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-rw-r--r--sound/soc/codecs/Kconfig11
-rw-r--r--sound/soc/codecs/Makefile4
-rw-r--r--sound/soc/codecs/ac97.c3
-rw-r--r--sound/soc/codecs/ad1980.c1
-rw-r--r--sound/soc/codecs/ad73311.c107
-rw-r--r--sound/soc/codecs/ad73311.h90
-rw-r--r--sound/soc/codecs/ak4535.c1
-rw-r--r--sound/soc/codecs/ssm2602.c1
-rw-r--r--sound/soc/codecs/tlv320aic23.c714
-rw-r--r--sound/soc/codecs/tlv320aic23.h122
-rw-r--r--sound/soc/codecs/tlv320aic3x.c5
-rw-r--r--sound/soc/codecs/uda1380.c1
-rw-r--r--sound/soc/codecs/wm8510.c111
-rw-r--r--sound/soc/codecs/wm8510.h1
-rw-r--r--sound/soc/codecs/wm8580.c2
-rw-r--r--sound/soc/codecs/wm8731.c1
-rw-r--r--sound/soc/codecs/wm8750.c1
-rw-r--r--sound/soc/codecs/wm8753.c75
-rw-r--r--sound/soc/codecs/wm8753.h4
-rw-r--r--sound/soc/codecs/wm8900.c1
-rw-r--r--sound/soc/codecs/wm8903.c4
-rw-r--r--sound/soc/codecs/wm8971.c1
-rw-r--r--sound/soc/codecs/wm8990.c1
-rw-r--r--sound/soc/codecs/wm9712.c3
-rw-r--r--sound/soc/codecs/wm9713.c3
25 files changed, 1223 insertions, 45 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index e0b9869df0f..4975d8573e4 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -3,9 +3,11 @@ config SND_SOC_ALL_CODECS
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
@@ -34,6 +36,9 @@ config SND_SOC_AC97_CODEC
config SND_SOC_AD1980
tristate
+config SND_SOC_AD73311
+ tristate
+
config SND_SOC_AK4535
tristate
@@ -58,9 +63,13 @@ config SND_SOC_CS4270_VD33_ERRATA
config SND_SOC_SSM2602
tristate
+config SND_SOC_TLV320AIC23
+ tristate
+ depends on I2C
+
config SND_SOC_TLV320AIC26
tristate "TI TLV320AIC26 Codec support"
- depends on SND_SOC && SPI
+ depends on SPI
config SND_SOC_TLV320AIC3X
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index f977978a340..90f0a585fc7 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,8 +1,10 @@
snd-soc-ac97-objs := ac97.o
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-ssm2602-objs := ssm2602.o
+snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
snd-soc-uda1380-objs := uda1380.o
@@ -20,9 +22,11 @@ snd-soc-wm9713-objs := wm9713.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
+obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(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_UDA1380) += snd-soc-uda1380.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 61fd96ca7bc..bd1ebdc6c86 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -2,8 +2,7 @@
* ac97.c -- ALSA Soc AC97 codec support
*
* Copyright 2005 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 4e09c1f2c06..1397b8e06c0 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -13,7 +13,6 @@
#include <linux/init.h>
#include <linux/module.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <sound/core.h>
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
new file mode 100644
index 00000000000..37af8607b00
--- /dev/null
+++ b/sound/soc/codecs/ad73311.c
@@ -0,0 +1,107 @@
+/*
+ * ad73311.c -- ALSA Soc AD73311 codec support
+ *
+ * Copyright: Analog Device Inc.
+ * Author: Cliff Cai <cliff.cai@analog.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.
+ *
+ * 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>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "ad73311.h"
+
+struct snd_soc_dai ad73311_dai = {
+ .name = "AD73311",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+};
+EXPORT_SYMBOL_GPL(ad73311_dai);
+
+static int ad73311_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+ mutex_init(&codec->mutex);
+ codec->name = "AD73311";
+ codec->owner = THIS_MODULE;
+ codec->dai = &ad73311_dai;
+ codec->num_dai = 1;
+ socdev->codec = codec;
+ 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 "ad73311: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "ad73311: failed to register card\n");
+ goto register_err;
+ }
+
+ return ret;
+
+register_err:
+ snd_soc_free_pcms(socdev);
+pcm_err:
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+ return ret;
+}
+
+static int ad73311_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec == NULL)
+ return 0;
+ snd_soc_free_pcms(socdev);
+ kfree(codec);
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ad73311 = {
+ .probe = ad73311_soc_probe,
+ .remove = ad73311_soc_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311);
+
+MODULE_DESCRIPTION("ASoC ad73311 driver");
+MODULE_AUTHOR("Cliff Cai ");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h
new file mode 100644
index 00000000000..507ce0c30ed
--- /dev/null
+++ b/sound/soc/codecs/ad73311.h
@@ -0,0 +1,90 @@
+/*
+ * File: sound/soc/codec/ad73311.h
+ * Based on:
+ * Author: Cliff Cai <cliff.cai@analog.com>
+ *
+ * Created: Thur Sep 25, 2008
+ * Description: definitions for AD73311 registers
+ *
+ *
+ * Modified:
+ * Copyright 2006 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef __AD73311_H__
+#define __AD73311_H__
+
+#define AD_CONTROL 0x8000
+#define AD_DATA 0x0000
+#define AD_READ 0x4000
+#define AD_WRITE 0x0000
+
+/* Control register A */
+#define CTRL_REG_A (0 << 8)
+
+#define REGA_MODE_PRO 0x00
+#define REGA_MODE_DATA 0x01
+#define REGA_MODE_MIXED 0x03
+#define REGA_DLB 0x04
+#define REGA_SLB 0x08
+#define REGA_DEVC(x) ((x & 0x7) << 4)
+#define REGA_RESET 0x80
+
+/* Control register B */
+#define CTRL_REG_B (1 << 8)
+
+#define REGB_DIRATE(x) (x & 0x3)
+#define REGB_SCDIV(x) ((x & 0x3) << 2)
+#define REGB_MCDIV(x) ((x & 0x7) << 4)
+#define REGB_CEE (1 << 7)
+
+/* Control register C */
+#define CTRL_REG_C (2 << 8)
+
+#define REGC_PUDEV (1 << 0)
+#define REGC_PUADC (1 << 3)
+#define REGC_PUDAC (1 << 4)
+#define REGC_PUREF (1 << 5)
+#define REGC_REFUSE (1 << 6)
+
+/* Control register D */
+#define CTRL_REG_D (3 << 8)
+
+#define REGD_IGS(x) (x & 0x7)
+#define REGD_RMOD (1 << 3)
+#define REGD_OGS(x) ((x & 0x7) << 4)
+#define REGD_MUTE (x << 7)
+
+/* Control register E */
+#define CTRL_REG_E (4 << 8)
+
+#define REGE_DA(x) (x & 0x1f)
+#define REGE_IBYP (1 << 5)
+
+/* Control register F */
+#define CTRL_REG_F (5 << 8)
+
+#define REGF_SEEN (1 << 5)
+#define REGF_INV (1 << 6)
+#define REGF_ALB (1 << 7)
+
+extern struct snd_soc_dai ad73311_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ad73311;
+#endif
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 088cf992772..2a89b5888e1 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -28,7 +28,6 @@
#include "ak4535.h"
-#define AUDIO_NAME "ak4535"
#define AK4535_VERSION "0.3"
struct snd_soc_codec_device soc_codec_dev_ak4535;
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 940ce1c3522..44ef0dacd56 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -42,7 +42,6 @@
#include "ssm2602.h"
-#define AUDIO_NAME "ssm2602"
#define SSM2602_VERSION "0.1"
struct snd_soc_codec_device soc_codec_dev_ssm2602;
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
new file mode 100644
index 00000000000..bac7815e00f
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -0,0 +1,714 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver
+ *
+ * Author: Arun KS, <arunks@mistralsolutions.com>
+ * Copyright: (C) 2008 Mistral Solutions Pvt Ltd.,
+ *
+ * Based on sound/soc/codecs/wm8731.c by Richard Purdie
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Notes:
+ * The AIC23 is a driver for a low power stereo audio
+ * codec tlv320aic23
+ *
+ * The machine layer should disable unsupported inputs/outputs by
+ * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+
+#include "tlv320aic23.h"
+
+#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
+ */
+static const u16 tlv320aic23_reg[] = {
+ 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */
+ 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */
+ 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */
+};
+
+/*
+ * read tlv320aic23 register cache
+ */
+static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec
+ *codec, unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= ARRAY_SIZE(tlv320aic23_reg))
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write tlv320aic23 register cache
+ */
+static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec,
+ u8 reg, u16 value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= ARRAY_SIZE(tlv320aic23_reg))
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the tlv320aic23 register space
+ */
+static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+
+ u8 data;
+
+ /* TLV320AIC23 has 7 bit address and 9 bits of data
+ * so we need to switch one data bit into reg and rest
+ * of data into val
+ */
+
+ if ((reg < 0 || reg > 9) && (reg != 15)) {
+ printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg);
+ return -1;
+ }
+
+ data = (reg << 1) | (value >> 8 & 0x01);
+
+ tlv320aic23_write_reg_cache(codec, reg, value);
+
+ if (codec->hw_write(codec->control_data, data,
+ (value & 0xff)) == 0)
+ return 0;
+
+ printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__,
+ value, reg);
+
+ return -EIO;
+}
+
+static const char *rec_src_text[] = { "Line", "Mic" };
+static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+
+static const struct soc_enum rec_src_enum =
+ SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
+
+static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
+SOC_DAPM_ENUM("Input Select", rec_src_enum);
+
+static const struct soc_enum tlv320aic23_rec_src =
+ SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
+static const struct soc_enum tlv320aic23_deemph =
+ SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text);
+
+static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0);
+static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0);
+static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0);
+
+static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ u16 val, reg;
+
+ val = (ucontrol->value.integer.value[0] & 0x07);
+
+ /* linear conversion to userspace
+ * 000 = -6db
+ * 001 = -9db
+ * 010 = -12db
+ * 011 = -18db (Min)
+ * 100 = 0db (Max)
+ */
+ val = (val >= 4) ? 4 : (3 - val);
+
+ reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0);
+ tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6));
+
+ return 0;
+}
+
+static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ u16 val;
+
+ val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0);
+ val = val >> 6;
+ val = (val >= 4) ? 4 : (3 - val);
+ ucontrol->value.integer.value[0] = val;
+ return 0;
+
+}
+
+#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\
+ .put = snd_soc_tlv320aic23_put_volsw, \
+ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL,
+ TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv),
+ SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1),
+ SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL,
+ TLV320AIC23_RINVOL, 7, 1, 0),
+ SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL,
+ TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv),
+ SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1),
+ SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0),
+ SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG,
+ 6, 4, 0, sidetone_vol_tlv),
+ SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
+};
+
+/* add non dapm controls */
+static int tlv320aic23_add_controls(struct snd_soc_codec *codec)
+{
+
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&tlv320aic23_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+
+}
+
+/* PGA Mixer controls for Line and Mic switch */
+static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
+ SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0),
+ SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1),
+ SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1),
+ SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0,
+ &tlv320aic23_rec_src_mux_controls),
+ SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1,
+ &tlv320aic23_output_mixer_controls[0],
+ ARRAY_SIZE(tlv320aic23_output_mixer_controls)),
+ SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0),
+ SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0),
+
+ SND_SOC_DAPM_OUTPUT("LHPOUT"),
+ SND_SOC_DAPM_OUTPUT("RHPOUT"),
+ SND_SOC_DAPM_OUTPUT("LOUT"),
+ SND_SOC_DAPM_OUTPUT("ROUT"),
+
+ SND_SOC_DAPM_INPUT("LLINEIN"),
+ SND_SOC_DAPM_INPUT("RLINEIN"),
+
+ SND_SOC_DAPM_INPUT("MICIN"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ /* Output Mixer */
+ {"Output Mixer", "Line Bypass Switch", "Line Input"},
+ {"Output Mixer", "Playback Switch", "DAC"},
+ {"Output Mixer", "Mic Sidetone Switch", "Mic Input"},
+
+ /* Outputs */
+ {"RHPOUT", NULL, "Output Mixer"},
+ {"LHPOUT", NULL, "Output Mixer"},
+ {"LOUT", NULL, "Output Mixer"},
+ {"ROUT", NULL, "Output Mixer"},
+
+ /* Inputs */
+ {"Line Input", "NULL", "LLINEIN"},
+ {"Line Input", "NULL", "RLINEIN"},
+
+ {"Mic Input", "NULL", "MICIN"},
+
+ /* input mux */
+ {"Capture Source", "Line", "Line Input"},
+ {"Capture Source", "Mic", "Mic Input"},
+ {"ADC", NULL, "Capture Source"},
+
+};
+
+/* 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 */
+};
+
+static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+ ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+ /* set up audio path interconnects */
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 iface_reg, data;
+ u8 count = 0;
+
+ 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;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface_reg |= (0x01 << 2);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface_reg |= (0x02 << 2);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface_reg |= (0x03 << 2);
+ break;
+ }
+ tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+
+ return 0;
+}
+
+static int tlv320aic23_pcm_prepare(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->codec;
+
+ /* set active */
+ tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001);
+
+ return 0;
+}
+
+static void tlv320aic23_shutdown(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->codec;
+
+ /* deactivate */
+ if (!codec->active) {
+ udelay(50);
+ tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+ }
+}
+
+static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 reg;
+
+ reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT);
+ if (mute)
+ reg |= TLV320AIC23_DACM_MUTE;
+
+ else
+ reg &= ~TLV320AIC23_DACM_MUTE;
+
+ tlv320aic23_write(codec, TLV320AIC23_DIGT, reg);
+
+ return 0;
+}
+
+static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface_reg;
+
+ iface_reg =
+ tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03);
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface_reg |= TLV320AIC23_MS_MASTER;
+ 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_reg |= TLV320AIC23_FOR_I2S;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface_reg |= TLV320AIC23_FOR_DSP;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface_reg |= TLV320AIC23_FOR_LJUST;
+ break;
+ default:
+ return -EINVAL;
+
+ }
+
+ tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+
+ return 0;
+}
+
+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;
+}
+
+static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* vref/mid, osc on, dac unmute */
+ 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);
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* everything off, dac mute, inactive */
+ tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+ tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define AIC23_RATES SNDRV_PCM_RATE_8000_96000
+#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai tlv320aic23_dai = {
+ .name = "tlv320aic23",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AIC23_RATES,
+ .formats = AIC23_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AIC23_RATES,
+ .formats = AIC23_FORMATS,},
+ .ops = {
+ .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,
+ }
+};
+EXPORT_SYMBOL_GPL(tlv320aic23_dai);
+
+static int tlv320aic23_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;
+
+ tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int tlv320aic23_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u16 reg;
+
+ /* Sync reg_cache with the hardware */
+ for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) {
+ u16 val = tlv320aic23_read_reg_cache(codec, reg);
+ tlv320aic23_write(codec, reg, val);
+ }
+
+ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ tlv320aic23_set_bias_level(codec, codec->suspend_bias_level);
+
+ return 0;
+}
+
+/*
+ * initialise the AIC23 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int tlv320aic23_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+ u16 reg;
+
+ codec->name = "tlv320aic23";
+ codec->owner = THIS_MODULE;
+ codec->read = tlv320aic23_read_reg_cache;
+ codec->write = tlv320aic23_write;
+ codec->set_bias_level = tlv320aic23_set_bias_level;
+ codec->dai = &tlv320aic23_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg);
+ codec->reg_cache =
+ kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ /* Reset codec */
+ tlv320aic23_write(codec, TLV320AIC23_RESET, 0);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "tlv320aic23: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* power on device */
+ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
+
+ /* Unmute input */
+ reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL);
+ tlv320aic23_write(codec, TLV320AIC23_LINVOL,
+ (reg & (~TLV320AIC23_LIM_MUTED)) |
+ (TLV320AIC23_LRS_ENABLED));
+
+ reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL);
+ tlv320aic23_write(codec, TLV320AIC23_RINVOL,
+ (reg & (~TLV320AIC23_LIM_MUTED)) |
+ TLV320AIC23_LRS_ENABLED);
+
+ reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG);
+ tlv320aic23_write(codec, TLV320AIC23_ANLG,
+ (reg) & (~TLV320AIC23_BYPASS_ON) &
+ (~TLV320AIC23_MICM_MUTED));
+
+ /* Default output volume */
+ tlv320aic23_write(codec, TLV320AIC23_LCHNVOL,
+ TLV320AIC23_DEFAULT_OUT_VOL &
+ TLV320AIC23_OUT_VOL_MASK);
+ tlv320aic23_write(codec, TLV320AIC23_RCHNVOL,
+ TLV320AIC23_DEFAULT_OUT_VOL &
+ TLV320AIC23_OUT_VOL_MASK);
+
+ tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1);
+
+ tlv320aic23_add_controls(codec);
+ tlv320aic23_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "tlv320aic23: 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 *tlv320aic23_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+/*
+ * If the i2c layer weren't so broken, we could pass this kind of data
+ * around
+ */
+static int tlv320aic23_codec_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *i2c_id)
+{
+ struct snd_soc_device *socdev = tlv320aic23_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
+ return -EINVAL;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = tlv320aic23_init(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n");
+ goto err;
+ }
+ return ret;
+
+err:
+ kfree(codec);
+ kfree(i2c);
+ return ret;
+}
+static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c)
+{
+ put_device(&i2c->dev);
+ return 0;
+}
+
+static const struct i2c_device_id tlv320aic23_id[] = {
+ {"tlv320aic23", 0},
+ {}
+};
+
+MODULE_DEVICE_TABLE(i2c, tlv320aic23_id);
+
+static struct i2c_driver tlv320aic23_i2c_driver = {
+ .driver = {
+ .name = "tlv320aic23",
+ },
+ .probe = tlv320aic23_codec_probe,
+ .remove = __exit_p(tlv320aic23_i2c_remove),
+ .id_table = tlv320aic23_id,
+};
+
+#endif
+
+static int tlv320aic23_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ 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)
+ return -ENOMEM;
+
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ tlv320aic23_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data;
+ codec->hw_read = NULL;
+ ret = i2c_add_driver(&tlv320aic23_i2c_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add i2c driver");
+#endif
+ return ret;
+}
+
+static int tlv320aic23_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)
+ tlv320aic23_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(&tlv320aic23_i2c_driver);
+#endif
+ kfree(codec->reg_cache);
+ kfree(codec);
+
+ return 0;
+}
+struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = {
+ .probe = tlv320aic23_probe,
+ .remove = tlv320aic23_remove,
+ .suspend = tlv320aic23_suspend,
+ .resume = tlv320aic23_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h
new file mode 100644
index 00000000000..79d1faf8e57
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23.h
@@ -0,0 +1,122 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver
+ *
+ * Author: Arun KS, <arunks@mistralsolutions.com>
+ * Copyright: (C) 2008 Mistral Solutions Pvt Ltd
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _TLV320AIC23_H
+#define _TLV320AIC23_H
+
+/* Codec TLV320AIC23 */
+#define TLV320AIC23_LINVOL 0x00
+#define TLV320AIC23_RINVOL 0x01
+#define TLV320AIC23_LCHNVOL 0x02
+#define TLV320AIC23_RCHNVOL 0x03
+#define TLV320AIC23_ANLG 0x04
+#define TLV320AIC23_DIGT 0x05
+#define TLV320AIC23_PWR 0x06
+#define TLV320AIC23_DIGT_FMT 0x07
+#define TLV320AIC23_SRATE 0x08
+#define TLV320AIC23_ACTIVE 0x09
+#define TLV320AIC23_RESET 0x0F
+
+/* Left (right) line input volume control register */
+#define TLV320AIC23_LRS_ENABLED 0x0100
+#define TLV320AIC23_LIM_MUTED 0x0080
+#define TLV320AIC23_LIV_DEFAULT 0x0017
+#define TLV320AIC23_LIV_MAX 0x001f
+#define TLV320AIC23_LIV_MIN 0x0000
+
+/* Left (right) channel headphone volume control register */
+#define TLV320AIC23_LZC_ON 0x0080
+#define TLV320AIC23_LHV_DEFAULT 0x0079
+#define TLV320AIC23_LHV_MAX 0x007f
+#define TLV320AIC23_LHV_MIN 0x0000
+
+/* Analog audio path control register */
+#define TLV320AIC23_STA_REG(x) ((x)<<6)
+#define TLV320AIC23_STE_ENABLED 0x0020
+#define TLV320AIC23_DAC_SELECTED 0x0010
+#define TLV320AIC23_BYPASS_ON 0x0008
+#define TLV320AIC23_INSEL_MIC 0x0004
+#define TLV320AIC23_MICM_MUTED 0x0002
+#define TLV320AIC23_MICB_20DB 0x0001
+
+/* Digital audio path control register */
+#define TLV320AIC23_DACM_MUTE 0x0008
+#define TLV320AIC23_DEEMP_32K 0x0002
+#define TLV320AIC23_DEEMP_44K 0x0004
+#define TLV320AIC23_DEEMP_48K 0x0006
+#define TLV320AIC23_ADCHP_ON 0x0001
+
+/* Power control down register */
+#define TLV320AIC23_DEVICE_PWR_OFF 0x0080
+#define TLV320AIC23_CLK_OFF 0x0040
+#define TLV320AIC23_OSC_OFF 0x0020
+#define TLV320AIC23_OUT_OFF 0x0010
+#define TLV320AIC23_DAC_OFF 0x0008
+#define TLV320AIC23_ADC_OFF 0x0004
+#define TLV320AIC23_MIC_OFF 0x0002
+#define TLV320AIC23_LINE_OFF 0x0001
+
+/* Digital audio interface register */
+#define TLV320AIC23_MS_MASTER 0x0040
+#define TLV320AIC23_LRSWAP_ON 0x0020
+#define TLV320AIC23_LRP_ON 0x0010
+#define TLV320AIC23_IWL_16 0x0000
+#define TLV320AIC23_IWL_20 0x0004
+#define TLV320AIC23_IWL_24 0x0008
+#define TLV320AIC23_IWL_32 0x000C
+#define TLV320AIC23_FOR_I2S 0x0002
+#define TLV320AIC23_FOR_DSP 0x0003
+#define TLV320AIC23_FOR_LJUST 0x0001
+
+/* Sample rate control register */
+#define TLV320AIC23_CLKOUT_HALF 0x0080
+#define TLV320AIC23_CLKIN_HALF 0x0040
+#define TLV320AIC23_BOSR_384fs 0x0002 /* BOSR_272fs in USB mode */
+#define TLV320AIC23_USB_CLK_ON 0x0001
+#define TLV320AIC23_SR_MASK 0xf
+#define TLV320AIC23_CLKOUT_SHIFT 7
+#define TLV320AIC23_CLKIN_SHIFT 6
+#define TLV320AIC23_SR_SHIFT 2
+#define TLV320AIC23_BOSR_SHIFT 1
+
+/* Digital interface register */
+#define TLV320AIC23_ACT_ON 0x0001
+
+/*
+ * AUDIO related MACROS
+ */
+
+#define TLV320AIC23_DEFAULT_OUT_VOL 0x70
+#define TLV320AIC23_DEFAULT_IN_VOLUME 0x10
+
+#define TLV320AIC23_OUT_VOL_MIN TLV320AIC23_LHV_MIN
+#define TLV320AIC23_OUT_VOL_MAX TLV320AIC23_LHV_MAX
+#define TLV320AIC23_OUT_VO_RANGE (TLV320AIC23_OUT_VOL_MAX - \
+ TLV320AIC23_OUT_VOL_MIN)
+#define TLV320AIC23_OUT_VOL_MASK TLV320AIC23_OUT_VOL_MAX
+
+#define TLV320AIC23_IN_VOL_MIN TLV320AIC23_LIV_MIN
+#define TLV320AIC23_IN_VOL_MAX TLV320AIC23_LIV_MAX
+#define TLV320AIC23_IN_VOL_RANGE (TLV320AIC23_IN_VOL_MAX - \
+ TLV320AIC23_IN_VOL_MIN)
+#define TLV320AIC23_IN_VOL_MASK TLV320AIC23_IN_VOL_MAX
+
+#define TLV320AIC23_SIDETONE_MASK 0x1c0
+#define TLV320AIC23_SIDETONE_0 0x100
+#define TLV320AIC23_SIDETONE_6 0x000
+#define TLV320AIC23_SIDETONE_9 0x040
+#define TLV320AIC23_SIDETONE_12 0x080
+#define TLV320AIC23_SIDETONE_18 0x0c0
+
+extern struct snd_soc_dai tlv320aic23_dai;
+extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23;
+
+#endif /* _TLV320AIC23_H */
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 566a427c928..05336ed7e49 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -48,7 +48,6 @@
#include "tlv320aic3x.h"
-#define AUDIO_NAME "aic3x"
#define AIC3X_VERSION "0.2"
/* codec private data */
@@ -991,7 +990,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected);
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
struct snd_soc_dai aic3x_dai = {
- .name = "aic3x",
+ .name = "tlv320aic3x",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
@@ -1055,7 +1054,7 @@ static int aic3x_init(struct snd_soc_device *socdev)
struct aic3x_setup_data *setup = socdev->codec_data;
int reg, ret = 0;
- codec->name = "aic3x";
+ codec->name = "tlv320aic3x";
codec->owner = THIS_MODULE;
codec->read = aic3x_read_reg_cache;
codec->write = aic3x_write;
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index d206d7f892b..a69ee72a7af 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -36,7 +36,6 @@
#include "uda1380.h"
#define UDA1380_VERSION "0.6"
-#define AUDIO_NAME "uda1380"
/*
* uda1380 register cache
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 9a37c8d95ed..d8ca2da8d63 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -3,7 +3,7 @@
*
* Copyright 2006 Wolfson Microelectronics PLC.
*
- * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
@@ -18,6 +18,7 @@
#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>
@@ -27,7 +28,6 @@
#include "wm8510.h"
-#define AUDIO_NAME "wm8510"
#define WM8510_VERSION "0.6"
struct snd_soc_codec_device soc_codec_dev_wm8510;
@@ -55,6 +55,9 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = {
0x0001,
};
+#define WM8510_POWER1_BIASEN 0x08
+#define WM8510_POWER1_BUFIOEN 0x10
+
/*
* read wm8510 register cache
*/
@@ -224,9 +227,9 @@ SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0),
-SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0,
- &wm8510_micpga_controls[0],
- ARRAY_SIZE(wm8510_micpga_controls)),
+SND_SOC_DAPM_MIXER("Mic PGA", WM8510_POWER2, 2, 0,
+ &wm8510_micpga_controls[0],
+ ARRAY_SIZE(wm8510_micpga_controls)),
SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0,
&wm8510_boost_controls[0],
ARRAY_SIZE(wm8510_boost_controls)),
@@ -526,23 +529,35 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute)
static int wm8510_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
+ u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3;
switch (level) {
case SND_SOC_BIAS_ON:
- wm8510_write(codec, WM8510_POWER1, 0x1ff);
- wm8510_write(codec, WM8510_POWER2, 0x1ff);
- wm8510_write(codec, WM8510_POWER3, 0x1ff);
- break;
case SND_SOC_BIAS_PREPARE:
+ power1 |= 0x1; /* VMID 50k */
+ wm8510_write(codec, WM8510_POWER1, power1);
+ break;
+
case SND_SOC_BIAS_STANDBY:
+ power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN;
+
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* Initial cap charge at VMID 5k */
+ wm8510_write(codec, WM8510_POWER1, power1 | 0x3);
+ mdelay(100);
+ }
+
+ power1 |= 0x2; /* VMID 500k */
+ wm8510_write(codec, WM8510_POWER1, power1);
break;
+
case SND_SOC_BIAS_OFF:
- /* everything off, dac mute, inactive */
- wm8510_write(codec, WM8510_POWER1, 0x0);
- wm8510_write(codec, WM8510_POWER2, 0x0);
- wm8510_write(codec, WM8510_POWER3, 0x0);
+ wm8510_write(codec, WM8510_POWER1, 0);
+ wm8510_write(codec, WM8510_POWER2, 0);
+ wm8510_write(codec, WM8510_POWER3, 0);
break;
}
+
codec->bias_level = level;
return 0;
}
@@ -640,6 +655,7 @@ static int wm8510_init(struct snd_soc_device *socdev)
}
/* power on device */
+ codec->bias_level = SND_SOC_BIAS_OFF;
wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
wm8510_add_controls(codec);
wm8510_add_widgets(codec);
@@ -747,6 +763,62 @@ err_driver:
}
#endif
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8510_spi_probe(struct spi_device *spi)
+{
+ struct snd_soc_device *socdev = wm8510_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ codec->control_data = spi;
+
+ ret = wm8510_init(socdev);
+ if (ret < 0)
+ dev_err(&spi->dev, "failed to initialise WM8510\n");
+
+ return ret;
+}
+
+static int __devexit wm8510_spi_remove(struct spi_device *spi)
+{
+ return 0;
+}
+
+static struct spi_driver wm8510_spi_driver = {
+ .driver = {
+ .name = "wm8510",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8510_spi_probe,
+ .remove = __devexit_p(wm8510_spi_remove),
+};
+
+static int wm8510_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 wm8510_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -772,8 +844,14 @@ static int wm8510_probe(struct platform_device *pdev)
codec->hw_write = (hw_write_t)i2c_master_send;
ret = wm8510_add_i2c_device(pdev, setup);
}
-#else
- /* Add other interfaces here */
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ if (setup->spi) {
+ codec->hw_write = (hw_write_t)wm8510_spi_write;
+ ret = spi_register_driver(&wm8510_spi_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add spi driver");
+ }
#endif
if (ret != 0)
@@ -796,6 +874,9 @@ static int wm8510_remove(struct platform_device *pdev)
i2c_unregister_device(codec->control_data);
i2c_del_driver(&wm8510_i2c_driver);
#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8510_spi_driver);
+#endif
kfree(codec);
return 0;
diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h
index c5368396045..bdefcf5c69f 100644
--- a/sound/soc/codecs/wm8510.h
+++ b/sound/soc/codecs/wm8510.h
@@ -94,6 +94,7 @@
#define WM8510_MCLKDIV_12 (7 << 5)
struct wm8510_setup_data {
+ int spi;
int i2c_bus;
unsigned short i2c_address;
};
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index df1ffbe305b..627ebfb4209 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -18,7 +18,6 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
@@ -36,7 +35,6 @@
#include "wm8580.h"
-#define AUDIO_NAME "wm8580"
#define WM8580_VERSION "0.1"
struct pll_state {
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 7b64d9a7ff7..7f8a7e36b33 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -29,7 +29,6 @@
#include "wm8731.h"
-#define AUDIO_NAME "wm8731"
#define WM8731_VERSION "0.13"
struct snd_soc_codec_device soc_codec_dev_wm8731;
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 4892e398a59..9b7296ee5b0 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -29,7 +29,6 @@
#include "wm8750.h"
-#define AUDIO_NAME "WM8750"
#define WM8750_VERSION "0.12"
/* codec private data */
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 8c4df44f334..d426eaa2218 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -2,8 +2,7 @@
* wm8753.c -- WM8753 ALSA Soc Audio driver
*
* Copyright 2003 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
@@ -40,6 +39,7 @@
#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>
@@ -51,7 +51,6 @@
#include "wm8753.h"
-#define AUDIO_NAME "wm8753"
#define WM8753_VERSION "0.16"
static int caps_charge = 2000;
@@ -1719,6 +1718,63 @@ err_driver:
}
#endif
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8753_spi_probe(struct spi_device *spi)
+{
+ struct snd_soc_device *socdev = wm8753_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ codec->control_data = spi;
+
+ ret = wm8753_init(socdev);
+ if (ret < 0)
+ dev_err(&spi->dev, "failed to initialise WM8753\n");
+
+ return ret;
+}
+
+static int __devexit wm8753_spi_remove(struct spi_device *spi)
+{
+ return 0;
+}
+
+static struct spi_driver wm8753_spi_driver = {
+ .driver = {
+ .name = "wm8753",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8753_spi_probe,
+ .remove = __devexit_p(wm8753_spi_remove),
+};
+
+static int wm8753_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
+
+
static int wm8753_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -1753,8 +1809,14 @@ static int wm8753_probe(struct platform_device *pdev)
codec->hw_write = (hw_write_t)i2c_master_send;
ret = wm8753_add_i2c_device(pdev, setup);
}
-#else
- /* Add other interfaces here */
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ if (setup->spi) {
+ codec->hw_write = (hw_write_t)wm8753_spi_write;
+ ret = spi_register_driver(&wm8753_spi_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add spi driver");
+ }
#endif
if (ret != 0) {
@@ -1798,6 +1860,9 @@ static int wm8753_remove(struct platform_device *pdev)
i2c_unregister_device(codec->control_data);
i2c_del_driver(&wm8753_i2c_driver);
#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8753_spi_driver);
+#endif
kfree(codec->private_data);
kfree(codec);
diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h
index 7defde069f1..f55704ce931 100644
--- a/sound/soc/codecs/wm8753.h
+++ b/sound/soc/codecs/wm8753.h
@@ -2,8 +2,7 @@
* wm8753.h -- audio driver for WM8753
*
* Copyright 2003 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
@@ -79,6 +78,7 @@
#define WM8753_ADCTL2 0x3f
struct wm8753_setup_data {
+ int spi;
int i2c_bus;
unsigned short i2c_address;
};
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 0b8c6d38b48..3b326c9b558 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -18,7 +18,6 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index a3f54ec4226..ce40d787760 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -653,14 +653,14 @@ static const struct snd_kcontrol_new wm8903_snd_controls[] = {
/* Input PGAs - No TLV since the scale depends on PGA mode */
SOC_SINGLE("Left Input PGA Switch", WM8903_ANALOGUE_LEFT_INPUT_0,
- 7, 1, 0),
+ 7, 1, 1),
SOC_SINGLE("Left Input PGA Volume", WM8903_ANALOGUE_LEFT_INPUT_0,
0, 31, 0),
SOC_SINGLE("Left Input PGA Common Mode Switch", WM8903_ANALOGUE_LEFT_INPUT_1,
6, 1, 0),
SOC_SINGLE("Right Input PGA Switch", WM8903_ANALOGUE_RIGHT_INPUT_0,
- 7, 1, 0),
+ 7, 1, 1),
SOC_SINGLE("Right Input PGA Volume", WM8903_ANALOGUE_RIGHT_INPUT_0,
0, 31, 0),
SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1,
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 974a4cd0f3f..f41a578ddd4 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -29,7 +29,6 @@
#include "wm8971.h"
-#define AUDIO_NAME "wm8971"
#define WM8971_VERSION "0.9"
#define WM8971_REG_COUNT 43
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 63410d7b5ef..572d22b0880 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -30,7 +30,6 @@
#include "wm8990.h"
-#define AUDIO_NAME "wm8990"
#define WM8990_VERSION "0.2"
/* codec private data */
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 2f1c91b1d55..ffb471e420e 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -2,8 +2,7 @@
* wm9712.c -- ALSA Soc WM9712 codec support
*
* Copyright 2006 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 441d0580db1..aba402b3c99 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -2,8 +2,7 @@
* wm9713.c -- ALSA Soc WM9713 codec support
*
* Copyright 2006 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-10 05:35:24 +0000