .hw_params = cs42l51_hw_params,
+ .set_sysclk = cs42l51_set_dai_sysclk,
+ .set_fmt = cs42l51_set_dai_fmt,
+ .digital_mute = cs42l51_dai_mute,
+};
+
+struct snd_soc_dai cs42l51_dai = {
+ .name = "CS42L51 HiFi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = CS42L51_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = CS42L51_FORMATS,
+ },
+ .ops = &cs42l51_dai_ops,
+};
+EXPORT_SYMBOL_GPL(cs42l51_dai);
+
+
+static int cs42l51_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (!cs42l51_codec) {
+ dev_err(&pdev->dev, "CS42L51 codec not yet registered\n");
+ return -EINVAL;
+ }
+
+ socdev->card->codec = cs42l51_codec;
+ codec = socdev->card->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");
+ return ret;
+ }
+
+ snd_soc_add_controls(codec, cs42l51_snd_controls,
+ ARRAY_SIZE(cs42l51_snd_controls));
+ snd_soc_dapm_new_controls(codec, cs42l51_dapm_widgets,
+ ARRAY_SIZE(cs42l51_dapm_widgets));
+ snd_soc_dapm_add_routes(codec, cs42l51_routes,
+ ARRAY_SIZE(cs42l51_routes));
+
+ return 0;
+}
+
+
+static int cs42l51_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_device_cs42l51 = {
+ .probe = cs42l51_probe,
+ .remove = cs42l51_remove
+};
+EXPORT_SYMBOL_GPL(soc_codec_device_cs42l51);
+
+static int __init cs42l51_init(void)
+{
+ int ret;
+
+ ret = i2c_add_driver(&cs42l51_i2c_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "%s: can't add i2c driver\n", __func__);
+ return ret;
+ }
+ return 0;
+}
+module_init(cs42l51_init);
+
+static void __exit cs42l51_exit(void)
+{
+ i2c_del_driver(&cs42l51_i2c_driver);
+}
+module_exit(cs42l51_exit);
+
+MODULE_AUTHOR("Arnaud Patard <apatard@mandriva.com>");
+MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h
new file mode 100644
index 0000000..8f0bd97
--- /dev/null
+++ b/sound/soc/codecs/cs42l51.h
@@ -0,0 +1,163 @@
+/*
+ * cs42l51.h
+ *
+ * ASoC Driver for Cirrus Logic CS42L51 codecs
+ *
+ * Copyright (c) 2010 Arnaud Patard <apatard@mandriva.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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.
+ */
+#ifndef _CS42L51_H
+#define _CS42L51_H
+
+#define CS42L51_CHIP_ID 0x1B
+#define CS42L51_CHIP_REV_A 0x00
+#define CS42L51_CHIP_REV_B 0x01
+
+#define CS42L51_CHIP_REV_ID 0x01
+#define CS42L51_MK_CHIP_REV(a, b) ((a)<<3|(b))
+
+#define CS42L51_POWER_CTL1 0x02
+#define CS42L51_POWER_CTL1_PDN_DACB (1<<6)
+#define CS42L51_POWER_CTL1_PDN_DACA (1<<5)
+#define CS42L51_POWER_CTL1_PDN_PGAB (1<<4)
+#define CS42L51_POWER_CTL1_PDN_PGAA (1<<3)
+#define CS42L51_POWER_CTL1_PDN_ADCB (1<<2)
+#define CS42L51_POWER_CTL1_PDN_ADCA (1<<1)
+#define CS42L51_POWER_CTL1_PDN (1<<0)
+
+#define CS42L51_MIC_POWER_CTL 0x03
+#define CS42L51_MIC_POWER_CTL_AUTO (1<<7)
+#define CS42L51_MIC_POWER_CTL_SPEED(x) (((x)&3)<<5)
+#define CS42L51_QSM_MODE 3
+#define CS42L51_HSM_MODE 2
+#define CS42L51_SSM_MODE 1
+#define CS42L51_DSM_MODE 0
+#define CS42L51_MIC_POWER_CTL_3ST_SP (1<<4)
+#define CS42L51_MIC_POWER_CTL_PDN_MICB (1<<3)
+#define CS42L51_MIC_POWER_CTL_PDN_MICA (1<<2)
+#define CS42L51_MIC_POWER_CTL_PDN_BIAS (1<<1)
+#define CS42L51_MIC_POWER_CTL_MCLK_DIV2 (1<<0)
+
+#define CS42L51_INTF_CTL 0x04
+#define CS42L51_INTF_CTL_LOOPBACK (1<<7)
+#define CS42L51_INTF_CTL_MASTER (1<<6)
+#define CS42L51_INTF_CTL_DAC_FORMAT(x) (((x)&7)<<3)
+#define CS42L51_DAC_DIF_LJ24 0x00
+#define CS42L51_DAC_DIF_I2S 0x01
+#define CS42L51_DAC_DIF_RJ24 0x02
+#define CS42L51_DAC_DIF_RJ20 0x03
+#define CS42L51_DAC_DIF_RJ18 0x04
+#define CS42L51_DAC_DIF_RJ16 0x05
+#define CS42L51_INTF_CTL_ADC_I2S (1<<2)
+#define CS42L51_INTF_CTL_DIGMIX (1<<1)
+#define CS42L51_INTF_CTL_MICMIX (1<<0)
+
+#define CS42L51_MIC_CTL 0x05
+#define CS42L51_MIC_CTL_ADC_SNGVOL (1<<7)
+#define CS42L51_MIC_CTL_ADCD_DBOOST (1<<6)
+#define CS42L51_MIC_CTL_ADCA_DBOOST (1<<5)
+#define CS42L51_MIC_CTL_MICBIAS_SEL (1<<4)
+#define CS42L51_MIC_CTL_MICBIAS_LVL(x) (((x)&3)<<2)
+#define CS42L51_MIC_CTL_MICB_BOOST (1<<1)
+#define CS42L51_MIC_CTL_MICA_BOOST (1<<0)
+
+#define CS42L51_ADC_CTL 0x06
+#define CS42L51_ADC_CTL_ADCB_HPFEN (1<<7)
+#define CS42L51_ADC_CTL_ADCB_HPFRZ (1<<6)
+#define CS42L51_ADC_CTL_ADCA_HPFEN (1<<5)
+#define CS42L51_ADC_CTL_ADCA_HPFRZ (1<<4)
+#define CS42L51_ADC_CTL_SOFTB (1<<3)
+#define CS42L51_ADC_CTL_ZCROSSB (1<<2)
+#define CS42L51_ADC_CTL_SOFTA (1<<1)
+#define CS42L51_ADC_CTL_ZCROSSA (1<<0)
+
+#define CS42L51_ADC_INPUT 0x07
+#define CS42L51_ADC_INPUT_AINB_MUX(x) (((x)&3)<<6)
+#define CS42L51_ADC_INPUT_AINA_MUX(x) (((x)&3)<<4)
+#define CS42L51_ADC_INPUT_INV_ADCB (1<<3)
+#define CS42L51_ADC_INPUT_INV_ADCA (1<<2)
+#define CS42L51_ADC_INPUT_ADCB_MUTE (1<<1)
+#define CS42L51_ADC_INPUT_ADCA_MUTE (1<<0)
+
+#define CS42L51_DAC_OUT_CTL 0x08
+#define CS42L51_DAC_OUT_CTL_HP_GAIN(x) (((x)&7)<<5)
+#define CS42L51_DAC_OUT_CTL_DAC_SNGVOL (1<<4)
+#define CS42L51_DAC_OUT_CTL_INV_PCMB (1<<3)
+#define CS42L51_DAC_OUT_CTL_INV_PCMA (1<<2)
+#define CS42L51_DAC_OUT_CTL_DACB_MUTE (1<<1)
+#define CS42L51_DAC_OUT_CTL_DACA_MUTE (1<<0)
+
+#define CS42L51_DAC_CTL 0x09
+#define CS42L51_DAC_CTL_DATA_SEL(x) (((x)&3)<<6)
+#define CS42L51_DAC_CTL_FREEZE (1<<5)
+#define CS42L51_DAC_CTL_DEEMPH (1<<3)
+#define CS42L51_DAC_CTL_AMUTE (1<<2)
+#define CS42L51_DAC_CTL_DACSZ(x) (((x)&3)<<0)
+
+#define CS42L51_ALC_PGA_CTL 0x0A
+#define CS42L51_ALC_PGB_CTL 0x0B
+#define CS42L51_ALC_PGX_ALCX_SRDIS (1<<7)
+#define CS42L51_ALC_PGX_ALCX_ZCDIS (1<<6)
+#define CS42L51_ALC_PGX_PGX_VOL(x) (((x)&0x1f)<<0)
+
+#define CS42L51_ADCA_ATT 0x0C
+#define CS42L51_ADCB_ATT 0x0D
+
+#define CS42L51_ADCA_VOL 0x0E
+#define CS42L51_ADCB_VOL 0x0F
+#define CS42L51_PCMA_VOL 0x10
+#define CS42L51_PCMB_VOL 0x11
+#define CS42L51_MIX_MUTE_ADCMIX (1<<7)
+#define CS42L51_MIX_VOLUME(x) (((x)&0x7f)<<0)
+
+#define CS42L51_BEEP_FREQ 0x12
+#define CS42L51_BEEP_VOL 0x13
+#define CS42L51_BEEP_CONF 0x14
+
+#define CS42L51_TONE_CTL 0x15
+#define CS42L51_TONE_CTL_TREB(x) (((x)&0xf)<<4)
+#define CS42L51_TONE_CTL_BASS(x) (((x)&0xf)<<0)
+
+#define CS42L51_AOUTA_VOL 0x16
+#define CS42L51_AOUTB_VOL 0x17
+#define CS42L51_PCM_MIXER 0x18
+#define CS42L51_LIMIT_THRES_DIS 0x19
+#define CS42L51_LIMIT_REL 0x1A
+#define CS42L51_LIMIT_ATT 0x1B
+#define CS42L51_ALC_EN 0x1C
+#define CS42L51_ALC_REL 0x1D
+#define CS42L51_ALC_THRES 0x1E
+#define CS42L51_NOISE_CONF 0x1F
+
+#define CS42L51_STATUS 0x20
+#define CS42L51_STATUS_SP_CLKERR (1<<6)
+#define CS42L51_STATUS_SPEA_OVFL (1<<5)
+#define CS42L51_STATUS_SPEB_OVFL (1<<4)
+#define CS42L51_STATUS_PCMA_OVFL (1<<3)
+#define CS42L51_STATUS_PCMB_OVFL (1<<2)
+#define CS42L51_STATUS_ADCA_OVFL (1<<1)
+#define CS42L51_STATUS_ADCB_OVFL (1<<0)
+
+#define CS42L51_CHARGE_FREQ 0x21
+
+#define CS42L51_FIRSTREG 0x01
+/*
+ * Hack: with register 0x21, it makes 33 registers. Looks like someone in the
+ * i2c layer doesn't like i2c smbus block read of 33 regs. Workaround by using
+ * 32 regs
+ */
+#define CS42L51_LASTREG 0x20
+#define CS42L51_NUMREGS (CS42L51_LASTREG - CS42L51_FIRSTREG + 1)
+
+extern struct snd_soc_dai cs42l51_dai;
+extern struct snd_soc_codec_device soc_codec_device_cs42l51;
+#endif
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 75af2d6..a83aa18 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -15,23 +15,14 @@
* option) any later version.
*/
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/kernel.h>
-#include <linux/init.h>
#include <linux/delay.h>
-#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
-#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
-#include <sound/soc.h>
#include <sound/soc-dapm.h>
-#include <sound/tlv.h>
#include <sound/initval.h>
-#include <asm/div64.h>
#include "da7210.h"
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
new file mode 100644
index 0000000..66557de
--- /dev/null
+++ b/sound/soc/codecs/jz4740.c
@@ -0,0 +1,511 @@
+/*
+ * Copyright (C) 2009-2010, Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * 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.
+ *
+ * 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.,
+ * 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+
+#include <linux/delay.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc-dapm.h>
+#include <sound/soc.h>
+
+#define JZ4740_REG_CODEC_1 0x0
+#define JZ4740_REG_CODEC_2 0x1
+
+#define JZ4740_CODEC_1_LINE_ENABLE BIT(29)
+#define JZ4740_CODEC_1_MIC_ENABLE BIT(28)
+#define JZ4740_CODEC_1_SW1_ENABLE BIT(27)
+#define JZ4740_CODEC_1_ADC_ENABLE BIT(26)
+#define JZ4740_CODEC_1_SW2_ENABLE BIT(25)
+#define JZ4740_CODEC_1_DAC_ENABLE BIT(24)
+#define JZ4740_CODEC_1_VREF_DISABLE BIT(20)
+#define JZ4740_CODEC_1_VREF_AMP_DISABLE BIT(19)
+#define JZ4740_CODEC_1_VREF_PULLDOWN BIT(18)
+#define JZ4740_CODEC_1_VREF_LOW_CURRENT BIT(17)
+#define JZ4740_CODEC_1_VREF_HIGH_CURRENT BIT(16)
+#define JZ4740_CODEC_1_HEADPHONE_DISABLE BIT(14)
+#define JZ4740_CODEC_1_HEADPHONE_AMP_CHANGE_ANY BIT(13)
+#define JZ4740_CODEC_1_HEADPHONE_CHARGE BIT(12)
+#define JZ4740_CODEC_1_HEADPHONE_PULLDOWN (BIT(11) | BIT(10))
+#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M BIT(9)
+#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN BIT(8)
+#define JZ4740_CODEC_1_SUSPEND BIT(1)
+#define JZ4740_CODEC_1_RESET BIT(0)
+
+#define JZ4740_CODEC_1_LINE_ENABLE_OFFSET 29
+#define JZ4740_CODEC_1_MIC_ENABLE_OFFSET 28
+#define JZ4740_CODEC_1_SW1_ENABLE_OFFSET 27
+#define JZ4740_CODEC_1_ADC_ENABLE_OFFSET 26
+#define JZ4740_CODEC_1_SW2_ENABLE_OFFSET 25
+#define JZ4740_CODEC_1_DAC_ENABLE_OFFSET 24
+#define JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET 14
+#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN_OFFSET 8
+
+#define JZ4740_CODEC_2_INPUT_VOLUME_MASK 0x1f0000
+#define JZ4740_CODEC_2_SAMPLE_RATE_MASK 0x000f00
+#define JZ4740_CODEC_2_MIC_BOOST_GAIN_MASK 0x000030
+#define JZ4740_CODEC_2_HEADPHONE_VOLUME_MASK 0x000003
+
+#define JZ4740_CODEC_2_INPUT_VOLUME_OFFSET 16
+#define JZ4740_CODEC_2_SAMPLE_RATE_OFFSET 8
+#define JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET 4
+#define JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET 0
+
+static const uint32_t jz4740_codec_regs[] = {
+ 0x021b2302, 0x00170803,
+};
+
+struct jz4740_codec {
+ void __iomem *base;
+ struct resource *mem;
+
+ uint32_t reg_cache[2];
+ struct snd_soc_codec codec;
+};
+
+static inline struct jz4740_codec *codec_to_jz4740(struct snd_soc_codec *codec)
+{
+ return container_of(codec, struct jz4740_codec, codec);
+}
+
+static unsigned int jz4740_codec_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec);
+ return readl(jz4740_codec->base + (reg << 2));
+}
+
+static int jz4740_codec_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val)
+{
+ struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec);
+
+ jz4740_codec->reg_cache[reg] = val;
+ writel(val, jz4740_codec->base + (reg << 2));
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new jz4740_codec_controls[] = {
+ SOC_SINGLE("Master Playback Volume", JZ4740_REG_CODEC_2,
+ JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET, 3, 0),
+ SOC_SINGLE("Master Capture Volume", JZ4740_REG_CODEC_2,
+ JZ4740_CODEC_2_INPUT_VOLUME_OFFSET, 31, 0),
+ SOC_SINGLE("Master Playback Switch", JZ4740_REG_CODEC_1,
+ JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET, 1, 1),
+ SOC_SINGLE("Mic Capture Volume", JZ4740_REG_CODEC_2,
+ JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET, 3, 0),
+};
+
+static const struct snd_kcontrol_new jz4740_codec_output_controls[] = {
+ SOC_DAPM_SINGLE("Bypass Switch", JZ4740_REG_CODEC_1,
+ JZ4740_CODEC_1_SW1_ENABLE_OFFSET, 1, 0),
+ SOC_DAPM_SINGLE("DAC Switch", JZ4740_REG_CODEC_1,
+ JZ4740_CODEC_1_SW2_ENABLE_OFFSET, 1, 0),
+};
+
+static const struct snd_kcontrol_new jz4740_codec_input_controls[] = {
+ SOC_DAPM_SINGLE("Line Capture Switch", JZ4740_REG_CODEC_1,
+ JZ4740_CODEC_1_LINE_ENABLE_OFFSET, 1, 0),
+ SOC_DAPM_SINGLE("Mic Capture Switch", JZ4740_REG_CODEC_1,
+ JZ4740_CODEC_1_MIC_ENABLE_OFFSET, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget jz4740_codec_dapm_widgets[] = {
+ SND_SOC_DAPM_ADC("ADC", "Capture", JZ4740_REG_CODEC_1,
+ JZ4740_CODEC_1_ADC_ENABLE_OFFSET, 0),
+ SND_SOC_DAPM_DAC("DAC", "Playback", JZ4740_REG_CODEC_1,
+ JZ4740_CODEC_1_DAC_ENABLE_OFFSET, 0),
+
+ SND_SOC_DAPM_MIXER("Output Mixer", JZ4740_REG_CODEC_1,
+ JZ4740_CODEC_1_HEADPHONE_POWERDOWN_OFFSET, 1,
+ jz4740_codec_output_controls,
+ ARRAY_SIZE(jz4740_codec_output_controls)),
+
+ SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0,
+ jz4740_codec_input_controls,
+ ARRAY_SIZE(jz4740_codec_input_controls)),
+ SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_OUTPUT("LOUT"),
+ SND_SOC_DAPM_OUTPUT("ROUT"),
+
+ SND_SOC_DAPM_INPUT("MIC"),
+ SND_SOC_DAPM_INPUT("LIN"),
+ SND_SOC_DAPM_INPUT("RIN"),
+};
+
+static const struct snd_soc_dapm_route jz4740_codec_dapm_routes[] = {
+ {"Line Input", NULL, "LIN"},
+ {"Line Input", NULL, "RIN"},
+
+ {"Input Mixer", "Line Capture Switch", "Line Input"},
+ {"Input Mixer", "Mic Capture Switch", "MIC"},
+
+ {"ADC", NULL, "Input Mixer"},
+
+ {"Output Mixer", "Bypass Switch", "Input Mixer"},
+ {"Output Mixer", "DAC Switch", "DAC"},
+
+ {"LOUT", NULL, "Output Mixer"},
+ {"ROUT", NULL, "Output Mixer"},
+};
+
+static int jz4740_codec_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ uint32_t val;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ switch (params_rate(params)) {
+ case 8000:
+ val = 0;
+ break;
+ case 11025:
+ val = 1;
+ break;
+ case 12000:
+ val = 2;
+ break;
+ case 16000:
+ val = 3;
+ break;
+ case 22050:
+ val = 4;
+ break;
+ case 24000:
+ val = 5;
+ break;
+ case 32000:
+ val = 6;
+ break;
+ case 44100:
+ val = 7;
+ break;
+ case 48000:
+ val = 8;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ val <<= JZ4740_CODEC_2_SAMPLE_RATE_OFFSET;
+
+ snd_soc_update_bits(codec, JZ4740_REG_CODEC_2,
+ JZ4740_CODEC_2_SAMPLE_RATE_MASK, val);
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops jz4740_codec_dai_ops = {
+ .hw_params = jz4740_codec_hw_params,
+};
+
+struct snd_soc_dai jz4740_codec_dai = {
+ .name = "jz4740",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
+ },
+ .ops = &jz4740_codec_dai_ops,
+ .symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(jz4740_codec_dai);
+
+static void jz4740_codec_wakeup(struct snd_soc_codec *codec)
+{
+ int i;
+ uint32_t *cache = codec->reg_cache;
+
+ snd_soc_update_bits(codec, JZ4740_REG_CODEC_1,
+ JZ4740_CODEC_1_RESET, JZ4740_CODEC_1_RESET);
+ udelay(2);
+
+ snd_soc_update_bits(codec, JZ4740_REG_CODEC_1,
+ JZ4740_CODEC_1_SUSPEND | JZ4740_CODEC_1_RESET, 0);
+
+ for (i = 0; i < ARRAY_SIZE(jz4740_codec_regs); ++i)
+ jz4740_codec_write(codec, i, cache[i]);
+}
+
+static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ unsigned int mask;
+ unsigned int value;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ mask = JZ4740_CODEC_1_VREF_DISABLE |
+ JZ4740_CODEC_1_VREF_AMP_DISABLE |
+ JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M;
+ value = 0;
+
+ snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* The only way to clear the suspend flag is to reset the codec */
+ if (codec->bias_level == SND_SOC_BIAS_OFF)
+ jz4740_codec_wakeup(codec);
+
+ mask = JZ4740_CODEC_1_VREF_DISABLE |
+ JZ4740_CODEC_1_VREF_AMP_DISABLE |
+ JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M;
+ value = JZ4740_CODEC_1_VREF_DISABLE |
+ JZ4740_CODEC_1_VREF_AMP_DISABLE |
+ JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M;
+
+ snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value);
+ break;
+ case SND_SOC_BIAS_OFF:
+ mask = JZ4740_CODEC_1_SUSPEND;
+ value = JZ4740_CODEC_1_SUSPEND;
+
+ snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value);
+ break;
+ default:
+ break;
+ }
+
+ codec->bias_level = level;
+
+ return 0;
+}
+
+static struct snd_soc_codec *jz4740_codec_codec;
+
+static int jz4740_codec_dev_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = jz4740_codec_codec;
+
+ BUG_ON(!codec);
+
+ socdev->card->codec = codec;
+
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to create pcms: %d\n", ret);
+ return ret;
+ }
+
+ snd_soc_add_controls(codec, jz4740_codec_controls,
+ ARRAY_SIZE(jz4740_codec_controls));
+
+ snd_soc_dapm_new_controls(codec, jz4740_codec_dapm_widgets,
+ ARRAY_SIZE(jz4740_codec_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, jz4740_codec_dapm_routes,
+ ARRAY_SIZE(jz4740_codec_dapm_routes));
+
+ snd_soc_dapm_new_widgets(codec);
+
+ return 0;
+}
+
+static int jz4740_codec_dev_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+
+static int jz4740_codec_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int jz4740_codec_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+}
+
+#else
+#define jz4740_codec_suspend NULL
+#define jz4740_codec_resume NULL
+#endif
+
+struct snd_soc_codec_device soc_codec_dev_jz4740_codec = {
+ .probe = jz4740_codec_dev_probe,
+ .remove = jz4740_codec_dev_remove,
+ .suspend = jz4740_codec_suspend,
+ .resume = jz4740_codec_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_jz4740_codec);
+
+static int __devinit jz4740_codec_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct jz4740_codec *jz4740_codec;
+ struct snd_soc_codec *codec;
+ struct resource *mem;
+
+ jz4740_codec = kzalloc(sizeof(*jz4740_codec), GFP_KERNEL);
+ if (!jz4740_codec)
+ return -ENOMEM;
+
+ mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!mem) {
+ dev_err(&pdev->dev, "Failed to get mmio memory resource\n");
+ ret = -ENOENT;
+ goto err_free_codec;
+ }
+
+ mem = request_mem_region(mem->start, resource_size(mem), pdev->name);
+ if (!mem) {
+ dev_err(&pdev->dev, "Failed to request mmio memory region\n");
+ ret = -EBUSY;
+ goto err_free_codec;
+ }
+
+ jz4740_codec->base = ioremap(mem->start, resource_size(mem));
+ if (!jz4740_codec->base) {
+ dev_err(&pdev->dev, "Failed to ioremap mmio memory\n");
+ ret = -EBUSY;
+ goto err_release_mem_region;
+ }
+ jz4740_codec->mem = mem;
+
+ jz4740_codec_dai.dev = &pdev->dev;
+
+ codec = &jz4740_codec->codec;
+
+ codec->dev = &pdev->dev;
+ codec->name = "jz4740";
+ codec->owner = THIS_MODULE;
+
+ codec->read = jz4740_codec_read;
+ codec->write = jz4740_codec_write;
+ codec->set_bias_level = jz4740_codec_set_bias_level;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+
+ codec->dai = &jz4740_codec_dai;
+ codec->num_dai = 1;
+
+ codec->reg_cache = jz4740_codec->reg_cache;
+ codec->reg_cache_size = 2;
+ memcpy(codec->reg_cache, jz4740_codec_regs, sizeof(jz4740_codec_regs));
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ jz4740_codec_codec = codec;
+
+ snd_soc_update_bits(codec, JZ4740_REG_CODEC_1,
+ JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE);
+
+ platform_set_drvdata(pdev, jz4740_codec);
+
+ ret = snd_soc_register_codec(codec);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register codec\n");
+ goto err_iounmap;
+ }
+
+ ret = snd_soc_register_dai(&jz4740_codec_dai);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register codec dai\n");
+ goto err_unregister_codec;
+ }
+
+ jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+
+err_unregister_codec:
+ snd_soc_unregister_codec(codec);
+err_iounmap:
+ iounmap(jz4740_codec->base);
+err_release_mem_region:
+ release_mem_region(mem->start, resource_size(mem));
+err_free_codec:
+ kfree(jz4740_codec);
+
+ return ret;
+}
+
+static int __devexit jz4740_codec_remove(struct platform_device *pdev)
+{
+ struct jz4740_codec *jz4740_codec = platform_get_drvdata(pdev);
+ struct resource *mem = jz4740_codec->mem;
+
+ snd_soc_unregister_dai(&jz4740_codec_dai);
+ snd_soc_unregister_codec(&jz4740_codec->codec);
+
+ iounmap(jz4740_codec->base);
+ release_mem_region(mem->start, resource_size(mem));
+
+ platform_set_drvdata(pdev, NULL);
+ kfree(jz4740_codec);
+
+ return 0;
+}
+
+static struct platform_driver jz4740_codec_driver = {
+ .probe = jz4740_codec_probe,
+ .remove = __devexit_p(jz4740_codec_remove),
+ .driver = {
+ .name = "jz4740-codec",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init jz4740_codec_init(void)
+{
+ return platform_driver_register(&jz4740_codec_driver);
+}
+module_init(jz4740_codec_init);
+
+static void __exit jz4740_codec_exit(void)
+{
+ platform_driver_unregister(&jz4740_codec_driver);
+}
+module_exit(jz4740_codec_exit);
+
+MODULE_DESCRIPTION("JZ4740 SoC internal codec driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:jz4740-codec");
diff --git a/sound/soc/codecs/jz4740.h b/sound/soc/codecs/jz4740.h
new file mode 100644
index 0000000..b5a0691
--- /dev/null
+++ b/sound/soc/codecs/jz4740.h
@@ -0,0 +1,20 @@
+/*
+ * Copyright (C) 2009, Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * 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.
+ *
+ * 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.,
+ * 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#ifndef __SND_SOC_CODECS_JZ4740_CODEC_H__
+#define __SND_SOC_CODECS_JZ4740_CODEC_H__
+
+extern struct snd_soc_dai jz4740_codec_dai;
+extern struct snd_soc_codec_device soc_codec_dev_jz4740_codec;
+
+#endif
diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c
index a631911..9119836 100644
--- a/sound/soc/codecs/spdif_transciever.c
+++ b/sound/soc/codecs/spdif_transciever.c
@@ -16,8 +16,10 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
+#include <linux/slab.h>
#include <sound/soc.h>
#include <sound/pcm.h>
+#include <sound/initval.h>
#include "spdif_transciever.h"
@@ -26,6 +28,48 @@ MODULE_LICENSE("GPL");
#define STUB_RATES SNDRV_PCM_RATE_8000_96000
#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+static struct snd_soc_codec *spdif_dit_codec;
+
+static int spdif_dit_codec_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret;
+
+ if (spdif_dit_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = spdif_dit_codec;
+ codec = spdif_dit_codec;
+
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto err_create_pcms;
+ }
+
+ return 0;
+
+err_create_pcms:
+ return ret;
+}
+
+static int spdif_dit_codec_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_spdif_dit = {
+ .probe = spdif_dit_codec_probe,
+ .remove = spdif_dit_codec_remove,
+}; EXPORT_SYMBOL_GPL(soc_codec_dev_spdif_dit);
+
struct snd_soc_dai dit_stub_dai = {
.name = "DIT",
.playback = {
@@ -40,13 +84,61 @@ EXPORT_SYMBOL_GPL(dit_stub_dai);
static int spdif_dit_probe(struct platform_device *pdev)
{
+ struct snd_soc_codec *codec;
+ int ret;
+
+ if (spdif_dit_codec) {
+ dev_err(&pdev->dev, "Another Codec is registered\n");
+ ret = -EINVAL;
+ goto err_reg_codec;
+ }
+
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ codec->dev = &pdev->dev;
+
+ mutex_init(&codec->mutex);
+
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->name = "spdif-dit";
+ codec->owner = THIS_MODULE;
+ codec->dai = &dit_stub_dai;
+ codec->num_dai = 1;
+
+ spdif_dit_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ goto err_reg_codec;
+ }
+
dit_stub_dai.dev = &pdev->dev;
- return snd_soc_register_dai(&dit_stub_dai);
+ ret = snd_soc_register_dai(&dit_stub_dai);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to register dai: %d\n", ret);
+ goto err_reg_dai;
+ }
+
+ return 0;
+
+err_reg_dai:
+ snd_soc_unregister_codec(codec);
+err_reg_codec:
+ kfree(spdif_dit_codec);
+ return ret;
}
static int spdif_dit_remove(struct platform_device *pdev)
{
snd_soc_unregister_dai(&dit_stub_dai);
+ snd_soc_unregister_codec(spdif_dit_codec);
+ kfree(spdif_dit_codec);
+ spdif_dit_codec = NULL;
return 0;
}
diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h
index 296f2eb..1e10212 100644
--- a/sound/soc/codecs/spdif_transciever.h
+++ b/sound/soc/codecs/spdif_transciever.h
@@ -12,6 +12,7 @@
#ifndef CODEC_STUBS_H
#define CODEC_STUBS_H
+extern struct snd_soc_codec_device soc_codec_dev_spdif_dit;
extern struct snd_soc_dai dit_stub_dai;
#endif /* CODEC_STUBS_H */
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index b0bae35..0a4b0fe 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -560,13 +560,16 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
switch (level) {
case SND_SOC_BIAS_ON:
/* vref/mid, osc on, dac unmute */
+ reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \
+ TLV320AIC23_DAC_OFF);
tlv320aic23_write(codec, TLV320AIC23_PWR, reg);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
/* everything off except vref/vmid, */
- tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040);
+ tlv320aic23_write(codec, TLV320AIC23_PWR, reg | \
+ TLV320AIC23_CLK_OFF);
break;
case SND_SOC_BIAS_OFF:
/* everything off, dac mute, inactive */
@@ -615,7 +618,6 @@ static int tlv320aic23_suspend(struct platform_device *pdev,
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
- tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
@@ -632,7 +634,6 @@ static int tlv320aic23_resume(struct platform_device *pdev)
u16 val = tlv320aic23_read_reg_cache(codec, reg);
tlv320aic23_write(codec, reg, val);
}
-
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 65adc77..2fa946c 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -120,6 +120,8 @@ struct tlv320dac33_priv {
* samples */
unsigned int mode7_us_to_lthr; /* Time to reach lthr from uthr */
+ unsigned int uthr;
+
enum dac33_state state;
};
@@ -442,6 +444,39 @@ static int dac33_set_nsample(struct snd_kcontrol *kcontrol,
return ret;
}
+static int dac33_get_uthr(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.integer.value[0] = dac33->uthr;
+
+ return 0;
+}
+
+static int dac33_set_uthr(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ if (dac33->substream)
+ return -EBUSY;
+
+ if (dac33->uthr == ucontrol->value.integer.value[0])
+ return 0;
+
+ if (ucontrol->value.integer.value[0] < (MODE7_LTHR + 10) ||
+ ucontrol->value.integer.value[0] > MODE7_UTHR)
+ ret = -EINVAL;
+ else
+ dac33->uthr = ucontrol->value.integer.value[0];
+
+ return ret;
+}
+
static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -506,6 +541,8 @@ static const struct snd_kcontrol_new dac33_snd_controls[] = {
static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = {
SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0,
dac33_get_nsample, dac33_set_nsample),
+ SOC_SINGLE_EXT("UTHR", 0, 0, MODE7_UTHR, 0,
+ dac33_get_uthr, dac33_set_uthr),
SOC_ENUM_EXT("FIFO Mode", dac33_fifo_mode_enum,
dac33_get_fifo_mode, dac33_set_fifo_mode),
};
@@ -985,7 +1022,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
* Configure the threshold levels, and leave 10 sample space
* at the bottom, and also at the top of the FIFO
*/
- dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(MODE7_UTHR));
+ dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(dac33->uthr));
dac33_write16(codec, DAC33_LTHR_MSB, DAC33_THRREG(MODE7_LTHR));
break;
default:
@@ -1052,8 +1089,8 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
break;
case DAC33_FIFO_MODE7:
dac33->mode7_us_to_lthr =
- SAMPLES_TO_US(substream->runtime->rate,
- MODE7_UTHR - MODE7_LTHR + 1);
+ SAMPLES_TO_US(substream->runtime->rate,
+ dac33->uthr - MODE7_LTHR + 1);
dac33->t_stamp1 = 0;
break;
default:
@@ -1104,7 +1141,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
struct snd_soc_codec *codec = socdev->card->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
unsigned long long t0, t1, t_now;
- unsigned int time_delta;
+ unsigned int time_delta, uthr;
int samples_out, samples_in, samples;
snd_pcm_sframes_t delay = 0;
@@ -1182,6 +1219,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
case DAC33_FIFO_MODE7:
spin_lock(&dac33->lock);
t0 = dac33->t_stamp1;
+ uthr = dac33->uthr;
spin_unlock(&dac33->lock);
t_now = ktime_to_us(ktime_get());
@@ -1194,7 +1232,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
* Either the timestamps are messed or equal. Report
* maximum delay
*/
- delay = MODE7_UTHR;
+ delay = uthr;
goto out;
}
@@ -1208,8 +1246,8 @@ static snd_pcm_sframes_t dac33_dai_delay(
substream->runtime->rate,
time_delta);
- if (likely(MODE7_UTHR > samples_out))
- delay = MODE7_UTHR - samples_out;
+ if (likely(uthr > samples_out))
+ delay = uthr - samples_out;
else
delay = 0;
} else {
@@ -1227,8 +1265,8 @@ static snd_pcm_sframes_t dac33_dai_delay(
time_delta);
delay = MODE7_LTHR + samples_in - samples_out;
- if (unlikely(delay > MODE7_UTHR))
- delay = MODE7_UTHR;
+ if (unlikely(delay > uthr))
+ delay = uthr;
}
break;
default:
@@ -1484,6 +1522,7 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
dac33->irq = client->irq;
dac33->nsample = NSAMPLE_MAX;
dac33->nsample_max = NSAMPLE_MAX;
+ dac33->uthr = MODE7_UTHR;
/* Disable FIFO use by default */
dac33->fifo_mode = DAC33_FIFO_BYPASS;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index b4fcdb0..8d36bfa 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -43,37 +43,37 @@
*/
static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
0x00, /* this register not used */
- 0x91, /* REG_CODEC_MODE (0x1) */
- 0xc3, /* REG_OPTION (0x2) */
+ 0x00, /* REG_CODEC_MODE (0x1) */
+ 0x00, /* REG_OPTION (0x2) */
0x00, /* REG_UNKNOWN (0x3) */
0x00, /* REG_MICBIAS_CTL (0x4) */
- 0x20, /* REG_ANAMICL (0x5) */
+ 0x00, /* 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) */
+ 0x0f, /* REG_ATXL1PGA (0xA) */
+ 0x0f, /* REG_ATXR1PGA (0xB) */
+ 0x0f, /* REG_AVTXL2PGA (0xC) */
+ 0x0f, /* REG_AVTXR2PGA (0xD) */
0x00, /* 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) */
+ 0x3f, /* REG_ARXR1PGA (0x10) */
+ 0x3f, /* REG_ARXL1PGA (0x11) */
+ 0x3f, /* REG_ARXR2PGA (0x12) */
+ 0x3f, /* REG_ARXL2PGA (0x13) */
+ 0x25, /* REG_VRXPGA (0x14) */
0x00, /* REG_VSTPGA (0x15) */
0x00, /* REG_VRX2ARXPGA (0x16) */
0x00, /* REG_AVDAC_CTL (0x17) */
0x00, /* REG_ARX2VTXPGA (0x18) */
- 0x00, /* REG_ARXL1_APGA_CTL (0x19) */
- 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */
- 0x4a, /* REG_ARXL2_APGA_CTL (0x1B) */
- 0x4a, /* REG_ARXR2_APGA_CTL (0x1C) */
+ 0x32, /* REG_ARXL1_APGA_CTL (0x19) */
+ 0x32, /* REG_ARXR1_APGA_CTL (0x1A) */
+ 0x32, /* REG_ARXL2_APGA_CTL (0x1B) */
+ 0x32, /* REG_ARXR2_APGA_CTL (0x1C) */
0x00, /* REG_ATX2ARXPGA (0x1D) */
0x00, /* REG_BT_IF (0x1E) */
- 0x00, /* REG_BTPGA (0x1F) */
+ 0x55, /* REG_BTPGA (0x1F) */
0x00, /* REG_BTSTPGA (0x20) */
0x00, /* REG_EAR_CTL (0x21) */
0x00, /* REG_HS_SEL (0x22) */
@@ -85,32 +85,32 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
0x00, /* REG_PRECKR_CTL (0x28) */
0x00, /* REG_HFL_CTL (0x29) */
0x00, /* REG_HFR_CTL (0x2A) */
- 0x00, /* REG_ALC_CTL (0x2B) */
+ 0x05, /* 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) */
+ 0x13, /* 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) */
+ 0x79, /* REG_DTMF_TONOFF (0x35) */
+ 0x11, /* 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) */
0x06, /* REG_APLL_CTL (0x3A) */
0x00, /* REG_DTMF_CTL (0x3B) */
- 0x00, /* REG_DTMF_PGA_CTL2 (0x3C) */
- 0x00, /* REG_DTMF_PGA_CTL1 (0x3D) */
+ 0x44, /* REG_DTMF_PGA_CTL2 (0x3C) */
+ 0x69, /* 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) */
+ 0x32, /* REG_VDL_APGA_CTL (0x44) */
0x00, /* REG_VIBRA_CTL (0x45) */
0x00, /* REG_VIBRA_SET (0x46) */
0x00, /* REG_VIBRA_PWM_SET (0x47) */
@@ -244,58 +244,93 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
udelay(10);
}
-static void twl4030_init_chip(struct snd_soc_codec *codec)
+static inline void twl4030_check_defaults(struct snd_soc_codec *codec)
{
- u8 *cache = codec->reg_cache;
- int i;
+ int i, difference = 0;
+ u8 val;
+
+ dev_dbg(codec->dev, "Checking TWL audio default configuration\n");
+ for (i = 1; i <= TWL4030_REG_MISC_SET_2; i++) {
+ twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &val, i);
+ if (val != twl4030_reg[i]) {
+ difference++;
+ dev_dbg(codec->dev,
+ "Reg 0x%02x: chip: 0x%02x driver: 0x%02x\n",
+ i, val, twl4030_reg[i]);
+ }
+ }
+ dev_dbg(codec->dev, "Found %d non maching registers. %s\n",
+ difference, difference ? "Not OK" : "OK");
+}
- /* clear CODECPDZ prior to setting register defaults */
- twl4030_codec_enable(codec, 0);
+static inline void twl4030_reset_registers(struct snd_soc_codec *codec)
+{
+ int i;
/* set all audio section registers to reasonable defaults */
for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
if (i != TWL4030_REG_APLL_CTL)
- twl4030_write(codec, i, cache[i]);
+ twl4030_write(codec, i, twl4030_reg[i]);
}
-static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
+static void twl4030_init_chip(struct platform_device *pdev)
{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct twl4030_setup_data *setup = socdev->codec_data;
+ struct snd_soc_codec *codec = socdev->card->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
- int status = -1;
+ u8 reg, byte;
+ int i = 0;
- if (enable) {
- twl4030->apll_enabled++;
- if (twl4030->apll_enabled == 1)
- status = twl4030_codec_enable_resource(
- TWL4030_CODEC_RES_APLL);
- } else {
- twl4030->apll_enabled--;
- if (!twl4030->apll_enabled)
- status = twl4030_codec_disable_resource(
- TWL4030_CODEC_RES_APLL);
- }
+ /* Check defaults, if instructed before anything else */
+ if (setup && setup->check_defaults)
+ twl4030_check_defaults(codec);
- if (status >= 0)
- twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status);
-}
+ /* Reset registers, if no setup data or if instructed to do so */
+ if (!setup || (setup && setup->reset_registers))
+ twl4030_reset_registers(codec);
-static void twl4030_power_up(struct snd_soc_codec *codec)
-{
- struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
- u8 anamicl, regmisc1, byte;
- int i = 0;
+ /* Refresh APLL_CTL register from HW */
+ twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
+ TWL4030_REG_APLL_CTL);
+ twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, byte);
- if (twl4030->codec_powered)
+ /* anti-pop when changing analog gain */
+ reg = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
+ twl4030_write(codec, TWL4030_REG_MISC_SET_1,
+ reg | TWL4030_SMOOTH_ANAVOL_EN);
+
+ twl4030_write(codec, TWL4030_REG_OPTION,
+ TWL4030_ATXL1_EN | TWL4030_ATXR1_EN |
+ TWL4030_ARXL2_EN | TWL4030_ARXR2_EN);
+
+ /* REG_ARXR2_APGA_CTL reset according to the TRM: 0dB, DA_EN */
+ twl4030_write(codec, TWL4030_REG_ARXR2_APGA_CTL, 0x32);
+
+ /* Machine dependent setup */
+ if (!setup)
return;
- /* set CODECPDZ to turn on codec */
- twl4030_codec_enable(codec, 1);
+ /* Configuration for headset ramp delay from setup data */
+ if (setup->sysclk != twl4030->sysclk)
+ dev_warn(codec->dev,
+ "Mismatch in APLL mclk: %u (configured: %u)\n",
+ setup->sysclk, twl4030->sysclk);
+
+ reg = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+ reg &= ~TWL4030_RAMP_DELAY;
+ reg |= (setup->ramp_delay_value << 2);
+ twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, reg);
/* initiate offset cancellation */
- anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+ twl4030_codec_enable(codec, 1);
+
+ reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+ reg &= ~TWL4030_OFFSET_CNCL_SEL;
+ reg |= setup->offset_cncl_path;
twl4030_write(codec, TWL4030_REG_ANAMICL,
- anamicl | TWL4030_CNCL_OFFSET_START);
+ reg | TWL4030_CNCL_OFFSET_START);
/* wait for offset cancellation to complete */
do {
@@ -310,23 +345,28 @@ static void twl4030_power_up(struct snd_soc_codec *codec)
/* Make sure that the reg_cache has the same value as the HW */
twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte);
- /* 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_codec_enable(codec, 0);
- twl4030_codec_enable(codec, 1);
}
-/*
- * Unconditional power down
- */
-static void twl4030_power_down(struct snd_soc_codec *codec)
+static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
{
- /* power down */
- twl4030_codec_enable(codec, 0);
+ struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
+ int status = -1;
+
+ if (enable) {
+ twl4030->apll_enabled++;
+ if (twl4030->apll_enabled == 1)
+ status = twl4030_codec_enable_resource(
+ TWL4030_CODEC_RES_APLL);
+ } else {
+ twl4030->apll_enabled--;
+ if (!twl4030->apll_enabled)
+ status = twl4030_codec_disable_resource(
+ TWL4030_CODEC_RES_APLL);
+ }
+
+ if (status >= 0)
+ twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status);
}
/* Earpiece */
@@ -1605,10 +1645,10 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_OFF)
- twl4030_power_up(codec);
+ twl4030_codec_enable(codec, 1);
break;
case SND_SOC_BIAS_OFF:
- twl4030_power_down(codec);
+ twl4030_codec_enable(codec, 0);
break;
}
codec->bias_level = level;
@@ -1794,13 +1834,6 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- if (mode != old_mode) {
- /* change rate and set CODECPDZ */
- twl4030_codec_enable(codec, 0);
- twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
- twl4030_codec_enable(codec, 1);
- }
-
/* sample size */
old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
format = old_format;
@@ -1818,16 +1851,20 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- if (format != old_format) {
-
- /* clear CODECPDZ before changing format (codec requirement) */
- twl4030_codec_enable(codec, 0);
-
- /* change format */
- twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
-
- /* set CODECPDZ afterwards */
- twl4030_codec_enable(codec, 1);
+ if (format != old_format || mode != old_mode) {
+ if (twl4030->codec_powered) {
+ /*
+ * If the codec is powered, than we need to toggle the
+ * codec power.
+ */
+ twl4030_codec_enable(codec, 0);
+ twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+ twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+ twl4030_codec_enable(codec, 1);
+ } else {
+ twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+ twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+ }
}
/* Store the important parameters for the DAI configuration and set
@@ -1877,6 +1914,7 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
+ struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
u8 old_format, format;
/* get format */
@@ -1911,15 +1949,17 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
}
if (format != old_format) {
-
- /* clear CODECPDZ before changing format (codec requirement) */
- twl4030_codec_enable(codec, 0);
-
- /* change format */
- twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
-
- /* set CODECPDZ afterwards */
- twl4030_codec_enable(codec, 1);
+ if (twl4030->codec_powered) {
+ /*
+ * If the codec is powered, than we need to toggle the
+ * codec power.
+ */
+ twl4030_codec_enable(codec, 0);
+ twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+ twl4030_codec_enable(codec, 1);
+ } else {
+ twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+ }
}
return 0;
@@ -2011,6 +2051,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
+ struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
u8 old_mode, mode;
/* Enable voice digital filters */
@@ -2035,10 +2076,17 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
}
if (mode != old_mode) {
- /* change rate and set CODECPDZ */
- twl4030_codec_enable(codec, 0);
- twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
- twl4030_codec_enable(codec, 1);
+ if (twl4030->codec_powered) {
+ /*
+ * If the codec is powered, than we need to toggle the
+ * codec power.
+ */
+ twl4030_codec_enable(codec, 0);
+ twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+ twl4030_codec_enable(codec, 1);
+ } else {
+ twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+ }
}
return 0;
@@ -2068,6 +2116,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
+ struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
u8 old_format, format;
/* get format */
@@ -2099,10 +2148,17 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
}
if (format != old_format) {
- /* change format and set CODECPDZ */
- twl4030_codec_enable(codec, 0);
- twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
- twl4030_codec_enable(codec, 1);
+ if (twl4030->codec_powered) {
+ /*
+ * If the codec is powered, than we need to toggle the
+ * codec power.
+ */
+ twl4030_codec_enable(codec, 0);
+ twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
+ twl4030_codec_enable(codec, 1);
+ } else {
+ twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
+ }
}
return 0;
@@ -2202,31 +2258,15 @@ static struct snd_soc_codec *twl4030_codec;
static int twl4030_soc_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct twl4030_setup_data *setup = socdev->codec_data;
struct snd_soc_codec *codec;
- struct twl4030_priv *twl4030;
int ret;
BUG_ON(!twl4030_codec);
codec = twl4030_codec;
- twl4030 = snd_soc_codec_get_drvdata(codec);
socdev->card->codec = codec;
- /* Configuration for headset ramp delay from setup data */
- if (setup) {
- unsigned char hs_pop;
-
- if (setup->sysclk != twl4030->sysclk)
- dev_warn(&pdev->dev,
- "Mismatch in APLL mclk: %u (configured: %u)\n",
- setup->sysclk, twl4030->sysclk);
-
- hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
- hs_pop &= ~TWL4030_RAMP_DELAY;
- hs_pop |= (setup->ramp_delay_value << 2);
- twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
- }
+ twl4030_init_chip(pdev);
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
@@ -2247,6 +2287,8 @@ static int twl4030_soc_remove(struct platform_device *pdev)
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
+ /* Reset registers to their chip default before leaving */
+ twl4030_reset_registers(codec);
twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
@@ -2287,6 +2329,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev)
codec->read = twl4030_read_reg_cache;
codec->write = twl4030_write;
codec->set_bias_level = twl4030_set_bias_level;
+ codec->idle_bias_off = 1;
codec->dai = twl4030_dai;
codec->num_dai = ARRAY_SIZE(twl4030_dai);
codec->reg_cache_size = sizeof(twl4030_reg);
@@ -2302,9 +2345,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev)
/* Set the defaults, and power up the codec */
twl4030->sysclk = twl4030_codec_get_mclk() / 1000;
- twl4030_init_chip(codec);
codec->bias_level = SND_SOC_BIAS_OFF;
- twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
ret = snd_soc_register_codec(codec);
if (ret != 0) {
@@ -2322,7 +2363,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev)
return 0;
error_codec:
- twl4030_power_down(codec);
+ twl4030_codec_enable(codec, 0);
kfree(codec->reg_cache);
error_cache:
kfree(twl4030);
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index f206d24..788e3d1 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -42,6 +42,9 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030;
struct twl4030_setup_data {
unsigned int ramp_delay_value;
unsigned int sysclk;
+ unsigned int offset_cncl_path;
+ unsigned int check_defaults:1;
+ unsigned int reset_registers:1;
unsigned int hs_extmute:1;
void (*set_hs_extmute)(int mute);
};
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index af36346..85dd4fb 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -928,7 +928,7 @@ static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai,
case 19200000:
/* mclk input, pll disabled */
hppllctl |= TWL6040_MCLK_19200KHZ |
- TWL6040_HPLLSQRBP |
+ TWL6040_HPLLSQRENA |
TWL6040_HPLLBP;
break;
case 26000000:
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 28aac53..f3b4c1d 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -28,19 +28,6 @@
#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)
@@ -58,7 +45,7 @@ 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,
+ 0x00, 0x83, 0x00, 0x40, 0x80, 0xC0, 0x00,
};
/*
@@ -117,6 +104,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
case UDA134X_DATA000:
case UDA134X_DATA001:
case UDA134X_DATA010:
+ case UDA134X_DATA011:
addr = UDA134X_DATA0_ADDR;
break;
case UDA134X_DATA1:
@@ -353,8 +341,22 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec,
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);
+ switch (pd->model) {
+ case UDA134X_UDA1340:
+ case UDA134X_UDA1344:
+ case UDA134X_UDA1345:
+ reg = uda134x_read_reg_cache(codec, UDA134X_DATA011);
+ uda134x_write(codec, UDA134X_DATA011, reg | 0x03);
+ break;
+ case UDA134X_UDA1341:
+ reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+ uda134x_write(codec, UDA134X_STATUS1, reg | 0x03);
+ break;
+ default:
+ printk(KERN_ERR "UDA134X SoC codec: "
+ "unsupported model %d\n", pd->model);
+ return -EINVAL;
+ }
break;
case SND_SOC_BIAS_PREPARE:
/* power on */
@@ -367,8 +369,22 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec,
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));
+ switch (pd->model) {
+ case UDA134X_UDA1340:
+ case UDA134X_UDA1344:
+ case UDA134X_UDA1345:
+ reg = uda134x_read_reg_cache(codec, UDA134X_DATA011);
+ uda134x_write(codec, UDA134X_DATA011, reg & ~(0x03));
+ break;
+ case UDA134X_UDA1341:
+ reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+ uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03));
+ break;
+ default:
+ printk(KERN_ERR "UDA134X SoC codec: "
+ "unsupported model %d\n", pd->model);
+ return -EINVAL;
+ }
break;
case SND_SOC_BIAS_OFF:
/* power off */
@@ -531,9 +547,7 @@ static int uda134x_soc_probe(struct platform_device *pdev)
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);
@@ -544,6 +558,14 @@ static int uda134x_soc_probe(struct platform_device *pdev)
uda134x_reset(codec);
+ if (pd->is_powered_on_standby) {
+ codec->set_bias_level = NULL;
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
+ } else {
+ codec->set_bias_level = uda134x_set_bias_level;
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ }
+
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h
index 94f4404..205f03b 100644
--- a/sound/soc/codecs/uda134x.h
+++ b/sound/soc/codecs/uda134x.h
@@ -23,9 +23,10 @@
#define UDA134X_DATA000 10
#define UDA134X_DATA001 11
#define UDA134X_DATA010 12
-#define UDA134X_DATA1 13
+#define UDA134X_DATA011 13
+#define UDA134X_DATA1 14
-#define UDA134X_REGS_NUM 14
+#define UDA134X_REGS_NUM 15
#define STATUS0_DAIFMT_MASK (~(7<<1))
#define STATUS0_SYSCLK_MASK (~(3<<4))
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 002e289..4bcd168 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -795,6 +795,8 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
dev_set_drvdata(&i2c->dev, wm2000);
wm2000->anc_eng_ena = 1;
+ wm2000->anc_active = 1;
+ wm2000->spk_ena = 1;
wm2000->i2c = i2c;
wm2000_reset(wm2000);
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 9407e19..e2c05e3 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -884,6 +884,7 @@ static int wm8750_i2c_remove(struct i2c_client *client)
static const struct i2c_device_id wm8750_i2c_id[] = {
{ "wm8750", 0 },
+ { "wm8987", 0 }, /* WM8987 is register compatible with WM8750 */
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id);
@@ -925,14 +926,22 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi)
return 0;
}
+static const struct spi_device_id wm8750_spi_id[] = {
+ { "wm8750", 0 },
+ { "wm8987", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(spi, wm8750_spi_id);
+
static struct spi_driver wm8750_spi_driver = {
.driver = {
- .name = "wm8750",
+ .name = "WM8750 SPI Codec",
.bus = &spi_bus_type,
.owner = THIS_MODULE,
},
.probe = wm8750_spi_probe,
.remove = __devexit_p(wm8750_spi_remove),
+ .id_table = wm8750_spi_id,
};
#endif
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 7233cc6..3c6ee61 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -79,12 +79,13 @@ struct wm8960_priv {
struct snd_soc_dapm_widget *lout1;
struct snd_soc_dapm_widget *rout1;
struct snd_soc_dapm_widget *out3;
+ bool deemph;
+ int playback_fs;
};
#define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0)
/* enumerated controls */
-static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted",
"Right Inverted", "Stereo Inversion"};
static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"};
@@ -93,7 +94,6 @@ static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"};
static const char *wm8960_alcmode[] = {"ALC", "Limiter"};
static const struct soc_enum wm8960_enum[] = {
- SOC_ENUM_SINGLE(WM8960_DACCTL1, 1, 4, wm8960_deemph),
SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity),
SOC_ENUM_SINGLE(WM8960_DACCTL2, 5, 4, wm8960_polarity),
SOC_ENUM_SINGLE(WM8960_3D, 6, 2, wm8960_3d_upper_cutoff),
@@ -102,6 +102,59 @@ static const struct soc_enum wm8960_enum[] = {
SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode),
};
+static const int deemph_settings[] = { 0, 32000, 44100, 48000 };
+
+static int wm8960_set_deemph(struct snd_soc_codec *codec)
+{
+ struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
+ int val, i, best;
+
+ /* If we're using deemphasis select the nearest available sample
+ * rate.
+ */
+ if (wm8960->deemph) {
+ best = 1;
+ for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) {
+ if (abs(deemph_settings[i] - wm8960->playback_fs) <
+ abs(deemph_settings[best] - wm8960->playback_fs))
+ best = i;
+ }
+
+ val = best << 1;
+ } else {
+ val = 0;
+ }
+
+ dev_dbg(codec->dev, "Set deemphasis %d\n", val);
+
+ return snd_soc_update_bits(codec, WM8960_DACCTL1,
+ 0x6, val);
+}
+
+static int wm8960_get_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
+
+ return wm8960->deemph;
+}
+
+static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
+ int deemph = ucontrol->value.enumerated.item[0];
+
+ if (deemph > 1)
+ return -EINVAL;
+
+ wm8960->deemph = deemph;
+
+ return wm8960_set_deemph(codec);
+}
+
static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0);
static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
@@ -131,23 +184,24 @@ SOC_SINGLE("Speaker DC Volume", WM8960_CLASSD3, 3, 5, 0),
SOC_SINGLE("Speaker AC Volume", WM8960_CLASSD3, 0, 5, 0),
SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0),
-SOC_ENUM("ADC Polarity", wm8960_enum[1]),
-SOC_ENUM("Playback De-emphasis", wm8960_enum[0]),
+SOC_ENUM("ADC Polarity", wm8960_enum[0]),
SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0),
SOC_ENUM("DAC Polarity", wm8960_enum[2]),
+SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0,
+ wm8960_get_deemph, wm8960_put_deemph),
-SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[3]),
-SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[4]),
+SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[2]),
+SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[3]),
SOC_SINGLE("3D Volume", WM8960_3D, 1, 15, 0),
SOC_SINGLE("3D Switch", WM8960_3D, 0, 1, 0),
-SOC_ENUM("ALC Function", wm8960_enum[5]),
+SOC_ENUM("ALC Function", wm8960_enum[4]),
SOC_SINGLE("ALC Max Gain", WM8960_ALC1, 4, 7, 0),
SOC_SINGLE("ALC Target", WM8960_ALC1, 0, 15, 1),
SOC_SINGLE("ALC Min Gain", WM8960_ALC2, 4, 7, 0),
SOC_SINGLE("ALC Hold Time", WM8960_ALC2, 0, 15, 0),
-SOC_ENUM("ALC Mode", wm8960_enum[6]),
+SOC_ENUM("ALC Mode", wm8960_enum[5]),
SOC_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0),
SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0),
@@ -433,6 +487,21 @@ static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai,
return 0;
}
+static struct {
+ int rate;
+ unsigned int val;
+} alc_rates[] = {
+ { 48000, 0 },
+ { 44100, 0 },
+ { 32000, 1 },
+ { 22050, 2 },
+ { 24000, 2 },
+ { 16000, 3 },
+ { 11250, 4 },
+ { 12000, 4 },
+ { 8000, 5 },
+};
+
static int wm8960_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -440,7 +509,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
+ struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3;
+ int i;
/* bit size */
switch (params_format(params)) {
@@ -454,6 +525,18 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
break;
}
+ /* Update filters for the new rate */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ wm8960->playback_fs = params_rate(params);
+ wm8960_set_deemph(codec);
+ } else {
+ for (i = 0; i < ARRAY_SIZE(alc_rates); i++)
+ if (alc_rates[i].rate == params_rate(params))
+ snd_soc_update_bits(codec,
+ WM8960_ADDCTL3, 0x7,
+ alc_rates[i].val);
+ }
+
/* set iface */
snd_soc_write(codec, WM8960_IFACE1, iface);
return 0;
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index c018772..dd8d909 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -30,8 +30,6 @@
#include "wm8990.h"
-#define WM8990_VERSION "0.2"
-
/* codec private data */
struct wm8990_priv {
unsigned int sysclk;
@@ -1511,8 +1509,6 @@ static int wm8990_probe(struct platform_device *pdev)
struct wm8990_priv *wm8990;
int ret;
- pr_info("WM8990 Audio Codec %s\n", WM8990_VERSION);
-
setup = socdev->codec_data;
codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (codec == NULL)
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index e84a117..c41cf47 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1677,6 +1677,26 @@ static struct {
static int wm8994_readable(unsigned int reg)
{
+ switch (reg) {
+ case WM8994_GPIO_1:
+ case WM8994_GPIO_2:
+ case WM8994_GPIO_3:
+ case WM8994_GPIO_4:
+ case WM8994_GPIO_5:
+ case WM8994_GPIO_6:
+ case WM8994_GPIO_7:
+ case WM8994_GPIO_8:
+ case WM8994_GPIO_9:
+ case WM8994_GPIO_10:
+ case WM8994_GPIO_11:
+ case WM8994_INTERRUPT_STATUS_1:
+ case WM8994_INTERRUPT_STATUS_2:
+ case WM8994_INTERRUPT_RAW_STATUS_2:
+ return 1;
+ default:
+ break;
+ }
+
if (reg >= ARRAY_SIZE(access_masks))
return 0;
return access_masks[reg].readable != 0;
@@ -2472,6 +2492,7 @@ static const struct snd_kcontrol_new aif3adc_mux =
static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("DMIC1DAT"),
SND_SOC_DAPM_INPUT("DMIC2DAT"),
+SND_SOC_DAPM_INPUT("Clock"),
SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -2946,11 +2967,14 @@ static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src,
return 0;
}
+static int opclk_divs[] = { 10, 20, 30, 40, 55, 60, 80, 120, 160 };
+
static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = dai->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ int i;
switch (dai->id) {
case 1:
@@ -2988,6 +3012,25 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
dev_dbg(dai->dev, "AIF%d using FLL2\n", dai->id);
break;
+ case WM8994_SYSCLK_OPCLK:
+ /* Special case - a division (times 10) is given and
+ * no effect on main clocking.
+ */
+ if (freq) {
+ for (i = 0; i < ARRAY_SIZE(opclk_divs); i++)
+ if (opclk_divs[i] == freq)
+ break;
+ if (i == ARRAY_SIZE(opclk_divs))
+ return -EINVAL;
+ snd_soc_update_bits(codec, WM8994_CLOCKING_2,
+ WM8994_OPCLK_DIV_MASK, i);
+ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_2,
+ WM8994_OPCLK_ENA, WM8994_OPCLK_ENA);
+ } else {
+ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_2,
+ WM8994_OPCLK_ENA, 0);
+ }
+
default:
return -EINVAL;
}
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index 7072dc5..2e0ca67 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -20,6 +20,9 @@ extern struct snd_soc_dai wm8994_dai[];
#define WM8994_SYSCLK_FLL1 3
#define WM8994_SYSCLK_FLL2 4
+/* OPCLK is also configured with set_dai_sysclk, specify division*10 as rate. */
+#define WM8994_SYSCLK_OPCLK 5
+
#define WM8994_FLL1 1
#define WM8994_FLL2 2
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